Setup Dial Agent Extension at Call Menu

All installation and configuration problems and questions

Moderators: gerski, enjay, williamconley, Op3r, Staydog, gardo, mflorell, MJCoate, mcargile, Kumba, Michael_N

Setup Dial Agent Extension at Call Menu

Postby hiadviser » Thu Jan 19, 2017 11:09 pm

Hey everyone,

Having some trouble setting up dialing an agent extension at the call menu.


I have a call menu setup (with no options right now), with this in the custom dialplan:
exten => _XXXX,1,Playback(/var/lib/asterisk/sounds/beep)
exten => _XXXX,1,AGI(agi-AGENT_route.agi,default---AGENTDIRECT---ACTIVE)

And yes, I have enabled Allow Custom Dialplans in System settings.


My call menu picks up, but when I dial the extension it plays me a beep and then hangs up. The phone is set to On-Hook Agent so I am not logged in to the interface. I also have that extension setup as a remote agent.

Here is my cli:

[Jan 19 21:57:48] -- Executing [111111111@trunkinbound:1] AGI("SIP/vitel-inbound2-00000019", "agi-DID_route.agi") in new stack
[Jan 19 21:57:48] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Jan 19 21:57:48] -- <SIP/vitel-inbound2-00000019>AGI Script agi-DID_route.agi completed, returning 0
[Jan 19 21:57:48] -- Executing [s@Main:1] Answer("SIP/vitel-inbound2-00000019", "") in new stack
[Jan 19 21:57:49] -- Executing [s@Main:2] AGI("SIP/vitel-inbound2-00000019", "agi-VDAD_inbound_calltime_check.agi,CALLMENU-----YES-----Main-------------------------NO") in new stack
[Jan 19 21:57:49] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_inbound_calltime_check.agi
[Jan 19 21:57:49] -- <SIP/vitel-inbound2-00000019> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Jan 19 21:57:49] -- <SIP/vitel-inbound2-00000019> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Jan 19 21:57:49] -- <SIP/vitel-inbound2-00000019>AGI Script agi-VDAD_inbound_calltime_check.agi completed, returning 0
[Jan 19 21:57:49] -- Executing [s@Main:3] Set("SIP/vitel-inbound2-00000019", "INVCOUNT=0") in new stack
[Jan 19 21:57:49] -- Executing [s@Main:4] BackGround("SIP/vitel-inbound2-00000019", "welcome") in new stack
[Jan 19 21:57:49] -- <SIP/vitel-inbound2-00000019> Playing 'welcome.gsm' (language 'en')
[Jan 19 21:57:50] -- Executing [s@Main:5] BackGround("SIP/vitel-inbound2-00000019", "for-sales") in new stack
[Jan 19 21:57:50] -- <SIP/vitel-inbound2-00000019> Playing 'for-sales.gsm' (language 'en')
[Jan 19 21:57:51] -- Executing [s@Main:6] BackGround("SIP/vitel-inbound2-00000019", "press") in new stack
[Jan 19 21:57:51] -- <SIP/vitel-inbound2-00000019> Playing 'press.gsm' (language 'en')
[Jan 19 21:57:51] -- Executing [s@Main:7] BackGround("SIP/vitel-inbound2-00000019", "./digits/1") in new stack
[Jan 19 21:57:51] -- <SIP/vitel-inbound2-00000019> Playing './digits/1.gsm' (language 'en')
[Jan 19 21:57:52] -- Executing [s@Main:8] BackGround("SIP/vitel-inbound2-00000019", "for-tech-support") in new stack
[Jan 19 21:57:52] -- <SIP/vitel-inbound2-00000019> Playing 'for-tech-support.gsm' (language 'en')
[Jan 19 21:57:53] -- Executing [s@Main:9] BackGround("SIP/vitel-inbound2-00000019", "press") in new stack
[Jan 19 21:57:53] -- <SIP/vitel-inbound2-00000019> Playing 'press.gsm' (language 'en')
[Jan 19 21:57:54] -- Executing [s@Main:10] BackGround("SIP/vitel-inbound2-00000019", "./digits/2") in new stack
