[OpenSIPS-Users] 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 <bogdan at voice-system.ro
> 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
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.opensips.org/pipermail/users/attachments/20090225/b781eaab/attachment.htm