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Posted to user@openmeetings.apache.org by Raju M K <mk...@gmail.com> on 2014/07/24 11:40:12 UTC
Pointer on WB
Dear all,
can i disable arrow pointer for all participants in restricted room on
Whiteboard??
--
Regards,
M K Raju.
RE: Pointer on WB
Posted by Horace Miles <Ho...@myit-solutions.com>.
I am still trying to integrate RED5SIP and VOIP into Openmeetings 3.0 The connection is be Declined as not authorized but I can not figure out why. Here are the relative log and debug files. Hopefully someone can help me figure this out.
Thanks Miles
Asterisk messages log
Aug 5 06:08:51] Asterisk 11.11.0 built by root @ vms on a i686 running Linux on 2014-07-26 19:25:45 UTC
[Aug 5 06:08:51] NOTICE[5128] loader.c: 2 modules will be loaded.
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: Connecting asterisk
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: res_odbc: Connected to asterisk [asterisk-connector]
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: Registered ODBC class 'asterisk' dsn->[asterisk-connector]
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: Connecting mysql2
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: res_odbc: Connected to mysql2 [asterisk-connector]
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: Registered ODBC class 'mysql2' dsn->[asterisk-connector]
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: res_odbc loaded.
[Aug 5 06:08:51] NOTICE[5128] config.c: Registered Config Engine odbc
[Aug 5 06:08:51] NOTICE[5128] cdr.c: CDR simple logging enabled.
[Aug 5 06:08:51] NOTICE[5128] loader.c: 201 modules will be loaded.
[Aug 5 06:08:51] NOTICE[5128] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener.
[Aug 5 06:08:51] NOTICE[5128] config.c: Registered Config Engine sqlite3
[Aug 5 06:08:52] NOTICE[5128] chan_skinny.c: Configuring skinny from skinny.conf
[Aug 5 06:08:52] WARNING[5128] chan_dahdi.c: Ignoring any changes to 'userbase' (on reload) at line 23.
[Aug 5 06:08:52] WARNING[5128] chan_dahdi.c: Ignoring any changes to 'vmsecret' (on reload) at line 31.
[Aug 5 06:08:52] WARNING[5128] chan_dahdi.c: Ignoring any changes to 'hassip' (on reload) at line 35.
[Aug 5 06:08:52] WARNING[5128] chan_dahdi.c: Ignoring any changes to 'hasiax' (on reload) at line 39.
[Aug 5 06:08:52] WARNING[5128] chan_dahdi.c: Ignoring any changes to 'hasmanager' (on reload) at line 47.
[Aug 5 06:08:52] NOTICE[5128] cel_custom.c: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Aug 5 06:08:52] WARNING[5128] pbx.c: Extension '_400X!' priority 5 in 'rooms', label 'ok' already in use at priority 2
[Aug 5 06:08:52] NOTICE[5128] pbx_ael.c: Starting AEL load process.
[Aug 5 06:08:52] NOTICE[5128] pbx_ael.c: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'.
[Aug 5 06:08:52] NOTICE[5128] pbx_ael.c: AEL load process: checked config file name '/etc/asterisk/extensions.ael'.
[Aug 5 06:08:52] NOTICE[5128] pbx_ael.c: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'.
[Aug 5 06:08:52] NOTICE[5128] pbx_ael.c: AEL load process: merged config file name '/etc/asterisk/extensions.ael'.
[Aug 5 06:08:52] NOTICE[5128] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'.
/var/log/asterisk# netstat -tlvn
Active Internet connections (only servers)
Proto Recv-Q Send-Q Local Address Foreign Address State
tcp 0 0 127.0.0.1:3306 0.0.0.0:* LISTEN
tcp 0 0 0.0.0.0:1935 0.0.0.0:* LISTEN
tcp 0 0 0.0.0.0:9999 0.0.0.0:* LISTEN
tcp 0 0 0.0.0.0:2000 0.0.0.0:* LISTEN
tcp 0 0 127.0.0.1:53 0.0.0.0:* LISTEN
tcp 0 0 127.0.0.1:631 0.0.0.0:* LISTEN
tcp 0 0 0.0.0.0:1720 0.0.0.0:* LISTEN
tcp 0 0 0.0.0.0:5080 0.0.0.0:* LISTEN
tcp 0 0 0.0.0.0:25 0.0.0.0:* LISTEN
tcp 0 0 0.0.0.0:39806 0.0.0.0:* LISTEN
tcp 0 0 0.0.0.0:5060 0.0.0.0:* LISTEN
tcp6 0 0 :::80 :::* LISTEN
tcp6 0 0 ::1:631 :::* LISTEN
tcp6 0 0 :::25 :::* LISTEN
asterisk -rvvvvvv
Connected to Asterisk 11.11.0 currently running on vms (pid = 5128)
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [40016@rooms-red5sip:1] GotoIf("SIP/red5sip_user-00000005", "0?ok:notavail") in new stack
-- Goto (rooms-red5sip,40016,3)
-- Executing [40016@rooms-red5sip:3] Hangup("SIP/red5sip_user-00000005", "") in new stack
== Spawn extension (rooms-red5sip, 40016, 3) exited non-zero on 'SIP/red5sip_user-00000005'
[Aug 5 06:14:19] WARNING[5164]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission 024312648651@127.0.1.1 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
sip debug logs
CLI> sip set debug on
SIP Debugging enabled
<--- SIP read from UDP:127.0.0.1:5070 --->
<------------->
Retransmitting #7 (no NAT) to 127.0.0.1:5070:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 127.0.1.1:5070;branch=z9hG4bK5224484;received=127.0.0.1;rport=5070
From: "red5sip_user" <si...@127.0.0.1>;tag=z9hG4bK74877027
To: <si...@127.0.0.1>;tag=as759e6d0c
Call-ID: 397099427934@127.0.1.1
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:127.0.0.1:5070 --->
ACK sip:40016@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK0858799
Max-Forwards: 70
To: <si...@127.0.0.1>
From: "red5sip_user" <si...@127.0.0.1>;tag=z9hG4bK29134357
Call-ID: 263147788729@127.0.1.1
CSeq: 1 ACK
Contact: <sip:red5sip_user@127.0.1.1:5070>
Expires: 3600
User-Agent: mjsip stack 1.6
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from UDP:127.0.0.1:5070 --->
INVITE sip:40016@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK19482100
Max-Forwards: 70
To: <si...@127.0.0.1>
From: "red5sip_user" <si...@127.0.0.1>;tag=z9hG4bK29134357
Call-ID: 263147788729@127.0.1.1
CSeq: 2 INVITE
Contact: <sip:red5sip_user@127.0.1.1:5070>
Expires: 3600
User-Agent: mjsip stack 1.6
Authorization: Digest username="red5sip_user", realm="asterisk", nonce="308fba53", uri="sip:40016@127.0.0.1", algorithm=MD5, response="9a2776ea6883adb1345d50eb1fed5d45"
Content-Length: 324
Content-Type: application/sdp
v=0
o=red5sip_user 0 0 IN IP4 127.0.1.1
s=Session SIP/SDP
c=IN IP4 127.0.1.1
t=0 0
m=audio 3010 RTP/AVP 8 18 0 111
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000/1
a=rtpmap:111 ILBC/8000/1
a=fmtp:111 mode=30
a=ptime:20
m=video 7010 RTP/AVP 35
a=rtpmap:35 H264/90000/1
<------------->
--- (13 headers 15 lines) ---
Sending to 127.0.0.1:5070 (no NAT)
Using INVITE request as basis request - 263147788729@127.0.1.1
Found peer 'red5sip_user' for 'red5sip_user' from 127.0.0.1:5070
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 111
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format ILBC for ID 111
Found RTP video format 35
Found video description format H264 for ID 35
Capabilities: us - (ulaw|h264), peer - audio=(ulaw|alaw|g729|ilbc)/video=(h264)/text=(nothing), combined - (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 127.0.1.1:3010
Peer video RTP is at port 127.0.1.1:7010
Looking for 40016 in rooms-red5sip (domain 127.0.0.1)
list_route: hop: <sip:red5sip_user@127.0.1.1:5070>
<--- Transmitting (no NAT) to 127.0.0.1:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.1.1:5070;branch=z9hG4bK19482100;received=127.0.0.1;rport=5070
From: "red5sip_user" <si...@127.0.0.1>;tag=z9hG4bK29134357
To: <si...@127.0.0.1>
Call-ID: 263147788729@127.0.1.1
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:40016@127.0.0.1:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:127.0.0.1:5070 --->
ACK sip:40016@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK9801398
Max-Forwards: 70
To: <si...@127.0.0.1>;tag=as6bc959f2
From: "red5sip_user" <si...@127.0.0.1>;tag=z9hG4bK29134357
Call-ID: 263147788729@127.0.1.1
CSeq: 1 ACK
User-Agent: mjsip stack 1.6
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '263147788729@127.0.1.1' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (no NAT) to 127.0.0.1:5070 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 127.0.1.1:5070;branch=z9hG4bK19482100;received=127.0.0.1;rport=5070
From: "red5sip_user" <si...@127.0.0.1>;tag=z9hG4bK29134357
To: <si...@127.0.0.1>;tag=as69b73555
Call-ID: 263147788729@127.0.1.1
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
<--- SIP read from UDP:127.0.0.1:5070 --->
ACK sip:40016@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK9801398
Max-Forwards: 70
To: <si...@127.0.0.1>;tag=as6bc959f2
From: "red5sip_user" <si...@127.0.0.1>;tag=z9hG4bK29134357
Call-ID: 263147788729@127.0.1.1
CSeq: 1 ACK
User-Agent: mjsip stack 1.6
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:127.0.0.1:5070 --->
ACK sip:40016@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK9801398
Max-Forwards: 70
To: <si...@127.0.0.1>;tag=as6bc959f2
From: "red5sip_user" <si...@127.0.0.1>;tag=z9hG4bK29134357
Call-ID: 263147788729@127.0.1.1
CSeq: 1 ACK
User-Agent: mjsip stack 1.6
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:127.0.0.1:5070 --->
ACK sip:40016@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK9801398
Max-Forwards: 70
To: <si...@127.0.0.1>;tag=as6bc959f2
From: "red5sip_user" <si...@127.0.0.1>;tag=z9hG4bK29134357
Call-ID: 263147788729@127.0.1.1
CSeq: 1 ACK
User-Agent: mjsip stack 1.6
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:127.0.0.1:5070 --->
ACK sip:40016@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK47244101
Max-Forwards: 70
To: <si...@127.0.0.1>
From: "red5sip_user" <si...@127.0.0.1>;tag=z9hG4bK29134357
Call-ID: 263147788729@127.0.1.1
CSeq: 1 ACK
Contact: <sip:red5sip_user@127.0.1.1:5070>
Expires: 3600
User-Agent: mjsip stack 1.6
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from UDP:127.0.0.1:5070 --->
INVITE sip:40016@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5070;rport;branch=z9hG4bK45513102
Max-Forwards: 70
To: <si...@127.0.0.1>
From: "red5sip_user" <si...@127.0.0.1>;tag=z9hG4bK16124726
Call-ID: 725338205557@127.0.1.1
CSeq: 1 INVITE
Contact: <sip:red5sip_user@127.0.1.1:5070>
Expires: 3600
User-Agent: mjsip stack 1.6
Content-Length: 324
Content-Type: application/sdp
v=0
o=red5sip_user 0 0 IN IP4 127.0.1.1
s=Session SIP/SDP
c=IN IP4 127.0.1.1
t=0 0
m=audio 3010 RTP/AVP 8 18 0 111
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000/1
a=rtpmap:111 ILBC/8000/1
a=fmtp:111 mode=30
a=ptime:20
m=video 7010 RTP/AVP 35
a=rtpmap:35 H264/90000/1
<------------->
--- (12 headers 15 lines) ---
Sending to 127.0.0.1:5070 (NAT)
Sending to 127.0.0.1:5070 (NAT)
Using INVITE request as basis request - 725338205557@127.0.1.1
Found peer 'red5sip_user' for 'red5sip_user' from 127.0.0.1:5070
<--- Reliably Transmitting (no NAT) to 127.0.0.1:5070 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 127.0.1.1:5070;branch=z9hG4bK45513102;received=127.0.0.1;rport=5070
From: "red5sip_user" <si...@127.0.0.1>;tag=z9hG4bK16124726
To: <si...@127.0.0.1>;tag=as65c60e65
Call-ID: 725338205557@127.0.1.1
CSeq: 1 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c80ccd0"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '725338205557@127.0.1.1' in 32000 ms (Method: INVITE)
Retransmitting #6 (no NAT) to 127.0.0.1:5070:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 127.0.1.1:5070;branch=z9hG4bK4721088;received=127.0.0.1;rport=5070
From: "red5sip_user" <si...@127.0.0.1>;tag=z9hG4bK79605539
To: <si...@127.0.0.1>;tag=as4a2fdbcb
Call-ID: 400642563986@127.0.1.1
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
Retransmitting #1 (no NAT) to 127.0.0.1:5070:
SIP/2.0 603 Declined
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, August 01, 2014 10:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
you can search red5sip in config :)
the key is "red5sip.enable"
On 1 August 2014 23:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the response.
I have confirmed everything but I am not sure where to find this setting. I am assuming Admin config is Openmeeting Admin->Configuration. If so I don’t a setting for Red5sip key.
3) red5sip* key should be enabled in Admin->Config – NOT SURE OF THIS STEP
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Wednesday, July 30, 2014 6:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
OM is accessible on all network interfaces by default
config.xml need to be modified only in case you need to restrict OM client.
According to red5sip enter-exit-enter-exit-.... it should be due to misconfiguration. Unfortunately this integration is not simple by design :( I'm using logs and debug to set it up properly.
Main steps are
1) asterisk should be configured to have access to OM DB
2) asterisk bean should be uncommented and configured properly in openmeetings-application.xml
3) red5sip* key should be enabled in Admin->Config
4) in case asterisk is integrated with OM user should be re-saved (to have password-hash being saved in asterisk DB table)
5) sip should be enabled in the room
this should be all (hope I haven't miss anything)
On 29 July 2014 08:29, Horace Miles <Ho...@myit-solutions.com> wrote:
Hi Maxim,
My box is connected directly to a public IP, no NAT. My understanding was that Openmeetings to be access from the internet needed to be on a public address. That address would be the one in the config.xml. If I a mistaken let me know.
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 2:42 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Only with code modification
On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
Dear all,
can i disable arrow pointer for all participants in restricted room on Whiteboard??
--
Regards,
M K Raju.
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
RE: Pointer on WB
Posted by Horace Miles <Ho...@myit-solutions.com>.
Thank Maxim, I thought I had cleared up all credential problems, I am probably wrong. I just don’t know where to go from here.. I appreciate you. I see you are busy and don’t mean to take up a lot of your time.
Miles
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Wednesday, August 06, 2014 10:46 PM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Hello Horace,
sorry for keeping silence, a little bit bit busy right now
SIP transport set up the bridge from asterisk to red5 and performs audio/video transcoding rtp <->rtmp
according to your issue it seems like creadentials specified in settings file are invalid for your Asterisk, can it be a problem?
Will try to reproduce your problem as soon as i will get some time
On 7 August 2014 02:53, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim,
Perhaps if I knew exactly what sip transport does, I might be able to figure this out. Can you tell me what it is suppose to do..
Miles
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, August 01, 2014 8:22 PM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Simple test if everything works is:
1) go to Admin->Conference rooms
2) select room
3) Check enable SIP
4) SIP number should appear in room panel (maybe after save)
is it works for you?
On 2 August 2014 00:36, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok found red5sip.enable value = yes
Asterisk is configured to access openmeeting database through asterisk-connector
Bean as been uncommented in openmeetings-application.xml and configure with matching values in asterisk/manager.conf
I have re-saved all users in Openmeetings to recreate password hashes in asterisk
Sip is enabled in rooms that have been created.
I can telnet to localhost 5080 and 1935
I am still having the following problems
Sip Transport will not stay in the room pops in and out every two seconds
It appears as though the sip transport can register but is unable to receive the invite message.
In the extension.conf
I get the following
n -- Executing [40016@rooms-red5sip:1] GotoIf(“SIP/red5sip_user-000000a6”,”0?ok:notavail”) in new stack
n -- Goto (rooms-red5sip,40016,3)
n --Executing [40016@rooms-red5sip:3] Hangup(“/red5sip_user-000000a6”,””) in new stack
n Spawn extension (rooms-red5sip, 40016,3) exited non-zero on ‘/red5sip_user-000000a6’
It appears to check the database not find the room and then hang up.
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, August 01, 2014 10:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
you can search red5sip in config :)
the key is "red5sip.enable"
On 1 August 2014 23:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the response.
I have confirmed everything but I am not sure where to find this setting. I am assuming Admin config is Openmeeting Admin->Configuration. If so I don’t a setting for Red5sip key.
3) red5sip* key should be enabled in Admin->Config – NOT SURE OF THIS STEP
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Wednesday, July 30, 2014 6:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
OM is accessible on all network interfaces by default
config.xml need to be modified only in case you need to restrict OM client.
According to red5sip enter-exit-enter-exit-.... it should be due to misconfiguration. Unfortunately this integration is not simple by design :( I'm using logs and debug to set it up properly.
Main steps are
1) asterisk should be configured to have access to OM DB
2) asterisk bean should be uncommented and configured properly in openmeetings-application.xml
3) red5sip* key should be enabled in Admin->Config
4) in case asterisk is integrated with OM user should be re-saved (to have password-hash being saved in asterisk DB table)
5) sip should be enabled in the room
this should be all (hope I haven't miss anything)
On 29 July 2014 08:29, Horace Miles <Ho...@myit-solutions.com> wrote:
Hi Maxim,
My box is connected directly to a public IP, no NAT. My understanding was that Openmeetings to be access from the internet needed to be on a public address. That address would be the one in the config.xml. If I a mistaken let me know.
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 2:42 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Only with code modification
On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
Dear all,
can i disable arrow pointer for all participants in restricted room on Whiteboard??
--
Regards,
M K Raju.
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
RE: Pointer on WB
Posted by Horace Miles <Ho...@myit-solutions.com>.
Sorry Maxim,
I thought I had sent this to te users/dev lists
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, August 18, 2014 8:41 PM
To: Тимур Тлеукенов
Cc: Horace Miles; Openmeetings user-list
Subject: Fwd: Pointer on WB
@Timur, could you please answer this question
@Horace, please send your questions to user/dev lists
---------- Forwarded message ----------
From: Horace Miles <Ho...@myit-solutions.com>
Date: 14 August 2014 20:45
Subject: RE: Pointer on WB
To: Maxim Solodovnik <so...@gmail.com>
I see that the VOIP integration will allow multiple people to dial into a conference room. Is there a way to automatically assign the ext for incoming calls based upon the conference ID?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, August 12, 2014 7:52 AM
To: Horace Miles
Subject: Re: Pointer on WB
this keys are created via JAVA code of OM
in case this part of config is made properly:
Update Openmeetings with creadentials for Asterisk manager. Modify /opt/red5/webapps/openmeetings/WEB-INF/classes/openmeetings-applicationContext.xml
find <bean id="sipDao" class="org.apache.openmeetings.db.dao.room.SipDao"> uncomment its parameters and set it to your custom values.
IMPORTANT: this step should be done BEFORE system install/restore otherwise all SIP related room information will be lost
On 12 August 2014 20:43, Horace Miles <Ho...@myit-solutions.com> wrote:
Good morning/afternoon or evening Maxim,
There was no databse entry in the astdb table for openmeetings/rooms
So I did the following
Database put openmeetings rooms 40000 as test and now it shows asterisk –rx “database show”
Has an entry openmeetings/rooms 40000.
The sip transport now stays in the room YEAH…..
Since this was a fresh install of Asterisk per the Integration instructions should there be an instruction to add the family openmeetings key rooms to the astDB or did I miss this somewhere?
Since I only used the red5sip-integration_3.0 guide. Is there anything else that I as a novice would need to know or install to make an outgoing call?
