Unavailable Dialplan Forward

[iboam]

I have setup the Unavailable Dialplan Forward for agents that are not available to take the call but caller id is the Outbound CallerID on the phone settings not the caller id of person that it's actually calling. Even when send voicemail to email to a non available agent the caller id is set to Phone Outbound CallerID

There is any way to fix this ??

[mflorell]

Please post an example with the call flow so we can better understand what is happening.

[Noah]

If you are using raw dialplan, you should verify you are getting both the DID number and the callerid number, then set that callerid number if you have to in the outbound dialplan.

Example string for asterisk to determine what the DID is and the Callerid Number
exten => _X.,n,noop("this is the DID" ${CALLERID(dnid)} "this is callerid number" ${CALLERID(num)}" )

[iboam]

I need to include (if you know the extension of the party you are trying to reach) before use the Unavailable Dialplan Forward
I already changed the field “Allow Custom Dialplan Entries” to “1” then click SUBMIT and Scroll to the bottom of the screen of the call menu and entered the following into the “Custom Dialplan Entry” field:
exten => _XXXX,1,Playback(/var/lib/asterisk/sounds/beep)
exten => _XXXX,x,AGI(agi-AGENT_route.agi,default---AGENTDIRECT---ACTIVE)

but it's not working ....

[mflorell]

What does Asterisk CLI output look like when you try to use that?

[iboam]

