fall into to ivr while dialing through siptrunk

[albokos]

hello!

i am working on an vicidial install in a contact center. everything work great but i cant pass call through the sip trunk (i'm dialing in france). when i did i fall into an ivr which ask me to choose my langage and to enter a card number.

could someone help me and answer my questions:

1- where is this ivr comming from ?
2- how can i dial directly through the siptrunk

please help!! i am running out of time to set up the system.

PS:
=> i'v install vicidial from scratch on a debian distri,

=>i have the default config files and just config sip trunking like in the vnow getting started manual
=> no zaptel card everything is done by SIP

=>and i'm in morocco does it matter in configuration??

[mflorell]

Asking for a card number is not part of the ViciDial system, not sure where that came from.

[albokos]

Thank you for your reply i was in a worry because i though i messed up something.

i'm on this for 2 days now and i came to the conclusion that it is probably my provider who route my calls to its IVR and not vicidial (as you says).

Do you think i'm true i mean could this be an explanation??
And is there a difference between an softphone account and an account designed for sip trunking.


Thank you

[albokos]

and sorry for my so messy english, i'm better in french

[williamconley]

If you watch the CLI in asterisk while you are being asked for your card number, you should find your answer. If you "set verbose 20" you will see the activity being generated by YOUR system. If nothing is happening, then YOUR system isn't asking for a card number. Which means it's theirs ...

How many "channels" are you allowed to use with your SIP account? If it's only ONE or TWO, you will not be able to use this with Vicidial very well without some special programming. In which case, you'll need to get a new account with more channels available or get SEVERAL more accounts ...

[albokos]

Thanks !!

I'v contact my sip provider and he promise to check the account and reconf it for sip trunking. but just in case i'll test what you said today, i 'll set verbosity 20 and tell you what i get in CLI.

thanks once more

[albokos]

Excuse me but i've got another probleme, i tried to test calls with verbosity 20 but when i launch calls, asterisk dial via zap/g2/ instead of SIP/siptrunk as i designed it. i've not change my config since yesterday when it seems to "work fine". how can i select the right channel. Thanks

[williamconley]

set your system up to use "dial 8" instead of "dial 9", in some versions of the software there is a "dial 9" in the hard coded extensions.conf that will interfere with calls and cause this problem (or you can delete the hard-coded dial 9).

To use dial 8 isntead of dial 9, in your carrier setup:

Code: Select all

exten => _81NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _81NXXNXXXXXX,2,Dial(${YOUR_CARRIER_GLOBAL_VARIABLE}/${EXTEN:1},,To)
exten => _81NXXNXXXXXX,3,Hangup

After this, you just put 8 instead of 9 in your Campaign dial prefix.

If you want to remove the annoying entry in extensions.conf instead, you are looking for:

Code: Select all

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

If you put a ";" in front of those three lines and then issue a "reload" command at the asterisk CLI, it should resolve your issue.

[albokos]

It's ok now

Nothing with verbosity 20
my provider check the account he gave me and designed it for sip trunking the matter was the type of authemtification he sets (something with a pin code i think) now it uses ip@ to authenticate my server. Hope this info will spare sleepless nights to someone.

Thank you for wasting your time on the newbie i am, maybe i'll later do the same for someone else (i hope so).

BYE!!!!!