Thanks for all your help
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, August 11, 2014 11:08 AM
To: Horace Miles
Cc: Openmeetings user-list; Тимур Тлеукенов
Subject: Re: Pointer on WB
please provide documentation change, I'll merge it :)
On 12 August 2014 00:46, Horace Miles <Ho...@myit-solutions.com> wrote:
My apologies to Turmik. I had a completely different understanding of how this was working. Thanks for your reply and info. I had read this information as well in the extension programming documentation for Asterisk, but thought Openmeetings was trying to use it in a different manner. The documentation should include some information on this to pre-clude future novice users as myself from making this same mistake.
Miles
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, August 11, 2014 8:20 AM
To: Horace Miles
Cc: Openmeetings user-list; Тимур Тлеукенов
Subject: Re: Pointer on WB
It is by desing, OM DB has table "room"
Asterisk key has name "/openmeetings/rooms" it doesn't need to match
On 11 August 2014 21:46, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Maxim,
Why would my openmeetings database not have a rooms table but have one name room?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, August 11, 2014 7:50 AM
To: Horace Miles
Cc: Openmeetings user-list; Тимур Тлеукенов
Subject: Re: Pointer on WB
sudo asterisk -rx "database show"
/openmeetings/rooms/4002 :
/openmeetings/rooms/40096 :
so in my case local asterisk key/value DB is quered
On 11 August 2014 20:19, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Timur,
That is what the Asterisk documentations says that dbexist does. However, as I am reading the code it is being used to query the openmeetings database and not the AstDB. The code below is from the Openmeetings SOAP and VOIP Integration wiki.
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
so according to your explaination openmeetings/rooms is being used to stored blacklisted numbers? Because if that is the case the bridge will never be reached because in my database the following is true:
table Rooms does not exist
table room does exist
table room holds all of the room information for all rooms created with the extension for that room
If this is being used for the blacklist then the following line will never be reached because the extensions are in this table
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user) - this line will never be reached if that is the way the GotoIF line is being used.
When I see the code above it is saying if the extension exist in the room table then connect to the bridge if it does not then hangup.
The problem is openmeetings/rooms table does not exist but table/room does exist and as far as I can tell there is no way for the query to look at the field with the extension in it, it is simply going to rooms or room table and not looking at a particular field. In this case not finding a field it returns a 0 does not exist and goes to the hangup line.
I can reverse the logic and say if 0 is returned then run the confbridge line but I don’t think it is designed to work that way.
So is there suppose to be a table with the name ‘rooms’?
If so where can I find the layout or schema for it?
What routines in Openmeetings will update the table. Because either something is missing OR there is a typo in the documentation AND/OR there is a typo and missing code in the DBEXIST line what would tell DBEXIST what field to check for the confo number?
Miles
From: Тимур Тлеукенов [mailto:timur.tleukenov@gmail.com]
Sent: Sunday, August 10, 2014 11:02 PM
To: Maxim Solodovnik
Cc: Openmeetings user-list
Subject: Re: Pointer on WB
Hello,
This extension code doesn't access openmeetings database, it uses internal asterisk database AstDB. It is key/value database.
If record "open30/room/${EXTEN})}" (where ${EXTEN} is name of calling extension) exists in asterisk database, then extension will be blocked.
Related asterisk database documentation:
http://www.voip-info.org/wiki/view/Asterisk+func+db
http://www.voip-info.org/wiki/view/Asterisk+func+db_exists
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
RE: Pointer on WB
Posted by Horace Miles <Ho...@myit-solutions.com>.
Disregard this request
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, August 18, 2014 8:41 PM
To: Тимур Тлеукенов
Cc: Horace Miles; Openmeetings user-list
Subject: Fwd: Pointer on WB
@Timur, could you please answer this question
@Horace, please send your questions to user/dev lists
---------- Forwarded message ----------
From: Horace Miles <Ho...@myit-solutions.com>
Date: 14 August 2014 20:45
Subject: RE: Pointer on WB
To: Maxim Solodovnik <so...@gmail.com>
I see that the VOIP integration will allow multiple people to dial into a conference room. Is there a way to automatically assign the ext for incoming calls based upon the conference ID?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Tuesday, August 12, 2014 7:52 AM
To: Horace Miles
Subject: Re: Pointer on WB
this keys are created via JAVA code of OM
in case this part of config is made properly:
Update Openmeetings with creadentials for Asterisk manager. Modify /opt/red5/webapps/openmeetings/WEB-INF/classes/openmeetings-applicationContext.xml
find <bean id="sipDao" class="org.apache.openmeetings.db.dao.room.SipDao"> uncomment its parameters and set it to your custom values.
IMPORTANT: this step should be done BEFORE system install/restore otherwise all SIP related room information will be lost
On 12 August 2014 20:43, Horace Miles <Ho...@myit-solutions.com> wrote:
Good morning/afternoon or evening Maxim,
There was no databse entry in the astdb table for openmeetings/rooms
So I did the following
Database put openmeetings rooms 40000 as test and now it shows asterisk –rx “database show”
Has an entry openmeetings/rooms 40000.
The sip transport now stays in the room YEAH…..
Since this was a fresh install of Asterisk per the Integration instructions should there be an instruction to add the family openmeetings key rooms to the astDB or did I miss this somewhere?
Since I only used the red5sip-integration_3.0 guide. Is there anything else that I as a novice would need to know or install to make an outgoing call?
Thanks for all your help
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, August 11, 2014 11:08 AM
To: Horace Miles
Cc: Openmeetings user-list; Тимур Тлеукенов
Subject: Re: Pointer on WB
please provide documentation change, I'll merge it :)
On 12 August 2014 00:46, Horace Miles <Ho...@myit-solutions.com> wrote:
My apologies to Turmik. I had a completely different understanding of how this was working. Thanks for your reply and info. I had read this information as well in the extension programming documentation for Asterisk, but thought Openmeetings was trying to use it in a different manner. The documentation should include some information on this to pre-clude future novice users as myself from making this same mistake.
Miles
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, August 11, 2014 8:20 AM
To: Horace Miles
Cc: Openmeetings user-list; Тимур Тлеукенов
Subject: Re: Pointer on WB
It is by desing, OM DB has table "room"
Asterisk key has name "/openmeetings/rooms" it doesn't need to match
On 11 August 2014 21:46, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Maxim,
Why would my openmeetings database not have a rooms table but have one name room?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, August 11, 2014 7:50 AM
To: Horace Miles
Cc: Openmeetings user-list; Тимур Тлеукенов
Subject: Re: Pointer on WB
sudo asterisk -rx "database show"
/openmeetings/rooms/4002 :
/openmeetings/rooms/40096 :
so in my case local asterisk key/value DB is quered
On 11 August 2014 20:19, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Timur,
That is what the Asterisk documentations says that dbexist does. However, as I am reading the code it is being used to query the openmeetings database and not the AstDB. The code below is from the Openmeetings SOAP and VOIP Integration wiki.
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
so according to your explaination openmeetings/rooms is being used to stored blacklisted numbers? Because if that is the case the bridge will never be reached because in my database the following is true:
table Rooms does not exist
table room does exist
table room holds all of the room information for all rooms created with the extension for that room
If this is being used for the blacklist then the following line will never be reached because the extensions are in this table
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user) - this line will never be reached if that is the way the GotoIF line is being used.
When I see the code above it is saying if the extension exist in the room table then connect to the bridge if it does not then hangup.
The problem is openmeetings/rooms table does not exist but table/room does exist and as far as I can tell there is no way for the query to look at the field with the extension in it, it is simply going to rooms or room table and not looking at a particular field. In this case not finding a field it returns a 0 does not exist and goes to the hangup line.
I can reverse the logic and say if 0 is returned then run the confbridge line but I don’t think it is designed to work that way.
So is there suppose to be a table with the name ‘rooms’?
If so where can I find the layout or schema for it?
What routines in Openmeetings will update the table. Because either something is missing OR there is a typo in the documentation AND/OR there is a typo and missing code in the DBEXIST line what would tell DBEXIST what field to check for the confo number?
Miles
From: Тимур Тлеукенов [mailto:timur.tleukenov@gmail.com]
Sent: Sunday, August 10, 2014 11:02 PM
To: Maxim Solodovnik
Cc: Openmeetings user-list
Subject: Re: Pointer on WB
Hello,
This extension code doesn't access openmeetings database, it uses internal asterisk database AstDB. It is key/value database.
If record "open30/room/${EXTEN})}" (where ${EXTEN} is name of calling extension) exists in asterisk database, then extension will be blocked.
Related asterisk database documentation:
http://www.voip-info.org/wiki/view/Asterisk+func+db
http://www.voip-info.org/wiki/view/Asterisk+func+db_exists
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Fwd: Pointer on WB
Posted by Maxim Solodovnik <so...@gmail.com>.
@Timur, could you please answer this question
@Horace, please send your questions to user/dev lists
---------- Forwarded message ----------
From: Horace Miles <Ho...@myit-solutions.com>
Date: 14 August 2014 20:45
Subject: RE: Pointer on WB
To: Maxim Solodovnik <so...@gmail.com>
I see that the VOIP integration will allow multiple people to dial into a
conference room. Is there a way to automatically assign the ext for
incoming calls based upon the conference ID?
*From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
*Sent:* Tuesday, August 12, 2014 7:52 AM
*To:* Horace Miles
*Subject:* Re: Pointer on WB
this keys are created via JAVA code of OM
in case this part of config is made properly:
Update Openmeetings with creadentials for Asterisk manager. Modify
/opt/red5/webapps/openmeetings/WEB-INF/classes/openmeetings-applicationContext.xml
find *<bean id="sipDao"
class="org.apache.openmeetings.db.dao.room.SipDao">* uncomment
its parameters and set it to your custom values.
IMPORTANT: this step should be done *BEFORE* system install/restore
otherwise all SIP related room information will be lost
On 12 August 2014 20:43, Horace Miles <Ho...@myit-solutions.com>
wrote:
Good morning/afternoon or evening Maxim,
There was no databse entry in the astdb table for openmeetings/rooms
So I did the following
Database put openmeetings rooms 40000 as test and now it shows asterisk –rx
“database show”
Has an entry openmeetings/rooms 40000.
The sip transport now stays in the room YEAH…..
Since this was a fresh install of Asterisk per the Integration instructions
should there be an instruction to add the family openmeetings key rooms to
the astDB or did I miss this somewhere?
Since I only used the red5sip-integration_3.0 guide. Is there anything
else that I as a novice would need to know or install to make an outgoing
call?
Thanks for all your help
*From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
*Sent:* Monday, August 11, 2014 11:08 AM
*To:* Horace Miles
*Cc:* Openmeetings user-list; Тимур Тлеукенов
*Subject:* Re: Pointer on WB
please provide documentation change, I'll merge it :)
On 12 August 2014 00:46, Horace Miles <Ho...@myit-solutions.com>
wrote:
My apologies to Turmik. I had a completely different understanding of how
this was working. Thanks for your reply and info. I had read this
information as well in the extension programming documentation for
Asterisk, but thought Openmeetings was trying to use it in a different
manner. The documentation should include some information on this to
pre-clude future novice users as myself from making this same mistake.
Miles
*From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
*Sent:* Monday, August 11, 2014 8:20 AM
*To:* Horace Miles
*Cc:* Openmeetings user-list; Тимур Тлеукенов
*Subject:* Re: Pointer on WB
It is by desing, OM DB has table "room"
Asterisk key has name "/openmeetings/rooms" it doesn't need to match
On 11 August 2014 21:46, Horace Miles <Ho...@myit-solutions.com>
wrote:
Thanks Maxim,
Why would my openmeetings database not have a rooms table but have one name
room?
*From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
*Sent:* Monday, August 11, 2014 7:50 AM
*To:* Horace Miles
*Cc:* Openmeetings user-list; Тимур Тлеукенов
*Subject:* Re: Pointer on WB
sudo asterisk -rx "database show"
/openmeetings/rooms/4002 :
/openmeetings/rooms/40096 :
so in my case local asterisk key/value DB is quered
On 11 August 2014 20:19, Horace Miles <Ho...@myit-solutions.com>
wrote:
Thanks Timur,
That is what the Asterisk documentations says that dbexist does. However,
as I am reading the code it is being used to query the openmeetings
database and not the AstDB. The code below is from the Openmeetings SOAP
and VOIP Integration wiki.
exten =>
_400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
so according to your explaination openmeetings/rooms is being used to
stored blacklisted numbers? Because if that is the case the bridge will
never be reached because in my database the following is true:
table Rooms does not exist
table room does exist
table room holds all of the room information for all rooms created with the
extension for that room
If this is being used for the blacklist then the following line will never
be reached because the extensions are in this table
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user) -
this line will never be reached if that is the way the GotoIF line is being
used.
When I see the code above it is saying if the extension exist in the room
table then connect to the bridge if it does not then hangup.
The problem is openmeetings/rooms table does not exist but table/room does
exist and as far as I can tell there is no way for the query to look at the
field with the extension in it, it is simply going to rooms or room table
and not looking at a particular field. In this case not finding a field it
returns a 0 does not exist and goes to the hangup line.
I can reverse the logic and say if 0 is returned then run the confbridge
line but I don’t think it is designed to work that way.
So is there suppose to be a table with the name ‘rooms’?
If so where can I find the layout or schema for it?
What routines in Openmeetings will update the table. Because either
something is missing OR there is a typo in the documentation AND/OR there
is a typo and missing code in the DBEXIST line what would tell DBEXIST what
field to check for the confo number?
Miles
*From:* Тимур Тлеукенов [mailto:timur.tleukenov@gmail.com]
*Sent:* Sunday, August 10, 2014 11:02 PM
*To:* Maxim Solodovnik
*Cc:* Openmeetings user-list
*Subject:* Re: Pointer on WB
Hello,
This extension code doesn't access openmeetings database, it uses internal
asterisk database AstDB. It is key/value database.
If record "open30/room/${EXTEN})}" (where ${EXTEN} is name of calling
extension) exists in asterisk database, then extension will be blocked.
Related asterisk database documentation:
http://www.voip-info.org/wiki/view/Asterisk+func+db
http://www.voip-info.org/wiki/view/Asterisk+func+db_exists
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: Pointer on WB
Posted by Maxim Solodovnik <so...@gmail.com>.
please provide documentation change, I'll merge it :)
On 12 August 2014 00:46, Horace Miles <Ho...@myit-solutions.com>
wrote:
> My apologies to Turmik. I had a completely different understanding of how
> this was working. Thanks for your reply and info. I had read this
> information as well in the extension programming documentation for
> Asterisk, but thought Openmeetings was trying to use it in a different
> manner. The documentation should include some information on this to
> pre-clude future novice users as myself from making this same mistake.
>
>
>
> Miles
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Monday, August 11, 2014 8:20 AM
> *To:* Horace Miles
> *Cc:* Openmeetings user-list; Тимур Тлеукенов
> *Subject:* Re: Pointer on WB
>
>
>
> It is by desing, OM DB has table "room"
>
> Asterisk key has name "/openmeetings/rooms" it doesn't need to match
>
>
>
> On 11 August 2014 21:46, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks Maxim,
>
>
>
> Why would my openmeetings database not have a rooms table but have one
> name room?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Monday, August 11, 2014 7:50 AM
> *To:* Horace Miles
> *Cc:* Openmeetings user-list; Тимур Тлеукенов
> *Subject:* Re: Pointer on WB
>
>
>
> sudo asterisk -rx "database show"
>
>
>
> /openmeetings/rooms/4002 :
>
>
> /openmeetings/rooms/40096 :
>
>
>
>
> so in my case local asterisk key/value DB is quered
>
>
>
> On 11 August 2014 20:19, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks Timur,
>
> That is what the Asterisk documentations says that dbexist does. However,
> as I am reading the code it is being used to query the openmeetings
> database and not the AstDB. The code below is from the Openmeetings SOAP
> and VOIP Integration wiki.
>
> exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
>
> so according to your explaination openmeetings/rooms is being used to
> stored blacklisted numbers? Because if that is the case the bridge will
> never be reached because in my database the following is true:
>
> table Rooms does not exist
>
> table room does exist
>
> table room holds all of the room information for all rooms created with
> the extension for that room
>
> If this is being used for the blacklist then the following line will never
> be reached because the extensions are in this table
>
> exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user) -
> this line will never be reached if that is the way the GotoIF line is being
> used.
>
> When I see the code above it is saying if the extension exist in the room
> table then connect to the bridge if it does not then hangup.
>
> The problem is openmeetings/rooms table does not exist but table/room does
> exist and as far as I can tell there is no way for the query to look at the
> field with the extension in it, it is simply going to rooms or room table
> and not looking at a particular field. In this case not finding a field it
> returns a 0 does not exist and goes to the hangup line.
>
> I can reverse the logic and say if 0 is returned then run the confbridge
> line but I don’t think it is designed to work that way.
>
> So is there suppose to be a table with the name ‘rooms’?
>
> If so where can I find the layout or schema for it?
>
> What routines in Openmeetings will update the table. Because either
> something is missing OR there is a typo in the documentation AND/OR there
> is a typo and missing code in the DBEXIST line what would tell DBEXIST what
> field to check for the confo number?
>
>
>
> Miles
>
>
>
>
>
>
>
> *From:* Тимур Тлеукенов [mailto:timur.tleukenov@gmail.com]
> *Sent:* Sunday, August 10, 2014 11:02 PM
> *To:* Maxim Solodovnik
> *Cc:* Openmeetings user-list
>
>
> *Subject:* Re: Pointer on WB
>
>
>
> Hello,
>
>
> This extension code doesn't access openmeetings database, it uses internal
> asterisk database AstDB. It is key/value database.
> If record "open30/room/${EXTEN})}" (where ${EXTEN} is name of calling
> extension) exists in asterisk database, then extension will be blocked.
>
> Related asterisk database documentation:
> http://www.voip-info.org/wiki/view/Asterisk+func+db
> http://www.voip-info.org/wiki/view/Asterisk+func+db_exists
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: Pointer on WB
Posted by Horace Miles <Ho...@myit-solutions.com>.
My apologies to Turmik. I had a completely different understanding of how this was working. Thanks for your reply and info. I had read this information as well in the extension programming documentation for Asterisk, but thought Openmeetings was trying to use it in a different manner. The documentation should include some information on this to pre-clude future novice users as myself from making this same mistake.
Miles
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, August 11, 2014 8:20 AM
To: Horace Miles
Cc: Openmeetings user-list; Тимур Тлеукенов
Subject: Re: Pointer on WB
It is by desing, OM DB has table "room"
Asterisk key has name "/openmeetings/rooms" it doesn't need to match
On 11 August 2014 21:46, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Maxim,
Why would my openmeetings database not have a rooms table but have one name room?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, August 11, 2014 7:50 AM
To: Horace Miles
Cc: Openmeetings user-list; Тимур Тлеукенов
Subject: Re: Pointer on WB
sudo asterisk -rx "database show"
/openmeetings/rooms/4002 :
/openmeetings/rooms/40096 :
so in my case local asterisk key/value DB is quered
On 11 August 2014 20:19, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Timur,
That is what the Asterisk documentations says that dbexist does. However, as I am reading the code it is being used to query the openmeetings database and not the AstDB. The code below is from the Openmeetings SOAP and VOIP Integration wiki.
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
so according to your explaination openmeetings/rooms is being used to stored blacklisted numbers? Because if that is the case the bridge will never be reached because in my database the following is true:
table Rooms does not exist
table room does exist
table room holds all of the room information for all rooms created with the extension for that room
If this is being used for the blacklist then the following line will never be reached because the extensions are in this table
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user) - this line will never be reached if that is the way the GotoIF line is being used.
When I see the code above it is saying if the extension exist in the room table then connect to the bridge if it does not then hangup.
The problem is openmeetings/rooms table does not exist but table/room does exist and as far as I can tell there is no way for the query to look at the field with the extension in it, it is simply going to rooms or room table and not looking at a particular field. In this case not finding a field it returns a 0 does not exist and goes to the hangup line.
I can reverse the logic and say if 0 is returned then run the confbridge line but I don’t think it is designed to work that way.
So is there suppose to be a table with the name ‘rooms’?
If so where can I find the layout or schema for it?
What routines in Openmeetings will update the table. Because either something is missing OR there is a typo in the documentation AND/OR there is a typo and missing code in the DBEXIST line what would tell DBEXIST what field to check for the confo number?
