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If your looking to link 2 asterisk boxes
might I suggest IAX. Much more efficient in the way bandwidth is utilized between the locations. Also
if you want to use your sip solution, have you setup the other end point in your SIP.CONF? I have never
got IP dialing to work in asterisk but it works fine when assigned in the conf file. .o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Systems Engineer From: 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/[EMAIL PROTECTED] ________________________________________________________ And this is what I get when I try to dial that user in
branch B _________________________________________________________ -- Executing
Dial("SIP/5001-66b1", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED] -- 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 This email was scanned by: Mcafee GroupShield |
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