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Posted to user@openmeetings.apache.org by Horace Miles <Ho...@myit-solutions.com> on 2014/07/18 15:52:54 UTC
VOIP and Sip Integration
It is nice that Openmeetings provided a way to integrate VOIP and Sip
with Asterisk. That being said, I can find no documentation that tells
the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how
to make a phone call out or in.
Did I miss something somewhere?
Miles
RE: VOIP and Sip Integration
Posted by Horace Miles <Ho...@myit-solutions.com>.
I am using the red5.3.0.2
Red5sip 3.0
I now see the sip dialer.
I have a sip transport that keeps popping in and out of the room.
I tried to make an outside call, but when I click call nothing happens.
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com> wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:07 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Do you have "Enable SIP transport in the room" checked for the room you are testing?
On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the reply, I went back and rechecked my setup. I have completed all the steps according to the integration document.
I found the following document on the wiki: https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to handle calls like onBWCheck, add a service provider. (it is my understanding this error is to be expect as it is not being used?)
So where would I start to try and figure out why there is no sip dialer available?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with Asterisk. That being said, I can find no documentation that tells the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how to make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
RE: VOIP and Sip Integration
Posted by Horace Miles <Ho...@myit-solutions.com>.
Maxim,
I hope you rested well. I shut everything down and deleted all the old red5sip logs and then restarted all the services. Here is a new log I hope it has everything you need to help. Thanks ahead of time.
Miles
New Red5sip log:
19 Jul 07:46:24 - [main]:[INFO ] o.r.c.n.r.BaseRTMPClientHandler: rtmp://98.0.0.0:1935/konnectme/0
19 Jul 07:46:24 - [NioProcessor-2]:[INFO ] o.r.s.n.r.RTMPHandshake: Processing server response for encryption
19 Jul 07:46:24 - [NioProcessor-2]:[INFO ] o.r.s.n.r.RTMPHandshake: Type 0 digest comparison success
19 Jul 07:46:24 - [NioProcessor-2]:[INFO ] o.r.s.n.r.RTMPHandshake: server response part 2 validation success, is Flash Player v9 handshake
19 Jul 07:46:25 - [NioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClientHandler: No service provider / method not found; to handle calls like onBWCheck, add a service provider
19 Jul 07:46:25 - [NioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClientHandler: No service provider / method not found; to handle calls like onBWCheck, add a service provider
19 Jul 07:46:25 - [NioProcessor-2]:[INFO ] o.r.s.n.r.RTMPControlClient: connect
19 Jul 07:46:25 - [NioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClientHandler: No service provider / method not found; to handle calls like onBWCheck, add a service provider
19 Jul 07:46:25 - [NioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClientHandler: No service provider / method not found; to handle calls like onBWCheck, add a service provider
19 Jul 07:46:25 - [NioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClientHandler: No service provider / method not found; to handle calls like onBWCheck, add a service provider
19 Jul 07:49:35 - [NioProcessor-2]:[WARN ] o.z.s.t.InviteTransactionServer: id: INVITE
19 Jul 07:49:35 - [NioProcessor-2]:[WARN ] o.z.s.t.InviteTransactionServer: created
19 Jul 07:49:35 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 674001139081@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK062340
19 Jul 07:49:35 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:49:35 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:49:35 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 674001139081@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK720231
19 Jul 07:49:35 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:49:35 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:49:38 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 066605218036@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK369952
19 Jul 07:49:38 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:49:38 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:49:38 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 066605218036@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK818583
19 Jul 07:49:38 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:49:38 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:49:40 - [Thread-8]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:49:41 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 602779158916@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK471634
19 Jul 07:49:41 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:49:41 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:49:41 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 602779158916@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK456885
19 Jul 07:49:41 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:49:41 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:49:43 - [Thread-10]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:49:43 - [Thread-15]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:49:44 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 731347605594@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK101626
19 Jul 07:49:44 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:49:44 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:49:44 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 731347605594@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK332867
19 Jul 07:49:44 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:49:44 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:49:46 - [Thread-17]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:49:46 - [Thread-22]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:49:47 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 489701648102@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK615698
19 Jul 07:49:47 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:49:47 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:49:47 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 489701648102@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK881259
19 Jul 07:49:47 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:49:47 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:49:49 - [Thread-24]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:49:49 - [Thread-29]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:49:50 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 820830485485@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK8212010
19 Jul 07:49:50 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:49:50 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:49:50 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 820830485485@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK2187611
19 Jul 07:49:50 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:49:50 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:49:52 - [Thread-31]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:49:52 - [Thread-36]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:49:53 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 334871348502@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK4501812
19 Jul 07:49:53 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:49:53 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:49:53 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 334871348502@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK8000013
19 Jul 07:49:53 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:49:53 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:49:55 - [Thread-38]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:49:55 - [Thread-43]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:49:56 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 394142085011@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK6665414
19 Jul 07:49:56 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:49:56 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:49:56 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 394142085011@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK8826615
19 Jul 07:49:56 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:49:56 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:49:58 - [Thread-45]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:49:58 - [Thread-50]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:49:59 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 529838978291@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK5086916
19 Jul 07:49:59 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:49:59 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:49:59 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 529838978291@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK1119617
19 Jul 07:49:59 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:49:59 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:01 - [Thread-52]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:01 - [Thread-57]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:02 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 611159612825@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK4633618
19 Jul 07:50:02 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:02 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:02 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 611159612825@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK6687319
19 Jul 07:50:02 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:02 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:04 - [Thread-59]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:04 - [Thread-64]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:05 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 416756980819@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK3818620
19 Jul 07:50:05 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:05 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:05 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 416756980819@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK0144921
19 Jul 07:50:05 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:05 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:07 - [Thread-66]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:07 - [Thread-71]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:08 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 998002207529@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK3584822
19 Jul 07:50:08 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:08 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:08 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 998002207529@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK1073023
19 Jul 07:50:08 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:08 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:10 - [Thread-73]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:10 - [Thread-78]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:11 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 437677059019@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK6474324
19 Jul 07:50:11 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:11 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:11 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 437677059019@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK4974025
19 Jul 07:50:11 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:11 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:13 - [Thread-80]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:13 - [Thread-85]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:14 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 124606691384@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK0560226
19 Jul 07:50:14 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:14 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:14 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 124606691384@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK4186627
19 Jul 07:50:14 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:14 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:16 - [Thread-87]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:16 - [Thread-92]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:17 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 099977203860@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK2021928
19 Jul 07:50:17 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:17 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:17 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 099977203860@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK9311329
19 Jul 07:50:17 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:17 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:19 - [Thread-94]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:19 - [Thread-99]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:20 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 956028910299@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK9674730
19 Jul 07:50:20 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:20 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:20 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 956028910299@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK4276731
19 Jul 07:50:20 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:20 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:22 - [Thread-101]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:22 - [Thread-106]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:23 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 770535056418@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK6288432
19 Jul 07:50:23 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:23 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:23 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 770535056418@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK4526333
19 Jul 07:50:23 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:23 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:25 - [Thread-108]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:25 - [Thread-113]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:26 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 494989214213@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK4239934
19 Jul 07:50:26 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:26 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:26 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 494989214213@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK3926235
19 Jul 07:50:26 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:26 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:28 - [Thread-115]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:28 - [Thread-120]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:29 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 726390791145@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK3189436
19 Jul 07:50:29 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:29 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:29 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 726390791145@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK2173837
19 Jul 07:50:29 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:29 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:31 - [Thread-122]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:31 - [Thread-127]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:32 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 708209983760@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK1077138
19 Jul 07:50:32 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:32 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:32 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 708209983760@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK4830239
19 Jul 07:50:32 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:32 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:34 - [Thread-129]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:34 - [Thread-134]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:35 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 436650025681@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK6817440
19 Jul 07:50:35 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:35 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:35 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 436650025681@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK5706041
19 Jul 07:50:35 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:35 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:37 - [Thread-136]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:37 - [Thread-141]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:38 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 559869653080@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK4845642
19 Jul 07:50:38 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:38 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:38 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 559869653080@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK9976143
19 Jul 07:50:38 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:38 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:40 - [Thread-143]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:40 - [Thread-148]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:41 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 544489598541@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK9633244
19 Jul 07:50:41 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:41 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:41 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 544489598541@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK7402645
19 Jul 07:50:41 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:41 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:43 - [Thread-150]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:43 - [Thread-155]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:44 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 148263956326@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK5937446
19 Jul 07:50:44 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:44 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:44 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 148263956326@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK3602147
19 Jul 07:50:44 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:44 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:46 - [Thread-157]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:46 - [Thread-162]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:47 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 685700184995@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK4492948
19 Jul 07:50:47 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:47 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:47 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 685700184995@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK4188949
19 Jul 07:50:47 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:47 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:49 - [Thread-164]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:49 - [Thread-169]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:50 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 878526398404@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK9597950
19 Jul 07:50:50 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:50 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:50 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 878526398404@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK8342751
19 Jul 07:50:50 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:50 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:52 - [Thread-171]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:52 - [Thread-176]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:53 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 959879120676@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK2109052
19 Jul 07:50:53 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:53 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:53 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 959879120676@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK5299453
19 Jul 07:50:53 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:53 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:55 - [Thread-178]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:55 - [Thread-183]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:56 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 600602858383@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK9230454
19 Jul 07:50:56 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:56 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:56 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 600602858383@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK3840855
19 Jul 07:50:56 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:56 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:58 - [Thread-185]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:58 - [Thread-190]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:50:59 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 103194981414@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK9898556
19 Jul 07:50:59 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:59 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:50:59 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 103194981414@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK6257357
19 Jul 07:50:59 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:50:59 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:01 - [Thread-192]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:01 - [Thread-197]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:02 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 551889042436@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK9897458
19 Jul 07:51:02 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:02 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:02 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 551889042436@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK2336259
19 Jul 07:51:02 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:02 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:04 - [Thread-199]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:04 - [Thread-204]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:05 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 691361700708@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK3680560
19 Jul 07:51:05 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:05 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:05 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 691361700708@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK8600461
19 Jul 07:51:05 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:05 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:07 - [Thread-206]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:07 - [Thread-211]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:08 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 157233887365@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK8398462
19 Jul 07:51:08 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:08 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:08 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 157233887365@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK1689763
19 Jul 07:51:08 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:08 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:10 - [Thread-213]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:10 - [Thread-218]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:11 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 965849942140@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK3402264
19 Jul 07:51:11 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:11 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:11 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 965849942140@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK9737865
19 Jul 07:51:11 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:11 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:13 - [Thread-220]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:13 - [Thread-225]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:14 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 004839630893@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK5741766
19 Jul 07:51:14 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:14 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:14 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 004839630893@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK5661667
19 Jul 07:51:14 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:14 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:16 - [Thread-227]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:16 - [Thread-232]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:17 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 449328756899@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK3981368
19 Jul 07:51:17 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:17 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:17 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 449328756899@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK3719969
19 Jul 07:51:17 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:17 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:19 - [Thread-234]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:19 - [Thread-239]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:20 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 351650682849@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK0087270
19 Jul 07:51:20 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:20 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:20 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 351650682849@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK7268871
19 Jul 07:51:20 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:20 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:22 - [Thread-241]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:22 - [Thread-246]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:23 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 404578393526@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK2325572
19 Jul 07:51:23 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:23 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:23 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 404578393526@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK2156273
19 Jul 07:51:23 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:23 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:25 - [Thread-248]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:25 - [Thread-253]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:26 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 229684248068@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK5204374
19 Jul 07:51:26 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:26 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:26 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 229684248068@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK9929275
19 Jul 07:51:26 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:26 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:28 - [Thread-255]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:28 - [Thread-260]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:29 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 061470285832@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK2480876
19 Jul 07:51:29 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:29 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:29 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 061470285832@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK3792177
19 Jul 07:51:29 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:29 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:31 - [Thread-262]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:31 - [Thread-267]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:32 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 861478786775@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK9016078
19 Jul 07:51:32 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:32 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:32 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 861478786775@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK4384979
19 Jul 07:51:32 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:32 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:34 - [Thread-269]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:34 - [Thread-274]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:35 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 730066088345@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK8016780
19 Jul 07:51:35 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:35 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:35 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 730066088345@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK6415681
19 Jul 07:51:35 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:35 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:37 - [Thread-276]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:37 - [Thread-281]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:38 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 359102505092@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK5390382
19 Jul 07:51:38 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:38 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:38 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 359102505092@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK9262583
19 Jul 07:51:38 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:38 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:40 - [Thread-283]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:40 - [Thread-288]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:41 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 960156182727@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK9204784
19 Jul 07:51:41 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:41 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:41 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 960156182727@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK4454485
19 Jul 07:51:41 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:41 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:43 - [Thread-290]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:43 - [Thread-295]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:44 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 636527215779@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK7131986
19 Jul 07:51:44 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:44 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:44 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 636527215779@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK1730987
19 Jul 07:51:44 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:44 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:46 - [Thread-297]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:46 - [Thread-302]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:47 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 037915448075@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK8396188
19 Jul 07:51:47 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:47 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:47 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 037915448075@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK9414189
19 Jul 07:51:47 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:47 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:49 - [Thread-304]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:49 - [Thread-309]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:50 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 711413722119@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK6755290
19 Jul 07:51:50 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:50 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:50 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 711413722119@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK7089991
19 Jul 07:51:50 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:50 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:52 - [Thread-311]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:52 - [Thread-316]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:53 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 219398941508@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK9063392
19 Jul 07:51:53 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:53 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:53 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 219398941508@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK9265093
19 Jul 07:51:53 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:53 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:55 - [Thread-318]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:55 - [Thread-323]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:56 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 604045435407@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK1151794
19 Jul 07:51:56 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:56 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:56 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 604045435407@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK0396595
19 Jul 07:51:56 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:56 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:58 - [Thread-325]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:58 - [Thread-330]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:51:59 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 365160205582@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK6787996
19 Jul 07:51:59 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:59 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:51:59 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 365160205582@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK2962297
19 Jul 07:51:59 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:51:59 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:01 - [Thread-332]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:01 - [Thread-337]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:02 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 637795488568@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK2034398
19 Jul 07:52:02 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:02 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:02 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 637795488568@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK0666199
19 Jul 07:52:02 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:02 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:04 - [Thread-339]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:04 - [Thread-344]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:05 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 865250039566@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK83615100
19 Jul 07:52:05 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:05 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:05 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 865250039566@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK33555101
19 Jul 07:52:05 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:05 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:07 - [Thread-346]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:07 - [Thread-351]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:08 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 632060431065@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK08334102
19 Jul 07:52:08 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:08 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:08 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 632060431065@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK38425103
19 Jul 07:52:08 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:08 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:10 - [Thread-353]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:10 - [Thread-358]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:11 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 045461340736@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK45061104
19 Jul 07:52:11 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:11 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:11 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 045461340736@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK07150105
19 Jul 07:52:11 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:11 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:13 - [Thread-360]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:13 - [Thread-365]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:14 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 872869776712@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK14037106
19 Jul 07:52:14 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:14 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:14 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 872869776712@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK91818107
19 Jul 07:52:14 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:14 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:16 - [Thread-367]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:16 - [Thread-372]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:17 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 183217279319@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK44077108
19 Jul 07:52:17 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:17 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:17 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 183217279319@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK57590109
19 Jul 07:52:17 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:17 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:19 - [Thread-374]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:19 - [Thread-379]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:20 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 805355501656@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK02092110
19 Jul 07:52:20 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:20 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:20 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 805355501656@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK88920111
19 Jul 07:52:20 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:20 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:22 - [Thread-381]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:22 - [Thread-386]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:23 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 270941805560@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK88626112
19 Jul 07:52:23 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:23 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:23 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 270941805560@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK17705113
19 Jul 07:52:23 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:23 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:25 - [Thread-388]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:25 - [Thread-393]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:26 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 415479783101@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK22824114
19 Jul 07:52:26 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:26 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:26 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 415479783101@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK45805115
19 Jul 07:52:26 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:26 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:28 - [Thread-395]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:28 - [Thread-400]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:29 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 368293471126@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK38238116
19 Jul 07:52:29 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:29 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:29 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 368293471126@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK41153117
19 Jul 07:52:29 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:29 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:31 - [Thread-402]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:31 - [Thread-407]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:32 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 900905159665@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK18180118
19 Jul 07:52:32 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:32 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:32 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 900905159665@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK79737119
19 Jul 07:52:32 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:32 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:34 - [Thread-409]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:34 - [Thread-414]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:35 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 256709592816@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK06964120
19 Jul 07:52:35 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:35 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:35 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 256709592816@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK24723121
19 Jul 07:52:35 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:35 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:37 - [Thread-416]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:37 - [Thread-421]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:38 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 481300866295@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK35278122
19 Jul 07:52:38 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:38 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:38 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 481300866295@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK89866123
19 Jul 07:52:38 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:38 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:40 - [Thread-423]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:40 - [Thread-428]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:41 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 466317395810@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK94796124
19 Jul 07:52:41 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:41 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:41 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 466317395810@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK12871125
19 Jul 07:52:41 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:41 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:43 - [Thread-430]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:43 - [Thread-435]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:44 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 009515276984@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK68524126
19 Jul 07:52:44 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:44 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:44 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 009515276984@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK35112127
19 Jul 07:52:44 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:44 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:46 - [Thread-437]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:46 - [Thread-442]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:47 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 992967724070@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK16672128
19 Jul 07:52:47 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:47 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:47 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 992967724070@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK03585129
19 Jul 07:52:47 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:47 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:49 - [Thread-444]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:49 - [Thread-449]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:50 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 726781566483@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK53701130
19 Jul 07:52:50 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:50 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:50 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 726781566483@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK45616131
19 Jul 07:52:50 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:50 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:52 - [Thread-451]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:52 - [Thread-456]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:53 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 094570430516@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK70466132
19 Jul 07:52:53 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:53 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:53 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 094570430516@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK59135133
19 Jul 07:52:53 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:53 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:55 - [Thread-458]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:55 - [Thread-463]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:56 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 824539444389@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK96765134
19 Jul 07:52:56 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:56 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:56 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 824539444389@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK21776135
19 Jul 07:52:56 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:56 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:58 - [Thread-465]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:58 - [Thread-470]:[WARN ] o.z.s.t.TransactionClient: Clearing timeout expired
19 Jul 07:52:59 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: id: 536171602859@127.0.1.1-1-REGISTER-127.0.1.1:5070-z9hG4bK81933136
19 Jul 07:52:59 - [org.red5.sip.app.SIPRegisterAgent]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:59 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
19 Jul 07:52:59 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: id: 536171602859@127.0.1.1-2-REGISTER-127.0.1.1:5070-z9hG4bK22223137
19 Jul 07:52:59 - [Thread-1]:[WARN ] o.z.s.t.TransactionClient: created
19 Jul 07:52:59 - [Thread-1]:[WARN ] o.z.s.p.SipProvider: received new SIP message
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 11:18 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
will try to take a look a look at it tomorrow, too late here ...
On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok I will restart red5sip service and red5 and then send a new log
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 10:06 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
got the full trace in other email, will try to check code
On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com> wrote:
Did miss understand what you were asking for?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:56 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
would be more helpful to get full stack instead of "Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com> wrote:
Openmeetings log says confBridgeList authentication is failing. I will check to make sure I didn’t change a password there..
Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:43 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
this "I have a sip transport that keeps popping in and out of the room." usually mean something configured wrong.
Any exceptions in the logs (openmeetings.log and red5sip.log
On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com> wrote:
Sorry also Asterisk 11
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com> wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:07 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Do you have "Enable SIP transport in the room" checked for the room you are testing?
On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the reply, I went back and rechecked my setup. I have completed all the steps according to the integration document.
I found the following document on the wiki: https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to handle calls like onBWCheck, add a service provider. (it is my understanding this error is to be expect as it is not being used?)
So where would I start to try and figure out why there is no sip dialer available?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with Asterisk. That being said, I can find no documentation that tells the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how to make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
RE: VOIP and Sip Integration
Posted by Horace Miles <Ho...@myit-solutions.com>.
Maxim that will work.. I don’t have a drop box account. But let me check my cloud server. Will get back to you by tomorrow 07/24/14 evening after I come from work.
I appreciate it Miles.
