vicidial.org

ViciBox_Redux.x86_64-3.1.15
Asterisk v.1.4.39.2-vici
ViciDial v.2.2.1
dual core 1.6
4 gig ram
hobbyist only

The installation seems smooth and easy, I make some extension sip and iax2 phones,can contact each other with no problem what so ever . The problem occur when I tried to install sip trunk.It can't call US number manually. I thought before because it was not configure that way it should be when dialing, within the vici system itself.I did tried that and still got the same problem.this is my trunk configuration. I am really confused because when I tried to check it in the asterisk using sip show peers it was there reachable. help please

[callcentric]
type=peer
qualify=yes
context=from-trunk
host=callcentric.com
defaultuser=17772672581
secret=xxxxxxxx
fromuser=17772672581
fromdomain=callcentric.com
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=ulaw
sendrpid=yes
trustrpid=no

GLOBALS STRING: SIPTRUNK = SIP/callcentric

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${callcentric}/${EXTEN:2},,tTor)
exten => _91NXXNXXXXXX,3,Hangup

continued from here http://vicidial.org/VICIDIALforum/viewt ... =8&t=25764 apparently.

good post btw.

change this line: (before/after)
exten => _91NXXNXXXXXX,2,Dial(${callcentric}/${EXTEN:2},,tTor)
exten => _91NXXNXXXXXX,2,Dial(${SIPTRUNK}/${EXTEN:2},,tTor)

Sorry my friend i kept on changing my configuration hoping this will solved my vici problem. Sad to say I didn't hit the right button yet . This is my new configuration.Since it's default configuration in my phone I supposed I can call outside vici manually.Thanks for trying to help me.

[testcarrier]
type=peer
qualify=yes
context=from-trunk
host=callcentric.com
defaultuser=17772672581
secret=xxxxxxx
fromuser=17772672581
fromdomain=callcentric.com
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=ulaw
sendrpid=yes
trustrpid=no

TESTSIPTRUNK = SIP/testcarrier

exten => _XXXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXXXXXXXXXX,2,Dial(${TESTSIPTRUNK}/${EXTEN:2},,tTor)
exten => _XXXXXXXXXXXX,3,Hangup

-- Executing [914099866700@default:1] AGI("SIP/gs102-0000000a", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 15 21:21:18] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Sep 15 21:21:18] -- Executing [914099866700@default:2] Dial("SIP/gs102-0000000a", "SIP/testcarrier/4099866700||tTor") in new stack
[Sep 15 21:21:18] -- Called testcarrier/4099866700
[Sep 15 21:21:19] NOTICE[4230]: chan_sip.c:13470 handle_response_invite: Failed to authenticate on INVITE to '"Test Admin Phone" <sip:17772672581@callcentric.com>;tag=as1b938c83'
[Sep 15 21:21:19] -- SIP/testcarrier-0000000b is circuit-busy
[Sep 15 21:21:19] == Everyone is busy/congested at this time (1:0/1/0)
[Sep 15 21:21:19] -- Executing [914099866700@default:3] Hangup("SIP/gs102-0000000a", "") in new stack
[Sep 15 21:21:19] == Spawn extension (default, 914099866700, 3) exited non-zero on 'SIP/gs102-0000000a'
[Sep 15 21:21:19] -- Executing [h@default:1] DeadAGI("SIP/gs102-0000000a", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CONGESTION----------") in new stack
[Sep 15 21:21:19] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

Sorry guys for bothering your peace it's my callcentric configuration setup. I thought we're using the latest asterisk, I set it up for the latest configuration.well well well for those who tried to help me I'm very grateful of you. For those who doesn't bothered to extend their hands ,you make me study the dialplan a little deeper.thnx

oh, so you manage to fix your problem?