[asterisk-users] sip prune realtime per issue

[Marc Smith]

On Tue, Jul 15, 2008 at 12:05 PM, Peder @ NetworkOblivion
<peder at networkoblivion.com> wrote:
> I am using realtime on two boxes, one running 1.4.10.1 and one running
> 1.4.11.  Everything works fine except for when I make a database change,
> such as a phones password.  I change the DB, I prune the peer, I see it
> is gone and then I see it show up again in "sip show peer xxxx", but
> everything is not being updated.  The phone will not register even
> though the DB and the phone have the correct password.  The only way to
> get it to register is to stop * and re-start it, then it works fine.  I
> even tried changing the callerid and pruned the peer.  A sip show peer
> shows the correct callerid, but when you call into voicemail, it is
> using the old callerid.  Again, if I stop * and restart, it works fine.
>
> Has anybody seen this bug and if so, know what the bug ID is?  We have a
> bunch of patches on these boxes and can't just upgrade to any old
> version to see if it fixes it.  I need to figure out what the bug is.  I
> did some research, but couldn't find it.
>
> Peder
>

Do the rt* options in sip.conf have any effect? Maybe one of those might help?

--Marc


> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
> !DSPAM:1,487ccb5365666785646901!
>
>
>