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Wednesday, July 23, 2014 7:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
I can privately send you all mine asterisk config files, so you can compare
additionally I can send both red5sip and OM, but I need some place like dropbox for this
On 23 July 2014 20:51, Horace Miles <Ho...@myit-solutions.com> wrote:
OK I will down load this evening and see what happens..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, July 22, 2014 11:28 PM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Hello Horace,
just have checked, 3.0.3 seems to work as expected (at least 'SIP Transport' sitting in the room)
There are some NPEs in logs (will take a looks at it as soon as will have some time)
On 23 July 2014 12:13, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok thanks Maxim
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, July 22, 2014 7:17 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
I'll try to find server with configured Asterisk and try to double-check
On 22 July 2014 20:37, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks I made the change prior to sending the email. There appears to be something else missing:
There appears to be a entry missing in the /etc/asterisk/func_odbc.conf: file for ${EXTEN}
I am probably wrong. But I can’t figure out how this is making the call to the database.
I don’t find any SQL statement in the /etc/asterisk/func_odbc.conf file and I am not sure how to construct one there that would work.
Would I simply add
[EXTEN]
dsn=asterisk
readsql=SELECT confno from room where confno = @EXTEN – NOT SURE HOW TO GET THE ROOMID INTO THIS VARIABLE
Thanks ahead of time
Miles
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, July 22, 2014 6:23 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
yes, this line need to be corrected
openmeetings/rooms -> openmeetings/room
guess this is the problem
On 22 July 2014 19:43, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Maxim,
I have been trying to figure this out, I am knew to it all but on a steep learning curve.
I do have a question about the asterisk extensions.conf
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
Does the above line check the openmeetings database rooms table for the confno and returns ok if it finds it and notavail if it doesn’t?
I am getting the following warning:
Chan_sip.c.:25184 handle_request_infite: Call from ‘red5sip_user’ (98.0.0.0:5070) to extension ‘40016’ rejected because extension not found in context “rooms-red5sip” I don’t recall seeing this error before. But if the “exten” line is checking the database openmeetings and looking for rooms table it does not exist. There is a table name room but no rooms.
Am I reading this correctly?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, July 22, 2014 4:27 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
AFAIK dial in/out conference room was working as expected
Not sure if we still have infrastructure test current version
Will try to ask someone
On 21 July 2014 20:17, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok on the sip transport, I will try to figure out why it keep popping in and out.
However, I am not understanding concerning the Asterisk config. The asterisk config I am using is from the install and modification as stated from the VOIP/SIP 3.0 integration package. The instructions don’t say a sip trunk or outside provider is required. However, I am unable to succesfull make calls from a conference room to a phone and visa versa. Which I thought I was suppose to be able to do after the integration. Per the below instructions I guess I am asking what am I missing from below?
Feature Matrix:
Feature
Description
1) Dial-In
A phone number is provided which you can give to anybody to "Dial-In" via usual landlane/phone into the conference room of OpenMeetings - Every room has its own phone number. Currently room gets number
like 400<Id of room>. Maybe should move phone prefix to settings,
currently it hardcoded.
2) Dial-Out
The users in the conference room can call anybody outside of the conference room by entering the phone number in the conference room - In room actions menu exist "SIP dialer". When user clicked dialer window appears.
Currently calls can't be dropped from Openmeetings, tbd
3) Multiple Dial-In
You can give away multiple numbers and do the same as described in case (1). Multiple Dial-In is achieved by configuring the SIP-server (Asterisk). It is possible to create multiple extensions (phone numbers) in Asterisk configuration that will be redirects to single conference room.
4) Multiple Dial-Out
You can dial multiple numbers from within the conference room - From within conference can be dialed multiple numbers.
Main difference to native Red5-Phone project
rom: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, July 21, 2014 5:36 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
If I do remember correctly
SIP transport should enter the room and be in room as long as there other users in it.
Possibility to call to phone numbers depends on your Asterisk config.
I'll try to fix documentation ASAP
On 21 July 2014 18:55, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Maxim,
Let me make sure I understand about the sip transport. It should not be popping in and out of the room? On my box I keep getting Sip Transport has exited the room.
When properly configured, should I be able to call land and cell phones without a need for another server or;
1. Do I need to subscribe to a VOIP service provider
2. Configure Asterisk as a sip trunk to use google voice or some other solution?
Also if you could have someone correct this line in the instructions of extensions.conf it will help to eliminate at least one error
[rooms-red5sip]
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil) <<<<<<<<<<<< should be “notavail”
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
exten => _400X!,n(notavail),Hangup
Thanks ahead of time
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, July 21, 2014 4:00 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Hello Horace,
jsvc can be used to start java application as service
I don't really like it (it was unstable when I used it)
I prefer to write init.d script
I see no errors in your log
If everything is OK SIP transport should be in the room
All 3 logs should be checked to have no errors
I usually run asterisk in debug mode while setting everything up
On 20 July 2014 23:55, Horace Miles <Ho...@myit-solutions.com> wrote:
Additonally the VOIP and SIP integration 3.0 instructions do not mention installing jsvc. Is it still a requirement to install jsvc under 3.0 as it was under 2.0?:
apt-get install jsvc
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 11:18 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
will try to take a look a look at it tomorrow, too late here ...
On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok I will restart red5sip service and red5 and then send a new log
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 10:06 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
got the full trace in other email, will try to check code
On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com> wrote:
Did miss understand what you were asking for?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:56 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
would be more helpful to get full stack instead of "Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com> wrote:
Openmeetings log says confBridgeList authentication is failing. I will check to make sure I didn’t change a password there..
Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:43 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
this "I have a sip transport that keeps popping in and out of the room." usually mean something configured wrong.
Any exceptions in the logs (openmeetings.log and red5sip.log
On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com> wrote:
Sorry also Asterisk 11
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com> wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:07 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Do you have "Enable SIP transport in the room" checked for the room you are testing?
On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the reply, I went back and rechecked my setup. I have completed all the steps according to the integration document.
I found the following document on the wiki: https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to handle calls like onBWCheck, add a service provider. (it is my understanding this error is to be expect as it is not being used?)
So where would I start to try and figure out why there is no sip dialer available?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with Asterisk. That being said, I can find no documentation that tells the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how to make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: VOIP and Sip Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
"When I configure config.xml as you have yours, I am not able to connect
to the server. I get the connection time out error."
this usually mean you have port 1935 closed closed on specific interface
i.e. "telnet localhost 1935" will fail
try to switch off your firewall and try again
On 30 July 2014 20:33, Horace Miles <Ho...@myit-solutions.com> wrote:
> Maxim,
>
>
>
> When I configure config.xml as you have yours, I am not able to connect
> to the server. I get the connection time out error.
>
> So my config.xml has my public IP address in it for rtmp and http settings.
>
>
>
> When I configure red5sip/ openmeetings properties settings
>
> Red5.host = ip address of rtmp and http settings in config.xml
>
> Sip.obproxy = ip address of rtmp and http settings in config.xml
>
> Sip.proxy= ip address of rtmp and http settings in config.xml
>
>
>
> Any other setting of 0.0.0.0 or 127.0.0.1 Red5sip fails to get the session
>
>
>
> The in Asterisk the openmeetings manager logs on and then right back up
> which seems to coincide with the sip transport popping in and out of the
> room.
>
> It appears that invite request done by the sip.transport is being
> refused. Which is bound to 127.0.0.1. (I am so confused as to what is
> going on.)
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Sunday, July 27, 2014 12:07 AM
> *To:* Horace Miles; Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Additionally red5sip connects to red5 server directly, not to the swf
> client, so contents of config.xml is ignored while connecting by red5sip
>
>
>
> On 27 July 2014 12:37, Maxim Solodovnik <so...@gmail.com> wrote:
>
> Hello Horas,
>
>
>
> Please write to user mailing list.
>
>
>
> I currently have no host configured in
> /webapps/openmeetings/public/config.xml file, all hosts are allowed
>
> Why do you need to limit host in this file?
>
>
>
> On 27 July 2014 04:44, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Hi Maxim,
>
>
>
> Can I have your thoughts on the following:
>
>
>
> I am unable to get the sip agent to bind to 127.0.0.1. It refuses to bind
> unless I have bind it to the same address that is in red5home
> /webapps/openmeetings/public/config.xml
>
>
>
> The problem appears to be either that the SIP protocol wants to use
> 127.0.0.1 for the subscribe or invites and SIP agent is bound to the Public
> IP address. Therefore it is generating the error for seqno 2 which would
> be the SIP Invite (I am assuming). I have not been able to get the SIP
> tansport to bind to 127.0.0.1 which would probably solve this problem.
>
>
>
> Your thoughts/
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 25, 2014 7:22 AM
> *To:* Horace Miles
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> hope you will be able to fix it, please let ne know if additional help is
> required
>
>
>
> On 25 July 2014 20:53, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Hey thanks for the files.
>
>
>
> I compared and I have found the following:
>
>
>
> It appears the integration is setup for for a box that is NAT’ed. I
> thought openmeetings had to be on a static public IP address?
>
>
>
> So I changed every place that is referencing 127.0.0.1 to my IP address.
>
>
>
> The Sip Agent/Openmeetings Manager does not come into the room until I
> restart Asterisk. I can see it successfully logging on and then
> immediately logging off. The room is successfully spawned.
>
>
>
> There seem to be a problem with the manager once it signs on with the sip
> handshake (again I am guessing)
>
>
>
> chan_sip.c:4164 retrans_pkt: Retransmission timeout reached on
> transmission #########@127.0.0.1 for seqno 2 (Critical Response) see……
> Packet timed out afer 32000ms with no response.
>
>
>
> I will load wireshark later today on the PBX to see what else I might find.
>
>
>
> Thanks for all your help.
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Thursday, July 24, 2014 8:44 AM
> *To:* Horace Miles
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> uploaded
>
>
>
> On 24 July 2014 20:40, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim,
>
>
>
> Thanks I appreciate it very much.
>
>
>
> I have created you an account on my cloud server :
> http://mycloud.myit-solutions.com
>
> Login: mmaxim
>
> Password: chief123
>
>
>
> There is a shared folder labeled openmeetings. You can upload the files
> there. You have 5 GB of space.
>
>
>
> Let me know if you have any problems with this. I won’t be available
> again until tonight but I will look at that time.
>
>
>
> Thanks a million
>
>
>
> Miles
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Wednesday, July 23, 2014 7:10 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> I can privately send you all mine asterisk config files, so you can compare
>
> additionally I can send both red5sip and OM, but I need some place like
> dropbox for this
>
>
>
>
>
> On 23 July 2014 20:51, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> OK I will down load this evening and see what happens..
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Tuesday, July 22, 2014 11:28 PM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Hello Horace,
>
>
>
> just have checked, 3.0.3 seems to work as expected (at least 'SIP
> Transport' sitting in the room)
>
> There are some NPEs in logs (will take a looks at it as soon as will have
> some time)
>
>
>
> On 23 July 2014 12:13, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok thanks Maxim
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Tuesday, July 22, 2014 7:17 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> I'll try to find server with configured Asterisk and try to double-check
>
>
>
> On 22 July 2014 20:37, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks I made the change prior to sending the email. There appears to be
> something else missing:
>
> There appears to be a entry missing in the /etc/asterisk/func_odbc.conf:
> file for ${EXTEN}
>
> I am probably wrong. But I can’t figure out how this is making the call
> to the database.
>
> I don’t find any SQL statement in the /etc/asterisk/func_odbc.conf file
> and I am not sure how to construct one there that would work.
>
> Would I simply add
>
> [EXTEN]
> dsn=asterisk
> readsql=SELECT confno from room where confno = @EXTEN – NOT SURE HOW TO
> GET THE ROOMID INTO THIS VARIABLE
>
>
>
> Thanks ahead of time
>
>
>
> Miles
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Tuesday, July 22, 2014 6:23 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> yes, this line need to be corrected
>
> openmeetings/rooms -> openmeetings/room
>
>
>
> guess this is the problem
>
>
>
> On 22 July 2014 19:43, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks Maxim,
>
> I have been trying to figure this out, I am knew to it all but on a steep
> learning curve.
>
>
>
> I do have a question about the asterisk extensions.conf
>
> exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
>
>
>
> Does the above line check the openmeetings database rooms table for the
> confno and returns ok if it finds it and notavail if it doesn’t?
>
>
>
> I am getting the following warning:
>
> Chan_sip.c.:25184 handle_request_infite: Call from ‘red5sip_user’ (
> 98.0.0.0:5070) to extension ‘40016’ rejected because extension not found
> in context “rooms-red5sip” I don’t recall seeing this error before. But
> if the “exten” line is checking the database openmeetings and looking for
> rooms table it does not exist. There is a table name room but no rooms.
>
>
>
> Am I reading this correctly?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Tuesday, July 22, 2014 4:27 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> AFAIK dial in/out conference room was working as expected
>
> Not sure if we still have infrastructure test current version
>
>
>
> Will try to ask someone
>
>
>
> On 21 July 2014 20:17, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok on the sip transport, I will try to figure out why it keep popping in
> and out.
>
> However, I am not understanding concerning the Asterisk config. The
> asterisk config I am using is from the install and modification as stated
> from the VOIP/SIP 3.0 integration package. The instructions don’t say a
> sip trunk or outside provider is required. However, I am unable to
> succesfull make calls from a conference room to a phone and visa versa.
> Which I thought I was suppose to be able to do after the integration. Per
> the below instructions I guess I am asking what am I missing from below?
>
> *Feature Matrix:*
>
> *Feature*
>
> *Description*
>
> 1) Dial-In
>
> A phone number is provided which you can give to anybody to "Dial-In" via
> usual landlane/phone into the conference room of OpenMeetings - Every room
> has its own phone number. Currently room gets number
> like 400<Id of room>. Maybe should move phone prefix to settings,
> currently it hardcoded.
>
> 2) Dial-Out
>
> The users in the conference room can call anybody outside of the
> conference room by entering the phone number in the conference room - In
> room actions menu exist "SIP dialer". When user clicked dialer window
> appears.
> Currently calls can't be dropped from Openmeetings, tbd
>
> 3) Multiple Dial-In
>
> You can give away multiple numbers and do the same as described in case
> (1). Multiple Dial-In is achieved by configuring the SIP-server (Asterisk).
> It is possible to create multiple extensions (phone numbers) in Asterisk
> configuration that will be redirects to single conference room.
>
> 4) Multiple Dial-Out
>
> You can dial multiple numbers from within the conference room - From
> within conference can be dialed multiple numbers.
>
> *Main difference to native Red5-Phone project*
>
>
>
>
>
>
>
> *rom:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Monday, July 21, 2014 5:36 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> If I do remember correctly
>
> SIP transport should enter the room and be in room as long as there other
> users in it.
>
>
>
> Possibility to call to phone numbers depends on your Asterisk config.
>
>
>
> I'll try to fix documentation ASAP
>
>
>
> On 21 July 2014 18:55, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks Maxim,
>
> Let me make sure I understand about the sip transport. It should not be
> popping in and out of the room? On my box I keep getting Sip Transport has
> exited the room.
>
>
>
> When properly configured, should I be able to call land and cell phones
> without a need for another server or;
>
> 1. Do I need to subscribe to a VOIP service provider
>
> 2. Configure Asterisk as a sip trunk to use google voice or some
> other solution?
>
> Also if you could have someone correct this line in the instructions of
> extensions.conf it will help to eliminate at least one error
>
>
>
>
> *[rooms-red5sip]exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil)
> <<<<<<<<<<<< should be “notavail”exten =>
> _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)exten =>
> _400X!,n(notavail),Hangup *
>
>
>
> *Thanks ahead of time*
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Monday, July 21, 2014 4:00 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Hello Horace,
>
>
>
> jsvc can be used to start java application as service
>
> I don't really like it (it was unstable when I used it)
>
> I prefer to write init.d script
>
>
>
> I see no errors in your log
>
> If everything is OK SIP transport should be in the room
>
> All 3 logs should be checked to have no errors
>
> I usually run asterisk in debug mode while setting everything up
>
>
>
>
>
>
>
> On 20 July 2014 23:55, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Additonally the VOIP and SIP integration 3.0 instructions do not mention
> installing jsvc. Is it still a requirement to install jsvc under 3.0 as it
> was under 2.0?:
>
> *apt-get install jsvc*
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 11:18 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> will try to take a look a look at it tomorrow, too late here ...
>
>
>
> On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok I will restart red5sip service and red5 and then send a new log
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 10:06 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> got the full trace in other email, will try to check code
>
>
>
> On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Did miss understand what you were asking for?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:56 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> would be more helpful to get full stack instead of "Red5sip log : Error
> o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
>
>
>
> On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Openmeetings log says confBridgeList authentication is failing. I will
> check to make sure I didn’t change a password there..
>
> Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException:
> Null
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:43 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> this "I have a sip transport that keeps popping in and out of the room."
> usually mean something configured wrong.
>
> Any exceptions in the logs (openmeetings.log and red5sip.log
>
>
>
> On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Sorry also Asterisk 11
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:10 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Additionally, what version are you using?
>
>
>
> On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Probably not, since I just went into a public room.. let me create a room..
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Do you have "*Enable SIP transport in the room*" checked for the room you
> are testing?
>
>
>
> On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the reply, I went back and rechecked my setup. I have
> completed all the steps according to the integration document.
>
> I found the following document on the wiki:
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
>
>
>
> According to this document I should get a sip dialer under the rooms
> actions menu. But I have no dialer there.
>
>
>
> The only error I see in the red5sip window is
>
> 18 Jul 07:50:11 . [nioProcessor-2]:[INFO ]
> o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to
> handle calls like onBWCheck, add a service provider. (it is my
> understanding this error is to be expect as it is not being used?)
>
>
>
> So where would I start to try and figure out why there is no sip dialer
> available?
>
>
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 7:15 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> http://openmeetings.apache.org/voip-sip-integration.html
>
>
>
> On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> It is nice that Openmeetings provided a way to integrate VOIP and Sip with
> Asterisk. That being said, I can find no documentation that tells the
> following:
>
> If the integration was successful?
>
> What icons should show up where etc.
>
> What actions can be taken by an admin or a user for that matter i.e. how
> to make a phone call out or in.
>
> Did I miss something somewhere?
>
>
>
> Miles
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: VOIP and Sip Integration
Posted by Horace Miles <Ho...@myit-solutions.com>.
Maxim,
When I configure config.xml as you have yours, I am not able to connect to the server. I get the connection time out error.
So my config.xml has my public IP address in it for rtmp and http settings.
When I configure red5sip/ openmeetings properties settings
Red5.host = ip address of rtmp and http settings in config.xml
Sip.obproxy = ip address of rtmp and http settings in config.xml
Sip.proxy= ip address of rtmp and http settings in config.xml
Any other setting of 0.0.0.0 or 127.0.0.1 Red5sip fails to get the session
The in Asterisk the openmeetings manager logs on and then right back up which seems to coincide with the sip transport popping in and out of the room.
It appears that invite request done by the sip.transport is being refused. Which is bound to 127.0.0.1. (I am so confused as to what is going on.)
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Sunday, July 27, 2014 12:07 AM
To: Horace Miles; Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally red5sip connects to red5 server directly, not to the swf client, so contents of config.xml is ignored while connecting by red5sip
On 27 July 2014 12:37, Maxim Solodovnik <so...@gmail.com> wrote:
Hello Horas,
Please write to user mailing list.
I currently have no host configured in /webapps/openmeetings/public/config.xml file, all hosts are allowed
Why do you need to limit host in this file?
On 27 July 2014 04:44, Horace Miles <Ho...@myit-solutions.com> wrote:
Hi Maxim,
Can I have your thoughts on the following:
I am unable to get the sip agent to bind to 127.0.0.1. It refuses to bind unless I have bind it to the same address that is in red5home /webapps/openmeetings/public/config.xml
The problem appears to be either that the SIP protocol wants to use 127.0.0.1 for the subscribe or invites and SIP agent is bound to the Public IP address. Therefore it is generating the error for seqno 2 which would be the SIP Invite (I am assuming). I have not been able to get the SIP tansport to bind to 127.0.0.1 which would probably solve this problem.
Your thoughts/
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 25, 2014 7:22 AM
To: Horace Miles
Subject: Re: VOIP and Sip Integration
hope you will be able to fix it, please let ne know if additional help is required
On 25 July 2014 20:53, Horace Miles <Ho...@myit-solutions.com> wrote:
Hey thanks for the files.
