Error in changing VOIP provider

[Milhouse28]

We just acquire new voip service, but after adding and enabling their settings to sip.conf and extensions.conf, we encountered problem. Vicidial interface shows "No Live Call" even though agents were able to talk to the customer. I dont know whats causing this problem.

Here is the settings of the old provider in extensions.conf:
; dial a local outbound number
exten => _9NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9NXXXXXX,2,Dial(${TRUNK}/1727${EXTEN:1},,To)
exten => _9NXXXXXX,3,Hangup

; dial a long distance outbound number
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,To)
exten => _91NXXNXXXXXX,3,Hangup


Here is the setting of the new provider in the extensions.conf:
;New VOIP
exten => _1.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1.,2,Dial(SIP/${EXTEN}@newvoip)
exten => _1.,3,Hungup

;New VOIP long distance
exten => _011.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _011.,2,Dial(SIP/${EXTEN}@newvoip)
exten => _011.,3,Hungup

[ykhan]

Here is the setting of the new provider in the extensions.conf:
;New VOIP
exten => _1.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1.,2,Dial(SIP/${EXTEN}@newvoip)
exten => _1.,3,Hungup

;New VOIP long distance
exten => _011.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _011.,2,Dial(SIP/${EXTEN}@newvoip)
exten => _011.,3,Hungup

Not sure if this will affect it, but change the bold text to Hangup instead of Hungup.

Also you have removed the 9 prefix, did you update the campaign with the same change?

[Milhouse28]

Yes, i replaced the prefix in the campaign settings. Also changed the 'Hungup' to Hangup' still having the problem. What could be causing this problem?

[cristian]

CLI output using screen -r please.

What dial mode are you in?

[Milhouse28]

Here is the output on screen -r

There are several suitable screens on:
3671.ASTVDauto (Detached)
3662.ASTupdate (Detached)
3608.asterisk (Detached)
3548.ASTVDadapt (Detached)
3674.ASTVDremote (Detached)
3668.ASTlisten (Detached)
3550.ASTfastlog (Detached)
3665.ASTsend (Detached)
Type "screen [-d] -r [pid.]tty.host" to resume one of them.

Were using predictive dial and also manual dial.

[okli]

He meant output of asterisk CLI, gathered by entering via screen interface.

Not sure if that's your problem but this is everywhere in the samples of extensions.conf :

; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls

Try adding ,,To , as it was set for your previous trunk:

Code: Select all

...
exten => _1.,2,Dial(SIP/${EXTEN}@newvoip,,To)
...

exten => _011.,2,Dial(SIP/${EXTEN}@newvoip,,To)

[Milhouse28]

seems its working now. ill post again if i see some problems again. thanks.

[williamconley]

Also, you should look at the [globals] section and notice that TRUNK is a variable used throughout the system. You define it in globals and then use it in any (as many) dial plan(s) you like. Then when you want to change providers, you just change TRUNK to be something else, instead of having to change it everywhere (especially useful if you use different providers for international calling to various countries).

then you would find that you could have changed TRUNK= to something else and that would have changed your existing provider, but you can also define another variable (TRUNK2?) and use that. Makes it easier to make revisions later on.

Also allows you to NOT mess up the ",,To) " portion by accident (because you're not in that area any more, you're just in globals).