[Asterisk-Users] Random SIP Phone Problem

[Asterisk List]

I got the same problem with 04/19/05 CVS version.  I am using
Grandstream phones.  I also noticed that when this happens, an already
hung-up call was still shown as bridged between a SIP phone and a Zap
channel.

On 4/18/05, Shaun Tierney <stierney at prairiestream.com> wrote:
> I am currently running CVS-HEAD-04/15/05-13:15:00 and I have an issue that
> just recently cropped up.  I upgraded to this version of Asterisk last
> Friday and now twice in the last two hours, all of my Aastra SIP phones lose
> service suddenly.  Network connectivity is still there between the phones
> and the PBX, and I have restart Asterisk to fix the issue.  Would it be
> worth my time to move to the latest CVS Asterisk release even though it has
> only been three days since I installed the version in operation?  Or would I
> be better off going with a previous CVS release to fix the problem?  I can't
> use the stable release because I use macro arguments in the dial command.
> Here are the error messages that seem to show up for the duration of the
> problem.
> 
> Apr 18 13:43:50 NOTICE[16997] chan_sip.c: Peer 'brettb' is now UNREACHABLE!
> Last qualify: 1045
> Apr 18 13:44:03 VERBOSE[16997] logger.c: Don't know what to do if second
> ROSE component is of type 0x6
> Apr 18 13:44:07 NOTICE[16997] app_queue.c: Added interface 'SIP/brettb' to
> queue 'psc'
> Apr 18 13:44:14 NOTICE[16997] chan_sip.c: Peer 'brettb' is now REACHABLE!
> (76ms / 2000ms)
> 
> Regards,
> 
> Shaun
> 
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