I compared and I have found the following:
It appears the integration is setup for for a box that is NAT’ed. I thought openmeetings had to be on a static public IP address?
So I changed every place that is referencing 127.0.0.1 to my IP address.
The Sip Agent/Openmeetings Manager does not come into the room until I restart Asterisk. I can see it successfully logging on and then immediately logging off. The room is successfully spawned.
There seem to be a problem with the manager once it signs on with the sip handshake (again I am guessing)
chan_sip.c:4164 retrans_pkt: Retransmission timeout reached on transmission #########@127.0.0.1 <ma...@127.0.0.1> for seqno 2 (Critical Response) see…… Packet timed out afer 32000ms with no response.
I will load wireshark later today on the PBX to see what else I might find.
Thanks for all your help.
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Thursday, July 24, 2014 8:44 AM
To: Horace Miles
Subject: Re: VOIP and Sip Integration
uploaded
On 24 July 2014 20:40, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim,
Thanks I appreciate it very much.
I have created you an account on my cloud server : http://mycloud.myit-solutions.com
Login: mmaxim
Password: chief123
There is a shared folder labeled openmeetings. You can upload the files there. You have 5 GB of space.
Let me know if you have any problems with this. I won’t be available again until tonight but I will look at that time.
Thanks a million
Miles
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Wednesday, July 23, 2014 7:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
I can privately send you all mine asterisk config files, so you can compare
additionally I can send both red5sip and OM, but I need some place like dropbox for this
On 23 July 2014 20:51, Horace Miles <Ho...@myit-solutions.com> wrote:
OK I will down load this evening and see what happens..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, July 22, 2014 11:28 PM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Hello Horace,
just have checked, 3.0.3 seems to work as expected (at least 'SIP Transport' sitting in the room)
There are some NPEs in logs (will take a looks at it as soon as will have some time)
On 23 July 2014 12:13, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok thanks Maxim
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, July 22, 2014 7:17 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
I'll try to find server with configured Asterisk and try to double-check
On 22 July 2014 20:37, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks I made the change prior to sending the email. There appears to be something else missing:
There appears to be a entry missing in the /etc/asterisk/func_odbc.conf: file for ${EXTEN}
I am probably wrong. But I can’t figure out how this is making the call to the database.
I don’t find any SQL statement in the /etc/asterisk/func_odbc.conf file and I am not sure how to construct one there that would work.
Would I simply add
[EXTEN]
dsn=asterisk
readsql=SELECT confno from room where confno = @EXTEN – NOT SURE HOW TO GET THE ROOMID INTO THIS VARIABLE
Thanks ahead of time
Miles
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, July 22, 2014 6:23 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
yes, this line need to be corrected
openmeetings/rooms -> openmeetings/room
guess this is the problem
On 22 July 2014 19:43, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Maxim,
I have been trying to figure this out, I am knew to it all but on a steep learning curve.
I do have a question about the asterisk extensions.conf
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
Does the above line check the openmeetings database rooms table for the confno and returns ok if it finds it and notavail if it doesn’t?
I am getting the following warning:
Chan_sip.c.:25184 handle_request_infite: Call from ‘red5sip_user’ (98.0.0.0:5070) to extension ‘40016’ rejected because extension not found in context “rooms-red5sip” I don’t recall seeing this error before. But if the “exten” line is checking the database openmeetings and looking for rooms table it does not exist. There is a table name room but no rooms.
Am I reading this correctly?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, July 22, 2014 4:27 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
AFAIK dial in/out conference room was working as expected
Not sure if we still have infrastructure test current version
Will try to ask someone
On 21 July 2014 20:17, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok on the sip transport, I will try to figure out why it keep popping in and out.
However, I am not understanding concerning the Asterisk config. The asterisk config I am using is from the install and modification as stated from the VOIP/SIP 3.0 integration package. The instructions don’t say a sip trunk or outside provider is required. However, I am unable to succesfull make calls from a conference room to a phone and visa versa. Which I thought I was suppose to be able to do after the integration. Per the below instructions I guess I am asking what am I missing from below?
Feature Matrix:
Feature
Description
1) Dial-In
A phone number is provided which you can give to anybody to "Dial-In" via usual landlane/phone into the conference room of OpenMeetings - Every room has its own phone number. Currently room gets number
like 400<Id of room>. Maybe should move phone prefix to settings,
currently it hardcoded.
2) Dial-Out
The users in the conference room can call anybody outside of the conference room by entering the phone number in the conference room - In room actions menu exist "SIP dialer". When user clicked dialer window appears.
Currently calls can't be dropped from Openmeetings, tbd
3) Multiple Dial-In
You can give away multiple numbers and do the same as described in case (1). Multiple Dial-In is achieved by configuring the SIP-server (Asterisk). It is possible to create multiple extensions (phone numbers) in Asterisk configuration that will be redirects to single conference room.
4) Multiple Dial-Out
You can dial multiple numbers from within the conference room - From within conference can be dialed multiple numbers.
Main difference to native Red5-Phone project
rom: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, July 21, 2014 5:36 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
If I do remember correctly
SIP transport should enter the room and be in room as long as there other users in it.
Possibility to call to phone numbers depends on your Asterisk config.
I'll try to fix documentation ASAP
On 21 July 2014 18:55, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Maxim,
Let me make sure I understand about the sip transport. It should not be popping in and out of the room? On my box I keep getting Sip Transport has exited the room.
When properly configured, should I be able to call land and cell phones without a need for another server or;
1. Do I need to subscribe to a VOIP service provider
2. Configure Asterisk as a sip trunk to use google voice or some other solution?
Also if you could have someone correct this line in the instructions of extensions.conf it will help to eliminate at least one error
[rooms-red5sip]
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil) <<<<<<<<<<<< should be “notavail”
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
exten => _400X!,n(notavail),Hangup
Thanks ahead of time
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, July 21, 2014 4:00 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Hello Horace,
jsvc can be used to start java application as service
I don't really like it (it was unstable when I used it)
I prefer to write init.d script
I see no errors in your log
If everything is OK SIP transport should be in the room
All 3 logs should be checked to have no errors
I usually run asterisk in debug mode while setting everything up
On 20 July 2014 23:55, Horace Miles <Ho...@myit-solutions.com> wrote:
Additonally the VOIP and SIP integration 3.0 instructions do not mention installing jsvc. Is it still a requirement to install jsvc under 3.0 as it was under 2.0?:
apt-get install jsvc
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 11:18 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
will try to take a look a look at it tomorrow, too late here ...
On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok I will restart red5sip service and red5 and then send a new log
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 10:06 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
got the full trace in other email, will try to check code
On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com> wrote:
Did miss understand what you were asking for?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:56 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
would be more helpful to get full stack instead of "Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com> wrote:
Openmeetings log says confBridgeList authentication is failing. I will check to make sure I didn’t change a password there..
Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:43 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
this "I have a sip transport that keeps popping in and out of the room." usually mean something configured wrong.
Any exceptions in the logs (openmeetings.log and red5sip.log
On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com> wrote:
Sorry also Asterisk 11
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com> wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:07 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Do you have "Enable SIP transport in the room" checked for the room you are testing?
On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the reply, I went back and rechecked my setup. I have completed all the steps according to the integration document.
I found the following document on the wiki: https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to handle calls like onBWCheck, add a service provider. (it is my understanding this error is to be expect as it is not being used?)
So where would I start to try and figure out why there is no sip dialer available?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with Asterisk. That being said, I can find no documentation that tells the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how to make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: VOIP and Sip Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
Additionally red5sip connects to red5 server directly, not to the swf
client, so contents of config.xml is ignored while connecting by red5sip
On 27 July 2014 12:37, Maxim Solodovnik <so...@gmail.com> wrote:
> Hello Horas,
>
> Please write to user mailing list.
>
> I currently have no host configured in /webapps/openmeetings/public/
> config.xml file, all hosts are allowed
> Why do you need to limit host in this file?
>
>
> On 27 July 2014 04:44, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
>> Hi Maxim,
>>
>>
>>
>> Can I have your thoughts on the following:
>>
>>
>>
>> I am unable to get the sip agent to bind to 127.0.0.1. It refuses to
>> bind unless I have bind it to the same address that is in red5home
>> /webapps/openmeetings/public/config.xml
>>
>>
>>
>> The problem appears to be either that the SIP protocol wants to use
>> 127.0.0.1 for the subscribe or invites and SIP agent is bound to the Public
>> IP address. Therefore it is generating the error for seqno 2 which would
>> be the SIP Invite (I am assuming). I have not been able to get the SIP
>> tansport to bind to 127.0.0.1 which would probably solve this problem.
>>
>>
>>
>> Your thoughts/
>>
>>
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Friday, July 25, 2014 7:22 AM
>> *To:* Horace Miles
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> hope you will be able to fix it, please let ne know if additional help is
>> required
>>
>>
>>
>> On 25 July 2014 20:53, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> Hey thanks for the files.
>>
>>
>>
>> I compared and I have found the following:
>>
>>
>>
>> It appears the integration is setup for for a box that is NAT’ed. I
>> thought openmeetings had to be on a static public IP address?
>>
>>
>>
>> So I changed every place that is referencing 127.0.0.1 to my IP address.
>>
>>
>>
>> The Sip Agent/Openmeetings Manager does not come into the room until I
>> restart Asterisk. I can see it successfully logging on and then
>> immediately logging off. The room is successfully spawned.
>>
>>
>>
>> There seem to be a problem with the manager once it signs on with the sip
>> handshake (again I am guessing)
>>
>>
>>
>> chan_sip.c:4164 retrans_pkt: Retransmission timeout reached on
>> transmission #########@127.0.0.1 for seqno 2 (Critical Response) see……
>> Packet timed out afer 32000ms with no response.
>>
>>
>>
>> I will load wireshark later today on the PBX to see what else I might
>> find.
>>
>>
>>
>> Thanks for all your help.
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Thursday, July 24, 2014 8:44 AM
>> *To:* Horace Miles
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> uploaded
>>
>>
>>
>> On 24 July 2014 20:40, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> Maxim,
>>
>>
>>
>> Thanks I appreciate it very much.
>>
>>
>>
>> I have created you an account on my cloud server :
>> http://mycloud.myit-solutions.com
>>
>> Login: mmaxim
>>
>> Password: chief123
>>
>>
>>
>> There is a shared folder labeled openmeetings. You can upload the files
>> there. You have 5 GB of space.
>>
>>
>>
>> Let me know if you have any problems with this. I won’t be available
>> again until tonight but I will look at that time.
>>
>>
>>
>> Thanks a million
>>
>>
>>
>> Miles
>>
>>
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Wednesday, July 23, 2014 7:10 AM
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> I can privately send you all mine asterisk config files, so you can
>> compare
>>
>> additionally I can send both red5sip and OM, but I need some place like
>> dropbox for this
>>
>>
>>
>>
>>
>> On 23 July 2014 20:51, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> OK I will down load this evening and see what happens..
>>
>>
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Tuesday, July 22, 2014 11:28 PM
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> Hello Horace,
>>
>>
>>
>> just have checked, 3.0.3 seems to work as expected (at least 'SIP
>> Transport' sitting in the room)
>>
>> There are some NPEs in logs (will take a looks at it as soon as will have
>> some time)
>>
>>
>>
>> On 23 July 2014 12:13, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> Ok thanks Maxim
>>
>>
>>
>>
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Tuesday, July 22, 2014 7:17 AM
>>
>>
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> I'll try to find server with configured Asterisk and try to double-check
>>
>>
>>
>> On 22 July 2014 20:37, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> Thanks I made the change prior to sending the email. There appears to be
>> something else missing:
>>
>> There appears to be a entry missing in the /etc/asterisk/func_odbc.conf:
>> file for ${EXTEN}
>>
>> I am probably wrong. But I can’t figure out how this is making the call
>> to the database.
>>
>> I don’t find any SQL statement in the /etc/asterisk/func_odbc.conf file
>> and I am not sure how to construct one there that would work.
>>
>> Would I simply add
>>
>> [EXTEN]
>> dsn=asterisk
>> readsql=SELECT confno from room where confno = @EXTEN – NOT SURE HOW TO
>> GET THE ROOMID INTO THIS VARIABLE
>>
>>
>>
>> Thanks ahead of time
>>
>>
>>
>> Miles
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Tuesday, July 22, 2014 6:23 AM
>>
>>
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> yes, this line need to be corrected
>>
>> openmeetings/rooms -> openmeetings/room
>>
>>
>>
>> guess this is the problem
>>
>>
>>
>> On 22 July 2014 19:43, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> Thanks Maxim,
>>
>> I have been trying to figure this out, I am knew to it all but on a steep
>> learning curve.
>>
>>
>>
>> I do have a question about the asterisk extensions.conf
>>
>> exten =>
>> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
>>
>>
>>
>> Does the above line check the openmeetings database rooms table for the
>> confno and returns ok if it finds it and notavail if it doesn’t?
>>
>>
>>
>> I am getting the following warning:
>>
>> Chan_sip.c.:25184 handle_request_infite: Call from ‘red5sip_user’ (
>> 98.0.0.0:5070) to extension ‘40016’ rejected because extension not found
>> in context “rooms-red5sip” I don’t recall seeing this error before. But
>> if the “exten” line is checking the database openmeetings and looking for
>> rooms table it does not exist. There is a table name room but no rooms.
>>
>>
>>
>> Am I reading this correctly?
>>
>>
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Tuesday, July 22, 2014 4:27 AM
>>
>>
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> AFAIK dial in/out conference room was working as expected
>>
>> Not sure if we still have infrastructure test current version
>>
>>
>>
>> Will try to ask someone
>>
>>
>>
>> On 21 July 2014 20:17, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> Ok on the sip transport, I will try to figure out why it keep popping in
>> and out.
>>
>> However, I am not understanding concerning the Asterisk config. The
>> asterisk config I am using is from the install and modification as stated
>> from the VOIP/SIP 3.0 integration package. The instructions don’t say a
>> sip trunk or outside provider is required. However, I am unable to
>> succesfull make calls from a conference room to a phone and visa versa.
>> Which I thought I was suppose to be able to do after the integration. Per
>> the below instructions I guess I am asking what am I missing from below?
>>
>> *Feature Matrix:*
>>
>> *Feature*
>>
>> *Description*
>>
>> 1) Dial-In
>>
>> A phone number is provided which you can give to anybody to "Dial-In" via
>> usual landlane/phone into the conference room of OpenMeetings - Every room
>> has its own phone number. Currently room gets number
>> like 400<Id of room>. Maybe should move phone prefix to settings,
>> currently it hardcoded.
>>
>> 2) Dial-Out
>>
>> The users in the conference room can call anybody outside of the
>> conference room by entering the phone number in the conference room - In
>> room actions menu exist "SIP dialer". When user clicked dialer window
>> appears.
>> Currently calls can't be dropped from Openmeetings, tbd
>>
>> 3) Multiple Dial-In
>>
>> You can give away multiple numbers and do the same as described in case
>> (1). Multiple Dial-In is achieved by configuring the SIP-server (Asterisk).
>> It is possible to create multiple extensions (phone numbers) in Asterisk
>> configuration that will be redirects to single conference room.
>>
>> 4) Multiple Dial-Out
>>
>> You can dial multiple numbers from within the conference room - From
>> within conference can be dialed multiple numbers.
>>
>> *Main difference to native Red5-Phone project*
>>
>>
>>
>>
>>
>>
>>
>> *rom:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Monday, July 21, 2014 5:36 AM
>>
>>
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> If I do remember correctly
>>
>> SIP transport should enter the room and be in room as long as there other
>> users in it.
>>
>>
>>
>> Possibility to call to phone numbers depends on your Asterisk config.
>>
>>
>>
>> I'll try to fix documentation ASAP
>>
>>
>>
>> On 21 July 2014 18:55, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> Thanks Maxim,
>>
>> Let me make sure I understand about the sip transport. It should not be
>> popping in and out of the room? On my box I keep getting Sip Transport has
>> exited the room.
>>
>>
>>
>> When properly configured, should I be able to call land and cell phones
>> without a need for another server or;
>>
>> 1. Do I need to subscribe to a VOIP service provider
>>
>> 2. Configure Asterisk as a sip trunk to use google voice or some
>> other solution?
>>
>> Also if you could have someone correct this line in the instructions of
>> extensions.conf it will help to eliminate at least one error
>>
>>
>>
>>
>> *[rooms-red5sip]exten =>
>> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil)
>> <<<<<<<<<<<< should be “notavail” exten =>
>> _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)exten =>
>> _400X!,n(notavail),Hangup *
>>
>>
>>
>> *Thanks ahead of time*
>>
>>
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Monday, July 21, 2014 4:00 AM
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> Hello Horace,
>>
>>
>>
>> jsvc can be used to start java application as service
>>
>> I don't really like it (it was unstable when I used it)
>>
>> I prefer to write init.d script
>>
>>
>>
>> I see no errors in your log
>>
>> If everything is OK SIP transport should be in the room
>>
>> All 3 logs should be checked to have no errors
>>
>> I usually run asterisk in debug mode while setting everything up
>>
>>
>>
>>
>>
>>
>>
>> On 20 July 2014 23:55, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> Additonally the VOIP and SIP integration 3.0 instructions do not
>> mention installing jsvc. Is it still a requirement to install jsvc under
>> 3.0 as it was under 2.0?:
>>
>> *apt-get install jsvc*
>>
>>
>>
>>
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Friday, July 18, 2014 11:18 AM
>>
>>
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> will try to take a look a look at it tomorrow, too late here ...
>>
>>
>>
>> On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> Ok I will restart red5sip service and red5 and then send a new log
>>
>>
>>
>>
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Friday, July 18, 2014 10:06 AM
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> got the full trace in other email, will try to check code
>>
>>
>>
>> On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> Did miss understand what you were asking for?
>>
>>
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Friday, July 18, 2014 8:56 AM
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> would be more helpful to get full stack instead of "Red5sip log : Error
>> o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
>>
>>
>>
>> On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> Openmeetings log says confBridgeList authentication is failing. I will
>> check to make sure I didn’t change a password there..
>>
>> Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException:
>> Null
>>
>>
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Friday, July 18, 2014 8:43 AM
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> this "I have a sip transport that keeps popping in and out of the room."
>> usually mean something configured wrong.
>>
>> Any exceptions in the logs (openmeetings.log and red5sip.log
>>
>>
>>
>> On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> Sorry also Asterisk 11
>>
>>
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Friday, July 18, 2014 8:10 AM
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> Additionally, what version are you using?
>>
>>
>>
>> On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> Probably not, since I just went into a public room.. let me create a
>> room..
>>
>>
>>
>>
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Friday, July 18, 2014 8:07 AM
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> Do you have "*Enable SIP transport in the room*" checked for the room
>> you are testing?
>>
>>
>>
>> On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> Maxim thanks for the reply, I went back and rechecked my setup. I have
>> completed all the steps according to the integration document.
>>
>> I found the following document on the wiki:
>> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
>>
>>
>>
>> According to this document I should get a sip dialer under the rooms
>> actions menu. But I have no dialer there.
>>
>>
>>
>> The only error I see in the red5sip window is
>>
>> 18 Jul 07:50:11 . [nioProcessor-2]:[INFO ]
>> o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to
>> handle calls like onBWCheck, add a service provider. (it is my
>> understanding this error is to be expect as it is not being used?)
>>
>>
>>
>> So where would I start to try and figure out why there is no sip dialer
>> available?
>>
>>
>>
>>
>>
>>
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Friday, July 18, 2014 7:15 AM
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> http://openmeetings.apache.org/voip-sip-integration.html
>>
>>
>>
>> On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> It is nice that Openmeetings provided a way to integrate VOIP and Sip
>> with Asterisk. That being said, I can find no documentation that tells the
>> following:
>>
>> If the integration was successful?
>>
>> What icons should show up where etc.
>>
>> What actions can be taken by an admin or a user for that matter i.e. how
>> to make a phone call out or in.
>>
>> Did I miss something somewhere?
>>
>>
>>
>> Miles
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
Re: VOIP and Sip Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
Hello Horas,
Please write to user mailing list.
I currently have no host configured in /webapps/openmeetings/public/
config.xml file, all hosts are allowed
Why do you need to limit host in this file?
