[Asterisk-Users] sip to sip 
    Quintin 
    quintin at kulweb.co.za
       
    Mon May 23 06:22:55 MST 2005
    
    
  
Hi B
 
Do you mean I must do this in my sip.conf file on eatch server
 
Branch A 
register => 3001:1234 at 192.168.0.200 /3001
 
Branch B
register => 5001:1234 at 192.168.0.227 /5001
 
thx
Q
 
  _____  
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Brian C.
Fertig
Sent: 23 May 2005 03:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] sip to sip 
 
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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Quintin
Sent: Monday, 23 May, 2005 08:08
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] 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
  _____  
This email was scanned by: Mcafee GroupShield 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/d03cb565/attachment.htm
    
    
More information about the asterisk-users
mailing list