AW: SIP connectivity

2013-02-13 Thread Naderi, Sascha 
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
 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.c...@infotech-enterprises.com]
Gesendet: Mittwoch, 13. Februar 2013 21:02
Bis: user@openmeetings.apache.org
Cc: Maxim Solodovnik [solomax...@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:snad...@datus.com]
Sent: Wednesday, February 13, 2013 2:00 PM
To: user@openmeetings.apache.org
Cc: Maxim Solodovnik [solomax...@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 [solomax...@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" 
mailto:bart.conin...@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 
mailto:solomax...@gmail.com>> wrote:

OK will add it and notify you
On Jan 31, 2013 5:05 PM, "Bart Coninckx" 
mailto:bart.conin...@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 
mailto:bart.conin...@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.conin...@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

AW: SIP connectivity

2013-02-13 Thread Naderi, Sascha 
Dear Maxim,





OK, thanks a lot. I will check it out and leave feedback.





Regards

Sascha



Von: Maxim Solodovnik [solomax...@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 
mailto:snad...@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 [solomax...@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" 
mailto:bart.conin...@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 
mailto:solomax...@gmail.com>> wrote:

OK will add it and notify you

On Jan 31, 2013 5:05 PM, "Bart Coninckx" 
mailto:bart.conin...@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 
mailto:bart.conin...@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.conin...@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.conin...@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 Coninck