[Jan 19 21:57:54] -- <SIP/vitel-inbound2-00000019> Playing './digits/2.gsm' (language 'en')
[Jan 19 21:57:55] -- Executing [s@Main:11] BackGround("SIP/vitel-inbound2-00000019", "silence") in new stack
[Jan 19 21:57:55] -- <SIP/vitel-inbound2-00000019> Playing 'silence.slin' (language 'en')
[Jan 19 21:57:55] DTMF[23053][C-0000001e]: channel.c:4215 __ast_read: DTMF begin '1' received on SIP/vitel-inbound2-00000019
[Jan 19 21:57:55] DTMF[23053][C-0000001e]: channel.c:4219 __ast_read: DTMF begin ignored '1' on SIP/vitel-inbound2-00000019
[Jan 19 21:57:55] -- Executing [s@Main:12] BackGround("SIP/vitel-inbound2-00000019", "if-u-know-ext-dial") in new stack
[Jan 19 21:57:55] -- <SIP/vitel-inbound2-00000019> Playing 'if-u-know-ext-dial.gsm' (language 'en')
[Jan 19 21:57:55] DTMF[23053][C-0000001e]: channel.c:4129 __ast_read: DTMF end '1' received on SIP/vitel-inbound2-00000019, duration 190 ms
[Jan 19 21:57:55] DTMF[23053][C-0000001e]: channel.c:4199 __ast_read: DTMF end passthrough '1' on SIP/vitel-inbound2-00000019
[Jan 19 21:57:55] DTMF[23053][C-0000001e]: channel.c:4215 __ast_read: DTMF begin '0' received on SIP/vitel-inbound2-00000019
[Jan 19 21:57:55] DTMF[23053][C-0000001e]: channel.c:4219 __ast_read: DTMF begin ignored '0' on SIP/vitel-inbound2-00000019
[Jan 19 21:57:56] DTMF[23053][C-0000001e]: channel.c:4129 __ast_read: DTMF end '0' received on SIP/vitel-inbound2-00000019, duration 190 ms
[Jan 19 21:57:56] DTMF[23053][C-0000001e]: channel.c:4199 __ast_read: DTMF end passthrough '0' on SIP/vitel-inbound2-00000019
[Jan 19 21:57:56] DTMF[23053][C-0000001e]: channel.c:4215 __ast_read: DTMF begin '1' received on SIP/vitel-inbound2-00000019
[Jan 19 21:57:56] DTMF[23053][C-0000001e]: channel.c:4219 __ast_read: DTMF begin ignored '1' on SIP/vitel-inbound2-00000019
[Jan 19 21:57:56] DTMF[23053][C-0000001e]: channel.c:4129 __ast_read: DTMF end '1' received on SIP/vitel-inbound2-00000019, duration 190 ms
[Jan 19 21:57:56] DTMF[23053][C-0000001e]: channel.c:4199 __ast_read: DTMF end passthrough '1' on SIP/vitel-inbound2-00000019
[Jan 19 21:57:56] DTMF[23053][C-0000001e]: channel.c:4215 __ast_read: DTMF begin '8' received on SIP/vitel-inbound2-00000019
[Jan 19 21:57:56] DTMF[23053][C-0000001e]: channel.c:4219 __ast_read: DTMF begin ignored '8' on SIP/vitel-inbound2-00000019
[Jan 19 21:57:56] DTMF[23053][C-0000001e]: channel.c:4129 __ast_read: DTMF end '8' received on SIP/vitel-inbound2-00000019, duration 190 ms
[Jan 19 21:57:56] DTMF[23053][C-0000001e]: channel.c:4199 __ast_read: DTMF end passthrough '8' on SIP/vitel-inbound2-00000019
[Jan 19 21:57:56] == CDR updated on SIP/vitel-inbound2-00000019
[Jan 19 21:57:56] -- Executing [1018@Main:1] Playback("SIP/vitel-inbound2-00000019", "/var/lib/asterisk/sounds/beep") in new stack
[Jan 19 21:57:56] -- <SIP/vitel-inbound2-00000019> Playing '/var/lib/asterisk/sounds/beep.gsm' (language 'en')
[Jan 19 21:57:57] -- Auto fallthrough, channel 'SIP/vitel-inbound2-00000019' status is 'UNKNOWN'
[Jan 19 21:57:57] -- Executing [h@Main:1] AGI("SIP/vitel-inbound2-00000019", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jan 19 21:57:57] -- <SIP/vitel-inbound2-00000019>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jan 19 21:58:01] -- Remote UNIX connection


