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

Marius Ciorecan marius.ciorecan at sarmet.com
Mon Sep 7 03:28:34 CDT 2009


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
>
>
>
> ---
> oej at edvina.net - http://edvina.net
> Open Unified Communication - building platforms with SIP and XMPP
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