problem with sip trunk

[mhido]

I have two servers VICIDIAL and I create a sip trunk between the two servers but I can not effectier a call between the server via the SIP trunk and below the configuration of two servers:

server 1:

Carrier ID: VICITRUNK
Carrier Name: VICITRUNK TEST
register => vicitrunk:VicT1234@192.168.1.a
[vicitrunk]
type=friend
disallow=all
allow=ulaw
allow=alaw
username=vicitrunk
secret=VicT1234
host=dynamic
dtmfmode=inband
qualify=yes
context=trunkinbound
insecure=very
permit=0.0.0.0/0.0.0.0

Protocol: SIP

Globals String: TRUNK = SIP/vicitrunk

exten => _9XX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XX.,2,Dial(${TRUNK}/${EXTEN:1},55,tTo)
exten => _9XX.,3,Hangup

server 2:

Carrier ID: VICITRUNK
Carrier Name: VICITRUNK TEST
register => vicitrunk:VicT1234@192.168.1.b
[vicitrunk]
type=friend
disallow=all
allow=ulaw
allow=alaw
username=vicitrunk
secret=VicT1234
host=dynamic
dtmfmode=inband
qualify=yes
context=trunkinbound
insecure=very
permit=0.0.0.0/0.0.0.0

Protocol: SIP

Globals String: TRUNK = SIP/vicitrunk

exten => _9XX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XX.,2,Dial(${TRUNK}/${EXTEN:1},55,tTo)
exten => _9XX.,3,Hangup

les deux trunk sont bien enregistrés.

SIP 101 in server 1
SIP160 in server 2

lorsque je fait un appel de 106(serveur 2 au serveur 1) au 101 j'ai les messages suivant sur la consol de deux serveur asterisk:

###serveur 2:

--executing AGI("SIP/cc160-087c3598","agi://127.0.0.1:4577/call_log") in new stack

--Called vicitrunk/101
--SIP/vicitrunk-087cb6f8 answered SIP/cc160-087c3598

==Spawn extension (default, 9101, 2) exited non-zero on 'SIP/cc160-087c3598'

--Executing DeadAGI("SIP/cc160-087c3598","agi//127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----8-----6") in new stack

--AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---8-----6 completes, returning 0

###serveur 1:

WARNING[5278]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0xb6e27d30', 9 retries!

--Executing AGI("SIP/vicitrunk-b6e01f00","agi-DID_route.agi") in new stack
--Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
--AGI Script agi-DID_route.agi completed, returning 0
--Executing Wait("SIP/vicitrunk-b6e01f00", "2") in new stack
--Executing Answer("SIP/vicitrunk-b6e01f00", "") in new stack
--Executing Playback("SIP/vicitrunk-b6e01f00", "ss-noservice") in new stack
--Playing 'ss-noservice' (language 'fr')
--Executing Playback("SIP/vicitrunk-b6e01f00", "vm-goodbye") in new stack
--Playing 'vm-goodbye' (language 'fr')
--Executing Hangup("SIP/vicitrunk-b6e01f00", "") in new stack




I need help I 1 months and I can not make a call between two servers thxxx

[williamconley]

1) it is a requirement that you post your configuration when you request assistance (so search engines will help others with similar configuration and problems, this helps everyone!). So post your Method of Installation, Vicidial Version and Build, Asterisk Version, and any Other software installed.

2) Why are you calling between the boxes? (trying to call an agent on the other box? trying to call out through the provider on the other box? want to check voice mail?)

3) ss-no service line in your post shows that the call DID get through, but it went to the inbound call agi script which defaults to "ss-no service" until you change it to go to an ingroup or elsewhere. read the manual for how to set up DIDs/Ingroups ...

4) Welcome to the party!

[mhido]

1--vicidialnow-1.3 second server with vicidial and astersik-1.2.30.4

2--I will make calls between two servers to test the SIP trunk by what I did with asterisk sip trunk without VICIDIAL simple and does not work when I tested with VAIT VICIDIAL using the graphical interface and I use vicidialnow-1.3

3--l'appel arrive de première serveur au deuxième mais il ne passe pas au numéro demander il tombe sur un message vocal est ce qu"il faut créer une route entrante pour atteint l'autre numéro?? s'il est nécessaire vous pouvez m'aider de faire ça??
comme je vous dite j'arrive pas à connecter deux serveur asterisk simple sans vicidial avec un trunk sip ou iax2

4--thank you

[williamconley]

1--please also post Vicidial Version and Build (vicidialNOW is good, but that's the installation method, vicidial is at 2.0.4, 2.0.5 or 2.2.0 and the build is located on almost any of the admin screens).

2--you are using the sip trunk to ... test the sip trunk? are you talking about server performance testing?

3--Sorry, my French from High School was 30 years ago.

[mhido]

1--server 1: vicidial-2.0.5 and server 2: 2.0.5

2--I do not want to test the performance of server I just want to create sip trunk between two servers and make a call between two SIP account different server that's my stumbles by eq I made with two asterisk without VICIDIAL and does not and also with freepbx Finally just does not work I do with VAIT VICIDIAL

3--I'm sorry I forgot to type the sentence in English so I said the first call comes from the second server but it does not apply to the number it lands on a voice message is what "we must create a route Inbound to reach another number? if necessary you can help me do that?
as I said I can not connect two asterisk server VICIDIAL simple without a sip or IAX2 trunk
4--thank u for help

[williamconley]

did you say you have freepbx on one of the boxes?

when your call "arrives" in the 2nd box, the method of routing the call is very tricky to deal with. vicidial, asterisk, contexts, freepbx, all these things can complicate the arrangements

it is usually NOT a good idea to route calls THROUGH a server (especially a FeePBX server) unless absolutely necessary, but it can be done.

[mhido]

I already used freepbx.
Now I removed freepbx and I installed VICIDIAL on two machines.
Now is that it is possible to call two SIP account on each server through a SIP trunk.
I configured a sip trunk between the two servers and call out the first server and is the second server but the call fails to account call to sip.
is what to call two accounts simply sip through VICIDIAL must create routes incoming and outgoing?

[williamconley]

please explain the route you would like the call to take through the second server. are you trying to generate a call to the outside world or to a phone/extension on the second server? (are you trying to call through it, or TO it?)