Miles
From: Тимур Тлеукенов [mailto:timur.tleukenov@gmail.com]
Sent: Sunday, August 10, 2014 11:02 PM
To: Maxim Solodovnik
Cc: Openmeetings user-list
Subject: Re: Pointer on WB
Hello,
This extension code doesn't access openmeetings database, it uses internal asterisk database AstDB. It is key/value database.
If record "open30/room/${EXTEN})}" (where ${EXTEN} is name of calling extension) exists in asterisk database, then extension will be blocked.
Related asterisk database documentation:
http://www.voip-info.org/wiki/view/Asterisk+func+db
http://www.voip-info.org/wiki/view/Asterisk+func+db_exists
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: Pointer on WB
Posted by Maxim Solodovnik <so...@gmail.com>.
It is by desing, OM DB has table "room"
Asterisk key has name "/openmeetings/rooms" it doesn't need to match
On 11 August 2014 21:46, Horace Miles <Ho...@myit-solutions.com>
wrote:
> Thanks Maxim,
>
>
>
> Why would my openmeetings database not have a rooms table but have one
> name room?
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Monday, August 11, 2014 7:50 AM
> *To:* Horace Miles
> *Cc:* Openmeetings user-list; Тимур Тлеукенов
> *Subject:* Re: Pointer on WB
>
>
>
> sudo asterisk -rx "database show"
>
>
>
> /openmeetings/rooms/4002 :
>
>
> /openmeetings/rooms/40096 :
>
>
>
>
> so in my case local asterisk key/value DB is quered
>
>
>
> On 11 August 2014 20:19, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Thanks Timur,
>
> That is what the Asterisk documentations says that dbexist does. However,
> as I am reading the code it is being used to query the openmeetings
> database and not the AstDB. The code below is from the Openmeetings SOAP
> and VOIP Integration wiki.
>
> exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
>
> so according to your explaination openmeetings/rooms is being used to
> stored blacklisted numbers? Because if that is the case the bridge will
> never be reached because in my database the following is true:
>
> table Rooms does not exist
>
> table room does exist
>
> table room holds all of the room information for all rooms created with
> the extension for that room
>
> If this is being used for the blacklist then the following line will never
> be reached because the extensions are in this table
>
> exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user) -
> this line will never be reached if that is the way the GotoIF line is being
> used.
>
> When I see the code above it is saying if the extension exist in the room
> table then connect to the bridge if it does not then hangup.
>
> The problem is openmeetings/rooms table does not exist but table/room does
> exist and as far as I can tell there is no way for the query to look at the
> field with the extension in it, it is simply going to rooms or room table
> and not looking at a particular field. In this case not finding a field it
> returns a 0 does not exist and goes to the hangup line.
>
> I can reverse the logic and say if 0 is returned then run the confbridge
> line but I don’t think it is designed to work that way.
>
> So is there suppose to be a table with the name ‘rooms’?
>
> If so where can I find the layout or schema for it?
>
> What routines in Openmeetings will update the table. Because either
> something is missing OR there is a typo in the documentation AND/OR there
> is a typo and missing code in the DBEXIST line what would tell DBEXIST what
> field to check for the confo number?
>
>
>
> Miles
>
>
>
>
>
>
>
> *From:* Тимур Тлеукенов [mailto:timur.tleukenov@gmail.com]
> *Sent:* Sunday, August 10, 2014 11:02 PM
> *To:* Maxim Solodovnik
> *Cc:* Openmeetings user-list
>
>
> *Subject:* Re: Pointer on WB
>
>
>
> Hello,
>
>
> This extension code doesn't access openmeetings database, it uses internal
> asterisk database AstDB. It is key/value database.
> If record "open30/room/${EXTEN})}" (where ${EXTEN} is name of calling
> extension) exists in asterisk database, then extension will be blocked.
>
> Related asterisk database documentation:
> http://www.voip-info.org/wiki/view/Asterisk+func+db
> http://www.voip-info.org/wiki/view/Asterisk+func+db_exists
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: Pointer on WB
Posted by Horace Miles <Ho...@myit-solutions.com>.
Thanks Maxim,
Why would my openmeetings database not have a rooms table but have one name room?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Monday, August 11, 2014 7:50 AM
To: Horace Miles
Cc: Openmeetings user-list; Тимур Тлеукенов
Subject: Re: Pointer on WB
sudo asterisk -rx "database show"
/openmeetings/rooms/4002 :
/openmeetings/rooms/40096 :
so in my case local asterisk key/value DB is quered
On 11 August 2014 20:19, Horace Miles <Ho...@myit-solutions.com> wrote:
Thanks Timur,
That is what the Asterisk documentations says that dbexist does. However, as I am reading the code it is being used to query the openmeetings database and not the AstDB. The code below is from the Openmeetings SOAP and VOIP Integration wiki.
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
so according to your explaination openmeetings/rooms is being used to stored blacklisted numbers? Because if that is the case the bridge will never be reached because in my database the following is true:
table Rooms does not exist
table room does exist
table room holds all of the room information for all rooms created with the extension for that room
If this is being used for the blacklist then the following line will never be reached because the extensions are in this table
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user) - this line will never be reached if that is the way the GotoIF line is being used.
When I see the code above it is saying if the extension exist in the room table then connect to the bridge if it does not then hangup.
The problem is openmeetings/rooms table does not exist but table/room does exist and as far as I can tell there is no way for the query to look at the field with the extension in it, it is simply going to rooms or room table and not looking at a particular field. In this case not finding a field it returns a 0 does not exist and goes to the hangup line.
I can reverse the logic and say if 0 is returned then run the confbridge line but I don’t think it is designed to work that way.
So is there suppose to be a table with the name ‘rooms’?
If so where can I find the layout or schema for it?
What routines in Openmeetings will update the table. Because either something is missing OR there is a typo in the documentation AND/OR there is a typo and missing code in the DBEXIST line what would tell DBEXIST what field to check for the confo number?
Miles
From: Тимур Тлеукенов [mailto:timur.tleukenov@gmail.com]
Sent: Sunday, August 10, 2014 11:02 PM
To: Maxim Solodovnik
Cc: Openmeetings user-list
Subject: Re: Pointer on WB
Hello,
This extension code doesn't access openmeetings database, it uses internal asterisk database AstDB. It is key/value database.
If record "open30/room/${EXTEN})}" (where ${EXTEN} is name of calling extension) exists in asterisk database, then extension will be blocked.
Related asterisk database documentation:
http://www.voip-info.org/wiki/view/Asterisk+func+db
http://www.voip-info.org/wiki/view/Asterisk+func+db_exists
--
WBR
Maxim aka solomax
Re: Pointer on WB
Posted by Maxim Solodovnik <so...@gmail.com>.
sudo asterisk -rx "database show"
/openmeetings/rooms/4002 :
/openmeetings/rooms/40096 :
so in my case local asterisk key/value DB is quered
On 11 August 2014 20:19, Horace Miles <Ho...@myit-solutions.com>
wrote:
> Thanks Timur,
>
> That is what the Asterisk documentations says that dbexist does. However,
> as I am reading the code it is being used to query the openmeetings
> database and not the AstDB. The code below is from the Openmeetings SOAP
> and VOIP Integration wiki.
>
> exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
>
> so according to your explaination openmeetings/rooms is being used to
> stored blacklisted numbers? Because if that is the case the bridge will
> never be reached because in my database the following is true:
>
> table Rooms does not exist
>
> table room does exist
>
> table room holds all of the room information for all rooms created with
> the extension for that room
>
> If this is being used for the blacklist then the following line will never
> be reached because the extensions are in this table
>
> exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user) -
> this line will never be reached if that is the way the GotoIF line is being
> used.
>
> When I see the code above it is saying if the extension exist in the room
> table then connect to the bridge if it does not then hangup.
>
> The problem is openmeetings/rooms table does not exist but table/room does
> exist and as far as I can tell there is no way for the query to look at the
> field with the extension in it, it is simply going to rooms or room table
> and not looking at a particular field. In this case not finding a field it
> returns a 0 does not exist and goes to the hangup line.
>
> I can reverse the logic and say if 0 is returned then run the confbridge
> line but I don’t think it is designed to work that way.
>
> So is there suppose to be a table with the name ‘rooms’?
>
> If so where can I find the layout or schema for it?
>
> What routines in Openmeetings will update the table. Because either
> something is missing OR there is a typo in the documentation AND/OR there
> is a typo and missing code in the DBEXIST line what would tell DBEXIST what
> field to check for the confo number?
>
>
>
> Miles
>
>
>
>
>
>
>
> *From:* Тимур Тлеукенов [mailto:timur.tleukenov@gmail.com]
> *Sent:* Sunday, August 10, 2014 11:02 PM
> *To:* Maxim Solodovnik
> *Cc:* Openmeetings user-list
>
> *Subject:* Re: Pointer on WB
>
>
>
> Hello,
>
> This extension code doesn't access openmeetings database, it uses internal
> asterisk database AstDB. It is key/value database.
> If record "open30/room/${EXTEN})}" (where ${EXTEN} is name of calling
> extension) exists in asterisk database, then extension will be blocked.
>
> Related asterisk database documentation:
> http://www.voip-info.org/wiki/view/Asterisk+func+db
> http://www.voip-info.org/wiki/view/Asterisk+func+db_exists
>
--
WBR
Maxim aka solomax
RE: Pointer on WB
Posted by Horace Miles <Ho...@myit-solutions.com>.
Thanks Timur,
That is what the Asterisk documentations says that dbexist does. However, as I am reading the code it is being used to query the openmeetings database and not the AstDB. The code below is from the Openmeetings SOAP and VOIP Integration wiki.
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
so according to your explaination openmeetings/rooms is being used to stored blacklisted numbers? Because if that is the case the bridge will never be reached because in my database the following is true:
table Rooms does not exist
table room does exist
table room holds all of the room information for all rooms created with the extension for that room
If this is being used for the blacklist then the following line will never be reached because the extensions are in this table
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user) - this line will never be reached if that is the way the GotoIF line is being used.
When I see the code above it is saying if the extension exist in the room table then connect to the bridge if it does not then hangup.
The problem is openmeetings/rooms table does not exist but table/room does exist and as far as I can tell there is no way for the query to look at the field with the extension in it, it is simply going to rooms or room table and not looking at a particular field. In this case not finding a field it returns a 0 does not exist and goes to the hangup line.
I can reverse the logic and say if 0 is returned then run the confbridge line but I don’t think it is designed to work that way.
So is there suppose to be a table with the name ‘rooms’?
If so where can I find the layout or schema for it?
What routines in Openmeetings will update the table. Because either something is missing OR there is a typo in the documentation AND/OR there is a typo and missing code in the DBEXIST line what would tell DBEXIST what field to check for the confo number?
Miles
From: Тимур Тлеукенов [mailto:timur.tleukenov@gmail.com]
Sent: Sunday, August 10, 2014 11:02 PM
To: Maxim Solodovnik
Cc: Openmeetings user-list
Subject: Re: Pointer on WB
Hello,
This extension code doesn't access openmeetings database, it uses internal asterisk database AstDB. It is key/value database.
If record "open30/room/${EXTEN})}" (where ${EXTEN} is name of calling extension) exists in asterisk database, then extension will be blocked.
Related asterisk database documentation:
http://www.voip-info.org/wiki/view/Asterisk+func+db
http://www.voip-info.org/wiki/view/Asterisk+func+db_exists
Re: Pointer on WB
Posted by Тимур Тлеукенов <ti...@gmail.com>.
Hello,
This extension code doesn't access openmeetings database, it uses internal
asterisk database AstDB. It is key/value database.
If record "open30/room/${EXTEN})}" (where ${EXTEN} is name of calling
extension) exists in asterisk database, then extension will be blocked.
Related asterisk database documentation:
http://www.voip-info.org/wiki/view/Asterisk+func+db
http://www.voip-info.org/wiki/view/Asterisk+func+db_exists
Re: Pointer on WB
Posted by Maxim Solodovnik <so...@gmail.com>.
Hello Timur,
can you please answer this?
Thanks :)
On 11 August 2014 05:39, Horace Miles <Ho...@myit-solutions.com>
wrote:
> Ok those credentials are correct.
>
>
>
> My next question would be is how does the following line of code now what
> field in the openmeetings/rooms to lookup?
>
> exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?notavail:ok)
>
> I can see the database - open30
>
> I can see the table – room
>
> What is telling it what field to look at in the table?
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Sunday, August 10, 2014 9:47 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> DB credentials used are configured in /etc/odbc.ini
>
> as described in
> http://openmeetings.apache.org/red5sip-integration_3.0.html
>
>
>
> On 10 August 2014 22:09, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> I took the firewall down. Still had the same problem.
>
>
>
> I reverse the logic in this line of the extensions.conf
>
> exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?notavail:ok)
> >>>>>>>>>> reverse the notavail and OK.
>
> It will then invoke the video bridge. Cant make a call but it shows me
> that it asterisk is not finding the record in openmeetings. I cant figure
> out wht account it is using to make the query. Openmeetings, root, or
> red5sip_user. I cant figure out the correlation of which is making the
> call to the openmeetings database.
>
>
>
> Miles
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Sunday, August 10, 2014 3:36 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Maybe it is network configuration issue as described here:
> http://stackoverflow.com/questions/22093328/asterisk-sip-retransmission-timeout
> ?
>
>
>
> can you check it with all firewalls disabled?
>
>
>
> On 8 August 2014 23:51, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim,
>
>
>
> Whenever you have time I understand. Here are all of my configurations by
> file name. I hope it will help.
>
> HERE IS THE CONFIGURATION FOR EACH FILE FOR THE ASTERISK INTEGRATION. I
> HAVE SEPPERATED EACH SECTION WITH “==========”
>
> HOPEFULLY SOMEONE CAN SEE THE ERROR I HAVE MADE
>
>
>
> Miles
>
> ================================================================
>
> CONFIGURATION for /etc/odbc.ini
>
> [asterisk-connector]
>
> Description = MySQL connection to 'openmeetings' database
>
> Driver = MySQL
>
> Database = open30
>
> Server = 127.0.0.1
>
> USER = root
>
> PASSWORD =******
>
> Port = 3306
>
> Socket = /var/run/mysqld/mysqld.sock
>
> ================================================================
>
> CONFIGURATION for /etc/odbcinst.ini
>
> [MySQL]
>
> Description = ODBC for MySQL
>
> Driver = /usr/lib/i386-linux-gnu/odbc/libmyodbc.so
>
> Setup = /usr/lib/i386-linux-gnu/odbc/libodbcmyS.so
>
> FileUsage = 1
>
> ================================================================
>
> CONFIGURATION for in /etc/asterisk/modules.conf
>
> [modules]
>
> autoload=yes
>
> ;
>
> ; Any modules that need to be loaded before the Asterisk core has been
>
> ; initialized (just after the logger has been initialized) can be loaded
>
> ; using 'preload'. This will frequently be needed if you wish to map all
>
> ; module configuration files into Realtime storage, since the Realtime
>
> ; driver will need to be loaded before the modules using those
> configuration
>
> ; files are initialized.
>
> ;
>
> ; An example of loading ODBC support would be:
>
> preload => res_odbc.so
>
> preload => res_config_odbc.so
>
> ================================================================
>
> CONFIGURATION for /etc/asterisk/res_odbc.conf
>
> ;;; odbc setup file
>
>
>
> ; ENV is a global set of environmental variables that will get set.
>
> ; Note that all environmental variables can be seen by all connections,
>
> ; so you can't have different values for different connections.
>
> [ENV]
>
> ;INFORMIXSERVER => my_special_database
>
> ;INFORMIXDIR => /opt/informix
>
> ;ORACLE_HOME => /home/oracle
>
>
>
> ; All other sections are arbitrary names for database connections.
>
>
>
> ;
>
> ; The context name is what will be used in other configuration files, such
>
> ; as extconfig.conf and func_odbc.conf, to reference this connection.
>
> [asterisk]
>
> ;
>
> ; Permit disabling sections without needing to comment them out.
>
> ; If not specified, it is assumed the section is enabled.
>
> enabled => yes
>
> ;
>
> ; This value should match an entry in /etc/odbc.ini
>
> ; (or /usr/local/etc/odbc.ini, on FreeBSD and similar systems).
>
> dsn => asterisk-connector
>
> ;
>
> ; Username for connecting to the database. The user defaults to the
> context
>
> ; name if unspecified.
>
> username => admin
>
> ;
>
> ; Password for authenticating the user to the database. The default
>
> ; password is blank.
>
> password => ******
>
> ;
>
> ; Build a connection at startup?
>
> pre-connect => yes
>
> ================================================================
>
> Configuration for /etc/asterisk/sip.conf
>
> ;
>
> ;
>
> ; SIP Configuration example for Asterisk
>
> ;
>
> ; Note: Please read the security documentation for Asterisk in order to
>
> ; understand the risks of installing Asterisk with the sample
>
> ; configuration. If your Asterisk is installed on a public
>
> ; IP address connected to the Internet, you will want to learn
>
> ; about the various security settings BEFORE you start
>
> ; Asterisk.
>
> ;
>
> ; Especially note the following settings:
>
> ; - allowguest (default enabled)
>
> ; - permit/deny/acl - IP address filters
>
> ; - contactpermit/contactdeny/contactacl - IP
> address filters for registrations
>
> ; - context - Which set of services you offer
> various users
>
> ;
>
>
>
> [general]
>
> context=public ; Default context for incoming calls.
> Defaults to 'default'
>
> allowoverlap=no ; Disable overlap dialing support.
> (Default is yes)
>
> realm=asterisk ; Realm for digest authentication
>
> udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to
> (0.0.0.0 binds to all)
>
> ; Optionally add a port number,
> 192.168.1.1:5062 (default is port 5060)
>
>
>
> tcpenable=yes ; Enable server for incoming TCP
> connections (default is no)
>
> tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to
> (0.0.0.0 binds to all interfaces)
>
> transport=udp ; Set the default transports. The order
> determines the primary default transport.
>
> ; If tcpenable=no and the transport set is
> tcp, we will fallback to UDP.
>
>
>
> srvlookup=yes ; Enable DNS SRV lookups on outbound calls
>
> maxexpiry=43200 ; Maximum allowed time of incoming
> registrations (seconds)
>
> videosupport=yes ; Turn on support for SIP video. You need
> to turn this
>
> rtcachefriends=yes ; Cache realtime friends by adding them to
> the internal list
>
>
>
> ;domain=mydomain.tld,mydomain-incoming
>
> ; Add domain and configure incoming context
>
> ; for external calls to this domain
>
> domain=127.0.0.1 ; Add IP address as local domain
>
> domain=98.174.244.232 ; You can have several "domain" settings
>
>
>
> [basic-options](!) ; a template
>
> dtmfmode=rfc2833
>
> context=from-office
>
> type=friend
>
>
>
> [natted-phone](!,basic-options) ; another template inheriting
> basic-options
>
> directmedia=no
>
> host=dynamic
>
>
>
> [public-phone](!,basic-options) ; another template inheriting
> basic-options
>
> directmedia=yes
>
>
>
> [my-codecs](!) ; a template for my preferred codecs
>
> disallow=all
>
> allow=ilbc
>
> allow=g729
>
> allow=gsm
>
> allow=g723
>
> allow=ulaw
>
> ; Or, more simply:
>
> ;allow=!all,ilbc,g729,gsm,g723,ulaw
>
>
>
> [ulaw-phone](!) ; and another one for ulaw-only
>
> disallow=all
>
> allow=ulaw
>
> ; Again, more simply:
>
> ;allow=!all,ulaw
>
>
>
> ; and finally instantiate a few phones
>
> ;
>
> ; [2133](natted-phone,my-codecs)
>
> ; secret = peekaboo
>
> ; [2134](natted-phone,ulaw-phone)
>
> ; secret = not_very_secret
>
> ; [2136](public-phone,ulaw-phone)
>
> ; secret = not_very_secret_either
>
> ; ...
>
> ;
>
> [red5sip_user]
>
> type=friend
>
> secret=12345
>
> disallow=all
>
> allow=ulaw
>
> allow=h264
>
> host=dynamic
>
> nat=no
>
> ;nat=force_rport,comedia
>
> context=rooms-red5sip
>
> ================================================================
>
> CONFIGURATION FOR /etc/asterisk/extconfig.conf
>
> ;
>
> ; Static and realtime external configuration
>
> ; engine configuration
>
> ;
>
> ; See
> https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
>
> ; for basic table formatting information.
