It could be that the phone is trying to (re)register too frequently and drops during the SIP negotiations.
-----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Carlos Chavez Sent: Monday, February 09, 2009 2:11 PM To: Asterisk Subject: Re: [asterisk-users] Call drops after a minute on 1.6.0.5 This problem only seems to occur when using Aastra phones. Calls to Polycom never drop. Anyone know of a setting for Aastra that could cause this? On Mon, 2009-02-09 at 13:22 -0600, Carlos Chavez wrote: > I upgraded my office PBX from 1.4.22 to 1.6.0.5 so we can start > evaluating and testing. I did not really test it over the weekend, just > made sure I could dial in and out. Today we are finding that incoming > calls to our POTS lines get dropped after a couple of minutes. All I > can see in the CLI is this: > > [Feb 9 13:00:22] WARNING[19831]: chan_sip.c:19266 proc_session_timer: > Session-Timer expired - [email protected] > > [Feb 9 13:00:22] NOTICE[19831]: chan_sip.c:17364 handle_request_invite: > Unable to create/find SIP channel for this INVITE > > Internal call between SIP phones are fine (for the moment at least I > can stay connected). All the session-* lines in sip.conf are commented > out so they are using the default values. Any ideas? > -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 _______________________________________________ -- 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
