[asterisk-users] Incoming call

[David fire]

if is a codec problem start putting all the systems in the same codec and
disallow all other.
put for example all in alaw and disalaw all other includeing ulaw.

you can make calls from asterisk to the sip extencion registered in opensip?
check that you can start the call from a soft phone or you can use orginate
and send the call to open sip one side and music on hold in the other side.

David

2009/2/24 michel freiha <michofr at gmail.com>

> Dera All,
>
> I have the following scenario,
> A customer dial a DID number...The call is routed to a PSTN GW that send
> the call to asterisk...
> On asterisk I created an AGI Script that send the call to an extension
> registered on OpenSIPS server...
> The extension is ringing successfully, but as soon as I accept the call on
> OpenSIPS side the call is hangd up...
> I checked rhe SIP debug and it seems that I have a Codec issue as you can
> see on http://pastebin.com/m767a2172
>
> Need some help please
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
(")_(")signature to help him gain world domination.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090225/cded9a68/attachment.htm