[OpenSIPS-Users] [asterisk-users] OpenSIPS with Asterisk Backend

Robert Borz robert.borz at web.de
Tue Jul 6 10:24:44 CEST 2010 

Solved this issue by setting "min_se" parameter for the SST module in OpenSIPS to 180 and setting "session-minse=180" in Asterisk sip.conf in the general section.

I just should have read the error message more carefully as "422 Session Interval Too Small." says it all... :-P

Nevertheless, thanks a lot. :-)


-----Ursprüngliche Nachricht-----
Von: Robert Borz <robert.borz at web.de>
Gesendet: 05.07.2010 17:10:08
An: OpenSIPS users mailling list <users at lists.opensips.org>,Asterisk Users
	Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
Betreff: Re: [OpenSIPS-Users] [asterisk-users] OpenSIPS with Asterisk Backend

>Hi Bogdan,
>
>thank you for your response. In the meantime I set the dialog timeout to three hours, this helps a bit. ;-)
>
>I wasn't able to catch a stuck call to get the state. Maybe in the near future...
>
>To get rid of stuck calls as fast as possible I want to use SIP Session Timers. For this I upgraded the Asterisk backend to version 1.6 which supports this and loaded and configured the SST module in OpenSIPS. With almost every user agent everything seems to work as expected.
>
>But I have a Problem with "AVM FRITZ!Box Fon WLAN 7050 (UI) 14.04.33 (May 10 2007)". When it receives an INVITE with "Session-Expires: 90" it just anwers with "422 Session Interval Too Small". At this point the whole thing doesn't get any further. I can't imagine that this behaviour is in accordance to the standard as this is the only UA I have problems with. Whatever, the UA hasn't any problems in placing outgoing calls... really strange.
>
>There are two options for me:
>a) Find a workaround: My idea is now not to enable SST on calls to this UA. I can't use the $ua scripting variable, as it contains "Asterisk PBX", which is absolutely right here... :-/
>
>b) Tell the customer to get a new UA as it is already EOL. ;-)
>
>What do you think?
>
>Regards,
>Robert.
>
>Here's the SIP-Trace (without SDP):
>OpenSIPS -> Customer:
>INVITE sip:10000 at XXX.XXX.XXX.124;uniq=0111AA28F74AE042C3CD6EB4C39F6 SIP/2.0.
>Record-Route: .
>Via: SIP/2.0/UDP XXX.XXX.XXX.8;branch=z9hG4bK9007.c4c6808.0.
>Via: SIP/2.0/UDP XXX.XXX.XXX.12:5060;received=XXX.XXX.XXX.12;branch=z9hG4bK144bb9fe;rport=5060.
>Max-Forwards: 69.
>From: ;tag=as152f5077.
>To: .
>Contact: .
>Call-ID: 24029a5240153c151015784f5736aef0 at XXX.XXX.XXX.12.
>CSeq: 102 INVITE.
>User-Agent: Asterisk PBX 1.6.2.6-1.
>Date: Mon, 05 Jul 2010 14:56:42 GMT.
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
>Supported: replaces, timer.
>Content-Type: application/sdp.
>Content-Length: 265.
>Session-Expires: 90.
>.
>
>
>Customer -> OpenSIPS:
>SIP/2.0 422 Session Interval Too Small.
>Via: SIP/2.0/UDP XXX.XXX.XXX.8;branch=z9hG4bK9007.c4c6808.0.
>Via: SIP/2.0/UDP XXX.XXX.XXX.12:5060;received=XXX.XXX.XXX.12;branch=z9hG4bK144bb9fe;rport=5060.
>From: ;tag=as152f5077.
>To: ;tag=D748EB0E786BFD44.
>Call-ID: 24029a5240153c151015784f5736aef0 at XXX.XXX.XXX.12.
>CSeq: 102 INVITE.
>User-Agent: AVM FRITZ!Box Fon WLAN 7050 (UI) 14.04.33 (May 10 2007).
>Content-Length: 0.
>.
>
>
>OpenSIPS -> Customer:
>ACK sip:10000 at XXX.XXX.XXX.124;uniq=0111AA28F74AE042C3CD6EB4C39F6 SIP/2.0.
