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Posted to user@openmeetings.apache.org by Bart Coninckx <ba...@telenet.be> on 2013/01/28 22:36:29 UTC
SIP connectivity
Hi,
I noticed some documentation on how to connect OM with a SIP proxy or
server, more particularly with the MeetMe application in Asterisk.
The exact goal or purpose is not mentionned however. Will OM callout to
a MeetMe conference? Or is it the other way round?
Cheers,
Bc
Re: SIP connectivity
Posted by Maxim Solodovnik <so...@gmail.com>.
I'm not sure if this is possible.
I guess it is easier to tell asterisk to use several databases than OM :(
On Thu, Jan 31, 2013 at 8:27 PM, Vieri <re...@yahoo.com> wrote:
> I have this in asterisk:
>
> TABLE meetme (
> bookid int(11) auto_increment,
> confno char(80) DEFAULT '0' NOT NULL,
> starttime datetime default '1900-01-01 12:00:00',
> endtime datetime default '2038-01-01 12:00:00',
> pin char(20) NULL,
> adminpin char(20) NULL,
> opts char(20) NULL,
> adminopts char(20) NULL,
> recordingfilename char(80) NULL,
> recordingformat char(10) NULL,
> maxusers int(11) NULL,
> members integer DEFAULT 0 NOT NULL,
> index confno (confno,starttime,endtime),
> PRIMARY KEY (bookid)
>
> By the way, the OM-Asterisk guide assumes that Asterisk realtime will link
> to the openmeetings database (to the meetme table and maybe also to the
> sipusers table - not sure). Could it be the other way around? ie. let
> openmeetings read/write to the meetme and/or sipusers table in an
> "asterisk" database but all other tables are left within the openmeetings
> database.
>
> Vieri
>
> --- On *Thu, 1/31/13, Maxim Solodovnik <so...@gmail.com>* wrote:
>
>
> From: Maxim Solodovnik <so...@gmail.com>
> Subject: Re: SIP connectivity
> To: "Bart Coninckx" <ba...@telenet.be>
> Cc: "user" <us...@openmeetings.apache.org>
> Date: Thursday, January 31, 2013, 7:05 AM
>
>
> Hello Bart,
>
> I just take a look at your URL ...
> OM does not create/use sipfriends DB table (at least from version 2.1)
> only meetme table is used
>
> so I'm afraid there is nothing to change here
>
> Here is the most recent instruction:
> http://openmeetings.apache.org/red5sip-integration_2.1.html
>
> Will ask our SIP guru to review it one more time :)
>
>
>
> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>
> > wrote:
>
> OK will add it and notify you
> On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>>
> wrote:
>
> It is for Asterisk 11 - don't know for other versions. You probably have
> no issues because of the 1.8 version. To be sure the .sql files in the
> Asterisk source should be compared across versions.
>
> this one is missing:
>
> `useragent` varchar(20) DEFAULT NULL,
>
> complete list (I think) is on:
> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>
>
> If I bump into others, I'll report ASAP,
>
>
> BC
>
>
>
> On 01/31/13 06:21, Maxim Solodovnik wrote:
>
> Is the OM meetme table incomplete?
> My asterisk reports no issues :(
>
> could you provide me with missing fields and I'll add it.
> My purpose was to create table with required fields only.
>
>
> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <ba...@telenet.be>
> > wrote:
>
> Openmeetings installed them for me, that's why I ended up with those.
> Using the Asterisk ones makes more sense to me. Maybe it's a good idea to
> have 'em removed from the install procedure.
>
> BC
>
>
> On 01/30/13 22:30, Jeff Clay wrote:
>
> Bart,
>
>
>
> If you look in the source directory of your asterisk tar file, under
> contrib/realtime/mysql you’ll find the .sql files required for all the
> realtime drivers. I never thought to use the ones with OM.
>
>
>
> Jeff Clay
>
> Network Administrator
>
> Infotech Enterprises America
>
> 870-215-5506
>
> Ext. 1506
>
>
>
> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ht...@telenet.be>]
>
> *Sent:* Wednesday, January 30, 2013 3:19 PM
> *To:* user@openmeetings.apache.org<ht...@openmeetings.apache.org>
> *Cc:* Jeff Clay
> *Subject:* Re: SIP connectivity
>
>
>
> Well,
>
> I might have found one difference though:
>
>
> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
> dictates how the table should look like. I obviously used the one in the
> openmeetings mysql database, but this one seems to miss the table
> "useragent". I discovered this because it showed up in the logfiles.
>
> BC
>
> On 01/29/13 14:41, Jeff Clay wrote:
>
> Bart,
>
>
>
> From an asterisk configuration standpoint there are very few differences
> between 1.8.x and 11.x. If memory serves, the only major changes that I ran
> into (in my production environment) was changes to SIP NAT values and the
> behavior of app_page() now uses confbridge instead of meetme to mix the
> audio. Also, TCP, TLS and app_confbridge got a major overhauling. There
> were of course many other changes and bug fixes, you can skim through the
> change log for full details, but I think that was the jist of it.
>
>
>
>
>
>
>
> Jeff Clay
>
> Network Administrator
>
> Infotech Enterprises America
>
> 870-215-5506
>
> Ext. 1506
>
>
>
> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ht...@telenet.be>]
>
> *Sent:* Tuesday, January 29, 2013 4:02 AM
> *To:* Maxim Solodovnik
> *Cc:* user
> *Subject:* Re: SIP connectivity
>
>
>
> I see - I'm willing to try the 11 version in the next fiew days if
> desired.
>
> BC
>
>
> On 01/29/13 10:57, Maxim Solodovnik wrote:
>
> I test the integration using
>
> Asterisk 1.8.13.1 (Ubuntu 12.10)
>
>
>
> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <ba...@telenet.be>>
> wrote:
>
> That is amazing - I initially tried to do the same thing by using the new
> chan_motif driver in Asterisk 11 which connects to a XMPP server.
>
> Are you guys using Asterisk 11? This version is the newest LTS version and
> has the best video capabilities.
>
> Cheers,
>
> BC
>
>
> On 01/29/13 02:44, Maxim Solodovnik wrote:
>
> red5sip will create special OM user in the room: "SIP Transport"
>
> after that you can call to the OM room using SIP hard or soft phone.
>
>
>
> We are currently testing it and trying to add video capabilities ...
>
>
>
> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <ba...@telenet.be>>
> wrote:
>
> Hi Jeff,
>
> In fact, I saw both pages, but none explain what they set up to do, just a
> bunch of command line instructions are given.
> Your "OM will create a meetme meeting as configured in the realtime meetme
> database" actually says it all in one go :-)
>
> cheers,
>
> BC
>
>
>
>
> On 01/28/13 22:38, Jeff Clay wrote:
>
> Bart,
>
> OM will create a meetme meeting as configured in the realtime meetme
> database. Have you read this page
> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
> http://openmeetings.apache.org/red5sip-integration.html but I assume this
> is the one you're already referring to.
>
> Jeff Clay
> Network Administrator
> Infotech Enterprises America
> 870-215-5506
> Ext. 1506
>
> -----Original Message-----
> From: Bart Coninckx [mailto:bart.coninckx@telenet.be<ht...@telenet.be>
> ]
> Sent: Monday, January 28, 2013 3:36 PM
> To: user@openmeetings.apache.org<ht...@openmeetings.apache.org>
> Subject: SIP connectivity
>
> Hi,
>
> I noticed some documentation on how to connect OM with a SIP proxy or
> server, more particularly with the MeetMe application in Asterisk.
>
> The exact goal or purpose is not mentionned however. Will OM callout to a
> MeetMe conference? Or is it the other way round?
>
>
> Cheers,
>
> Bc
>
> ________________________________
>
> DISCLAIMER:
>
> This email may contain confidential information and is intended only for
> the use of the specific individual(s) to which it is addressed. If you are
> not the intended recipient of this email, you are hereby notified that any
> unauthorized use, dissemination or copying of this email or the information
> contained in it or attached to it is strictly prohibited. If you received
> this message in error, please immediately notify the sender at Infotech and
> delete the original message.
>
>
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
--
WBR
Maxim aka solomax
AW: SIP connectivity
Posted by "Naderi, Sascha" <SN...@datus.com>.
Dear Jeff, dear all,
not the asterisk but the red5sip. I get the following error message from red5sip after the directory name (url) of openmeetings was changed.
13 Feb 08:29:39 - [INFO ] o.r.s.n.r.BaseRTMPClientHandler: rtmp://127.0.0.1:1935/openmeetings/0
13 Feb 08:29:39 - [INFO ] o.r.s.n.r.c.RTMPProtocolDecoder: Action _result
13 Feb 08:29:39 - [INFO ] o.r.s.a.RTMPControlClient: service call result: Service: null Method: connect No params
13 Feb 08:29:39 - [INFO ] o.r.s.a.RTMPControlClient: connect
13 Feb 08:29:39 - [ERROR] o.r.s.n.r.BaseRTMPHandler: Error while executing callback org.red5.sip.app.Application$2@3debe8ab<ma...@3debe8ab> java.lang.IllegalThreadStateException
13 Feb 08:29:39 - [WARN ] o.r.s.n.r.RTMPMinaIoHandler: Exception caught Connection reset by peer
13 Feb 08:29:39 - [INFO ] o.r.s.a.RTMPControlClient: service call result: Service: null Method: getActiveRoomIds No params
13 Feb 08:29:39 - [INFO ] o.r.s.a.RTMPControlClient: getActiveRoomIds
Regards
Sascha
________________________________
Von: Jeff Clay [Jeff.Clay@infotech-enterprises.com]
Gesendet: Mittwoch, 13. Februar 2013 21:02
Bis: user@openmeetings.apache.org
Cc: Maxim Solodovnik [solomax666@gmail.com]
Betreff: RE: SIP connectivity
I do not believe that the asterisk context is related to the url of openmeetings.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
From: Naderi, Sascha [mailto:SNaderi@datus.com]
Sent: Wednesday, February 13, 2013 2:00 PM
To: user@openmeetings.apache.org
Cc: Maxim Solodovnik [solomax666@gmail.com]
Subject: Re: SIP connectivity
Dear all,
i have tested the asterisk sip integration as documented with the most recent instruction (http://openmeetings.apache.org/red5sip-integration_2.1.html) and it works just fine.
The only thing i am missing is a way to get this working when i choose to rename the openmeetings context from http://yourcorp.com:5080/openmeetings to http://yourcorp.com:5080/yourmeetings
Which settings do i have to modify so that red5sip functions even if the context name is changed?
Regards
Sascha Naderi
________________________________
Von: Maxim Solodovnik [solomax666@gmail.com]
Gesendet: Samstag, 9. Februar 2013 02:32
Bis: Bart Coninckx
Cc: user
Betreff: Re: SIP connectivity
All tables are created by OM automatically
On Feb 9, 2013 5:46 AM, "Bart Coninckx" <ba...@telenet.be>> wrote:
May I add that a portion is missing, since one explains how to configure Asterisk for Realtime, but one does not stipulate how to create the necessary tables.
It's in my CentOS docs however (which I hope to post shortly).
BC
On 01/31/13 13:05, Maxim Solodovnik wrote:
Hello Bart,
I just take a look at your URL ...
OM does not create/use sipfriends DB table (at least from version 2.1)
only meetme table is used
so I'm afraid there is nothing to change here
Here is the most recent instruction:
http://openmeetings.apache.org/red5sip-integration_2.1.html
Will ask our SIP guru to review it one more time :)
On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>> wrote:
OK will add it and notify you
On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>> wrote:
It is for Asterisk 11 - don't know for other versions. You probably have no issues because of the 1.8 version. To be sure the .sql files in the Asterisk source should be compared across versions.
this one is missing:
`useragent` varchar(20) DEFAULT NULL,
complete list (I think) is on:
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
If I bump into others, I'll report ASAP,
BC
On 01/31/13 06:21, Maxim Solodovnik wrote:
Is the OM meetme table incomplete?
My asterisk reports no issues :(
could you provide me with missing fields and I'll add it.
My purpose was to create table with required fields only.
On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <ba...@telenet.be>> wrote:
Openmeetings installed them for me, that's why I ended up with those. Using the Asterisk ones makes more sense to me. Maybe it's a good idea to have 'em removed from the install procedure.
BC
On 01/30/13 22:30, Jeff Clay wrote:
Bart,
If you look in the source directory of your asterisk tar file, under contrib/realtime/mysql you’ll find the .sql files required for all the realtime drivers. I never thought to use the ones with OM.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
Sent: Wednesday, January 30, 2013 3:19 PM
To: user@openmeetings.apache.org<ma...@openmeetings.apache.org>
Cc: Jeff Clay
Subject: Re: SIP connectivity
Well,
I might have found one difference though:
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure dictates how the table should look like. I obviously used the one in the openmeetings mysql database, but this one seems to miss the table "useragent". I discovered this because it showed up in the logfiles.
BC
On 01/29/13 14:41, Jeff Clay wrote:
Bart,
>From an asterisk configuration standpoint there are very few differences between 1.8.x and 11.x. If memory serves, the only major changes that I ran into (in my production environment) was changes to SIP NAT values and the behavior of app_page() now uses confbridge instead of meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major overhauling. There were of course many other changes and bug fixes, you can skim through the change log for full details, but I think that was the jist of it.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
Sent: Tuesday, January 29, 2013 4:02 AM
To: Maxim Solodovnik
Cc: user
Subject: Re: SIP connectivity
I see - I'm willing to try the 11 version in the next fiew days if desired.
BC
On 01/29/13 10:57, Maxim Solodovnik wrote:
I test the integration using
Asterisk 1.8.13.1 (Ubuntu 12.10)
On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <ba...@telenet.be>> wrote:
That is amazing - I initially tried to do the same thing by using the new chan_motif driver in Asterisk 11 which connects to a XMPP server.
Are you guys using Asterisk 11? This version is the newest LTS version and has the best video capabilities.
Cheers,
BC
On 01/29/13 02:44, Maxim Solodovnik wrote:
red5sip will create special OM user in the room: "SIP Transport"
after that you can call to the OM room using SIP hard or soft phone.
We are currently testing it and trying to add video capabilities ...
On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <ba...@telenet.be>> wrote:
Hi Jeff,
In fact, I saw both pages, but none explain what they set up to do, just a bunch of command line instructions are given.
Your "OM will create a meetme meeting as configured in the realtime meetme database" actually says it all in one go :-)
cheers,
BC
On 01/28/13 22:38, Jeff Clay wrote:
Bart,
OM will create a meetme meeting as configured in the realtime meetme database. Have you read this page https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out http://openmeetings.apache.org/red5sip-integration.html but I assume this is the one you're already referring to.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
-----Original Message-----
From: Bart Coninckx [mailto:bart.coninckx@telenet.be<ma...@telenet.be>]
Sent: Monday, January 28, 2013 3:36 PM
To: user@openmeetings.apache.org<ma...@openmeetings.apache.org>
Subject: SIP connectivity
Hi,
I noticed some documentation on how to connect OM with a SIP proxy or server, more particularly with the MeetMe application in Asterisk.
The exact goal or purpose is not mentionned however. Will OM callout to a MeetMe conference? Or is it the other way round?
Cheers,
Bc
________________________________
DISCLAIMER:
This email may contain confidential information and is intended only for the use of the specific individual(s) to which it is addressed. If you are not the intended recipient of this email, you are hereby notified that any unauthorized use, dissemination or copying of this email or the information contained in it or attached to it is strictly prohibited. If you received this message in error, please immediately notify the sender at Infotech and delete the original message.
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
RE: SIP connectivity
Posted by Jeff Clay <Je...@infotech-enterprises.com>.
I do not believe that the asterisk context is related to the url of openmeetings.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
From: Naderi, Sascha [mailto:SNaderi@datus.com]
Sent: Wednesday, February 13, 2013 2:00 PM
To: user@openmeetings.apache.org
Cc: Maxim Solodovnik [solomax666@gmail.com]
Subject: Re: SIP connectivity
Dear all,
i have tested the asterisk sip integration as documented with the most recent instruction (http://openmeetings.apache.org/red5sip-integration_2.1.html) and it works just fine.
The only thing i am missing is a way to get this working when i choose to rename the openmeetings context from http://yourcorp.com:5080/openmeetings to http://yourcorp.com:5080/yourmeetings
Which settings do i have to modify so that red5sip functions even if the context name is changed?
Regards
Sascha Naderi
________________________________
Von: Maxim Solodovnik [solomax666@gmail.com]
Gesendet: Samstag, 9. Februar 2013 02:32
Bis: Bart Coninckx
Cc: user
Betreff: Re: SIP connectivity
All tables are created by OM automatically
On Feb 9, 2013 5:46 AM, "Bart Coninckx" <ba...@telenet.be>> wrote:
May I add that a portion is missing, since one explains how to configure Asterisk for Realtime, but one does not stipulate how to create the necessary tables.
It's in my CentOS docs however (which I hope to post shortly).
BC
On 01/31/13 13:05, Maxim Solodovnik wrote:
Hello Bart,
I just take a look at your URL ...
OM does not create/use sipfriends DB table (at least from version 2.1)
only meetme table is used
so I'm afraid there is nothing to change here
Here is the most recent instruction:
http://openmeetings.apache.org/red5sip-integration_2.1.html
Will ask our SIP guru to review it one more time :)
On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>> wrote:
OK will add it and notify you
On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>> wrote:
It is for Asterisk 11 - don't know for other versions. You probably have no issues because of the 1.8 version. To be sure the .sql files in the Asterisk source should be compared across versions.
this one is missing:
`useragent` varchar(20) DEFAULT NULL,
complete list (I think) is on:
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
If I bump into others, I'll report ASAP,
BC
On 01/31/13 06:21, Maxim Solodovnik wrote:
Is the OM meetme table incomplete?
My asterisk reports no issues :(
could you provide me with missing fields and I'll add it.
My purpose was to create table with required fields only.
On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <ba...@telenet.be>> wrote:
Openmeetings installed them for me, that's why I ended up with those. Using the Asterisk ones makes more sense to me. Maybe it's a good idea to have 'em removed from the install procedure.
BC
On 01/30/13 22:30, Jeff Clay wrote:
Bart,
If you look in the source directory of your asterisk tar file, under contrib/realtime/mysql you'll find the .sql files required for all the realtime drivers. I never thought to use the ones with OM.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
Sent: Wednesday, January 30, 2013 3:19 PM
To: user@openmeetings.apache.org<ma...@openmeetings.apache.org>
Cc: Jeff Clay
Subject: Re: SIP connectivity
Well,
I might have found one difference though:
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure dictates how the table should look like. I obviously used the one in the openmeetings mysql database, but this one seems to miss the table "useragent". I discovered this because it showed up in the logfiles.
BC
On 01/29/13 14:41, Jeff Clay wrote:
Bart,
>From an asterisk configuration standpoint there are very few differences between 1.8.x and 11.x. If memory serves, the only major changes that I ran into (in my production environment) was changes to SIP NAT values and the behavior of app_page() now uses confbridge instead of meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major overhauling. There were of course many other changes and bug fixes, you can skim through the change log for full details, but I think that was the jist of it.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
Sent: Tuesday, January 29, 2013 4:02 AM
To: Maxim Solodovnik
Cc: user
Subject: Re: SIP connectivity
I see - I'm willing to try the 11 version in the next fiew days if desired.
BC
On 01/29/13 10:57, Maxim Solodovnik wrote:
I test the integration using
Asterisk 1.8.13.1 (Ubuntu 12.10)
On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <ba...@telenet.be>> wrote:
That is amazing - I initially tried to do the same thing by using the new chan_motif driver in Asterisk 11 which connects to a XMPP server.
Are you guys using Asterisk 11? This version is the newest LTS version and has the best video capabilities.
Cheers,
BC
On 01/29/13 02:44, Maxim Solodovnik wrote:
red5sip will create special OM user in the room: "SIP Transport"
after that you can call to the OM room using SIP hard or soft phone.
We are currently testing it and trying to add video capabilities ...
On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <ba...@telenet.be>> wrote:
Hi Jeff,
In fact, I saw both pages, but none explain what they set up to do, just a bunch of command line instructions are given.
Your "OM will create a meetme meeting as configured in the realtime meetme database" actually says it all in one go :-)
cheers,
BC
On 01/28/13 22:38, Jeff Clay wrote:
Bart,
OM will create a meetme meeting as configured in the realtime meetme database. Have you read this page https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out http://openmeetings.apache.org/red5sip-integration.html but I assume this is the one you're already referring to.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
-----Original Message-----
From: Bart Coninckx [mailto:bart.coninckx@telenet.be<ma...@telenet.be>]
Sent: Monday, January 28, 2013 3:36 PM
To: user@openmeetings.apache.org<ma...@openmeetings.apache.org>
Subject: SIP connectivity
Hi,
I noticed some documentation on how to connect OM with a SIP proxy or server, more particularly with the MeetMe application in Asterisk.
The exact goal or purpose is not mentionned however. Will OM callout to a MeetMe conference? Or is it the other way round?
Cheers,
Bc
________________________________
DISCLAIMER:
This email may contain confidential information and is intended only for the use of the specific individual(s) to which it is addressed. If you are not the intended recipient of this email, you are hereby notified that any unauthorized use, dissemination or copying of this email or the information contained in it or attached to it is strictly prohibited. If you received this message in error, please immediately notify the sender at Infotech and delete the original message.
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: SIP connectivity
Posted by Alexei Fedotov <al...@gmail.com>.
Hello folks,
Timur succesfully demonstrated Asterisk integration with Openmeetings 1.9
after three months if work at the end of 2011.
We failed to make the same things working for om 2.0 by spending two weeks
because integration success strongly deoends on asterisk & flash versions,
network & asterisk configuration.
Another two months and I've seen SIP working again, though reliability and
sound quality were not good enough.
All are welcome to test the cuurent trunk and provide more details because
these two types of problems are hard to localise.
14.02.2013 3:04 пользователь "Maxim Solodovnik" <so...@gmail.com>
написал:
> I never tried asterisk integration in 2.0, it was reported to be very
> unstable
>
> 2.1 can be downloaded from here
> https://builds.apache.org/job/openmeetings/ (it is not released yet)
>
>
> On Thu, Feb 14, 2013 at 5:04 AM, Bakko <as...@gmail.com> wrote:
>
>> Hello,
>>
>> two questions.
>>
>> Asterisk integration working on openmeetings 2.0?
>>
>> Where can I download 2.1 version
>>
>> Thank you
>>
>> Regards
>>
>>
>> El 13/02/2013 14:59, Naderi, Sascha escribió:
>>
>> Dear all, ****
>>
>> ****
>>
>> ****
>>
>> ****
>>
>> i have tested the asterisk sip integration as documented with the most recent instruction
>> (http://openmeetings.apache.org/red5sip-integration_2.1.html) and it works
>> just fine.****
>>
>> ****
>>
>> The only thing i am missing is a way to get this working when
>> i choose to rename the openmeetings context from
>> http://yourcorp.com:5080/openmeetings to
>> http://yourcorp.com:5080/yourmeetings ****
>>
>> ****
>>
>> Which settings do i have to modify
>> so that red5sip functions even if the context name is changed?****
>>
>> ** **
>>
>>
>> Regards
>> Sascha Naderi
>>
>>
>> ------------------------------
>> *Von:* Maxim Solodovnik [solomax666@gmail.com]
>> *Gesendet:* Samstag, 9. Februar 2013 02:32
>> *Bis:* Bart Coninckx
>> *Cc:* user
>> *Betreff:* Re: SIP connectivity
>>
>> All tables are created by OM automatically
>> On Feb 9, 2013 5:46 AM, "Bart Coninckx" <ba...@telenet.be> wrote:
>>
>>> May I add that a portion is missing, since one explains how to
>>> configure Asterisk for Realtime, but one does not stipulate how to create
>>> the necessary tables.
>>> It's in my CentOS docs however (which I hope to post shortly).
>>>
>>> BC
>>>
>>> On 01/31/13 13:05, Maxim Solodovnik wrote:
>>>
>>> Hello Bart,
>>>
>>> I just take a look at your URL ...
>>> OM does not create/use sipfriends DB table (at least from version 2.1)
>>> only meetme table is used
>>>
>>> so I'm afraid there is nothing to change here
>>>
>>> Here is the most recent instruction:
>>> http://openmeetings.apache.org/red5sip-integration_2.1.html
>>>
>>> Will ask our SIP guru to review it one more time :)
>>>
>>>
>>>
>>> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>>>
>>>> OK will add it and notify you
>>>> On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>
>>>> wrote:
>>>>
>>>>> It is for Asterisk 11 - don't know for other versions. You probably
>>>>> have no issues because of the 1.8 version. To be sure the .sql files in the
>>>>> Asterisk source should be compared across versions.
>>>>>
>>>>> this one is missing:
>>>>>
>>>>> `useragent` varchar(20) DEFAULT NULL,
>>>>>
>>>>> complete list (I think) is on:
>>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>>>
>>>>>
>>>>> If I bump into others, I'll report ASAP,
>>>>>
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>>
>>>>> On 01/31/13 06:21, Maxim Solodovnik wrote:
>>>>>
>>>>> Is the OM meetme table incomplete?
>>>>> My asterisk reports no issues :(
>>>>>
>>>>> could you provide me with missing fields and I'll add it.
>>>>> My purpose was to create table with required fields only.
>>>>>
>>>>>
>>>>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <
>>>>> bart.coninckx@telenet.be> wrote:
>>>>>
>>>>>> Openmeetings installed them for me, that's why I ended up with
>>>>>> those. Using the Asterisk ones makes more sense to me. Maybe it's a good
>>>>>> idea to have 'em removed from the install procedure.
>>>>>>
>>>>>> BC
>>>>>>
>>>>>>
>>>>>> On 01/30/13 22:30, Jeff Clay wrote:
>>>>>>
>>>>>> Bart,
>>>>>>
>>>>>>
>>>>>>
>>>>>> If you look in the source directory of your asterisk tar file, under
>>>>>> contrib/realtime/mysql you’ll find the .sql files required for all the
>>>>>> realtime drivers. I never thought to use the ones with OM.
>>>>>>
>>>>>>
>>>>>>
>>>>>> Jeff Clay
>>>>>>
>>>>>> Network Administrator
>>>>>>
>>>>>> Infotech Enterprises America
>>>>>>
>>>>>> 870-215-5506
>>>>>>
>>>>>> Ext. 1506
>>>>>>
>>>>>>
>>>>>>
>>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>>>
>>>>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>>>>> *To:* user@openmeetings.apache.org
>>>>>> *Cc:* Jeff Clay
>>>>>> *Subject:* Re: SIP connectivity
>>>>>>
>>>>>>
>>>>>>
>>>>>> Well,
>>>>>>
>>>>>> I might have found one difference though:
>>>>>>
>>>>>>
>>>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>>>> dictates how the table should look like. I obviously used the one in the
>>>>>> openmeetings mysql database, but this one seems to miss the table
>>>>>> "useragent". I discovered this because it showed up in the logfiles.
>>>>>>
>>>>>> BC
>>>>>>
>>>>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>>>>
>>>>>> Bart,
>>>>>>
>>>>>>
>>>>>>
>>>>>> From an asterisk configuration standpoint there are very few
>>>>>> differences between 1.8.x and 11.x. If memory serves, the only major
>>>>>> changes that I ran into (in my production environment) was changes to SIP
>>>>>> NAT values and the behavior of app_page() now uses confbridge instead of
>>>>>> meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major
>>>>>> overhauling. There were of course many other changes and bug fixes, you can
>>>>>> skim through the change log for full details, but I think that was the jist
>>>>>> of it.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> Jeff Clay
>>>>>>
>>>>>> Network Administrator
>>>>>>
>>>>>> Infotech Enterprises America
>>>>>>
>>>>>> 870-215-5506
>>>>>>
>>>>>> Ext. 1506
>>>>>>
>>>>>>
>>>>>>
>>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>>>
>>>>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>>>>> *To:* Maxim Solodovnik
>>>>>> *Cc:* user
>>>>>> *Subject:* Re: SIP connectivity
>>>>>>
>>>>>>
>>>>>>
>>>>>> I see - I'm willing to try the 11 version in the next fiew days if
>>>>>> desired.