On 27 July 2014 04:44, Horace Miles <Ho...@myit-solutions.com> wrote:
> Hi Maxim,
>
>
>
> Can I have your thoughts on the following:
>
>
>
> I am unable to get the sip agent to bind to 127.0.0.1. It refuses to bind
> unless I have bind it to the same address that is in red5home
> /webapps/openmeetings/public/config.xml
>
>
>
> The problem appears to be either that the SIP protocol wants to use
> 127.0.0.1 for the subscribe or invites and SIP agent is bound to the Public
> IP address. Therefore it is generating the error for seqno 2 which would
> be the SIP Invite (I am assuming). I have not been able to get the SIP
> tansport to bind to 127.0.0.1 which would probably solve this problem.
>
>
>
> Your thoughts/
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 25, 2014 7:22 AM
> *To:* Horace Miles
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> hope you will be able to fix it, please let ne know if additional help is
> required
>
>
>
> On 25 July 2014 20:53, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Hey thanks for the files.
>
>
>
> I compared and I have found the following:
>
>
>
> It appears the integration is setup for for a box that is NAT’ed. I
> thought openmeetings had to be on a static public IP address?
>
>
>
> So I changed every place that is referencing 127.0.0.1 to my IP address.
>
>
>
> The Sip Agent/Openmeetings Manager does not come into the room until I
> restart Asterisk. I can see it successfully logging on and then
> immediately logging off. The room is successfully spawned.
>
>
>
> There seem to be a problem with the manager once it signs on with the sip
> handshake (again I am guessing)
>
>
>
> chan_sip.c:4164 retrans_pkt: Retransmission timeout reached on
> transmission #########@127.0.0.1 for seqno 2 (Critical Response) see……
> Packet timed out afer 32000ms with no response.
>
>
>
> I will load wireshark later today on the PBX to see what else I might find.
>
>
>
> Thanks for all your help.
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Thursday, July 24, 2014 8:44 AM
> *To:* Horace Miles
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> uploaded
>
>
>
> On 24 July 2014 20:40, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim,
>
>
>
> Thanks I appreciate it very much.
>
>
>
> I have created you an account on my cloud server :
> http://mycloud.myit-solutions.com
>
> Login: mmaxim
>
> Password: chief123
>
>
>
> There is a shared folder labeled openmeetings. You can upload the files
> there. You have 5 GB of space.
>
>
>
> Let me know if you have any problems with this. I won’t be available
> again until tonight but I will look at that time.
>
>
>
> Thanks a million
>
>
>
> Miles
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Wednesday, July 23, 2014 7:10 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> I can privately send you all mine asterisk config files, so you can compare
>
> additionally I can send both red5sip and OM, but I need some place like
> dropbox for this
>
>
>
>
>
> On 23 July 2014 20:51, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> OK I will down load this evening and see what happens..
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Tuesday, July 22, 2014 11:28 PM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Hello Horace,
>
>
>
> just have checked, 3.0.3 seems to work as expected (at least 'SIP
> Transport' sitting in the room)
>
> There are some NPEs in logs (will take a looks at it as soon as will have
> some time)
>
>
>
> On 23 July 2014 12:13, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok thanks Maxim
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Tuesday, July 22, 2014 7:17 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> I'll try to find server with configured Asterisk and try to double-check
>
>
>
> On 22 July 2014 20:37, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks I made the change prior to sending the email. There appears to be
> something else missing:
>
> There appears to be a entry missing in the /etc/asterisk/func_odbc.conf:
> file for ${EXTEN}
>
> I am probably wrong. But I can’t figure out how this is making the call
> to the database.
>
> I don’t find any SQL statement in the /etc/asterisk/func_odbc.conf file
> and I am not sure how to construct one there that would work.
>
> Would I simply add
>
> [EXTEN]
> dsn=asterisk
> readsql=SELECT confno from room where confno = @EXTEN – NOT SURE HOW TO
> GET THE ROOMID INTO THIS VARIABLE
>
>
>
> Thanks ahead of time
>
>
>
> Miles
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Tuesday, July 22, 2014 6:23 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> yes, this line need to be corrected
>
> openmeetings/rooms -> openmeetings/room
>
>
>
> guess this is the problem
>
>
>
> On 22 July 2014 19:43, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks Maxim,
>
> I have been trying to figure this out, I am knew to it all but on a steep
> learning curve.
>
>
>
> I do have a question about the asterisk extensions.conf
>
> exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
>
>
>
> Does the above line check the openmeetings database rooms table for the
> confno and returns ok if it finds it and notavail if it doesn’t?
>
>
>
> I am getting the following warning:
>
> Chan_sip.c.:25184 handle_request_infite: Call from ‘red5sip_user’ (
> 98.0.0.0:5070) to extension ‘40016’ rejected because extension not found
> in context “rooms-red5sip” I don’t recall seeing this error before. But
> if the “exten” line is checking the database openmeetings and looking for
> rooms table it does not exist. There is a table name room but no rooms.
>
>
>
> Am I reading this correctly?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Tuesday, July 22, 2014 4:27 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> AFAIK dial in/out conference room was working as expected
>
> Not sure if we still have infrastructure test current version
>
>
>
> Will try to ask someone
>
>
>
> On 21 July 2014 20:17, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok on the sip transport, I will try to figure out why it keep popping in
> and out.
>
> However, I am not understanding concerning the Asterisk config. The
> asterisk config I am using is from the install and modification as stated
> from the VOIP/SIP 3.0 integration package. The instructions don’t say a
> sip trunk or outside provider is required. However, I am unable to
> succesfull make calls from a conference room to a phone and visa versa.
> Which I thought I was suppose to be able to do after the integration. Per
> the below instructions I guess I am asking what am I missing from below?
>
> *Feature Matrix:*
>
> *Feature*
>
> *Description*
>
> 1) Dial-In
>
> A phone number is provided which you can give to anybody to "Dial-In" via
> usual landlane/phone into the conference room of OpenMeetings - Every room
> has its own phone number. Currently room gets number
> like 400<Id of room>. Maybe should move phone prefix to settings,
> currently it hardcoded.
>
> 2) Dial-Out
>
> The users in the conference room can call anybody outside of the
> conference room by entering the phone number in the conference room - In
> room actions menu exist "SIP dialer". When user clicked dialer window
> appears.
> Currently calls can't be dropped from Openmeetings, tbd
>
> 3) Multiple Dial-In
>
> You can give away multiple numbers and do the same as described in case
> (1). Multiple Dial-In is achieved by configuring the SIP-server (Asterisk).
> It is possible to create multiple extensions (phone numbers) in Asterisk
> configuration that will be redirects to single conference room.
>
> 4) Multiple Dial-Out
>
> You can dial multiple numbers from within the conference room - From
> within conference can be dialed multiple numbers.
>
> *Main difference to native Red5-Phone project*
>
>
>
>
>
>
>
> *rom:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Monday, July 21, 2014 5:36 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> If I do remember correctly
>
> SIP transport should enter the room and be in room as long as there other
> users in it.
>
>
>
> Possibility to call to phone numbers depends on your Asterisk config.
>
>
>
> I'll try to fix documentation ASAP
>
>
>
> On 21 July 2014 18:55, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks Maxim,
>
> Let me make sure I understand about the sip transport. It should not be
> popping in and out of the room? On my box I keep getting Sip Transport has
> exited the room.
>
>
>
> When properly configured, should I be able to call land and cell phones
> without a need for another server or;
>
> 1. Do I need to subscribe to a VOIP service provider
>
> 2. Configure Asterisk as a sip trunk to use google voice or some
> other solution?
>
> Also if you could have someone correct this line in the instructions of
> extensions.conf it will help to eliminate at least one error
>
>
>
>
> *[rooms-red5sip]exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil)
> <<<<<<<<<<<< should be “notavail”exten =>
> _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)exten =>
> _400X!,n(notavail),Hangup *
>
>
>
> *Thanks ahead of time*
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Monday, July 21, 2014 4:00 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Hello Horace,
>
>
>
> jsvc can be used to start java application as service
>
> I don't really like it (it was unstable when I used it)
>
> I prefer to write init.d script
>
>
>
> I see no errors in your log
>
> If everything is OK SIP transport should be in the room
>
> All 3 logs should be checked to have no errors
>
> I usually run asterisk in debug mode while setting everything up
>
>
>
>
>
>
>
> On 20 July 2014 23:55, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Additonally the VOIP and SIP integration 3.0 instructions do not mention
> installing jsvc. Is it still a requirement to install jsvc under 3.0 as it
> was under 2.0?:
>
> *apt-get install jsvc*
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 11:18 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> will try to take a look a look at it tomorrow, too late here ...
>
>
>
> On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok I will restart red5sip service and red5 and then send a new log
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 10:06 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> got the full trace in other email, will try to check code
>
>
>
> On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Did miss understand what you were asking for?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:56 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> would be more helpful to get full stack instead of "Red5sip log : Error
> o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
>
>
>
> On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Openmeetings log says confBridgeList authentication is failing. I will
> check to make sure I didn’t change a password there..
>
> Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException:
> Null
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:43 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> this "I have a sip transport that keeps popping in and out of the room."
> usually mean something configured wrong.
>
> Any exceptions in the logs (openmeetings.log and red5sip.log
>
>
>
> On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Sorry also Asterisk 11
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:10 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Additionally, what version are you using?
>
>
>
> On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Probably not, since I just went into a public room.. let me create a room..
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Do you have "*Enable SIP transport in the room*" checked for the room you
> are testing?
>
>
>
> On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the reply, I went back and rechecked my setup. I have
> completed all the steps according to the integration document.
>
> I found the following document on the wiki:
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
>
>
>
> According to this document I should get a sip dialer under the rooms
> actions menu. But I have no dialer there.
>
>
>
> The only error I see in the red5sip window is
>
> 18 Jul 07:50:11 . [nioProcessor-2]:[INFO ]
> o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to
> handle calls like onBWCheck, add a service provider. (it is my
> understanding this error is to be expect as it is not being used?)
>
>
>
> So where would I start to try and figure out why there is no sip dialer
> available?
>
>
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 7:15 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> http://openmeetings.apache.org/voip-sip-integration.html
>
>
>
> On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> It is nice that Openmeetings provided a way to integrate VOIP and Sip with
> Asterisk. That being said, I can find no documentation that tells the
> following:
>
> If the integration was successful?
>
> What icons should show up where etc.
>
> What actions can be taken by an admin or a user for that matter i.e. how
> to make a phone call out or in.
>
> Did I miss something somewhere?
>
>
>
> Miles
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
Re: VOIP and Sip Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
I can privately send you all mine asterisk config files, so you can compare
additionally I can send both red5sip and OM, but I need some place like
dropbox for this
On 23 July 2014 20:51, Horace Miles <Ho...@myit-solutions.com> wrote:
> OK I will down load this evening and see what happens..
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Tuesday, July 22, 2014 11:28 PM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Hello Horace,
>
>
>
> just have checked, 3.0.3 seems to work as expected (at least 'SIP
> Transport' sitting in the room)
>
> There are some NPEs in logs (will take a looks at it as soon as will have
> some time)
>
>
>
> On 23 July 2014 12:13, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok thanks Maxim
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Tuesday, July 22, 2014 7:17 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> I'll try to find server with configured Asterisk and try to double-check
>
>
>
> On 22 July 2014 20:37, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks I made the change prior to sending the email. There appears to be
> something else missing:
>
> There appears to be a entry missing in the /etc/asterisk/func_odbc.conf:
> file for ${EXTEN}
>
> I am probably wrong. But I can’t figure out how this is making the call
> to the database.
>
> I don’t find any SQL statement in the /etc/asterisk/func_odbc.conf file
> and I am not sure how to construct one there that would work.
>
> Would I simply add
>
> [EXTEN]
> dsn=asterisk
> readsql=SELECT confno from room where confno = @EXTEN – NOT SURE HOW TO
> GET THE ROOMID INTO THIS VARIABLE
>
>
>
> Thanks ahead of time
>
>
>
> Miles
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Tuesday, July 22, 2014 6:23 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> yes, this line need to be corrected
>
> openmeetings/rooms -> openmeetings/room
>
>
>
> guess this is the problem
>
>
>
> On 22 July 2014 19:43, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks Maxim,
>
> I have been trying to figure this out, I am knew to it all but on a steep
> learning curve.
>
>
>
> I do have a question about the asterisk extensions.conf
>
> exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
>
>
>
> Does the above line check the openmeetings database rooms table for the
> confno and returns ok if it finds it and notavail if it doesn’t?
>
>
>
> I am getting the following warning:
>
> Chan_sip.c.:25184 handle_request_infite: Call from ‘red5sip_user’ (
> 98.0.0.0:5070) to extension ‘40016’ rejected because extension not found
> in context “rooms-red5sip” I don’t recall seeing this error before. But
> if the “exten” line is checking the database openmeetings and looking for
> rooms table it does not exist. There is a table name room but no rooms.
>
>
>
> Am I reading this correctly?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Tuesday, July 22, 2014 4:27 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> AFAIK dial in/out conference room was working as expected
>
> Not sure if we still have infrastructure test current version
>
>
>
> Will try to ask someone
>
>
>
> On 21 July 2014 20:17, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok on the sip transport, I will try to figure out why it keep popping in
> and out.
>
> However, I am not understanding concerning the Asterisk config. The
> asterisk config I am using is from the install and modification as stated
> from the VOIP/SIP 3.0 integration package. The instructions don’t say a
> sip trunk or outside provider is required. However, I am unable to
> succesfull make calls from a conference room to a phone and visa versa.
> Which I thought I was suppose to be able to do after the integration. Per
> the below instructions I guess I am asking what am I missing from below?
>
> *Feature Matrix:*
>
> *Feature*
>
> *Description*
>
> 1) Dial-In
>
> A phone number is provided which you can give to anybody to "Dial-In" via
> usual landlane/phone into the conference room of OpenMeetings - Every room
> has its own phone number. Currently room gets number
> like 400<Id of room>. Maybe should move phone prefix to settings,
> currently it hardcoded.
>
> 2) Dial-Out
>
> The users in the conference room can call anybody outside of the
> conference room by entering the phone number in the conference room - In
> room actions menu exist "SIP dialer". When user clicked dialer window
> appears.
> Currently calls can't be dropped from Openmeetings, tbd
>
> 3) Multiple Dial-In
>
> You can give away multiple numbers and do the same as described in case
> (1). Multiple Dial-In is achieved by configuring the SIP-server (Asterisk).
> It is possible to create multiple extensions (phone numbers) in Asterisk
> configuration that will be redirects to single conference room.
>
> 4) Multiple Dial-Out
>
> You can dial multiple numbers from within the conference room - From
> within conference can be dialed multiple numbers.
>
> *Main difference to native Red5-Phone project*
>
>
>
>
>
>
>
> *rom:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Monday, July 21, 2014 5:36 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> If I do remember correctly
>
> SIP transport should enter the room and be in room as long as there other
> users in it.
>
>
>
> Possibility to call to phone numbers depends on your Asterisk config.
>
>
>
> I'll try to fix documentation ASAP
>
>
>
> On 21 July 2014 18:55, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks Maxim,
>
> Let me make sure I understand about the sip transport. It should not be
> popping in and out of the room? On my box I keep getting Sip Transport has
> exited the room.
>
>
>
> When properly configured, should I be able to call land and cell phones
> without a need for another server or;
>
> 1. Do I need to subscribe to a VOIP service provider
>
> 2. Configure Asterisk as a sip trunk to use google voice or some
> other solution?
>
> Also if you could have someone correct this line in the instructions of
> extensions.conf it will help to eliminate at least one error
>
>
>
>
> *[rooms-red5sip]exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil)
> <<<<<<<<<<<< should be “notavail”exten =>
> _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)exten =>
> _400X!,n(notavail),Hangup *
>
>
>
> *Thanks ahead of time*
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Monday, July 21, 2014 4:00 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Hello Horace,
>
>
>
> jsvc can be used to start java application as service
>
> I don't really like it (it was unstable when I used it)
>
> I prefer to write init.d script
>
>
>
> I see no errors in your log
>
> If everything is OK SIP transport should be in the room
>
> All 3 logs should be checked to have no errors
>
> I usually run asterisk in debug mode while setting everything up
>
>
>
>
>
>
>
> On 20 July 2014 23:55, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Additonally the VOIP and SIP integration 3.0 instructions do not mention
> installing jsvc. Is it still a requirement to install jsvc under 3.0 as it
> was under 2.0?:
>
> *apt-get install jsvc*
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 11:18 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> will try to take a look a look at it tomorrow, too late here ...
>
>
>
> On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok I will restart red5sip service and red5 and then send a new log
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 10:06 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> got the full trace in other email, will try to check code
>
>
>
> On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Did miss understand what you were asking for?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:56 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> would be more helpful to get full stack instead of "Red5sip log : Error
> o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
>
>
>
> On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Openmeetings log says confBridgeList authentication is failing. I will
> check to make sure I didn’t change a password there..
>
> Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException:
> Null
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:43 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> this "I have a sip transport that keeps popping in and out of the room."
> usually mean something configured wrong.
>
> Any exceptions in the logs (openmeetings.log and red5sip.log
>
>
>
> On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Sorry also Asterisk 11
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:10 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Additionally, what version are you using?
>
>
>
> On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Probably not, since I just went into a public room.. let me create a room..
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Do you have "*Enable SIP transport in the room*" checked for the room you
> are testing?
>
>
>
> On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the reply, I went back and rechecked my setup. I have
> completed all the steps according to the integration document.
>
> I found the following document on the wiki:
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
>
>
>
> According to this document I should get a sip dialer under the rooms
> actions menu. But I have no dialer there.
>
>
>
> The only error I see in the red5sip window is
>
> 18 Jul 07:50:11 . [nioProcessor-2]:[INFO ]
> o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to
> handle calls like onBWCheck, add a service provider. (it is my
> understanding this error is to be expect as it is not being used?)
>
>
>
> So where would I start to try and figure out why there is no sip dialer
> available?
>
>
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 7:15 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> http://openmeetings.apache.org/voip-sip-integration.html
>
>
>
> On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> It is nice that Openmeetings provided a way to integrate VOIP and Sip with
> Asterisk. That being said, I can find no documentation that tells the
> following:
>
> If the integration was successful?
>
> What icons should show up where etc.
>
> What actions can be taken by an admin or a user for that matter i.e. how
> to make a phone call out or in.
>
> Did I miss something somewhere?
>
>
>
> Miles
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: VOIP and Sip Integration
Posted by Horace Miles <Ho...@myit-solutions.com>.
OK I will down load this evening and see what happens..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, July 22, 2014 11:28 PM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Hello Horace,
just have checked, 3.0.3 seems to work as expected (at least 'SIP Transport' sitting in the room)
There are some NPEs in logs (will take a looks at it as soon as will have some time)
On 23 July 2014 12:13, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok thanks Maxim
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, July 22, 2014 7:17 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
I'll try to find server with configured Asterisk and try to double-check
On 22 July 2014 20:37, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks I made the change prior to sending the email. There appears to be something else missing:
There appears to be a entry missing in the /etc/asterisk/func_odbc.conf: file for ${EXTEN}
I am probably wrong. But I can’t figure out how this is making the call to the database.
I don’t find any SQL statement in the /etc/asterisk/func_odbc.conf file and I am not sure how to construct one there that would work.
Would I simply add
[EXTEN]
dsn=asterisk
readsql=SELECT confno from room where confno = @EXTEN – NOT SURE HOW TO GET THE ROOMID INTO THIS VARIABLE
Thanks ahead of time
Miles
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, July 22, 2014 6:23 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
yes, this line need to be corrected
openmeetings/rooms -> openmeetings/room
guess this is the problem
On 22 July 2014 19:43, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Maxim,
I have been trying to figure this out, I am knew to it all but on a steep learning curve.
I do have a question about the asterisk extensions.conf
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
Does the above line check the openmeetings database rooms table for the confno and returns ok if it finds it and notavail if it doesn’t?
I am getting the following warning:
Chan_sip.c.:25184 handle_request_infite: Call from ‘red5sip_user’ (98.0.0.0:5070) to extension ‘40016’ rejected because extension not found in context “rooms-red5sip” I don’t recall seeing this error before. But if the “exten” line is checking the database openmeetings and looking for rooms table it does not exist. There is a table name room but no rooms.
