[asterisk-users] Multiple Asterisk SIP Server/client connections

[Ken Williams]

When I said "same config" I meant same with minor differences of account
information :D

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

[104]
type=friend
secret=1234
dial=SIP/104
callerid=Video<104>
allowsubscribe=yes
host=dynamic
context=from-internal
insecure=port,invite 

I get no errors on the "main server" side, I get a lot of retransmitting
messages on the remotes that don't get a connection.  I'll post a
portion of the log file later.  The strange part to me is that it seems
as if it's the first person in that wins.  I can shutdown remote servers
and bring them up individually and they work, but after that initial one
it's as if the main server's ignoring port 5060 from other locations.  I
thought perhaps it was a firewall/router problem on the main server, so
I swapped out a netgear router for a linksys wrt54g, same problem occurs
on both routers.  All 4 servers are listed as DMZ on their local
firewalls.

Ken

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Noah
Miller
Sent: Wednesday, July 30, 2008 2:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Multiple Asterisk SIP Server/client
connections

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

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