>>>>>>
>>>>>> BC
>>>>>>
>>>>>>
>>>>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>>>>
>>>>>> I test the integration using
>>>>>>
>>>>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <
>>>>>> bart.coninckx@telenet.be> wrote:
>>>>>>
>>>>>> That is amazing - I initially tried to do the same thing by using the
>>>>>> new chan_motif driver in Asterisk 11 which connects to a XMPP server.
>>>>>>
>>>>>> Are you guys using Asterisk 11? This version is the newest LTS
>>>>>> version and has the best video capabilities.
>>>>>>
>>>>>> Cheers,
>>>>>>
>>>>>> BC
>>>>>>
>>>>>>
>>>>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>>>>
>>>>>> red5sip will create special OM user in the room: "SIP Transport"
>>>>>>
>>>>>> after that you can call to the OM room using SIP hard or soft phone.
>>>>>>
>>>>>>
>>>>>>
>>>>>> We are currently testing it and trying to add video capabilities ...
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <
>>>>>> bart.coninckx@telenet.be> wrote:
>>>>>>
>>>>>> Hi Jeff,
>>>>>>
>>>>>> In fact, I saw both pages, but none explain what they set up to do,
>>>>>> just a bunch of command line instructions are given.
>>>>>> Your "OM will create a meetme meeting as configured in the realtime
>>>>>> meetme database" actually says it all in one go :-)
>>>>>>
>>>>>> cheers,
>>>>>>
>>>>>> BC
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>>>>
>>>>>> Bart,
>>>>>>
>>>>>> OM will create a meetme meeting as configured in the realtime meetme
>>>>>> database. Have you read this page
>>>>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
>>>>>> http://openmeetings.apache.org/red5sip-integration.html but I assume
>>>>>> this is the one you're already referring to.
>>>>>>
>>>>>> Jeff Clay
>>>>>> Network Administrator
>>>>>> Infotech Enterprises America
>>>>>> 870-215-5506
>>>>>> Ext. 1506
>>>>>>
>>>>>> -----Original Message-----
>>>>>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>>>>> Sent: Monday, January 28, 2013 3:36 PM
>>>>>> To: user@openmeetings.apache.org
>>>>>> Subject: SIP connectivity
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I noticed some documentation on how to connect OM with a SIP proxy or
>>>>>> server, more particularly with the MeetMe application in Asterisk.
>>>>>>
>>>>>> The exact goal or purpose is not mentionned however. Will OM callout
>>>>>> to a MeetMe conference? Or is it the other way round?
>>>>>>
>>>>>>
>>>>>> Cheers,
>>>>>>
>>>>>> Bc
>>>>>>
>>>>>> ________________________________
>>>>>>
>>>>>> DISCLAIMER:
>>>>>>
>>>>>> This email may contain confidential information and is intended only
>>>>>> for the use of the specific individual(s) to which it is addressed. If you
>>>>>> are not the intended recipient of this email, you are hereby notified that
>>>>>> any unauthorized use, dissemination or copying of this email or the
>>>>>> information contained in it or attached to it is strictly prohibited. If
>>>>>> you received this message in error, please immediately notify the sender at
>>>>>> Infotech and delete the original message.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> WBR
>>>>>> Maxim aka solomax
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> WBR
>>>>>> Maxim aka solomax
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> WBR
>>>>> Maxim aka solomax
>>>>>
>>>>>
>>>>>
>>>
>>>
>>> --
>>> WBR
>>> Maxim aka solomax
>>>
>>>
>>>
>>
>
>
> --
> WBR
> Maxim aka solomax
>
Re: SIP connectivity
Posted by Maxim Solodovnik <so...@gmail.com>.
I never tried asterisk integration in 2.0, it was reported to be very
unstable
2.1 can be downloaded from here https://builds.apache.org/job/openmeetings/ (it
is not released yet)
On Thu, Feb 14, 2013 at 5:04 AM, Bakko <as...@gmail.com> wrote:
> Hello,
>
> two questions.
>
> Asterisk integration working on openmeetings 2.0?
>
> Where can I download 2.1 version
>
> Thank you
>
> Regards
>
>
> El 13/02/2013 14:59, Naderi, Sascha escribió:
>
> Dear all, ****
>
> ****
>
> ****
>
> ****
>
> i have tested the asterisk sip integration as documented with the most recent instruction
> (http://openmeetings.apache.org/red5sip-integration_2.1.html) and it works
> just fine.****
>
> ****
>
> The only thing i am missing is a way to get this working when
> i choose to rename the openmeetings context from
> http://yourcorp.com:5080/openmeetings to
> http://yourcorp.com:5080/yourmeetings ****
>
> ****
>
> Which settings do i have to modify
> so that red5sip functions even if the context name is changed?****
>
> ** **
>
>
> Regards
> Sascha Naderi
>
>
> ------------------------------
> *Von:* Maxim Solodovnik [solomax666@gmail.com]
> *Gesendet:* Samstag, 9. Februar 2013 02:32
> *Bis:* Bart Coninckx
> *Cc:* user
> *Betreff:* Re: SIP connectivity
>
> All tables are created by OM automatically
> On Feb 9, 2013 5:46 AM, "Bart Coninckx" <ba...@telenet.be> wrote:
>
>> May I add that a portion is missing, since one explains how to
>> configure Asterisk for Realtime, but one does not stipulate how to create
>> the necessary tables.
>> It's in my CentOS docs however (which I hope to post shortly).
>>
>> BC
>>
>> On 01/31/13 13:05, Maxim Solodovnik wrote:
>>
>> Hello Bart,
>>
>> I just take a look at your URL ...
>> OM does not create/use sipfriends DB table (at least from version 2.1)
>> only meetme table is used
>>
>> so I'm afraid there is nothing to change here
>>
>> Here is the most recent instruction:
>> http://openmeetings.apache.org/red5sip-integration_2.1.html
>>
>> Will ask our SIP guru to review it one more time :)
>>
>>
>>
>> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>>
>>> OK will add it and notify you
>>> On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>
>>> wrote:
>>>
>>>> It is for Asterisk 11 - don't know for other versions. You probably
>>>> have no issues because of the 1.8 version. To be sure the .sql files in the
>>>> Asterisk source should be compared across versions.
>>>>
>>>> this one is missing:
>>>>
>>>> `useragent` varchar(20) DEFAULT NULL,
>>>>
>>>> complete list (I think) is on:
>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>>
>>>>
>>>> If I bump into others, I'll report ASAP,
>>>>
>>>>
>>>> BC
>>>>
>>>>
>>>>
>>>> On 01/31/13 06:21, Maxim Solodovnik wrote:
>>>>
>>>> Is the OM meetme table incomplete?
>>>> My asterisk reports no issues :(
>>>>
>>>> could you provide me with missing fields and I'll add it.
>>>> My purpose was to create table with required fields only.
>>>>
>>>>
>>>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <
>>>> bart.coninckx@telenet.be> wrote:
>>>>
>>>>> Openmeetings installed them for me, that's why I ended up with
>>>>> those. Using the Asterisk ones makes more sense to me. Maybe it's a good
>>>>> idea to have 'em removed from the install procedure.
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>> On 01/30/13 22:30, Jeff Clay wrote:
>>>>>
>>>>> Bart,
>>>>>
>>>>>
>>>>>
>>>>> If you look in the source directory of your asterisk tar file, under
>>>>> contrib/realtime/mysql you’ll find the .sql files required for all the
>>>>> realtime drivers. I never thought to use the ones with OM.
>>>>>
>>>>>
>>>>>
>>>>> Jeff Clay
>>>>>
>>>>> Network Administrator
>>>>>
>>>>> Infotech Enterprises America
>>>>>
>>>>> 870-215-5506
>>>>>
>>>>> Ext. 1506
>>>>>
>>>>>
>>>>>
>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>>
>>>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>>>> *To:* user@openmeetings.apache.org
>>>>> *Cc:* Jeff Clay
>>>>> *Subject:* Re: SIP connectivity
>>>>>
>>>>>
>>>>>
>>>>> Well,
>>>>>
>>>>> I might have found one difference though:
>>>>>
>>>>>
>>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>>> dictates how the table should look like. I obviously used the one in the
>>>>> openmeetings mysql database, but this one seems to miss the table
>>>>> "useragent". I discovered this because it showed up in the logfiles.
>>>>>
>>>>> BC
>>>>>
>>>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>>>
>>>>> Bart,
>>>>>
>>>>>
>>>>>
>>>>> From an asterisk configuration standpoint there are very few
>>>>> differences between 1.8.x and 11.x. If memory serves, the only major
>>>>> changes that I ran into (in my production environment) was changes to SIP
>>>>> NAT values and the behavior of app_page() now uses confbridge instead of
>>>>> meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major
>>>>> overhauling. There were of course many other changes and bug fixes, you can
>>>>> skim through the change log for full details, but I think that was the jist
>>>>> of it.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Jeff Clay
>>>>>
>>>>> Network Administrator
>>>>>
>>>>> Infotech Enterprises America
>>>>>
>>>>> 870-215-5506
>>>>>
>>>>> Ext. 1506
>>>>>
>>>>>
>>>>>
>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>>
>>>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>>>> *To:* Maxim Solodovnik
>>>>> *Cc:* user
>>>>> *Subject:* Re: SIP connectivity
>>>>>
>>>>>
>>>>>
>>>>> I see - I'm willing to try the 11 version in the next fiew days if
>>>>> desired.
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>>>
>>>>> I test the integration using
>>>>>
>>>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <
>>>>> bart.coninckx@telenet.be> wrote:
>>>>>
>>>>> That is amazing - I initially tried to do the same thing by using the
>>>>> new chan_motif driver in Asterisk 11 which connects to a XMPP server.
>>>>>
>>>>> Are you guys using Asterisk 11? This version is the newest LTS version
>>>>> and has the best video capabilities.
>>>>>
>>>>> Cheers,
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>>>
>>>>> red5sip will create special OM user in the room: "SIP Transport"
>>>>>
>>>>> after that you can call to the OM room using SIP hard or soft phone.
>>>>>
>>>>>
>>>>>
>>>>> We are currently testing it and trying to add video capabilities ...
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <
>>>>> bart.coninckx@telenet.be> wrote:
>>>>>
>>>>> Hi Jeff,
>>>>>
>>>>> In fact, I saw both pages, but none explain what they set up to do,
>>>>> just a bunch of command line instructions are given.
>>>>> Your "OM will create a meetme meeting as configured in the realtime
>>>>> meetme database" actually says it all in one go :-)
>>>>>
>>>>> cheers,
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>>>
>>>>> Bart,
>>>>>
>>>>> OM will create a meetme meeting as configured in the realtime meetme
>>>>> database. Have you read this page
>>>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
>>>>> http://openmeetings.apache.org/red5sip-integration.html but I assume
>>>>> this is the one you're already referring to.
>>>>>
>>>>> Jeff Clay
>>>>> Network Administrator
>>>>> Infotech Enterprises America
>>>>> 870-215-5506
>>>>> Ext. 1506
>>>>>
>>>>> -----Original Message-----
>>>>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>>>> Sent: Monday, January 28, 2013 3:36 PM
>>>>> To: user@openmeetings.apache.org
>>>>> Subject: SIP connectivity
>>>>>
>>>>> Hi,
>>>>>
>>>>> I noticed some documentation on how to connect OM with a SIP proxy or
>>>>> server, more particularly with the MeetMe application in Asterisk.
>>>>>
>>>>> The exact goal or purpose is not mentionned however. Will OM callout
>>>>> to a MeetMe conference? Or is it the other way round?
>>>>>
>>>>>
>>>>> Cheers,
>>>>>
>>>>> Bc
>>>>>
>>>>> ________________________________
>>>>>
>>>>> DISCLAIMER:
>>>>>
>>>>> This email may contain confidential information and is intended only
>>>>> for the use of the specific individual(s) to which it is addressed. If you
>>>>> are not the intended recipient of this email, you are hereby notified that
>>>>> any unauthorized use, dissemination or copying of this email or the
>>>>> information contained in it or attached to it is strictly prohibited. If
>>>>> you received this message in error, please immediately notify the sender at
>>>>> Infotech and delete the original message.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> WBR
>>>>> Maxim aka solomax
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> WBR
>>>>> Maxim aka solomax
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>>
>>>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>
--
WBR
Maxim aka solomax
Re: SIP connectivity
Posted by Bakko <as...@gmail.com>.
Hello,
two questions.
Asterisk integration working on openmeetings 2.0?
Where can I download 2.1 version
Thank you
Regards
El 13/02/2013 14:59, Naderi, Sascha escribió:
>
> Dear all,
>
> i have tested the asterisk sip integration as documented with the most recent instruction
> (http://openmeetings.apache.org/red5sip-integration_2.1.html) and it works
> just fine.
>
> The only thing i am missing is a way to get this working when
> i choose to rename the openmeetings context from
> http://yourcorp.com:5080/openmeetings to
> http://yourcorp.com:5080/yourmeetings
>
> Which settings do i have to modify
> so that red5sip functions even if the context name is changed?
>
> Regards
> Sascha Naderi
>
>
> ------------------------------------------------------------------------
> *Von:* Maxim Solodovnik [solomax666@gmail.com]
> *Gesendet:* Samstag, 9. Februar 2013 02:32
> *Bis:* Bart Coninckx
> *Cc:* user
> *Betreff:* Re: SIP connectivity
>
> All tables are created by OM automatically
>
> On Feb 9, 2013 5:46 AM, "Bart Coninckx" <bart.coninckx@telenet.be
> <ma...@telenet.be>> wrote:
>
> May I add that a portion is missing, since one explains how to
> configure Asterisk for Realtime, but one does not stipulate how to
> create the necessary tables.
> It's in my CentOS docs however (which I hope to post shortly).
>
> BC
>
> On 01/31/13 13:05, Maxim Solodovnik wrote:
>> Hello Bart,
>>
>> I just take a look at your URL ...
>> OM does not create/use sipfriends DB table (at least from version
>> 2.1)
>> only meetme table is used
>>
>> so I'm afraid there is nothing to change here
>>
>> Here is the most recent instruction:
>> http://openmeetings.apache.org/red5sip-integration_2.1.html
>>
>> Will ask our SIP guru to review it one more time :)
>>
>>
>>
>> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik
>> <solomax666@gmail.com <ma...@gmail.com>> wrote:
>>
>> OK will add it and notify you
>>
>> On Jan 31, 2013 5:05 PM, "Bart Coninckx"
>> <bart.coninckx@telenet.be <ma...@telenet.be>>
>> wrote:
>>
>> It is for Asterisk 11 - don't know for other versions.
>> You probably have no issues because of the 1.8 version.
>> To be sure the .sql files in the Asterisk source should
>> be compared across versions.
>>
>> this one is missing:
>>
>> `useragent` varchar(20) DEFAULT NULL,
>>
>> complete list (I think) is on:
>>
>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>
>>
>> If I bump into others, I'll report ASAP,
>>
>>
>> BC
>>
>>
>>
>> On 01/31/13 06:21, Maxim Solodovnik wrote:
>>> Is the OM meetme table incomplete?
>>> My asterisk reports no issues :(
>>>
>>> could you provide me with missing fields and I'll add it.
>>> My purpose was to create table with required fields only.
>>>
>>>
>>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx
>>> <bart.coninckx@telenet.be
>>> <ma...@telenet.be>> wrote:
>>>
>>> Openmeetings installed them for me, that's why I
>>> ended up with those. Using the Asterisk ones makes
>>> more sense to me. Maybe it's a good idea to have 'em
>>> removed from the install procedure.
>>>
>>> BC
>>>
>>>
>>> On 01/30/13 22:30, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>> If you look in the source directory of your
>>>> asterisk tar file, under contrib/realtime/mysql
>>>> you’ll find the .sql files required for all the
>>>> realtime drivers. I never thought to use the ones
>>>> with OM.
>>>>
>>>> Jeff Clay
>>>>
>>>> Network Administrator
>>>>
>>>> Infotech Enterprises America
>>>>
>>>> 870-215-5506
>>>>
>>>> Ext. 1506
>>>>
>>>> *From:*Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>>> *To:* user@openmeetings.apache.org
>>>> <ma...@openmeetings.apache.org>
>>>> *Cc:* Jeff Clay
>>>> *Subject:* Re: SIP connectivity
>>>>
>>>> Well,
>>>>
>>>> I might have found one difference though:
>>>>
>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>> dictates how the table should look like. I
>>>> obviously used the one in the openmeetings mysql
>>>> database, but this one seems to miss the table
>>>> "useragent". I discovered this because it showed up
>>>> in the logfiles.
>>>>
>>>> BC
>>>>
>>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>> From an asterisk configuration standpoint there
>>>> are very few differences between 1.8.x and
>>>> 11.x. If memory serves, the only major changes
>>>> that I ran into (in my production environment)
>>>> was changes to SIP NAT values and the behavior
>>>> of app_page() now uses confbridge instead of
>>>> meetme to mix the audio. Also, TCP, TLS and
>>>> app_confbridge got a major overhauling. There
>>>> were of course many other changes and bug
>>>> fixes, you can skim through the change log for
>>>> full details, but I think that was the jist of it.
>>>>
>>>> Jeff Clay
>>>>
>>>> Network Administrator
>>>>
>>>> Infotech Enterprises America
>>>>
>>>> 870-215-5506
>>>>
>>>> Ext. 1506
>>>>
>>>> *From:*Bart Coninckx
>>>> [mailto:bart.coninckx@telenet.be]
>>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>>> *To:* Maxim Solodovnik
>>>> *Cc:* user
>>>> *Subject:* Re: SIP connectivity
>>>>
>>>> I see - I'm willing to try the 11 version in
>>>> the next fiew days if desired.
>>>>
>>>> BC
>>>>
>>>>
>>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>>
>>>> I test the integration using
>>>>
>>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>>
>>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart
>>>> Coninckx <bart.coninckx@telenet.be
>>>> <ma...@telenet.be>> wrote:
>>>>
>>>> That is amazing - I initially tried to do
>>>> the same thing by using the new chan_motif
>>>> driver in Asterisk 11 which connects to a
>>>> XMPP server.
>>>>
>>>> Are you guys using Asterisk 11? This
>>>> version is the newest LTS version and has
>>>> the best video capabilities.
>>>>
>>>> Cheers,
>>>>
>>>> BC
>>>>
>>>>
>>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>>
>>>> red5sip will create special OM user in
>>>> the room: "SIP Transport"
>>>>
>>>> after that you can call to the OM room
>>>> using SIP hard or soft phone.
>>>>
>>>> We are currently testing it and trying
>>>> to add video capabilities ...
>>>>
>>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart
>>>> Coninckx <bart.coninckx@telenet.be
>>>> <ma...@telenet.be>> wrote:
>>>>
>>>> Hi Jeff,
>>>>
>>>> In fact, I saw both pages, but none
>>>> explain what they set up to do, just a
>>>> bunch of command line instructions are
>>>> given.
>>>> Your "OM will create a meetme meeting
>>>> as configured in the realtime meetme
>>>> database" actually says it all in one
>>>> go :-)
>>>>
>>>> cheers,
>>>>
>>>> BC
>>>>
>>>>
>>>>
>>>>
>>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>> OM will create a meetme meeting as
>>>> configured in the realtime meetme
>>>> database. Have you read this page
>>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html
>>>> ? You might also check out
>>>> http://openmeetings.apache.org/red5sip-integration.html
>>>> but I assume this is the one you're
>>>> already referring to.
>>>>
>>>> Jeff Clay
>>>> Network Administrator
>>>> Infotech Enterprises America
>>>> 870-215-5506
>>>> Ext. 1506
>>>>
>>>> -----Original Message-----
>>>> From: Bart Coninckx
>>>> [mailto:bart.coninckx@telenet.be
>>>> <ma...@telenet.be>]
>>>> Sent: Monday, January 28, 2013 3:36 PM
>>>> To: user@openmeetings.apache.org
>>>> <ma...@openmeetings.apache.org>
>>>> Subject: SIP connectivity
>>>>
>>>> Hi,
>>>>
>>>> I noticed some documentation on how to
>>>> connect OM with a SIP proxy or server,
>>>> more particularly with the MeetMe
>>>> application in Asterisk.
>>>>
>>>> The exact goal or purpose is not
>>>> mentionned however. Will OM callout to
>>>> a MeetMe conference? Or is it the other
>>>> way round?
>>>>
>>>>
>>>> Cheers,
>>>>
>>>> Bc
>>>>
>>>> ________________________________
>>>>
>>>> DISCLAIMER:
>>>>
>>>> This email may contain confidential
>>>> information and is intended only for
>>>> the use of the specific individual(s)
>>>> to which it is addressed. If you are
>>>> not the intended recipient of this
>>>> email, you are hereby notified that any
>>>> unauthorized use, dissemination or
>>>> copying of this email or the
>>>> information contained in it or attached
>>>> to it is strictly prohibited. If you
>>>> received this message in error, please
>>>> immediately notify the sender at
>>>> Infotech and delete the original message.
>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>
>>>
>>>
>>>
>>> --
>>> WBR
>>> Maxim aka solomax
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>
Re: SIP connectivity
Posted by Maxim Solodovnik <so...@gmail.com>.
You will need to set up red5sip separately (it licence is incompatible with
Apache)
On Tue, Feb 19, 2013 at 8:42 PM, BBS Technik <do...@gmx.de> wrote:
> Hello,
>
> that are very good news. Could one of the sip experts document the finaly
> needed steps in the VOIP / SIP Integration documentation in the wiki?
>
> Will the upcomming 2.1 release support the sip integration out of the box?
>
> Best regards
> Ed
>
>
>
> -------- Original-Nachricht --------
> > Datum: Tue, 19 Feb 2013 13:01:03 +0000
> > Von: "Naderi, Sascha" <SN...@datus.com>
> > An: "user@openmeetings.apache.org" <us...@openmeetings.apache.org>
> > CC: "solomax666@gmail.com" <so...@gmail.com>
> > Betreff: Re: SIP connectivity
>
> > Dear Maxim, dear all,
> >
> >
> >
> >
> >
> > i tried it with the latest red5sip rev. (91) and it worked fine with a
> > changed openmeetings context.
> >
> > Thank you!
> >
> >
> >
> >
> >
> >
> >
> > Regards
> >
> > Sascha
> >
> > ________________________________
> >
> > Von: Naderi, Sascha
> > Gesendet: Donnerstag, 14. Februar 2013 08:09
> > Bis: Maxim Solodovnik
> > Cc: user@openmeetings.apache.org
> > Betreff: AW: SIP connectivity
> >
> >
> > Dear Maxim,
> >
> >
> >
> >
> >
> > OK, thanks a lot. I will check it out and leave feedback.
> >
> >
> >
> >
> >
> > Regards
> >
> > Sascha
> >
> > ________________________________
> >
> > Von: Maxim Solodovnik [solomax666@gmail.com]
> > Gesendet: Mittwoch, 13. Februar 2013 23:58
> > Bis: Naderi, Sascha
> > Cc: user@openmeetings.apache.org
> > Betreff: Re: SIP connectivity
> >
> > please try red5sip rev. 76
> > it has additional parameter: om.context
> >
> >
> > On Thu, Feb 14, 2013 at 2:59 AM, Naderi, Sascha
> > <SN...@datus.com>> wrote:
> >
> > Dear all,
> >
> >
> >
> >
> >
> >
> >
> > i have tested the asterisk sip integration as documented with the most
> > recent instruction
> > (http://openmeetings.apache.org/red5sip-integration_2.1.html) and it
> works just fine.
> >
> > The only thing i am missing is a way to get this working when i choose to
> > rename the openmeetings context from
> http://yourcorp.com:5080/openmeetings
> > to http://yourcorp.com:5080/yourmeetings
> >
> > Which settings do i have to modify so that red5sip functions even if the
> > context name is changed?
> >
> >
> >
> >
> > Regards
> > Sascha Naderi
> >
> >
> > ________________________________
> >
> > Von: Maxim Solodovnik [solomax666@gmail.com<mailto:solomax666@gmail.com
> >]
> > Gesendet: Samstag, 9. Februar 2013 02:32
> > Bis: Bart Coninckx
> > Cc: user
> > Betreff: Re: SIP connectivity
> >
> >
> > All tables are created by OM automatically
> >
> > On Feb 9, 2013 5:46 AM, "Bart Coninckx"
> > <ba...@telenet.be>> wrote:
> > May I add that a portion is missing, since one explains how to configure
> > Asterisk for Realtime, but one does not stipulate how to create the
> > necessary tables.
> > It's in my CentOS docs however (which I hope to post shortly).
> >
> > BC
> >
> > On 01/31/13 13:05, Maxim Solodovnik wrote:
> > Hello Bart,
> >
> > I just take a look at your URL ...
> > OM does not create/use sipfriends DB table (at least from version 2.1)
> > only meetme table is used
> >
> > so I'm afraid there is nothing to change here
> >
> > Here is the most recent instruction:
> > http://openmeetings.apache.org/red5sip-integration_2.1.html
> >
> > Will ask our SIP guru to review it one more time :)
> >
> >
> >
> > On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik
> > <so...@gmail.com>> wrote:
> >
> > OK will add it and notify you
> >
> > On Jan 31, 2013 5:05 PM, "Bart Coninckx"
> > <ba...@telenet.be>> wrote:
> > It is for Asterisk 11 - don't know for other versions. You probably have
> > no issues because of the 1.8 version. To be sure the .sql files in the
> > Asterisk source should be compared across versions.
> >
> > this one is missing:
> >
> >
> > `useragent` varchar(20) DEFAULT NULL,
> >
> > complete list (I think) is on:
> >
> >
> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
> >
> >
> > If I bump into others, I'll report ASAP,
> >
> >
> > BC
> >
> >
> >
> > On 01/31/13 06:21, Maxim Solodovnik wrote:
> > Is the OM meetme table incomplete?
> > My asterisk reports no issues :(
> >
> > could you provide me with missing fields and I'll add it.
> > My purpose was to create table with required fields only.
> >
> >
> > On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx
> > <ba...@telenet.be>> wrote:
> > Openmeetings installed them for me, that's why I ended up with those.
> > Using the Asterisk ones makes more sense to me. Maybe it's a good idea
> to have
> > 'em removed from the install procedure.
> >
> > BC
> >
> >
> > On 01/30/13 22:30, Jeff Clay wrote:
> > Bart,
> >
> > If you look in the source directory of your asterisk tar file, under
> > contrib/realtime/mysql you’ll find the .sql files required for all the
> > realtime drivers. I never thought to use the ones with OM.
> >
> > Jeff Clay
> > Network Administrator
> > Infotech Enterprises America
> > 870-215-5506
> > Ext. 1506
> >
> > From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
> > Sent: Wednesday, January 30, 2013 3:19 PM
> > To: user@openmeetings.apache.org<ma...@openmeetings.apache.org>
> > Cc: Jeff Clay
> > Subject: Re: SIP connectivity
> >
> > Well,
> >
> > I might have found one difference though:
> >
> >
> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
> > dictates how the table should look like. I obviously used the one in the
> > openmeetings mysql database, but this one seems to miss the table
> > "useragent". I discovered this because it showed up in the logfiles.
> >
> > BC
> >
> > On 01/29/13 14:41, Jeff Clay wrote:
> > Bart,
> >
> > From an asterisk configuration standpoint there are very few differences
> > between 1.8.x and 11.x. If memory serves, the only major changes that I
> ran
> > into (in my production environment) was changes to SIP NAT values and the
> > behavior of app_page() now uses confbridge instead of meetme to mix the
> > audio. Also, TCP, TLS and app_confbridge got a major overhauling. There
> were of
> > course many other changes and bug fixes, you can skim through the change
> > log for full details, but I think that was the jist of it.
> >
> >
> >
> > Jeff Clay
> > Network Administrator
> > Infotech Enterprises America
> > 870-215-5506
> > Ext. 1506
> >
> > From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
> > Sent: Tuesday, January 29, 2013 4:02 AM
> > To: Maxim Solodovnik
> > Cc: user
> > Subject: Re: SIP connectivity
> >
> > I see - I'm willing to try the 11 version in the next fiew days if
> > desired.
> >
> > BC
> >
> >
> > On 01/29/13 10:57, Maxim Solodovnik wrote:
> > I test the integration using
> > Asterisk 1.8.13.1 (Ubuntu 12.10)
> >
> > On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx
> > <ba...@telenet.be>> wrote:
> > That is amazing - I initially tried to do the same thing by using the new
> > chan_motif driver in Asterisk 11 which connects to a XMPP server.
