[asterisk-users] Dial timeout with SIP - how to set timeout for INVITE ACK

[Julien Chavanton]

>So why not use 30 and let Asterisk take care of the SIP details/
>timeouts?
 
Asterisk will wait the until it receive "answer" or timeout
 
I need to timeout a SIP call on SIP INVITE ACK, in ISDN for exmaple this is translated to PROCEEDING 
Meaning "I have received the call, now I will look what to do with it"
 
The result with the suggested timeout is not good enought, you may wait for the whole timeout even if the other side as not sent anything, this will be the case for all your calls, depending on the timeout this would be killing the traffic.
 
 

________________________________

From: asterisk-users-bounces at lists.digium.com on behalf of Philipp Kempgen
Sent: Thu 18/12/2008 4:17 PM
To: Asterisk Users
Subject: Re: [asterisk-users] Dial timeout with SIP - how to set timeout for INVITE ACK



Julien Chavanton schrieb:
> I have a concern with Dial command, I want to enable a secondary route with a remote partner, if the first route fails then we use the second one :

> Solution1: it will try both (there will be 2 simultanious actives calls ringing) this is not clean when calling an endusers
> 
>  exten => _X.,1,Dial(SIP/${EXTEN}@remote-sip1,5 <SIP/${EXTEN}@remote-sip1,5 <mailto:SIP/${EXTEN}@remote-sip1,5> > )
>  exten => _X.,1,Dial(SIP/${EXTEN}@remote-sip2,5 <SIP/${EXTEN}@remote-sip2,5 <mailto:SIP/${EXTEN}@remote-sip2,5> > )

You can't have the same "priority" (1) more than once per
extension (_X.).

> Solution2: it will wait until 5 seconds of timeout (on answer) and then try the second alternative "n"
> 
>  exten => _X.,1,Dial(SIP/${EXTEN}@remote-sip1,5 <SIP/${EXTEN}@remote-sip1,5 <mailto:SIP/${EXTEN}@remote-sip1,5> > )
>  exten => _X.,n,Dial(SIP/${EXTEN}@remote-sip2,5 <SIP/${EXTEN}@remote-sip2,5 <mailto:SIP/${EXTEN}@remote-sip2,5> > )
> 
> the problem is we can not select what timeout represents, timeout on ACK from INVITE would be perfect I think (1 second for example), timeout for answer ? this is to hard to predict, some mobile phone can ring for 30 seconds, etc.

So why not use 30 and let Asterisk take care of the SIP details/
timeouts?

And just to be sure: Don't put those "mailto" things in
extensions.conf.  :-)


   Philipp Kempgen

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