[asterisk-users] Call drops after a minute on 1.6.0.5

Danny Nicholas danny at debsinc.com
Mon Feb 9 14:17:38 CST 2009


It could be that the phone is trying to (re)register too frequently and
drops during the SIP negotiations.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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 - 5514a9165907723e0131451e5e540951 at 192.168.2.50
> 
> [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




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