> >
> > Are you guys using Asterisk 11? This version is the newest LTS version
> and
> > has the best video capabilities.
> >
> > Cheers,
> >
> > BC
> >
> >
> > On 01/29/13 02:44, Maxim Solodovnik wrote:
> > red5sip will create special OM user in the room: "SIP Transport"
> > after that you can call to the OM room using SIP hard or soft phone.
> >
> > We are currently testing it and trying to add video capabilities ...
> >
> > On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx
> > <ba...@telenet.be>> wrote:
> > Hi Jeff,
> >
> > In fact, I saw both pages, but none explain what they set up to do, just
> a
> > bunch of command line instructions are given.
> > Your "OM will create a meetme meeting as configured in the realtime
> meetme
> > database" actually says it all in one go :-)
> >
> > cheers,
> >
> > BC
> >
> >
> >
> > On 01/28/13 22:38, Jeff Clay wrote:
> > Bart,
> >
> > OM will create a meetme meeting as configured in the realtime meetme
> > database. Have you read this page
> >
> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
> > http://openmeetings.apache.org/red5sip-integration.html but I assume
> this is the one
> > you're already referring to.
> >
> > Jeff Clay
> > Network Administrator
> > Infotech Enterprises America
> > 870-215-5506
> > Ext. 1506
> >
> > -----Original Message-----
> > From: Bart Coninckx
> > [mailto:bart.coninckx@telenet.be<ma...@telenet.be>]
> > Sent: Monday, January 28, 2013 3:36 PM
> > To: user@openmeetings.apache.org<ma...@openmeetings.apache.org>
> > Subject: SIP connectivity
> >
> > Hi,
> >
> > I noticed some documentation on how to connect OM with a SIP proxy or
> > server, more particularly with the MeetMe application in Asterisk.
> >
> > The exact goal or purpose is not mentionned however. Will OM callout to a
> > MeetMe conference? Or is it the other way round?
> >
> >
> > Cheers,
> >
> > Bc
> >
> > ________________________________
> >
> > DISCLAIMER:
> >
> > This email may contain confidential information and is intended only for
> > the use of the specific individual(s) to which it is addressed. If you
> are
> > not the intended recipient of this email, you are hereby notified that
> any
> > unauthorized use, dissemination or copying of this email or the
> information
> > contained in it or attached to it is strictly prohibited. If you received
> > this message in error, please immediately notify the sender at Infotech
> and
> > delete the original message.
> >
> >
> >
> >
> > --
> > 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: SIP connectivity
Posted by BBS Technik <do...@gmx.de>.
Hello,
that are very good news. Could one of the sip experts document the finaly needed steps in the VOIP / SIP Integration documentation in the wiki?
Will the upcomming 2.1 release support the sip integration out of the box?
Best regards
Ed
-------- Original-Nachricht --------
> Datum: Tue, 19 Feb 2013 13:01:03 +0000
> Von: "Naderi, Sascha" <SN...@datus.com>
> An: "user@openmeetings.apache.org" <us...@openmeetings.apache.org>
> CC: "solomax666@gmail.com" <so...@gmail.com>
> Betreff: Re: SIP connectivity
> Dear Maxim, dear all,
>
>
>
>
>
> i tried it with the latest red5sip rev. (91) and it worked fine with a
> changed openmeetings context.
>
> Thank you!
>
>
>
>
>
>
>
> Regards
>
> Sascha
>
> ________________________________
>
> Von: Naderi, Sascha
> Gesendet: Donnerstag, 14. Februar 2013 08:09
> Bis: Maxim Solodovnik
> Cc: user@openmeetings.apache.org
> Betreff: AW: SIP connectivity
>
>
> Dear Maxim,
>
>
>
>
>
> OK, thanks a lot. I will check it out and leave feedback.
>
>
>
>
>
> Regards
>
> Sascha
>
> ________________________________
>
> Von: Maxim Solodovnik [solomax666@gmail.com]
> Gesendet: Mittwoch, 13. Februar 2013 23:58
> Bis: Naderi, Sascha
> Cc: user@openmeetings.apache.org
> Betreff: Re: SIP connectivity
>
> please try red5sip rev. 76
> it has additional parameter: om.context
>
>
> On Thu, Feb 14, 2013 at 2:59 AM, Naderi, Sascha
> <SN...@datus.com>> wrote:
>
> Dear all,
>
>
>
>
>
>
>
> i have tested the asterisk sip integration as documented with the most
> recent instruction
> (http://openmeetings.apache.org/red5sip-integration_2.1.html) and it works just fine.
>
> The only thing i am missing is a way to get this working when i choose to
> rename the openmeetings context from http://yourcorp.com:5080/openmeetings
> to http://yourcorp.com:5080/yourmeetings
>
> Which settings do i have to modify so that red5sip functions even if the
> context name is changed?
>
>
>
>
> Regards
> Sascha Naderi
>
>
> ________________________________
>
> Von: Maxim Solodovnik [solomax666@gmail.com<ma...@gmail.com>]
> Gesendet: Samstag, 9. Februar 2013 02:32
> Bis: Bart Coninckx
> Cc: user
> Betreff: Re: SIP connectivity
>
>
> All tables are created by OM automatically
>
> On Feb 9, 2013 5:46 AM, "Bart Coninckx"
> <ba...@telenet.be>> wrote:
> May I add that a portion is missing, since one explains how to configure
> Asterisk for Realtime, but one does not stipulate how to create the
> necessary tables.
> It's in my CentOS docs however (which I hope to post shortly).
>
> BC
>
> On 01/31/13 13:05, Maxim Solodovnik wrote:
> Hello Bart,
>
> I just take a look at your URL ...
> OM does not create/use sipfriends DB table (at least from version 2.1)
> only meetme table is used
>
> so I'm afraid there is nothing to change here
>
> Here is the most recent instruction:
> http://openmeetings.apache.org/red5sip-integration_2.1.html
>
> Will ask our SIP guru to review it one more time :)
>
>
>
> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik
> <so...@gmail.com>> wrote:
>
> OK will add it and notify you
>
> On Jan 31, 2013 5:05 PM, "Bart Coninckx"
> <ba...@telenet.be>> wrote:
> It is for Asterisk 11 - don't know for other versions. You probably have
> no issues because of the 1.8 version. To be sure the .sql files in the
> Asterisk source should be compared across versions.
>
> this one is missing:
>
>
> `useragent` varchar(20) DEFAULT NULL,
>
> complete list (I think) is on:
>
> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>
>
> If I bump into others, I'll report ASAP,
>
>
> BC
>
>
>
> On 01/31/13 06:21, Maxim Solodovnik wrote:
> Is the OM meetme table incomplete?
> My asterisk reports no issues :(
>
> could you provide me with missing fields and I'll add it.
> My purpose was to create table with required fields only.
>
>
> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx
> <ba...@telenet.be>> wrote:
> Openmeetings installed them for me, that's why I ended up with those.
> Using the Asterisk ones makes more sense to me. Maybe it's a good idea to have
> 'em removed from the install procedure.
>
> BC
>
>
> On 01/30/13 22:30, Jeff Clay wrote:
> Bart,
>
> If you look in the source directory of your asterisk tar file, under
> contrib/realtime/mysql you’ll find the .sql files required for all the
> realtime drivers. I never thought to use the ones with OM.
>
> Jeff Clay
> Network Administrator
> Infotech Enterprises America
> 870-215-5506
> Ext. 1506
>
> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
> Sent: Wednesday, January 30, 2013 3:19 PM
> To: user@openmeetings.apache.org<ma...@openmeetings.apache.org>
> Cc: Jeff Clay
> Subject: Re: SIP connectivity
>
> Well,
>
> I might have found one difference though:
>
> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
> dictates how the table should look like. I obviously used the one in the
> openmeetings mysql database, but this one seems to miss the table
> "useragent". I discovered this because it showed up in the logfiles.
>
> BC
>
> On 01/29/13 14:41, Jeff Clay wrote:
> Bart,
>
> From an asterisk configuration standpoint there are very few differences
> between 1.8.x and 11.x. If memory serves, the only major changes that I ran
> into (in my production environment) was changes to SIP NAT values and the
> behavior of app_page() now uses confbridge instead of meetme to mix the
> audio. Also, TCP, TLS and app_confbridge got a major overhauling. There were of
> course many other changes and bug fixes, you can skim through the change
> log for full details, but I think that was the jist of it.
>
>
>
> Jeff Clay
> Network Administrator
> Infotech Enterprises America
> 870-215-5506
> Ext. 1506
>
> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
> Sent: Tuesday, January 29, 2013 4:02 AM
> To: Maxim Solodovnik
> Cc: user
> Subject: Re: SIP connectivity
>
> I see - I'm willing to try the 11 version in the next fiew days if
> desired.
>
> BC
>
>
> On 01/29/13 10:57, Maxim Solodovnik wrote:
> I test the integration using
> Asterisk 1.8.13.1 (Ubuntu 12.10)
>
> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx
> <ba...@telenet.be>> wrote:
> That is amazing - I initially tried to do the same thing by using the new
> chan_motif driver in Asterisk 11 which connects to a XMPP server.
>
> Are you guys using Asterisk 11? This version is the newest LTS version and
> has the best video capabilities.
>
> Cheers,
>
> BC
>
>
> On 01/29/13 02:44, Maxim Solodovnik wrote:
> red5sip will create special OM user in the room: "SIP Transport"
> after that you can call to the OM room using SIP hard or soft phone.
>
> We are currently testing it and trying to add video capabilities ...
>
> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx
> <ba...@telenet.be>> wrote:
> Hi Jeff,
>
> In fact, I saw both pages, but none explain what they set up to do, just a
> bunch of command line instructions are given.
> Your "OM will create a meetme meeting as configured in the realtime meetme
> database" actually says it all in one go :-)
>
> cheers,
>
> BC
>
>
>
> On 01/28/13 22:38, Jeff Clay wrote:
> Bart,
>
> OM will create a meetme meeting as configured in the realtime meetme
> database. Have you read this page
> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
> http://openmeetings.apache.org/red5sip-integration.html but I assume this is the one
> you're already referring to.
>
> Jeff Clay
> Network Administrator
> Infotech Enterprises America
> 870-215-5506
> Ext. 1506
>
> -----Original Message-----
> From: Bart Coninckx
> [mailto:bart.coninckx@telenet.be<ma...@telenet.be>]
> Sent: Monday, January 28, 2013 3:36 PM
> To: user@openmeetings.apache.org<ma...@openmeetings.apache.org>
> Subject: SIP connectivity
>
> Hi,
>
> I noticed some documentation on how to connect OM with a SIP proxy or
> server, more particularly with the MeetMe application in Asterisk.
>
> The exact goal or purpose is not mentionned however. Will OM callout to a
> MeetMe conference? Or is it the other way round?
>
>
> Cheers,
>
> Bc
>
> ________________________________
>
> DISCLAIMER:
>
> This email may contain confidential information and is intended only for
> the use of the specific individual(s) to which it is addressed. If you are
> not the intended recipient of this email, you are hereby notified that any
> unauthorized use, dissemination or copying of this email or the information
> contained in it or attached to it is strictly prohibited. If you received
> this message in error, please immediately notify the sender at Infotech and
> delete the original message.
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
> --
> WBR
> Maxim aka solomax
Re: SIP connectivity
Posted by "Naderi, Sascha" <SN...@datus.com>.
Dear Maxim, dear all,
i tried it with the latest red5sip rev. (91) and it worked fine with a changed openmeetings context.
Thank you!
Regards
Sascha
________________________________
Von: Naderi, Sascha
Gesendet: Donnerstag, 14. Februar 2013 08:09
Bis: Maxim Solodovnik
Cc: user@openmeetings.apache.org
Betreff: AW: SIP connectivity
Dear Maxim,
OK, thanks a lot. I will check it out and leave feedback.
Regards
Sascha
________________________________
Von: Maxim Solodovnik [solomax666@gmail.com]
Gesendet: Mittwoch, 13. Februar 2013 23:58
Bis: Naderi, Sascha
Cc: user@openmeetings.apache.org
Betreff: Re: SIP connectivity
please try red5sip rev. 76
it has additional parameter: om.context
On Thu, Feb 14, 2013 at 2:59 AM, Naderi, Sascha <SN...@datus.com>> wrote:
Dear all,
i have tested the asterisk sip integration as documented with the most recent instruction (http://openmeetings.apache.org/red5sip-integration_2.1.html) and it works just fine.
The only thing i am missing is a way to get this working when i choose to rename the openmeetings context from http://yourcorp.com:5080/openmeetings to http://yourcorp.com:5080/yourmeetings
Which settings do i have to modify so that red5sip functions even if the context name is changed?
Regards
Sascha Naderi
________________________________
Von: Maxim Solodovnik [solomax666@gmail.com<ma...@gmail.com>]
Gesendet: Samstag, 9. Februar 2013 02:32
Bis: Bart Coninckx
Cc: user
Betreff: Re: SIP connectivity
All tables are created by OM automatically
On Feb 9, 2013 5:46 AM, "Bart Coninckx" <ba...@telenet.be>> wrote:
May I add that a portion is missing, since one explains how to configure Asterisk for Realtime, but one does not stipulate how to create the necessary tables.
It's in my CentOS docs however (which I hope to post shortly).
BC
On 01/31/13 13:05, Maxim Solodovnik wrote:
Hello Bart,
I just take a look at your URL ...
OM does not create/use sipfriends DB table (at least from version 2.1)
only meetme table is used
so I'm afraid there is nothing to change here
Here is the most recent instruction:
http://openmeetings.apache.org/red5sip-integration_2.1.html
Will ask our SIP guru to review it one more time :)
On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>> wrote:
OK will add it and notify you
On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>> wrote:
It is for Asterisk 11 - don't know for other versions. You probably have no issues because of the 1.8 version. To be sure the .sql files in the Asterisk source should be compared across versions.
this one is missing:
`useragent` varchar(20) DEFAULT NULL,
complete list (I think) is on:
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
If I bump into others, I'll report ASAP,
BC
On 01/31/13 06:21, Maxim Solodovnik wrote:
Is the OM meetme table incomplete?
My asterisk reports no issues :(
could you provide me with missing fields and I'll add it.
My purpose was to create table with required fields only.
On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <ba...@telenet.be>> wrote:
Openmeetings installed them for me, that's why I ended up with those. Using the Asterisk ones makes more sense to me. Maybe it's a good idea to have 'em removed from the install procedure.
BC
On 01/30/13 22:30, Jeff Clay wrote:
Bart,
If you look in the source directory of your asterisk tar file, under contrib/realtime/mysql you’ll find the .sql files required for all the realtime drivers. I never thought to use the ones with OM.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
Sent: Wednesday, January 30, 2013 3:19 PM
To: user@openmeetings.apache.org<ma...@openmeetings.apache.org>
Cc: Jeff Clay
Subject: Re: SIP connectivity
Well,
I might have found one difference though:
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure dictates how the table should look like. I obviously used the one in the openmeetings mysql database, but this one seems to miss the table "useragent". I discovered this because it showed up in the logfiles.
BC
On 01/29/13 14:41, Jeff Clay wrote:
Bart,
>From an asterisk configuration standpoint there are very few differences between 1.8.x and 11.x. If memory serves, the only major changes that I ran into (in my production environment) was changes to SIP NAT values and the behavior of app_page() now uses confbridge instead of meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major overhauling. There were of course many other changes and bug fixes, you can skim through the change log for full details, but I think that was the jist of it.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
Sent: Tuesday, January 29, 2013 4:02 AM
To: Maxim Solodovnik
Cc: user
Subject: Re: SIP connectivity
I see - I'm willing to try the 11 version in the next fiew days if desired.
BC
On 01/29/13 10:57, Maxim Solodovnik wrote:
I test the integration using
Asterisk 1.8.13.1 (Ubuntu 12.10)
On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <ba...@telenet.be>> wrote:
That is amazing - I initially tried to do the same thing by using the new chan_motif driver in Asterisk 11 which connects to a XMPP server.
Are you guys using Asterisk 11? This version is the newest LTS version and has the best video capabilities.
Cheers,
BC
On 01/29/13 02:44, Maxim Solodovnik wrote:
red5sip will create special OM user in the room: "SIP Transport"
after that you can call to the OM room using SIP hard or soft phone.
We are currently testing it and trying to add video capabilities ...
On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <ba...@telenet.be>> wrote:
Hi Jeff,
In fact, I saw both pages, but none explain what they set up to do, just a bunch of command line instructions are given.
Your "OM will create a meetme meeting as configured in the realtime meetme database" actually says it all in one go :-)
cheers,
BC
On 01/28/13 22:38, Jeff Clay wrote:
Bart,
OM will create a meetme meeting as configured in the realtime meetme database. Have you read this page https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out http://openmeetings.apache.org/red5sip-integration.html but I assume this is the one you're already referring to.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
-----Original Message-----
From: Bart Coninckx [mailto:bart.coninckx@telenet.be<ma...@telenet.be>]
Sent: Monday, January 28, 2013 3:36 PM
To: user@openmeetings.apache.org<ma...@openmeetings.apache.org>
Subject: SIP connectivity
Hi,
I noticed some documentation on how to connect OM with a SIP proxy or server, more particularly with the MeetMe application in Asterisk.
The exact goal or purpose is not mentionned however. Will OM callout to a MeetMe conference? Or is it the other way round?
Cheers,
Bc
________________________________
DISCLAIMER:
This email may contain confidential information and is intended only for the use of the specific individual(s) to which it is addressed. If you are not the intended recipient of this email, you are hereby notified that any unauthorized use, dissemination or copying of this email or the information contained in it or attached to it is strictly prohibited. If you received this message in error, please immediately notify the sender at Infotech and delete the original message.
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
AW: SIP connectivity
Posted by "Naderi, Sascha" <SN...@datus.com>.
Dear Maxim,
OK, thanks a lot. I will check it out and leave feedback.
Regards
Sascha
________________________________
Von: Maxim Solodovnik [solomax666@gmail.com]
Gesendet: Mittwoch, 13. Februar 2013 23:58
Bis: Naderi, Sascha
Cc: user@openmeetings.apache.org
Betreff: Re: SIP connectivity
please try red5sip rev. 76
it has additional parameter: om.context
On Thu, Feb 14, 2013 at 2:59 AM, Naderi, Sascha <SN...@datus.com>> wrote:
Dear all,
i have tested the asterisk sip integration as documented with the most recent instruction (http://openmeetings.apache.org/red5sip-integration_2.1.html) and it works just fine.
The only thing i am missing is a way to get this working when i choose to rename the openmeetings context from http://yourcorp.com:5080/openmeetings to http://yourcorp.com:5080/yourmeetings
Which settings do i have to modify so that red5sip functions even if the context name is changed?
Regards
Sascha Naderi
________________________________
Von: Maxim Solodovnik [solomax666@gmail.com<ma...@gmail.com>]
Gesendet: Samstag, 9. Februar 2013 02:32
Bis: Bart Coninckx
Cc: user
Betreff: Re: SIP connectivity
All tables are created by OM automatically
On Feb 9, 2013 5:46 AM, "Bart Coninckx" <ba...@telenet.be>> wrote:
May I add that a portion is missing, since one explains how to configure Asterisk for Realtime, but one does not stipulate how to create the necessary tables.
It's in my CentOS docs however (which I hope to post shortly).
BC
On 01/31/13 13:05, Maxim Solodovnik wrote:
Hello Bart,
I just take a look at your URL ...
OM does not create/use sipfriends DB table (at least from version 2.1)
only meetme table is used
so I'm afraid there is nothing to change here
Here is the most recent instruction:
http://openmeetings.apache.org/red5sip-integration_2.1.html
Will ask our SIP guru to review it one more time :)
On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>> wrote:
OK will add it and notify you
On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>> wrote:
It is for Asterisk 11 - don't know for other versions. You probably have no issues because of the 1.8 version. To be sure the .sql files in the Asterisk source should be compared across versions.
this one is missing:
`useragent` varchar(20) DEFAULT NULL,
complete list (I think) is on:
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
If I bump into others, I'll report ASAP,
BC
On 01/31/13 06:21, Maxim Solodovnik wrote:
Is the OM meetme table incomplete?
My asterisk reports no issues :(
could you provide me with missing fields and I'll add it.
My purpose was to create table with required fields only.
On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <ba...@telenet.be>> wrote:
Openmeetings installed them for me, that's why I ended up with those. Using the Asterisk ones makes more sense to me. Maybe it's a good idea to have 'em removed from the install procedure.
BC
On 01/30/13 22:30, Jeff Clay wrote:
Bart,
If you look in the source directory of your asterisk tar file, under contrib/realtime/mysql you’ll find the .sql files required for all the realtime drivers. I never thought to use the ones with OM.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
Sent: Wednesday, January 30, 2013 3:19 PM
To: user@openmeetings.apache.org<ma...@openmeetings.apache.org>
Cc: Jeff Clay
Subject: Re: SIP connectivity
Well,
I might have found one difference though:
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure dictates how the table should look like. I obviously used the one in the openmeetings mysql database, but this one seems to miss the table "useragent". I discovered this because it showed up in the logfiles.
BC
On 01/29/13 14:41, Jeff Clay wrote:
Bart,
>From an asterisk configuration standpoint there are very few differences between 1.8.x and 11.x. If memory serves, the only major changes that I ran into (in my production environment) was changes to SIP NAT values and the behavior of app_page() now uses confbridge instead of meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major overhauling. There were of course many other changes and bug fixes, you can skim through the change log for full details, but I think that was the jist of it.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
Sent: Tuesday, January 29, 2013 4:02 AM
To: Maxim Solodovnik
Cc: user
Subject: Re: SIP connectivity
I see - I'm willing to try the 11 version in the next fiew days if desired.
BC
On 01/29/13 10:57, Maxim Solodovnik wrote:
I test the integration using
Asterisk 1.8.13.1 (Ubuntu 12.10)
On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <ba...@telenet.be>> wrote:
That is amazing - I initially tried to do the same thing by using the new chan_motif driver in Asterisk 11 which connects to a XMPP server.
Are you guys using Asterisk 11? This version is the newest LTS version and has the best video capabilities.
Cheers,
BC
On 01/29/13 02:44, Maxim Solodovnik wrote:
red5sip will create special OM user in the room: "SIP Transport"
after that you can call to the OM room using SIP hard or soft phone.
We are currently testing it and trying to add video capabilities ...
On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <ba...@telenet.be>> wrote:
Hi Jeff,
In fact, I saw both pages, but none explain what they set up to do, just a bunch of command line instructions are given.
Your "OM will create a meetme meeting as configured in the realtime meetme database" actually says it all in one go :-)
cheers,
BC
On 01/28/13 22:38, Jeff Clay wrote:
Bart,
OM will create a meetme meeting as configured in the realtime meetme database. Have you read this page https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out http://openmeetings.apache.org/red5sip-integration.html but I assume this is the one you're already referring to.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
-----Original Message-----
From: Bart Coninckx [mailto:bart.coninckx@telenet.be<ma...@telenet.be>]
Sent: Monday, January 28, 2013 3:36 PM
To: user@openmeetings.apache.org<ma...@openmeetings.apache.org>
Subject: SIP connectivity
Hi,
I noticed some documentation on how to connect OM with a SIP proxy or server, more particularly with the MeetMe application in Asterisk.
The exact goal or purpose is not mentionned however. Will OM callout to a MeetMe conference? Or is it the other way round?
Cheers,
Bc
________________________________
DISCLAIMER:
This email may contain confidential information and is intended only for the use of the specific individual(s) to which it is addressed. If you are not the intended recipient of this email, you are hereby notified that any unauthorized use, dissemination or copying of this email or the information contained in it or attached to it is strictly prohibited. If you received this message in error, please immediately notify the sender at Infotech and delete the original message.
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: SIP connectivity
Posted by Maxim Solodovnik <so...@gmail.com>.
please try red5sip rev. 76
it has additional parameter: om.context
On Thu, Feb 14, 2013 at 2:59 AM, Naderi, Sascha <SN...@datus.com> wrote:
> Dear all, ** ****
>
> ****
>
> ****
>
> ****
>
> i have tested the asterisk sip integration as documented with the most
> recent instruction (
> http://openmeetings.apache.org/red5sip-integration_2.1.html) and it worksjust fine
> .****
>
> ****
>
> The only thing i am missing is a way to get this working when i choose to
> rename the openmeetings context from
> http://yourcorp.com:5080/openmeetings to
> http://yourcorp.com:5080/yourmeetings ****
>
> ****
>
> Which settings do i have to modify so that red5sip functions even if the
> context name is changed?****
>
> ** **
>
>
> Regards
> Sascha Naderi
>
>
> ------------------------------
>
> *Von:* Maxim Solodovnik [solomax666@gmail.com]
> *Gesendet:* Samstag, 9. Februar 2013 02:32
> *Bis:* Bart Coninckx
> *Cc:* user
> *Betreff:* Re: SIP connectivity
>
> All tables are created by OM automatically
> On Feb 9, 2013 5:46 AM, "Bart Coninckx" <ba...@telenet.be> wrote:
>
>> May I add that a portion is missing, since one explains how to
>> configure Asterisk for Realtime, but one does not stipulate how to create
>> the necessary tables.
>> It's in my CentOS docs however (which I hope to post shortly).
>>
>> BC
>>
>> On 01/31/13 13:05, Maxim Solodovnik wrote:
>>
>> Hello Bart,
>>
>> I just take a look at your URL ...
>> OM does not create/use sipfriends DB table (at least from version 2.1)
>> only meetme table is used
>>
>> so I'm afraid there is nothing to change here
>>
>> Here is the most recent instruction:
>> http://openmeetings.apache.org/red5sip-integration_2.1.html
>>
>> Will ask our SIP guru to review it one more time :)
>>
>>
>>
>> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>>
>>> OK will add it and notify you
>>> On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>
>>> wrote:
>>>
>>>> It is for Asterisk 11 - don't know for other versions. You probably
>>>> have no issues because of the 1.8 version. To be sure the .sql files in the
>>>> Asterisk source should be compared across versions.
>>>>
>>>> this one is missing:
>>>>
>>>> `useragent` varchar(20) DEFAULT NULL,
>>>>
>>>> complete list (I think) is on:
>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>>
>>>>
>>>> If I bump into others, I'll report ASAP,
>>>>
>>>>
>>>> BC
>>>>
>>>>
>>>>
>>>> On 01/31/13 06:21, Maxim Solodovnik wrote:
>>>>
>>>> Is the OM meetme table incomplete?
>>>> My asterisk reports no issues :(
>>>>
>>>> could you provide me with missing fields and I'll add it.
>>>> My purpose was to create table with required fields only.
>>>>
>>>>
>>>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <
>>>> bart.coninckx@telenet.be> wrote:
>>>>
>>>>> Openmeetings installed them for me, that's why I ended up with
>>>>> those. Using the Asterisk ones makes more sense to me. Maybe it's a good
>>>>> idea to have 'em removed from the install procedure.
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>> On 01/30/13 22:30, Jeff Clay wrote:
>>>>>
>>>>> Bart,
>>>>>
>>>>>
>>>>>
>>>>> If you look in the source directory of your asterisk tar file, under
>>>>> contrib/realtime/mysql you’ll find the .sql files required for all the
>>>>> realtime drivers. I never thought to use the ones with OM.
>>>>>
>>>>>
>>>>>
>>>>> Jeff Clay
>>>>>
>>>>> Network Administrator
>>>>>
>>>>> Infotech Enterprises America
>>>>>
>>>>> 870-215-5506
>>>>>
>>>>> Ext. 1506
>>>>>
>>>>>
>>>>>
>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>>
>>>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>>>> *To:* user@openmeetings.apache.org
>>>>> *Cc:* Jeff Clay
>>>>> *Subject:* Re: SIP connectivity
>>>>>
>>>>>
>>>>>
>>>>> Well,
>>>>>
>>>>> I might have found one difference though:
>>>>>
>>>>>
>>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>>> dictates how the table should look like. I obviously used the one in the
>>>>> openmeetings mysql database, but this one seems to miss the table
>>>>> "useragent". I discovered this because it showed up in the logfiles.
