[Asterisk-Users] sip to sip
Quintin
quintin at kulweb.co.za
Mon May 23 07:41:02 MST 2005
_____
From: Quintin [mailto:quintin at kulweb.co.za]
Sent: 23 May 2005 02:08 PM
To: 'asterisk-users at lists.digium.com'
Subject: sip to sip
Hi
I'm trying to put up an sip pbx system for my company but i'm getting some
problems when I'm trying to call from server ( branch A ) to server ( branch
B ).
This is my extentions.conf :
exten => 3003,1,Dial,SIP/3003 at 192.168.0.200
________________________________________________________
And this is what I get when I try to dial that user in branch B
_________________________________________________________
-- Executing Dial("SIP/5001-66b1", "SIP/3003 at 192.168.0.200") in new
stack
-- Called 3003 at 192.168.0.200
-- Got SIP response 404 "Not Found" back from 192.168.0.200
-- SIP/192.168.0.200-e638 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/5001-66b1' status is 'CONGESTION'
Both servers are exactly the same...
What can the problem be, that branch B server doesn't route the call through
Thx
Quintin
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