Manual Dial Issue

[Srinivasa Rao]

Hi Every One,
Need Help,

Following are my server details

1. Asterisk 1.4.42
2. vicidial 2.2.1
3. Redfone PRI terminator
4. Redhat 6.5
5. Soft Phone Zoiper

Installation Method : Vicidialnow
$version = '2.2.1-259';
$build = '100510-2014';
Asterisk version: 1.4.42-vici

I am using the vicidial application from past five years, it is very nice. Suddenly in my agent application I am getting issue with outbound dialing. Most of the times when an agent uses manual dial option, it is not getting connected to asterisk, when I check the asterisk log I didnt find any log for that number.

The same configuration is working after sometime with the same number.


Success Case:

Whenever user dials the number, user conference created in meetme and after that it searches for pattern and it redirects to matching pattern then the call will be routed to dialed number.

Code: Select all

[default]
exten => _86000[5-9]X,1,Meetme,${EXTEN}|q

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,n,Set(test3=${EXTEN})
exten => _91XXXXXXXXXX,n,Set(str3=${test3:2:11})
exten => _91XXXXXXXXXX,n,Dial(Zap/g2/${str3},30)
exten => _91XXXXXXXXXX,n,Hangup

Success Log:

-- Executing [8600069@default:1] MeetMe("Local/8600069@default-42a8,2", "8600069|q") in new stack
> Channel Local/8600069@default-42a8,1 was answered.
-- Executing [919885098850@default:1] AGI("Local/8600069@default-42a8,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [919885098850@default:2] Playback("Local/8600069@default-42a8,1", "/var/UPCL/english/wel2") in new stack
-- Executing [919885098850@default:3] Set("Local/8600069@default-42a8,1", "test3=919885098850") in new stack
-- Executing [919885098850@default:4] Set("Local/8600069@default-42a8,1", "str3=9885098850") in new stack
-- Executing [919885098850@default:5] Dial("Local/8600069@default-42a8,1", "Zap/g2/9885098850|30") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g2/9885098850
-- Zap/29-1 is proceeding passing it to Local/8600069@default-42a8,1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [58600069@default:3] MeetMe("Local/58600069@default-3dd8,2", "58600069|Fmq") in new stack
> Channel Local/58600069@default-3dd8,1 was answered.
== Parsing '/etc/asterisk/meetme.conf': Found
-- Executing [8309@default:1] Answer("Local/58600069@default-3dd8,1", "") in new stack
-- Executing [8309@default:2] Set("Local/58600069@default-3dd8,1", "FILENAME=") in new stack
-- Executing [8309@default:3] NoOp("Local/58600069@default-3dd8,1", "") in new stack
-- Executing [8309@default:4] AGI("Local/58600069@default-3dd8,1", "agi://localhost/UpdateRecordings.agi") in new stack
-- <Local/58600069@default-3dd8,2> Playing 'conf-invalid' (language 'en')
-- Zap/29-1 is making progress passing it to Local/8600069@default-42a8,1
-- Zap/29-1 is making progress passing it to Local/8600069@default-42a8,1
-- Zap/29-1 is ringing


Failure Case:

With the same configuration if I dial the same number after some time it is not routing to first meetme extension in the default context.
My number is not at all routing to asterisk .


Please help me on this. Your help will be very much appreciated.

Thanks,
Srinivas

[williamconley]

If you dial the wrong number (by accident) by missing a digit, it may never show in the CLI.

Try adding this to your dialplan:

Code: Select all

exten => s,1,AGI(agi://127.0.0.1:4577/call_log)
exten => s,n,Hangup

This should catch any misdials, show them on the screen, and kill the call.

What is your method of manual dial? Direct from the SIP phone or in the Vicidial Agent Screen's Manual Dial interface?

[Srinivasa Rao]

Sorry for the delay in update.

We have identified the root cause for this issue. We have implemented our solution in a cluster mode with two servers. So the issue is, we are using the manual dial option from the agent interface sometimes the call is reaching to asterisk and sometimes call is not reaching to asterisk(Nothing is showing in asterisk console).

Root cause we have identified is, We have two servers which are running in active-passive mode. So when we dial out a number from agent desk, the perl script which is running from my secondary server updating the status to db.(AST_manager_send.pl, AST_manager_listen.pl) due to this my calls are not routing to asterisk. I have commented out the required pl's in secondary server cron tab and killed the respective processes from operating system, my issue has been resolved.

I just want to update this, as this may help to someone in future.

Thank you.

[williamconley]

I'm not sure I understood that, but it's good that you updated (even a bit late) so the next user has some help.

I even applaud you for the effort if the only reason you did it is so you can use this forum as a handy storage locker for your support solutions for Vicidial (for easy reference, LOL).