>>>>>
>>>>> BC
>>>>>
>>>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>>>
>>>>> Bart,
>>>>>
>>>>>
>>>>>
>>>>> From an asterisk configuration standpoint there are very few
>>>>> differences between 1.8.x and 11.x. If memory serves, the only major
>>>>> changes that I ran into (in my production environment) was changes to SIP
>>>>> NAT values and the behavior of app_page() now uses confbridge instead of
>>>>> meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major
>>>>> overhauling. There were of course many other changes and bug fixes, you can
>>>>> skim through the change log for full details, but I think that was the jist
>>>>> of it.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Jeff Clay
>>>>>
>>>>> Network Administrator
>>>>>
>>>>> Infotech Enterprises America
>>>>>
>>>>> 870-215-5506
>>>>>
>>>>> Ext. 1506
>>>>>
>>>>>
>>>>>
>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>>
>>>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>>>> *To:* Maxim Solodovnik
>>>>> *Cc:* user
>>>>> *Subject:* Re: SIP connectivity
>>>>>
>>>>>
>>>>>
>>>>> I see - I'm willing to try the 11 version in the next fiew days if
>>>>> desired.
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>>>
>>>>> I test the integration using
>>>>>
>>>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <
>>>>> bart.coninckx@telenet.be> wrote:
>>>>>
>>>>> That is amazing - I initially tried to do the same thing by using the
>>>>> new chan_motif driver in Asterisk 11 which connects to a XMPP server.
>>>>>
>>>>> Are you guys using Asterisk 11? This version is the newest LTS version
>>>>> and has the best video capabilities.
>>>>>
>>>>> Cheers,
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>>>
>>>>> red5sip will create special OM user in the room: "SIP Transport"
>>>>>
>>>>> after that you can call to the OM room using SIP hard or soft phone.
>>>>>
>>>>>
>>>>>
>>>>> We are currently testing it and trying to add video capabilities ...
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <
>>>>> bart.coninckx@telenet.be> wrote:
>>>>>
>>>>> Hi Jeff,
>>>>>
>>>>> In fact, I saw both pages, but none explain what they set up to do,
>>>>> just a bunch of command line instructions are given.
>>>>> Your "OM will create a meetme meeting as configured in the realtime
>>>>> meetme database" actually says it all in one go :-)
>>>>>
>>>>> cheers,
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>>>
>>>>> Bart,
>>>>>
>>>>> OM will create a meetme meeting as configured in the realtime meetme
>>>>> database. Have you read this page
>>>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
>>>>> http://openmeetings.apache.org/red5sip-integration.html but I assume
>>>>> this is the one you're already referring to.
>>>>>
>>>>> Jeff Clay
>>>>> Network Administrator
>>>>> Infotech Enterprises America
>>>>> 870-215-5506
>>>>> Ext. 1506
>>>>>
>>>>> -----Original Message-----
>>>>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>>>> Sent: Monday, January 28, 2013 3:36 PM
>>>>> To: user@openmeetings.apache.org
>>>>> Subject: SIP connectivity
>>>>>
>>>>> Hi,
>>>>>
>>>>> I noticed some documentation on how to connect OM with a SIP proxy or
>>>>> server, more particularly with the MeetMe application in Asterisk.
>>>>>
>>>>> The exact goal or purpose is not mentionned however. Will OM callout
>>>>> to a MeetMe conference? Or is it the other way round?
>>>>>
>>>>>
>>>>> Cheers,
>>>>>
>>>>> Bc
>>>>>
>>>>> ________________________________
>>>>>
>>>>> DISCLAIMER:
>>>>>
>>>>> This email may contain confidential information and is intended only
>>>>> for the use of the specific individual(s) to which it is addressed. If you
>>>>> are not the intended recipient of this email, you are hereby notified that
>>>>> any unauthorized use, dissemination or copying of this email or the
>>>>> information contained in it or attached to it is strictly prohibited. If
>>>>> you received this message in error, please immediately notify the sender at
>>>>> Infotech and delete the original message.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> WBR
>>>>> Maxim aka solomax
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> WBR
>>>>> Maxim aka solomax
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>>
>>>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
--
WBR
Maxim aka solomax
Re: SIP connectivity
Posted by "Naderi, Sascha" <SN...@datus.com>.
Dear all,
i have tested the asterisk sip integration as documented with the most recent instruction (http://openmeetings.apache.org/red5sip-integration_2.1.html) and it works just fine.
The only thing i am missing is a way to get this working when i choose to rename the openmeetings context from http://yourcorp.com:5080/openmeetings to http://yourcorp.com:5080/yourmeetings
Which settings do i have to modify so that red5sip functions even if the context name is changed?
Regards
Sascha Naderi
________________________________
Von: Maxim Solodovnik [solomax666@gmail.com]
Gesendet: Samstag, 9. Februar 2013 02:32
Bis: Bart Coninckx
Cc: user
Betreff: Re: SIP connectivity
All tables are created by OM automatically
On Feb 9, 2013 5:46 AM, "Bart Coninckx" <ba...@telenet.be>> wrote:
May I add that a portion is missing, since one explains how to configure Asterisk for Realtime, but one does not stipulate how to create the necessary tables.
It's in my CentOS docs however (which I hope to post shortly).
BC
On 01/31/13 13:05, Maxim Solodovnik wrote:
Hello Bart,
I just take a look at your URL ...
OM does not create/use sipfriends DB table (at least from version 2.1)
only meetme table is used
so I'm afraid there is nothing to change here
Here is the most recent instruction:
http://openmeetings.apache.org/red5sip-integration_2.1.html
Will ask our SIP guru to review it one more time :)
On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>> wrote:
OK will add it and notify you
On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>> wrote:
It is for Asterisk 11 - don't know for other versions. You probably have no issues because of the 1.8 version. To be sure the .sql files in the Asterisk source should be compared across versions.
this one is missing:
`useragent` varchar(20) DEFAULT NULL,
complete list (I think) is on:
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
If I bump into others, I'll report ASAP,
BC
On 01/31/13 06:21, Maxim Solodovnik wrote:
Is the OM meetme table incomplete?
My asterisk reports no issues :(
could you provide me with missing fields and I'll add it.
My purpose was to create table with required fields only.
On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <ba...@telenet.be>> wrote:
Openmeetings installed them for me, that's why I ended up with those. Using the Asterisk ones makes more sense to me. Maybe it's a good idea to have 'em removed from the install procedure.
BC
On 01/30/13 22:30, Jeff Clay wrote:
Bart,
If you look in the source directory of your asterisk tar file, under contrib/realtime/mysql you’ll find the .sql files required for all the realtime drivers. I never thought to use the ones with OM.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
Sent: Wednesday, January 30, 2013 3:19 PM
To: user@openmeetings.apache.org<ma...@openmeetings.apache.org>
Cc: Jeff Clay
Subject: Re: SIP connectivity
Well,
I might have found one difference though:
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure dictates how the table should look like. I obviously used the one in the openmeetings mysql database, but this one seems to miss the table "useragent". I discovered this because it showed up in the logfiles.
BC
On 01/29/13 14:41, Jeff Clay wrote:
Bart,
>From an asterisk configuration standpoint there are very few differences between 1.8.x and 11.x. If memory serves, the only major changes that I ran into (in my production environment) was changes to SIP NAT values and the behavior of app_page() now uses confbridge instead of meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major overhauling. There were of course many other changes and bug fixes, you can skim through the change log for full details, but I think that was the jist of it.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
Sent: Tuesday, January 29, 2013 4:02 AM
To: Maxim Solodovnik
Cc: user
Subject: Re: SIP connectivity
I see - I'm willing to try the 11 version in the next fiew days if desired.
BC
On 01/29/13 10:57, Maxim Solodovnik wrote:
I test the integration using
Asterisk 1.8.13.1 (Ubuntu 12.10)
On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <ba...@telenet.be>> wrote:
That is amazing - I initially tried to do the same thing by using the new chan_motif driver in Asterisk 11 which connects to a XMPP server.
Are you guys using Asterisk 11? This version is the newest LTS version and has the best video capabilities.
Cheers,
BC
On 01/29/13 02:44, Maxim Solodovnik wrote:
red5sip will create special OM user in the room: "SIP Transport"
after that you can call to the OM room using SIP hard or soft phone.
We are currently testing it and trying to add video capabilities ...
On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <ba...@telenet.be>> wrote:
Hi Jeff,
In fact, I saw both pages, but none explain what they set up to do, just a bunch of command line instructions are given.
Your "OM will create a meetme meeting as configured in the realtime meetme database" actually says it all in one go :-)
cheers,
BC
On 01/28/13 22:38, Jeff Clay wrote:
Bart,
OM will create a meetme meeting as configured in the realtime meetme database. Have you read this page https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out http://openmeetings.apache.org/red5sip-integration.html but I assume this is the one you're already referring to.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
-----Original Message-----
From: Bart Coninckx [mailto:bart.coninckx@telenet.be<ma...@telenet.be>]
Sent: Monday, January 28, 2013 3:36 PM
To: user@openmeetings.apache.org<ma...@openmeetings.apache.org>
Subject: SIP connectivity
Hi,
I noticed some documentation on how to connect OM with a SIP proxy or server, more particularly with the MeetMe application in Asterisk.
The exact goal or purpose is not mentionned however. Will OM callout to a MeetMe conference? Or is it the other way round?
Cheers,
Bc
________________________________
DISCLAIMER:
This email may contain confidential information and is intended only for the use of the specific individual(s) to which it is addressed. If you are not the intended recipient of this email, you are hereby notified that any unauthorized use, dissemination or copying of this email or the information contained in it or attached to it is strictly prohibited. If you received this message in error, please immediately notify the sender at Infotech and delete the original message.
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: SIP connectivity
Posted by Maxim Solodovnik <so...@gmail.com>.
All tables are created by OM automatically
On Feb 9, 2013 5:46 AM, "Bart Coninckx" <ba...@telenet.be> wrote:
> May I add that a portion is missing, since one explains how to configure
> Asterisk for Realtime, but one does not stipulate how to create the
> necessary tables.
> It's in my CentOS docs however (which I hope to post shortly).
>
> BC
>
> On 01/31/13 13:05, Maxim Solodovnik wrote:
>
> Hello Bart,
>
> I just take a look at your URL ...
> OM does not create/use sipfriends DB table (at least from version 2.1)
> only meetme table is used
>
> so I'm afraid there is nothing to change here
>
> Here is the most recent instruction:
> http://openmeetings.apache.org/red5sip-integration_2.1.html
>
> Will ask our SIP guru to review it one more time :)
>
>
>
> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>
>> OK will add it and notify you
>> On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>
>> wrote:
>>
>>> It is for Asterisk 11 - don't know for other versions. You probably
>>> have no issues because of the 1.8 version. To be sure the .sql files in the
>>> Asterisk source should be compared across versions.
>>>
>>> this one is missing:
>>>
>>> `useragent` varchar(20) DEFAULT NULL,
>>>
>>> complete list (I think) is on:
>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>
>>>
>>> If I bump into others, I'll report ASAP,
>>>
>>>
>>> BC
>>>
>>>
>>>
>>> On 01/31/13 06:21, Maxim Solodovnik wrote:
>>>
>>> Is the OM meetme table incomplete?
>>> My asterisk reports no issues :(
>>>
>>> could you provide me with missing fields and I'll add it.
>>> My purpose was to create table with required fields only.
>>>
>>>
>>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <bart.coninckx@telenet.be
>>> > wrote:
>>>
>>>> Openmeetings installed them for me, that's why I ended up with those.
>>>> Using the Asterisk ones makes more sense to me. Maybe it's a good idea to
>>>> have 'em removed from the install procedure.
>>>>
>>>> BC
>>>>
>>>>
>>>> On 01/30/13 22:30, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>>
>>>>
>>>> If you look in the source directory of your asterisk tar file, under
>>>> contrib/realtime/mysql you’ll find the .sql files required for all the
>>>> realtime drivers. I never thought to use the ones with OM.
>>>>
>>>>
>>>>
>>>> Jeff Clay
>>>>
>>>> Network Administrator
>>>>
>>>> Infotech Enterprises America
>>>>
>>>> 870-215-5506
>>>>
>>>> Ext. 1506
>>>>
>>>>
>>>>
>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>
>>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>>> *To:* user@openmeetings.apache.org
>>>> *Cc:* Jeff Clay
>>>> *Subject:* Re: SIP connectivity
>>>>
>>>>
>>>>
>>>> Well,
>>>>
>>>> I might have found one difference though:
>>>>
>>>>
>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>> dictates how the table should look like. I obviously used the one in the
>>>> openmeetings mysql database, but this one seems to miss the table
>>>> "useragent". I discovered this because it showed up in the logfiles.
>>>>
>>>> BC
>>>>
>>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>>
>>>>
>>>> From an asterisk configuration standpoint there are very few
>>>> differences between 1.8.x and 11.x. If memory serves, the only major
>>>> changes that I ran into (in my production environment) was changes to SIP
>>>> NAT values and the behavior of app_page() now uses confbridge instead of
>>>> meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major
>>>> overhauling. There were of course many other changes and bug fixes, you can
>>>> skim through the change log for full details, but I think that was the jist
>>>> of it.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Jeff Clay
>>>>
>>>> Network Administrator
>>>>
>>>> Infotech Enterprises America
>>>>
>>>> 870-215-5506
>>>>
>>>> Ext. 1506
>>>>
>>>>
>>>>
>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>
>>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>>> *To:* Maxim Solodovnik
>>>> *Cc:* user
>>>> *Subject:* Re: SIP connectivity
>>>>
>>>>
>>>>
>>>> I see - I'm willing to try the 11 version in the next fiew days if
>>>> desired.
>>>>
>>>> BC
>>>>
>>>>
>>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>>
>>>> I test the integration using
>>>>
>>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>>
>>>>
>>>>
>>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <
>>>> bart.coninckx@telenet.be> wrote:
>>>>
>>>> That is amazing - I initially tried to do the same thing by using the
>>>> new chan_motif driver in Asterisk 11 which connects to a XMPP server.
>>>>
>>>> Are you guys using Asterisk 11? This version is the newest LTS version
>>>> and has the best video capabilities.
>>>>
>>>> Cheers,
>>>>
>>>> BC
>>>>
>>>>
>>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>>
>>>> red5sip will create special OM user in the room: "SIP Transport"
>>>>
>>>> after that you can call to the OM room using SIP hard or soft phone.
>>>>
>>>>
>>>>
>>>> We are currently testing it and trying to add video capabilities ...
>>>>
>>>>
>>>>
>>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <
>>>> bart.coninckx@telenet.be> wrote:
>>>>
>>>> Hi Jeff,
>>>>
>>>> In fact, I saw both pages, but none explain what they set up to do,
>>>> just a bunch of command line instructions are given.
>>>> Your "OM will create a meetme meeting as configured in the realtime
>>>> meetme database" actually says it all in one go :-)
>>>>
>>>> cheers,
>>>>
>>>> BC
>>>>
>>>>
>>>>
>>>>
>>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>> OM will create a meetme meeting as configured in the realtime meetme
>>>> database. Have you read this page
>>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
>>>> http://openmeetings.apache.org/red5sip-integration.html but I assume
>>>> this is the one you're already referring to.
>>>>
>>>> Jeff Clay
>>>> Network Administrator
>>>> Infotech Enterprises America
>>>> 870-215-5506
>>>> Ext. 1506
>>>>
>>>> -----Original Message-----
>>>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>>> Sent: Monday, January 28, 2013 3:36 PM
>>>> To: user@openmeetings.apache.org
>>>> Subject: SIP connectivity
>>>>
>>>> Hi,
>>>>
>>>> I noticed some documentation on how to connect OM with a SIP proxy or
>>>> server, more particularly with the MeetMe application in Asterisk.
>>>>
>>>> The exact goal or purpose is not mentionned however. Will OM callout to
>>>> a MeetMe conference? Or is it the other way round?
>>>>
>>>>
>>>> Cheers,
>>>>
>>>> Bc
>>>>
>>>> ________________________________
>>>>
>>>> DISCLAIMER:
>>>>
>>>> This email may contain confidential information and is intended only
>>>> for the use of the specific individual(s) to which it is addressed. If you
>>>> are not the intended recipient of this email, you are hereby notified that
>>>> any unauthorized use, dissemination or copying of this email or the
>>>> information contained in it or attached to it is strictly prohibited. If
>>>> you received this message in error, please immediately notify the sender at
>>>> Infotech and delete the original message.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>
>>>
>>> --
>>> WBR
>>> Maxim aka solomax
>>>
>>>
>>>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
Re: SIP connectivity
Posted by Bart Coninckx <ba...@telenet.be>.
May I add that a portion is missing, since one explains how to configure
Asterisk for Realtime, but one does not stipulate how to create the
necessary tables.
It's in my CentOS docs however (which I hope to post shortly).
BC
On 01/31/13 13:05, Maxim Solodovnik wrote:
> Hello Bart,
>
> I just take a look at your URL ...
> OM does not create/use sipfriends DB table (at least from version 2.1)
> only meetme table is used
>
> so I'm afraid there is nothing to change here
>
> Here is the most recent instruction:
> http://openmeetings.apache.org/red5sip-integration_2.1.html
>
> Will ask our SIP guru to review it one more time :)
>
>
>
> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik
> <solomax666@gmail.com <ma...@gmail.com>> wrote:
>
> OK will add it and notify you
>
> On Jan 31, 2013 5:05 PM, "Bart Coninckx" <bart.coninckx@telenet.be
> <ma...@telenet.be>> wrote:
>
> It is for Asterisk 11 - don't know for other versions. You
> probably have no issues because of the 1.8 version. To be sure
> the .sql files in the Asterisk source should be compared
> across versions.
>
> this one is missing:
>
> `useragent` varchar(20) DEFAULT NULL,
>
> complete list (I think) is on:
>
> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>
>
> If I bump into others, I'll report ASAP,
>
>
> BC
>
>
>
> On 01/31/13 06:21, Maxim Solodovnik wrote:
>> Is the OM meetme table incomplete?
>> My asterisk reports no issues :(
>>
>> could you provide me with missing fields and I'll add it.
>> My purpose was to create table with required fields only.
>>
>>
>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx
>> <bart.coninckx@telenet.be <ma...@telenet.be>>
>> wrote:
>>
>> Openmeetings installed them for me, that's why I ended up
>> with those. Using the Asterisk ones makes more sense to
>> me. Maybe it's a good idea to have 'em removed from the
>> install procedure.
>>
>> BC
>>
>>
>> On 01/30/13 22:30, Jeff Clay wrote:
>>>
>>> Bart,
>>>
>>> If you look in the source directory of your asterisk tar
>>> file, under contrib/realtime/mysql you’ll find the .sql
>>> files required for all the realtime drivers. I never
>>> thought to use the ones with OM.
>>>
>>> Jeff Clay
>>>
>>> Network Administrator
>>>
>>> Infotech Enterprises America
>>>
>>> 870-215-5506
>>>
>>> Ext. 1506
>>>
>>> *From:*Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>> *To:* user@openmeetings.apache.org
>>> <ma...@openmeetings.apache.org>
>>> *Cc:* Jeff Clay
>>> *Subject:* Re: SIP connectivity
>>>
>>> Well,
>>>
>>> I might have found one difference though:
>>>
>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>> dictates how the table should look like. I obviously
>>> used the one in the openmeetings mysql database, but
>>> this one seems to miss the table "useragent". I
>>> discovered this because it showed up in the logfiles.
>>>
>>> BC
>>>
>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>
>>> Bart,
>>>
>>> From an asterisk configuration standpoint there are
>>> very few differences between 1.8.x and 11.x. If
>>> memory serves, the only major changes that I ran
>>> into (in my production environment) was changes to
>>> SIP NAT values and the behavior of app_page() now
>>> uses confbridge instead of meetme to mix the audio.
>>> Also, TCP, TLS and app_confbridge got a major
>>> overhauling. There were of course many other changes
>>> and bug fixes, you can skim through the change log
>>> for full details, but I think that was the jist of it.
>>>
>>> Jeff Clay
>>>
>>> Network Administrator
>>>
>>> Infotech Enterprises America
>>>
>>> 870-215-5506
>>>
>>> Ext. 1506
>>>
>>> *From:*Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>> *To:* Maxim Solodovnik
>>> *Cc:* user
>>> *Subject:* Re: SIP connectivity
>>>
>>> I see - I'm willing to try the 11 version in the
>>> next fiew days if desired.
>>>
>>> BC
>>>
>>>
>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>
>>> I test the integration using
>>>
>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>
>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx
>>> <bart.coninckx@telenet.be
>>> <ma...@telenet.be>> wrote:
>>>
>>> That is amazing - I initially tried to do the
>>> same thing by using the new chan_motif driver in
>>> Asterisk 11 which connects to a XMPP server.
>>>
>>> Are you guys using Asterisk 11? This version is
>>> the newest LTS version and has the best video
>>> capabilities.
>>>
>>> Cheers,
>>>
>>> BC
>>>
>>>
>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>
>>> red5sip will create special OM user in the
>>> room: "SIP Transport"
>>>
>>> after that you can call to the OM room using
>>> SIP hard or soft phone.
>>>
>>> We are currently testing it and trying to
>>> add video capabilities ...
>>>
>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart
>>> Coninckx <bart.coninckx@telenet.be
>>> <ma...@telenet.be>> wrote:
>>>
>>> Hi Jeff,
>>>
>>> In fact, I saw both pages, but none explain
>>> what they set up to do, just a bunch of
>>> command line instructions are given.
>>> Your "OM will create a meetme meeting as
>>> configured in the realtime meetme database"
>>> actually says it all in one go :-)
>>>
>>> cheers,
>>>
>>> BC
>>>
>>>
>>>
>>>
>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>
>>> Bart,
>>>
>>> OM will create a meetme meeting as
>>> configured in the realtime meetme database.
>>> Have you read this page
>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html
>>> ? You might also check out
>>> http://openmeetings.apache.org/red5sip-integration.html
>>> but I assume this is the one you're already
>>> referring to.
>>>
>>> Jeff Clay
>>> Network Administrator
>>> Infotech Enterprises America
>>> 870-215-5506
>>> Ext. 1506
>>>
>>> -----Original Message-----
>>> From: Bart Coninckx
>>> [mailto:bart.coninckx@telenet.be
>>> <ma...@telenet.be>]
>>> Sent: Monday, January 28, 2013 3:36 PM
>>> To: user@openmeetings.apache.org
>>> <ma...@openmeetings.apache.org>
>>> Subject: SIP connectivity
>>>
>>> Hi,
>>>
>>> I noticed some documentation on how to
>>> connect OM with a SIP proxy or server, more
>>> particularly with the MeetMe application in
>>> Asterisk.
>>>
>>> The exact goal or purpose is not mentionned
>>> however. Will OM callout to a MeetMe
>>> conference? Or is it the other way round?
>>>
>>>
>>> Cheers,
>>>
>>> Bc
>>>
>>> ________________________________
>>>
>>> DISCLAIMER:
>>>
>>> This email may contain confidential
>>> information and is intended only for the use
>>> of the specific individual(s) to which it is
>>> addressed. If you are not the intended
>>> recipient of this email, you are hereby
>>> notified that any unauthorized use,
>>> dissemination or copying of this email or
>>> the information contained in it or attached
>>> to it is strictly prohibited. If you
>>> received this message in error, please
>>> immediately notify the sender at Infotech
>>> and delete the original message.
>>>
>>>
>>>
>>> --
>>> WBR
>>> Maxim aka solomax
>>>
>>>
>>>
>>> --
>>> WBR
>>> Maxim aka solomax
>>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>
>
>
>
> --
> WBR
> Maxim aka solomax
Re: SIP connectivity
Posted by Maxim Solodovnik <so...@gmail.com>.
We are currently working on SIP and cluster ....
I believe it is stable :)
On Thu, Jan 31, 2013 at 7:12 PM, Bart Coninckx <ba...@telenet.be>wrote:
> OK, I suppose these instructions supersede the ones on:
>
>
> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html?
>
> what is the status of OM 2.1? Is it production stable?
>
> cheers,
>
>
> BC
>
>
>
> On 01/31/13 13:05, Maxim Solodovnik wrote:
>
> Hello Bart,
>
> I just take a look at your URL ...
> OM does not create/use sipfriends DB table (at least from version 2.1)
> only meetme table is used
>
> so I'm afraid there is nothing to change here
>
> Here is the most recent instruction:
> http://openmeetings.apache.org/red5sip-integration_2.1.html
>
> Will ask our SIP guru to review it one more time :)
>
>
>
> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>
>> OK will add it and notify you
>> On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>
>> wrote:
>>
>>> It is for Asterisk 11 - don't know for other versions. You probably
>>> have no issues because of the 1.8 version. To be sure the .sql files in the
>>> Asterisk source should be compared across versions.
>>>
>>> this one is missing:
>>>
>>> `useragent` varchar(20) DEFAULT NULL,
>>>
>>> complete list (I think) is on:
>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>
>>>
>>> If I bump into others, I'll report ASAP,
>>>
>>>
>>> BC
>>>
>>>
>>>
>>> On 01/31/13 06:21, Maxim Solodovnik wrote:
>>>
>>> Is the OM meetme table incomplete?
>>> My asterisk reports no issues :(
>>>
>>> could you provide me with missing fields and I'll add it.
>>> My purpose was to create table with required fields only.
>>>
>>>
>>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <bart.coninckx@telenet.be
>>> > wrote:
>>>
>>>> Openmeetings installed them for me, that's why I ended up with those.
>>>> Using the Asterisk ones makes more sense to me. Maybe it's a good idea to
>>>> have 'em removed from the install procedure.
>>>>
>>>> BC
>>>>
>>>>
>>>> On 01/30/13 22:30, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>>
>>>>
>>>> If you look in the source directory of your asterisk tar file, under
>>>> contrib/realtime/mysql you’ll find the .sql files required for all the
>>>> realtime drivers. I never thought to use the ones with OM.
>>>>
>>>>
>>>>
>>>> Jeff Clay
>>>>
>>>> Network Administrator
>>>>
>>>> Infotech Enterprises America
>>>>
>>>> 870-215-5506
>>>>
>>>> Ext. 1506
>>>>
>>>>
>>>>
>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>
>>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>>> *To:* user@openmeetings.apache.org
>>>> *Cc:* Jeff Clay
>>>> *Subject:* Re: SIP connectivity
>>>>
>>>>
>>>>
>>>> Well,
>>>>
>>>> I might have found one difference though:
>>>>
>>>>
>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>> dictates how the table should look like. I obviously used the one in the
>>>> openmeetings mysql database, but this one seems to miss the table
>>>> "useragent". I discovered this because it showed up in the logfiles.
>>>>
>>>> BC
>>>>
>>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>>
>>>>
>>>> From an asterisk configuration standpoint there are very few
>>>> differences between 1.8.x and 11.x. If memory serves, the only major
>>>> changes that I ran into (in my production environment) was changes to SIP
>>>> NAT values and the behavior of app_page() now uses confbridge instead of
>>>> meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major
>>>> overhauling. There were of course many other changes and bug fixes, you can
>>>> skim through the change log for full details, but I think that was the jist
>>>> of it.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Jeff Clay
>>>>
>>>> Network Administrator
>>>>
>>>> Infotech Enterprises America
>>>>
>>>> 870-215-5506
>>>>
>>>> Ext. 1506
>>>>
>>>>
>>>>
>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>
>>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>>> *To:* Maxim Solodovnik
>>>> *Cc:* user
>>>> *Subject:* Re: SIP connectivity
>>>>
>>>>
>>>>
>>>> I see - I'm willing to try the 11 version in the next fiew days if
>>>> desired.
>>>>
>>>> BC
>>>>
>>>>
>>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>>
>>>> I test the integration using
>>>>
>>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>>
>>>>
>>>>
>>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <
>>>> bart.coninckx@telenet.be> wrote:
>>>>
>>>> That is amazing - I initially tried to do the same thing by using the
>>>> new chan_motif driver in Asterisk 11 which connects to a XMPP server.
>>>>
>>>> Are you guys using Asterisk 11? This version is the newest LTS version
>>>> and has the best video capabilities.
>>>>
>>>> Cheers,
>>>>
>>>> BC
>>>>
>>>>
>>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>>
>>>> red5sip will create special OM user in the room: "SIP Transport"
>>>>
>>>> after that you can call to the OM room using SIP hard or soft phone.
