sip trunk problem

[villbonzload]

I have a problem with my vicibox_redux.x86_64 3.1.15 .I already put the sip trunk and it says there's no problem with it (see paste below) but every time I call outside cannot connect (see paste below).Their must me a configuration error in it.I try to not to put dialpaln entry coz it's says optional but same result.I have a little back ground in elastix but never helps.Please help.

vicibox_redux.x86_64 3.1.15 suse
dual core 1.6
asus pss-mx sse
3 g ram

gs102/gs102 192.168.254.7 D N 42552 OK (1 ms)
callcentric/17772672581 204.11.192.39 N 5080 OK (326 ms)

[Sep 5 15:26:53] NOTICE[7786]: chan_sip.c:15566 handle_request_invite: Call from 'gs102' to extension '00639166962508' rejected because extension not found.

00-011 prefix for non US number for callcentric.

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(17772672581:xxxxxx@callcentric.xxx/17772672581:5060/${EXTEN:2},,tTor)
exten => _91NXXNXXXXXX,3,Hangup

[villbonzload]

what's wrong with my post?

[villbonzload]

grabeha pud ninyo oi.bag-uhan ko ani ni pud mo motubang sa akong mga pangutana.bisag gamay ra gud.unsa man diay ni nga forum pa sikat ra.

[DomeDan]

I dont usually answer post regarding dial-plan setup and stuff like that.
but I will give it a try because no one else seams to do it.

the dialed number '00639166962508' can be called because you have no part in the dial-plan to take care of it.

if you put a '9' in 'Dial Prefix:' on your campaigns then asterisk will try to call '900639166962508'

to handle that phone number you should edit you dialplan to something like this:

exten => _90XZXXXXXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _90XZXXXXXXXXX.,2,Dial(17772672581:xxxxxx@callcentric.xxx/17772672581:5060/${EXTEN:3},,tTor)
exten => _90XZXXXXXXXXX.,3,Hangup


and btw this "17772672581:xxxxxx@callcentric.xxx" should probably be in the registration string
"register => 17772672581:xxxxxx@callcentric.xxx:5060"
and global string like this
"SIPTRUNK = SIP/somename"
where somename is the name of the account entry "[somename]"
and in the end the dial plan would like like this instead:

exten => _90XZXXXXXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _90XZXXXXXXXXX.,2,Dial(${SIPTRUNK}/${EXTEN:3},,tTor)
exten => _90XZXXXXXXXXX.,3,Hangup

with "${EXTEN:3}" this number "900639166962508" will make asterisk call this "639166962508" number

[villbonzload]

Thanks Bro for your answer. I'm hoping this dialplan configuration will solve my problem to run vici. I really appreciated the effort.mabuhay ka pare.

[villbonzload]

Since the configuration intended for US call so I make a few try call to make it sure the setup is ok. I did direct dial it to eyebeam manually.I have new result below..I know you guys out there you have a lot of experiences when it comes to vici,please give me a little piece of your mind.

[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

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

[Sep 12 22:22:40] == Spawn extension (default, 916077462135, 3) exited non-zero on 'SIP/gs102-0000000f'
[Sep 12 22:22:40] -- Executing [h@default:1] DeadAGI("SIP/gs102-0000000f", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----66-----CHANUNAVAIL----------") in new stack
[Sep 12 22:22:40] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

[williamconley]

1) the "context=" entry should ALWAYS be "context=trunkinbound" (Not particularly important today, but when you want inbound ... no reason to go back and fix all of them).

2) You did not show us what happened above this. "default, 916077462135, 3" represents line 3 in the extension. Lines 1 and 2 would be handy, along with anything else directly resulting from this phone call. You showed only the result (like showing a flat tire while omitting the "stop sticks" you ran over that may be an obvious cause for the flat tire). Also: You may have no contact with the carrier or a long qualify time (either of which will cause asterisk to refuse to build the channel with the nonexistent or unworthy peer) or the call may have failed for some other reason.

3) "sip show peers" what does this particular sip connection say? (OK is good, unreachable is ... not so good, LOL).