>
> ;
>
> [settings]
>
> sippeers => odbc,asterisk,sipusers
>
> ================================================================
>
> CONFIGURATION FOR /etc/asterisk/extensions.conf
>
> [rooms]
>
> exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?ok:notavail)
>
> exten => _400X!,n(ok),SET(PIN=${DB(open30/room/${EXTEN})})
>
> exten => _400X!,n,Set(CONFBRIDGE(user,template)=sip_user)
>
> exten => _400X!,n,Set(CONFBRIDGE(user,pin)=${PIN})
>
> exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,)
>
> exten => _400X!,n,Hangup
>
> exten => _400X!,n(notavail),Answer()
>
> exten => _400X!,n,Playback(invalid)
>
> exten => _400X!,n,Hangup
>
>
>
> [rooms-originate]
>
> exten => _400X!,1,Confbridge(${EXTEN},default_bridge,sip_user)
>
> exten => _400X!,n,Hangup
>
>
>
> [rooms-out]
>
> ; *****************************************************
>
> ; Extensions for outgoing calls from Openmeetings room.
>
> ; *****************************************************
>
>
>
> [rooms-red5sip]
>
> exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?ok:notavail)
>
> exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
>
> exten => _400X!,n(notavail),Hangup
>
> ================================================================
>
> CONFIGURATION for /etc/asterisk/confbridge.conf
>
> [red5sip_user]
>
> type=user
>
> marked=yes
>
> dsp_drop_silence=yes
>
> denoise=true
>
>
>
> [sip_user]
>
> type=user
>
> end_marked=yes
>
> wait_marked=yes
>
> music_on_hold_when_empty=yes
>
> dsp_drop_silence=yes
>
> denoise=true
>
>
>
> [default_bridge]
>
> type=bridge
>
> video_mode=follow_talker
>
> ;video_mode=last_marked
>
> ;video_mode=first_marked
>
> ================================================================
>
> CONFIGURATION /etc/asterisk/manager.conf
>
> [general]
>
> ;enabled = no
>
> ;webenabled = yes
>
> enabled = yes
>
> webenabled = no
>
> port = 5038
>
> bindaddr = 127.0.0.1
>
>
>
> [openmeetings]
>
> secret = 12345
>
> deny=0.0.0.0/0.0.0.0
>
> permit=127.0.0.1/255.255.255.0
>
> read = all
>
> write = all
>
> ================================================================
>
> CONFIGURATION for
> /usr/lib/red53/webapps/konnectme/WEB-INF/classes/openmeetings-applicationContext.xml
>
>
>
> class="org.apache.openmeetings.db.dao.calendar.AppointmentCategoryDao" />
>
> <bean id="roommanagement"
> class="org.apache.openmeetings.data.conference.RoomManager" />
>
> <bean id="roomDao"
> class="org.apache.openmeetings.db.dao.room.RoomDao"/>
>
> <bean id="sipDao"
> class="org.apache.openmeetings.db.dao.room.SipDao">
>
> <!-- Should be uncommented and updated with real values for
> Asterisk -->
>
>
> <constructor-arg><value>127.0.0.1</value></constructor-arg>
>
>
> <constructor-arg><value>5038</value></constructor-arg>
>
>
> <constructor-arg><value>openmeetings</value></constructor-arg>
>
>
> <constructor-arg><value>12345</value></constructor-arg>
>
> ================================================================
>
> CONFIGURATION for /opt/red5sip/red5sip_3.0/settings.properties
>
> red5.host=127.0.0.1
>
> om.context=konnectme
>
> red5.codec=asao
>
> red5.codec.rate=22
>
> sip.obproxy=127.0.0.1
>
> sip.phone=red5sip_user
>
> sip.authid=red5sip_user
>
> sip.secret=12345
>
> sip.realm=asterisk
>
> sip.proxy=127.0.0.1
>
> rooms.forceStart=no
>
> rooms=1
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Wednesday, August 06, 2014 10:46 PM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Hello Horace,
>
>
>
> sorry for keeping silence, a little bit bit busy right now
>
> SIP transport set up the bridge from asterisk to red5 and performs
> audio/video transcoding rtp <->rtmp
>
>
>
> according to your issue it seems like creadentials specified in settings
> file are invalid for your Asterisk, can it be a problem?
>
> Will try to reproduce your problem as soon as i will get some time
>
>
>
> On 7 August 2014 02:53, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim,
>
> Perhaps if I knew exactly what sip transport does, I might be able to
> figure this out. Can you tell me what it is suppose to do..
>
>
>
>
>
> Miles
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, August 01, 2014 8:22 PM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Simple test if everything works is:
>
> 1) go to Admin->Conference rooms
>
> 2) select room
>
> 3) Check enable SIP
>
> 4) SIP number should appear in room panel (maybe after save)
>
>
>
> is it works for you?
>
>
>
> On 2 August 2014 00:36, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok found red5sip.enable value = yes
>
> Asterisk is configured to access openmeeting database through
> asterisk-connector
>
> Bean as been uncommented in openmeetings-application.xml and configure
> with matching values in asterisk/manager.conf
>
> I have re-saved all users in Openmeetings to recreate password hashes in
> asterisk
>
> Sip is enabled in rooms that have been created.
>
>
>
> I can telnet to localhost 5080 and 1935
>
>
>
> I am still having the following problems
>
> Sip Transport will not stay in the room pops in and out every two seconds
>
> It appears as though the sip transport can register but is unable to
> receive the invite message.
>
> In the extension.conf
>
> I get the following
>
> n -- Executing [40016@rooms-red5sip:1]
> GotoIf(“SIP/red5sip_user-000000a6”,”0?ok:notavail”) in new stack
>
> n -- Goto (rooms-red5sip,40016,3)
>
> n --Executing [40016@rooms-red5sip:3]
> Hangup(“/red5sip_user-000000a6”,””) in new stack
>
> n Spawn extension (rooms-red5sip, 40016,3) exited non-zero on
> ‘/red5sip_user-000000a6’
>
> It appears to check the database not find the room and then hang up.
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, August 01, 2014 10:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> you can search red5sip in config :)
>
> the key is "red5sip.enable"
>
>
>
> On 1 August 2014 23:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the response.
>
> I have confirmed everything but I am not sure where to find this setting.
> I am assuming Rootconfig is Openmeeting Admin->Configuration. If so I
> don’t a setting for Red5sip key.
>
> 3) red5sip* key should be enabled in Admin->Config – NOT SURE OF THIS STEP
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Wednesday, July 30, 2014 6:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> OM is accessible on all network interfaces by default
>
> config.xml need to be modified only in case you need to restrict OM client.
>
>
>
> According to red5sip enter-exit-enter-exit-.... it should be due to
> misconfiguration. Unfortunately this integration is not simple by design :(
> I'm using logs and debug to set it up properly.
>
>
>
> Main steps are
>
> 1) asterisk should be configured to have access to OM DB
>
> 2) asterisk bean should be uncommented and configured properly in
> openmeetings-application.xml
>
> 3) red5sip* key should be enabled in Admin->Config
>
> 4) in case asterisk is integrated with OM user should be re-saved (to have
> password-hash being saved in asterisk DB table)
>
> 5) sip should be enabled in the room
>
>
>
> this should be all (hope I haven't miss anything)
>
>
>
> On 29 July 2014 08:29, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Hi Maxim,
>
> My box is connected directly to a public IP, no NAT. My understanding
> was that Openmeetings to be access from the internet needed to be on a
> public address. That address would be the one in the config.xml. If I a
> mistaken let me know.
>
> 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 2:42 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Only with code modification
>
> On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
>
> Dear all,
>
> can i disable arrow pointer for all participants in restricted room on
> Whiteboard??
>
>
> --
> Regards,
> M K Raju.
>
>
>
>
>
> --
> 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: Pointer on WB
Posted by Horace Miles <Ho...@myit-solutions.com>.
Ok those credentials are correct.
My next question would be is how does the following line of code now what field in the openmeetings/rooms to lookup?
exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?notavail:ok)
I can see the database - open30
I can see the table – room
What is telling it what field to look at in the table?
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Sunday, August 10, 2014 9:47 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
DB credentials used are configured in /etc/odbc.ini
as described in http://openmeetings.apache.org/red5sip-integration_3.0.html
On 10 August 2014 22:09, Horace Miles <Ho...@myit-solutions.com> wrote:
I took the firewall down. Still had the same problem.
I reverse the logic in this line of the extensions.conf
exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?notavail:ok) >>>>>>>>>> reverse the notavail and OK.
It will then invoke the video bridge. Cant make a call but it shows me that it asterisk is not finding the record in openmeetings. I cant figure out wht account it is using to make the query. Openmeetings, root, or red5sip_user. I cant figure out the correlation of which is making the call to the openmeetings database.
Miles
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Sunday, August 10, 2014 3:36 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Maybe it is network configuration issue as described here: http://stackoverflow.com/questions/22093328/asterisk-sip-retransmission-timeout ?
can you check it with all firewalls disabled?
On 8 August 2014 23:51, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim,
Whenever you have time I understand. Here are all of my configurations by file name. I hope it will help.
HERE IS THE CONFIGURATION FOR EACH FILE FOR THE ASTERISK INTEGRATION. I HAVE SEPPERATED EACH SECTION WITH “==========”
HOPEFULLY SOMEONE CAN SEE THE ERROR I HAVE MADE
Miles
================================================================
CONFIGURATION for /etc/odbc.ini
[asterisk-connector]
Description = MySQL connection to 'openmeetings' database
Driver = MySQL
Database = open30
Server = 127.0.0.1
USER = root
PASSWORD =******
Port = 3306
Socket = /var/run/mysqld/mysqld.sock
================================================================
CONFIGURATION for /etc/odbcinst.ini
[MySQL]
Description = ODBC for MySQL
Driver = /usr/lib/i386-linux-gnu/odbc/libmyodbc.so
Setup = /usr/lib/i386-linux-gnu/odbc/libodbcmyS.so
FileUsage = 1
================================================================
CONFIGURATION for in /etc/asterisk/modules.conf
[modules]
autoload=yes
;
; Any modules that need to be loaded before the Asterisk core has been
; initialized (just after the logger has been initialized) can be loaded
; using 'preload'. This will frequently be needed if you wish to map all
; module configuration files into Realtime storage, since the Realtime
; driver will need to be loaded before the modules using those configuration
; files are initialized.
;
; An example of loading ODBC support would be:
preload => res_odbc.so
preload => res_config_odbc.so
================================================================
CONFIGURATION for /etc/asterisk/res_odbc.conf
;;; odbc setup file
; ENV is a global set of environmental variables that will get set.
; Note that all environmental variables can be seen by all connections,
; so you can't have different values for different connections.
[ENV]
;INFORMIXSERVER => my_special_database
;INFORMIXDIR => /opt/informix
;ORACLE_HOME => /home/oracle
; All other sections are arbitrary names for database connections.
;
; The context name is what will be used in other configuration files, such
; as extconfig.conf and func_odbc.conf, to reference this connection.
[asterisk]
;
; Permit disabling sections without needing to comment them out.
; If not specified, it is assumed the section is enabled.
enabled => yes
;
; This value should match an entry in /etc/odbc.ini
; (or /usr/local/etc/odbc.ini, on FreeBSD and similar systems).
dsn => asterisk-connector
;
; Username for connecting to the database. The user defaults to the context
; name if unspecified.
username => admin
;
; Password for authenticating the user to the database. The default
; password is blank.
password => ******
;
; Build a connection at startup?
pre-connect => yes
================================================================
Configuration for /etc/asterisk/sip.conf
;
;
; SIP Configuration example for Asterisk
;
; Note: Please read the security documentation for Asterisk in order to
; understand the risks of installing Asterisk with the sample
; configuration. If your Asterisk is installed on a public
; IP address connected to the Internet, you will want to learn
; about the various security settings BEFORE you start
; Asterisk.
;
; Especially note the following settings:
; - allowguest (default enabled)
; - permit/deny/acl - IP address filters
; - contactpermit/contactdeny/contactacl - IP address filters for registrations
; - context - Which set of services you offer various users
;
[general]
context=public ; Default context for incoming calls. Defaults to 'default'
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
realm=asterisk ; Realm for digest authentication
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
tcpenable=yes ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
transport=udp ; Set the default transports. The order determines the primary default transport.
; If tcpenable=no and the transport set is tcp, we will fallback to UDP.
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
maxexpiry=43200 ; Maximum allowed time of incoming registrations (seconds)
videosupport=yes ; Turn on support for SIP video. You need to turn this
rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
;domain=mydomain.tld,mydomain-incoming
; Add domain and configure incoming context
; for external calls to this domain
domain=127.0.0.1 ; Add IP address as local domain
domain=98.174.244.232 ; You can have several "domain" settings
[basic-options](!) ; a template
dtmfmode=rfc2833
context=from-office
type=friend
[natted-phone](!,basic-options) ; another template inheriting basic-options
directmedia=no
host=dynamic
[public-phone](!,basic-options) ; another template inheriting basic-options
directmedia=yes
[my-codecs](!) ; a template for my preferred codecs
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g723
allow=ulaw
; Or, more simply:
;allow=!all,ilbc,g729,gsm,g723,ulaw
[ulaw-phone](!) ; and another one for ulaw-only
disallow=all
allow=ulaw
; Again, more simply:
;allow=!all,ulaw
; and finally instantiate a few phones
;
; [2133](natted-phone,my-codecs)
; secret = peekaboo
; [2134](natted-phone,ulaw-phone)
; secret = not_very_secret
; [2136](public-phone,ulaw-phone)
; secret = not_very_secret_either
; ...
;
[red5sip_user]
type=friend
secret=12345
disallow=all
allow=ulaw
allow=h264
host=dynamic
nat=no
;nat=force_rport,comedia
context=rooms-red5sip
================================================================
CONFIGURATION FOR /etc/asterisk/extconfig.conf
;
; Static and realtime external configuration
; engine configuration
;
; See https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
; for basic table formatting information.
;
[settings]
sippeers => odbc,asterisk,sipusers
================================================================
CONFIGURATION FOR /etc/asterisk/extensions.conf
[rooms]
exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?ok:notavail)
exten => _400X!,n(ok),SET(PIN=${DB(open30/room/${EXTEN})})
exten => _400X!,n,Set(CONFBRIDGE(user,template)=sip_user)
exten => _400X!,n,Set(CONFBRIDGE(user,pin)=${PIN})
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,)
exten => _400X!,n,Hangup
exten => _400X!,n(notavail),Answer()
exten => _400X!,n,Playback(invalid)
exten => _400X!,n,Hangup
[rooms-originate]
exten => _400X!,1,Confbridge(${EXTEN},default_bridge,sip_user)
exten => _400X!,n,Hangup
[rooms-out]
; *****************************************************
; Extensions for outgoing calls from Openmeetings room.
; *****************************************************
[rooms-red5sip]
exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?ok:notavail)
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
exten => _400X!,n(notavail),Hangup
================================================================
CONFIGURATION for /etc/asterisk/confbridge.conf
[red5sip_user]
type=user
marked=yes
dsp_drop_silence=yes
denoise=true
[sip_user]
type=user
end_marked=yes
wait_marked=yes
music_on_hold_when_empty=yes
dsp_drop_silence=yes
denoise=true
[default_bridge]
type=bridge
video_mode=follow_talker
;video_mode=last_marked
;video_mode=first_marked
================================================================
CONFIGURATION /etc/asterisk/manager.conf
[general]
;enabled = no
;webenabled = yes
enabled = yes
webenabled = no
port = 5038
bindaddr = 127.0.0.1
[openmeetings]
secret = 12345
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = all
write = all
================================================================
CONFIGURATION for /usr/lib/red53/webapps/konnectme/WEB-INF/classes/openmeetings-applicationContext.xml
class="org.apache.openmeetings.db.dao.calendar.AppointmentCategoryDao" />
<bean id="roommanagement" class="org.apache.openmeetings.data.conference.RoomManager" />
<bean id="roomDao" class="org.apache.openmeetings.db.dao.room.RoomDao"/>
<bean id="sipDao" class="org.apache.openmeetings.db.dao.room.SipDao">
<!-- Should be uncommented and updated with real values for Asterisk -->
<constructor-arg><value>127.0.0.1</value></constructor-arg>
<constructor-arg><value>5038</value></constructor-arg>
<constructor-arg><value>openmeetings</value></constructor-arg>
<constructor-arg><value>12345</value></constructor-arg>
================================================================
CONFIGURATION for /opt/red5sip/red5sip_3.0/settings.properties
red5.host=127.0.0.1
om.context=konnectme
red5.codec=asao
red5.codec.rate=22
sip.obproxy=127.0.0.1
sip.phone=red5sip_user
sip.authid=red5sip_user
sip.secret=12345
sip.realm=asterisk
sip.proxy=127.0.0.1
rooms.forceStart=no
rooms=1
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Wednesday, August 06, 2014 10:46 PM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Hello Horace,
sorry for keeping silence, a little bit bit busy right now
SIP transport set up the bridge from asterisk to red5 and performs audio/video transcoding rtp <->rtmp
according to your issue it seems like creadentials specified in settings file are invalid for your Asterisk, can it be a problem?
Will try to reproduce your problem as soon as i will get some time
On 7 August 2014 02:53, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim,
Perhaps if I knew exactly what sip transport does, I might be able to figure this out. Can you tell me what it is suppose to do..
Miles
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, August 01, 2014 8:22 PM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Simple test if everything works is:
1) go to Admin->Conference rooms
2) select room
3) Check enable SIP
4) SIP number should appear in room panel (maybe after save)
is it works for you?
On 2 August 2014 00:36, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok found red5sip.enable value = yes
Asterisk is configured to access openmeeting database through asterisk-connector
Bean as been uncommented in openmeetings-application.xml and configure with matching values in asterisk/manager.conf
I have re-saved all users in Openmeetings to recreate password hashes in asterisk
Sip is enabled in rooms that have been created.
I can telnet to localhost 5080 and 1935
I am still having the following problems
Sip Transport will not stay in the room pops in and out every two seconds
It appears as though the sip transport can register but is unable to receive the invite message.
In the extension.conf
I get the following
n -- Executing [40016@rooms-red5sip:1] GotoIf(“SIP/red5sip_user-000000a6”,”0?ok:notavail”) in new stack
n -- Goto (rooms-red5sip,40016,3)
n --Executing [40016@rooms-red5sip:3] Hangup(“/red5sip_user-000000a6”,””) in new stack
n Spawn extension (rooms-red5sip, 40016,3) exited non-zero on ‘/red5sip_user-000000a6’
It appears to check the database not find the room and then hang up.
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, August 01, 2014 10:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
you can search red5sip in config :)
the key is "red5sip.enable"
On 1 August 2014 23:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the response.
I have confirmed everything but I am not sure where to find this setting. I am assuming Rootconfig is Openmeeting Admin->Configuration. If so I don’t a setting for Red5sip key.
3) red5sip* key should be enabled in Admin->Config – NOT SURE OF THIS STEP
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Wednesday, July 30, 2014 6:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
OM is accessible on all network interfaces by default
config.xml need to be modified only in case you need to restrict OM client.
According to red5sip enter-exit-enter-exit-.... it should be due to misconfiguration. Unfortunately this integration is not simple by design :( I'm using logs and debug to set it up properly.
Main steps are
1) asterisk should be configured to have access to OM DB
2) asterisk bean should be uncommented and configured properly in openmeetings-application.xml
3) red5sip* key should be enabled in Admin->Config
4) in case asterisk is integrated with OM user should be re-saved (to have password-hash being saved in asterisk DB table)
5) sip should be enabled in the room
this should be all (hope I haven't miss anything)
On 29 July 2014 08:29, Horace Miles <Ho...@myit-solutions.com> wrote:
Hi Maxim,
My box is connected directly to a public IP, no NAT. My understanding was that Openmeetings to be access from the internet needed to be on a public address. That address would be the one in the config.xml. If I a mistaken let me know.
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 2:42 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Only with code modification
On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
Dear all,
can i disable arrow pointer for all participants in restricted room on Whiteboard??
--
Regards,
M K Raju.
--
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: Pointer on WB
Posted by Maxim Solodovnik <so...@gmail.com>.
