Unable to transfer live call to external

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Unable to transfer live call to external

Postby kein » Tue Oct 29, 2013 10:41 pm

Good day all, first time poster here.

I'm having a bit of a problem getting live call transfers to work, and I think it's my dial plan that's not working right - not entirely sure on this though.

What I'm doing:
1. Agent logs in and begins outbound campaign.
2. Agent attends to call, needs to transfer call to external party.
3. Agent transfers call using TRANSFER-CONF button
4. Agent uses Consultative and Dial Override, enters phone number in "NUMBER TO CALL" field, dials third party using "DIAL WITH CUSTOMER" button.

I'm assuming the conference call is supposed to be made to the number in the "NUMBER TO CALL" field at this point, but it doesn't happen.

This is the Asterisk CLI output. 04YYYYYYYY is the customer's number, 04XXXXXXXX is the external number I'm trying to transfer to.

Code: Select all
[Oct 30 09:34:01]     -- Executing [90009**CXFER*36819**04YYYYYYYY*james*04XXXXXXXX*10*@default:1] Answer("Local/90009**CXFER*36819**04YYYYYYYY*james*04XXXXXXXX*10*@default-427", "") in new stack
[Oct 30 09:34:01]        > Channel Local/90009**CXFER*36819**04YYYYYYYY*james*04XXXXXXXX*10*@default-4270,1 was answered.
[Oct 30 09:34:01]     -- Executing [90009**CXFER*36819**04YYYYYYYY*james*04XXXXXXXX*10*@default:2] Dial("Local/90009**CXFER*36819**04YYYYYYYY*james*04XXXXXXXX*10*@default-427", "IAX2/ASTloop:direct@127.0.0.1:40569/990009**CXFER*36819**04YYYYYYYY*james*04XXXXXXXX*10*||to") in new stack
[Oct 30 09:34:01]     -- Called ASTloop:direct@127.0.0.1:40569/990009**CXFER*36819**04YYYYYYYY*james*04XXXXXXXX*10*
[Oct 30 09:34:01]     -- Executing [8600052@default:1] MeetMe("Local/90009**CXFER*36819**04YYYYYYYY*james*04XXXXXXXX*10*@default-427", "8600052|F") in new stack
[Oct 30 09:34:01]     -- Accepting AUTHENTICATED call from 127.0.0.1:
       > requested format = gsm,
       > requested prefs = (gsm|ulaw),
       > actual format = ulaw,
       > host prefs = (ulaw),
       > priority = mine
[Oct 30 09:34:01]     -- Executing [990009**CXFER*36819**04YYYYYYYY*james*04XXXXXXXX*10*@default:1] Answer("IAX2/ASTloop-926", "") in new stack
[Oct 30 09:34:01]     -- Executing [990009**CXFER*36819**04YYYYYYYY*james*04XXXXXXXX*10*@default:2] AGI("IAX2/ASTloop-926", "agi-VDAD_ALL_inbound.agi|CLOSER-----LB-----CL_TESTCAMP-----7275551212-----Closer-----park----------999-----1") in new stack
[Oct 30 09:34:01]     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Oct 30 09:34:01]     -- Call accepted by 127.0.0.1 (format ulaw)
[Oct 30 09:34:01]     -- Format for call is ulaw
[Oct 30 09:34:01]     -- IAX2/127.0.0.1:40569-8352 answered Local/90009**CXFER*36819**04YYYYYYYY*james*04XXXXXXXX*10*@default-4270,2
[Oct 30 09:34:01]   == Parsing '/etc/asterisk/manager.conf': [Oct 30 09:34:01] Found
[Oct 30 09:34:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct 30 09:34:01]   == Parsing '/etc/asterisk/manager.conf': [Oct 30 09:34:01] Found
[Oct 30 09:34:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct 30 09:34:01]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct 30 09:34:01]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Oct 30 09:34:01]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Oct 30 09:34:01]     -- Executing [h@default:1] DeadAGI("Local/90009**CXFER*36819**04YYYYYYYY*james*04XXXXXXXX*10*@default-427", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----0-----0") in new stack
[Oct 30 09:34:01]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----0-----0 completed, returning 0
[Oct 30 09:34:01]   == Spawn extension (default, 90009**CXFER*36819**04YYYYYYYY*james*04XXXXXXXX*10*, 2) exited non-zero on 'Local/90009**CXFER*36819**04YYYYYYYY*james*04XXXXXXXX*10*@default-4270,2'
[Oct 30 09:34:01]     -- AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
[Oct 30 09:34:01]     -- Executing [990009**CXFER*36819**04YYYYYYYY*james*04XXXXXXXX*10*@default:3] Hangup("IAX2/ASTloop-926", "") in new stack
[Oct 30 09:34:01]   == Spawn extension (default, 990009**CXFER*36819**04YYYYYYYY*james*04XXXXXXXX*10*, 3) exited non-zero on 'IAX2/ASTloop-926'
[Oct 30 09:34:01]     -- Executing [h@default:1] DeadAGI("IAX2/ASTloop-926", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Oct 30 09:34:01]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Oct 30 09:34:01]     -- Hungup 'IAX2/ASTloop-926'
[Oct 30 09:34:01]   == Spawn extension (default, 8600052, 1) exited non-zero on 'IAX2/127.0.0.1:40569-8352'
[Oct 30 09:34:01]     -- Executing [h@default:1] DeadAGI("IAX2/127.0.0.1:40569-8352", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Oct 30 09:34:01]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Oct 30 09:34:01]     -- Hungup 'IAX2/127.0.0.1:40569-8352'


If anyone has any idea on what I'm missing out at all, comments would be very welcome. I've looked at the Vicidial manager manual, read through most of the threads here and I'm not coming up with any fresh ideas on things I should be looking at, have to admit I'm quite raw with Asterisk and Vicidial. Thanks in advance everyone cheers.
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Re: Unable to transfer live call to external

Postby williamconley » Sat Nov 02, 2013 8:30 pm

There is an exercise for transfer of calls in the Vicidial Manager's Manual. You may want to try it from beginning to end (not looking through, but actually performing the exercise if there is one for external transfers).

Do not push "consultative" or "dial override". Avoid touching the local closer button or the dropdown next to it.

Just push transfer-conf, put in a 10 digit us Phone number and then hit either dial with customer or blind transfer (or park customer dial). It will dial using the manual dial prefix for the campaign (so be sure that has been set to the proper carrier).

And test it with a manual dial or call to your own cell phone with no other traffic on the dialer to be sure you are not rushed and looking only at debug output from this attempt.
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