>>>>
>>>>
>>>>
>>>> We are currently testing it and trying to add video capabilities ...
>>>>
>>>>
>>>>
>>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <
>>>> bart.coninckx@telenet.be> wrote:
>>>>
>>>> Hi Jeff,
>>>>
>>>> In fact, I saw both pages, but none explain what they set up to do,
>>>> just a bunch of command line instructions are given.
>>>> Your "OM will create a meetme meeting as configured in the realtime
>>>> meetme database" actually says it all in one go :-)
>>>>
>>>> cheers,
>>>>
>>>> BC
>>>>
>>>>
>>>>
>>>>
>>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>> OM will create a meetme meeting as configured in the realtime meetme
>>>> database. Have you read this page
>>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
>>>> http://openmeetings.apache.org/red5sip-integration.html but I assume
>>>> this is the one you're already referring to.
>>>>
>>>> Jeff Clay
>>>> Network Administrator
>>>> Infotech Enterprises America
>>>> 870-215-5506
>>>> Ext. 1506
>>>>
>>>> -----Original Message-----
>>>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>>> Sent: Monday, January 28, 2013 3:36 PM
>>>> To: user@openmeetings.apache.org
>>>> Subject: SIP connectivity
>>>>
>>>> Hi,
>>>>
>>>> I noticed some documentation on how to connect OM with a SIP proxy or
>>>> server, more particularly with the MeetMe application in Asterisk.
>>>>
>>>> The exact goal or purpose is not mentionned however. Will OM callout to
>>>> a MeetMe conference? Or is it the other way round?
>>>>
>>>>
>>>> Cheers,
>>>>
>>>> Bc
>>>>
>>>> ________________________________
>>>>
>>>> DISCLAIMER:
>>>>
>>>> This email may contain confidential information and is intended only
>>>> for the use of the specific individual(s) to which it is addressed. If you
>>>> are not the intended recipient of this email, you are hereby notified that
>>>> any unauthorized use, dissemination or copying of this email or the
>>>> information contained in it or attached to it is strictly prohibited. If
>>>> you received this message in error, please immediately notify the sender at
>>>> Infotech and delete the original message.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>
>>>
>>> --
>>> WBR
>>> Maxim aka solomax
>>>
>>>
>>>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
--
WBR
Maxim aka solomax
Re: SIP connectivity
Posted by Bart Coninckx <ba...@telenet.be>.
OK, I suppose these instructions supersede the ones on:
https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html
?
what is the status of OM 2.1? Is it production stable?
cheers,
BC
On 01/31/13 13:05, Maxim Solodovnik wrote:
> Hello Bart,
>
> I just take a look at your URL ...
> OM does not create/use sipfriends DB table (at least from version 2.1)
> only meetme table is used
>
> so I'm afraid there is nothing to change here
>
> Here is the most recent instruction:
> http://openmeetings.apache.org/red5sip-integration_2.1.html
>
> Will ask our SIP guru to review it one more time :)
>
>
>
> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik
> <solomax666@gmail.com <ma...@gmail.com>> wrote:
>
> OK will add it and notify you
>
> On Jan 31, 2013 5:05 PM, "Bart Coninckx" <bart.coninckx@telenet.be
> <ma...@telenet.be>> wrote:
>
> It is for Asterisk 11 - don't know for other versions. You
> probably have no issues because of the 1.8 version. To be sure
> the .sql files in the Asterisk source should be compared
> across versions.
>
> this one is missing:
>
> `useragent` varchar(20) DEFAULT NULL,
>
> complete list (I think) is on:
>
> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>
>
> If I bump into others, I'll report ASAP,
>
>
> BC
>
>
>
> On 01/31/13 06:21, Maxim Solodovnik wrote:
>> Is the OM meetme table incomplete?
>> My asterisk reports no issues :(
>>
>> could you provide me with missing fields and I'll add it.
>> My purpose was to create table with required fields only.
>>
>>
>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx
>> <bart.coninckx@telenet.be <ma...@telenet.be>>
>> wrote:
>>
>> Openmeetings installed them for me, that's why I ended up
>> with those. Using the Asterisk ones makes more sense to
>> me. Maybe it's a good idea to have 'em removed from the
>> install procedure.
>>
>> BC
>>
>>
>> On 01/30/13 22:30, Jeff Clay wrote:
>>>
>>> Bart,
>>>
>>> If you look in the source directory of your asterisk tar
>>> file, under contrib/realtime/mysql you’ll find the .sql
>>> files required for all the realtime drivers. I never
>>> thought to use the ones with OM.
>>>
>>> Jeff Clay
>>>
>>> Network Administrator
>>>
>>> Infotech Enterprises America
>>>
>>> 870-215-5506
>>>
>>> Ext. 1506
>>>
>>> *From:*Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>> *To:* user@openmeetings.apache.org
>>> <ma...@openmeetings.apache.org>
>>> *Cc:* Jeff Clay
>>> *Subject:* Re: SIP connectivity
>>>
>>> Well,
>>>
>>> I might have found one difference though:
>>>
>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>> dictates how the table should look like. I obviously
>>> used the one in the openmeetings mysql database, but
>>> this one seems to miss the table "useragent". I
>>> discovered this because it showed up in the logfiles.
>>>
>>> BC
>>>
>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>
>>> Bart,
>>>
>>> From an asterisk configuration standpoint there are
>>> very few differences between 1.8.x and 11.x. If
>>> memory serves, the only major changes that I ran
>>> into (in my production environment) was changes to
>>> SIP NAT values and the behavior of app_page() now
>>> uses confbridge instead of meetme to mix the audio.
>>> Also, TCP, TLS and app_confbridge got a major
>>> overhauling. There were of course many other changes
>>> and bug fixes, you can skim through the change log
>>> for full details, but I think that was the jist of it.
>>>
>>> Jeff Clay
>>>
>>> Network Administrator
>>>
>>> Infotech Enterprises America
>>>
>>> 870-215-5506
>>>
>>> Ext. 1506
>>>
>>> *From:*Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>> *To:* Maxim Solodovnik
>>> *Cc:* user
>>> *Subject:* Re: SIP connectivity
>>>
>>> I see - I'm willing to try the 11 version in the
>>> next fiew days if desired.
>>>
>>> BC
>>>
>>>
>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>
>>> I test the integration using
>>>
>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>
>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx
>>> <bart.coninckx@telenet.be
>>> <ma...@telenet.be>> wrote:
>>>
>>> That is amazing - I initially tried to do the
>>> same thing by using the new chan_motif driver in
>>> Asterisk 11 which connects to a XMPP server.
>>>
>>> Are you guys using Asterisk 11? This version is
>>> the newest LTS version and has the best video
>>> capabilities.
>>>
>>> Cheers,
>>>
>>> BC
>>>
>>>
>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>
>>> red5sip will create special OM user in the
>>> room: "SIP Transport"
>>>
>>> after that you can call to the OM room using
>>> SIP hard or soft phone.
>>>
>>> We are currently testing it and trying to
>>> add video capabilities ...
>>>
>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart
>>> Coninckx <bart.coninckx@telenet.be
>>> <ma...@telenet.be>> wrote:
>>>
>>> Hi Jeff,
>>>
>>> In fact, I saw both pages, but none explain
>>> what they set up to do, just a bunch of
>>> command line instructions are given.
>>> Your "OM will create a meetme meeting as
>>> configured in the realtime meetme database"
>>> actually says it all in one go :-)
>>>
>>> cheers,
>>>
>>> BC
>>>
>>>
>>>
>>>
>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>
>>> Bart,
>>>
>>> OM will create a meetme meeting as
>>> configured in the realtime meetme database.
>>> Have you read this page
>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html
>>> ? You might also check out
>>> http://openmeetings.apache.org/red5sip-integration.html
>>> but I assume this is the one you're already
>>> referring to.
>>>
>>> Jeff Clay
>>> Network Administrator
>>> Infotech Enterprises America
>>> 870-215-5506
>>> Ext. 1506
>>>
>>> -----Original Message-----
>>> From: Bart Coninckx
>>> [mailto:bart.coninckx@telenet.be
>>> <ma...@telenet.be>]
>>> Sent: Monday, January 28, 2013 3:36 PM
>>> To: user@openmeetings.apache.org
>>> <ma...@openmeetings.apache.org>
>>> Subject: SIP connectivity
>>>
>>> Hi,
>>>
>>> I noticed some documentation on how to
>>> connect OM with a SIP proxy or server, more
>>> particularly with the MeetMe application in
>>> Asterisk.
>>>
>>> The exact goal or purpose is not mentionned
>>> however. Will OM callout to a MeetMe
>>> conference? Or is it the other way round?
>>>
>>>
>>> Cheers,
>>>
>>> Bc
>>>
>>> ________________________________
>>>
>>> DISCLAIMER:
>>>
>>> This email may contain confidential
>>> information and is intended only for the use
>>> of the specific individual(s) to which it is
>>> addressed. If you are not the intended
>>> recipient of this email, you are hereby
>>> notified that any unauthorized use,
>>> dissemination or copying of this email or
>>> the information contained in it or attached
>>> to it is strictly prohibited. If you
>>> received this message in error, please
>>> immediately notify the sender at Infotech
>>> and delete the original message.
>>>
>>>
>>>
>>> --
>>> WBR
>>> Maxim aka solomax
>>>
>>>
>>>
>>> --
>>> WBR
>>> Maxim aka solomax
>>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>
>
>
>
> --
> WBR
> Maxim aka solomax
Re: SIP connectivity
Posted by Maxim Solodovnik <so...@gmail.com>.
We already set two columns: name and defaultuser:
https://svn.apache.org/repos/asf/openmeetings/trunk/singlewebapp/src/org/apache/openmeetings/persistence/beans/sip/asterisk/AsteriskSipUser.java
On Thu, Feb 7, 2013 at 10:07 PM, Bakko <as...@gmail.com> wrote:
> Other think,
>
> On the openmeetings table sipuser, field username is deprecated. Change to
> defaultuser.
>
> Regards
>
> El 07/02/2013 09:59, Maxim Solodovnik escribió:
>
> Just have tested it and it it works!
> Thanks a lot!
> I'll update the documentation
>
>
> On Fri, Feb 1, 2013 at 7:35 PM, Bakko <as...@gmail.com> wrote:
>
>> Hello,
>>
>> on Asterisk 1.8, extconfig.conf this line:
>>
>> *sipusers => odbc,asterisk,sipusers
>>
>> *is deprecated.
>>
>> Now only use:
>>
>> *sippeers => odbc,asterisk,sipusers
>>
>> *Regards*
>>
>> *
>> El 01/02/2013 00:00, Maxim Solodovnik escribió:
>>
>> I have updated the instruction (minor update)
>>
>>
>> On Thu, Jan 31, 2013 at 7:05 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>>
>>> Hello Bart,
>>>
>>> I just take a look at your URL ...
>>> OM does not create/use sipfriends DB table (at least from version 2.1)
>>> only meetme table is used
>>>
>>> so I'm afraid there is nothing to change here
>>>
>>> Here is the most recent instruction:
>>> http://openmeetings.apache.org/red5sip-integration_2.1.html
>>>
>>> Will ask our SIP guru to review it one more time :)
>>>
>>>
>>>
>>> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>>>
>>>> OK will add it and notify you
>>>> On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>
>>>> wrote:
>>>>
>>>>> It is for Asterisk 11 - don't know for other versions. You probably
>>>>> have no issues because of the 1.8 version. To be sure the .sql files in the
>>>>> Asterisk source should be compared across versions.
>>>>>
>>>>> this one is missing:
>>>>>
>>>>> `useragent` varchar(20) DEFAULT NULL,
>>>>>
>>>>> complete list (I think) is on:
>>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>>>
>>>>>
>>>>> If I bump into others, I'll report ASAP,
>>>>>
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>>
>>>>> On 01/31/13 06:21, Maxim Solodovnik wrote:
>>>>>
>>>>> Is the OM meetme table incomplete?
>>>>> My asterisk reports no issues :(
>>>>>
>>>>> could you provide me with missing fields and I'll add it.
>>>>> My purpose was to create table with required fields only.
>>>>>
>>>>>
>>>>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <
>>>>> bart.coninckx@telenet.be> wrote:
>>>>>
>>>>>> Openmeetings installed them for me, that's why I ended up with
>>>>>> those. Using the Asterisk ones makes more sense to me. Maybe it's a good
>>>>>> idea to have 'em removed from the install procedure.
>>>>>>
>>>>>> BC
>>>>>>
>>>>>>
>>>>>> On 01/30/13 22:30, Jeff Clay wrote:
>>>>>>
>>>>>> Bart,
>>>>>>
>>>>>>
>>>>>>
>>>>>> If you look in the source directory of your asterisk tar file, under
>>>>>> contrib/realtime/mysql you’ll find the .sql files required for all the
>>>>>> realtime drivers. I never thought to use the ones with OM.
>>>>>>
>>>>>>
>>>>>>
>>>>>> Jeff Clay
>>>>>>
>>>>>> Network Administrator
>>>>>>
>>>>>> Infotech Enterprises America
>>>>>>
>>>>>> 870-215-5506
>>>>>>
>>>>>> Ext. 1506
>>>>>>
>>>>>>
>>>>>>
>>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>>>
>>>>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>>>>> *To:* user@openmeetings.apache.org
>>>>>> *Cc:* Jeff Clay
>>>>>> *Subject:* Re: SIP connectivity
>>>>>>
>>>>>>
>>>>>>
>>>>>> Well,
>>>>>>
>>>>>> I might have found one difference though:
>>>>>>
>>>>>>
>>>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>>>> dictates how the table should look like. I obviously used the one in the
>>>>>> openmeetings mysql database, but this one seems to miss the table
>>>>>> "useragent". I discovered this because it showed up in the logfiles.
>>>>>>
>>>>>> BC
>>>>>>
>>>>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>>>>
>>>>>> Bart,
>>>>>>
>>>>>>
>>>>>>
>>>>>> From an asterisk configuration standpoint there are very few
>>>>>> differences between 1.8.x and 11.x. If memory serves, the only major
>>>>>> changes that I ran into (in my production environment) was changes to SIP
>>>>>> NAT values and the behavior of app_page() now uses confbridge instead of
>>>>>> meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major
>>>>>> overhauling. There were of course many other changes and bug fixes, you can
>>>>>> skim through the change log for full details, but I think that was the jist
>>>>>> of it.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> Jeff Clay
>>>>>>
>>>>>> Network Administrator
>>>>>>
>>>>>> Infotech Enterprises America
>>>>>>
>>>>>> 870-215-5506
>>>>>>
>>>>>> Ext. 1506
>>>>>>
>>>>>>
>>>>>>
>>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>>>
>>>>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>>>>> *To:* Maxim Solodovnik
>>>>>> *Cc:* user
>>>>>> *Subject:* Re: SIP connectivity
>>>>>>
>>>>>>
>>>>>>
>>>>>> I see - I'm willing to try the 11 version in the next fiew days if
>>>>>> desired.
>>>>>>
>>>>>> BC
>>>>>>
>>>>>>
>>>>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>>>>
>>>>>> I test the integration using
>>>>>>
>>>>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <
>>>>>> bart.coninckx@telenet.be> wrote:
>>>>>>
>>>>>> That is amazing - I initially tried to do the same thing by using the
>>>>>> new chan_motif driver in Asterisk 11 which connects to a XMPP server.
>>>>>>
>>>>>> Are you guys using Asterisk 11? This version is the newest LTS
>>>>>> version and has the best video capabilities.
>>>>>>
>>>>>> Cheers,
>>>>>>
>>>>>> BC
>>>>>>
>>>>>>
>>>>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>>>>
>>>>>> red5sip will create special OM user in the room: "SIP Transport"
>>>>>>
>>>>>> after that you can call to the OM room using SIP hard or soft phone.
>>>>>>
>>>>>>
>>>>>>
>>>>>> We are currently testing it and trying to add video capabilities ...
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <
>>>>>> bart.coninckx@telenet.be> wrote:
>>>>>>
>>>>>> Hi Jeff,
>>>>>>
>>>>>> In fact, I saw both pages, but none explain what they set up to do,
>>>>>> just a bunch of command line instructions are given.
>>>>>> Your "OM will create a meetme meeting as configured in the realtime
>>>>>> meetme database" actually says it all in one go :-)
>>>>>>
>>>>>> cheers,
>>>>>>
>>>>>> BC
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>>>>
>>>>>> Bart,
>>>>>>
>>>>>> OM will create a meetme meeting as configured in the realtime meetme
>>>>>> database. Have you read this page
>>>>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
>>>>>> http://openmeetings.apache.org/red5sip-integration.html but I assume
>>>>>> this is the one you're already referring to.
>>>>>>
>>>>>> Jeff Clay
>>>>>> Network Administrator
>>>>>> Infotech Enterprises America
>>>>>> 870-215-5506
>>>>>> Ext. 1506
>>>>>>
>>>>>> -----Original Message-----
>>>>>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>>>>> Sent: Monday, January 28, 2013 3:36 PM
>>>>>> To: user@openmeetings.apache.org
>>>>>> Subject: SIP connectivity
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I noticed some documentation on how to connect OM with a SIP proxy or
>>>>>> server, more particularly with the MeetMe application in Asterisk.
>>>>>>
>>>>>> The exact goal or purpose is not mentionned however. Will OM callout
>>>>>> to a MeetMe conference? Or is it the other way round?
>>>>>>
>>>>>>
>>>>>> Cheers,
>>>>>>
>>>>>> Bc
>>>>>>
>>>>>> ________________________________
>>>>>>
>>>>>> DISCLAIMER:
>>>>>>
>>>>>> This email may contain confidential information and is intended only
>>>>>> for the use of the specific individual(s) to which it is addressed. If you
>>>>>> are not the intended recipient of this email, you are hereby notified that
>>>>>> any unauthorized use, dissemination or copying of this email or the
>>>>>> information contained in it or attached to it is strictly prohibited. If
>>>>>> you received this message in error, please immediately notify the sender at
>>>>>> Infotech and delete the original message.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> 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: SIP connectivity
Posted by Bakko <as...@gmail.com>.
Other think,
On the openmeetings table sipuser, field username is deprecated. Change
to defaultuser.
Regards
El 07/02/2013 09:59, Maxim Solodovnik escribió:
> Just have tested it and it it works!
> Thanks a lot!
> I'll update the documentation
>
>
> On Fri, Feb 1, 2013 at 7:35 PM, Bakko <asannucci@gmail.com
> <ma...@gmail.com>> wrote:
>
> Hello,
>
> on Asterisk 1.8, extconfig.conf this line:
>
> /sipusers => odbc,asterisk,sipusers
>
> /is deprecated.
>
> Now only use:
>
> /sippeers => odbc,asterisk,sipusers
>
> /Regards/
>
> /
> El 01/02/2013 00:00, Maxim Solodovnik escribió:
>> I have updated the instruction (minor update)
>>
>>
>> On Thu, Jan 31, 2013 at 7:05 PM, Maxim Solodovnik
>> <solomax666@gmail.com <ma...@gmail.com>> wrote:
>>
>> Hello Bart,
>>
>> I just take a look at your URL ...
>> OM does not create/use sipfriends DB table (at least from
>> version 2.1)
>> only meetme table is used
>>
>> so I'm afraid there is nothing to change here
>>
>> Here is the most recent instruction:
>> http://openmeetings.apache.org/red5sip-integration_2.1.html
>>
>> Will ask our SIP guru to review it one more time :)
>>
>>
>>
>> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik
>> <solomax666@gmail.com <ma...@gmail.com>> wrote:
>>
>> OK will add it and notify you
>>
>> On Jan 31, 2013 5:05 PM, "Bart Coninckx"
>> <bart.coninckx@telenet.be
>> <ma...@telenet.be>> wrote:
>>
>> It is for Asterisk 11 - don't know for other
>> versions. You probably have no issues because of the
>> 1.8 version. To be sure the .sql files in the
>> Asterisk source should be compared across versions.
>>
>> this one is missing:
>>
>> `useragent` varchar(20) DEFAULT NULL,
>>
>> complete list (I think) is on:
>>
>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>
>>
>> If I bump into others, I'll report ASAP,
>>
>>
>> BC
>>
>>
>>
>> On 01/31/13 06:21, Maxim Solodovnik wrote:
>>> Is the OM meetme table incomplete?
>>> My asterisk reports no issues :(
>>>
>>> could you provide me with missing fields and I'll
>>> add it.
>>> My purpose was to create table with required fields
>>> only.
>>>
>>>
>>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx
>>> <bart.coninckx@telenet.be
>>> <ma...@telenet.be>> wrote:
>>>
>>> Openmeetings installed them for me, that's why I
>>> ended up with those. Using the Asterisk ones
>>> makes more sense to me. Maybe it's a good idea
>>> to have 'em removed from the install procedure.
>>>
>>> BC
>>>
>>>
>>> On 01/30/13 22:30, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>> If you look in the source directory of your
>>>> asterisk tar file, under contrib/realtime/mysql
>>>> you’ll find the .sql files required for all the
>>>> realtime drivers. I never thought to use the
>>>> ones with OM.
>>>>
>>>> Jeff Clay
>>>>
>>>> Network Administrator
>>>>
>>>> Infotech Enterprises America
>>>>
>>>> 870-215-5506
>>>>
>>>> Ext. 1506
>>>>
>>>> *From:*Bart Coninckx
>>>> [mailto:bart.coninckx@telenet.be]
>>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>>> *To:* user@openmeetings.apache.org
>>>> <ma...@openmeetings.apache.org>
>>>> *Cc:* Jeff Clay
>>>> *Subject:* Re: SIP connectivity
>>>>
>>>> Well,
>>>>
>>>> I might have found one difference though:
>>>>
>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>> dictates how the table should look like. I
>>>> obviously used the one in the openmeetings
>>>> mysql database, but this one seems to miss the
>>>> table "useragent". I discovered this because it
>>>> showed up in the logfiles.
>>>>
>>>> BC
>>>>
>>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>> From an asterisk configuration standpoint
>>>> there are very few differences between
>>>> 1.8.x and 11.x. If memory serves, the only
>>>> major changes that I ran into (in my
>>>> production environment) was changes to SIP
>>>> NAT values and the behavior of app_page()
>>>> now uses confbridge instead of meetme to
>>>> mix the audio. Also, TCP, TLS and
>>>> app_confbridge got a major overhauling.
>>>> There were of course many other changes and
>>>> bug fixes, you can skim through the change
>>>> log for full details, but I think that was
>>>> the jist of it.
>>>>
>>>> Jeff Clay
>>>>
>>>> Network Administrator
>>>>
>>>> Infotech Enterprises America
>>>>
>>>> 870-215-5506
>>>>
>>>> Ext. 1506
>>>>
>>>> *From:*Bart Coninckx
>>>> [mailto:bart.coninckx@telenet.be]
>>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>>> *To:* Maxim Solodovnik
>>>> *Cc:* user
>>>> *Subject:* Re: SIP connectivity
>>>>
>>>> I see - I'm willing to try the 11 version
>>>> in the next fiew days if desired.
>>>>
>>>> BC
>>>>
>>>>
>>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>>
>>>> I test the integration using
>>>>
>>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>>
>>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart
>>>> Coninckx <bart.coninckx@telenet.be
>>>> <ma...@telenet.be>> wrote:
>>>>
>>>> That is amazing - I initially tried to
>>>> do the same thing by using the new
>>>> chan_motif driver in Asterisk 11 which
>>>> connects to a XMPP server.
>>>>
>>>> Are you guys using Asterisk 11? This
>>>> version is the newest LTS version and
>>>> has the best video capabilities.
>>>>
>>>> Cheers,
>>>>
>>>> BC
>>>>
>>>>
>>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>>
>>>> red5sip will create special OM user
>>>> in the room: "SIP Transport"
>>>>
>>>> after that you can call to the OM
>>>> room using SIP hard or soft phone.
>>>>
>>>> We are currently testing it and
>>>> trying to add video capabilities ...
>>>>
>>>> On Tue, Jan 29, 2013 at 4:44 AM,
>>>> Bart Coninckx
>>>> <bart.coninckx@telenet.be
>>>> <ma...@telenet.be>>
>>>> wrote:
>>>>
>>>> Hi Jeff,
>>>>
>>>> In fact, I saw both pages, but none
>>>> explain what they set up to do,
>>>> just a bunch of command line
>>>> instructions are given.
>>>> Your "OM will create a meetme
>>>> meeting as configured in the
>>>> realtime meetme database" actually
>>>> says it all in one go :-)
>>>>
>>>> cheers,
>>>>
>>>> BC
>>>>
>>>>
>>>>
>>>>
>>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>> OM will create a meetme meeting as
>>>> configured in the realtime meetme
>>>> database. Have you read this page
>>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html
>>>> ? You might also check out
>>>> http://openmeetings.apache.org/red5sip-integration.html
>>>> but I assume this is the one you're
>>>> already referring to.
>>>>
>>>> Jeff Clay
>>>> Network Administrator
>>>> Infotech Enterprises America
>>>> 870-215-5506
>>>> Ext. 1506
>>>>
>>>> -----Original Message-----
>>>> From: Bart Coninckx
>>>> [mailto:bart.coninckx@telenet.be
>>>> <ma...@telenet.be>]
>>>> Sent: Monday, January 28, 2013 3:36 PM
>>>> To: user@openmeetings.apache.org
>>>> <ma...@openmeetings.apache.org>
>>>> Subject: SIP connectivity
>>>>
>>>> Hi,
>>>>
>>>> I noticed some documentation on how
>>>> to connect OM with a SIP proxy or
>>>> server, more particularly with the
>>>> MeetMe application in Asterisk.
>>>>
>>>> The exact goal or purpose is not
>>>> mentionned however. Will OM callout
>>>> to a MeetMe conference? Or is it
>>>> the other way round?
>>>>
>>>>
>>>> Cheers,
>>>>
>>>> Bc
>>>>
>>>> ________________________________
>>>>
>>>> DISCLAIMER:
>>>>
>>>> This email may contain confidential
>>>> information and is intended only
>>>> for the use of the specific
>>>> individual(s) to which it is
>>>> addressed. If you are not the
>>>> intended recipient of this email,
>>>> you are hereby notified that any
>>>> unauthorized use, dissemination or
>>>> copying of this email or the
>>>> information contained in it or
>>>> attached to it is strictly
>>>> prohibited. If you received this
>>>> message in error, please
>>>> immediately notify the sender at
>>>> Infotech and delete the original
>>>> message.
>>>>
>>>>
>>>>
>>>> --
>>>> 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: SIP connectivity
Posted by Maxim Solodovnik <so...@gmail.com>.
Just have tested it and it it works!
Thanks a lot!
I'll update the documentation
On Fri, Feb 1, 2013 at 7:35 PM, Bakko <as...@gmail.com> wrote:
> Hello,
>
> on Asterisk 1.8, extconfig.conf this line:
>
> *sipusers => odbc,asterisk,sipusers
>
> *is deprecated.
>
> Now only use:
>
> *sippeers => odbc,asterisk,sipusers
>
> *Regards*
>
> *
> El 01/02/2013 00:00, Maxim Solodovnik escribió:
>
> I have updated the instruction (minor update)
>
>
> On Thu, Jan 31, 2013 at 7:05 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>
>> Hello Bart,
>>
>> I just take a look at your URL ...
>> OM does not create/use sipfriends DB table (at least from version 2.1)
>> only meetme table is used
>>
>> so I'm afraid there is nothing to change here
>>
>> Here is the most recent instruction:
>> http://openmeetings.apache.org/red5sip-integration_2.1.html
>>
>> Will ask our SIP guru to review it one more time :)
>>
>>
>>
>> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>>
>>> OK will add it and notify you
>>> On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>
>>> wrote:
>>>
>>>> It is for Asterisk 11 - don't know for other versions. You probably
>>>> have no issues because of the 1.8 version. To be sure the .sql files in the
>>>> Asterisk source should be compared across versions.
>>>>
>>>> this one is missing:
>>>>
>>>> `useragent` varchar(20) DEFAULT NULL,
>>>>
>>>> complete list (I think) is on:
>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>>
>>>>
>>>> If I bump into others, I'll report ASAP,
>>>>
>>>>
>>>> BC
>>>>
>>>>
>>>>
>>>> On 01/31/13 06:21, Maxim Solodovnik wrote:
>>>>
>>>> Is the OM meetme table incomplete?