Am I reading this correctly?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, July 22, 2014 4:27 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
AFAIK dial in/out conference room was working as expected
Not sure if we still have infrastructure test current version
Will try to ask someone
On 21 July 2014 20:17, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok on the sip transport, I will try to figure out why it keep popping in and out.
However, I am not understanding concerning the Asterisk config. The asterisk config I am using is from the install and modification as stated from the VOIP/SIP 3.0 integration package. The instructions don’t say a sip trunk or outside provider is required. However, I am unable to succesfull make calls from a conference room to a phone and visa versa. Which I thought I was suppose to be able to do after the integration. Per the below instructions I guess I am asking what am I missing from below?
Feature Matrix:
Feature
Description
1) Dial-In
A phone number is provided which you can give to anybody to "Dial-In" via usual landlane/phone into the conference room of OpenMeetings - Every room has its own phone number. Currently room gets number
like 400<Id of room>. Maybe should move phone prefix to settings,
currently it hardcoded.
2) Dial-Out
The users in the conference room can call anybody outside of the conference room by entering the phone number in the conference room - In room actions menu exist "SIP dialer". When user clicked dialer window appears.
Currently calls can't be dropped from Openmeetings, tbd
3) Multiple Dial-In
You can give away multiple numbers and do the same as described in case (1). Multiple Dial-In is achieved by configuring the SIP-server (Asterisk). It is possible to create multiple extensions (phone numbers) in Asterisk configuration that will be redirects to single conference room.
4) Multiple Dial-Out
You can dial multiple numbers from within the conference room - From within conference can be dialed multiple numbers.
Main difference to native Red5-Phone project
rom: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, July 21, 2014 5:36 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
If I do remember correctly
SIP transport should enter the room and be in room as long as there other users in it.
Possibility to call to phone numbers depends on your Asterisk config.
I'll try to fix documentation ASAP
On 21 July 2014 18:55, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Maxim,
Let me make sure I understand about the sip transport. It should not be popping in and out of the room? On my box I keep getting Sip Transport has exited the room.
When properly configured, should I be able to call land and cell phones without a need for another server or;
1. Do I need to subscribe to a VOIP service provider
2. Configure Asterisk as a sip trunk to use google voice or some other solution?
Also if you could have someone correct this line in the instructions of extensions.conf it will help to eliminate at least one error
[rooms-red5sip]
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil) <<<<<<<<<<<< should be “notavail”
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
exten => _400X!,n(notavail),Hangup
Thanks ahead of time
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, July 21, 2014 4:00 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Hello Horace,
jsvc can be used to start java application as service
I don't really like it (it was unstable when I used it)
I prefer to write init.d script
I see no errors in your log
If everything is OK SIP transport should be in the room
All 3 logs should be checked to have no errors
I usually run asterisk in debug mode while setting everything up
On 20 July 2014 23:55, Horace Miles <Ho...@myit-solutions.com> wrote:
Additonally the VOIP and SIP integration 3.0 instructions do not mention installing jsvc. Is it still a requirement to install jsvc under 3.0 as it was under 2.0?:
apt-get install jsvc
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 11:18 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
will try to take a look a look at it tomorrow, too late here ...
On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok I will restart red5sip service and red5 and then send a new log
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 10:06 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
got the full trace in other email, will try to check code
On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com> wrote:
Did miss understand what you were asking for?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:56 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
would be more helpful to get full stack instead of "Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com> wrote:
Openmeetings log says confBridgeList authentication is failing. I will check to make sure I didn’t change a password there..
Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:43 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
this "I have a sip transport that keeps popping in and out of the room." usually mean something configured wrong.
Any exceptions in the logs (openmeetings.log and red5sip.log
On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com> wrote:
Sorry also Asterisk 11
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com> wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:07 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Do you have "Enable SIP transport in the room" checked for the room you are testing?
On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the reply, I went back and rechecked my setup. I have completed all the steps according to the integration document.
I found the following document on the wiki: https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to handle calls like onBWCheck, add a service provider. (it is my understanding this error is to be expect as it is not being used?)
So where would I start to try and figure out why there is no sip dialer available?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with Asterisk. That being said, I can find no documentation that tells the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how to make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: VOIP and Sip Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
Hello Horace,
just have checked, 3.0.3 seems to work as expected (at least 'SIP
Transport' sitting in the room)
There are some NPEs in logs (will take a looks at it as soon as will have
some time)
On 23 July 2014 12:13, Horace Miles <Ho...@myit-solutions.com> wrote:
> Ok thanks Maxim
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Tuesday, July 22, 2014 7:17 AM
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> I'll try to find server with configured Asterisk and try to double-check
>
>
>
> On 22 July 2014 20:37, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks I made the change prior to sending the email. There appears to be
> something else missing:
>
> There appears to be a entry missing in the /etc/asterisk/func_odbc.conf:
> file for ${EXTEN}
>
> I am probably wrong. But I can’t figure out how this is making the call
> to the database.
>
> I don’t find any SQL statement in the /etc/asterisk/func_odbc.conf file
> and I am not sure how to construct one there that would work.
>
> Would I simply add
>
> [EXTEN]
> dsn=asterisk
> readsql=SELECT confno from room where confno = @EXTEN – NOT SURE HOW TO
> GET THE ROOMID INTO THIS VARIABLE
>
>
>
> Thanks ahead of time
>
>
>
> Miles
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Tuesday, July 22, 2014 6:23 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> yes, this line need to be corrected
>
> openmeetings/rooms -> openmeetings/room
>
>
>
> guess this is the problem
>
>
>
> On 22 July 2014 19:43, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks Maxim,
>
> I have been trying to figure this out, I am knew to it all but on a steep
> learning curve.
>
>
>
> I do have a question about the asterisk extensions.conf
>
> exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
>
>
>
> Does the above line check the openmeetings database rooms table for the
> confno and returns ok if it finds it and notavail if it doesn’t?
>
>
>
> I am getting the following warning:
>
> Chan_sip.c.:25184 handle_request_infite: Call from ‘red5sip_user’ (
> 98.0.0.0:5070) to extension ‘40016’ rejected because extension not found
> in context “rooms-red5sip” I don’t recall seeing this error before. But
> if the “exten” line is checking the database openmeetings and looking for
> rooms table it does not exist. There is a table name room but no rooms.
>
>
>
> Am I reading this correctly?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Tuesday, July 22, 2014 4:27 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> AFAIK dial in/out conference room was working as expected
>
> Not sure if we still have infrastructure test current version
>
>
>
> Will try to ask someone
>
>
>
> On 21 July 2014 20:17, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok on the sip transport, I will try to figure out why it keep popping in
> and out.
>
> However, I am not understanding concerning the Asterisk config. The
> asterisk config I am using is from the install and modification as stated
> from the VOIP/SIP 3.0 integration package. The instructions don’t say a
> sip trunk or outside provider is required. However, I am unable to
> succesfull make calls from a conference room to a phone and visa versa.
> Which I thought I was suppose to be able to do after the integration. Per
> the below instructions I guess I am asking what am I missing from below?
>
> *Feature Matrix:*
>
> *Feature*
>
> *Description*
>
> 1) Dial-In
>
> A phone number is provided which you can give to anybody to "Dial-In" via
> usual landlane/phone into the conference room of OpenMeetings - Every room
> has its own phone number. Currently room gets number
> like 400<Id of room>. Maybe should move phone prefix to settings,
> currently it hardcoded.
>
> 2) Dial-Out
>
> The users in the conference room can call anybody outside of the
> conference room by entering the phone number in the conference room - In
> room actions menu exist "SIP dialer". When user clicked dialer window
> appears.
> Currently calls can't be dropped from Openmeetings, tbd
>
> 3) Multiple Dial-In
>
> You can give away multiple numbers and do the same as described in case
> (1). Multiple Dial-In is achieved by configuring the SIP-server (Asterisk).
> It is possible to create multiple extensions (phone numbers) in Asterisk
> configuration that will be redirects to single conference room.
>
> 4) Multiple Dial-Out
>
> You can dial multiple numbers from within the conference room - From
> within conference can be dialed multiple numbers.
>
> *Main difference to native Red5-Phone project*
>
>
>
>
>
>
>
> *rom:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Monday, July 21, 2014 5:36 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> If I do remember correctly
>
> SIP transport should enter the room and be in room as long as there other
> users in it.
>
>
>
> Possibility to call to phone numbers depends on your Asterisk config.
>
>
>
> I'll try to fix documentation ASAP
>
>
>
> On 21 July 2014 18:55, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks Maxim,
>
> Let me make sure I understand about the sip transport. It should not be
> popping in and out of the room? On my box I keep getting Sip Transport has
> exited the room.
>
>
>
> When properly configured, should I be able to call land and cell phones
> without a need for another server or;
>
> 1. Do I need to subscribe to a VOIP service provider
>
> 2. Configure Asterisk as a sip trunk to use google voice or some
> other solution?
>
> Also if you could have someone correct this line in the instructions of
> extensions.conf it will help to eliminate at least one error
>
>
>
>
> *[rooms-red5sip]exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil)
> <<<<<<<<<<<< should be “notavail”exten =>
> _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)exten =>
> _400X!,n(notavail),Hangup *
>
>
>
> *Thanks ahead of time*
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Monday, July 21, 2014 4:00 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Hello Horace,
>
>
>
> jsvc can be used to start java application as service
>
> I don't really like it (it was unstable when I used it)
>
> I prefer to write init.d script
>
>
>
> I see no errors in your log
>
> If everything is OK SIP transport should be in the room
>
> All 3 logs should be checked to have no errors
>
> I usually run asterisk in debug mode while setting everything up
>
>
>
>
>
>
>
> On 20 July 2014 23:55, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Additonally the VOIP and SIP integration 3.0 instructions do not mention
> installing jsvc. Is it still a requirement to install jsvc under 3.0 as it
> was under 2.0?:
>
> *apt-get install jsvc*
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 11:18 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> will try to take a look a look at it tomorrow, too late here ...
>
>
>
> On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok I will restart red5sip service and red5 and then send a new log
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 10:06 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> got the full trace in other email, will try to check code
>
>
>
> On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Did miss understand what you were asking for?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:56 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> would be more helpful to get full stack instead of "Red5sip log : Error
> o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
>
>
>
> On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Openmeetings log says confBridgeList authentication is failing. I will
> check to make sure I didn’t change a password there..
>
> Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException:
> Null
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:43 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> this "I have a sip transport that keeps popping in and out of the room."
> usually mean something configured wrong.
>
> Any exceptions in the logs (openmeetings.log and red5sip.log
>
>
>
> On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Sorry also Asterisk 11
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:10 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Additionally, what version are you using?
>
>
>
> On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Probably not, since I just went into a public room.. let me create a room..
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Do you have "*Enable SIP transport in the room*" checked for the room you
> are testing?
>
>
>
> On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the reply, I went back and rechecked my setup. I have
> completed all the steps according to the integration document.
>
> I found the following document on the wiki:
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
>
>
>
> According to this document I should get a sip dialer under the rooms
> actions menu. But I have no dialer there.
>
>
>
> The only error I see in the red5sip window is
>
> 18 Jul 07:50:11 . [nioProcessor-2]:[INFO ]
> o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to
> handle calls like onBWCheck, add a service provider. (it is my
> understanding this error is to be expect as it is not being used?)
>
>
>
> So where would I start to try and figure out why there is no sip dialer
> available?
>
>
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 7:15 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> http://openmeetings.apache.org/voip-sip-integration.html
>
>
>
> On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> It is nice that Openmeetings provided a way to integrate VOIP and Sip with
> Asterisk. That being said, I can find no documentation that tells the
> following:
>
> If the integration was successful?
>
> What icons should show up where etc.
>
> What actions can be taken by an admin or a user for that matter i.e. how
> to make a phone call out or in.
>
> Did I miss something somewhere?
>
>
>
> Miles
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: VOIP and Sip Integration
Posted by Horace Miles <Ho...@myit-solutions.com>.
Ok thanks Maxim
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, July 22, 2014 7:17 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
I'll try to find server with configured Asterisk and try to double-check
On 22 July 2014 20:37, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks I made the change prior to sending the email. There appears to be something else missing:
There appears to be a entry missing in the /etc/asterisk/func_odbc.conf: file for ${EXTEN}
I am probably wrong. But I can’t figure out how this is making the call to the database.
I don’t find any SQL statement in the /etc/asterisk/func_odbc.conf file and I am not sure how to construct one there that would work.
Would I simply add
[EXTEN]
dsn=asterisk
readsql=SELECT confno from room where confno = @EXTEN – NOT SURE HOW TO GET THE ROOMID INTO THIS VARIABLE
Thanks ahead of time
Miles
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, July 22, 2014 6:23 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
yes, this line need to be corrected
openmeetings/rooms -> openmeetings/room
guess this is the problem
On 22 July 2014 19:43, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Maxim,
I have been trying to figure this out, I am knew to it all but on a steep learning curve.
I do have a question about the asterisk extensions.conf
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
Does the above line check the openmeetings database rooms table for the confno and returns ok if it finds it and notavail if it doesn’t?
I am getting the following warning:
Chan_sip.c.:25184 handle_request_infite: Call from ‘red5sip_user’ (98.0.0.0:5070) to extension ‘40016’ rejected because extension not found in context “rooms-red5sip” I don’t recall seeing this error before. But if the “exten” line is checking the database openmeetings and looking for rooms table it does not exist. There is a table name room but no rooms.
Am I reading this correctly?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, July 22, 2014 4:27 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
AFAIK dial in/out conference room was working as expected
Not sure if we still have infrastructure test current version
Will try to ask someone
On 21 July 2014 20:17, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok on the sip transport, I will try to figure out why it keep popping in and out.
However, I am not understanding concerning the Asterisk config. The asterisk config I am using is from the install and modification as stated from the VOIP/SIP 3.0 integration package. The instructions don’t say a sip trunk or outside provider is required. However, I am unable to succesfull make calls from a conference room to a phone and visa versa. Which I thought I was suppose to be able to do after the integration. Per the below instructions I guess I am asking what am I missing from below?
Feature Matrix:
Feature
Description
1) Dial-In
A phone number is provided which you can give to anybody to "Dial-In" via usual landlane/phone into the conference room of OpenMeetings - Every room has its own phone number. Currently room gets number
like 400<Id of room>. Maybe should move phone prefix to settings,
currently it hardcoded.
2) Dial-Out
The users in the conference room can call anybody outside of the conference room by entering the phone number in the conference room - In room actions menu exist "SIP dialer". When user clicked dialer window appears.
Currently calls can't be dropped from Openmeetings, tbd
3) Multiple Dial-In
You can give away multiple numbers and do the same as described in case (1). Multiple Dial-In is achieved by configuring the SIP-server (Asterisk). It is possible to create multiple extensions (phone numbers) in Asterisk configuration that will be redirects to single conference room.
4) Multiple Dial-Out
You can dial multiple numbers from within the conference room - From within conference can be dialed multiple numbers.
Main difference to native Red5-Phone project
rom: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, July 21, 2014 5:36 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
If I do remember correctly
SIP transport should enter the room and be in room as long as there other users in it.
Possibility to call to phone numbers depends on your Asterisk config.
I'll try to fix documentation ASAP
On 21 July 2014 18:55, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Maxim,
Let me make sure I understand about the sip transport. It should not be popping in and out of the room? On my box I keep getting Sip Transport has exited the room.
When properly configured, should I be able to call land and cell phones without a need for another server or;
1. Do I need to subscribe to a VOIP service provider
2. Configure Asterisk as a sip trunk to use google voice or some other solution?
Also if you could have someone correct this line in the instructions of extensions.conf it will help to eliminate at least one error
[rooms-red5sip]
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil) <<<<<<<<<<<< should be “notavail”
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
exten => _400X!,n(notavail),Hangup
Thanks ahead of time
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, July 21, 2014 4:00 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Hello Horace,
jsvc can be used to start java application as service
I don't really like it (it was unstable when I used it)
I prefer to write init.d script
I see no errors in your log
If everything is OK SIP transport should be in the room
All 3 logs should be checked to have no errors
I usually run asterisk in debug mode while setting everything up
On 20 July 2014 23:55, Horace Miles <Ho...@myit-solutions.com> wrote:
Additonally the VOIP and SIP integration 3.0 instructions do not mention installing jsvc. Is it still a requirement to install jsvc under 3.0 as it was under 2.0?:
apt-get install jsvc
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 11:18 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
will try to take a look a look at it tomorrow, too late here ...
On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok I will restart red5sip service and red5 and then send a new log
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 10:06 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
got the full trace in other email, will try to check code
On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com> wrote:
Did miss understand what you were asking for?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:56 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
would be more helpful to get full stack instead of "Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com> wrote:
Openmeetings log says confBridgeList authentication is failing. I will check to make sure I didn’t change a password there..
Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:43 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
this "I have a sip transport that keeps popping in and out of the room." usually mean something configured wrong.
Any exceptions in the logs (openmeetings.log and red5sip.log
On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com> wrote:
Sorry also Asterisk 11
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com> wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:07 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Do you have "Enable SIP transport in the room" checked for the room you are testing?
On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the reply, I went back and rechecked my setup. I have completed all the steps according to the integration document.
I found the following document on the wiki: https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to handle calls like onBWCheck, add a service provider. (it is my understanding this error is to be expect as it is not being used?)
So where would I start to try and figure out why there is no sip dialer available?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with Asterisk. That being said, I can find no documentation that tells the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how to make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: VOIP and Sip Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
I'll try to find server with configured Asterisk and try to double-check
On 22 July 2014 20:37, Horace Miles <Ho...@myit-solutions.com> wrote:
> Thanks I made the change prior to sending the email. There appears to be
> something else missing:
>
> There appears to be a entry missing in the /etc/asterisk/func_odbc.conf:
> file for ${EXTEN}
>
> I am probably wrong. But I can’t figure out how this is making the call
> to the database.
>
> I don’t find any SQL statement in the /etc/asterisk/func_odbc.conf file
> and I am not sure how to construct one there that would work.
>
> Would I simply add
>
> [EXTEN]
> dsn=asterisk
> readsql=SELECT confno from room where confno = @EXTEN – NOT SURE HOW TO
> GET THE ROOMID INTO THIS VARIABLE
>
>
>
> Thanks ahead of time
>
>
>
> Miles
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Tuesday, July 22, 2014 6:23 AM
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> yes, this line need to be corrected
>
> openmeetings/rooms -> openmeetings/room
>
>
>
> guess this is the problem
>
>
>
> On 22 July 2014 19:43, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks Maxim,
>
> I have been trying to figure this out, I am knew to it all but on a steep
> learning curve.
>
>
>
> I do have a question about the asterisk extensions.conf
>
> exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
>
>
>
> Does the above line check the openmeetings database rooms table for the
> confno and returns ok if it finds it and notavail if it doesn’t?
>
>
>
> I am getting the following warning:
>
> Chan_sip.c.:25184 handle_request_infite: Call from ‘red5sip_user’ (
> 98.0.0.0:5070) to extension ‘40016’ rejected because extension not found
> in context “rooms-red5sip” I don’t recall seeing this error before. But
> if the “exten” line is checking the database openmeetings and looking for
> rooms table it does not exist. There is a table name room but no rooms.
>
>
>
> Am I reading this correctly?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Tuesday, July 22, 2014 4:27 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> AFAIK dial in/out conference room was working as expected
>
> Not sure if we still have infrastructure test current version
>
>
>
> Will try to ask someone
>
>
>
> On 21 July 2014 20:17, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok on the sip transport, I will try to figure out why it keep popping in
> and out.
>
> However, I am not understanding concerning the Asterisk config. The
> asterisk config I am using is from the install and modification as stated
> from the VOIP/SIP 3.0 integration package. The instructions don’t say a
> sip trunk or outside provider is required. However, I am unable to
> succesfull make calls from a conference room to a phone and visa versa.
> Which I thought I was suppose to be able to do after the integration. Per
> the below instructions I guess I am asking what am I missing from below?
>
> *Feature Matrix:*
>
> *Feature*
>
> *Description*
>
> 1) Dial-In
>
> A phone number is provided which you can give to anybody to "Dial-In" via
> usual landlane/phone into the conference room of OpenMeetings - Every room
> has its own phone number. Currently room gets number
> like 400<Id of room>. Maybe should move phone prefix to settings,
> currently it hardcoded.
