Thank you Florain, for your reply. My answers are inline.
On Wed, Jun 22, 2011 at 17:08, <[email protected]> wrote:
> Hassan, I think I have a contribution to your problem:****
>
> ** **
>
> As of Release 1.6, you need to make an explicit ****
>
> ** **
>
> exten => 1234,n,Progress()
>
Oh, did not know that. So, I need to put this at the top of the dialplan,
before I put the "dialplan", right?
My current dialplan is:
[ss7out]
exten => _919.,1,Dial(DAHDI/g1/${EXTEN:2})
exten => _919.,n,Hangup()
So, change this to:
[ss7out]
exten => _919.,1,Progress()
exten => _919.,n,Dial(DAHDI/g1/${EXTEN:2})
exten => _919.,n,Hangup()
**
>
> else Asterisk will not proceed using SIP/183 with SDP. Can you show the
> signaling data of the SIP session? It would help to understand what call
> vector you are having issues with since the routing (aka dialplan) has
> different requirements on an incoming (SS7->SIP), respectively outgoing call
> (SIP->SS7).****
>
> **
>
Can you tell me what data you want? Do I need to do a SIP Trace? Or SS7
Trace? I've never done the trace on LibSS7 earlier. Which command do I
need to run?
Regards
HASSAN
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