[OpenSIPS-Users] OpenSIPS with Asterisk Backend

Bogdan-Andrei Iancu bogdan at voice-system.ro
Tue Apr 20 11:25:25 CEST 2010


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.
>
>   
>
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>
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-- 
Bogdan-Andrei Iancu
www.voice-system.ro




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