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 <[email protected]> Gesendet: 05.07.2010 17:10:08 An: OpenSIPS users mailling list <[email protected]>,Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> 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:[email protected];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: [email protected]. >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: [email protected]. >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:[email protected];uniq=0111AA28F74AE042C3CD6EB4C39F6 SIP/2.0. >Via: SIP/2.0/UDP XXX.XXX.XXX.8;branch=z9hG4bK9007.c4c6808.0. >From: ;tag=as152f5077. >Call-ID: [email protected]. >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 [email protected] 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 >>> [email protected] >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >>-- >>Bogdan-Andrei Iancu >>www.voice-system.ro >> >> >>-- >>_____________________________________________________________________ >>-- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>New to Asterisk? 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