can't dialout

[manickam]

hi,

i've installed vicidial and followed all instructions.

my sip trunk is registered, i can see it by sip show registry.

but when i try diling it allways keeps telling i'm sorry that is not a valid extension.

this is my dialplanentry as entered in gui

exten => _9NXXXXXXXXX,AGI(agi://127.0.0.1:4577/call_log)
exten => _9NXXXXXXXXX,2,Dial(${trunk12}/${EXTEN:1},,To)
exten => _9NXXXXXXXXX,3,Hangup

and this is what i get on cli

-- Executing MeetMe("Local/8600051@default-ecaa,2", "8600051|F") in new stack
> Channel Local/8600051@default-ecaa,1 was answered.
== Starting Local/8600051@default-ecaa,1 at default,90561570882,1 failed so falling back to exten 's'
== Starting Local/8600051@default-ecaa,1 at default,s,1 still failed so falling back to context 'default'
-- Sent into invalid extension 's' in context 'default' on Local/8600051@default-ecaa,1
-- Executing Playback("Local/8600051@default-ecaa,1", "invalid") in new stack
-- Playing 'invalid' (language 'en')
Sep 29 21:58:35 WARNING[5865]: file.c:1045 ast_waitstream: Unexpected control subclass '-1'

thanks for help

[gardo]

The dialplan 9NXXXXXXXXX doesn't match the number you called (90561570882). N matches any digit from 2-9 and 0 doesn't belong to N.

This might help you create the right dialplan:

X matches any digit from 0-9
Z matches any digit from 1-9
N matches any digit from 2-9

[manickam]

thanks gardo

i've changed the dial plan through gui as this to avoid conflicts with _9 in extension.conf, but still having the same problem

exten => _1XXXXXXXXXX,AGI(agi://127.0.0.1:4577/call_log)
exten => _1XXXXXXXXXX,2,Dial(${trunk12}/${EXTEN},,To)
exten => _1XXXXXXXXXX,3,Hangup

[manickam]

[resolved]done
i'e downloaded the new version istaaled it followed the getting started
but instead og _9 in the gui installation of dialplan i put _7

downloaded the free g729 and made calls through web interface on test camaign.

Now i'll be trying the predictive.
sure i'll be back giving you headaches lolll

thanks a lot
[/resolved]

[shobhit29]

Hi i am also facing the same problem can you please tell me how you have configured it .. and what changes i should do on extensions.conf and same do we have to make in carrier kindly help

[manickam]

hi

you make a fresh install, and follow the instructions of the guide step by step, when you install your sip provider, ind the dial plan window in the gui interface you should use _1 or _7 but not the _9 as in the manual because it would create a conflict with the dial plan in extensions.conf

try out this way and notify if any problem

[williamconley]

or _8 or _6, or all of them if you have multiple carriers (use one for each carrier)

also you CAN delete the _9NXXNXXXXXX lines from the extensions.conf file (just put ; in front of them if you're nervous about deleting). afterwards you'll need to "reload" at the asterisk cli (or reboot).

[shobhit29]

Hi please see my sip show peers status

cc101/cc101 192.168.1.6 D N 5060 Unmonitored
cc100/cc100 192.168.1.5 D N 44598 Unmonitored
trunk12/crosslink 64.132.217.195 N 5060 OK (797 ms)
53 sip peers [53 online , 0 offline]

But when i make a call it says
error
Got SIP response 481 "Call/Transaction Does Not Exist" back from 192.168.1.5

Please advice