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
