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-boun...@lists.digium.com [mailto:asterisk-users-boun...@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 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users