I'm not sure what the problem. Any help is appreciated.
ViciDial VERSION: 2.14-583a | BUILD: 161226-2224 | Asterisk 11.22.0-vici | Cluster Setp - 3 Web, 1 DB| No Digium/Sangoma Hardware | No Extra Software | HP ProLiant DL360 G6, 48GB RAM, 16 x Intel Xeon X5550 2.67GHz, 480GB SSD
hiadviser
 
Posts: 71
Joined: Wed Dec 05, 2012 9:29 am

Re: Setup Dial Agent Extension at Call Menu

Postby mflorell » Thu Jan 19, 2017 11:27 pm

You gave the second line a priority of "1", it should be either "n" or "2".
mflorell
Site Admin
 
Posts: 18339
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida

Re: Setup Dial Agent Extension at Call Menu

Postby hiadviser » Thu Jan 19, 2017 11:32 pm

Ok setting it to 2 worked...at least it sent me to the hold music.

It never reached the phone though.

[Jan 19 22:31:16] -- <SIP/vitel-inbound2-00000022> Playing './digits/1.gsm' (language 'en')
[Jan 19 22:31:17] DTMF[27064][C-00000027]: channel.c:4215 __ast_read: DTMF begin '1' received on SIP/vitel-inbound2-00000022
[Jan 19 22:31:17] DTMF[27064][C-00000027]: channel.c:4219 __ast_read: DTMF begin ignored '1' on SIP/vitel-inbound2-00000022
[Jan 19 22:31:17] DTMF[27064][C-00000027]: channel.c:4129 __ast_read: DTMF end '1' received on SIP/vitel-inbound2-00000022, duration 190 ms
[Jan 19 22:31:17] DTMF[27064][C-00000027]: channel.c:4199 __ast_read: DTMF end passthrough '1' on SIP/vitel-inbound2-00000022
[Jan 19 22:31:17] DTMF[27064][C-00000027]: channel.c:4215 __ast_read: DTMF begin '0' received on SIP/vitel-inbound2-00000022
[Jan 19 22:31:17] DTMF[27064][C-00000027]: channel.c:4219 __ast_read: DTMF begin ignored '0' on SIP/vitel-inbound2-00000022
[Jan 19 22:31:17] DTMF[27064][C-00000027]: channel.c:4129 __ast_read: DTMF end '0' received on SIP/vitel-inbound2-00000022, duration 190 ms
[Jan 19 22:31:17] DTMF[27064][C-00000027]: channel.c:4199 __ast_read: DTMF end passthrough '0' on SIP/vitel-inbound2-00000022
[Jan 19 22:31:18] DTMF[27064][C-00000027]: channel.c:4215 __ast_read: DTMF begin '1' received on SIP/vitel-inbound2-00000022
[Jan 19 22:31:18] DTMF[27064][C-00000027]: channel.c:4219 __ast_read: DTMF begin ignored '1' on SIP/vitel-inbound2-00000022
[Jan 19 22:31:18] DTMF[27064][C-00000027]: channel.c:4129 __ast_read: DTMF end '1' received on SIP/vitel-inbound2-00000022, duration 190 ms
[Jan 19 22:31:18] DTMF[27064][C-00000027]: channel.c:4199 __ast_read: DTMF end passthrough '1' on SIP/vitel-inbound2-00000022
[Jan 19 22:31:18] DTMF[27064][C-00000027]: channel.c:4215 __ast_read: DTMF begin '8' received on SIP/vitel-inbound2-00000022
[Jan 19 22:31:18] DTMF[27064][C-00000027]: channel.c:4219 __ast_read: DTMF begin ignored '8' on SIP/vitel-inbound2-00000022
[Jan 19 22:31:18] DTMF[27064][C-00000027]: channel.c:4129 __ast_read: DTMF end '8' received on SIP/vitel-inbound2-00000022, duration 190 ms
[Jan 19 22:31:18] DTMF[27064][C-00000027]: channel.c:4199 __ast_read: DTMF end passthrough '8' on SIP/vitel-inbound2-00000022
[Jan 19 22:31:18] == CDR updated on SIP/vitel-inbound2-00000022