>
> 2) Dial-Out
>
> The users in the conference room can call anybody outside of the
> conference room by entering the phone number in the conference room - In
> room actions menu exist "SIP dialer". When user clicked dialer window
> appears.
> Currently calls can't be dropped from Openmeetings, tbd
>
> 3) Multiple Dial-In
>
> You can give away multiple numbers and do the same as described in case
> (1). Multiple Dial-In is achieved by configuring the SIP-server (Asterisk).
> It is possible to create multiple extensions (phone numbers) in Asterisk
> configuration that will be redirects to single conference room.
>
> 4) Multiple Dial-Out
>
> You can dial multiple numbers from within the conference room - From
> within conference can be dialed multiple numbers.
>
> *Main difference to native Red5-Phone project*
>
>
>
>
>
>
>
> *rom:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Monday, July 21, 2014 5:36 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> If I do remember correctly
>
> SIP transport should enter the room and be in room as long as there other
> users in it.
>
>
>
> Possibility to call to phone numbers depends on your Asterisk config.
>
>
>
> I'll try to fix documentation ASAP
>
>
>
> On 21 July 2014 18:55, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks Maxim,
>
> Let me make sure I understand about the sip transport. It should not be
> popping in and out of the room? On my box I keep getting Sip Transport has
> exited the room.
>
>
>
> When properly configured, should I be able to call land and cell phones
> without a need for another server or;
>
> 1. Do I need to subscribe to a VOIP service provider
>
> 2. Configure Asterisk as a sip trunk to use google voice or some
> other solution?
>
> Also if you could have someone correct this line in the instructions of
> extensions.conf it will help to eliminate at least one error
>
>
>
>
> *[rooms-red5sip]exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil)
> <<<<<<<<<<<< should be “notavail”exten =>
> _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)exten =>
> _400X!,n(notavail),Hangup *
>
>
>
> *Thanks ahead of time*
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Monday, July 21, 2014 4:00 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Hello Horace,
>
>
>
> jsvc can be used to start java application as service
>
> I don't really like it (it was unstable when I used it)
>
> I prefer to write init.d script
>
>
>
> I see no errors in your log
>
> If everything is OK SIP transport should be in the room
>
> All 3 logs should be checked to have no errors
>
> I usually run asterisk in debug mode while setting everything up
>
>
>
>
>
>
>
> On 20 July 2014 23:55, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Additonally the VOIP and SIP integration 3.0 instructions do not mention
> installing jsvc. Is it still a requirement to install jsvc under 3.0 as it
> was under 2.0?:
>
> *apt-get install jsvc*
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 11:18 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> will try to take a look a look at it tomorrow, too late here ...
>
>
>
> On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok I will restart red5sip service and red5 and then send a new log
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 10:06 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> got the full trace in other email, will try to check code
>
>
>
> On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Did miss understand what you were asking for?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:56 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> would be more helpful to get full stack instead of "Red5sip log : Error
> o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
>
>
>
> On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Openmeetings log says confBridgeList authentication is failing. I will
> check to make sure I didn’t change a password there..
>
> Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException:
> Null
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:43 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> this "I have a sip transport that keeps popping in and out of the room."
> usually mean something configured wrong.
>
> Any exceptions in the logs (openmeetings.log and red5sip.log
>
>
>
> On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Sorry also Asterisk 11
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:10 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Additionally, what version are you using?
>
>
>
> On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Probably not, since I just went into a public room.. let me create a room..
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Do you have "*Enable SIP transport in the room*" checked for the room you
> are testing?
>
>
>
> On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the reply, I went back and rechecked my setup. I have
> completed all the steps according to the integration document.
>
> I found the following document on the wiki:
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
>
>
>
> According to this document I should get a sip dialer under the rooms
> actions menu. But I have no dialer there.
>
>
>
> The only error I see in the red5sip window is
>
> 18 Jul 07:50:11 . [nioProcessor-2]:[INFO ]
> o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to
> handle calls like onBWCheck, add a service provider. (it is my
> understanding this error is to be expect as it is not being used?)
>
>
>
> So where would I start to try and figure out why there is no sip dialer
> available?
>
>
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 7:15 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> http://openmeetings.apache.org/voip-sip-integration.html
>
>
>
> On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> It is nice that Openmeetings provided a way to integrate VOIP and Sip with
> Asterisk. That being said, I can find no documentation that tells the
> following:
>
> If the integration was successful?
>
> What icons should show up where etc.
>
> What actions can be taken by an admin or a user for that matter i.e. how
> to make a phone call out or in.
>
> Did I miss something somewhere?
>
>
>
> Miles
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: VOIP and Sip Integration
Posted by Horace Miles <Ho...@myit-solutions.com>.
Thanks I made the change prior to sending the email. There appears to be something else missing:
There appears to be a entry missing in the /etc/asterisk/func_odbc.conf: file for ${EXTEN}
I am probably wrong. But I can’t figure out how this is making the call to the database.
I don’t find any SQL statement in the /etc/asterisk/func_odbc.conf file and I am not sure how to construct one there that would work.
Would I simply add
[EXTEN]
dsn=asterisk
readsql=SELECT confno from room where confno = @EXTEN – NOT SURE HOW TO GET THE ROOMID INTO THIS VARIABLE
Thanks ahead of time
Miles
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, July 22, 2014 6:23 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
yes, this line need to be corrected
openmeetings/rooms -> openmeetings/room
guess this is the problem
On 22 July 2014 19:43, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Maxim,
I have been trying to figure this out, I am knew to it all but on a steep learning curve.
I do have a question about the asterisk extensions.conf
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
Does the above line check the openmeetings database rooms table for the confno and returns ok if it finds it and notavail if it doesn’t?
I am getting the following warning:
Chan_sip.c.:25184 handle_request_infite: Call from ‘red5sip_user’ (98.0.0.0:5070) to extension ‘40016’ rejected because extension not found in context “rooms-red5sip” I don’t recall seeing this error before. But if the “exten” line is checking the database openmeetings and looking for rooms table it does not exist. There is a table name room but no rooms.
Am I reading this correctly?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, July 22, 2014 4:27 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
AFAIK dial in/out conference room was working as expected
Not sure if we still have infrastructure test current version
Will try to ask someone
On 21 July 2014 20:17, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok on the sip transport, I will try to figure out why it keep popping in and out.
However, I am not understanding concerning the Asterisk config. The asterisk config I am using is from the install and modification as stated from the VOIP/SIP 3.0 integration package. The instructions don’t say a sip trunk or outside provider is required. However, I am unable to succesfull make calls from a conference room to a phone and visa versa. Which I thought I was suppose to be able to do after the integration. Per the below instructions I guess I am asking what am I missing from below?
Feature Matrix:
Feature
Description
1) Dial-In
A phone number is provided which you can give to anybody to "Dial-In" via usual landlane/phone into the conference room of OpenMeetings - Every room has its own phone number. Currently room gets number
like 400<Id of room>. Maybe should move phone prefix to settings,
currently it hardcoded.
2) Dial-Out
The users in the conference room can call anybody outside of the conference room by entering the phone number in the conference room - In room actions menu exist "SIP dialer". When user clicked dialer window appears.
Currently calls can't be dropped from Openmeetings, tbd
3) Multiple Dial-In
You can give away multiple numbers and do the same as described in case (1). Multiple Dial-In is achieved by configuring the SIP-server (Asterisk). It is possible to create multiple extensions (phone numbers) in Asterisk configuration that will be redirects to single conference room.
4) Multiple Dial-Out
You can dial multiple numbers from within the conference room - From within conference can be dialed multiple numbers.
Main difference to native Red5-Phone project
rom: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, July 21, 2014 5:36 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
If I do remember correctly
SIP transport should enter the room and be in room as long as there other users in it.
Possibility to call to phone numbers depends on your Asterisk config.
I'll try to fix documentation ASAP
On 21 July 2014 18:55, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Maxim,
Let me make sure I understand about the sip transport. It should not be popping in and out of the room? On my box I keep getting Sip Transport has exited the room.
When properly configured, should I be able to call land and cell phones without a need for another server or;
1. Do I need to subscribe to a VOIP service provider
2. Configure Asterisk as a sip trunk to use google voice or some other solution?
Also if you could have someone correct this line in the instructions of extensions.conf it will help to eliminate at least one error
[rooms-red5sip]
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil) <<<<<<<<<<<< should be “notavail”
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
exten => _400X!,n(notavail),Hangup
Thanks ahead of time
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, July 21, 2014 4:00 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Hello Horace,
jsvc can be used to start java application as service
I don't really like it (it was unstable when I used it)
I prefer to write init.d script
I see no errors in your log
If everything is OK SIP transport should be in the room
All 3 logs should be checked to have no errors
I usually run asterisk in debug mode while setting everything up
On 20 July 2014 23:55, Horace Miles <Ho...@myit-solutions.com> wrote:
Additonally the VOIP and SIP integration 3.0 instructions do not mention installing jsvc. Is it still a requirement to install jsvc under 3.0 as it was under 2.0?:
apt-get install jsvc
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 11:18 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
will try to take a look a look at it tomorrow, too late here ...
On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok I will restart red5sip service and red5 and then send a new log
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 10:06 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
got the full trace in other email, will try to check code
On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com> wrote:
Did miss understand what you were asking for?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:56 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
would be more helpful to get full stack instead of "Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com> wrote:
Openmeetings log says confBridgeList authentication is failing. I will check to make sure I didn’t change a password there..
Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:43 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
this "I have a sip transport that keeps popping in and out of the room." usually mean something configured wrong.
Any exceptions in the logs (openmeetings.log and red5sip.log
On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com> wrote:
Sorry also Asterisk 11
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com> wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:07 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Do you have "Enable SIP transport in the room" checked for the room you are testing?
On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the reply, I went back and rechecked my setup. I have completed all the steps according to the integration document.
I found the following document on the wiki: https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to handle calls like onBWCheck, add a service provider. (it is my understanding this error is to be expect as it is not being used?)
So where would I start to try and figure out why there is no sip dialer available?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with Asterisk. That being said, I can find no documentation that tells the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how to make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: VOIP and Sip Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
yes, this line need to be corrected
openmeetings/rooms -> openmeetings/room
guess this is the problem
On 22 July 2014 19:43, Horace Miles <Ho...@myit-solutions.com> wrote:
> Thanks Maxim,
>
> I have been trying to figure this out, I am knew to it all but on a steep
> learning curve.
>
>
>
> I do have a question about the asterisk extensions.conf
>
> exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
>
>
>
> Does the above line check the openmeetings database rooms table for the
> confno and returns ok if it finds it and notavail if it doesn’t?
>
>
>
> I am getting the following warning:
>
> Chan_sip.c.:25184 handle_request_infite: Call from ‘red5sip_user’ (
> 98.0.0.0:5070) to extension ‘40016’ rejected because extension not found
> in context “rooms-red5sip” I don’t recall seeing this error before. But
> if the “exten” line is checking the database openmeetings and looking for
> rooms table it does not exist. There is a table name room but no rooms.
>
>
>
> Am I reading this correctly?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Tuesday, July 22, 2014 4:27 AM
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> AFAIK dial in/out conference room was working as expected
>
> Not sure if we still have infrastructure test current version
>
>
>
> Will try to ask someone
>
>
>
> On 21 July 2014 20:17, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok on the sip transport, I will try to figure out why it keep popping in
> and out.
>
> However, I am not understanding concerning the Asterisk config. The
> asterisk config I am using is from the install and modification as stated
> from the VOIP/SIP 3.0 integration package. The instructions don’t say a
> sip trunk or outside provider is required. However, I am unable to
> succesfull make calls from a conference room to a phone and visa versa.
> Which I thought I was suppose to be able to do after the integration. Per
> the below instructions I guess I am asking what am I missing from below?
>
> *Feature Matrix:*
>
> *Feature*
>
> *Description*
>
> 1) Dial-In
>
> A phone number is provided which you can give to anybody to "Dial-In" via
> usual landlane/phone into the conference room of OpenMeetings - Every room
> has its own phone number. Currently room gets number
> like 400<Id of room>. Maybe should move phone prefix to settings,
> currently it hardcoded.
>
> 2) Dial-Out
>
> The users in the conference room can call anybody outside of the
> conference room by entering the phone number in the conference room - In
> room actions menu exist "SIP dialer". When user clicked dialer window
> appears.
> Currently calls can't be dropped from Openmeetings, tbd
>
> 3) Multiple Dial-In
>
> You can give away multiple numbers and do the same as described in case
> (1). Multiple Dial-In is achieved by configuring the SIP-server (Asterisk).
> It is possible to create multiple extensions (phone numbers) in Asterisk
> configuration that will be redirects to single conference room.
>
> 4) Multiple Dial-Out
>
> You can dial multiple numbers from within the conference room - From
> within conference can be dialed multiple numbers.
>
> *Main difference to native Red5-Phone project*
>
>
>
>
>
>
>
> *rom:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Monday, July 21, 2014 5:36 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> If I do remember correctly
>
> SIP transport should enter the room and be in room as long as there other
> users in it.
>
>
>
> Possibility to call to phone numbers depends on your Asterisk config.
>
>
>
> I'll try to fix documentation ASAP
>
>
>
> On 21 July 2014 18:55, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks Maxim,
>
> Let me make sure I understand about the sip transport. It should not be
> popping in and out of the room? On my box I keep getting Sip Transport has
> exited the room.
>
>
>
> When properly configured, should I be able to call land and cell phones
> without a need for another server or;
>
> 1. Do I need to subscribe to a VOIP service provider
>
> 2. Configure Asterisk as a sip trunk to use google voice or some
> other solution?
>
> Also if you could have someone correct this line in the instructions of
> extensions.conf it will help to eliminate at least one error
>
>
>
>
> *[rooms-red5sip]exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil)
> <<<<<<<<<<<< should be “notavail”exten =>
> _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)exten =>
> _400X!,n(notavail),Hangup *
>
>
>
> *Thanks ahead of time*
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Monday, July 21, 2014 4:00 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Hello Horace,
>
>
>
> jsvc can be used to start java application as service
>
> I don't really like it (it was unstable when I used it)
>
> I prefer to write init.d script
>
>
>
> I see no errors in your log
>
> If everything is OK SIP transport should be in the room
>
> All 3 logs should be checked to have no errors
>
> I usually run asterisk in debug mode while setting everything up
>
>
>
>
>
>
>
> On 20 July 2014 23:55, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Additonally the VOIP and SIP integration 3.0 instructions do not mention
> installing jsvc. Is it still a requirement to install jsvc under 3.0 as it
> was under 2.0?:
>
> *apt-get install jsvc*
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 11:18 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> will try to take a look a look at it tomorrow, too late here ...
>
>
>
> On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok I will restart red5sip service and red5 and then send a new log
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 10:06 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> got the full trace in other email, will try to check code
>
>
>
> On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Did miss understand what you were asking for?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:56 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> would be more helpful to get full stack instead of "Red5sip log : Error
> o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
>
>
>
> On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Openmeetings log says confBridgeList authentication is failing. I will
> check to make sure I didn’t change a password there..
>
> Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException:
> Null
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:43 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> this "I have a sip transport that keeps popping in and out of the room."
> usually mean something configured wrong.
>
> Any exceptions in the logs (openmeetings.log and red5sip.log
>
>
>
> On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Sorry also Asterisk 11
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:10 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Additionally, what version are you using?
>
>
>
> On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Probably not, since I just went into a public room.. let me create a room..
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Do you have "*Enable SIP transport in the room*" checked for the room you
> are testing?
>
>
>
> On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the reply, I went back and rechecked my setup. I have
> completed all the steps according to the integration document.
>
> I found the following document on the wiki:
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
>
>
>
> According to this document I should get a sip dialer under the rooms
> actions menu. But I have no dialer there.
>
>
>
> The only error I see in the red5sip window is
>
> 18 Jul 07:50:11 . [nioProcessor-2]:[INFO ]
> o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to
> handle calls like onBWCheck, add a service provider. (it is my
> understanding this error is to be expect as it is not being used?)
>
>
>
> So where would I start to try and figure out why there is no sip dialer
> available?
>
>
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 7:15 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> http://openmeetings.apache.org/voip-sip-integration.html
>
>
>
> On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> It is nice that Openmeetings provided a way to integrate VOIP and Sip with
> Asterisk. That being said, I can find no documentation that tells the
> following:
>
> If the integration was successful?
>
> What icons should show up where etc.
>
> What actions can be taken by an admin or a user for that matter i.e. how
> to make a phone call out or in.
>
> Did I miss something somewhere?
>
>
>
> Miles
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: VOIP and Sip Integration
Posted by Horace Miles <Ho...@myit-solutions.com>.
Thanks Maxim,
I have been trying to figure this out, I am knew to it all but on a steep learning curve.
I do have a question about the asterisk extensions.conf
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
Does the above line check the openmeetings database rooms table for the confno and returns ok if it finds it and notavail if it doesn’t?
I am getting the following warning:
Chan_sip.c.:25184 handle_request_infite: Call from ‘red5sip_user’ (98.0.0.0:5070) to extension ‘40016’ rejected because extension not found in context “rooms-red5sip” I don’t recall seeing this error before. But if the “exten” line is checking the database openmeetings and looking for rooms table it does not exist. There is a table name room but no rooms.
Am I reading this correctly?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, July 22, 2014 4:27 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
AFAIK dial in/out conference room was working as expected
Not sure if we still have infrastructure test current version
Will try to ask someone
On 21 July 2014 20:17, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok on the sip transport, I will try to figure out why it keep popping in and out.
However, I am not understanding concerning the Asterisk config. The asterisk config I am using is from the install and modification as stated from the VOIP/SIP 3.0 integration package. The instructions don’t say a sip trunk or outside provider is required. However, I am unable to succesfull make calls from a conference room to a phone and visa versa. Which I thought I was suppose to be able to do after the integration. Per the below instructions I guess I am asking what am I missing from below?
Feature Matrix:
Feature
Description
1) Dial-In
A phone number is provided which you can give to anybody to "Dial-In" via usual landlane/phone into the conference room of OpenMeetings - Every room has its own phone number. Currently room gets number
like 400<Id of room>. Maybe should move phone prefix to settings,
currently it hardcoded.
2) Dial-Out
The users in the conference room can call anybody outside of the conference room by entering the phone number in the conference room - In room actions menu exist "SIP dialer". When user clicked dialer window appears.
Currently calls can't be dropped from Openmeetings, tbd
3) Multiple Dial-In
You can give away multiple numbers and do the same as described in case (1). Multiple Dial-In is achieved by configuring the SIP-server (Asterisk). It is possible to create multiple extensions (phone numbers) in Asterisk configuration that will be redirects to single conference room.
4) Multiple Dial-Out
You can dial multiple numbers from within the conference room - From within conference can be dialed multiple numbers.
Main difference to native Red5-Phone project
rom: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, July 21, 2014 5:36 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
If I do remember correctly
SIP transport should enter the room and be in room as long as there other users in it.
Possibility to call to phone numbers depends on your Asterisk config.
I'll try to fix documentation ASAP
On 21 July 2014 18:55, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Maxim,
Let me make sure I understand about the sip transport. It should not be popping in and out of the room? On my box I keep getting Sip Transport has exited the room.
When properly configured, should I be able to call land and cell phones without a need for another server or;
1. Do I need to subscribe to a VOIP service provider
2. Configure Asterisk as a sip trunk to use google voice or some other solution?
Also if you could have someone correct this line in the instructions of extensions.conf it will help to eliminate at least one error
[rooms-red5sip]
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil) <<<<<<<<<<<< should be “notavail”
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
exten => _400X!,n(notavail),Hangup
Thanks ahead of time
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, July 21, 2014 4:00 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Hello Horace,
jsvc can be used to start java application as service
I don't really like it (it was unstable when I used it)
I prefer to write init.d script
I see no errors in your log
If everything is OK SIP transport should be in the room
All 3 logs should be checked to have no errors
I usually run asterisk in debug mode while setting everything up
On 20 July 2014 23:55, Horace Miles <Ho...@myit-solutions.com> wrote:
Additonally the VOIP and SIP integration 3.0 instructions do not mention installing jsvc. Is it still a requirement to install jsvc under 3.0 as it was under 2.0?:
apt-get install jsvc
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 11:18 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
will try to take a look a look at it tomorrow, too late here ...
