GOautodial Open Source Omni-channel Contact Center Suite

@Disconnect problem :
From asterisk logs - Unable to request channel SIP/8001

No SIP phone attached, ofc. Any way to simulate the presence up until agent clicks Hang Up ?

Yes. You can use GOautodial as a mini CRM or caller management software even if you're not using it's dialing features.

Can you post the output of your Asterisk CLI when dialing? You don't need to comment out anything in Asterisk (unless you need to configure some advance parameters).

Hello. This is the output from CLI when i do the following

- Login with the agent001
- Press dial next

[Oct  4 03:58:48]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  4 03:58:48] NOTICE[32497]: channel.c:5429 __ast_request_and_dial: Unable to request channel SIP/8001
[Oct  4 03:58:48]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct  4 03:59:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  4 03:59:01]     -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000002;2", "8600051,F") in new stack
[Oct  4 03:59:01]        > Channel Local/8600051@default-00000002;1 was answered.
[Oct  4 03:59:01]   == Starting Local/8600051@default-00000002;1 at default,9386025511314,1 failed so falling back to exten 's'
[Oct  4 03:59:01]   == Starting Local/8600051@default-00000002;1 at default,s,1 still failed so falling back to context 'default'
[Oct  4 03:59:01]     -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-00000002;1
[Oct  4 03:59:01]     -- Executing [i@default:1] Playback("Local/8600051@default-00000002;1", "invalid") in new stack
[Oct  4 03:59:01]     -- <Local/8600051@default-00000002;1> Playing 'invalid.gsm' (language 'en')
[Oct  4 03:59:01]   == Parsing '/etc/asterisk/meetme.conf': [Oct  4 03:59:01]   == Found
[Oct  4 03:59:01]   == Parsing '/etc/asterisk/meetme-vicidial.conf': [Oct  4 03:59:01]   == Found
[Oct  4 03:59:01]     -- Created MeetMe conference 1023 for conference '8600051'
[Oct  4 03:59:01]     -- <Local/8600051@default-00000002;2> Playing 'conf-onlyperson.gsm' (language 'en')
[Oct  4 03:59:01]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct  4 03:59:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  4 03:59:01]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct  4 03:59:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  4 03:59:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct  4 03:59:05]     -- Executing [i@default:2] Hangup("Local/8600051@default-00000002;1", "") in new stack
[Oct  4 03:59:05]   == Spawn extension (default, i, 2) exited non-zero on 'Local/8600051@default-00000002;1'
[Oct  4 03:59:05]     -- Executing [h@default:1] AGI("Local/8600051@default-00000002;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Oct  4 03:59:05]     -- <Local/8600051@default-00000002;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Oct  4 03:59:05]     -- Hungup 'DAHDI/pseudo-2087551056'
[Oct  4 03:59:05]   == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000002;2'
[Oct  4 03:59:05]     -- Executing [h@default:1] AGI("Local/8600051@default-00000002;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Oct  4 03:59:05]     -- <Local/8600051@default-00000002;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Oct  4 03:59:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  4 03:59:06]   == Manager 'sendcron' logged off from 127.0.0.1

I see the notification with phone number , customer data is displayed and status "Waiting for ring ...."
Then Agent diisconnects after approx. 20 seconds.

Thank you.
Damijan

On the softphone side, make sure that any RTP timeout settings are disabled. Some softphones disconnects after X number of seconds if this is enabled. Also make sure that Ulaw or Alaw is set as the primary codec on the softphone (if on the same LAN as the server). GSM is recommended if the softphone is going to connect to your server over the internet.

Looks like you don't have a dialplan configured for 9386025511314. Can you post a screenshot of your carrier configurations?

[Oct  4 03:59:01]   == Starting Local/8600051@default-00000002;1 at default,9386025511314,1 failed so falling back to exten 's'
[Oct  4 03:59:01]   == Starting Local/8600051@default-00000002;1 at default,s,1 still failed so falling back to context 'default'

Well, yes. That's because there is no softphone installed.
We want to use goautodial as a caller management software without any dialing.

That is ....
When the agent presses Dial Next he/she gets the lead data on screen (name , surname , address , phone number).
Agent dials number by hand on his prehistoric phone.
After the (possible) conversation agent presses Hang Up and can choose one of the post-dial options (Like Busy, Sale made , No sale , Callback , etc ....).

This already works for 20 seconds , then agent gets disconnected.

So, instead of "Waiting for ring ..." after "Dial Next" , the response should be like the connection is already established (simulated) or something ... sorry, don't know the right terminology to describe this behavior in less words :)

YOu need to have a softphone installed and registered properly (to the server) to use the GOautodial agent application.

It's actually possible to integrate GOautodial on a legacy PBX system. And you can also use a "prehistoric" phone to connect to GOautodial as the agent's phone.