>>>> My asterisk reports no issues :(
>>>>
>>>> could you provide me with missing fields and I'll add it.
>>>> My purpose was to create table with required fields only.
>>>>
>>>>
>>>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <
>>>> bart.coninckx@telenet.be> wrote:
>>>>
>>>>> Openmeetings installed them for me, that's why I ended up with
>>>>> those. Using the Asterisk ones makes more sense to me. Maybe it's a good
>>>>> idea to have 'em removed from the install procedure.
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>> On 01/30/13 22:30, Jeff Clay wrote:
>>>>>
>>>>> Bart,
>>>>>
>>>>>
>>>>>
>>>>> If you look in the source directory of your asterisk tar file, under
>>>>> contrib/realtime/mysql you’ll find the .sql files required for all the
>>>>> realtime drivers. I never thought to use the ones with OM.
>>>>>
>>>>>
>>>>>
>>>>> Jeff Clay
>>>>>
>>>>> Network Administrator
>>>>>
>>>>> Infotech Enterprises America
>>>>>
>>>>> 870-215-5506
>>>>>
>>>>> Ext. 1506
>>>>>
>>>>>
>>>>>
>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>>
>>>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>>>> *To:* user@openmeetings.apache.org
>>>>> *Cc:* Jeff Clay
>>>>> *Subject:* Re: SIP connectivity
>>>>>
>>>>>
>>>>>
>>>>> Well,
>>>>>
>>>>> I might have found one difference though:
>>>>>
>>>>>
>>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>>> dictates how the table should look like. I obviously used the one in the
>>>>> openmeetings mysql database, but this one seems to miss the table
>>>>> "useragent". I discovered this because it showed up in the logfiles.
>>>>>
>>>>> BC
>>>>>
>>>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>>>
>>>>> Bart,
>>>>>
>>>>>
>>>>>
>>>>> From an asterisk configuration standpoint there are very few
>>>>> differences between 1.8.x and 11.x. If memory serves, the only major
>>>>> changes that I ran into (in my production environment) was changes to SIP
>>>>> NAT values and the behavior of app_page() now uses confbridge instead of
>>>>> meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major
>>>>> overhauling. There were of course many other changes and bug fixes, you can
>>>>> skim through the change log for full details, but I think that was the jist
>>>>> of it.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Jeff Clay
>>>>>
>>>>> Network Administrator
>>>>>
>>>>> Infotech Enterprises America
>>>>>
>>>>> 870-215-5506
>>>>>
>>>>> Ext. 1506
>>>>>
>>>>>
>>>>>
>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>>
>>>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>>>> *To:* Maxim Solodovnik
>>>>> *Cc:* user
>>>>> *Subject:* Re: SIP connectivity
>>>>>
>>>>>
>>>>>
>>>>> I see - I'm willing to try the 11 version in the next fiew days if
>>>>> desired.
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>>>
>>>>> I test the integration using
>>>>>
>>>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <
>>>>> bart.coninckx@telenet.be> wrote:
>>>>>
>>>>> That is amazing - I initially tried to do the same thing by using the
>>>>> new chan_motif driver in Asterisk 11 which connects to a XMPP server.
>>>>>
>>>>> Are you guys using Asterisk 11? This version is the newest LTS version
>>>>> and has the best video capabilities.
>>>>>
>>>>> Cheers,
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>>>
>>>>> red5sip will create special OM user in the room: "SIP Transport"
>>>>>
>>>>> after that you can call to the OM room using SIP hard or soft phone.
>>>>>
>>>>>
>>>>>
>>>>> We are currently testing it and trying to add video capabilities ...
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <
>>>>> bart.coninckx@telenet.be> wrote:
>>>>>
>>>>> Hi Jeff,
>>>>>
>>>>> In fact, I saw both pages, but none explain what they set up to do,
>>>>> just a bunch of command line instructions are given.
>>>>> Your "OM will create a meetme meeting as configured in the realtime
>>>>> meetme database" actually says it all in one go :-)
>>>>>
>>>>> cheers,
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>>>
>>>>> Bart,
>>>>>
>>>>> OM will create a meetme meeting as configured in the realtime meetme
>>>>> database. Have you read this page
>>>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
>>>>> http://openmeetings.apache.org/red5sip-integration.html but I assume
>>>>> this is the one you're already referring to.
>>>>>
>>>>> Jeff Clay
>>>>> Network Administrator
>>>>> Infotech Enterprises America
>>>>> 870-215-5506
>>>>> Ext. 1506
>>>>>
>>>>> -----Original Message-----
>>>>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>>>> Sent: Monday, January 28, 2013 3:36 PM
>>>>> To: user@openmeetings.apache.org
>>>>> Subject: SIP connectivity
>>>>>
>>>>> Hi,
>>>>>
>>>>> I noticed some documentation on how to connect OM with a SIP proxy or
>>>>> server, more particularly with the MeetMe application in Asterisk.
>>>>>
>>>>> The exact goal or purpose is not mentionned however. Will OM callout
>>>>> to a MeetMe conference? Or is it the other way round?
>>>>>
>>>>>
>>>>> Cheers,
>>>>>
>>>>> Bc
>>>>>
>>>>> ________________________________
>>>>>
>>>>> DISCLAIMER:
>>>>>
>>>>> This email may contain confidential information and is intended only
>>>>> for the use of the specific individual(s) to which it is addressed. If you
>>>>> are not the intended recipient of this email, you are hereby notified that
>>>>> any unauthorized use, dissemination or copying of this email or the
>>>>> information contained in it or attached to it is strictly prohibited. If
>>>>> you received this message in error, please immediately notify the sender at
>>>>> Infotech and delete the original message.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> 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: SIP connectivity
Posted by Maxim Solodovnik <so...@gmail.com>.
Thanks!
I'll try to test it and modify the doc.
(I leave it since I'm not "asterisk guru" and try not to experiment :)) )
On Fri, Feb 1, 2013 at 7:35 PM, Bakko <as...@gmail.com> wrote:
> Hello,
>
> on Asterisk 1.8, extconfig.conf this line:
>
> *sipusers => odbc,asterisk,sipusers
>
> *is deprecated.
>
> Now only use:
>
> *sippeers => odbc,asterisk,sipusers
>
> *Regards*
>
> *
> El 01/02/2013 00:00, Maxim Solodovnik escribió:
>
> I have updated the instruction (minor update)
>
>
> On Thu, Jan 31, 2013 at 7:05 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>
>> Hello Bart,
>>
>> I just take a look at your URL ...
>> OM does not create/use sipfriends DB table (at least from version 2.1)
>> only meetme table is used
>>
>> so I'm afraid there is nothing to change here
>>
>> Here is the most recent instruction:
>> http://openmeetings.apache.org/red5sip-integration_2.1.html
>>
>> Will ask our SIP guru to review it one more time :)
>>
>>
>>
>> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>>
>>> OK will add it and notify you
>>> On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>
>>> wrote:
>>>
>>>> It is for Asterisk 11 - don't know for other versions. You probably
>>>> have no issues because of the 1.8 version. To be sure the .sql files in the
>>>> Asterisk source should be compared across versions.
>>>>
>>>> this one is missing:
>>>>
>>>> `useragent` varchar(20) DEFAULT NULL,
>>>>
>>>> complete list (I think) is on:
>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>>
>>>>
>>>> If I bump into others, I'll report ASAP,
>>>>
>>>>
>>>> BC
>>>>
>>>>
>>>>
>>>> On 01/31/13 06:21, Maxim Solodovnik wrote:
>>>>
>>>> Is the OM meetme table incomplete?
>>>> My asterisk reports no issues :(
>>>>
>>>> could you provide me with missing fields and I'll add it.
>>>> My purpose was to create table with required fields only.
>>>>
>>>>
>>>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <
>>>> bart.coninckx@telenet.be> wrote:
>>>>
>>>>> Openmeetings installed them for me, that's why I ended up with
>>>>> those. Using the Asterisk ones makes more sense to me. Maybe it's a good
>>>>> idea to have 'em removed from the install procedure.
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>> On 01/30/13 22:30, Jeff Clay wrote:
>>>>>
>>>>> Bart,
>>>>>
>>>>>
>>>>>
>>>>> If you look in the source directory of your asterisk tar file, under
>>>>> contrib/realtime/mysql you’ll find the .sql files required for all the
>>>>> realtime drivers. I never thought to use the ones with OM.
>>>>>
>>>>>
>>>>>
>>>>> Jeff Clay
>>>>>
>>>>> Network Administrator
>>>>>
>>>>> Infotech Enterprises America
>>>>>
>>>>> 870-215-5506
>>>>>
>>>>> Ext. 1506
>>>>>
>>>>>
>>>>>
>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>>
>>>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>>>> *To:* user@openmeetings.apache.org
>>>>> *Cc:* Jeff Clay
>>>>> *Subject:* Re: SIP connectivity
>>>>>
>>>>>
>>>>>
>>>>> Well,
>>>>>
>>>>> I might have found one difference though:
>>>>>
>>>>>
>>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>>> dictates how the table should look like. I obviously used the one in the
>>>>> openmeetings mysql database, but this one seems to miss the table
>>>>> "useragent". I discovered this because it showed up in the logfiles.
>>>>>
>>>>> BC
>>>>>
>>>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>>>
>>>>> Bart,
>>>>>
>>>>>
>>>>>
>>>>> From an asterisk configuration standpoint there are very few
>>>>> differences between 1.8.x and 11.x. If memory serves, the only major
>>>>> changes that I ran into (in my production environment) was changes to SIP
>>>>> NAT values and the behavior of app_page() now uses confbridge instead of
>>>>> meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major
>>>>> overhauling. There were of course many other changes and bug fixes, you can
>>>>> skim through the change log for full details, but I think that was the jist
>>>>> of it.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Jeff Clay
>>>>>
>>>>> Network Administrator
>>>>>
>>>>> Infotech Enterprises America
>>>>>
>>>>> 870-215-5506
>>>>>
>>>>> Ext. 1506
>>>>>
>>>>>
>>>>>
>>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>>
>>>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>>>> *To:* Maxim Solodovnik
>>>>> *Cc:* user
>>>>> *Subject:* Re: SIP connectivity
>>>>>
>>>>>
>>>>>
>>>>> I see - I'm willing to try the 11 version in the next fiew days if
>>>>> desired.
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>>>
>>>>> I test the integration using
>>>>>
>>>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <
>>>>> bart.coninckx@telenet.be> wrote:
>>>>>
>>>>> That is amazing - I initially tried to do the same thing by using the
>>>>> new chan_motif driver in Asterisk 11 which connects to a XMPP server.
>>>>>
>>>>> Are you guys using Asterisk 11? This version is the newest LTS version
>>>>> and has the best video capabilities.
>>>>>
>>>>> Cheers,
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>>>
>>>>> red5sip will create special OM user in the room: "SIP Transport"
>>>>>
>>>>> after that you can call to the OM room using SIP hard or soft phone.
>>>>>
>>>>>
>>>>>
>>>>> We are currently testing it and trying to add video capabilities ...
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <
>>>>> bart.coninckx@telenet.be> wrote:
>>>>>
>>>>> Hi Jeff,
>>>>>
>>>>> In fact, I saw both pages, but none explain what they set up to do,
>>>>> just a bunch of command line instructions are given.
>>>>> Your "OM will create a meetme meeting as configured in the realtime
>>>>> meetme database" actually says it all in one go :-)
>>>>>
>>>>> cheers,
>>>>>
>>>>> BC
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>>>
>>>>> Bart,
>>>>>
>>>>> OM will create a meetme meeting as configured in the realtime meetme
>>>>> database. Have you read this page
>>>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
>>>>> http://openmeetings.apache.org/red5sip-integration.html but I assume
>>>>> this is the one you're already referring to.
>>>>>
>>>>> Jeff Clay
>>>>> Network Administrator
>>>>> Infotech Enterprises America
>>>>> 870-215-5506
>>>>> Ext. 1506
>>>>>
>>>>> -----Original Message-----
>>>>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>>>> Sent: Monday, January 28, 2013 3:36 PM
>>>>> To: user@openmeetings.apache.org
>>>>> Subject: SIP connectivity
>>>>>
>>>>> Hi,
>>>>>
>>>>> I noticed some documentation on how to connect OM with a SIP proxy or
>>>>> server, more particularly with the MeetMe application in Asterisk.
>>>>>
>>>>> The exact goal or purpose is not mentionned however. Will OM callout
>>>>> to a MeetMe conference? Or is it the other way round?
>>>>>
>>>>>
>>>>> Cheers,
>>>>>
>>>>> Bc
>>>>>
>>>>> ________________________________
>>>>>
>>>>> DISCLAIMER:
>>>>>
>>>>> This email may contain confidential information and is intended only
>>>>> for the use of the specific individual(s) to which it is addressed. If you
>>>>> are not the intended recipient of this email, you are hereby notified that
>>>>> any unauthorized use, dissemination or copying of this email or the
>>>>> information contained in it or attached to it is strictly prohibited. If
>>>>> you received this message in error, please immediately notify the sender at
>>>>> Infotech and delete the original message.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> 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: SIP connectivity
Posted by Bakko <as...@gmail.com>.
Hello,
on Asterisk 1.8, extconfig.conf this line:
/sipusers => odbc,asterisk,sipusers
/is deprecated.
Now only use:
/sippeers => odbc,asterisk,sipusers
/Regards/
/
El 01/02/2013 00:00, Maxim Solodovnik escribió:
> I have updated the instruction (minor update)
>
>
> On Thu, Jan 31, 2013 at 7:05 PM, Maxim Solodovnik
> <solomax666@gmail.com <ma...@gmail.com>> wrote:
>
> Hello Bart,
>
> I just take a look at your URL ...
> OM does not create/use sipfriends DB table (at least from version 2.1)
> only meetme table is used
>
> so I'm afraid there is nothing to change here
>
> Here is the most recent instruction:
> http://openmeetings.apache.org/red5sip-integration_2.1.html
>
> Will ask our SIP guru to review it one more time :)
>
>
>
> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik
> <solomax666@gmail.com <ma...@gmail.com>> wrote:
>
> OK will add it and notify you
>
> On Jan 31, 2013 5:05 PM, "Bart Coninckx"
> <bart.coninckx@telenet.be <ma...@telenet.be>>
> wrote:
>
> It is for Asterisk 11 - don't know for other versions. You
> probably have no issues because of the 1.8 version. To be
> sure the .sql files in the Asterisk source should be
> compared across versions.
>
> this one is missing:
>
> `useragent` varchar(20) DEFAULT NULL,
>
> complete list (I think) is on:
>
> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>
>
> If I bump into others, I'll report ASAP,
>
>
> BC
>
>
>
> On 01/31/13 06:21, Maxim Solodovnik wrote:
>> Is the OM meetme table incomplete?
>> My asterisk reports no issues :(
>>
>> could you provide me with missing fields and I'll add it.
>> My purpose was to create table with required fields only.
>>
>>
>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx
>> <bart.coninckx@telenet.be
>> <ma...@telenet.be>> wrote:
>>
>> Openmeetings installed them for me, that's why I
>> ended up with those. Using the Asterisk ones makes
>> more sense to me. Maybe it's a good idea to have 'em
>> removed from the install procedure.
>>
>> BC
>>
>>
>> On 01/30/13 22:30, Jeff Clay wrote:
>>>
>>> Bart,
>>>
>>> If you look in the source directory of your asterisk
>>> tar file, under contrib/realtime/mysql you’ll find
>>> the .sql files required for all the realtime
>>> drivers. I never thought to use the ones with OM.
>>>
>>> Jeff Clay
>>>
>>> Network Administrator
>>>
>>> Infotech Enterprises America
>>>
>>> 870-215-5506
>>>
>>> Ext. 1506
>>>
>>> *From:*Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>> *To:* user@openmeetings.apache.org
>>> <ma...@openmeetings.apache.org>
>>> *Cc:* Jeff Clay
>>> *Subject:* Re: SIP connectivity
>>>
>>> Well,
>>>
>>> I might have found one difference though:
>>>
>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>> dictates how the table should look like. I obviously
>>> used the one in the openmeetings mysql database, but
>>> this one seems to miss the table "useragent". I
>>> discovered this because it showed up in the logfiles.
>>>
>>> BC
>>>
>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>
>>> Bart,
>>>
>>> From an asterisk configuration standpoint there
>>> are very few differences between 1.8.x and 11.x.
>>> If memory serves, the only major changes that I
>>> ran into (in my production environment) was
>>> changes to SIP NAT values and the behavior of
>>> app_page() now uses confbridge instead of meetme
>>> to mix the audio. Also, TCP, TLS and
>>> app_confbridge got a major overhauling. There
>>> were of course many other changes and bug fixes,
>>> you can skim through the change log for full
>>> details, but I think that was the jist of it.
>>>
>>> Jeff Clay
>>>
>>> Network Administrator
>>>
>>> Infotech Enterprises America
>>>
>>> 870-215-5506
>>>
>>> Ext. 1506
>>>
>>> *From:*Bart Coninckx
>>> [mailto:bart.coninckx@telenet.be]
>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>> *To:* Maxim Solodovnik
>>> *Cc:* user
>>> *Subject:* Re: SIP connectivity
>>>
>>> I see - I'm willing to try the 11 version in the
>>> next fiew days if desired.
>>>
>>> BC
>>>
>>>
>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>
>>> I test the integration using
>>>
>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>
>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart
>>> Coninckx <bart.coninckx@telenet.be
>>> <ma...@telenet.be>> wrote:
>>>
>>> That is amazing - I initially tried to do
>>> the same thing by using the new chan_motif
>>> driver in Asterisk 11 which connects to a
>>> XMPP server.
>>>
>>> Are you guys using Asterisk 11? This version
>>> is the newest LTS version and has the best
>>> video capabilities.
>>>
>>> Cheers,
>>>
>>> BC
>>>
>>>
>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>
>>> red5sip will create special OM user in
>>> the room: "SIP Transport"
>>>
>>> after that you can call to the OM room
>>> using SIP hard or soft phone.
>>>
>>> We are currently testing it and trying
>>> to add video capabilities ...
>>>
>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart
>>> Coninckx <bart.coninckx@telenet.be
>>> <ma...@telenet.be>> wrote:
>>>
>>> Hi Jeff,
>>>
>>> In fact, I saw both pages, but none
>>> explain what they set up to do, just a
>>> bunch of command line instructions are
>>> given.
>>> Your "OM will create a meetme meeting as
>>> configured in the realtime meetme
>>> database" actually says it all in one go
>>> :-)
>>>
>>> cheers,
>>>
>>> BC
>>>
>>>
>>>
>>>
>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>
>>> Bart,
>>>
>>> OM will create a meetme meeting as
>>> configured in the realtime meetme
>>> database. Have you read this page
>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html
>>> ? You might also check out
>>> http://openmeetings.apache.org/red5sip-integration.html
>>> but I assume this is the one you're
>>> already referring to.
>>>
>>> Jeff Clay
>>> Network Administrator
>>> Infotech Enterprises America
>>> 870-215-5506
>>> Ext. 1506
>>>
>>> -----Original Message-----
>>> From: Bart Coninckx
>>> [mailto:bart.coninckx@telenet.be
>>> <ma...@telenet.be>]
>>> Sent: Monday, January 28, 2013 3:36 PM
>>> To: user@openmeetings.apache.org
>>> <ma...@openmeetings.apache.org>
>>> Subject: SIP connectivity
>>>
>>> Hi,
>>>
>>> I noticed some documentation on how to
>>> connect OM with a SIP proxy or server,
>>> more particularly with the MeetMe
>>> application in Asterisk.
>>>
>>> The exact goal or purpose is not
>>> mentionned however. Will OM callout to a
>>> MeetMe conference? Or is it the other
>>> way round?
>>>
>>>
>>> Cheers,
>>>
>>> Bc
>>>
>>> ________________________________
>>>
>>> DISCLAIMER:
>>>
>>> This email may contain confidential
>>> information and is intended only for the
>>> use of the specific individual(s) to
>>> which it is addressed. If you are not
>>> the intended recipient of this email,
>>> you are hereby notified that any
>>> unauthorized use, dissemination or
>>> copying of this email or the information
>>> contained in it or attached to it is
>>> strictly prohibited. If you received
>>> this message in error, please
>>> immediately notify the sender at
>>> Infotech and delete the original message.
>>>
>>>
>>>
>>> --
>>> WBR
>>> Maxim aka solomax
>>>
>>>
>>>
>>> --
>>> WBR
>>> Maxim aka solomax
>>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
> --
> WBR
> Maxim aka solomax
Re: SIP connectivity
Posted by Maxim Solodovnik <so...@gmail.com>.
I have updated the instruction (minor update)
On Thu, Jan 31, 2013 at 7:05 PM, Maxim Solodovnik <so...@gmail.com>wrote:
> Hello Bart,
>
> I just take a look at your URL ...
> OM does not create/use sipfriends DB table (at least from version 2.1)
> only meetme table is used
>
> so I'm afraid there is nothing to change here
>
> Here is the most recent instruction:
> http://openmeetings.apache.org/red5sip-integration_2.1.html
>
> Will ask our SIP guru to review it one more time :)
>
>
>
> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>wrote:
>
>> OK will add it and notify you
>> On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be>
>> wrote:
>>
>>> It is for Asterisk 11 - don't know for other versions. You probably
>>> have no issues because of the 1.8 version. To be sure the .sql files in the
>>> Asterisk source should be compared across versions.
>>>
>>> this one is missing:
>>>
>>> `useragent` varchar(20) DEFAULT NULL,
>>>
>>> complete list (I think) is on:
>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>
>>>
>>> If I bump into others, I'll report ASAP,
>>>
>>>
>>> BC
>>>
>>>
>>>
>>> On 01/31/13 06:21, Maxim Solodovnik wrote:
>>>
>>> Is the OM meetme table incomplete?
>>> My asterisk reports no issues :(
>>>
>>> could you provide me with missing fields and I'll add it.
>>> My purpose was to create table with required fields only.
>>>
>>>
>>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <bart.coninckx@telenet.be
>>> > wrote:
>>>
>>>> Openmeetings installed them for me, that's why I ended up with those.
>>>> Using the Asterisk ones makes more sense to me. Maybe it's a good idea to
>>>> have 'em removed from the install procedure.
>>>>
>>>> BC
>>>>
>>>>
>>>> On 01/30/13 22:30, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>>
>>>>
>>>> If you look in the source directory of your asterisk tar file, under
>>>> contrib/realtime/mysql you’ll find the .sql files required for all the
>>>> realtime drivers. I never thought to use the ones with OM.
>>>>
>>>>
>>>>
>>>> Jeff Clay
>>>>
>>>> Network Administrator
>>>>
>>>> Infotech Enterprises America
>>>>
>>>> 870-215-5506
>>>>
>>>> Ext. 1506
>>>>
>>>>
>>>>
>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>
>>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>>> *To:* user@openmeetings.apache.org
>>>> *Cc:* Jeff Clay
>>>> *Subject:* Re: SIP connectivity
>>>>
>>>>
>>>>
>>>> Well,
>>>>
>>>> I might have found one difference though:
>>>>
>>>>
>>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>>> dictates how the table should look like. I obviously used the one in the
>>>> openmeetings mysql database, but this one seems to miss the table
>>>> "useragent". I discovered this because it showed up in the logfiles.
>>>>
>>>> BC
>>>>
>>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>>
>>>>
>>>> From an asterisk configuration standpoint there are very few
>>>> differences between 1.8.x and 11.x. If memory serves, the only major
>>>> changes that I ran into (in my production environment) was changes to SIP
>>>> NAT values and the behavior of app_page() now uses confbridge instead of
>>>> meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major
>>>> overhauling. There were of course many other changes and bug fixes, you can
>>>> skim through the change log for full details, but I think that was the jist
>>>> of it.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Jeff Clay
>>>>
>>>> Network Administrator
>>>>
>>>> Infotech Enterprises America
>>>>
>>>> 870-215-5506
>>>>
>>>> Ext. 1506
>>>>
>>>>
>>>>
>>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>>
>>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>>> *To:* Maxim Solodovnik
>>>> *Cc:* user
>>>> *Subject:* Re: SIP connectivity
>>>>
>>>>
>>>>
>>>> I see - I'm willing to try the 11 version in the next fiew days if
>>>> desired.
>>>>
>>>> BC
>>>>
>>>>
>>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>>
>>>> I test the integration using
>>>>
>>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>>
>>>>
>>>>
>>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <
>>>> bart.coninckx@telenet.be> wrote:
>>>>
>>>> That is amazing - I initially tried to do the same thing by using the
>>>> new chan_motif driver in Asterisk 11 which connects to a XMPP server.
>>>>
>>>> Are you guys using Asterisk 11? This version is the newest LTS version
>>>> and has the best video capabilities.
>>>>
>>>> Cheers,
>>>>
>>>> BC
>>>>
>>>>
>>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>>
>>>> red5sip will create special OM user in the room: "SIP Transport"
>>>>
>>>> after that you can call to the OM room using SIP hard or soft phone.
>>>>
>>>>
>>>>
>>>> We are currently testing it and trying to add video capabilities ...
>>>>
>>>>
>>>>
>>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <
>>>> bart.coninckx@telenet.be> wrote:
>>>>
>>>> Hi Jeff,
>>>>
>>>> In fact, I saw both pages, but none explain what they set up to do,
>>>> just a bunch of command line instructions are given.
>>>> Your "OM will create a meetme meeting as configured in the realtime
>>>> meetme database" actually says it all in one go :-)
>>>>
>>>> cheers,
>>>>
>>>> BC
>>>>
>>>>
>>>>
>>>>
>>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>>
>>>> Bart,
>>>>
>>>> OM will create a meetme meeting as configured in the realtime meetme
>>>> database. Have you read this page
>>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
>>>> http://openmeetings.apache.org/red5sip-integration.html but I assume
>>>> this is the one you're already referring to.
>>>>
>>>> Jeff Clay
>>>> Network Administrator
>>>> Infotech Enterprises America
>>>> 870-215-5506
>>>> Ext. 1506
>>>>
>>>> -----Original Message-----
>>>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>>> Sent: Monday, January 28, 2013 3:36 PM
>>>> To: user@openmeetings.apache.org
>>>> Subject: SIP connectivity
>>>>
>>>> Hi,
>>>>
>>>> I noticed some documentation on how to connect OM with a SIP proxy or
>>>> server, more particularly with the MeetMe application in Asterisk.
>>>>
>>>> The exact goal or purpose is not mentionned however. Will OM callout to
>>>> a MeetMe conference? Or is it the other way round?
>>>>
>>>>
>>>> Cheers,
>>>>
>>>> Bc
>>>>
>>>> ________________________________
>>>>
>>>> DISCLAIMER:
>>>>
>>>> This email may contain confidential information and is intended only
>>>> for the use of the specific individual(s) to which it is addressed. If you
>>>> are not the intended recipient of this email, you are hereby notified that
>>>> any unauthorized use, dissemination or copying of this email or the
>>>> information contained in it or attached to it is strictly prohibited. If
>>>> you received this message in error, please immediately notify the sender at
>>>> Infotech and delete the original message.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>
>>>
>>> --
>>> WBR
>>> Maxim aka solomax
>>>
>>>
>>>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
Re: SIP connectivity
Posted by Vieri <re...@yahoo.com>.
I have this in asterisk:
TABLE meetme (
bookid int(11) auto_increment,
confno char(80) DEFAULT '0' NOT NULL,
starttime datetime default '1900-01-01 12:00:00',
endtime datetime default '2038-01-01 12:00:00',
pin char(20) NULL,
adminpin char(20) NULL,
opts char(20) NULL,
adminopts char(20) NULL,
recordingfilename char(80) NULL,
recordingformat char(10) NULL,
maxusers int(11) NULL,
members integer DEFAULT 0 NOT NULL,
index confno (confno,starttime,endtime),
PRIMARY KEY (bookid)
By the way, the OM-Asterisk guide assumes that Asterisk realtime will link to the openmeetings database (to the meetme table and maybe also to the sipusers table - not sure). Could it be the other way around? ie. let openmeetings read/write to the meetme and/or sipusers table in an "asterisk" database but all other tables are left within the openmeetings database.