On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok I will restart red5sip service and red5 and then send a new log
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 10:06 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
got the full trace in other email, will try to check code
On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com> wrote:
Did miss understand what you were asking for?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:56 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
would be more helpful to get full stack instead of "Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com> wrote:
Openmeetings log says confBridgeList authentication is failing. I will check to make sure I didn’t change a password there..
Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:43 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
this "I have a sip transport that keeps popping in and out of the room." usually mean something configured wrong.
Any exceptions in the logs (openmeetings.log and red5sip.log
On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com> wrote:
Sorry also Asterisk 11
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com> wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:07 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Do you have "Enable SIP transport in the room" checked for the room you are testing?
On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the reply, I went back and rechecked my setup. I have completed all the steps according to the integration document.
I found the following document on the wiki: https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to handle calls like onBWCheck, add a service provider. (it is my understanding this error is to be expect as it is not being used?)
So where would I start to try and figure out why there is no sip dialer available?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with Asterisk. That being said, I can find no documentation that tells the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how to make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: VOIP and Sip Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
AFAIK dial in/out conference room was working as expected
Not sure if we still have infrastructure test current version
Will try to ask someone
On 21 July 2014 20:17, Horace Miles <Ho...@myit-solutions.com> wrote:
> Ok on the sip transport, I will try to figure out why it keep popping in
> and out.
>
> However, I am not understanding concerning the Asterisk config. The
> asterisk config I am using is from the install and modification as stated
> from the VOIP/SIP 3.0 integration package. The instructions don’t say a
> sip trunk or outside provider is required. However, I am unable to
> succesfull make calls from a conference room to a phone and visa versa.
> Which I thought I was suppose to be able to do after the integration. Per
> the below instructions I guess I am asking what am I missing from below?
>
> *Feature Matrix:*
>
> *Feature*
>
> *Description*
>
> 1) Dial-In
>
> A phone number is provided which you can give to anybody to "Dial-In" via
> usual landlane/phone into the conference room of OpenMeetings - Every room
> has its own phone number. Currently room gets number
> like 400<Id of room>. Maybe should move phone prefix to settings,
> currently it hardcoded.
>
> 2) Dial-Out
>
> The users in the conference room can call anybody outside of the
> conference room by entering the phone number in the conference room - In
> room actions menu exist "SIP dialer". When user clicked dialer window
> appears.
> Currently calls can't be dropped from Openmeetings, tbd
>
> 3) Multiple Dial-In
>
> You can give away multiple numbers and do the same as described in case
> (1). Multiple Dial-In is achieved by configuring the SIP-server (Asterisk).
> It is possible to create multiple extensions (phone numbers) in Asterisk
> configuration that will be redirects to single conference room.
>
> 4) Multiple Dial-Out
>
> You can dial multiple numbers from within the conference room - From
> within conference can be dialed multiple numbers.
>
> *Main difference to native Red5-Phone project*
>
>
>
>
>
>
>
> *rom:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Monday, July 21, 2014 5:36 AM
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> If I do remember correctly
>
> SIP transport should enter the room and be in room as long as there other
> users in it.
>
>
>
> Possibility to call to phone numbers depends on your Asterisk config.
>
>
>
> I'll try to fix documentation ASAP
>
>
>
> On 21 July 2014 18:55, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks Maxim,
>
> Let me make sure I understand about the sip transport. It should not be
> popping in and out of the room? On my box I keep getting Sip Transport has
> exited the room.
>
>
>
> When properly configured, should I be able to call land and cell phones
> without a need for another server or;
>
> 1. Do I need to subscribe to a VOIP service provider
>
> 2. Configure Asterisk as a sip trunk to use google voice or some
> other solution?
>
> Also if you could have someone correct this line in the instructions of
> extensions.conf it will help to eliminate at least one error
>
>
>
>
> *[rooms-red5sip]exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil)
> <<<<<<<<<<<< should be “notavail”exten =>
> _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)exten =>
> _400X!,n(notavail),Hangup *
>
>
>
> *Thanks ahead of time*
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Monday, July 21, 2014 4:00 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Hello Horace,
>
>
>
> jsvc can be used to start java application as service
>
> I don't really like it (it was unstable when I used it)
>
> I prefer to write init.d script
>
>
>
> I see no errors in your log
>
> If everything is OK SIP transport should be in the room
>
> All 3 logs should be checked to have no errors
>
> I usually run asterisk in debug mode while setting everything up
>
>
>
>
>
>
>
> On 20 July 2014 23:55, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Additonally the VOIP and SIP integration 3.0 instructions do not mention
> installing jsvc. Is it still a requirement to install jsvc under 3.0 as it
> was under 2.0?:
>
> *apt-get install jsvc*
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 11:18 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> will try to take a look a look at it tomorrow, too late here ...
>
>
>
> On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok I will restart red5sip service and red5 and then send a new log
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 10:06 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> got the full trace in other email, will try to check code
>
>
>
> On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Did miss understand what you were asking for?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:56 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> would be more helpful to get full stack instead of "Red5sip log : Error
> o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
>
>
>
> On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Openmeetings log says confBridgeList authentication is failing. I will
> check to make sure I didn’t change a password there..
>
> Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException:
> Null
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:43 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> this "I have a sip transport that keeps popping in and out of the room."
> usually mean something configured wrong.
>
> Any exceptions in the logs (openmeetings.log and red5sip.log
>
>
>
> On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Sorry also Asterisk 11
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:10 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Additionally, what version are you using?
>
>
>
> On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Probably not, since I just went into a public room.. let me create a room..
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Do you have "*Enable SIP transport in the room*" checked for the room you
> are testing?
>
>
>
> On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the reply, I went back and rechecked my setup. I have
> completed all the steps according to the integration document.
>
> I found the following document on the wiki:
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
>
>
>
> According to this document I should get a sip dialer under the rooms
> actions menu. But I have no dialer there.
>
>
>
> The only error I see in the red5sip window is
>
> 18 Jul 07:50:11 . [nioProcessor-2]:[INFO ]
> o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to
> handle calls like onBWCheck, add a service provider. (it is my
> understanding this error is to be expect as it is not being used?)
>
>
>
> So where would I start to try and figure out why there is no sip dialer
> available?
>
>
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 7:15 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> http://openmeetings.apache.org/voip-sip-integration.html
>
>
>
> On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> It is nice that Openmeetings provided a way to integrate VOIP and Sip with
> Asterisk. That being said, I can find no documentation that tells the
> following:
>
> If the integration was successful?
>
> What icons should show up where etc.
>
> What actions can be taken by an admin or a user for that matter i.e. how
> to make a phone call out or in.
>
> Did I miss something somewhere?
>
>
>
> Miles
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: VOIP and Sip Integration
Posted by Horace Miles <Ho...@myit-solutions.com>.
Ok on the sip transport, I will try to figure out why it keep popping in and out.
However, I am not understanding concerning the Asterisk config. The asterisk config I am using is from the install and modification as stated from the VOIP/SIP 3.0 integration package. The instructions don’t say a sip trunk or outside provider is required. However, I am unable to succesfull make calls from a conference room to a phone and visa versa. Which I thought I was suppose to be able to do after the integration. Per the below instructions I guess I am asking what am I missing from below?
Feature Matrix:
Feature
Description
1) Dial-In
A phone number is provided which you can give to anybody to "Dial-In" via usual landlane/phone into the conference room of OpenMeetings - Every room has its own phone number. Currently room gets number
like 400<Id of room>. Maybe should move phone prefix to settings,
currently it hardcoded.
2) Dial-Out
The users in the conference room can call anybody outside of the conference room by entering the phone number in the conference room - In room actions menu exist "SIP dialer". When user clicked dialer window appears.
Currently calls can't be dropped from Openmeetings, tbd
3) Multiple Dial-In
You can give away multiple numbers and do the same as described in case (1). Multiple Dial-In is achieved by configuring the SIP-server (Asterisk). It is possible to create multiple extensions (phone numbers) in Asterisk configuration that will be redirects to single conference room.
4) Multiple Dial-Out
You can dial multiple numbers from within the conference room - From within conference can be dialed multiple numbers.
Main difference to native Red5-Phone project
rom: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, July 21, 2014 5:36 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
If I do remember correctly
SIP transport should enter the room and be in room as long as there other users in it.
Possibility to call to phone numbers depends on your Asterisk config.
I'll try to fix documentation ASAP
On 21 July 2014 18:55, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Maxim,
Let me make sure I understand about the sip transport. It should not be popping in and out of the room? On my box I keep getting Sip Transport has exited the room.
When properly configured, should I be able to call land and cell phones without a need for another server or;
1. Do I need to subscribe to a VOIP service provider
2. Configure Asterisk as a sip trunk to use google voice or some other solution?
Also if you could have someone correct this line in the instructions of extensions.conf it will help to eliminate at least one error
[rooms-red5sip]
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil) <<<<<<<<<<<< should be “notavail”
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
exten => _400X!,n(notavail),Hangup
Thanks ahead of time
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, July 21, 2014 4:00 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Hello Horace,
jsvc can be used to start java application as service
I don't really like it (it was unstable when I used it)
I prefer to write init.d script
I see no errors in your log
If everything is OK SIP transport should be in the room
All 3 logs should be checked to have no errors
I usually run asterisk in debug mode while setting everything up
On 20 July 2014 23:55, Horace Miles <Ho...@myit-solutions.com> wrote:
Additonally the VOIP and SIP integration 3.0 instructions do not mention installing jsvc. Is it still a requirement to install jsvc under 3.0 as it was under 2.0?:
apt-get install jsvc
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 11:18 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
will try to take a look a look at it tomorrow, too late here ...
On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok I will restart red5sip service and red5 and then send a new log
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 10:06 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
got the full trace in other email, will try to check code
On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com> wrote:
Did miss understand what you were asking for?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:56 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
would be more helpful to get full stack instead of "Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com> wrote:
Openmeetings log says confBridgeList authentication is failing. I will check to make sure I didn’t change a password there..
Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:43 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
this "I have a sip transport that keeps popping in and out of the room." usually mean something configured wrong.
Any exceptions in the logs (openmeetings.log and red5sip.log
On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com> wrote:
Sorry also Asterisk 11
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com> wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:07 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Do you have "Enable SIP transport in the room" checked for the room you are testing?
On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the reply, I went back and rechecked my setup. I have completed all the steps according to the integration document.
I found the following document on the wiki: https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to handle calls like onBWCheck, add a service provider. (it is my understanding this error is to be expect as it is not being used?)
So where would I start to try and figure out why there is no sip dialer available?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with Asterisk. That being said, I can find no documentation that tells the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how to make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: VOIP and Sip Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
If I do remember correctly
SIP transport should enter the room and be in room as long as there other
users in it.
Possibility to call to phone numbers depends on your Asterisk config.
I'll try to fix documentation ASAP
On 21 July 2014 18:55, Horace Miles <Ho...@myit-solutions.com> wrote:
> Thanks Maxim,
>
> Let me make sure I understand about the sip transport. It should not be
> popping in and out of the room? On my box I keep getting Sip Transport has
> exited the room.
>
>
>
> When properly configured, should I be able to call land and cell phones
> without a need for another server or;
>
> 1. Do I need to subscribe to a VOIP service provider
>
> 2. Configure Asterisk as a sip trunk to use google voice or some
> other solution?
>
> Also if you could have someone correct this line in the instructions of
> extensions.conf it will help to eliminate at least one error
>
>
>
>
> *[rooms-red5sip]exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil)
> <<<<<<<<<<<< should be “notavail”exten =>
> _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)exten =>
> _400X!,n(notavail),Hangup *
>
>
>
> *Thanks ahead of time*
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Monday, July 21, 2014 4:00 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Hello Horace,
>
>
>
> jsvc can be used to start java application as service
>
> I don't really like it (it was unstable when I used it)
>
> I prefer to write init.d script
>
>
>
> I see no errors in your log
>
> If everything is OK SIP transport should be in the room
>
> All 3 logs should be checked to have no errors
>
> I usually run asterisk in debug mode while setting everything up
>
>
>
>
>
>
>
> On 20 July 2014 23:55, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Additonally the VOIP and SIP integration 3.0 instructions do not mention
> installing jsvc. Is it still a requirement to install jsvc under 3.0 as it
> was under 2.0?:
>
> *apt-get install jsvc*
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 11:18 AM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> will try to take a look a look at it tomorrow, too late here ...
>
>
>
> On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok I will restart red5sip service and red5 and then send a new log
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 10:06 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> got the full trace in other email, will try to check code
>
>
>
> On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Did miss understand what you were asking for?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:56 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> would be more helpful to get full stack instead of "Red5sip log : Error
> o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
>
>
>
> On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Openmeetings log says confBridgeList authentication is failing. I will
> check to make sure I didn’t change a password there..
>
> Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException:
> Null
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:43 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> this "I have a sip transport that keeps popping in and out of the room."
> usually mean something configured wrong.
>
> Any exceptions in the logs (openmeetings.log and red5sip.log
>
>
>
> On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Sorry also Asterisk 11
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:10 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Additionally, what version are you using?
>
>
>
> On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Probably not, since I just went into a public room.. let me create a room..
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Do you have "*Enable SIP transport in the room*" checked for the room you
> are testing?
>
>
>
> On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the reply, I went back and rechecked my setup. I have
> completed all the steps according to the integration document.
>
> I found the following document on the wiki:
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
>
>
>
> According to this document I should get a sip dialer under the rooms
> actions menu. But I have no dialer there.
>
>
>
> The only error I see in the red5sip window is
>
> 18 Jul 07:50:11 . [nioProcessor-2]:[INFO ]
> o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to
> handle calls like onBWCheck, add a service provider. (it is my
> understanding this error is to be expect as it is not being used?)
>
>
>
> So where would I start to try and figure out why there is no sip dialer
> available?
>
>
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 7:15 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> http://openmeetings.apache.org/voip-sip-integration.html
>
>
>
> On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> It is nice that Openmeetings provided a way to integrate VOIP and Sip with
> Asterisk. That being said, I can find no documentation that tells the
> following:
>
> If the integration was successful?
>
> What icons should show up where etc.
>
> What actions can be taken by an admin or a user for that matter i.e. how
> to make a phone call out or in.
>
> Did I miss something somewhere?
>
>
>
> Miles
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: VOIP and Sip Integration
Posted by Horace Miles <Ho...@myit-solutions.com>.
Thanks Maxim,
Let me make sure I understand about the sip transport. It should not be popping in and out of the room? On my box I keep getting Sip Transport has exited the room.
When properly configured, should I be able to call land and cell phones without a need for another server or;
1. Do I need to subscribe to a VOIP service provider
2. Configure Asterisk as a sip trunk to use google voice or some other solution?
Also if you could have someone correct this line in the instructions of extensions.conf it will help to eliminate at least one error
[rooms-red5sip]
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil) <<<<<<<<<<<< should be “notavail”
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
exten => _400X!,n(notavail),Hangup
Thanks ahead of time
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, July 21, 2014 4:00 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Hello Horace,
jsvc can be used to start java application as service
I don't really like it (it was unstable when I used it)
I prefer to write init.d script
I see no errors in your log
If everything is OK SIP transport should be in the room
All 3 logs should be checked to have no errors
I usually run asterisk in debug mode while setting everything up
On 20 July 2014 23:55, Horace Miles <Ho...@myit-solutions.com> wrote:
Additonally the VOIP and SIP integration 3.0 instructions do not mention installing jsvc. Is it still a requirement to install jsvc under 3.0 as it was under 2.0?:
apt-get install jsvc
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 11:18 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
will try to take a look a look at it tomorrow, too late here ...
On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok I will restart red5sip service and red5 and then send a new log
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 10:06 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
got the full trace in other email, will try to check code
On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com> wrote:
Did miss understand what you were asking for?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:56 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
would be more helpful to get full stack instead of "Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com> wrote:
Openmeetings log says confBridgeList authentication is failing. I will check to make sure I didn’t change a password there..
Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:43 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
this "I have a sip transport that keeps popping in and out of the room." usually mean something configured wrong.
Any exceptions in the logs (openmeetings.log and red5sip.log
On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com> wrote:
Sorry also Asterisk 11
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com> wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:07 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Do you have "Enable SIP transport in the room" checked for the room you are testing?
On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the reply, I went back and rechecked my setup. I have completed all the steps according to the integration document.
I found the following document on the wiki: https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to handle calls like onBWCheck, add a service provider. (it is my understanding this error is to be expect as it is not being used?)
So where would I start to try and figure out why there is no sip dialer available?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with Asterisk. That being said, I can find no documentation that tells the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how to make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: VOIP and Sip Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
Hello Horace,
jsvc can be used to start java application as service
I don't really like it (it was unstable when I used it)
I prefer to write init.d script
I see no errors in your log
If everything is OK SIP transport should be in the room
All 3 logs should be checked to have no errors
I usually run asterisk in debug mode while setting everything up
On 20 July 2014 23:55, Horace Miles <Ho...@myit-solutions.com> wrote:
> Additonally the VOIP and SIP integration 3.0 instructions do not mention
> installing jsvc. Is it still a requirement to install jsvc under 3.0 as it
> was under 2.0?:
>
> *apt-get install jsvc*
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 11:18 AM
>
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> will try to take a look a look at it tomorrow, too late here ...
>
>
>
> On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok I will restart red5sip service and red5 and then send a new log
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 10:06 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> got the full trace in other email, will try to check code
>
>
>
> On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Did miss understand what you were asking for?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:56 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> would be more helpful to get full stack instead of "Red5sip log : Error
> o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
>
>
>
> On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Openmeetings log says confBridgeList authentication is failing. I will
> check to make sure I didn’t change a password there..
>
> Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException:
> Null
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:43 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> this "I have a sip transport that keeps popping in and out of the room."
> usually mean something configured wrong.
>
> Any exceptions in the logs (openmeetings.log and red5sip.log
>
>
>
> On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Sorry also Asterisk 11
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:10 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Additionally, what version are you using?
>
>
>
> On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Probably not, since I just went into a public room.. let me create a room..
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Do you have "*Enable SIP transport in the room*" checked for the room you
> are testing?
>
>
>
> On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the reply, I went back and rechecked my setup. I have
> completed all the steps according to the integration document.
>
> I found the following document on the wiki:
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
>
>
>
> According to this document I should get a sip dialer under the rooms
> actions menu. But I have no dialer there.
>
>
>
> The only error I see in the red5sip window is
>
> 18 Jul 07:50:11 . [nioProcessor-2]:[INFO ]
> o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to
> handle calls like onBWCheck, add a service provider. (it is my
> understanding this error is to be expect as it is not being used?)
>
>
>
> So where would I start to try and figure out why there is no sip dialer
> available?
>
>
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 7:15 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> http://openmeetings.apache.org/voip-sip-integration.html
>
>
>
> On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> It is nice that Openmeetings provided a way to integrate VOIP and Sip with
> Asterisk. That being said, I can find no documentation that tells the
> following:
>
> If the integration was successful?
>
> What icons should show up where etc.
>
> What actions can be taken by an admin or a user for that matter i.e. how
> to make a phone call out or in.
>
> Did I miss something somewhere?
>
>
>
> Miles
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: VOIP and Sip Integration
Posted by Horace Miles <Ho...@myit-solutions.com>.
Additonally the VOIP and SIP integration 3.0 instructions do not mention installing jsvc. Is it still a requirement to install jsvc under 3.0 as it was under 2.0?:
apt-get install jsvc
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 11:18 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
will try to take a look a look at it tomorrow, too late here ...
On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok I will restart red5sip service and red5 and then send a new log
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 10:06 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
got the full trace in other email, will try to check code
On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com> wrote:
Did miss understand what you were asking for?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:56 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
would be more helpful to get full stack instead of "Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com> wrote:
Openmeetings log says confBridgeList authentication is failing. I will check to make sure I didn’t change a password there..
Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:43 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
this "I have a sip transport that keeps popping in and out of the room." usually mean something configured wrong.
Any exceptions in the logs (openmeetings.log and red5sip.log
On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com> wrote:
Sorry also Asterisk 11
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com> wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:07 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Do you have "Enable SIP transport in the room" checked for the room you are testing?
On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the reply, I went back and rechecked my setup. I have completed all the steps according to the integration document.