DB credentials used are configured in /etc/odbc.ini
as described in http://openmeetings.apache.org/red5sip-integration_3.0.html
On 10 August 2014 22:09, Horace Miles <Ho...@myit-solutions.com>
wrote:
> I took the firewall down. Still had the same problem.
>
>
>
> I reverse the logic in this line of the extensions.conf
>
> exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?notavail:ok)
> >>>>>>>>>> reverse the notavail and OK.
>
> It will then invoke the video bridge. Cant make a call but it shows me
> that it asterisk is not finding the record in openmeetings. I cant figure
> out wht account it is using to make the query. Openmeetings, root, or
> red5sip_user. I cant figure out the correlation of which is making the
> call to the openmeetings database.
>
>
>
> Miles
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Sunday, August 10, 2014 3:36 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Maybe it is network configuration issue as described here:
> http://stackoverflow.com/questions/22093328/asterisk-sip-retransmission-timeout
> ?
>
>
>
> can you check it with all firewalls disabled?
>
>
>
> On 8 August 2014 23:51, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim,
>
>
>
> Whenever you have time I understand. Here are all of my configurations by
> file name. I hope it will help.
>
> HERE IS THE CONFIGURATION FOR EACH FILE FOR THE ASTERISK INTEGRATION. I
> HAVE SEPPERATED EACH SECTION WITH “==========”
>
> HOPEFULLY SOMEONE CAN SEE THE ERROR I HAVE MADE
>
>
>
> Miles
>
> ================================================================
>
> CONFIGURATION for /etc/odbc.ini
>
> [asterisk-connector]
>
> Description = MySQL connection to 'openmeetings' database
>
> Driver = MySQL
>
> Database = open30
>
> Server = 127.0.0.1
>
> USER = root
>
> PASSWORD =******
>
> Port = 3306
>
> Socket = /var/run/mysqld/mysqld.sock
>
> ================================================================
>
> CONFIGURATION for /etc/odbcinst.ini
>
> [MySQL]
>
> Description = ODBC for MySQL
>
> Driver = /usr/lib/i386-linux-gnu/odbc/libmyodbc.so
>
> Setup = /usr/lib/i386-linux-gnu/odbc/libodbcmyS.so
>
> FileUsage = 1
>
> ================================================================
>
> CONFIGURATION for in /etc/asterisk/modules.conf
>
> [modules]
>
> autoload=yes
>
> ;
>
> ; Any modules that need to be loaded before the Asterisk core has been
>
> ; initialized (just after the logger has been initialized) can be loaded
>
> ; using 'preload'. This will frequently be needed if you wish to map all
>
> ; module configuration files into Realtime storage, since the Realtime
>
> ; driver will need to be loaded before the modules using those
> configuration
>
> ; files are initialized.
>
> ;
>
> ; An example of loading ODBC support would be:
>
> preload => res_odbc.so
>
> preload => res_config_odbc.so
>
> ================================================================
>
> CONFIGURATION for /etc/asterisk/res_odbc.conf
>
> ;;; odbc setup file
>
>
>
> ; ENV is a global set of environmental variables that will get set.
>
> ; Note that all environmental variables can be seen by all connections,
>
> ; so you can't have different values for different connections.
>
> [ENV]
>
> ;INFORMIXSERVER => my_special_database
>
> ;INFORMIXDIR => /opt/informix
>
> ;ORACLE_HOME => /home/oracle
>
>
>
> ; All other sections are arbitrary names for database connections.
>
>
>
> ;
>
> ; The context name is what will be used in other configuration files, such
>
> ; as extconfig.conf and func_odbc.conf, to reference this connection.
>
> [asterisk]
>
> ;
>
> ; Permit disabling sections without needing to comment them out.
>
> ; If not specified, it is assumed the section is enabled.
>
> enabled => yes
>
> ;
>
> ; This value should match an entry in /etc/odbc.ini
>
> ; (or /usr/local/etc/odbc.ini, on FreeBSD and similar systems).
>
> dsn => asterisk-connector
>
> ;
>
> ; Username for connecting to the database. The user defaults to the
> context
>
> ; name if unspecified.
>
> username => admin
>
> ;
>
> ; Password for authenticating the user to the database. The default
>
> ; password is blank.
>
> password => ******
>
> ;
>
> ; Build a connection at startup?
>
> pre-connect => yes
>
> ================================================================
>
> Configuration for /etc/asterisk/sip.conf
>
> ;
>
> ;
>
> ; SIP Configuration example for Asterisk
>
> ;
>
> ; Note: Please read the security documentation for Asterisk in order to
>
> ; understand the risks of installing Asterisk with the sample
>
> ; configuration. If your Asterisk is installed on a public
>
> ; IP address connected to the Internet, you will want to learn
>
> ; about the various security settings BEFORE you start
>
> ; Asterisk.
>
> ;
>
> ; Especially note the following settings:
>
> ; - allowguest (default enabled)
>
> ; - permit/deny/acl - IP address filters
>
> ; - contactpermit/contactdeny/contactacl - IP
> address filters for registrations
>
> ; - context - Which set of services you offer
> various users
>
> ;
>
>
>
> [general]
>
> context=public ; Default context for incoming calls.
> Defaults to 'default'
>
> allowoverlap=no ; Disable overlap dialing support.
> (Default is yes)
>
> realm=asterisk ; Realm for digest authentication
>
> udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to
> (0.0.0.0 binds to all)
>
> ; Optionally add a port number,
> 192.168.1.1:5062 (default is port 5060)
>
>
>
> tcpenable=yes ; Enable server for incoming TCP
> connections (default is no)
>
> tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to
> (0.0.0.0 binds to all interfaces)
>
> transport=udp ; Set the default transports. The order
> determines the primary default transport.
>
> ; If tcpenable=no and the transport set is
> tcp, we will fallback to UDP.
>
>
>
> srvlookup=yes ; Enable DNS SRV lookups on outbound calls
>
> maxexpiry=43200 ; Maximum allowed time of incoming
> registrations (seconds)
>
> videosupport=yes ; Turn on support for SIP video. You need
> to turn this
>
> rtcachefriends=yes ; Cache realtime friends by adding them to
> the internal list
>
>
>
> ;domain=mydomain.tld,mydomain-incoming
>
> ; Add domain and configure incoming context
>
> ; for external calls to this domain
>
> domain=127.0.0.1 ; Add IP address as local domain
>
> domain=98.174.244.232 ; You can have several "domain" settings
>
>
>
> [basic-options](!) ; a template
>
> dtmfmode=rfc2833
>
> context=from-office
>
> type=friend
>
>
>
> [natted-phone](!,basic-options) ; another template inheriting
> basic-options
>
> directmedia=no
>
> host=dynamic
>
>
>
> [public-phone](!,basic-options) ; another template inheriting
> basic-options
>
> directmedia=yes
>
>
>
> [my-codecs](!) ; a template for my preferred codecs
>
> disallow=all
>
> allow=ilbc
>
> allow=g729
>
> allow=gsm
>
> allow=g723
>
> allow=ulaw
>
> ; Or, more simply:
>
> ;allow=!all,ilbc,g729,gsm,g723,ulaw
>
>
>
> [ulaw-phone](!) ; and another one for ulaw-only
>
> disallow=all
>
> allow=ulaw
>
> ; Again, more simply:
>
> ;allow=!all,ulaw
>
>
>
> ; and finally instantiate a few phones
>
> ;
>
> ; [2133](natted-phone,my-codecs)
>
> ; secret = peekaboo
>
> ; [2134](natted-phone,ulaw-phone)
>
> ; secret = not_very_secret
>
> ; [2136](public-phone,ulaw-phone)
>
> ; secret = not_very_secret_either
>
> ; ...
>
> ;
>
> [red5sip_user]
>
> type=friend
>
> secret=12345
>
> disallow=all
>
> allow=ulaw
>
> allow=h264
>
> host=dynamic
>
> nat=no
>
> ;nat=force_rport,comedia
>
> context=rooms-red5sip
>
> ================================================================
>
> CONFIGURATION FOR /etc/asterisk/extconfig.conf
>
> ;
>
> ; Static and realtime external configuration
>
> ; engine configuration
>
> ;
>
> ; See
> https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
>
> ; for basic table formatting information.
>
> ;
>
> [settings]
>
> sippeers => odbc,asterisk,sipusers
>
> ================================================================
>
> CONFIGURATION FOR /etc/asterisk/extensions.conf
>
> [rooms]
>
> exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?ok:notavail)
>
> exten => _400X!,n(ok),SET(PIN=${DB(open30/room/${EXTEN})})
>
> exten => _400X!,n,Set(CONFBRIDGE(user,template)=sip_user)
>
> exten => _400X!,n,Set(CONFBRIDGE(user,pin)=${PIN})
>
> exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,)
>
> exten => _400X!,n,Hangup
>
> exten => _400X!,n(notavail),Answer()
>
> exten => _400X!,n,Playback(invalid)
>
> exten => _400X!,n,Hangup
>
>
>
> [rooms-originate]
>
> exten => _400X!,1,Confbridge(${EXTEN},default_bridge,sip_user)
>
> exten => _400X!,n,Hangup
>
>
>
> [rooms-out]
>
> ; *****************************************************
>
> ; Extensions for outgoing calls from Openmeetings room.
>
> ; *****************************************************
>
>
>
> [rooms-red5sip]
>
> exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?ok:notavail)
>
> exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
>
> exten => _400X!,n(notavail),Hangup
>
> ================================================================
>
> CONFIGURATION for /etc/asterisk/confbridge.conf
>
> [red5sip_user]
>
> type=user
>
> marked=yes
>
> dsp_drop_silence=yes
>
> denoise=true
>
>
>
> [sip_user]
>
> type=user
>
> end_marked=yes
>
> wait_marked=yes
>
> music_on_hold_when_empty=yes
>
> dsp_drop_silence=yes
>
> denoise=true
>
>
>
> [default_bridge]
>
> type=bridge
>
> video_mode=follow_talker
>
> ;video_mode=last_marked
>
> ;video_mode=first_marked
>
> ================================================================
>
> CONFIGURATION /etc/asterisk/manager.conf
>
> [general]
>
> ;enabled = no
>
> ;webenabled = yes
>
> enabled = yes
>
> webenabled = no
>
> port = 5038
>
> bindaddr = 127.0.0.1
>
>
>
> [openmeetings]
>
> secret = 12345
>
> deny=0.0.0.0/0.0.0.0
>
> permit=127.0.0.1/255.255.255.0
>
> read = all
>
> write = all
>
> ================================================================
>
> CONFIGURATION for
> /usr/lib/red53/webapps/konnectme/WEB-INF/classes/openmeetings-applicationContext.xml
>
>
>
> class="org.apache.openmeetings.db.dao.calendar.AppointmentCategoryDao" />
>
> <bean id="roommanagement"
> class="org.apache.openmeetings.data.conference.RoomManager" />
>
> <bean id="roomDao"
> class="org.apache.openmeetings.db.dao.room.RoomDao"/>
>
> <bean id="sipDao"
> class="org.apache.openmeetings.db.dao.room.SipDao">
>
> <!-- Should be uncommented and updated with real values for
> Asterisk -->
>
>
> <constructor-arg><value>127.0.0.1</value></constructor-arg>
>
>
> <constructor-arg><value>5038</value></constructor-arg>
>
>
> <constructor-arg><value>openmeetings</value></constructor-arg>
>
>
> <constructor-arg><value>12345</value></constructor-arg>
>
> ================================================================
>
> CONFIGURATION for /opt/red5sip/red5sip_3.0/settings.properties
>
> red5.host=127.0.0.1
>
> om.context=konnectme
>
> red5.codec=asao
>
> red5.codec.rate=22
>
> sip.obproxy=127.0.0.1
>
> sip.phone=red5sip_user
>
> sip.authid=red5sip_user
>
> sip.secret=12345
>
> sip.realm=asterisk
>
> sip.proxy=127.0.0.1
>
> rooms.forceStart=no
>
> rooms=1
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Wednesday, August 06, 2014 10:46 PM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Hello Horace,
>
>
>
> sorry for keeping silence, a little bit bit busy right now
>
> SIP transport set up the bridge from asterisk to red5 and performs
> audio/video transcoding rtp <->rtmp
>
>
>
> according to your issue it seems like creadentials specified in settings
> file are invalid for your Asterisk, can it be a problem?
>
> Will try to reproduce your problem as soon as i will get some time
>
>
>
> On 7 August 2014 02:53, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim,
>
> Perhaps if I knew exactly what sip transport does, I might be able to
> figure this out. Can you tell me what it is suppose to do..
>
>
>
>
>
> Miles
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, August 01, 2014 8:22 PM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Simple test if everything works is:
>
> 1) go to Admin->Conference rooms
>
> 2) select room
>
> 3) Check enable SIP
>
> 4) SIP number should appear in room panel (maybe after save)
>
>
>
> is it works for you?
>
>
>
> On 2 August 2014 00:36, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok found red5sip.enable value = yes
>
> Asterisk is configured to access openmeeting database through
> asterisk-connector
>
> Bean as been uncommented in openmeetings-application.xml and configure
> with matching values in asterisk/manager.conf
>
> I have re-saved all users in Openmeetings to recreate password hashes in
> asterisk
>
> Sip is enabled in rooms that have been created.
>
>
>
> I can telnet to localhost 5080 and 1935
>
>
>
> I am still having the following problems
>
> Sip Transport will not stay in the room pops in and out every two seconds
>
> It appears as though the sip transport can register but is unable to
> receive the invite message.
>
> In the extension.conf
>
> I get the following
>
> n -- Executing [40016@rooms-red5sip:1]
> GotoIf(“SIP/red5sip_user-000000a6”,”0?ok:notavail”) in new stack
>
> n -- Goto (rooms-red5sip,40016,3)
>
> n --Executing [40016@rooms-red5sip:3]
> Hangup(“/red5sip_user-000000a6”,””) in new stack
>
> n Spawn extension (rooms-red5sip, 40016,3) exited non-zero on
> ‘/red5sip_user-000000a6’
>
> It appears to check the database not find the room and then hang up.
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, August 01, 2014 10:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> you can search red5sip in config :)
>
> the key is "red5sip.enable"
>
>
>
> On 1 August 2014 23:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the response.
>
> I have confirmed everything but I am not sure where to find this setting.
> I am assuming Rootconfig is Openmeeting Admin->Configuration. If so I
> don’t a setting for Red5sip key.
>
> 3) red5sip* key should be enabled in Admin->Config – NOT SURE OF THIS STEP
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Wednesday, July 30, 2014 6:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> OM is accessible on all network interfaces by default
>
> config.xml need to be modified only in case you need to restrict OM client.
>
>
>
> According to red5sip enter-exit-enter-exit-.... it should be due to
> misconfiguration. Unfortunately this integration is not simple by design :(
> I'm using logs and debug to set it up properly.
>
>
>
> Main steps are
>
> 1) asterisk should be configured to have access to OM DB
>
> 2) asterisk bean should be uncommented and configured properly in
> openmeetings-application.xml
>
> 3) red5sip* key should be enabled in Admin->Config
>
> 4) in case asterisk is integrated with OM user should be re-saved (to have
> password-hash being saved in asterisk DB table)
>
> 5) sip should be enabled in the room
>
>
>
> this should be all (hope I haven't miss anything)
>
>
>
> On 29 July 2014 08:29, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Hi Maxim,
>
> My box is connected directly to a public IP, no NAT. My understanding
> was that Openmeetings to be access from the internet needed to be on a
> public address. That address would be the one in the config.xml. If I a
> mistaken let me know.
>
> 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 2:42 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Only with code modification
>
> On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
>
> Dear all,
>
> can i disable arrow pointer for all participants in restricted room on
> Whiteboard??
>
>
> --
> Regards,
> M K Raju.
>
>
>
>
>
> --
> 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: Pointer on WB
Posted by Horace Miles <Ho...@myit-solutions.com>.
I took the firewall down. Still had the same problem.
I reverse the logic in this line of the extensions.conf
exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?notavail:ok) >>>>>>>>>> reverse the notavail and OK.
It will then invoke the video bridge. Cant make a call but it shows me that it asterisk is not finding the record in openmeetings. I cant figure out wht account it is using to make the query. Openmeetings, root, or red5sip_user. I cant figure out the correlation of which is making the call to the openmeetings database.
Miles
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Sunday, August 10, 2014 3:36 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Maybe it is network configuration issue as described here: http://stackoverflow.com/questions/22093328/asterisk-sip-retransmission-timeout ?
can you check it with all firewalls disabled?
On 8 August 2014 23:51, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim,
Whenever you have time I understand. Here are all of my configurations by file name. I hope it will help.
HERE IS THE CONFIGURATION FOR EACH FILE FOR THE ASTERISK INTEGRATION. I HAVE SEPPERATED EACH SECTION WITH “==========”
HOPEFULLY SOMEONE CAN SEE THE ERROR I HAVE MADE
Miles
================================================================
CONFIGURATION for /etc/odbc.ini
[asterisk-connector]
Description = MySQL connection to 'openmeetings' database
Driver = MySQL
Database = open30
Server = 127.0.0.1
USER = root
PASSWORD =******
Port = 3306
Socket = /var/run/mysqld/mysqld.sock
================================================================
CONFIGURATION for /etc/odbcinst.ini
[MySQL]
Description = ODBC for MySQL
Driver = /usr/lib/i386-linux-gnu/odbc/libmyodbc.so
Setup = /usr/lib/i386-linux-gnu/odbc/libodbcmyS.so
FileUsage = 1
================================================================
CONFIGURATION for in /etc/asterisk/modules.conf
[modules]
autoload=yes
;
; Any modules that need to be loaded before the Asterisk core has been
; initialized (just after the logger has been initialized) can be loaded
; using 'preload'. This will frequently be needed if you wish to map all
; module configuration files into Realtime storage, since the Realtime
; driver will need to be loaded before the modules using those configuration
; files are initialized.
;
; An example of loading ODBC support would be:
preload => res_odbc.so
preload => res_config_odbc.so
================================================================
CONFIGURATION for /etc/asterisk/res_odbc.conf
;;; odbc setup file
; ENV is a global set of environmental variables that will get set.
; Note that all environmental variables can be seen by all connections,
; so you can't have different values for different connections.
[ENV]
;INFORMIXSERVER => my_special_database
;INFORMIXDIR => /opt/informix
;ORACLE_HOME => /home/oracle
; All other sections are arbitrary names for database connections.
;
; The context name is what will be used in other configuration files, such
; as extconfig.conf and func_odbc.conf, to reference this connection.
[asterisk]
;
; Permit disabling sections without needing to comment them out.
; If not specified, it is assumed the section is enabled.
enabled => yes
;
; This value should match an entry in /etc/odbc.ini
; (or /usr/local/etc/odbc.ini, on FreeBSD and similar systems).
dsn => asterisk-connector
;
; Username for connecting to the database. The user defaults to the context
; name if unspecified.
username => admin
;
; Password for authenticating the user to the database. The default
; password is blank.
password => ******
;
; Build a connection at startup?
pre-connect => yes
================================================================
Configuration for /etc/asterisk/sip.conf
;
;
; SIP Configuration example for Asterisk
;
; Note: Please read the security documentation for Asterisk in order to
; understand the risks of installing Asterisk with the sample
; configuration. If your Asterisk is installed on a public
; IP address connected to the Internet, you will want to learn
; about the various security settings BEFORE you start
; Asterisk.
;
; Especially note the following settings:
; - allowguest (default enabled)
; - permit/deny/acl - IP address filters
; - contactpermit/contactdeny/contactacl - IP address filters for registrations
; - context - Which set of services you offer various users
;
[general]
context=public ; Default context for incoming calls. Defaults to 'default'
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
realm=asterisk ; Realm for digest authentication
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
tcpenable=yes ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
transport=udp ; Set the default transports. The order determines the primary default transport.
; If tcpenable=no and the transport set is tcp, we will fallback to UDP.