[Jan 19 22:31:18] -- Executing [1018@Main:1] Playback("SIP/vitel-inbound2-00000022", "/var/lib/asterisk/sounds/beep") in new stack
[Jan 19 22:31:18] -- <SIP/vitel-inbound2-00000022> Playing '/var/lib/asterisk/sounds/beep.gsm' (language 'en')
[Jan 19 22:31:19] -- Executing [1018@Main:2] AGI("SIP/vitel-inbound2-00000022", "agi-AGENT_route.agi,default---AGENTDIRECT---ACTIVE") in new stack
[Jan 19 22:31:19] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-AGENT_route.agi
[Jan 19 22:31:19] -- <SIP/vitel-inbound2-00000022>AGI Script agi-AGENT_route.agi completed, returning 0
[Jan 19 22:31:19] -- Executing [99909*1*AGENTDIRECT*1018*@default:1] Answer("SIP/vitel-inbound2-00000022", "") in new stack
[Jan 19 22:31:19] -- Executing [99909*1*AGENTDIRECT*1018*@default:2] AGI("SIP/vitel-inbound2-00000022", "agi-VDAD_ALL_inbound.agi") in new stack
[Jan 19 22:31:19] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Jan 19 22:31:19] -- <SIP/vitel-inbound2-00000022> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Jan 19 22:31:19] -- <SIP/vitel-inbound2-00000022> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Jan 19 22:31:21] -- Started music on hold, class 'default', on SIP/vitel-inbound2-00000022
[Jan 19 22:31:24] -- Stopped music on hold on SIP/vitel-inbound2-00000022
[Jan 19 22:31:24] -- <SIP/vitel-inbound2-00000022> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Jan 19 22:31:24] -- <SIP/vitel-inbound2-00000022> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Jan 19 22:31:24] -- <SIP/vitel-inbound2-00000022> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Jan 19 22:31:24] -- <SIP/vitel-inbound2-00000022> Playing 'generic_hold.gsm' (escape_digits=) (sample_offset 0) (language 'en')
ViciDial VERSION: 2.14-583a | BUILD: 161226-2224 | Asterisk 11.22.0-vici | Cluster Setp - 3 Web, 1 DB| No Digium/Sangoma Hardware | No Extra Software | HP ProLiant DL360 G6, 48GB RAM, 16 x Intel Xeon X5550 2.67GHz, 480GB SSD
hiadviser
 
Posts: 71
Joined: Wed Dec 05, 2012 9:29 am

Re: Setup Dial Agent Extension at Call Menu

Postby mflorell » Fri Jan 20, 2017 6:34 am

The call went into the queue for that one agent, it's not going to ring a phone, it's going to go to their logged-in VICIdial agent screen session.
mflorell
Site Admin
 
Posts: 18339
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida

Re: Setup Dial Agent Extension at Call Menu

Postby carpenox » Thu Oct 28, 2021 10:41 am

try following my latest article for this:

https://cyburdial.net/how-to-create-a-c ... ension-on/


Chris
Alma Linux 9.3 | SVN Version: 3822 | DB Schema Version: 1711 | Asterisk 18.18.1
www.dialer.one -:- 1-833-DIALER-1 -:- https://linktr.ee/CyburDial -:- WhatsApp: +19549477572 -:- Skype: live:carpenox_3 | Discord: https://discord.gg/DVktk6smbh
carpenox
 
Posts: 2250
Joined: Wed Apr 08, 2020 2:02 am
Location: St Petersburg, FL


Return to Support

Who is online

Users browsing this forum: No registered users and 127 guests