vicidial.org

Hi Everyone,

I am using Vicidialnow 2.0.5 and i can see these files inside asterisk folder which files did vicidial runs? sip.conf or sip-vicidial.conf and extensions.conf or extensions-vicidial.conf?

Thanks,

Julz

Vicidial includes the vicidial-extensions.conf and vicidial-sip.conf in the original Asterisk files (extensions.conf and sip.conf). The updates you make through the web pages are written to the vicidial conf files.

Hi ykhan,

Can you give the sample configuration of your SIP and Extension that is working good?

Many Thanks,

Julz

With the latest version of Astguiclient/Vicidial, you don't need to edit the Asterisk configuration files by hand. You can do it via the Vici admin interface. You can add SIP carrier, phone extensions and etc all without touching the Asterisk backend.

Hi Gardo,

Can you give a sample configuration of your vicidial? Just a screen shot of how to setup trunk and dial plan, you can omit the sensitive information. All i need is the actual configuration. I have already tried the configuration but it's not working.

Thanks

There's already an example SIP and IAX entry in VicidialNOW 1.2. If you're still having problems, check the getting started guide: http://vicidialnow.org/wiki/vnow/GettingStartedGuide .

Please post your configuration on the Carriers page. We can then assist you in troubleshooting errors if any.

ykhan wrote:Please post your configuration on the Carriers page. We can then assist you in troubleshooting errors if any.

the link he gave above HAS a screen shot of the page you are asking for. Open it and look for "Create an outbound trunk:". The image below that is what you are asking.

I have found that changing "_91" to _81" on all three lines in the dialplan section is often necessary as there is already a _91 defined in the hard coded extensions.conf of some releases which can confuse asterisk. Using _81 bypasses that problem.

Alternatively, comment out the existing _91 files from your extensions.conf file. Especially if your setup is _91. in the Vicidial Carriers pages.

Actually, yes, we comment them out on all our builds.

In SVN trunk we now comment out the 91 extensions as well by default.

On the Dial plan sections of the extensions.conf, there are '_91' which is the case for the US. Here in Rwanda, phone code is 250 then 9 digits number, e.g. if you want to call me in Rwanda, you d type "+250 78 XXX XX XX" or my other phone of another carrier "+250 72 XXXXXXX",
but calling locally, you have simply to put '0' in front of those 9 digits, e.g. to call my friend locally i simply type "078 XXXXXXX"

Any idea (help) how to set up my extensions.conf

everything is working well so far except the inbound part, i can call leads



[May 31 21:19:31] == Parsing '/etc/asterisk/manager.conf': [May 31 21:19:31] Found
[May 31 21:19:31] == Manager 'sendcron' logged on from 127.0.0.1
[May 31 21:19:31] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-cacc,2", "8600051|F") in new stack
[May 31 21:19:31] > Channel Local/8600051@default-cacc,1 was answered.
[May 31 21:19:31] -- Executing [0788537919@default:1] AGI("Local/8600051@default-cacc,1", "agi://127.0.0.1:4577/call_log") in new stack
[May 31 21:19:31] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 31 21:19:31] -- Executing [0788537919@default:2] Dial("Local/8600051@default-cacc,1", "ZAP/g1/0788XXX919||tTor") in new stack
[May 31 21:19:31] -- Requested transfer capability: 0x00 - SPEECH
[May 31 21:19:31] -- Called g1/0788XXX919
[May 31 21:19:31] -- Zap/32-1 is proceeding passing it to Local/8600051@default-cacc,1
[May 31 21:19:33] == Manager 'sendcron' logged off from 127.0.0.1
[May 31 21:19:34] -- Zap/32-1 is ringing

it rings and people can pick up the call and talk with an agent


I created new In-Group, new DID entry, new inbound campaign... when i login i get to the conference (i use Xlite), but when i try to call the DID number the telco provided 0788XXX443, it fails:

[May 31 21:05:23] -- Executing [0788XXX443@default:1] Ringing("Zap/32-1", "") in new stack
[May 31 21:05:23] -- Executing [0788XXX443@default:2] Wait("Zap/32-1", "1") in new stack
[May 31 21:05:23] -- Accepting call from '788XXX919' to '0788XXX443' on channel 0/1, span 2
[May 31 21:05:24] -- Executing [0788XXX443@default:3] AGI("Zap/32-1", "agi://127.0.0.1:4577/call_log--fullCID--0788143000-----"" <788XXX919>-----788537919-----") in new stack
[May 31 21:05:24] -- AGI Script agi://127.0.0.1:4577/call_log--fullCID--0788143000-----"" <788537919>-----788537919----- completed, returning 0
[May 31 21:05:24] -- Executing [0788XXX443@default:4] Answer("Zap/32-1", "") in new stack
[May 31 21:05:24] == Auto fallthrough, channel 'Zap/32-1' status is 'UNKNOWN'
[May 31 21:05:24] -- Executing [h@default:1] DeadAGI("Zap/32-1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[May 31 21:05:24] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[May 31 21:05:24] -- Hungup 'Zap/32-1'


Any idea how to solve this problem, i m getting there, i feel it,.... just that inbound part which is giv me headaches