Incoming call

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Incoming call

michel freiha
Hi all,

I have an opensips server installed on my network and used for registration on local call...When a customer dial a PSTN call, it'll be routed to an asterisk server that route it to a PSTN gateway...Tis scenario is working smoothly...

The problem occurs when receiving a call from asterisk...The call is sent from asterisk to an online endpoint on Opensips...The extension is ringing but as soon as I open accept the call on the extension registered on opensips, the call is hanged up direcly...

I checked logs and found out that asterisk send INVITE packets to opensips and OpenSip replies by <Call/Transaction Does Not Exist>

Please chek the opensips log at http://pastebin.com/d27ae4ee9

Thanks for help

Regards

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Re: Incoming call

Bogdan-Andrei Iancu
Hi Michel,

can you post also the SIP trace of the call ? So far, from the log I can
see the call is established (I see 200OK and ACK) and then I see a
BYE... but the trace will be more helpful.

Regards,
Bogdan

michel freiha wrote:

> Hi all,
>
> I have an opensips server installed on my network and used for
> registration on local call...When a customer dial a PSTN call, it'll
> be routed to an asterisk server that route it to a PSTN gateway...Tis
> scenario is working smoothly...
>
> The problem occurs when receiving a call from asterisk...The call is
> sent from asterisk to an online endpoint on Opensips...The extension
> is ringing but as soon as I open accept the call on the extension
> registered on opensips, the call is hanged up direcly...
>
> I checked logs and found out that asterisk send INVITE packets to
> opensips and OpenSip replies by <Call/Transaction Does Not Exist>
>
> Please chek the opensips log at http://pastebin.com/d27ae4ee9
>
> Thanks for help
>
> Regards
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>  


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Re: Incoming call

michel freiha
Dear Bogdan,

Thanks for help...I solved it...It was an asterisk issue

Regards

On Wed, Feb 25, 2009 at 4:00 PM, Bogdan-Andrei Iancu <[hidden email]> wrote:
Hi Michel,

can you post also the SIP trace of the call ? So far, from the log I can see the call is established (I see 200OK and ACK) and then I see a BYE... but the trace will be more helpful.

Regards,
Bogdan

michel freiha wrote:
Hi all,

I have an opensips server installed on my network and used for registration on local call...When a customer dial a PSTN call, it'll be routed to an asterisk server that route it to a PSTN gateway...Tis scenario is working smoothly...

The problem occurs when receiving a call from asterisk...The call is sent from asterisk to an online endpoint on Opensips...The extension is ringing but as soon as I open accept the call on the extension registered on opensips, the call is hanged up direcly...

I checked logs and found out that asterisk send INVITE packets to opensips and OpenSip replies by <Call/Transaction Does Not Exist>

Please chek the opensips log at http://pastebin.com/d27ae4ee9

Thanks for help

Regards
------------------------------------------------------------------------

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Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
 



_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users