Vieri
--- On Thu, 1/31/13, Maxim Solodovnik <so...@gmail.com> wrote:
From: Maxim Solodovnik <so...@gmail.com>
Subject: Re: SIP connectivity
To: "Bart Coninckx" <ba...@telenet.be>
Cc: "user" <us...@openmeetings.apache.org>
Date: Thursday, January 31, 2013, 7:05 AM
Hello Bart,
I just take a look at your URL ...OM does not create/use sipfriends DB table (at least from version 2.1)only meetme table is used
so I'm afraid there is nothing to change here
Here is the most recent instruction:http://openmeetings.apache.org/red5sip-integration_2.1.html
Will ask our SIP guru to review it one more time :)
On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com> wrote:
OK will add it and notify you
On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be> wrote:
It is for Asterisk 11 - don't know for
other versions. You probably have no issues because of the 1.8
version. To be sure the .sql files in the Asterisk source should
be compared across versions.
this one is missing:
`useragent` varchar(20) DEFAULT NULL,
complete list (I think) is on:
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
If I bump into others, I'll report ASAP,
BC
On 01/31/13 06:21, Maxim Solodovnik wrote:
Is the OM meetme table incomplete?
My asterisk reports no issues :(
could you provide me with missing fields and I'll
add it.
My purpose was to create table with required
fields only.
On Thu, Jan 31, 2013 at 4:45 AM, Bart
Coninckx <ba...@telenet.be>
wrote:
Openmeetings installed them for me, that's why I
ended up with those. Using the Asterisk ones makes more
sense to me. Maybe it's a good idea to have 'em removed
from the install procedure.
BC
On 01/30/13 22:30, Jeff Clay wrote:
Bart,
If
you look in the source directory of your
asterisk tar file, under
contrib/realtime/mysql you’ll find the .sql
files required for all the realtime drivers. I
never thought to use the ones with OM.
Jeff
Clay
Network
Administrator
Infotech
Enterprises America
870-215-5506
Ext.
1506
From:
Bart Coninckx [mailto:bart.coninckx@telenet.be]
Sent: Wednesday, January 30, 2013
3:19 PM
To: user@openmeetings.apache.org
Cc: Jeff Clay
Subject: Re: SIP connectivity
Well,
I might have found one difference though:
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
dictates how the table should look like. I
obviously used the one in the openmeetings
mysql database, but this one seems to miss the
table "useragent". I discovered this because
it showed up in the logfiles.
BC
On 01/29/13 14:41, Jeff Clay wrote:
Bart,
From
an asterisk configuration standpoint there
are very few differences between 1.8.x and
11.x. If memory serves, the only major
changes that I ran into (in my production
environment) was changes to SIP NAT values
and the behavior of app_page() now uses
confbridge instead of meetme to mix the
audio. Also, TCP, TLS and app_confbridge got
a major overhauling. There were of course
many other changes and bug fixes, you can
skim through the change log for full
details, but I think that was the jist of
it.
Jeff
Clay
Network
Administrator
Infotech
Enterprises America
870-215-5506
Ext.
1506
From:
Bart Coninckx [mailto:bart.coninckx@telenet.be]
Sent: Tuesday, January 29, 2013
4:02 AM
To: Maxim Solodovnik
Cc: user
Subject: Re: SIP connectivity
I see - I'm willing to
try the 11 version in the next fiew days if
desired.
BC
On 01/29/13 10:57, Maxim Solodovnik wrote:
I test the integration
using
Asterisk 1.8.13.1
(Ubuntu 12.10)
On Tue, Jan 29, 2013
at 4:51 PM, Bart Coninckx <ba...@telenet.be>
wrote:
That is amazing -
I initially tried to do the same
thing by using the new chan_motif
driver in Asterisk 11 which connects
to a XMPP server.
Are you guys using Asterisk 11? This
version is the newest LTS version
and has the best video capabilities.
Cheers,
BC
On 01/29/13 02:44, Maxim Solodovnik
wrote:
red5sip will
create special OM user in the
room: "SIP Transport"
after that
you can call to the OM room
using SIP hard or soft phone.
We are
currently testing it and trying
to add video capabilities ...
On Tue, Jan
29, 2013 at 4:44 AM, Bart
Coninckx <ba...@telenet.be>
wrote:
Hi Jeff,
In fact, I saw both pages, but
none explain what they set up to
do, just a bunch of command line
instructions are given.
Your "OM will create a meetme
meeting as configured in the
realtime meetme database"
actually says it all in one go
:-)
cheers,
BC
On 01/28/13 22:38, Jeff Clay
wrote:
Bart,
OM will create a meetme
meeting as configured in the
realtime meetme database.
Have you read this page https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html
? You might also check
out
http://openmeetings.apache.org/red5sip-integration.html but I assume
this is the one you're
already referring to.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
-----Original Message-----
From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
Sent: Monday, January 28,
2013 3:36 PM
To: user@openmeetings.apache.org
Subject: SIP connectivity
Hi,
I noticed some documentation
on how to connect OM with a
SIP proxy or server, more
particularly with the MeetMe
application in Asterisk.
The exact goal or purpose is
not mentionned however. Will
OM callout to a MeetMe
conference? Or is it the
other way round?
Cheers,
Bc
________________________________
DISCLAIMER:
This email may contain
confidential information and
is intended only for the use
of the specific
individual(s) to which it is
addressed. If you are not
the intended recipient of
this email, you are hereby
notified that any
unauthorized use,
dissemination or copying of
this email or the
information contained in it
or attached to it is
strictly prohibited. If you
received this message in
error, please immediately
notify the sender at
Infotech and delete the
original message.
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: SIP connectivity
Posted by Maxim Solodovnik <so...@gmail.com>.
Hello Bart,
I just take a look at your URL ...
OM does not create/use sipfriends DB table (at least from version 2.1)
only meetme table is used
so I'm afraid there is nothing to change here
Here is the most recent instruction:
http://openmeetings.apache.org/red5sip-integration_2.1.html
Will ask our SIP guru to review it one more time :)
On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik <so...@gmail.com>wrote:
> OK will add it and notify you
> On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be> wrote:
>
>> It is for Asterisk 11 - don't know for other versions. You probably
>> have no issues because of the 1.8 version. To be sure the .sql files in the
>> Asterisk source should be compared across versions.
>>
>> this one is missing:
>>
>> `useragent` varchar(20) DEFAULT NULL,
>>
>> complete list (I think) is on:
>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>
>>
>> If I bump into others, I'll report ASAP,
>>
>>
>> BC
>>
>>
>>
>> On 01/31/13 06:21, Maxim Solodovnik wrote:
>>
>> Is the OM meetme table incomplete?
>> My asterisk reports no issues :(
>>
>> could you provide me with missing fields and I'll add it.
>> My purpose was to create table with required fields only.
>>
>>
>> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <ba...@telenet.be>wrote:
>>
>>> Openmeetings installed them for me, that's why I ended up with those.
>>> Using the Asterisk ones makes more sense to me. Maybe it's a good idea to
>>> have 'em removed from the install procedure.
>>>
>>> BC
>>>
>>>
>>> On 01/30/13 22:30, Jeff Clay wrote:
>>>
>>> Bart,
>>>
>>>
>>>
>>> If you look in the source directory of your asterisk tar file, under
>>> contrib/realtime/mysql you’ll find the .sql files required for all the
>>> realtime drivers. I never thought to use the ones with OM.
>>>
>>>
>>>
>>> Jeff Clay
>>>
>>> Network Administrator
>>>
>>> Infotech Enterprises America
>>>
>>> 870-215-5506
>>>
>>> Ext. 1506
>>>
>>>
>>>
>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>
>>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>>> *To:* user@openmeetings.apache.org
>>> *Cc:* Jeff Clay
>>> *Subject:* Re: SIP connectivity
>>>
>>>
>>>
>>> Well,
>>>
>>> I might have found one difference though:
>>>
>>>
>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>>> dictates how the table should look like. I obviously used the one in the
>>> openmeetings mysql database, but this one seems to miss the table
>>> "useragent". I discovered this because it showed up in the logfiles.
>>>
>>> BC
>>>
>>> On 01/29/13 14:41, Jeff Clay wrote:
>>>
>>> Bart,
>>>
>>>
>>>
>>> From an asterisk configuration standpoint there are very few differences
>>> between 1.8.x and 11.x. If memory serves, the only major changes that I ran
>>> into (in my production environment) was changes to SIP NAT values and the
>>> behavior of app_page() now uses confbridge instead of meetme to mix the
>>> audio. Also, TCP, TLS and app_confbridge got a major overhauling. There
>>> were of course many other changes and bug fixes, you can skim through the
>>> change log for full details, but I think that was the jist of it.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> Jeff Clay
>>>
>>> Network Administrator
>>>
>>> Infotech Enterprises America
>>>
>>> 870-215-5506
>>>
>>> Ext. 1506
>>>
>>>
>>>
>>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>>
>>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>>> *To:* Maxim Solodovnik
>>> *Cc:* user
>>> *Subject:* Re: SIP connectivity
>>>
>>>
>>>
>>> I see - I'm willing to try the 11 version in the next fiew days if
>>> desired.
>>>
>>> BC
>>>
>>>
>>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>>
>>> I test the integration using
>>>
>>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>>
>>>
>>>
>>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <ba...@telenet.be>
>>> wrote:
>>>
>>> That is amazing - I initially tried to do the same thing by using the
>>> new chan_motif driver in Asterisk 11 which connects to a XMPP server.
>>>
>>> Are you guys using Asterisk 11? This version is the newest LTS version
>>> and has the best video capabilities.
>>>
>>> Cheers,
>>>
>>> BC
>>>
>>>
>>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>>
>>> red5sip will create special OM user in the room: "SIP Transport"
>>>
>>> after that you can call to the OM room using SIP hard or soft phone.
>>>
>>>
>>>
>>> We are currently testing it and trying to add video capabilities ...
>>>
>>>
>>>
>>> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <ba...@telenet.be>
>>> wrote:
>>>
>>> Hi Jeff,
>>>
>>> In fact, I saw both pages, but none explain what they set up to do, just
>>> a bunch of command line instructions are given.
>>> Your "OM will create a meetme meeting as configured in the realtime
>>> meetme database" actually says it all in one go :-)
>>>
>>> cheers,
>>>
>>> BC
>>>
>>>
>>>
>>>
>>> On 01/28/13 22:38, Jeff Clay wrote:
>>>
>>> Bart,
>>>
>>> OM will create a meetme meeting as configured in the realtime meetme
>>> database. Have you read this page
>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
>>> http://openmeetings.apache.org/red5sip-integration.html but I assume
>>> this is the one you're already referring to.
>>>
>>> Jeff Clay
>>> Network Administrator
>>> Infotech Enterprises America
>>> 870-215-5506
>>> Ext. 1506
>>>
>>> -----Original Message-----
>>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>> Sent: Monday, January 28, 2013 3:36 PM
>>> To: user@openmeetings.apache.org
>>> Subject: SIP connectivity
>>>
>>> Hi,
>>>
>>> I noticed some documentation on how to connect OM with a SIP proxy or
>>> server, more particularly with the MeetMe application in Asterisk.
>>>
>>> The exact goal or purpose is not mentionned however. Will OM callout to
>>> a MeetMe conference? Or is it the other way round?
>>>
>>>
>>> Cheers,
>>>
>>> Bc
>>>
>>> ________________________________
>>>
>>> DISCLAIMER:
>>>
>>> This email may contain confidential information and is intended only for
>>> the use of the specific individual(s) to which it is addressed. If you are
>>> not the intended recipient of this email, you are hereby notified that any
>>> unauthorized use, dissemination or copying of this email or the information
>>> contained in it or attached to it is strictly prohibited. If you received
>>> this message in error, please immediately notify the sender at Infotech and
>>> delete the original message.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> --
>>> WBR
>>> Maxim aka solomax
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> --
>>> WBR
>>> Maxim aka solomax
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
--
WBR
Maxim aka solomax
Re: SIP connectivity
Posted by Maxim Solodovnik <so...@gmail.com>.
OK will add it and notify you
On Jan 31, 2013 5:05 PM, "Bart Coninckx" <ba...@telenet.be> wrote:
> It is for Asterisk 11 - don't know for other versions. You probably have
> no issues because of the 1.8 version. To be sure the .sql files in the
> Asterisk source should be compared across versions.
>
> this one is missing:
>
> `useragent` varchar(20) DEFAULT NULL,
>
> complete list (I think) is on:
> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>
>
> If I bump into others, I'll report ASAP,
>
>
> BC
>
>
>
> On 01/31/13 06:21, Maxim Solodovnik wrote:
>
> Is the OM meetme table incomplete?
> My asterisk reports no issues :(
>
> could you provide me with missing fields and I'll add it.
> My purpose was to create table with required fields only.
>
>
> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <ba...@telenet.be>wrote:
>
>> Openmeetings installed them for me, that's why I ended up with those.
>> Using the Asterisk ones makes more sense to me. Maybe it's a good idea to
>> have 'em removed from the install procedure.
>>
>> BC
>>
>>
>> On 01/30/13 22:30, Jeff Clay wrote:
>>
>> Bart,
>>
>>
>>
>> If you look in the source directory of your asterisk tar file, under
>> contrib/realtime/mysql you’ll find the .sql files required for all the
>> realtime drivers. I never thought to use the ones with OM.
>>
>>
>>
>> Jeff Clay
>>
>> Network Administrator
>>
>> Infotech Enterprises America
>>
>> 870-215-5506
>>
>> Ext. 1506
>>
>>
>>
>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>
>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>> *To:* user@openmeetings.apache.org
>> *Cc:* Jeff Clay
>> *Subject:* Re: SIP connectivity
>>
>>
>>
>> Well,
>>
>> I might have found one difference though:
>>
>>
>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>> dictates how the table should look like. I obviously used the one in the
>> openmeetings mysql database, but this one seems to miss the table
>> "useragent". I discovered this because it showed up in the logfiles.
>>
>> BC
>>
>> On 01/29/13 14:41, Jeff Clay wrote:
>>
>> Bart,
>>
>>
>>
>> From an asterisk configuration standpoint there are very few differences
>> between 1.8.x and 11.x. If memory serves, the only major changes that I ran
>> into (in my production environment) was changes to SIP NAT values and the
>> behavior of app_page() now uses confbridge instead of meetme to mix the
>> audio. Also, TCP, TLS and app_confbridge got a major overhauling. There
>> were of course many other changes and bug fixes, you can skim through the
>> change log for full details, but I think that was the jist of it.
>>
>>
>>
>>
>>
>>
>>
>> Jeff Clay
>>
>> Network Administrator
>>
>> Infotech Enterprises America
>>
>> 870-215-5506
>>
>> Ext. 1506
>>
>>
>>
>> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>>
>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>> *To:* Maxim Solodovnik
>> *Cc:* user
>> *Subject:* Re: SIP connectivity
>>
>>
>>
>> I see - I'm willing to try the 11 version in the next fiew days if
>> desired.
>>
>> BC
>>
>>
>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>
>> I test the integration using
>>
>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>
>>
>>
>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <ba...@telenet.be>
>> wrote:
>>
>> That is amazing - I initially tried to do the same thing by using the new
>> chan_motif driver in Asterisk 11 which connects to a XMPP server.
>>
>> Are you guys using Asterisk 11? This version is the newest LTS version
>> and has the best video capabilities.
>>
>> Cheers,
>>
>> BC
>>
>>
>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>
>> red5sip will create special OM user in the room: "SIP Transport"
>>
>> after that you can call to the OM room using SIP hard or soft phone.
>>
>>
>>
>> We are currently testing it and trying to add video capabilities ...
>>
>>
>>
>> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <ba...@telenet.be>
>> wrote:
>>
>> Hi Jeff,
>>
>> In fact, I saw both pages, but none explain what they set up to do, just
>> a bunch of command line instructions are given.
>> Your "OM will create a meetme meeting as configured in the realtime
>> meetme database" actually says it all in one go :-)
>>
>> cheers,
>>
>> BC
>>
>>
>>
>>
>> On 01/28/13 22:38, Jeff Clay wrote:
>>
>> Bart,
>>
>> OM will create a meetme meeting as configured in the realtime meetme
>> database. Have you read this page
>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
>> http://openmeetings.apache.org/red5sip-integration.html but I assume
>> this is the one you're already referring to.
>>
>> Jeff Clay
>> Network Administrator
>> Infotech Enterprises America
>> 870-215-5506
>> Ext. 1506
>>
>> -----Original Message-----
>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
>> Sent: Monday, January 28, 2013 3:36 PM
>> To: user@openmeetings.apache.org
>> Subject: SIP connectivity
>>
>> Hi,
>>
>> I noticed some documentation on how to connect OM with a SIP proxy or
>> server, more particularly with the MeetMe application in Asterisk.
>>
>> The exact goal or purpose is not mentionned however. Will OM callout to a
>> MeetMe conference? Or is it the other way round?
>>
>>
>> Cheers,
>>
>> Bc
>>
>> ________________________________
>>
>> DISCLAIMER:
>>
>> This email may contain confidential information and is intended only for
>> the use of the specific individual(s) to which it is addressed. If you are
>> not the intended recipient of this email, you are hereby notified that any
>> unauthorized use, dissemination or copying of this email or the information
>> contained in it or attached to it is strictly prohibited. If you received
>> this message in error, please immediately notify the sender at Infotech and
>> delete the original message.
>>
>>
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>>
>>
>>
>>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
Re: SIP connectivity
Posted by Bart Coninckx <ba...@telenet.be>.
It is for Asterisk 11 - don't know for other versions. You probably have
no issues because of the 1.8 version. To be sure the .sql files in the
Asterisk source should be compared across versions.
this one is missing:
`useragent` varchar(20) DEFAULT NULL,
complete list (I think) is on:
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
If I bump into others, I'll report ASAP,
BC
On 01/31/13 06:21, Maxim Solodovnik wrote:
> Is the OM meetme table incomplete?
> My asterisk reports no issues :(
>
> could you provide me with missing fields and I'll add it.
> My purpose was to create table with required fields only.
>
>
> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx
> <bart.coninckx@telenet.be <ma...@telenet.be>> wrote:
>
> Openmeetings installed them for me, that's why I ended up with
> those. Using the Asterisk ones makes more sense to me. Maybe it's
> a good idea to have 'em removed from the install procedure.
>
> BC
>
>
> On 01/30/13 22:30, Jeff Clay wrote:
>>
>> Bart,
>>
>> If you look in the source directory of your asterisk tar file,
>> under contrib/realtime/mysql you’ll find the .sql files required
>> for all the realtime drivers. I never thought to use the ones
>> with OM.
>>
>> Jeff Clay
>>
>> Network Administrator
>>
>> Infotech Enterprises America
>>
>> 870-215-5506
>>
>> Ext. 1506
>>
>> *From:*Bart Coninckx [mailto:bart.coninckx@telenet.be]
>> *Sent:* Wednesday, January 30, 2013 3:19 PM
>> *To:* user@openmeetings.apache.org
>> <ma...@openmeetings.apache.org>
>> *Cc:* Jeff Clay
>> *Subject:* Re: SIP connectivity
>>
>> Well,
>>
>> I might have found one difference though:
>>
>> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
>> dictates how the table should look like. I obviously used the one
>> in the openmeetings mysql database, but this one seems to miss
>> the table "useragent". I discovered this because it showed up in
>> the logfiles.
>>
>> BC
>>
>> On 01/29/13 14:41, Jeff Clay wrote:
>>
>> Bart,
>>
>> From an asterisk configuration standpoint there are very few
>> differences between 1.8.x and 11.x. If memory serves, the
>> only major changes that I ran into (in my production
>> environment) was changes to SIP NAT values and the behavior
>> of app_page() now uses confbridge instead of meetme to mix
>> the audio. Also, TCP, TLS and app_confbridge got a major
>> overhauling. There were of course many other changes and bug
>> fixes, you can skim through the change log for full details,
>> but I think that was the jist of it.
>>
>> Jeff Clay
>>
>> Network Administrator
>>
>> Infotech Enterprises America
>>
>> 870-215-5506
>>
>> Ext. 1506
>>
>> *From:*Bart Coninckx [mailto:bart.coninckx@telenet.be]
>> *Sent:* Tuesday, January 29, 2013 4:02 AM
>> *To:* Maxim Solodovnik
>> *Cc:* user
>> *Subject:* Re: SIP connectivity
>>
>> I see - I'm willing to try the 11 version in the next fiew
>> days if desired.
>>
>> BC
>>
>>
>> On 01/29/13 10:57, Maxim Solodovnik wrote:
>>
>> I test the integration using
>>
>> Asterisk 1.8.13.1 (Ubuntu 12.10)
>>
>> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx
>> <bart.coninckx@telenet.be
>> <ma...@telenet.be>> wrote:
>>
>> That is amazing - I initially tried to do the same thing
>> by using the new chan_motif driver in Asterisk 11 which
>> connects to a XMPP server.
>>
>> Are you guys using Asterisk 11? This version is the
>> newest LTS version and has the best video capabilities.
>>
>> Cheers,
>>
>> BC
>>
>>
>> On 01/29/13 02:44, Maxim Solodovnik wrote:
>>
>> red5sip will create special OM user in the room: "SIP
>> Transport"
>>
>> after that you can call to the OM room using SIP hard
>> or soft phone.
>>
>> We are currently testing it and trying to add video
>> capabilities ...
>>
>> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx
>> <bart.coninckx@telenet.be
>> <ma...@telenet.be>> wrote:
>>
>> Hi Jeff,
>>
>> In fact, I saw both pages, but none explain what they
>> set up to do, just a bunch of command line
>> instructions are given.
>> Your "OM will create a meetme meeting as configured
>> in the realtime meetme database" actually says it all
>> in one go :-)
>>
>> cheers,
>>
>> BC
>>
>>
>>
>>
>> On 01/28/13 22:38, Jeff Clay wrote:
>>
>> Bart,
>>
>> OM will create a meetme meeting as configured in the
>> realtime meetme database. Have you read this page
>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html
>> ? You might also check out
>> http://openmeetings.apache.org/red5sip-integration.html
>> but I assume this is the one you're already referring to.
>>
>> Jeff Clay
>> Network Administrator
>> Infotech Enterprises America
>> 870-215-5506
>> Ext. 1506
>>
>> -----Original Message-----
>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be
>> <ma...@telenet.be>]
>> Sent: Monday, January 28, 2013 3:36 PM
>> To: user@openmeetings.apache.org
>> <ma...@openmeetings.apache.org>
>> Subject: SIP connectivity
>>
>> Hi,
>>
>> I noticed some documentation on how to connect OM
>> with a SIP proxy or server, more particularly with
>> the MeetMe application in Asterisk.
>>
>> The exact goal or purpose is not mentionned however.
>> Will OM callout to a MeetMe conference? Or is it the
>> other way round?
>>
>>
>> Cheers,
>>
>> Bc
>>
>> ________________________________
>>
>> DISCLAIMER:
>>
>> This email may contain confidential information and
>> is intended only for the use of the specific
>> individual(s) to which it is addressed. If you are
>> not the intended recipient of this email, you are
>> hereby notified that any unauthorized use,
>> dissemination or copying of this email or the
>> information contained in it or attached to it is
>> strictly prohibited. If you received this message in
>> error, please immediately notify the sender at
>> Infotech and delete the original message.
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>
>
>
>
> --
> WBR
> Maxim aka solomax
Re: SIP connectivity
Posted by Maxim Solodovnik <so...@gmail.com>.
Is the OM meetme table incomplete?
My asterisk reports no issues :(
could you provide me with missing fields and I'll add it.
My purpose was to create table with required fields only.
On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx <ba...@telenet.be>wrote:
> Openmeetings installed them for me, that's why I ended up with those.
> Using the Asterisk ones makes more sense to me. Maybe it's a good idea to
> have 'em removed from the install procedure.
>
> BC
>
>
> On 01/30/13 22:30, Jeff Clay wrote:
>
> Bart,****
>
> ** **
>
> If you look in the source directory of your asterisk tar file, under
> contrib/realtime/mysql you’ll find the .sql files required for all the
> realtime drivers. I never thought to use the ones with OM.****
>
> ** **
>
> Jeff Clay****
>
> Network Administrator****
>
> Infotech Enterprises America****
>
> 870-215-5506****
>
> Ext. 1506****
>
> ** **
>
> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>
> *Sent:* Wednesday, January 30, 2013 3:19 PM
> *To:* user@openmeetings.apache.org
> *Cc:* Jeff Clay
> *Subject:* Re: SIP connectivity****
>
> ** **
>
> Well,
>
> I might have found one difference though:
>
>
> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
> dictates how the table should look like. I obviously used the one in the
> openmeetings mysql database, but this one seems to miss the table
> "useragent". I discovered this because it showed up in the logfiles.
>
> BC
>
> On 01/29/13 14:41, Jeff Clay wrote:****
>
> Bart,****
>
> ****
>
> From an asterisk configuration standpoint there are very few differences
> between 1.8.x and 11.x. If memory serves, the only major changes that I ran
> into (in my production environment) was changes to SIP NAT values and the
> behavior of app_page() now uses confbridge instead of meetme to mix the
> audio. Also, TCP, TLS and app_confbridge got a major overhauling. There
> were of course many other changes and bug fixes, you can skim through the
> change log for full details, but I think that was the jist of it.****
>
> ****
>
> ****
>
> ****
>
> Jeff Clay****
>
> Network Administrator****
>
> Infotech Enterprises America****
>
> 870-215-5506****
>
> Ext. 1506****
>
> ****
>
> *From:* Bart Coninckx [mailto:bart.coninckx@telenet.be<ba...@telenet.be>]
>
> *Sent:* Tuesday, January 29, 2013 4:02 AM
> *To:* Maxim Solodovnik
> *Cc:* user
> *Subject:* Re: SIP connectivity****
>
> ****
>
> I see - I'm willing to try the 11 version in the next fiew days if
> desired.
>
> BC
>
>
> On 01/29/13 10:57, Maxim Solodovnik wrote:****
>
> I test the integration using ****
>
> Asterisk 1.8.13.1 (Ubuntu 12.10)****
>
> ****
>
> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <ba...@telenet.be>
> wrote:****
>
> That is amazing - I initially tried to do the same thing by using the new
> chan_motif driver in Asterisk 11 which connects to a XMPP server.
>
> Are you guys using Asterisk 11? This version is the newest LTS version and
> has the best video capabilities.
>
> Cheers,
>
> BC
>
>
> On 01/29/13 02:44, Maxim Solodovnik wrote:****
>
> red5sip will create special OM user in the room: "SIP Transport" ****
>
> after that you can call to the OM room using SIP hard or soft phone.****
>
> ****
>
> We are currently testing it and trying to add video capabilities ...****
>
> ****
>
> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <ba...@telenet.be>
> wrote:****
>
> Hi Jeff,
>
> In fact, I saw both pages, but none explain what they set up to do, just a
> bunch of command line instructions are given.
> Your "OM will create a meetme meeting as configured in the realtime meetme
> database" actually says it all in one go :-)
>
> cheers,
>
> BC ****
>
>
>
>
> On 01/28/13 22:38, Jeff Clay wrote:****
>
> Bart,
>
> OM will create a meetme meeting as configured in the realtime meetme
> database. Have you read this page
> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
> http://openmeetings.apache.org/red5sip-integration.html but I assume this
> is the one you're already referring to.
>
> Jeff Clay
> Network Administrator
> Infotech Enterprises America
> 870-215-5506
> Ext. 1506
>
> -----Original Message-----
> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
> Sent: Monday, January 28, 2013 3:36 PM
> To: user@openmeetings.apache.org
> Subject: SIP connectivity
>
> Hi,
>
> I noticed some documentation on how to connect OM with a SIP proxy or
> server, more particularly with the MeetMe application in Asterisk.
>
> The exact goal or purpose is not mentionned however. Will OM callout to a
> MeetMe conference? Or is it the other way round?
