[OpenSIPS-Users] 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
> ------------------------------------------------------------------------
>
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