Code: Select all

[Nov 11 15:03:33]   == WebSocket connection from '104.7.46.160:50371' for protocol 'sip' accepted using version '13'
[Nov 11 15:03:33] DTMF[23906][C-0000006f]: channel.c:4215 __ast_read: DTMF begin '2' received on SIP/HOSTEDPBX-0000001c
[Nov 11 15:03:33] DTMF[23906][C-0000006f]: channel.c:4219 __ast_read: DTMF begin ignored '2' on SIP/HOSTEDPBX-0000001c
[Nov 11 15:03:33]     -- Registered SIP '2000' at 104.7.46.160:50371
[Nov 11 15:03:33] NOTICE[23988]: chan_sip.c:23861 handle_response_peerpoke: Peer '2000' is now Reachable. (84ms / 2000ms)
[Nov 11 15:03:33] DTMF[23906][C-0000006f]: channel.c:4129 __ast_read: DTMF end '2' received on SIP/HOSTEDPBX-0000001c, duration 395 ms
[Nov 11 15:03:33] DTMF[23906][C-0000006f]: channel.c:4199 __ast_read: DTMF end passthrough '2' on SIP/HOSTEDPBX-0000001c
[Nov 11 15:03:34] DTMF[23906][C-0000006f]: channel.c:4215 __ast_read: DTMF begin '0' received on SIP/HOSTEDPBX-0000001c
[Nov 11 15:03:34] DTMF[23906][C-0000006f]: channel.c:4219 __ast_read: DTMF begin ignored '0' on SIP/HOSTEDPBX-0000001c
[Nov 11 15:03:34] DTMF[23906][C-0000006f]: channel.c:4129 __ast_read: DTMF end '0' received on SIP/HOSTEDPBX-0000001c, duration 315 ms
[Nov 11 15:03:34] DTMF[23906][C-0000006f]: channel.c:4199 __ast_read: DTMF end passthrough '0' on SIP/HOSTEDPBX-0000001c
[Nov 11 15:03:34] DTMF[23906][C-0000006f]: channel.c:4215 __ast_read: DTMF begin '0' received on SIP/HOSTEDPBX-0000001c
[Nov 11 15:03:34] DTMF[23906][C-0000006f]: channel.c:4219 __ast_read: DTMF begin ignored '0' on SIP/HOSTEDPBX-0000001c
[Nov 11 15:03:34] DTMF[23906][C-0000006f]: channel.c:4129 __ast_read: DTMF end '0' received on SIP/HOSTEDPBX-0000001c, duration 235 ms
[Nov 11 15:03:34] DTMF[23906][C-0000006f]: channel.c:4199 __ast_read: DTMF end passthrough '0' on SIP/HOSTEDPBX-0000001c
[Nov 11 15:03:35] DTMF[23906][C-0000006f]: channel.c:4215 __ast_read: DTMF begin '0' received on SIP/HOSTEDPBX-0000001c
[Nov 11 15:03:35] DTMF[23906][C-0000006f]: channel.c:4219 __ast_read: DTMF begin ignored '0' on SIP/HOSTEDPBX-0000001c
[Nov 11 15:03:35] DTMF[23906][C-0000006f]: channel.c:4129 __ast_read: DTMF end '0' received on SIP/HOSTEDPBX-0000001c, duration 235 ms
[Nov 11 15:03:35] DTMF[23906][C-0000006f]: channel.c:4199 __ast_read: DTMF end passthrough '0' on SIP/HOSTEDPBX-0000001c
[Nov 11 15:03:35]   == CDR updated on SIP/HOSTEDPBX-0000001c
[Nov 11 15:03:35]     -- Executing [2000@CALLMENU_CLOUDPBX:1] SayDigits("SIP/HOSTEDPBX-0000001c", "2000") in new stack
[Nov 11 15:03:35]     -- <SIP/HOSTEDPBX-0000001c> Playing 'digits/2.gsm' (language 'en')
[Nov 11 15:03:35]     -- <SIP/HOSTEDPBX-0000001c> Playing 'digits/0.gsm' (language 'en')
[Nov 11 15:03:36]     -- <SIP/HOSTEDPBX-0000001c> Playing 'digits/0.gsm' (language 'en')
[Nov 11 15:03:37]     -- <SIP/HOSTEDPBX-0000001c> Playing 'digits/0.gsm' (language 'en')
[Nov 11 15:03:38]   == Manager 'sendcron' logged on from 127.0.0.1
[Nov 11 15:03:38]   == Using SIP RTP CoS mark 5
[Nov 11 15:03:38]        > Channel SIP/2000-0000001d was answered
[Nov 11 15:03:38]     -- Executing [8600051@default:1] MeetMe("SIP/2000-0000001d", "8600051,F") in new stack
[Nov 11 15:03:38]   == Parsing '/etc/asterisk/meetme.conf': Found
[Nov 11 15:03:38]   == Parsing '/etc/asterisk/meetme-vicidial.conf': Found
[Nov 11 15:03:38]     -- Created MeetMe conference 1023 for conference '8600051'
[Nov 11 15:03:38]     -- <SIP/2000-0000001d> Playing 'conf-onlyperson.gsm' (language 'en')
[Nov 11 15:03:38]     -- Executing [2000@CALLMENU_CLOUDPBX:2] Goto("SIP/HOSTEDPBX-0000001c", "trunkinbound,2000,1") in new stack
[Nov 11 15:03:38]     -- Goto (trunkinbound,2000,1)
[Nov 11 15:03:38]     -- Executing [2000@trunkinbound:1] AGI("SIP/HOSTEDPBX-0000001c", "agi-DID_route.agi") in new stack
[Nov 11 15:03:38]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Nov 11 15:03:38]        > 0x2041c90 -- Probation passed - setting RTP source address to 104.7.46.160:52320
[Nov 11 15:03:38]     -- <SIP/HOSTEDPBX-0000001c>AGI Script agi-DID_route.agi completed, returning 0
[Nov 11 15:03:38]     -- Executing [9998811112@default:1] Wait("SIP/HOSTEDPBX-0000001c", "2") in new stack
[Nov 11 15:03:39]   == Manager 'sendcron' logged off from 127.0.0.1
[Nov 11 15:03:40]     -- Executing [9998811112@default:2] Answer("SIP/HOSTEDPBX-0000001c", "") in new stack
[Nov 11 15:03:40]     -- Executing [9998811112@default:3] Playback("SIP/HOSTEDPBX-0000001c", "ss-noservice") in new stack
[Nov 11 15:03:40]     -- <SIP/HOSTEDPBX-0000001c> Playing 'ss-noservice.gsm' (language 'en')
[Nov 11 15:03:45]     -- Executing [9998811112@default:4] Playback("SIP/HOSTEDPBX-0000001c", "vm-goodbye") in new stack
[Nov 11 15:03:45]     -- <SIP/HOSTEDPBX-0000001c> Playing 'vm-goodbye.gsm' (language 'en')
[Nov 11 15:03:46]     -- Executing [9998811112@default:5] Hangup("SIP/HOSTEDPBX-0000001c", "") in new stack
[Nov 11 15:03:46]   == Spawn extension (default, 9998811112, 5) exited non-zero on 'SIP/HOSTEDPBX-0000001c'
[Nov 11 15:03:46]     -- Executing [h@default:1] AGI("SIP/HOSTEDPBX-0000001c", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Nov 11 15:03:46]     -- <SIP/HOSTEDPBX-0000001c>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
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