[asterisk-ss7] Asterisk 1.8.4.2 + LibSS7 1.0.2 : Early Media Problem

[florian at gruendler.net]

Hassan, how did the story go on? 

 

Did the Progress() enable the early media transmission in chan_sip to the calling user agent as desired? 

 

If not, what did you find regarding LibSS7 and its influence on early media bridging to chan_sip? I believe the control stack handling media (which could well be chan_dahdi) does send received media along to any other involved channel constantly and the receiver chan_sip just discards it (while busy sending media from internal tone generator = fake ringing, treatment tone or comfort noise) unless early media capability is explicitly requested instead of those alternatives, latest when the B-channel enters a connected state when the signaling stack updates the channel variable to twoway audio after receiving the SS7 ANM message from the far end. 

 

Kind regards, Florian

 

Von: asterisk-ss7-bounces at lists.digium.com [mailto:asterisk-ss7-bounces at lists.digium.com] Im Auftrag von Nyamul Hassan
Gesendet: Mittwoch, 22. Juni 2011 13:20
An: asterisk-ss7 at lists.digium.com
Betreff: Re: [asterisk-ss7] Asterisk 1.8.4.2 + LibSS7 1.0.2 : Early Media Problem

 

Thank you Florain, for your reply.  My answers are inline.

 

On Wed, Jun 22, 2011 at 17:08, <florian at gruendler.net> 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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