>Via: SIP/2.0/UDP XXX.XXX.XXX.8;branch=z9hG4bK9007.c4c6808.0.
>From: ;tag=as152f5077.
>Call-ID: 24029a5240153c151015784f5736aef0 at XXX.XXX.XXX.12.
>To: ;tag=D748EB0E786BFD44.
>CSeq: 102 ACK.
>Max-Forwards: 70.
>User-Agent: OpenSIPS (1.5.1-notls (x86_64/linux)).
>Content-Length: 0.
>.
>
>
>-----Ursprüngliche Nachricht-----
>Von: Bogdan-Andrei Iancu 
>Gesendet: 20.04.2010 11:25:25
>An: OpenSIPS users mailling list 
>Betreff: Re: [asterisk-users] [OpenSIPS-Users] OpenSIPS with Asterisk Backend
>
>>Hi Robert,
>>
>>
>>The opensips dialog module mainly does dialog monitoring and has limited 
>>capability when comes to checking dialog health (like it the call is not 
>>zombie and it is really ongoing). The dialog module can just expire too 
>>long calls (using a timeout for call duration).
>>
>>First of all, dealing with the cause : what is the state of that zombie 
>>calls (see the dlg_list output) - maybe it is a bogus setup call or a 
>>call without BYE.
>>
>>Now about how do deal with these calls: first reduce the timeout to 2-3 
>>hours, so even if you have a bogus call, it will be automatically 
>>removed. There are other options, but it highly depends on the state of 
>>the zombie call.
>>
>>A basic idea is also to have an external script (simple bash) to 
>>correlate the dialogs from Asterisk with the ones from OpenSIPS - like 
>>OpenSIPS claim to have an ongoing call C1 via Asterisk A1, but A1 does 
>>not report it -> use the MI of OpenSIPS (dlg_end_dlg command) to 
>>terminate the bogus call on OpenSIPS.
>>
>>BTW, is any kind of call keepalive ? like SST with re-INVITEs ? does 
>>Asterisk do media timeout  ?
>>
>>Regards,
>>Bogdan
>>
>>Robert Borz wrote:
>>>
>>>  Hi,
>>>
>>>  
>>>
>>> sorry for cross-posting on both mailing lists, but I think a setup of 
>>> Asterisk with OpenSIPS as frontend isn't unusual. So maybe both 
>>> parties would be interested in this.
>>>
>>>  
>>>
>>> I'm using Asterisk (v1.4.21) to connect my OpenSIPS (v1.5.1) server to 
>>> the PSTN (Asterisk connects to a local SIP provider doing the PSTN 
>>> termination) so the Asterisk just acts as an PSTN gateway here. For 
>>> doing some call control stuff (channel limitation) I'm using the 
>>> dialog module on OpenSIPS.
>>>
>>>  
>>>
>>> Generally everything works well with about 250 users at the moment. 
>>> But sometimes there are stuck dialogs on the OpenSIPS server (seen by 
>>> #opensipsctl fifo dlg_list). At the same time in Asterisk messages 
>>> there is this:
>>>
>>> [Apr 19 07:21:50] WARNING[13498] chan_sip.c: Maximum retries exceeded 
>>> on transmission 5CE33D1A20E54183 at XXX.XXX.XXX.XXX for seqno 7912 
>>> (Critical Response)
>>>
>>>  
>>>
>>> As I'm doing channel limitation to a single channel by using the 
>>> dialog module a stuck dialog leads to the user not being able to do 
>>> any further calls until the dialog is destroyed by dialog timeout.
>>>
>>>  
>>>
>>> Any ideas how to solve this issue?
>>>
>>>  
>>>
>>>  
>>>
>>> Regards, Robert.
>>>
>>>   
>>>
>>> NEU: WEB.DE DSL für 19,99 EUR/mtl. und ohne Mindest-Laufzeit!   
>>> http://produkte.web.de/go/02/
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>   
>>
>>
>>-- 
>>Bogdan-Andrei Iancu
>>www.voice-system.ro
>>
>>
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