[asterisk-users] Timeout when dialing dead peer

[Danny Nicholas]

There is a timeout function in the Dial command.  The folks who wrote the
command obviously felt that setting a programmatic limit on this would cause
somebody a problem.  If you expect a reply from your SIP peer in 30 seconds,
just do Dial(SIP/peer,30) and the line will disconnect in 30 seconds.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Stefan Schmidt
Sent: Monday, June 08, 2009 7:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Timeout when dialing dead peer

Benny Amorsen schrieb:
> A regular Dial(somepeer) to a SIP peer which doesn't reply at all seems
> to not time out, or at least have a very long time out.
> 
> We have a set up where we can dial two different peers, a primary and a
> backup peer. If the first one dies completely, so that no error messages
> (no ICMP unreachables or anything) are returned, Asterisk does not
> continue in the dial plan but just gets stuck on that one Dial(). I can
> of course put a time out in the Dial(), but then one call will turn into
> two calls if the person at the other end is too slow to answer their
> phone, so this is not very handy.
> 
> It is possible that qualify would help, but it is not a very nice
> answer -- Asterisk's use of SIP OPTIONs is non-standard, and it can
> impose a significant load on the peer.

What kind of client cant handle one packet per minute without getting a
higher load? The interval asterisk sends an Options packet is 60 seconds
 and the default timeout is 2 s for this packet. So i believe this
coudnt be a problem, or do you have a problem with the peer when a
second invite arrives during an active call?


> It would be good if Asterisk would give up after not receving any reply
> after a configurable interval.

What your are searching for is called Sip T1 Timeout and i´ve seen that
in asterisk 1.6.1.x you can set this timeout in sip.conf. Iam not sure
about changing this in other versions.

> 
> /Benny
> 

best regards

steve

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