asterisk users - Sep 2009 - Send 200 OK with SDP instead of 183 with SDP when ringing starts

Hello, all. I have an asterisk 2.3.2 and a Sangoma interface through 
which I connected an external PSTN line. I use it as carrier for VoIP 
calls. I can make successfully calls, but there's one problem, I receive 
200 OK with SDP with delay (sometimes more than 30 seconds).
So when I make a call through asterisk I receive intially:
- 100 Trying
- 183 Session Progress, with SDP
when the called number respond, I start receiving RTP with voice, also 
the called receives voice from me, but only after a while asterisk sends 
200 OK with SDP.

I'm not sure if the problem is from asterisk or from the telephony 
provider (I think the provider). Is there a posibility to replace 183 
with 200 OK ? I mean from the moment when ringing starts to receive 200 
OK with SDP instead of 183 ?

Thank you,
Marius

4 sep 2009 kl. 13.40 skrev Marius Ciorecan:
> Hello, all. I have an asterisk 2.3.2 and a Sangoma interface through
> which I connected an external PSTN line. I use it as carrier for VoIP
> calls. I can make successfully calls, but there's one problem, I  
> receive
> 200 OK with SDP with delay (sometimes more than 30 seconds).
> So when I make a call through asterisk I receive intially:
> - 100 Trying
> - 183 Session Progress, with SDP
> when the called number respond, I start receiving RTP with voice, also
> the called receives voice from me, but only after a while asterisk  
> sends
> 200 OK with SDP.
>
> I'm not sure if the problem is from asterisk or from the telephony
> provider (I think the provider). Is there a posibility to replace 183
> with 200 OK ? I mean from the moment when ringing starts to receive  
> 200
> OK with SDP instead of 183 ?
>
You can answer() at any point in the dialplan - and that will generate  
a 200 OK.

Like

exten => marius,1,answer()
exten => marius,n,dial(sip/mariusphone)

This will generate an immediate 200 ok, regardless if mariusphone is  
busy or gone from the network.
It's propably not what you want.

Asterisk sends 200 OK on the incoming call as soon as we get a  
connection reply, a 200 OK or something similar in other protocols on  
the outbound call. For some reason, this happens very late for you and  
causes your problem. Could be some issue with the service provider,  
your ISDN connection or -even worse - your IAX2 trunk... (could not  
resist)

Please start with debugging that and solving the real issue, instead  
of trying to change the functionality in Asterisk :-)

Regards,
/O



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