setting SIP trunk between 2 asterisk servers

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setting SIP trunk between 2 asterisk servers

Postby bored_to_death » Thu Apr 05, 2012 3:04 am

hi guys,

i am new to asterisk. my problem is that i want to connect two asterisk servers (i named them TX182 and TX183) with a SIP trunk. TX182 is for clients with 1XX pattern, and TX183 is for clients with 2XX pattern. i followed a tutorial for this.
now everything seems OK, and my two servers recognize eachother. but when i try to make a call from one client of TX183 to a client of TX182, i get error. in asterisk debug in TX183, when i try to connect, it says:

Code: Select all
-- Executing [103@home:1] Dial("SIP/201-00000002", "SIP/1-sip-trunk/103") in new stack
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Called 1-sip-trunk/103
    -- SIP/1-sip-trunk-00000003 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [103@home:2] Hangup("SIP/201-00000002", "") in new stack


what am i doing wrong?

thank you.

here is what i added in config files:

in TX182 in "sip.conf" i added:
Code: Select all
[2-sip-trunk]
type=peer
context=internal
host=192.168.0.183
disallow=all
allow=g729
canreinvite=yes
qualify=yes
;


in TX182 in "extensions.conf" i added:
Code: Select all
exten => _2XX,1,Dial(SIP/2-sip-trunk/${EXTEN})
exten => _2XX,2,Hangup()


similarly, in TX183, in "sip.conf" i added:
Code: Select all
[1-sip-trunk]
type=peer
context=internal
host=192.168.0.182
disallow=all
allow=g729
canreinvite=yes
qualify=yes
;


and in TX183 in "extensions.conf" i added:
Code: Select all
exten => _1XX,1,Dial(SIP/1-sip-trunk/${EXTEN})
exten => _1XX,2,Hangup()
bored_to_death
Newsterisk
 
Posts: 5
Joined: Thu Apr 05, 2012 3:00 am

Re: setting SIP trunk between 2 asterisk servers

Postby dejanst » Sat Apr 07, 2012 12:22 am

Have you bought a license for g729 on both Asterisk servers?

Do a "sip set debug on" in Asterisk CLI on one of the servers and copy/paste the output.
dejanst
Astmaster
 
Posts: 849
Joined: Tue Apr 27, 2010 7:14 am
Location: Ljubljana, Slovenia

Re: setting SIP trunk between 2 asterisk servers

Postby david55 » Sat Apr 07, 2012 3:28 am

It is much easier to debug this sort of thing when one knows what the rejection reason was. You should use sip set history ... to obtain the SIP response codes for the rejection. Better, you should use sip set debug on, with appropriate debug and verbose levels, but this can produce a lot of output, which may discourage people from looking, so you should try to identify the key points before posting it.

Also, the best place to get debugging information is the server that rejected the call, not the one that originated it.
david55
Moves Like Spencer
 
Posts: 9734
Joined: Fri Sep 26, 2008 5:03 am


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