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
maxexpiry=43200 ; Maximum allowed time of incoming registrations (seconds)
videosupport=yes ; Turn on support for SIP video. You need to turn this
rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
;domain=mydomain.tld,mydomain-incoming
; Add domain and configure incoming context
; for external calls to this domain
domain=127.0.0.1 ; Add IP address as local domain
domain=98.174.244.232 ; You can have several "domain" settings
[basic-options](!) ; a template
dtmfmode=rfc2833
context=from-office
type=friend
[natted-phone](!,basic-options) ; another template inheriting basic-options
directmedia=no
host=dynamic
[public-phone](!,basic-options) ; another template inheriting basic-options
directmedia=yes
[my-codecs](!) ; a template for my preferred codecs
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g723
allow=ulaw
; Or, more simply:
;allow=!all,ilbc,g729,gsm,g723,ulaw
[ulaw-phone](!) ; and another one for ulaw-only
disallow=all
allow=ulaw
; Again, more simply:
;allow=!all,ulaw
; and finally instantiate a few phones
;
; [2133](natted-phone,my-codecs)
; secret = peekaboo
; [2134](natted-phone,ulaw-phone)
; secret = not_very_secret
; [2136](public-phone,ulaw-phone)
; secret = not_very_secret_either
; ...
;
[red5sip_user]
type=friend
secret=12345
disallow=all
allow=ulaw
allow=h264
host=dynamic
nat=no
;nat=force_rport,comedia
context=rooms-red5sip
================================================================
CONFIGURATION FOR /etc/asterisk/extconfig.conf
;
; Static and realtime external configuration
; engine configuration
;
; See https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
; for basic table formatting information.
;
[settings]
sippeers => odbc,asterisk,sipusers
================================================================
CONFIGURATION FOR /etc/asterisk/extensions.conf
[rooms]
exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?ok:notavail)
exten => _400X!,n(ok),SET(PIN=${DB(open30/room/${EXTEN})})
exten => _400X!,n,Set(CONFBRIDGE(user,template)=sip_user)
exten => _400X!,n,Set(CONFBRIDGE(user,pin)=${PIN})
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,)
exten => _400X!,n,Hangup
exten => _400X!,n(notavail),Answer()
exten => _400X!,n,Playback(invalid)
exten => _400X!,n,Hangup
[rooms-originate]
exten => _400X!,1,Confbridge(${EXTEN},default_bridge,sip_user)
exten => _400X!,n,Hangup
[rooms-out]
; *****************************************************
; Extensions for outgoing calls from Openmeetings room.
; *****************************************************
[rooms-red5sip]
exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?ok:notavail)
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
exten => _400X!,n(notavail),Hangup
================================================================
CONFIGURATION for /etc/asterisk/confbridge.conf
[red5sip_user]
type=user
marked=yes
dsp_drop_silence=yes
denoise=true
[sip_user]
type=user
end_marked=yes
wait_marked=yes
music_on_hold_when_empty=yes
dsp_drop_silence=yes
denoise=true
[default_bridge]
type=bridge
video_mode=follow_talker
;video_mode=last_marked
;video_mode=first_marked
================================================================
CONFIGURATION /etc/asterisk/manager.conf
[general]
;enabled = no
;webenabled = yes
enabled = yes
webenabled = no
port = 5038
bindaddr = 127.0.0.1
[openmeetings]
secret = 12345
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = all
write = all
================================================================
CONFIGURATION for /usr/lib/red53/webapps/konnectme/WEB-INF/classes/openmeetings-applicationContext.xml
class="org.apache.openmeetings.db.dao.calendar.AppointmentCategoryDao" />
<bean id="roommanagement" class="org.apache.openmeetings.data.conference.RoomManager" />
<bean id="roomDao" class="org.apache.openmeetings.db.dao.room.RoomDao"/>
<bean id="sipDao" class="org.apache.openmeetings.db.dao.room.SipDao">
<!-- Should be uncommented and updated with real values for Asterisk -->
<constructor-arg><value>127.0.0.1</value></constructor-arg>
<constructor-arg><value>5038</value></constructor-arg>
<constructor-arg><value>openmeetings</value></constructor-arg>
<constructor-arg><value>12345</value></constructor-arg>
================================================================
CONFIGURATION for /opt/red5sip/red5sip_3.0/settings.properties
red5.host=127.0.0.1
om.context=konnectme
red5.codec=asao
red5.codec.rate=22
sip.obproxy=127.0.0.1
sip.phone=red5sip_user
sip.authid=red5sip_user
sip.secret=12345
sip.realm=asterisk
sip.proxy=127.0.0.1
rooms.forceStart=no
rooms=1
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Wednesday, August 06, 2014 10:46 PM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Hello Horace,
sorry for keeping silence, a little bit bit busy right now
SIP transport set up the bridge from asterisk to red5 and performs audio/video transcoding rtp <->rtmp
according to your issue it seems like creadentials specified in settings file are invalid for your Asterisk, can it be a problem?
Will try to reproduce your problem as soon as i will get some time
On 7 August 2014 02:53, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim,
Perhaps if I knew exactly what sip transport does, I might be able to figure this out. Can you tell me what it is suppose to do..
Miles
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, August 01, 2014 8:22 PM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Simple test if everything works is:
1) go to Admin->Conference rooms
2) select room
3) Check enable SIP
4) SIP number should appear in room panel (maybe after save)
is it works for you?
On 2 August 2014 00:36, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok found red5sip.enable value = yes
Asterisk is configured to access openmeeting database through asterisk-connector
Bean as been uncommented in openmeetings-application.xml and configure with matching values in asterisk/manager.conf
I have re-saved all users in Openmeetings to recreate password hashes in asterisk
Sip is enabled in rooms that have been created.
I can telnet to localhost 5080 and 1935
I am still having the following problems
Sip Transport will not stay in the room pops in and out every two seconds
It appears as though the sip transport can register but is unable to receive the invite message.
In the extension.conf
I get the following
n -- Executing [40016@rooms-red5sip:1] GotoIf(“SIP/red5sip_user-000000a6”,”0?ok:notavail”) in new stack
n -- Goto (rooms-red5sip,40016,3)
n --Executing [40016@rooms-red5sip:3] Hangup(“/red5sip_user-000000a6”,””) in new stack
n Spawn extension (rooms-red5sip, 40016,3) exited non-zero on ‘/red5sip_user-000000a6’
It appears to check the database not find the room and then hang up.
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, August 01, 2014 10:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
you can search red5sip in config :)
the key is "red5sip.enable"
On 1 August 2014 23:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the response.
I have confirmed everything but I am not sure where to find this setting. I am assuming Rootconfig is Openmeeting Admin->Configuration. If so I don’t a setting for Red5sip key.
3) red5sip* key should be enabled in Admin->Config – NOT SURE OF THIS STEP
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Wednesday, July 30, 2014 6:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
OM is accessible on all network interfaces by default
config.xml need to be modified only in case you need to restrict OM client.
According to red5sip enter-exit-enter-exit-.... it should be due to misconfiguration. Unfortunately this integration is not simple by design :( I'm using logs and debug to set it up properly.
Main steps are
1) asterisk should be configured to have access to OM DB
2) asterisk bean should be uncommented and configured properly in openmeetings-application.xml
3) red5sip* key should be enabled in Admin->Config
4) in case asterisk is integrated with OM user should be re-saved (to have password-hash being saved in asterisk DB table)
5) sip should be enabled in the room
this should be all (hope I haven't miss anything)
On 29 July 2014 08:29, Horace Miles <Ho...@myit-solutions.com> wrote:
Hi Maxim,
My box is connected directly to a public IP, no NAT. My understanding was that Openmeetings to be access from the internet needed to be on a public address. That address would be the one in the config.xml. If I a mistaken let me know.
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 2:42 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Only with code modification
On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
Dear all,
can i disable arrow pointer for all participants in restricted room on Whiteboard??
--
Regards,
M K Raju.
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: Pointer on WB
Posted by Maxim Solodovnik <so...@gmail.com>.
Maybe it is network configuration issue as described here:
http://stackoverflow.com/questions/22093328/asterisk-sip-retransmission-timeout
?
can you check it with all firewalls disabled?
On 8 August 2014 23:51, Horace Miles <Ho...@myit-solutions.com>
wrote:
> Maxim,
>
>
>
> Whenever you have time I understand. Here are all of my configurations by
> file name. I hope it will help.
>
> HERE IS THE CONFIGURATION FOR EACH FILE FOR THE ASTERISK INTEGRATION. I
> HAVE SEPPERATED EACH SECTION WITH “==========”
>
> HOPEFULLY SOMEONE CAN SEE THE ERROR I HAVE MADE
>
>
>
> Miles
>
> ================================================================
>
> CONFIGURATION for /etc/odbc.ini
>
> [asterisk-connector]
>
> Description = MySQL connection to 'openmeetings' database
>
> Driver = MySQL
>
> Database = open30
>
> Server = 127.0.0.1
>
> USER = root
>
> PASSWORD =******
>
> Port = 3306
>
> Socket = /var/run/mysqld/mysqld.sock
>
> ================================================================
>
> CONFIGURATION for /etc/odbcinst.ini
>
> [MySQL]
>
> Description = ODBC for MySQL
>
> Driver = /usr/lib/i386-linux-gnu/odbc/libmyodbc.so
>
> Setup = /usr/lib/i386-linux-gnu/odbc/libodbcmyS.so
>
> FileUsage = 1
>
> ================================================================
>
> CONFIGURATION for in /etc/asterisk/modules.conf
>
> [modules]
>
> autoload=yes
>
> ;
>
> ; Any modules that need to be loaded before the Asterisk core has been
>
> ; initialized (just after the logger has been initialized) can be loaded
>
> ; using 'preload'. This will frequently be needed if you wish to map all
>
> ; module configuration files into Realtime storage, since the Realtime
>
> ; driver will need to be loaded before the modules using those
> configuration
>
> ; files are initialized.
>
> ;
>
> ; An example of loading ODBC support would be:
>
> preload => res_odbc.so
>
> preload => res_config_odbc.so
>
> ================================================================
>
> CONFIGURATION for /etc/asterisk/res_odbc.conf
>
> ;;; odbc setup file
>
>
>
> ; ENV is a global set of environmental variables that will get set.
>
> ; Note that all environmental variables can be seen by all connections,
>
> ; so you can't have different values for different connections.
>
> [ENV]
>
> ;INFORMIXSERVER => my_special_database
>
> ;INFORMIXDIR => /opt/informix
>
> ;ORACLE_HOME => /home/oracle
>
>
>
> ; All other sections are arbitrary names for database connections.
>
>
>
> ;
>
> ; The context name is what will be used in other configuration files, such
>
> ; as extconfig.conf and func_odbc.conf, to reference this connection.
>
> [asterisk]
>
> ;
>
> ; Permit disabling sections without needing to comment them out.
>
> ; If not specified, it is assumed the section is enabled.
>
> enabled => yes
>
> ;
>
> ; This value should match an entry in /etc/odbc.ini
>
> ; (or /usr/local/etc/odbc.ini, on FreeBSD and similar systems).
>
> dsn => asterisk-connector
>
> ;
>
> ; Username for connecting to the database. The user defaults to the
> context
>
> ; name if unspecified.
>
> username => admin
>
> ;
>
> ; Password for authenticating the user to the database. The default
>
> ; password is blank.
>
> password => ******
>
> ;
>
> ; Build a connection at startup?
>
> pre-connect => yes
>
> ================================================================
>
> Configuration for /etc/asterisk/sip.conf
>
> ;
>
> ;
>
> ; SIP Configuration example for Asterisk
>
> ;
>
> ; Note: Please read the security documentation for Asterisk in order to
>
> ; understand the risks of installing Asterisk with the sample
>
> ; configuration. If your Asterisk is installed on a public
>
> ; IP address connected to the Internet, you will want to learn
>
> ; about the various security settings BEFORE you start
>
> ; Asterisk.
>
> ;
>
> ; Especially note the following settings:
>
> ; - allowguest (default enabled)
>
> ; - permit/deny/acl - IP address filters
>
> ; - contactpermit/contactdeny/contactacl - IP
> address filters for registrations
>
> ; - context - Which set of services you offer
> various users
>
> ;
>
>
>
> [general]
>
> context=public ; Default context for incoming calls.
> Defaults to 'default'
>
> allowoverlap=no ; Disable overlap dialing support.
> (Default is yes)
>
> realm=asterisk ; Realm for digest authentication
>
> udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to
> (0.0.0.0 binds to all)
>
> ; Optionally add a port number,
> 192.168.1.1:5062 (default is port 5060)
>
>
>
> tcpenable=yes ; Enable server for incoming TCP
> connections (default is no)
>
> tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to
> (0.0.0.0 binds to all interfaces)
>
> transport=udp ; Set the default transports. The order
> determines the primary default transport.
>
> ; If tcpenable=no and the transport set is
> tcp, we will fallback to UDP.
>
>
>
> srvlookup=yes ; Enable DNS SRV lookups on outbound calls
>
> maxexpiry=43200 ; Maximum allowed time of incoming
> registrations (seconds)
>
> videosupport=yes ; Turn on support for SIP video. You need
> to turn this
>
> rtcachefriends=yes ; Cache realtime friends by adding them to
> the internal list
>
>
>
> ;domain=mydomain.tld,mydomain-incoming
>
> ; Add domain and configure incoming context
>
> ; for external calls to this domain
>
> domain=127.0.0.1 ; Add IP address as local domain
>
> domain=98.174.244.232 ; You can have several "domain" settings
>
>
>
> [basic-options](!) ; a template
>
> dtmfmode=rfc2833
>
> context=from-office
>
> type=friend
>
>
>
> [natted-phone](!,basic-options) ; another template inheriting
> basic-options
>
> directmedia=no
>
> host=dynamic
>
>
>
> [public-phone](!,basic-options) ; another template inheriting
> basic-options
>
> directmedia=yes
>
>
>
> [my-codecs](!) ; a template for my preferred codecs
>
> disallow=all
>
> allow=ilbc
>
> allow=g729
>
> allow=gsm
>
> allow=g723
>
> allow=ulaw
>
> ; Or, more simply:
>
> ;allow=!all,ilbc,g729,gsm,g723,ulaw
>
>
>
> [ulaw-phone](!) ; and another one for ulaw-only
>
> disallow=all
>
> allow=ulaw
>
> ; Again, more simply:
>
> ;allow=!all,ulaw
>
>
>
> ; and finally instantiate a few phones
>
> ;
>
> ; [2133](natted-phone,my-codecs)
>
> ; secret = peekaboo
>
> ; [2134](natted-phone,ulaw-phone)
>
> ; secret = not_very_secret
>
> ; [2136](public-phone,ulaw-phone)
>
> ; secret = not_very_secret_either
>
> ; ...
>
> ;
>
> [red5sip_user]
>
> type=friend
>
> secret=12345
>
> disallow=all
>
> allow=ulaw
>
> allow=h264
>
> host=dynamic
>
> nat=no
>
> ;nat=force_rport,comedia
>
> context=rooms-red5sip
>
> ================================================================
>
> CONFIGURATION FOR /etc/asterisk/extconfig.conf
>
> ;
>
> ; Static and realtime external configuration
>
> ; engine configuration
>
> ;
>
> ; See
> https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
>
> ; for basic table formatting information.
>
> ;
>
> [settings]
>
> sippeers => odbc,asterisk,sipusers
>
> ================================================================
>
> CONFIGURATION FOR /etc/asterisk/extensions.conf
>
> [rooms]
>
> exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?ok:notavail)
>
> exten => _400X!,n(ok),SET(PIN=${DB(open30/room/${EXTEN})})
>
> exten => _400X!,n,Set(CONFBRIDGE(user,template)=sip_user)
>
> exten => _400X!,n,Set(CONFBRIDGE(user,pin)=${PIN})
>
> exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,)
>
> exten => _400X!,n,Hangup
>
> exten => _400X!,n(notavail),Answer()
>
> exten => _400X!,n,Playback(invalid)
>
> exten => _400X!,n,Hangup
>
>
>
> [rooms-originate]
>
> exten => _400X!,1,Confbridge(${EXTEN},default_bridge,sip_user)
>
> exten => _400X!,n,Hangup
>
>
>
> [rooms-out]
>
> ; *****************************************************
>
> ; Extensions for outgoing calls from Openmeetings room.
>
> ; *****************************************************
>
>
>
> [rooms-red5sip]
>
> exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?ok:notavail)
>
> exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
>
> exten => _400X!,n(notavail),Hangup
>
> ================================================================
>
> CONFIGURATION for /etc/asterisk/confbridge.conf
>
> [red5sip_user]
>
> type=user
>
> marked=yes
>
> dsp_drop_silence=yes
>
> denoise=true
>
>
>
> [sip_user]
>
> type=user
>
> end_marked=yes
>
> wait_marked=yes
>
> music_on_hold_when_empty=yes
>
> dsp_drop_silence=yes
>
> denoise=true
>
>
>
> [default_bridge]
>
> type=bridge
>
> video_mode=follow_talker
>
> ;video_mode=last_marked
>
> ;video_mode=first_marked
>
> ================================================================
>
> CONFIGURATION /etc/asterisk/manager.conf
>
> [general]
>
> ;enabled = no
>
> ;webenabled = yes
>
> enabled = yes
>
> webenabled = no
>
> port = 5038
>
> bindaddr = 127.0.0.1
>
>
>
> [openmeetings]
>
> secret = 12345
>
> deny=0.0.0.0/0.0.0.0
>
> permit=127.0.0.1/255.255.255.0
>
> read = all
>
> write = all
>
> ================================================================
>
> CONFIGURATION for
> /usr/lib/red53/webapps/konnectme/WEB-INF/classes/openmeetings-applicationContext.xml
>
>
>
> class="org.apache.openmeetings.db.dao.calendar.AppointmentCategoryDao" />
>
> <bean id="roommanagement"
> class="org.apache.openmeetings.data.conference.RoomManager" />
>
> <bean id="roomDao"
> class="org.apache.openmeetings.db.dao.room.RoomDao"/>
>
> <bean id="sipDao"
> class="org.apache.openmeetings.db.dao.room.SipDao">
>
> <!-- Should be uncommented and updated with real values for
> Asterisk -->
>
>
> <constructor-arg><value>127.0.0.1</value></constructor-arg>
>
>
> <constructor-arg><value>5038</value></constructor-arg>
>
>
> <constructor-arg><value>openmeetings</value></constructor-arg>
>
>
> <constructor-arg><value>12345</value></constructor-arg>
>
> ================================================================
>
> CONFIGURATION for /opt/red5sip/red5sip_3.0/settings.properties
>
> red5.host=127.0.0.1
>
> om.context=konnectme
>
> red5.codec=asao
>
> red5.codec.rate=22
>
> sip.obproxy=127.0.0.1
>
> sip.phone=red5sip_user
>
> sip.authid=red5sip_user
>
> sip.secret=12345
>
> sip.realm=asterisk
>
> sip.proxy=127.0.0.1
>
> rooms.forceStart=no
>
> rooms=1
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Wednesday, August 06, 2014 10:46 PM
>
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Hello Horace,
>
>
>
> sorry for keeping silence, a little bit bit busy right now
>
> SIP transport set up the bridge from asterisk to red5 and performs
> audio/video transcoding rtp <->rtmp
>
>
>
> according to your issue it seems like creadentials specified in settings
> file are invalid for your Asterisk, can it be a problem?
>
> Will try to reproduce your problem as soon as i will get some time
>
>
>
> On 7 August 2014 02:53, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim,
>
> Perhaps if I knew exactly what sip transport does, I might be able to
> figure this out. Can you tell me what it is suppose to do..
>
>
>
>
>
> Miles
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, August 01, 2014 8:22 PM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Simple test if everything works is:
>
> 1) go to Admin->Conference rooms
>
> 2) select room
>
> 3) Check enable SIP
>
> 4) SIP number should appear in room panel (maybe after save)
>
>
>
> is it works for you?
>
>
>
> On 2 August 2014 00:36, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok found red5sip.enable value = yes
>
> Asterisk is configured to access openmeeting database through
> asterisk-connector
>
> Bean as been uncommented in openmeetings-application.xml and configure
> with matching values in asterisk/manager.conf
>
> I have re-saved all users in Openmeetings to recreate password hashes in
> asterisk
>
> Sip is enabled in rooms that have been created.
>
>
>
> I can telnet to localhost 5080 and 1935
>
>
>
> I am still having the following problems
>
> Sip Transport will not stay in the room pops in and out every two seconds
>
> It appears as though the sip transport can register but is unable to
> receive the invite message.
