asterisk users - Jul 2008 - Multiple Asterisk SIP Server/client connections

I have 4 asterisk servers.  They all have local phones on their local
network they manage for SIP based conversations.  We then have IAX
between them all for inter-asterisk connections.
 
This setup has worked well for nearly 2 years now, minor problems here
and there but overall very nice.
 
Recently we acquired some Polycom video conference units.  I was able to
setup our main server to host all the video coordination using video
over SIP.  I was able to configure the video conference units on the
local network, have all 4 of them (one going to each remote server)
displaying 4 videos on the local network.
 
I then sent them out to their remote facilities and setup Asterisk with
as a SIP client on the 3 remote locations to talk to the main server.
One at a time we tested them and they worked one on one.
 
Recently we tried to get two going, and I noticed there seems to be an
issue with the SIP registration if one of the 3 remote SIP clients has
already registered.  That is, the other requests are unanswered or not
fully registered for some reason or another.  At very random times I've
actually managed to get 2 of the 3 connected, but inevitably I lose one
of those 2 shortly after.
 
The SIP.CONF has been made identical across all 3 remote locations, and
the main server has the same config for each remote site connecting.
 
I first want to confirm that it's possible to have 3 remote Asterisk
servers setup as a SIP client connected to a 4th Asterisk server.  
 
Assuming it is possible, here is the SIP Client SIP.CONF:
 
[general]
register => 103:1234 at yy.yy.yy.yy/699
defaultexpirey=1800
maxexpirey=3600
relaxdtmf=yes
videosupport=yes
disallow=all
allow=ulaw
allow=gsm
allow=h263p
canreinvite=no
limitonpeer=yes
notifyringing=yes
notifyhold=yes
externip=xx.xx.xx.xx.xx
fromdomain=xx.xx.xx.xx
localnet=192.168.0.0/255.255.255.0

[yy.yy.yy.yy]
type=friend
host=yy.yy.yy.yy
insecure=port,invite

[699]
type=friend
secret=1234
dial=SIP/699
callerid=Video <699>
allowsubscribe=yes
host=dynamic
context=from-internal
insecure=port,invite

In addition here's the relevant portions of the SIP.CONF from the main
server:
 
[general]
videosupport=yes
disallow=all
allow=ulaw
allow=gsm
allow=h263p
canreinvite=no
fromdomain=yy.yy.yy.yy
externip=yy.yy.yy.yy
localnet=10.200.26.0/255.255.255.0
nat=yes
bindport=5060

[103]
type=friend
secret=1234
dial=SIP/103
callerid=Video<103>
allowsubscribe=yes
host=dynamic
context=from-internal
insecure=port,invite

Please, any suggestions would be great.  I've been bashing my head
against the keyboard all day trying to find why it's acting in this way.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20080729/8cb94fdb/attachment.htm

Hi Ken -
> The SIP.CONF has been made identical across all 3 remote locations, and the
> main server has the same config for each remote site connecting.
>
> I first want to confirm that it's possible to have 3 remote Asterisk
servers
> setup as a SIP client connected to a 4th Asterisk server.
I just want to double-check the setup you have:  you say the main
server has the "same config" for each remote site connecting.  Does
that mean they're all connecting to the same SIP user/friend account?
If so, that wouldn't work.  You need to have a unique SIP account for
each SIP device that's connecting.

If that's not the case, and you have a unique sip account for each of
your Polycom devices, can you show us the relevant part of your
sip.conf from the main asterisk server?  Also, do you get any
particular messages on the console regarding this?  Have you tried
turning on SIP debugging?

Thanks,
Noah