[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.

Rich