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

[John Todd]

On Dec 18, 2008, at 7:22 AM, Julien Chavanton wrote:

> 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/${EXTEN}@remote-sip1,5)
>  exten => _X.,1,Dial(SIP/${EXTEN}@remote-sip2,5)
>
>
>
> Solution2: it will wait until 5 seconds of timeout (on answer) and  
> then try the second alternative "n"
>
>  exten => _X.,1,Dial(SIP/${EXTEN}@remote-sip1,5)
>  exten => _X.,n,Dial(SIP/${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.


You should look at the configurable T1 timers in sip.conf, which allow  
you to specify the retransmit intervals.  I think this will do what  
you want, but it is a very dangerous setting that can lead to  
significant unintended consequences.  First, do some reading on what  
the T1 timer does - Google can help there.

;--------------------------- SIP timers  
----------------------------------------------------
; These timers are used primarily in INVITE transactions.
; The default for Timer T1 is 500 ms or the measured run-trip time  
between
; Asterisk and the device if you have qualify=yes for the device.
;
;t1min=100                      ; Minimum roundtrip time for messages  
to monitored hosts
                                 ; Defaults to 100 ms
;timert1=500                    ; Default T1 timer
                                 ; Defaults to 500 ms or the measured  
round-trip
                                 ; time to a peer (qualify=yes).
;timerb=32000                   ; Call setup timer. If a provisional  
response is not received
                                 ; in this amount of time, the call  
will autocongest
                                 ; Defaults to 64*timert1

JT


---
John Todd                       email:jtodd at digium.com
Digium, Inc. | Asterisk Open Source Community Director
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