Redirect IB call to VOICEMAIL via IB DID

[donX]

Hi,

VERSION: 2.12-552a
BUILD: 160429-0835

Setup:

I have a blended campaign (INBOUND_MAN).
Inbound DID route is set to AGENT.
User Unavailable is set to VOICEMAIL.
VOICEMAIL # is the VOICEMAIL BOX set in the PHONE.

Dialplan:
exten => XXXXXXXXXX,1,Ringing
exten => XXXXXXXXXX,2,Wait(1)
exten => XXXXXXXXXX,3,Answer
exten => XXXXXXXXXX,4,AGI(agi://127.0.0.1:4577/call_log)
exten => XXXXXXXXXX,5,AGI(agi-DID_route.agi)
exten => XXXXXXXXXX,6,Hangup

So, the idea obviously is when someone calls the DID it will then get rerouted to the agent automatically.

Issue:
The IB is working fine and calls are coming in, however, when the agent is PAUSED or not LOGGED in and you try to call the DID it gets stuck playing the WAIT MUSIC and does not offer VOICEMAIL OPTIONS.

Here's the TRACE:

Code: Select all

[Aug 25 04:16:33]   == Using SIP RTP CoS mark 5
[Aug 25 04:16:33]     -- Executing [8643495683@default:1] Ringing("SIP/WadeAst1016525-00000381", "") in new stack
[Aug 25 04:16:33]     -- Executing [8643495683@default:2] Wait("SIP/WadeAst1016525-00000381", "1") in new stack
[Aug 25 04:16:34]     -- Executing [8643495683@default:3] Answer("SIP/WadeAst1016525-00000381", "") in new stack
[Aug 25 04:16:34]     -- Executing [8643495683@default:4] AGI("SIP/WadeAst1016525-00000381", "agi-DID_route.agi") in new stack
[Aug 25 04:16:34]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Aug 25 04:16:34]     -- AGI Script Executing Application: (Monitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20180825041634_8643495683_357426)
[Aug 25 04:16:34] WARNING[19578][C-000005a9]: pbx.c:1675 pbx_exec: The application delimiter is now the comma, not the pipe.  Did you forget to convert your dialplan?  (Monitor(wav|/var/spool/asterisk/monitor/MIX/20180825041634_8643495683_357426))
[Aug 25 04:16:34] WARNING[19578][C-000005a9]: file.c:1242 ast_writefile: No such format 'wav|/var/spool/asterisk/monitor/MIX/20180825041634_8643495683_357426'
[Aug 25 04:16:34] WARNING[19578][C-000005a9]: res_monitor.c:363 __ast_monitor_start: Could not create file /var/spool/asterisk/monitor/audio-in-56
[Aug 25 04:16:34] WARNING[19578][C-000005a9]: res_monitor.c:571 __ast_monitor_change_fname: Cannot change monitor filename of channel SIP/WadeAst1016525-00000381 to null
[Aug 25 04:16:34]     -- <SIP/WadeAst1016525-00000381>AGI Script agi-DID_route.agi completed, returning 0
[Aug 25 04:16:34]     -- Executing [99909*2***DID@default:1] Answer("SIP/WadeAst1016525-00000381", "") in new stack
[Aug 25 04:16:34]     -- Executing [99909*2***DID@default:2] AGI("SIP/WadeAst1016525-00000381", "agi-VDAD_ALL_inbound.agi") in new stack
[Aug 25 04:16:34]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Aug 25 04:16:34]        > 0x7f5bb8248f00 -- Probation passed - setting RTP source address to 10.1.65.25:15834
[Aug 25 04:16:34]     -- <SIP/WadeAst1016525-00000381> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Aug 25 04:16:34]     -- <SIP/WadeAst1016525-00000381> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Aug 25 04:16:36]     -- Started music on hold, class 'default', on SIP/WadeAst1016525-00000381
[Aug 25 04:16:39]     -- Stopped music on hold on SIP/WadeAst1016525-00000381
[Aug 25 04:16:39]     -- <SIP/WadeAst1016525-00000381> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Aug 25 04:16:39]     -- <SIP/WadeAst1016525-00000381> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Aug 25 04:16:39]     -- <SIP/WadeAst1016525-00000381> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Aug 25 04:16:39]     -- <SIP/WadeAst1016525-00000381> Playing 'generic_hold.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Aug 25 04:16:45]     -- Started music on hold, class 'default', on SIP/WadeAst1016525-00000381
[Aug 25 04:16:58]     -- Stopped music on hold on SIP/WadeAst1016525-00000381


It goes in a loop through WAIT MUSIC.

How do I instruct vicidial to redirect to voicemail if no agent available?

PLEASE HELP!

[donX]

This is now working. I just removed the In-group ID from the IB DID. Changed the In-group WAIT TIME to VM and set the audio file to vm-hangup,digits/1.

Is there a way to customize VM message per agent tho? cause it seems that the audio file for VM is looking at the VM box set in the Phone extension.

Is there a way to edit that?

[donX]

By the way, is there a way to customize messages for each agent's voicemail?

[williamconley]

Voicemail is an asterisk module. Of course it can be customized per user. Log in to the VM module as each user to record a new message for each. Of course, there are other methods to manage this without directly accessing the module by logging in, but that's an Asterisk question, outside the realm of Vicidial unless you want to have someone add an "Unavailable" message option to the Vicidial Voicemail modify page under "Admin->Voicemail". Could be done. Nobody's paid for it yet. I think FreePBX can do it, though, so the code exists outside Vicidial.