>
>
> Cheers,
>
> Bc
>
> ________________________________
>
> DISCLAIMER:
>
> This email may contain confidential information and is intended only for
> the use of the specific individual(s) to which it is addressed. If you are
> not the intended recipient of this email, you are hereby notified that any
> unauthorized use, dissemination or copying of this email or the information
> contained in it or attached to it is strictly prohibited. If you received
> this message in error, please immediately notify the sender at Infotech and
> delete the original message.****
>
> ****
>
>
>
> ****
>
> ****
>
> --
> WBR
> Maxim aka solomax ****
>
> ****
>
>
>
> ****
>
> ****
>
> --
> WBR
> Maxim aka solomax ****
>
> ****
>
> ** **
>
>
>
--
WBR
Maxim aka solomax
Re: SIP connectivity
Posted by Bart Coninckx <ba...@telenet.be>.
Openmeetings installed them for me, that's why I ended up with those.
Using the Asterisk ones makes more sense to me. Maybe it's a good idea
to have 'em removed from the install procedure.
BC
On 01/30/13 22:30, Jeff Clay wrote:
>
> Bart,
>
> If you look in the source directory of your asterisk tar file, under
> contrib/realtime/mysql you’ll find the .sql files required for all the
> realtime drivers. I never thought to use the ones with OM.
>
> Jeff Clay
>
> Network Administrator
>
> Infotech Enterprises America
>
> 870-215-5506
>
> Ext. 1506
>
> *From:*Bart Coninckx [mailto:bart.coninckx@telenet.be]
> *Sent:* Wednesday, January 30, 2013 3:19 PM
> *To:* user@openmeetings.apache.org
> *Cc:* Jeff Clay
> *Subject:* Re: SIP connectivity
>
> Well,
>
> I might have found one difference though:
>
> https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
> dictates how the table should look like. I obviously used the one in
> the openmeetings mysql database, but this one seems to miss the table
> "useragent". I discovered this because it showed up in the logfiles.
>
> BC
>
> On 01/29/13 14:41, Jeff Clay wrote:
>
> Bart,
>
> From an asterisk configuration standpoint there are very few
> differences between 1.8.x and 11.x. If memory serves, the only
> major changes that I ran into (in my production environment) was
> changes to SIP NAT values and the behavior of app_page() now uses
> confbridge instead of meetme to mix the audio. Also, TCP, TLS and
> app_confbridge got a major overhauling. There were of course many
> other changes and bug fixes, you can skim through the change log
> for full details, but I think that was the jist of it.
>
> Jeff Clay
>
> Network Administrator
>
> Infotech Enterprises America
>
> 870-215-5506
>
> Ext. 1506
>
> *From:*Bart Coninckx [mailto:bart.coninckx@telenet.be]
> *Sent:* Tuesday, January 29, 2013 4:02 AM
> *To:* Maxim Solodovnik
> *Cc:* user
> *Subject:* Re: SIP connectivity
>
> I see - I'm willing to try the 11 version in the next fiew days if
> desired.
>
> BC
>
>
> On 01/29/13 10:57, Maxim Solodovnik wrote:
>
> I test the integration using
>
> Asterisk 1.8.13.1 (Ubuntu 12.10)
>
> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx
> <bart.coninckx@telenet.be <ma...@telenet.be>>
> wrote:
>
> That is amazing - I initially tried to do the same thing by
> using the new chan_motif driver in Asterisk 11 which connects
> to a XMPP server.
>
> Are you guys using Asterisk 11? This version is the newest LTS
> version and has the best video capabilities.
>
> Cheers,
>
> BC
>
>
> On 01/29/13 02:44, Maxim Solodovnik wrote:
>
> red5sip will create special OM user in the room: "SIP
> Transport"
>
> after that you can call to the OM room using SIP hard or
> soft phone.
>
> We are currently testing it and trying to add video
> capabilities ...
>
> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx
> <bart.coninckx@telenet.be
> <ma...@telenet.be>> wrote:
>
> Hi Jeff,
>
> In fact, I saw both pages, but none explain what they set
> up to do, just a bunch of command line instructions are given.
> Your "OM will create a meetme meeting as configured in the
> realtime meetme database" actually says it all in one go :-)
>
> cheers,
>
> BC
>
>
>
>
> On 01/28/13 22:38, Jeff Clay wrote:
>
> Bart,
>
> OM will create a meetme meeting as configured in the
> realtime meetme database. Have you read this page
> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html
> ? You might also check out
> http://openmeetings.apache.org/red5sip-integration.html
> but I assume this is the one you're already referring to.
>
> Jeff Clay
> Network Administrator
> Infotech Enterprises America
> 870-215-5506
> Ext. 1506
>
> -----Original Message-----
> From: Bart Coninckx [mailto:bart.coninckx@telenet.be
> <ma...@telenet.be>]
> Sent: Monday, January 28, 2013 3:36 PM
> To: user@openmeetings.apache.org
> <ma...@openmeetings.apache.org>
> Subject: SIP connectivity
>
> Hi,
>
> I noticed some documentation on how to connect OM with a
> SIP proxy or server, more particularly with the MeetMe
> application in Asterisk.
>
> The exact goal or purpose is not mentionned however. Will
> OM callout to a MeetMe conference? Or is it the other way
> round?
>
>
> Cheers,
>
> Bc
>
> ________________________________
>
> DISCLAIMER:
>
> This email may contain confidential information and is
> intended only for the use of the specific individual(s) to
> which it is addressed. If you are not the intended
> recipient of this email, you are hereby notified that any
> unauthorized use, dissemination or copying of this email
> or the information contained in it or attached to it is
> strictly prohibited. If you received this message in
> error, please immediately notify the sender at Infotech
> and delete the original message.
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
> --
> WBR
> Maxim aka solomax
>
RE: SIP connectivity
Posted by Jeff Clay <Je...@infotech-enterprises.com>.
Bart,
If you look in the source directory of your asterisk tar file, under contrib/realtime/mysql you’ll find the .sql files required for all the realtime drivers. I never thought to use the ones with OM.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
Sent: Wednesday, January 30, 2013 3:19 PM
To: user@openmeetings.apache.org
Cc: Jeff Clay
Subject: Re: SIP connectivity
Well,
I might have found one difference though:
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure dictates how the table should look like. I obviously used the one in the openmeetings mysql database, but this one seems to miss the table "useragent". I discovered this because it showed up in the logfiles.
BC
On 01/29/13 14:41, Jeff Clay wrote:
Bart,
From an asterisk configuration standpoint there are very few differences between 1.8.x and 11.x. If memory serves, the only major changes that I ran into (in my production environment) was changes to SIP NAT values and the behavior of app_page() now uses confbridge instead of meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major overhauling. There were of course many other changes and bug fixes, you can skim through the change log for full details, but I think that was the jist of it.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
Sent: Tuesday, January 29, 2013 4:02 AM
To: Maxim Solodovnik
Cc: user
Subject: Re: SIP connectivity
I see - I'm willing to try the 11 version in the next fiew days if desired.
BC
On 01/29/13 10:57, Maxim Solodovnik wrote:
I test the integration using
Asterisk 1.8.13.1 (Ubuntu 12.10)
On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <ba...@telenet.be>> wrote:
That is amazing - I initially tried to do the same thing by using the new chan_motif driver in Asterisk 11 which connects to a XMPP server.
Are you guys using Asterisk 11? This version is the newest LTS version and has the best video capabilities.
Cheers,
BC
On 01/29/13 02:44, Maxim Solodovnik wrote:
red5sip will create special OM user in the room: "SIP Transport"
after that you can call to the OM room using SIP hard or soft phone.
We are currently testing it and trying to add video capabilities ...
On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <ba...@telenet.be>> wrote:
Hi Jeff,
In fact, I saw both pages, but none explain what they set up to do, just a bunch of command line instructions are given.
Your "OM will create a meetme meeting as configured in the realtime meetme database" actually says it all in one go :-)
cheers,
BC
On 01/28/13 22:38, Jeff Clay wrote:
Bart,
OM will create a meetme meeting as configured in the realtime meetme database. Have you read this page https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out http://openmeetings.apache.org/red5sip-integration.html but I assume this is the one you're already referring to.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
-----Original Message-----
From: Bart Coninckx [mailto:bart.coninckx@telenet.be<ma...@telenet.be>]
Sent: Monday, January 28, 2013 3:36 PM
To: user@openmeetings.apache.org<ma...@openmeetings.apache.org>
Subject: SIP connectivity
Hi,
I noticed some documentation on how to connect OM with a SIP proxy or server, more particularly with the MeetMe application in Asterisk.
The exact goal or purpose is not mentionned however. Will OM callout to a MeetMe conference? Or is it the other way round?
Cheers,
Bc
________________________________
DISCLAIMER:
This email may contain confidential information and is intended only for the use of the specific individual(s) to which it is addressed. If you are not the intended recipient of this email, you are hereby notified that any unauthorized use, dissemination or copying of this email or the information contained in it or attached to it is strictly prohibited. If you received this message in error, please immediately notify the sender at Infotech and delete the original message.
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: SIP connectivity
Posted by Bart Coninckx <ba...@telenet.be>.
Well,
I might have found one difference though:
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
dictates how the table should look like. I obviously used the one in the
openmeetings mysql database, but this one seems to miss the table
"useragent". I discovered this because it showed up in the logfiles.
BC
On 01/29/13 14:41, Jeff Clay wrote:
>
> Bart,
>
> From an asterisk configuration standpoint there are very few
> differences between 1.8.x and 11.x. If memory serves, the only major
> changes that I ran into (in my production environment) was changes to
> SIP NAT values and the behavior of app_page() now uses confbridge
> instead of meetme to mix the audio. Also, TCP, TLS and app_confbridge
> got a major overhauling. There were of course many other changes and
> bug fixes, you can skim through the change log for full details, but I
> think that was the jist of it.
>
> Jeff Clay
>
> Network Administrator
>
> Infotech Enterprises America
>
> 870-215-5506
>
> Ext. 1506
>
> *From:*Bart Coninckx [mailto:bart.coninckx@telenet.be]
> *Sent:* Tuesday, January 29, 2013 4:02 AM
> *To:* Maxim Solodovnik
> *Cc:* user
> *Subject:* Re: SIP connectivity
>
> I see - I'm willing to try the 11 version in the next fiew days if
> desired.
>
> BC
>
>
> On 01/29/13 10:57, Maxim Solodovnik wrote:
>
> I test the integration using
>
> Asterisk 1.8.13.1 (Ubuntu 12.10)
>
> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx
> <bart.coninckx@telenet.be <ma...@telenet.be>> wrote:
>
> That is amazing - I initially tried to do the same thing by using
> the new chan_motif driver in Asterisk 11 which connects to a XMPP
> server.
>
> Are you guys using Asterisk 11? This version is the newest LTS
> version and has the best video capabilities.
>
> Cheers,
>
> BC
>
>
> On 01/29/13 02:44, Maxim Solodovnik wrote:
>
> red5sip will create special OM user in the room: "SIP Transport"
>
> after that you can call to the OM room using SIP hard or soft
> phone.
>
> We are currently testing it and trying to add video
> capabilities ...
>
> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx
> <bart.coninckx@telenet.be <ma...@telenet.be>>
> wrote:
>
> Hi Jeff,
>
> In fact, I saw both pages, but none explain what they set up
> to do, just a bunch of command line instructions are given.
> Your "OM will create a meetme meeting as configured in the
> realtime meetme database" actually says it all in one go :-)
>
> cheers,
>
> BC
>
>
>
>
> On 01/28/13 22:38, Jeff Clay wrote:
>
> Bart,
>
> OM will create a meetme meeting as configured in the realtime
> meetme database. Have you read this page
> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html
> ? You might also check out
> http://openmeetings.apache.org/red5sip-integration.html but I
> assume this is the one you're already referring to.
>
> Jeff Clay
> Network Administrator
> Infotech Enterprises America
> 870-215-5506
> Ext. 1506
>
> -----Original Message-----
> From: Bart Coninckx [mailto:bart.coninckx@telenet.be
> <ma...@telenet.be>]
> Sent: Monday, January 28, 2013 3:36 PM
> To: user@openmeetings.apache.org
> <ma...@openmeetings.apache.org>
> Subject: SIP connectivity
>
> Hi,
>
> I noticed some documentation on how to connect OM with a SIP
> proxy or server, more particularly with the MeetMe application
> in Asterisk.
>
> The exact goal or purpose is not mentionned however. Will OM
> callout to a MeetMe conference? Or is it the other way round?
>
>
> Cheers,
>
> Bc
>
> ________________________________
>
> DISCLAIMER:
>
> This email may contain confidential information and is
> intended only for the use of the specific individual(s) to
> which it is addressed. If you are not the intended recipient
> of this email, you are hereby notified that any unauthorized
> use, dissemination or copying of this email or the information
> contained in it or attached to it is strictly prohibited. If
> you received this message in error, please immediately notify
> the sender at Infotech and delete the original message.
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
> --
> WBR
> Maxim aka solomax
>
RE: SIP connectivity
Posted by Jeff Clay <Je...@infotech-enterprises.com>.
Bart,
From an asterisk configuration standpoint there are very few differences between 1.8.x and 11.x. If memory serves, the only major changes that I ran into (in my production environment) was changes to SIP NAT values and the behavior of app_page() now uses confbridge instead of meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major overhauling. There were of course many other changes and bug fixes, you can skim through the change log for full details, but I think that was the jist of it.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
Sent: Tuesday, January 29, 2013 4:02 AM
To: Maxim Solodovnik
Cc: user
Subject: Re: SIP connectivity
I see - I'm willing to try the 11 version in the next fiew days if desired.
BC
On 01/29/13 10:57, Maxim Solodovnik wrote:
I test the integration using
Asterisk 1.8.13.1 (Ubuntu 12.10)
On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <ba...@telenet.be>> wrote:
That is amazing - I initially tried to do the same thing by using the new chan_motif driver in Asterisk 11 which connects to a XMPP server.
Are you guys using Asterisk 11? This version is the newest LTS version and has the best video capabilities.
Cheers,
BC
On 01/29/13 02:44, Maxim Solodovnik wrote:
red5sip will create special OM user in the room: "SIP Transport"
after that you can call to the OM room using SIP hard or soft phone.
We are currently testing it and trying to add video capabilities ...
On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <ba...@telenet.be>> wrote:
Hi Jeff,
In fact, I saw both pages, but none explain what they set up to do, just a bunch of command line instructions are given.
Your "OM will create a meetme meeting as configured in the realtime meetme database" actually says it all in one go :-)
cheers,
BC
On 01/28/13 22:38, Jeff Clay wrote:
Bart,
OM will create a meetme meeting as configured in the realtime meetme database. Have you read this page https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out http://openmeetings.apache.org/red5sip-integration.html but I assume this is the one you're already referring to.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
-----Original Message-----
From: Bart Coninckx [mailto:bart.coninckx@telenet.be<ma...@telenet.be>]
Sent: Monday, January 28, 2013 3:36 PM
To: user@openmeetings.apache.org<ma...@openmeetings.apache.org>
Subject: SIP connectivity
Hi,
I noticed some documentation on how to connect OM with a SIP proxy or server, more particularly with the MeetMe application in Asterisk.
The exact goal or purpose is not mentionned however. Will OM callout to a MeetMe conference? Or is it the other way round?
Cheers,
Bc
________________________________
DISCLAIMER:
This email may contain confidential information and is intended only for the use of the specific individual(s) to which it is addressed. If you are not the intended recipient of this email, you are hereby notified that any unauthorized use, dissemination or copying of this email or the information contained in it or attached to it is strictly prohibited. If you received this message in error, please immediately notify the sender at Infotech and delete the original message.
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
Re: SIP connectivity
Posted by Bart Coninckx <ba...@telenet.be>.
I see - I'm willing to try the 11 version in the next fiew days if desired.
BC
On 01/29/13 10:57, Maxim Solodovnik wrote:
> I test the integration using
> Asterisk 1.8.13.1 (Ubuntu 12.10)
>
>
> On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx
> <bart.coninckx@telenet.be <ma...@telenet.be>> wrote:
>
> That is amazing - I initially tried to do the same thing by using
> the new chan_motif driver in Asterisk 11 which connects to a XMPP
> server.
>
> Are you guys using Asterisk 11? This version is the newest LTS
> version and has the best video capabilities.
>
> Cheers,
>
> BC
>
>
> On 01/29/13 02:44, Maxim Solodovnik wrote:
>> red5sip will create special OM user in the room: "SIP Transport"
>> after that you can call to the OM room using SIP hard or soft phone.
>>
>> We are currently testing it and trying to add video capabilities ...
>>
>>
>> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx
>> <bart.coninckx@telenet.be <ma...@telenet.be>> wrote:
>>
>> Hi Jeff,
>>
>> In fact, I saw both pages, but none explain what they set up
>> to do, just a bunch of command line instructions are given.
>> Your "OM will create a meetme meeting as configured in the
>> realtime meetme database" actually says it all in one go :-)
>>
>> cheers,
>>
>> BC
>>
>>
>>
>> On 01/28/13 22:38, Jeff Clay wrote:
>>
>> Bart,
>>
>> OM will create a meetme meeting as configured in the
>> realtime meetme database. Have you read this page
>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html
>> ? You might also check out
>> http://openmeetings.apache.org/red5sip-integration.html
>> but I assume this is the one you're already referring to.
>>
>> Jeff Clay
>> Network Administrator
>> Infotech Enterprises America
>> 870-215-5506
>> Ext. 1506
>>
>> -----Original Message-----
>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be
>> <ma...@telenet.be>]
>> Sent: Monday, January 28, 2013 3:36 PM
>> To: user@openmeetings.apache.org
>> <ma...@openmeetings.apache.org>
>> Subject: SIP connectivity
>>
>> Hi,
>>
>> I noticed some documentation on how to connect OM with a
>> SIP proxy or server, more particularly with the MeetMe
>> application in Asterisk.
>>
>> The exact goal or purpose is not mentionned however. Will
>> OM callout to a MeetMe conference? Or is it the other way
>> round?
>>
>>
>> Cheers,
>>
>> Bc
>>
>> ________________________________
>>
>> DISCLAIMER:
>>
>> This email may contain confidential information and is
>> intended only for the use of the specific individual(s)
>> to which it is addressed. If you are not the intended
>> recipient of this email, you are hereby notified that any
>> unauthorized use, dissemination or copying of this email
>> or the information contained in it or attached to it is
>> strictly prohibited. If you received this message in
>> error, please immediately notify the sender at Infotech
>> and delete the original message.
>>
>>
>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>
>
>
>
> --
> WBR
> Maxim aka solomax
Re: SIP connectivity
Posted by Maxim Solodovnik <so...@gmail.com>.
I test the integration using
Asterisk 1.8.13.1 (Ubuntu 12.10)
On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx <ba...@telenet.be>wrote:
> That is amazing - I initially tried to do the same thing by using the
> new chan_motif driver in Asterisk 11 which connects to a XMPP server.
>
> Are you guys using Asterisk 11? This version is the newest LTS version and
> has the best video capabilities.
>
> Cheers,
>
> BC
>
>
> On 01/29/13 02:44, Maxim Solodovnik wrote:
>
> red5sip will create special OM user in the room: "SIP Transport"
> after that you can call to the OM room using SIP hard or soft phone.
>
> We are currently testing it and trying to add video capabilities ...
>
>
> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <ba...@telenet.be>wrote:
>
>> Hi Jeff,
>>
>> In fact, I saw both pages, but none explain what they set up to do, just
>> a bunch of command line instructions are given.
>> Your "OM will create a meetme meeting as configured in the realtime
>> meetme database" actually says it all in one go :-)
>>
>> cheers,
>>
>> BC
>>
>>
>>
>> On 01/28/13 22:38, Jeff Clay wrote:
>>
>>> Bart,
>>>
>>> OM will create a meetme meeting as configured in the realtime meetme
>>> database. Have you read this page
>>> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out
>>> http://openmeetings.apache.org/red5sip-integration.html but I assume
>>> this is the one you're already referring to.
>>>
>>> Jeff Clay
>>> Network Administrator
>>> Infotech Enterprises America
>>> 870-215-5506
>>> Ext. 1506
>>>
>>> -----Original Message-----
>>> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
>>> Sent: Monday, January 28, 2013 3:36 PM
>>> To: user@openmeetings.apache.org
>>> Subject: SIP connectivity
>>>
>>> Hi,
>>>
>>> I noticed some documentation on how to connect OM with a SIP proxy or
>>> server, more particularly with the MeetMe application in Asterisk.
>>>
>>> The exact goal or purpose is not mentionned however. Will OM callout to
>>> a MeetMe conference? Or is it the other way round?
>>>
>>>
>>> Cheers,
>>>
>>> Bc
>>>
>>> ________________________________
>>>
>>> DISCLAIMER:
>>>
>>> This email may contain confidential information and is intended only for
>>> the use of the specific individual(s) to which it is addressed. If you are
>>> not the intended recipient of this email, you are hereby notified that any
>>> unauthorized use, dissemination or copying of this email or the information
>>> contained in it or attached to it is strictly prohibited. If you received
>>> this message in error, please immediately notify the sender at Infotech and
>>> delete the original message.
>>>
>>
>>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
--
WBR
Maxim aka solomax
Re: SIP connectivity
Posted by Bart Coninckx <ba...@telenet.be>.
That is amazing - I initially tried to do the same thing by using the
new chan_motif driver in Asterisk 11 which connects to a XMPP server.
Are you guys using Asterisk 11? This version is the newest LTS version
and has the best video capabilities.
Cheers,
BC
On 01/29/13 02:44, Maxim Solodovnik wrote:
> red5sip will create special OM user in the room: "SIP Transport"
> after that you can call to the OM room using SIP hard or soft phone.
>
> We are currently testing it and trying to add video capabilities ...
>
>
> On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx
> <bart.coninckx@telenet.be <ma...@telenet.be>> wrote:
>
> Hi Jeff,
>
> In fact, I saw both pages, but none explain what they set up to
> do, just a bunch of command line instructions are given.
> Your "OM will create a meetme meeting as configured in the
> realtime meetme database" actually says it all in one go :-)
>
> cheers,
>
> BC
>
>
>
> On 01/28/13 22:38, Jeff Clay wrote:
>
> Bart,
>
> OM will create a meetme meeting as configured in the realtime
> meetme database. Have you read this page
> https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html
> ? You might also check out
> http://openmeetings.apache.org/red5sip-integration.html but I
> assume this is the one you're already referring to.
>
> Jeff Clay
> Network Administrator
> Infotech Enterprises America
> 870-215-5506
> Ext. 1506
>
> -----Original Message-----
> From: Bart Coninckx [mailto:bart.coninckx@telenet.be
> <ma...@telenet.be>]
> Sent: Monday, January 28, 2013 3:36 PM
> To: user@openmeetings.apache.org
> <ma...@openmeetings.apache.org>
> Subject: SIP connectivity
>
> Hi,
>
> I noticed some documentation on how to connect OM with a SIP
> proxy or server, more particularly with the MeetMe application
> in Asterisk.
>
> The exact goal or purpose is not mentionned however. Will OM
> callout to a MeetMe conference? Or is it the other way round?
>
>
> Cheers,
>
> Bc
>
> ________________________________
>
> DISCLAIMER:
>
> This email may contain confidential information and is
> intended only for the use of the specific individual(s) to
> which it is addressed. If you are not the intended recipient
> of this email, you are hereby notified that any unauthorized
> use, dissemination or copying of this email or the information
> contained in it or attached to it is strictly prohibited. If
> you received this message in error, please immediately notify
> the sender at Infotech and delete the original message.
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
Re: SIP connectivity
Posted by Maxim Solodovnik <so...@gmail.com>.
red5sip will create special OM user in the room: "SIP Transport"
after that you can call to the OM room using SIP hard or soft phone.
We are currently testing it and trying to add video capabilities ...
On Tue, Jan 29, 2013 at 4:44 AM, Bart Coninckx <ba...@telenet.be>wrote:
> Hi Jeff,
>
> In fact, I saw both pages, but none explain what they set up to do, just a
> bunch of command line instructions are given.
> Your "OM will create a meetme meeting as configured in the realtime meetme
> database" actually says it all in one go :-)
>
> cheers,
>
> BC
>
>
>
> On 01/28/13 22:38, Jeff Clay wrote:
>
>> Bart,
>>
>> OM will create a meetme meeting as configured in the realtime meetme
>> database. Have you read this page https://cwiki.apache.org/**
>> OPENMEETINGS/openmeetings-**asterisk-integration.html<https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html> ? You might also check out
>> http://openmeetings.apache.**org/red5sip-integration.html<http://openmeetings.apache.org/red5sip-integration.html>but I assume this is the one you're already referring to.
>>
>> Jeff Clay
>> Network Administrator
>> Infotech Enterprises America
>> 870-215-5506
>> Ext. 1506
>>
>> -----Original Message-----
>> From: Bart Coninckx [mailto:bart.coninckx@telenet.**be<ba...@telenet.be>
>> ]
>> Sent: Monday, January 28, 2013 3:36 PM
>> To: user@openmeetings.apache.org
>> Subject: SIP connectivity
>>
>> Hi,
>>
>> I noticed some documentation on how to connect OM with a SIP proxy or
>> server, more particularly with the MeetMe application in Asterisk.
>>
>> The exact goal or purpose is not mentionned however. Will OM callout to a
>> MeetMe conference? Or is it the other way round?
>>
>>
>> Cheers,
>>
>> Bc
>>
>> ______________________________**__
>>
>> DISCLAIMER:
>>
>> This email may contain confidential information and is intended only for
>> the use of the specific individual(s) to which it is addressed. If you are
>> not the intended recipient of this email, you are hereby notified that any
>> unauthorized use, dissemination or copying of this email or the information
>> contained in it or attached to it is strictly prohibited. If you received
>> this message in error, please immediately notify the sender at Infotech and
>> delete the original message.
>>
>
>
--
WBR
Maxim aka solomax
Re: SIP connectivity
Posted by Bart Coninckx <ba...@telenet.be>.
Hi Jeff,
In fact, I saw both pages, but none explain what they set up to do, just
a bunch of command line instructions are given.
Your "OM will create a meetme meeting as configured in the realtime
meetme database" actually says it all in one go :-)
cheers,
BC
On 01/28/13 22:38, Jeff Clay wrote:
> Bart,
>
> OM will create a meetme meeting as configured in the realtime meetme database. Have you read this page https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out http://openmeetings.apache.org/red5sip-integration.html but I assume this is the one you're already referring to.
>
> Jeff Clay
> Network Administrator
> Infotech Enterprises America
> 870-215-5506
> Ext. 1506
>
> -----Original Message-----
> From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
> Sent: Monday, January 28, 2013 3:36 PM
> To: user@openmeetings.apache.org
> Subject: SIP connectivity
>
> Hi,
>
> I noticed some documentation on how to connect OM with a SIP proxy or server, more particularly with the MeetMe application in Asterisk.
>
> The exact goal or purpose is not mentionned however. Will OM callout to a MeetMe conference? Or is it the other way round?
>
>
> Cheers,
>
> Bc
>
> ________________________________
>
> DISCLAIMER:
>
> This email may contain confidential information and is intended only for the use of the specific individual(s) to which it is addressed. If you are not the intended recipient of this email, you are hereby notified that any unauthorized use, dissemination or copying of this email or the information contained in it or attached to it is strictly prohibited. If you received this message in error, please immediately notify the sender at Infotech and delete the original message.
RE: SIP connectivity
Posted by Jeff Clay <Je...@infotech-enterprises.com>.
Bart,
OM will create a meetme meeting as configured in the realtime meetme database. Have you read this page https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html ? You might also check out http://openmeetings.apache.org/red5sip-integration.html but I assume this is the one you're already referring to.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
-----Original Message-----
From: Bart Coninckx [mailto:bart.coninckx@telenet.be]
Sent: Monday, January 28, 2013 3:36 PM
To: user@openmeetings.apache.org
Subject: SIP connectivity
Hi,
I noticed some documentation on how to connect OM with a SIP proxy or server, more particularly with the MeetMe application in Asterisk.
The exact goal or purpose is not mentionned however. Will OM callout to a MeetMe conference? Or is it the other way round?
Cheers,
Bc
________________________________
DISCLAIMER:
This email may contain confidential information and is intended only for the use of the specific individual(s) to which it is addressed. If you are not the intended recipient of this email, you are hereby notified that any unauthorized use, dissemination or copying of this email or the information contained in it or attached to it is strictly prohibited. If you received this message in error, please immediately notify the sender at Infotech and delete the original message.