vicidial.org

Hi All,

Is anybody know with the callmenu "If you know your parties extension please dial it now" process? I already followed with this agi "agi-AGENT_route.agi" and applied in callmenu but the AGENTDIRECT wont go through on the agents screen it will just stock on the onhold music even though the agent is already in READY status and selected the AGENTDIRECT queue.

Please let me know what i missing on this. Thanks


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Callmenu


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Asterisk logs:
May 20 11:08:05] DTMF[1198][C-00000052]: channel.c:4108 __ast_read: DTMF end passthrough '1' on SIP/microtalk-00000035
[May 20 11:08:05] -- Executing [101@IVRparchment:1] AGI("SIP/microtalk-00000035", "agi-AGENT_route.agi,default---AGENTDIRECT---ACTIVE") in new stack
[May 20 11:08:05] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-AGENT_route.agi
[May 20 11:08:05] -- <SIP/microtalk-00000035>AGI Script agi-AGENT_route.agi completed, returning 0
[May 20 11:08:05] -- Executing [99909**AGENTDIRECT*101*@default:1] Answer("SIP/microtalk-00000035", "") in new stack
[May 20 11:08:05] -- Executing [99909**AGENTDIRECT*101*@default:2] AGI("SIP/microtalk-00000035", "agi-VDAD_ALL_inbound.agi") in new stack
[May 20 11:08:05] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[May 20 11:08:05] -- <SIP/microtalk-00000035> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[May 20 11:08:05] -- <SIP/microtalk-00000035> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[May 20 11:08:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 20 11:08:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 20 11:08:07] -- Started music on hold, class 'default', on channel 'SIP/microtalk-00000035'
[May 20 11:08:10] -- Stopped music on hold on SIP/microtalk-00000035
[May 20 11:08:10] -- <SIP/microtalk-00000035> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[May 20 11:08:10] -- <SIP/microtalk-00000035> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[May 20 11:08:10] -- <SIP/microtalk-00000035> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[May 20 11:08:10] -- <SIP/microtalk-00000035> Playing 'generic_hold.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[May 20 11:08:16] -- Started music on hold, class 'default', on channel 'SIP/microtalk-00000035'

This is good now. This is because the phone number caller id wont display on the phone numbers field. And made some adjustment in agi-VDAD_ALL_inbound.agi extension

I'm also having this same issue. Do you remember what changes you made to make it work?

try this out:

https://dialer.one/how-to-create-a-call ... on-on/?amp

carpenox wrote:try this out:

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

I pretty much did that as indicated in the manual. The issue is that the call goes to the agent queue, but never reaches the agent. it stays in the queue. just like the OP showed in this thread.

rmiranda wrote:I'm also having this same issue. Do you remember what changes you made to make it work?

Please share your version information. Vicidial Version with Build, installer filename or full version (or link to instructions).

Share your related configuration settings.

Perhaps show the asterisk CLI and/or log output from a test call (with nothing else on the system, so no extraneous traffic, and so you're not tempted to delete something you THINK is unrelated but is actually the problem, lol, happens).

Obviously we don't need actual phone numbers or IPs or carrier names.

Also verify that the agent in question is actually IN the queue to take the call. In the real-time-screen click the (+) next to the agent name and verify that the ingroup is highlighted. Amazing how often the agent just isn't in the ingroup to take the calls.

OK thx for your postback rmiranda, if you're still having trouble join my gc on Skype we've got over 150 experts in there now who can prolly help