asterisk users - Jul 2008 - sip prune realtime per issue

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

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

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