[asterisk-dev] trunk problem today with sip?

[Rich Adamson]

> > > FYI...
> > >
> > > Looks like svn trunk as of this afternoon has an issue maintaining sip
> > > registration.
> > >
> > > Several Cisco, Snom, ata's, etc, disappear from 'show sip peers' after
> > > an hour or two, and calls cannot be completed to them. Haven't attempted
> > > to narrow down the issue as yet.
> > >
> > 
> >  Unless I'm missing messages from svn-commit (which seems always a
> > possibility with gmail lately), it doesn't look like chan_sip.c has
> > changed today. What was the last version that was working for you?
> 
> I reverted to SVN-trunk-r8852 (Jan 29th) and everything seems to have
> been stable for the last 63 minutes. Will need to wait a little
> while longer to validate for sure.
> 
> This afternoon was fresh svn trunk (zaptel, libpri, asterisk) with
> no patches or other changes. Had to clean up left over modules in
> /usr/lib/asterisk/modules, and started asterisk with '-cddvvvvv' to
> ensure no unresolved errors/changes; then safe_asterisk. Review of 
> logs did not indicate any issues; CLI did not indicate any issues, only
> 'sip show peers' as mentioned. This is a small psuedo-production fc3 
> box that I don't mind testing/working with trunk code. No realtime
> or database config'ed; all static config files.
> 
> Stopping, restarting asterisk, and waiting a few minutes would bring
> the majority of the sip peers back online. Sometime later (unknown
> exactly how long) the sip peers would disappear again.

This morning I switched back to the svn trunk from yesterday's attempts,
did a make update, recompiled & installed, and so far no dropped sip
registrations. Not sure if the difference was related to the changes
to iax2 and sip that markster applied last night or something else.
Seems to be stable now.

Rich