>
> In the extension.conf
>
> I get the following
>
> n -- Executing [40016@rooms-red5sip:1]
> GotoIf(“SIP/red5sip_user-000000a6”,”0?ok:notavail”) in new stack
>
> n -- Goto (rooms-red5sip,40016,3)
>
> n --Executing [40016@rooms-red5sip:3]
> Hangup(“/red5sip_user-000000a6”,””) in new stack
>
> n Spawn extension (rooms-red5sip, 40016,3) exited non-zero on
> ‘/red5sip_user-000000a6’
>
> It appears to check the database not find the room and then hang up.
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, August 01, 2014 10:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> you can search red5sip in config :)
>
> the key is "red5sip.enable"
>
>
>
> On 1 August 2014 23:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the response.
>
> I have confirmed everything but I am not sure where to find this setting.
> I am assuming Rootconfig is Openmeeting Admin->Configuration. If so I
> don’t a setting for Red5sip key.
>
> 3) red5sip* key should be enabled in Admin->Config – NOT SURE OF THIS STEP
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Wednesday, July 30, 2014 6:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> OM is accessible on all network interfaces by default
>
> config.xml need to be modified only in case you need to restrict OM client.
>
>
>
> According to red5sip enter-exit-enter-exit-.... it should be due to
> misconfiguration. Unfortunately this integration is not simple by design :(
> I'm using logs and debug to set it up properly.
>
>
>
> Main steps are
>
> 1) asterisk should be configured to have access to OM DB
>
> 2) asterisk bean should be uncommented and configured properly in
> openmeetings-application.xml
>
> 3) red5sip* key should be enabled in Admin->Config
>
> 4) in case asterisk is integrated with OM user should be re-saved (to have
> password-hash being saved in asterisk DB table)
>
> 5) sip should be enabled in the room
>
>
>
> this should be all (hope I haven't miss anything)
>
>
>
> On 29 July 2014 08:29, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Hi Maxim,
>
> My box is connected directly to a public IP, no NAT. My understanding
> was that Openmeetings to be access from the internet needed to be on a
> public address. That address would be the one in the config.xml. If I a
> mistaken let me know.
>
> 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 2:42 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Only with code modification
>
> On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
>
> Dear all,
>
> can i disable arrow pointer for all participants in restricted room on
> Whiteboard??
>
>
> --
> Regards,
> M K Raju.
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: Pointer on WB
Posted by Horace Miles <Ho...@myit-solutions.com>.
Maxim,
Whenever you have time I understand. Here are all of my configurations by file name. I hope it will help.
HERE IS THE CONFIGURATION FOR EACH FILE FOR THE ASTERISK INTEGRATION. I HAVE SEPPERATED EACH SECTION WITH “==========”
HOPEFULLY SOMEONE CAN SEE THE ERROR I HAVE MADE
Miles
================================================================
CONFIGURATION for /etc/odbc.ini
[asterisk-connector]
Description = MySQL connection to 'openmeetings' database
Driver = MySQL
Database = open30
Server = 127.0.0.1
USER = root
PASSWORD =******
Port = 3306
Socket = /var/run/mysqld/mysqld.sock
================================================================
CONFIGURATION for /etc/odbcinst.ini
[MySQL]
Description = ODBC for MySQL
Driver = /usr/lib/i386-linux-gnu/odbc/libmyodbc.so
Setup = /usr/lib/i386-linux-gnu/odbc/libodbcmyS.so
FileUsage = 1
================================================================
CONFIGURATION for in /etc/asterisk/modules.conf
[modules]
autoload=yes
;
; Any modules that need to be loaded before the Asterisk core has been
; initialized (just after the logger has been initialized) can be loaded
; using 'preload'. This will frequently be needed if you wish to map all
; module configuration files into Realtime storage, since the Realtime
; driver will need to be loaded before the modules using those configuration
; files are initialized.
;
; An example of loading ODBC support would be:
preload => res_odbc.so
preload => res_config_odbc.so
================================================================
CONFIGURATION for /etc/asterisk/res_odbc.conf
;;; odbc setup file
; ENV is a global set of environmental variables that will get set.
; Note that all environmental variables can be seen by all connections,
; so you can't have different values for different connections.
[ENV]
;INFORMIXSERVER => my_special_database
;INFORMIXDIR => /opt/informix
;ORACLE_HOME => /home/oracle
; All other sections are arbitrary names for database connections.
;
; The context name is what will be used in other configuration files, such
; as extconfig.conf and func_odbc.conf, to reference this connection.
[asterisk]
;
; Permit disabling sections without needing to comment them out.
; If not specified, it is assumed the section is enabled.
enabled => yes
;
; This value should match an entry in /etc/odbc.ini
; (or /usr/local/etc/odbc.ini, on FreeBSD and similar systems).
dsn => asterisk-connector
;
; Username for connecting to the database. The user defaults to the context
; name if unspecified.
username => admin
;
; Password for authenticating the user to the database. The default
; password is blank.
password => ******
;
; Build a connection at startup?
pre-connect => yes
================================================================
Configuration for /etc/asterisk/sip.conf
;
;
; SIP Configuration example for Asterisk
;
; Note: Please read the security documentation for Asterisk in order to
; understand the risks of installing Asterisk with the sample
; configuration. If your Asterisk is installed on a public
; IP address connected to the Internet, you will want to learn
; about the various security settings BEFORE you start
; Asterisk.
;
; Especially note the following settings:
; - allowguest (default enabled)
; - permit/deny/acl - IP address filters
; - contactpermit/contactdeny/contactacl - IP address filters for registrations
; - context - Which set of services you offer various users
;
[general]
context=public ; Default context for incoming calls. Defaults to 'default'
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
realm=asterisk ; Realm for digest authentication
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
tcpenable=yes ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
transport=udp ; Set the default transports. The order determines the primary default transport.
; If tcpenable=no and the transport set is tcp, we will fallback to UDP.
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
maxexpiry=43200 ; Maximum allowed time of incoming registrations (seconds)
videosupport=yes ; Turn on support for SIP video. You need to turn this
rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
;domain=mydomain.tld,mydomain-incoming
; Add domain and configure incoming context
; for external calls to this domain
domain=127.0.0.1 ; Add IP address as local domain
domain=98.174.244.232 ; You can have several "domain" settings
[basic-options](!) ; a template
dtmfmode=rfc2833
context=from-office
type=friend
[natted-phone](!,basic-options) ; another template inheriting basic-options
directmedia=no
host=dynamic
[public-phone](!,basic-options) ; another template inheriting basic-options
directmedia=yes
[my-codecs](!) ; a template for my preferred codecs
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g723
allow=ulaw
; Or, more simply:
;allow=!all,ilbc,g729,gsm,g723,ulaw
[ulaw-phone](!) ; and another one for ulaw-only
disallow=all
allow=ulaw
; Again, more simply:
;allow=!all,ulaw
; and finally instantiate a few phones
;
; [2133](natted-phone,my-codecs)
; secret = peekaboo
; [2134](natted-phone,ulaw-phone)
; secret = not_very_secret
; [2136](public-phone,ulaw-phone)
; secret = not_very_secret_either
; ...
;
[red5sip_user]
type=friend
secret=12345
disallow=all
allow=ulaw
allow=h264
host=dynamic
nat=no
;nat=force_rport,comedia
context=rooms-red5sip
================================================================
CONFIGURATION FOR /etc/asterisk/extconfig.conf
;
; Static and realtime external configuration
; engine configuration
;
; See https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
; for basic table formatting information.
;
[settings]
sippeers => odbc,asterisk,sipusers
================================================================
CONFIGURATION FOR /etc/asterisk/extensions.conf
[rooms]
exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?ok:notavail)
exten => _400X!,n(ok),SET(PIN=${DB(open30/room/${EXTEN})})
exten => _400X!,n,Set(CONFBRIDGE(user,template)=sip_user)
exten => _400X!,n,Set(CONFBRIDGE(user,pin)=${PIN})
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,)
exten => _400X!,n,Hangup
exten => _400X!,n(notavail),Answer()
exten => _400X!,n,Playback(invalid)
exten => _400X!,n,Hangup
[rooms-originate]
exten => _400X!,1,Confbridge(${EXTEN},default_bridge,sip_user)
exten => _400X!,n,Hangup
[rooms-out]
; *****************************************************
; Extensions for outgoing calls from Openmeetings room.
; *****************************************************
[rooms-red5sip]
exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?ok:notavail)
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
exten => _400X!,n(notavail),Hangup
================================================================
CONFIGURATION for /etc/asterisk/confbridge.conf
[red5sip_user]
type=user
marked=yes
dsp_drop_silence=yes
denoise=true
[sip_user]
type=user
end_marked=yes
wait_marked=yes
music_on_hold_when_empty=yes
dsp_drop_silence=yes
denoise=true
[default_bridge]
type=bridge
video_mode=follow_talker
;video_mode=last_marked
;video_mode=first_marked
================================================================
CONFIGURATION /etc/asterisk/manager.conf
[general]
;enabled = no
;webenabled = yes
enabled = yes
webenabled = no
port = 5038
bindaddr = 127.0.0.1
[openmeetings]
secret = 12345
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = all
write = all
================================================================
CONFIGURATION for /usr/lib/red53/webapps/konnectme/WEB-INF/classes/openmeetings-applicationContext.xml
class="org.apache.openmeetings.db.dao.calendar.AppointmentCategoryDao" />
<bean id="roommanagement" class="org.apache.openmeetings.data.conference.RoomManager" />
<bean id="roomDao" class="org.apache.openmeetings.db.dao.room.RoomDao"/>
<bean id="sipDao" class="org.apache.openmeetings.db.dao.room.SipDao">
<!-- Should be uncommented and updated with real values for Asterisk -->
<constructor-arg><value>127.0.0.1</value></constructor-arg>
<constructor-arg><value>5038</value></constructor-arg>
<constructor-arg><value>openmeetings</value></constructor-arg>
<constructor-arg><value>12345</value></constructor-arg>
================================================================
CONFIGURATION for /opt/red5sip/red5sip_3.0/settings.properties
red5.host=127.0.0.1
om.context=konnectme
red5.codec=asao
red5.codec.rate=22
sip.obproxy=127.0.0.1
sip.phone=red5sip_user
sip.authid=red5sip_user
sip.secret=12345
sip.realm=asterisk
sip.proxy=127.0.0.1
rooms.forceStart=no
rooms=1
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Wednesday, August 06, 2014 10:46 PM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Hello Horace,
sorry for keeping silence, a little bit bit busy right now
SIP transport set up the bridge from asterisk to red5 and performs audio/video transcoding rtp <->rtmp
according to your issue it seems like creadentials specified in settings file are invalid for your Asterisk, can it be a problem?
Will try to reproduce your problem as soon as i will get some time
On 7 August 2014 02:53, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim,
Perhaps if I knew exactly what sip transport does, I might be able to figure this out. Can you tell me what it is suppose to do..
Miles
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, August 01, 2014 8:22 PM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Simple test if everything works is:
1) go to Admin->Conference rooms
2) select room
3) Check enable SIP
4) SIP number should appear in room panel (maybe after save)
is it works for you?
On 2 August 2014 00:36, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok found red5sip.enable value = yes
Asterisk is configured to access openmeeting database through asterisk-connector
Bean as been uncommented in openmeetings-application.xml and configure with matching values in asterisk/manager.conf
I have re-saved all users in Openmeetings to recreate password hashes in asterisk
Sip is enabled in rooms that have been created.
I can telnet to localhost 5080 and 1935
I am still having the following problems
Sip Transport will not stay in the room pops in and out every two seconds
It appears as though the sip transport can register but is unable to receive the invite message.
In the extension.conf
I get the following
n -- Executing [40016@rooms-red5sip:1] GotoIf(“SIP/red5sip_user-000000a6”,”0?ok:notavail”) in new stack
n -- Goto (rooms-red5sip,40016,3)
n --Executing [40016@rooms-red5sip:3] Hangup(“/red5sip_user-000000a6”,””) in new stack
n Spawn extension (rooms-red5sip, 40016,3) exited non-zero on ‘/red5sip_user-000000a6’
It appears to check the database not find the room and then hang up.
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, August 01, 2014 10:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
you can search red5sip in config :)
the key is "red5sip.enable"
On 1 August 2014 23:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the response.
I have confirmed everything but I am not sure where to find this setting. I am assuming Rootconfig is Openmeeting Admin->Configuration. If so I don’t a setting for Red5sip key.
3) red5sip* key should be enabled in Admin->Config – NOT SURE OF THIS STEP
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Wednesday, July 30, 2014 6:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
OM is accessible on all network interfaces by default
config.xml need to be modified only in case you need to restrict OM client.
According to red5sip enter-exit-enter-exit-.... it should be due to misconfiguration. Unfortunately this integration is not simple by design :( I'm using logs and debug to set it up properly.
Main steps are
1) asterisk should be configured to have access to OM DB
2) asterisk bean should be uncommented and configured properly in openmeetings-application.xml
3) red5sip* key should be enabled in Admin->Config
4) in case asterisk is integrated with OM user should be re-saved (to have password-hash being saved in asterisk DB table)
5) sip should be enabled in the room
this should be all (hope I haven't miss anything)
On 29 July 2014 08:29, Horace Miles <Ho...@myit-solutions.com> wrote:
Hi Maxim,
My box is connected directly to a public IP, no NAT. My understanding was that Openmeetings to be access from the internet needed to be on a public address. That address would be the one in the config.xml. If I a mistaken let me know.
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 2:42 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Only with code modification
On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
Dear all,
can i disable arrow pointer for all participants in restricted room on Whiteboard??
--
Regards,
M K Raju.
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
RE: Pointer on WB
Posted by Horace Miles <Ho...@myit-solutions.com>.
Maxim here is an entry out of the asterisk messages log:
Asterisk messages log
Aug 5 06:08:51] Asterisk 11.11.0 built by root @ vms on a i686 running Linux on 2014-07-26 19:25:45 UTC
[Aug 5 06:08:51] NOTICE[5128] loader.c: 2 modules will be loaded.
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: Connecting asterisk
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: res_odbc: Connected to asterisk [asterisk-connector]
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: Registered ODBC class 'asterisk' dsn->[asterisk-connector]
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: Connecting mysql2
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: res_odbc: Connected to mysql2 [asterisk-connector]
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: Registered ODBC class 'mysql2' dsn->[asterisk-connector]
[Aug 5 06:08:51] NOTICE[5128] res_odbc.c: res_odbc loaded.
[Aug 5 06:08:51] NOTICE[5128] config.c: Registered Config Engine odbc
As far as I can tell everything is stating that it is connected.
[rooms-red5sip]
exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?ok:notavil)
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)>>>>>>>>>>>>>>>>>>>>> This line never gets ran for the Confrencebridge
exten => _400X!,n(notavail),Hangup
As per the asterisk debug It appears to be checking the openmeetings database for the rooom and it does not find it. It then hangs up. Or am I looking at the wrong code for this? See Below.
What account is actually making the call to the openmeetings database? I thought it was the openmeetings user as configured in the manager.conf file?
Connected to Asterisk 11.11.0 currently running on vms (pid = 5128)
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [40016@rooms-red5sip:1] GotoIf("SIP/red5sip_user-00000005", "0?ok:notavail") in new stack “WHAT CREDENTIALS ARE BEING USED HERE”
-- Goto (rooms-red5sip,40016,3)
-- Executing [40016@rooms-red5sip:3] Hangup("SIP/red5sip_user-00000005", "") in new stack
== Spawn extension (rooms-red5sip, 40016, 3) exited non-zero on 'SIP/red5sip_user-00000005'
[Aug 5 06:14:19] WARNING[5164]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission 024312648651@127.0.1.1 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Wednesday, August 06, 2014 10:46 PM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Hello Horace,
sorry for keeping silence, a little bit bit busy right now
SIP transport set up the bridge from asterisk to red5 and performs audio/video transcoding rtp <->rtmp
according to your issue it seems like creadentials specified in settings file are invalid for your Asterisk, can it be a problem?
Will try to reproduce your problem as soon as i will get some time
On 7 August 2014 02:53, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim,
Perhaps if I knew exactly what sip transport does, I might be able to figure this out. Can you tell me what it is suppose to do..
Miles
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, August 01, 2014 8:22 PM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Simple test if everything works is:
1) go to Admin->Conference rooms
2) select room
3) Check enable SIP
4) SIP number should appear in room panel (maybe after save)
is it works for you?
On 2 August 2014 00:36, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok found red5sip.enable value = yes
Asterisk is configured to access openmeeting database through asterisk-connector
Bean as been uncommented in openmeetings-application.xml and configure with matching values in asterisk/manager.conf
I have re-saved all users in Openmeetings to recreate password hashes in asterisk
Sip is enabled in rooms that have been created.
I can telnet to localhost 5080 and 1935
I am still having the following problems
Sip Transport will not stay in the room pops in and out every two seconds
It appears as though the sip transport can register but is unable to receive the invite message.
In the extension.conf
I get the following
n -- Executing [40016@rooms-red5sip:1] GotoIf(“SIP/red5sip_user-000000a6”,”0?ok:notavail”) in new stack
n -- Goto (rooms-red5sip,40016,3)
n --Executing [40016@rooms-red5sip:3] Hangup(“/red5sip_user-000000a6”,””) in new stack
n Spawn extension (rooms-red5sip, 40016,3) exited non-zero on ‘/red5sip_user-000000a6’
It appears to check the database not find the room and then hang up.
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, August 01, 2014 10:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
you can search red5sip in config :)
the key is "red5sip.enable"
On 1 August 2014 23:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the response.
I have confirmed everything but I am not sure where to find this setting. I am assuming Admin config is Openmeeting Admin->Configuration. If so I don’t a setting for Red5sip key.
3) red5sip* key should be enabled in Admin->Config – NOT SURE OF THIS STEP
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Wednesday, July 30, 2014 6:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
OM is accessible on all network interfaces by default
config.xml need to be modified only in case you need to restrict OM client.
According to red5sip enter-exit-enter-exit-.... it should be due to misconfiguration. Unfortunately this integration is not simple by design :( I'm using logs and debug to set it up properly.
Main steps are
1) asterisk should be configured to have access to OM DB
2) asterisk bean should be uncommented and configured properly in openmeetings-application.xml
3) red5sip* key should be enabled in Admin->Config
4) in case asterisk is integrated with OM user should be re-saved (to have password-hash being saved in asterisk DB table)
5) sip should be enabled in the room
this should be all (hope I haven't miss anything)
On 29 July 2014 08:29, Horace Miles <Ho...@myit-solutions.com> wrote:
Hi Maxim,
My box is connected directly to a public IP, no NAT. My understanding was that Openmeetings to be access from the internet needed to be on a public address. That address would be the one in the config.xml. If I a mistaken let me know.
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 2:42 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Only with code modification
On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
Dear all,
can i disable arrow pointer for all participants in restricted room on Whiteboard??
--
Regards,
M K Raju.
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: Pointer on WB
Posted by Maxim Solodovnik <so...@gmail.com>.
Hello Horace,
sorry for keeping silence, a little bit bit busy right now
SIP transport set up the bridge from asterisk to red5 and performs
audio/video transcoding rtp <->rtmp
according to your issue it seems like creadentials specified in settings
file are invalid for your Asterisk, can it be a problem?
Will try to reproduce your problem as soon as i will get some time
On 7 August 2014 02:53, Horace Miles <Ho...@myit-solutions.com>
wrote:
> Maxim,
>
> Perhaps if I knew exactly what sip transport does, I might be able to
> figure this out. Can you tell me what it is suppose to do..
>
>
>
>
>
> Miles
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, August 01, 2014 8:22 PM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Simple test if everything works is:
>
> 1) go to Admin->Conference rooms
>
> 2) select room
>
> 3) Check enable SIP
>
> 4) SIP number should appear in room panel (maybe after save)
>
>
>
> is it works for you?
>
>
>
> On 2 August 2014 00:36, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Ok found red5sip.enable value = yes
>
> Asterisk is configured to access openmeeting database through
> asterisk-connector
>
> Bean as been uncommented in openmeetings-application.xml and configure
> with matching values in asterisk/manager.conf
>
> I have re-saved all users in Openmeetings to recreate password hashes in
> asterisk
>
> Sip is enabled in rooms that have been created.