I found the following document on the wiki: https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to handle calls like onBWCheck, add a service provider. (it is my understanding this error is to be expect as it is not being used?)
So where would I start to try and figure out why there is no sip dialer available?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with Asterisk. That being said, I can find no documentation that tells the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how to make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
RE: VOIP and Sip Integration
Posted by Horace Miles <Ho...@myit-solutions.com>.
I am no longer getting the sip transport popping in and out of the room.
1. Should I see any instance of the sip transport in the room
2. When using the sip dialer upon clicking call nothing happens.
a. I check the openmeetings log and I can see the following:
1. getSipNumber: room_ID: 13, sipNumber: 40013
2. Asterisk –rx “originate Local/6234123251@rooms-out extension 40013@rooms-originate”
Should I now be looking at the Asterisk logs to find out why the call is not being placed? If so where do I check?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 11:18 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
will try to take a look a look at it tomorrow, too late here ...
On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok I will restart red5sip service and red5 and then send a new log
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 10:06 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
got the full trace in other email, will try to check code
On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com> wrote:
Did miss understand what you were asking for?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:56 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
would be more helpful to get full stack instead of "Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com> wrote:
Openmeetings log says confBridgeList authentication is failing. I will check to make sure I didn’t change a password there..
Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:43 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
this "I have a sip transport that keeps popping in and out of the room." usually mean something configured wrong.
Any exceptions in the logs (openmeetings.log and red5sip.log
On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com> wrote:
Sorry also Asterisk 11
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com> wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:07 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Do you have "Enable SIP transport in the room" checked for the room you are testing?
On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the reply, I went back and rechecked my setup. I have completed all the steps according to the integration document.
I found the following document on the wiki: https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to handle calls like onBWCheck, add a service provider. (it is my understanding this error is to be expect as it is not being used?)
So where would I start to try and figure out why there is no sip dialer available?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with Asterisk. That being said, I can find no documentation that tells the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how to make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: VOIP and Sip Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
will try to take a look a look at it tomorrow, too late here ...
On 19 July 2014 00:59, Horace Miles <Ho...@myit-solutions.com> wrote:
> Ok I will restart red5sip service and red5 and then send a new log
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 10:06 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> got the full trace in other email, will try to check code
>
>
>
> On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Did miss understand what you were asking for?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:56 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> would be more helpful to get full stack instead of "Red5sip log : Error
> o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
>
>
>
> On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Openmeetings log says confBridgeList authentication is failing. I will
> check to make sure I didn’t change a password there..
>
> Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException:
> Null
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:43 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> this "I have a sip transport that keeps popping in and out of the room."
> usually mean something configured wrong.
>
> Any exceptions in the logs (openmeetings.log and red5sip.log
>
>
>
> On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Sorry also Asterisk 11
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:10 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Additionally, what version are you using?
>
>
>
> On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Probably not, since I just went into a public room.. let me create a room..
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Do you have "*Enable SIP transport in the room*" checked for the room you
> are testing?
>
>
>
> On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the reply, I went back and rechecked my setup. I have
> completed all the steps according to the integration document.
>
> I found the following document on the wiki:
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
>
>
>
> According to this document I should get a sip dialer under the rooms
> actions menu. But I have no dialer there.
>
>
>
> The only error I see in the red5sip window is
>
> 18 Jul 07:50:11 . [nioProcessor-2]:[INFO ]
> o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to
> handle calls like onBWCheck, add a service provider. (it is my
> understanding this error is to be expect as it is not being used?)
>
>
>
> So where would I start to try and figure out why there is no sip dialer
> available?
>
>
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 7:15 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> http://openmeetings.apache.org/voip-sip-integration.html
>
>
>
> On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> It is nice that Openmeetings provided a way to integrate VOIP and Sip with
> Asterisk. That being said, I can find no documentation that tells the
> following:
>
> If the integration was successful?
>
> What icons should show up where etc.
>
> What actions can be taken by an admin or a user for that matter i.e. how
> to make a phone call out or in.
>
> Did I miss something somewhere?
>
>
>
> Miles
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: VOIP and Sip Integration
Posted by Horace Miles <Ho...@myit-solutions.com>.
Ok I will restart red5sip service and red5 and then send a new log
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 10:06 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
got the full trace in other email, will try to check code
On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com> wrote:
Did miss understand what you were asking for?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:56 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
would be more helpful to get full stack instead of "Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com> wrote:
Openmeetings log says confBridgeList authentication is failing. I will check to make sure I didn’t change a password there..
Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:43 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
this "I have a sip transport that keeps popping in and out of the room." usually mean something configured wrong.
Any exceptions in the logs (openmeetings.log and red5sip.log
On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com> wrote:
Sorry also Asterisk 11
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com> wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:07 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Do you have "Enable SIP transport in the room" checked for the room you are testing?
On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the reply, I went back and rechecked my setup. I have completed all the steps according to the integration document.
I found the following document on the wiki: https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to handle calls like onBWCheck, add a service provider. (it is my understanding this error is to be expect as it is not being used?)
So where would I start to try and figure out why there is no sip dialer available?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with Asterisk. That being said, I can find no documentation that tells the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how to make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: VOIP and Sip Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
According to the code you should have more exceptions in red5sip log.
this NullPointer is not the initial one
On 19 July 2014 00:05, Maxim Solodovnik <so...@gmail.com> wrote:
> got the full trace in other email, will try to check code
>
>
> On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
>> Did miss understand what you were asking for?
>>
>>
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Friday, July 18, 2014 8:56 AM
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> would be more helpful to get full stack instead of "Red5sip log : Error
>> o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
>>
>>
>>
>> On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> Openmeetings log says confBridgeList authentication is failing. I will
>> check to make sure I didn’t change a password there..
>>
>> Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException:
>> Null
>>
>>
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Friday, July 18, 2014 8:43 AM
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> this "I have a sip transport that keeps popping in and out of the room."
>> usually mean something configured wrong.
>>
>> Any exceptions in the logs (openmeetings.log and red5sip.log
>>
>>
>>
>> On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> Sorry also Asterisk 11
>>
>>
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Friday, July 18, 2014 8:10 AM
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> Additionally, what version are you using?
>>
>>
>>
>> On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> Probably not, since I just went into a public room.. let me create a
>> room..
>>
>>
>>
>>
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Friday, July 18, 2014 8:07 AM
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> Do you have "*Enable SIP transport in the room*" checked for the room
>> you are testing?
>>
>>
>>
>> On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> Maxim thanks for the reply, I went back and rechecked my setup. I have
>> completed all the steps according to the integration document.
>>
>> I found the following document on the wiki:
>> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
>>
>>
>>
>> According to this document I should get a sip dialer under the rooms
>> actions menu. But I have no dialer there.
>>
>>
>>
>> The only error I see in the red5sip window is
>>
>> 18 Jul 07:50:11 . [nioProcessor-2]:[INFO ]
>> o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to
>> handle calls like onBWCheck, add a service provider. (it is my
>> understanding this error is to be expect as it is not being used?)
>>
>>
>>
>> So where would I start to try and figure out why there is no sip dialer
>> available?
>>
>>
>>
>>
>>
>>
>>
>> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
>> *Sent:* Friday, July 18, 2014 7:15 AM
>> *To:* Openmeetings user-list
>> *Subject:* Re: VOIP and Sip Integration
>>
>>
>>
>> http://openmeetings.apache.org/voip-sip-integration.html
>>
>>
>>
>> On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com>
>> wrote:
>>
>> It is nice that Openmeetings provided a way to integrate VOIP and Sip
>> with Asterisk. That being said, I can find no documentation that tells the
>> following:
>>
>> If the integration was successful?
>>
>> What icons should show up where etc.
>>
>> What actions can be taken by an admin or a user for that matter i.e. how
>> to make a phone call out or in.
>>
>> Did I miss something somewhere?
>>
>>
>>
>> Miles
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
Re: VOIP and Sip Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
got the full trace in other email, will try to check code
On 18 July 2014 23:22, Horace Miles <Ho...@myit-solutions.com> wrote:
> Did miss understand what you were asking for?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:56 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> would be more helpful to get full stack instead of "Red5sip log : Error
> o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
>
>
>
> On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Openmeetings log says confBridgeList authentication is failing. I will
> check to make sure I didn’t change a password there..
>
> Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException:
> Null
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:43 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> this "I have a sip transport that keeps popping in and out of the room."
> usually mean something configured wrong.
>
> Any exceptions in the logs (openmeetings.log and red5sip.log
>
>
>
> On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Sorry also Asterisk 11
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:10 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Additionally, what version are you using?
>
>
>
> On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Probably not, since I just went into a public room.. let me create a room..
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Do you have "*Enable SIP transport in the room*" checked for the room you
> are testing?
>
>
>
> On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the reply, I went back and rechecked my setup. I have
> completed all the steps according to the integration document.
>
> I found the following document on the wiki:
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
>
>
>
> According to this document I should get a sip dialer under the rooms
> actions menu. But I have no dialer there.
>
>
>
> The only error I see in the red5sip window is
>
> 18 Jul 07:50:11 . [nioProcessor-2]:[INFO ]
> o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to
> handle calls like onBWCheck, add a service provider. (it is my
> understanding this error is to be expect as it is not being used?)
>
>
>
> So where would I start to try and figure out why there is no sip dialer
> available?
>
>
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 7:15 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> http://openmeetings.apache.org/voip-sip-integration.html
>
>
>
> On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> It is nice that Openmeetings provided a way to integrate VOIP and Sip with
> Asterisk. That being said, I can find no documentation that tells the
> following:
>
> If the integration was successful?
>
> What icons should show up where etc.
>
> What actions can be taken by an admin or a user for that matter i.e. how
> to make a phone call out or in.
>
> Did I miss something somewhere?
>
>
>
> Miles
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: VOIP and Sip Integration
Posted by Horace Miles <Ho...@myit-solutions.com>.
Did miss understand what you were asking for?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:56 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
would be more helpful to get full stack instead of "Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com> wrote:
Openmeetings log says confBridgeList authentication is failing. I will check to make sure I didn’t change a password there..
Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:43 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
this "I have a sip transport that keeps popping in and out of the room." usually mean something configured wrong.
Any exceptions in the logs (openmeetings.log and red5sip.log
On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com> wrote:
Sorry also Asterisk 11
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com> wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:07 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Do you have "Enable SIP transport in the room" checked for the room you are testing?
On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the reply, I went back and rechecked my setup. I have completed all the steps according to the integration document.
I found the following document on the wiki: https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to handle calls like onBWCheck, add a service provider. (it is my understanding this error is to be expect as it is not being used?)
So where would I start to try and figure out why there is no sip dialer available?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with Asterisk. That being said, I can find no documentation that tells the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how to make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: VOIP and Sip Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
would be more helpful to get full stack instead of "Red5sip log : Error
o.z.s.p.SipProvider: java.lang.NullPointerException: Null"
On 18 July 2014 22:37, Horace Miles <Ho...@myit-solutions.com> wrote:
> Openmeetings log says confBridgeList authentication is failing. I will
> check to make sure I didn’t change a password there..
>
> Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException:
> Null
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:43 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> this "I have a sip transport that keeps popping in and out of the room."
> usually mean something configured wrong.
>
> Any exceptions in the logs (openmeetings.log and red5sip.log
>
>
>
> On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Sorry also Asterisk 11
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:10 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Additionally, what version are you using?
>
>
>
> On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Probably not, since I just went into a public room.. let me create a room..
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Do you have "*Enable SIP transport in the room*" checked for the room you
> are testing?
>
>
>
> On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the reply, I went back and rechecked my setup. I have
> completed all the steps according to the integration document.
>
> I found the following document on the wiki:
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
>
>
>
> According to this document I should get a sip dialer under the rooms
> actions menu. But I have no dialer there.
>
>
>
> The only error I see in the red5sip window is
>
> 18 Jul 07:50:11 . [nioProcessor-2]:[INFO ]
> o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to
> handle calls like onBWCheck, add a service provider. (it is my
> understanding this error is to be expect as it is not being used?)
>
>
>
> So where would I start to try and figure out why there is no sip dialer
> available?
>
>
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 7:15 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> http://openmeetings.apache.org/voip-sip-integration.html
>
>
>
> On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> It is nice that Openmeetings provided a way to integrate VOIP and Sip with
> Asterisk. That being said, I can find no documentation that tells the
> following:
>
> If the integration was successful?
>
> What icons should show up where etc.
>
> What actions can be taken by an admin or a user for that matter i.e. how
> to make a phone call out or in.
>
> Did I miss something somewhere?
>
>
>
> Miles
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: VOIP and Sip Integration
Posted by Horace Miles <Ho...@myit-solutions.com>.
Openmeetings log says confBridgeList authentication is failing. I will check to make sure I didn’t change a password there..
Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: Null
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:43 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
this "I have a sip transport that keeps popping in and out of the room." usually mean something configured wrong.
Any exceptions in the logs (openmeetings.log and red5sip.log
On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com> wrote:
Sorry also Asterisk 11
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com> wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:07 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Do you have "Enable SIP transport in the room" checked for the room you are testing?
On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the reply, I went back and rechecked my setup. I have completed all the steps according to the integration document.
I found the following document on the wiki: https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to handle calls like onBWCheck, add a service provider. (it is my understanding this error is to be expect as it is not being used?)
So where would I start to try and figure out why there is no sip dialer available?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with Asterisk. That being said, I can find no documentation that tells the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how to make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: VOIP and Sip Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
this "I have a sip transport that keeps popping in and out of the room."
usually mean something configured wrong.
Any exceptions in the logs (openmeetings.log and red5sip.log
On 18 July 2014 22:15, Horace Miles <Ho...@myit-solutions.com> wrote:
> Sorry also Asterisk 11
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:10 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Additionally, what version are you using?
>
>
>
> On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Probably not, since I just went into a public room.. let me create a room..
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Do you have "*Enable SIP transport in the room*" checked for the room you
> are testing?
>
>
>
> On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the reply, I went back and rechecked my setup. I have
> completed all the steps according to the integration document.
>
> I found the following document on the wiki:
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
>
>
>
> According to this document I should get a sip dialer under the rooms
> actions menu. But I have no dialer there.
>
>
>
> The only error I see in the red5sip window is
>
> 18 Jul 07:50:11 . [nioProcessor-2]:[INFO ]
> o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to
> handle calls like onBWCheck, add a service provider. (it is my
> understanding this error is to be expect as it is not being used?)
>
>
>
> So where would I start to try and figure out why there is no sip dialer
> available?
>
>
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 7:15 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> http://openmeetings.apache.org/voip-sip-integration.html
>
>
>
> On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> It is nice that Openmeetings provided a way to integrate VOIP and Sip with
> Asterisk. That being said, I can find no documentation that tells the
> following:
>
> If the integration was successful?
>
> What icons should show up where etc.
>
> What actions can be taken by an admin or a user for that matter i.e. how
> to make a phone call out or in.
>
> Did I miss something somewhere?
>
>
>
> Miles
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: VOIP and Sip Integration
Posted by Horace Miles <Ho...@myit-solutions.com>.
Sorry also Asterisk 11
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:10 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com> wrote:
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:07 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Do you have "Enable SIP transport in the room" checked for the room you are testing?
On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the reply, I went back and rechecked my setup. I have completed all the steps according to the integration document.
I found the following document on the wiki: https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to handle calls like onBWCheck, add a service provider. (it is my understanding this error is to be expect as it is not being used?)
So where would I start to try and figure out why there is no sip dialer available?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with Asterisk. That being said, I can find no documentation that tells the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how to make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: VOIP and Sip Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
Additionally, what version are you using?
On 18 July 2014 21:52, Horace Miles <Ho...@myit-solutions.com> wrote:
> Probably not, since I just went into a public room.. let me create a room..
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 8:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> Do you have "*Enable SIP transport in the room*" checked for the room you
> are testing?
>
>
>
> On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the reply, I went back and rechecked my setup. I have
> completed all the steps according to the integration document.
>
> I found the following document on the wiki:
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
>
>
>
> According to this document I should get a sip dialer under the rooms
> actions menu. But I have no dialer there.
>
>
>
> The only error I see in the red5sip window is
>
> 18 Jul 07:50:11 . [nioProcessor-2]:[INFO ]
> o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to
> handle calls like onBWCheck, add a service provider. (it is my
> understanding this error is to be expect as it is not being used?)
>
>
>
> So where would I start to try and figure out why there is no sip dialer
> available?
>
>
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 7:15 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> http://openmeetings.apache.org/voip-sip-integration.html
>
>
>
> On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> It is nice that Openmeetings provided a way to integrate VOIP and Sip with
> Asterisk. That being said, I can find no documentation that tells the
> following:
>
> If the integration was successful?
>
> What icons should show up where etc.
>
> What actions can be taken by an admin or a user for that matter i.e. how
> to make a phone call out or in.
>
> Did I miss something somewhere?
>
>
>
> Miles
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: VOIP and Sip Integration
Posted by Horace Miles <Ho...@myit-solutions.com>.
Probably not, since I just went into a public room.. let me create a room..
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 8:07 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
Do you have "Enable SIP transport in the room" checked for the room you are testing?
On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the reply, I went back and rechecked my setup. I have completed all the steps according to the integration document.
I found the following document on the wiki: https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to handle calls like onBWCheck, add a service provider. (it is my understanding this error is to be expect as it is not being used?)
So where would I start to try and figure out why there is no sip dialer available?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with Asterisk. That being said, I can find no documentation that tells the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how to make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: VOIP and Sip Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
Do you have "Enable SIP transport in the room" checked for the room you are
testing?
On 18 July 2014 21:48, Horace Miles <Ho...@myit-solutions.com> wrote:
> Maxim thanks for the reply, I went back and rechecked my setup. I have
> completed all the steps according to the integration document.
>
> I found the following document on the wiki:
> https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
>
>
>
> According to this document I should get a sip dialer under the rooms
> actions menu. But I have no dialer there.
>
>
>
> The only error I see in the red5sip window is
>
> 18 Jul 07:50:11 . [nioProcessor-2]:[INFO ]
> o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to
> handle calls like onBWCheck, add a service provider. (it is my
> understanding this error is to be expect as it is not being used?)
>
>
>
> So where would I start to try and figure out why there is no sip dialer
> available?
>
>
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, July 18, 2014 7:15 AM
> *To:* Openmeetings user-list
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> http://openmeetings.apache.org/voip-sip-integration.html
>
>
>
> On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> It is nice that Openmeetings provided a way to integrate VOIP and Sip with
> Asterisk. That being said, I can find no documentation that tells the
> following:
>
> If the integration was successful?
>
> What icons should show up where etc.
>
> What actions can be taken by an admin or a user for that matter i.e. how
> to make a phone call out or in.
>
> Did I miss something somewhere?
>
>
>
> Miles
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: VOIP and Sip Integration
Posted by Horace Miles <Ho...@myit-solutions.com>.
Maxim thanks for the reply, I went back and rechecked my setup. I have completed all the steps according to the integration document.
I found the following document on the wiki: https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description
According to this document I should get a sip dialer under the rooms actions menu. But I have no dialer there.
The only error I see in the red5sip window is
18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to handle calls like onBWCheck, add a service provider. (it is my understanding this error is to be expect as it is not being used?)
So where would I start to try and figure out why there is no sip dialer available?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, July 18, 2014 7:15 AM
To: Openmeetings user-list
Subject: Re: VOIP and Sip Integration
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
It is nice that Openmeetings provided a way to integrate VOIP and Sip with Asterisk. That being said, I can find no documentation that tells the following:
If the integration was successful?
What icons should show up where etc.
What actions can be taken by an admin or a user for that matter i.e. how to make a phone call out or in.
Did I miss something somewhere?
Miles
--
WBR
Maxim aka solomax
Re: VOIP and Sip Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
http://openmeetings.apache.org/voip-sip-integration.html
On 18 July 2014 20:52, Horace Miles <Ho...@myit-solutions.com> wrote:
> It is nice that Openmeetings provided a way to integrate VOIP and Sip with
> Asterisk. That being said, I can find no documentation that tells the
> following:
>
> If the integration was successful?
>
> What icons should show up where etc.
>
> What actions can be taken by an admin or a user for that matter i.e. how
> to make a phone call out or in.
>
> Did I miss something somewhere?
>
>
>
> Miles
>
--
WBR
Maxim aka solomax