Tilghman Lesher
2007-Nov-08 05:58 UTC
[asterisk-users] weird 185 secs timeout call problem
On Thursday 08 November 2007 00:13:44 Andre Quintaes wrote:> On our tests using asterisk, some calls have been terminated > abruptely with exact 185 seconds. This is happening with all our > incoming calls from a trunk from 1 of my DID providers ( other > providers or trunks are fine) and I could reproduce it by calling a > queue from my Wengophone Softphone and letting the MoH play for 185 > secs. If I make the same call from my WRTP54G on the same place, the > call doest not get hung up after 185 secs. > The incoming calls go trhough a queue and get mixmonitored. I will > make further tests but I tried changing several timeout and keepalive > parameters on sip.conf but nothing got effect. Even tried with > reinvites enabled and disabled. > > Does any one have a clue?Yes, you probably have a firewall which is timing out your UDP sessions. You wouldn't happen to be using a SonicWall, would you? Those are the most infamous devices among VOIP users for shutting down calls without warning. -- Tilghman
On our tests using asterisk, some calls have been terminated abruptely with exact 185 seconds. This is happening with all our incoming calls from a trunk from 1 of my DID providers ( other providers or trunks are fine) and I could reproduce it by calling a queue from my Wengophone Softphone and letting the MoH play for 185 secs. If I make the same call from my WRTP54G on the same place, the call doest not get hung up after 185 secs. The incoming calls go trhough a queue and get mixmonitored. I will make further tests but I tried changing several timeout and keepalive parameters on sip.conf but nothing got effect. Even tried with reinvites enabled and disabled. Does any one have a clue? Thanks
exten => whatever,1,Answer() rest of your dialplan for the queue Thanks, Steve Totaro Andre Quintaes wrote:> On our tests using asterisk, some calls have been terminated > abruptely with exact 185 seconds. This is happening with all our > incoming calls from a trunk from 1 of my DID providers ( other > providers or trunks are fine) and I could reproduce it by calling a > queue from my Wengophone Softphone and letting the MoH play for 185 > secs. If I make the same call from my WRTP54G on the same place, the > call doest not get hung up after 185 secs. > The incoming calls go trhough a queue and get mixmonitored. I will > make further tests but I tried changing several timeout and keepalive > parameters on sip.conf but nothing got effect. Even tried with > reinvites enabled and disabled. > > Does any one have a clue? > > Thanks > > _______________________________________________ > --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 > >