>
>
>
> I can telnet to localhost 5080 and 1935
>
>
>
> I am still having the following problems
>
> Sip Transport will not stay in the room pops in and out every two seconds
>
> It appears as though the sip transport can register but is unable to
> receive the invite message.
>
> In the extension.conf
>
> I get the following
>
> n -- Executing [40016@rooms-red5sip:1]
> GotoIf(“SIP/red5sip_user-000000a6”,”0?ok:notavail”) in new stack
>
> n -- Goto (rooms-red5sip,40016,3)
>
> n --Executing [40016@rooms-red5sip:3]
> Hangup(“/red5sip_user-000000a6”,””) in new stack
>
> n Spawn extension (rooms-red5sip, 40016,3) exited non-zero on
> ‘/red5sip_user-000000a6’
>
> It appears to check the database not find the room and then hang up.
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, August 01, 2014 10:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> you can search red5sip in config :)
>
> the key is "red5sip.enable"
>
>
>
> On 1 August 2014 23:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the response.
>
> I have confirmed everything but I am not sure where to find this setting.
> I am assuming Admin config is Openmeeting Admin->Configuration. If so I
> don’t a setting for Red5sip key.
>
> 3) red5sip* key should be enabled in Admin->Config – NOT SURE OF THIS STEP
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Wednesday, July 30, 2014 6:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> OM is accessible on all network interfaces by default
>
> config.xml need to be modified only in case you need to restrict OM client.
>
>
>
> According to red5sip enter-exit-enter-exit-.... it should be due to
> misconfiguration. Unfortunately this integration is not simple by design :(
> I'm using logs and debug to set it up properly.
>
>
>
> Main steps are
>
> 1) asterisk should be configured to have access to OM DB
>
> 2) asterisk bean should be uncommented and configured properly in
> openmeetings-application.xml
>
> 3) red5sip* key should be enabled in Admin->Config
>
> 4) in case asterisk is integrated with OM user should be re-saved (to have
> password-hash being saved in asterisk DB table)
>
> 5) sip should be enabled in the room
>
>
>
> this should be all (hope I haven't miss anything)
>
>
>
> On 29 July 2014 08:29, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Hi Maxim,
>
> My box is connected directly to a public IP, no NAT. My understanding
> was that Openmeetings to be access from the internet needed to be on a
> public address. That address would be the one in the config.xml. If I a
> mistaken let me know.
>
> 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 2:42 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Only with code modification
>
> On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
>
> Dear all,
>
> can i disable arrow pointer for all participants in restricted room on
> Whiteboard??
>
>
> --
> Regards,
> M K Raju.
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: Pointer on WB
Posted by Horace Miles <Ho...@myit-solutions.com>.
Maxim,
Perhaps if I knew exactly what sip transport does, I might be able to figure this out. Can you tell me what it is suppose to do..
Miles
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, August 01, 2014 8:22 PM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Simple test if everything works is:
1) go to Admin->Conference rooms
2) select room
3) Check enable SIP
4) SIP number should appear in room panel (maybe after save)
is it works for you?
On 2 August 2014 00:36, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok found red5sip.enable value = yes
Asterisk is configured to access openmeeting database through asterisk-connector
Bean as been uncommented in openmeetings-application.xml and configure with matching values in asterisk/manager.conf
I have re-saved all users in Openmeetings to recreate password hashes in asterisk
Sip is enabled in rooms that have been created.
I can telnet to localhost 5080 and 1935
I am still having the following problems
Sip Transport will not stay in the room pops in and out every two seconds
It appears as though the sip transport can register but is unable to receive the invite message.
In the extension.conf
I get the following
n -- Executing [40016@rooms-red5sip:1] GotoIf(“SIP/red5sip_user-000000a6”,”0?ok:notavail”) in new stack
n -- Goto (rooms-red5sip,40016,3)
n --Executing [40016@rooms-red5sip:3] Hangup(“/red5sip_user-000000a6”,””) in new stack
n Spawn extension (rooms-red5sip, 40016,3) exited non-zero on ‘/red5sip_user-000000a6’
It appears to check the database not find the room and then hang up.
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, August 01, 2014 10:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
you can search red5sip in config :)
the key is "red5sip.enable"
On 1 August 2014 23:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the response.
I have confirmed everything but I am not sure where to find this setting. I am assuming Admin config is Openmeeting Admin->Configuration. If so I don’t a setting for Red5sip key.
3) red5sip* key should be enabled in Admin->Config – NOT SURE OF THIS STEP
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Wednesday, July 30, 2014 6:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
OM is accessible on all network interfaces by default
config.xml need to be modified only in case you need to restrict OM client.
According to red5sip enter-exit-enter-exit-.... it should be due to misconfiguration. Unfortunately this integration is not simple by design :( I'm using logs and debug to set it up properly.
Main steps are
1) asterisk should be configured to have access to OM DB
2) asterisk bean should be uncommented and configured properly in openmeetings-application.xml
3) red5sip* key should be enabled in Admin->Config
4) in case asterisk is integrated with OM user should be re-saved (to have password-hash being saved in asterisk DB table)
5) sip should be enabled in the room
this should be all (hope I haven't miss anything)
On 29 July 2014 08:29, Horace Miles <Ho...@myit-solutions.com> wrote:
Hi Maxim,
My box is connected directly to a public IP, no NAT. My understanding was that Openmeetings to be access from the internet needed to be on a public address. That address would be the one in the config.xml. If I a mistaken let me know.
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 2:42 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Only with code modification
On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
Dear all,
can i disable arrow pointer for all participants in restricted room on Whiteboard??
--
Regards,
M K Raju.
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
RE: Pointer on WB
Posted by Horace Miles <Ho...@myit-solutions.com>.
Maxim,
Yes I have always received the SIP number in the room. My problem is explained in the last email. SIP INVITE message being declined.
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, August 01, 2014 8:22 PM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Simple test if everything works is:
1) go to Admin->Conference rooms
2) select room
3) Check enable SIP
4) SIP number should appear in room panel (maybe after save)
is it works for you?
On 2 August 2014 00:36, Horace Miles <Ho...@myit-solutions.com> wrote:
Ok found red5sip.enable value = yes
Asterisk is configured to access openmeeting database through asterisk-connector
Bean as been uncommented in openmeetings-application.xml and configure with matching values in asterisk/manager.conf
I have re-saved all users in Openmeetings to recreate password hashes in asterisk
Sip is enabled in rooms that have been created.
I can telnet to localhost 5080 and 1935
I am still having the following problems
Sip Transport will not stay in the room pops in and out every two seconds
It appears as though the sip transport can register but is unable to receive the invite message.
In the extension.conf
I get the following
n -- Executing [40016@rooms-red5sip:1] GotoIf(“SIP/red5sip_user-000000a6”,”0?ok:notavail”) in new stack
n -- Goto (rooms-red5sip,40016,3)
n --Executing [40016@rooms-red5sip:3] Hangup(“/red5sip_user-000000a6”,””) in new stack
n Spawn extension (rooms-red5sip, 40016,3) exited non-zero on ‘/red5sip_user-000000a6’
It appears to check the database not find the room and then hang up.
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, August 01, 2014 10:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
you can search red5sip in config :)
the key is "red5sip.enable"
On 1 August 2014 23:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the response.
I have confirmed everything but I am not sure where to find this setting. I am assuming Admin config is Openmeeting Admin->Configuration. If so I don’t a setting for Red5sip key.
3) red5sip* key should be enabled in Admin->Config – NOT SURE OF THIS STEP
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Wednesday, July 30, 2014 6:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
OM is accessible on all network interfaces by default
config.xml need to be modified only in case you need to restrict OM client.
According to red5sip enter-exit-enter-exit-.... it should be due to misconfiguration. Unfortunately this integration is not simple by design :( I'm using logs and debug to set it up properly.
Main steps are
1) asterisk should be configured to have access to OM DB
2) asterisk bean should be uncommented and configured properly in openmeetings-application.xml
3) red5sip* key should be enabled in Admin->Config
4) in case asterisk is integrated with OM user should be re-saved (to have password-hash being saved in asterisk DB table)
5) sip should be enabled in the room
this should be all (hope I haven't miss anything)
On 29 July 2014 08:29, Horace Miles <Ho...@myit-solutions.com> wrote:
Hi Maxim,
My box is connected directly to a public IP, no NAT. My understanding was that Openmeetings to be access from the internet needed to be on a public address. That address would be the one in the config.xml. If I a mistaken let me know.
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 2:42 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Only with code modification
On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
Dear all,
can i disable arrow pointer for all participants in restricted room on Whiteboard??
--
Regards,
M K Raju.
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: Pointer on WB
Posted by Maxim Solodovnik <so...@gmail.com>.
Simple test if everything works is:
1) go to Admin->Conference rooms
2) select room
3) Check enable SIP
4) SIP number should appear in room panel (maybe after save)
is it works for you?
On 2 August 2014 00:36, Horace Miles <Ho...@myit-solutions.com>
wrote:
> Ok found red5sip.enable value = yes
>
> Asterisk is configured to access openmeeting database through
> asterisk-connector
>
> Bean as been uncommented in openmeetings-application.xml and configure
> with matching values in asterisk/manager.conf
>
> I have re-saved all users in Openmeetings to recreate password hashes in
> asterisk
>
> Sip is enabled in rooms that have been created.
>
>
>
> I can telnet to localhost 5080 and 1935
>
>
>
> I am still having the following problems
>
> Sip Transport will not stay in the room pops in and out every two seconds
>
> It appears as though the sip transport can register but is unable to
> receive the invite message.
>
> In the extension.conf
>
> I get the following
>
> n -- Executing [40016@rooms-red5sip:1]
> GotoIf(“SIP/red5sip_user-000000a6”,”0?ok:notavail”) in new stack
>
> n -- Goto (rooms-red5sip,40016,3)
>
> n --Executing [40016@rooms-red5sip:3]
> Hangup(“/red5sip_user-000000a6”,””) in new stack
>
> n Spawn extension (rooms-red5sip, 40016,3) exited non-zero on
> ‘/red5sip_user-000000a6’
>
> It appears to check the database not find the room and then hang up.
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Friday, August 01, 2014 10:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> you can search red5sip in config :)
>
> the key is "red5sip.enable"
>
>
>
> On 1 August 2014 23:48, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the response.
>
> I have confirmed everything but I am not sure where to find this setting.
> I am assuming Admin config is Openmeeting Admin->Configuration. If so I
> don’t a setting for Red5sip key.
>
> 3) red5sip* key should be enabled in Admin->Config – NOT SURE OF THIS STEP
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Wednesday, July 30, 2014 6:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> OM is accessible on all network interfaces by default
>
> config.xml need to be modified only in case you need to restrict OM client.
>
>
>
> According to red5sip enter-exit-enter-exit-.... it should be due to
> misconfiguration. Unfortunately this integration is not simple by design :(
> I'm using logs and debug to set it up properly.
>
>
>
> Main steps are
>
> 1) asterisk should be configured to have access to OM DB
>
> 2) asterisk bean should be uncommented and configured properly in
> openmeetings-application.xml
>
> 3) red5sip* key should be enabled in Admin->Config
>
> 4) in case asterisk is integrated with OM user should be re-saved (to have
> password-hash being saved in asterisk DB table)
>
> 5) sip should be enabled in the room
>
>
>
> this should be all (hope I haven't miss anything)
>
>
>
> On 29 July 2014 08:29, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Hi Maxim,
>
> My box is connected directly to a public IP, no NAT. My understanding
> was that Openmeetings to be access from the internet needed to be on a
> public address. That address would be the one in the config.xml. If I a
> mistaken let me know.
>
> 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 2:42 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Only with code modification
>
> On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
>
> Dear all,
>
> can i disable arrow pointer for all participants in restricted room on
> Whiteboard??
>
>
> --
> Regards,
> M K Raju.
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: Pointer on WB
Posted by Horace Miles <Ho...@myit-solutions.com>.
Ok found red5sip.enable value = yes
Asterisk is configured to access openmeeting database through asterisk-connector
Bean as been uncommented in openmeetings-application.xml and configure with matching values in asterisk/manager.conf
I have re-saved all users in Openmeetings to recreate password hashes in asterisk
Sip is enabled in rooms that have been created.
I can telnet to localhost 5080 and 1935
I am still having the following problems
Sip Transport will not stay in the room pops in and out every two seconds
It appears as though the sip transport can register but is unable to receive the invite message.
In the extension.conf
I get the following
n -- Executing [40016@rooms-red5sip:1] GotoIf(“SIP/red5sip_user-000000a6”,”0?ok:notavail”) in new stack
n -- Goto (rooms-red5sip,40016,3)
n --Executing [40016@rooms-red5sip:3] Hangup(“/red5sip_user-000000a6”,””) in new stack
n Spawn extension (rooms-red5sip, 40016,3) exited non-zero on ‘/red5sip_user-000000a6’
It appears to check the database not find the room and then hang up.
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Friday, August 01, 2014 10:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
you can search red5sip in config :)
the key is "red5sip.enable"
On 1 August 2014 23:48, Horace Miles <Ho...@myit-solutions.com> wrote:
Maxim thanks for the response.
I have confirmed everything but I am not sure where to find this setting. I am assuming Admin config is Openmeeting Admin->Configuration. If so I don’t a setting for Red5sip key.
3) red5sip* key should be enabled in Admin->Config – NOT SURE OF THIS STEP
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Wednesday, July 30, 2014 6:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
OM is accessible on all network interfaces by default
config.xml need to be modified only in case you need to restrict OM client.
According to red5sip enter-exit-enter-exit-.... it should be due to misconfiguration. Unfortunately this integration is not simple by design :( I'm using logs and debug to set it up properly.
Main steps are
1) asterisk should be configured to have access to OM DB
2) asterisk bean should be uncommented and configured properly in openmeetings-application.xml
3) red5sip* key should be enabled in Admin->Config
4) in case asterisk is integrated with OM user should be re-saved (to have password-hash being saved in asterisk DB table)
5) sip should be enabled in the room
this should be all (hope I haven't miss anything)
On 29 July 2014 08:29, Horace Miles <Ho...@myit-solutions.com> wrote:
Hi Maxim,
My box is connected directly to a public IP, no NAT. My understanding was that Openmeetings to be access from the internet needed to be on a public address. That address would be the one in the config.xml. If I a mistaken let me know.
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 2:42 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Only with code modification
On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
Dear all,
can i disable arrow pointer for all participants in restricted room on Whiteboard??
--
Regards,
M K Raju.
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: Pointer on WB
Posted by Maxim Solodovnik <so...@gmail.com>.
you can search red5sip in config :)
the key is "red5sip.enable"
On 1 August 2014 23:48, Horace Miles <Ho...@myit-solutions.com>
wrote:
> Maxim thanks for the response.
>
> I have confirmed everything but I am not sure where to find this setting.
> I am assuming Admin config is Openmeeting Admin->Configuration. If so I
> don’t a setting for Red5sip key.
>
> 3) red5sip* key should be enabled in Admin->Config – NOT SURE OF THIS STEP
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax666@gmail.com]
> *Sent:* Wednesday, July 30, 2014 6:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> OM is accessible on all network interfaces by default
>
> config.xml need to be modified only in case you need to restrict OM client.
>
>
>
> According to red5sip enter-exit-enter-exit-.... it should be due to
> misconfiguration. Unfortunately this integration is not simple by design :(
> I'm using logs and debug to set it up properly.
>
>
>
> Main steps are
>
> 1) asterisk should be configured to have access to OM DB
>
> 2) asterisk bean should be uncommented and configured properly in
> openmeetings-application.xml
>
> 3) red5sip* key should be enabled in Admin->Config
>
> 4) in case asterisk is integrated with OM user should be re-saved (to have
> password-hash being saved in asterisk DB table)
>
> 5) sip should be enabled in the room
>
>
>
> this should be all (hope I haven't miss anything)
>
>
>
> On 29 July 2014 08:29, Horace Miles <Ho...@myit-solutions.com>
> wrote:
>
> Hi Maxim,
>
> My box is connected directly to a public IP, no NAT. My understanding
> was that Openmeetings to be access from the internet needed to be on a
> public address. That address would be the one in the config.xml. If I a
> mistaken let me know.
>
> 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 2:42 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Only with code modification
>
> On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
>
> Dear all,
>
> can i disable arrow pointer for all participants in restricted room on
> Whiteboard??
>
>
> --
> Regards,
> M K Raju.
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
RE: Pointer on WB
Posted by Horace Miles <Ho...@myit-solutions.com>.
Maxim thanks for the response.
I have confirmed everything but I am not sure where to find this setting. I am assuming Admin config is Openmeeting Admin->Configuration. If so I don’t a setting for Red5sip key.
3) red5sip* key should be enabled in Admin->Config – NOT SURE OF THIS STEP
From: Maxim Solodovnik [mailto:solomax666@gmail.com]
Sent: Wednesday, July 30, 2014 6:07 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
OM is accessible on all network interfaces by default
config.xml need to be modified only in case you need to restrict OM client.
According to red5sip enter-exit-enter-exit-.... it should be due to misconfiguration. Unfortunately this integration is not simple by design :( I'm using logs and debug to set it up properly.
Main steps are
1) asterisk should be configured to have access to OM DB
2) asterisk bean should be uncommented and configured properly in openmeetings-application.xml
3) red5sip* key should be enabled in Admin->Config
4) in case asterisk is integrated with OM user should be re-saved (to have password-hash being saved in asterisk DB table)
5) sip should be enabled in the room
this should be all (hope I haven't miss anything)
On 29 July 2014 08:29, Horace Miles <Ho...@myit-solutions.com> wrote:
Hi Maxim,
My box is connected directly to a public IP, no NAT. My understanding was that Openmeetings to be access from the internet needed to be on a public address. That address would be the one in the config.xml. If I a mistaken let me know.
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 2:42 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Only with code modification
On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
Dear all,
can i disable arrow pointer for all participants in restricted room on Whiteboard??
--
Regards,
M K Raju.
--
WBR
Maxim aka solomax
Re: Pointer on WB
Posted by Maxim Solodovnik <so...@gmail.com>.
OM is accessible on all network interfaces by default
config.xml need to be modified only in case you need to restrict OM client.
According to red5sip enter-exit-enter-exit-.... it should be due to
misconfiguration. Unfortunately this integration is not simple by design :(
I'm using logs and debug to set it up properly.
Main steps are
1) asterisk should be configured to have access to OM DB
2) asterisk bean should be uncommented and configured properly in
openmeetings-application.xml
3) red5sip* key should be enabled in Admin->Config
4) in case asterisk is integrated with OM user should be re-saved (to have
password-hash being saved in asterisk DB table)
5) sip should be enabled in the room
this should be all (hope I haven't miss anything)
On 29 July 2014 08:29, Horace Miles <Ho...@myit-solutions.com> wrote:
> Hi Maxim,
>
> My box is connected directly to a public IP, no NAT. My understanding
> was that Openmeetings to be access from the internet needed to be on a
> public address. That address would be the one in the config.xml. If I a
> mistaken let me know.
>
> 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 2:42 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Only with code modification
>
> On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
>
> Dear all,
>
> can i disable arrow pointer for all participants in restricted room on
> Whiteboard??
>
>
> --
> Regards,
> M K Raju.
>
--
WBR
Maxim aka solomax
RE: Pointer on WB
Posted by Horace Miles <Ho...@myit-solutions.com>.
Hi Maxim,
My box is connected directly to a public IP, no NAT. My understanding was that Openmeetings to be access from the internet needed to be on a public address. That address would be the one in the config.xml. If I a mistaken let me know.
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 2:42 AM
To: Openmeetings user-list
Subject: Re: Pointer on WB
Only with code modification
On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
Dear all,
can i disable arrow pointer for all participants in restricted room on Whiteboard??
--
Regards,
M K Raju.
Re: Pointer on WB
Posted by Maxim Solodovnik <so...@gmail.com>.
Only with code modification
On Jul 24, 2014 4:40 PM, "Raju M K" <mk...@gmail.com> wrote:
> Dear all,
> can i disable arrow pointer for all participants in restricted room on
> Whiteboard??
>
>
> --
> Regards,
> M K Raju.
>
>