Re: [asterisk-users] Send 200 OK with SDP instead of 183 with SDP when ringing starts

2009-09-07 Thread Marius Ciorecan
You are right, I think the provider has some problems, and this should 
be fixed.
But is also good to know that I can do this workaround for the worse 
case scenario.

thank you !

Olle E. Johansson wrote:
 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



 ---
 o...@edvina.net - http://edvina.net
 Open Unified Communication - building platforms with SIP and XMPP
  From PBX to large scale implementations for carriers. Contact us today!




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[asterisk-users] Send 200 OK with SDP instead of 183 with SDP when ringing starts

2009-09-04 Thread 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 ?

Thank you,
Marius
 

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Re: [asterisk-users] Send 200 OK with SDP instead of 183 with SDP when ringing starts

2009-09-04 Thread Olle E. Johansson

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



---
o...@edvina.net - http://edvina.net
Open Unified Communication - building platforms with SIP and XMPP
 From PBX to large scale implementations for carriers. Contact us today!




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Register Now: http://www.astricon.net

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   http://lists.digium.com/mailman/listinfo/asterisk-users