General Advice on Vicidial

[badoudnace]

Sorry If I posted on the wrong forum but I would like some general advice.

Goautodial CE 2.1 | Asterisk 1.2.30.2 | Single Server | | No Extra Software After Installation | (still trying to find more info)

I have been asked to manage a new call centre that is using vicidialnow CE 2.1. The Call Centre is going to be taking calls for a GSM operator and using voip internally. The problem is that the call centre recently moved locations and now cannot receive any inbound or outbound calls. Unfortunately the system admin has disappeared with all the documentation for the installation.

I am no technical wiz but can pick things up relatively quickly. I have check the forums here and read the manager and agent manuals but still have no luck understanding what the problem is. Can login to the manager and agent interfaces, agents can login to phones but then that's where it ends. The Carrier, In-group, DID and campaign settings appear to be ok but not really sure where else to look for problem solving.

1)Any tips on where else to look for troubleshooting?
2)Is it wrong to assume that if it worked in the old location it should work in the new one?
3) Should I start from scratch and install latest GoAutodial?

I am leaning towards (3) above as I feel I will understand the installation better if I install it from scratch.

Again apologies if this is in the wrong forum.

[rrb555]

you can check for the asterisk logs to see what happens to the supposedly inbound calls or any other outgoing calls.

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asterisk -R

Check also your Carrier configured on the dialer and see if there are any issues with configuration. There are many post on this forum regarding with dial plan entry

Cheers!!

[badoudnace]

Thanks for response rrb555, I will check the logs and the carrier configuration

[badoudnace]

Using this carrier

Carrier ID: GSMCARRIER
Carrier Name: GSM GATEWAY
Registration String:
Template ID: -None-

Account Entry:
[gsmcarrier]
disallow=all
allow=ulaw
type=friend
host=192.168.0.176 (IP of Matrix SETU VGFX System)
dtmfmode=rfc2833
context-trunkinbound

Protocol: SIP
Global string: gsmgambia=SIP/gsmcarrier

Dialplan Entry:

exten => _000.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _000.,2,Dial(${gsmgambia}/${EXTEN:3},,tTor)
exten => _000.,3,Hangup

Server IP:192.168.2.201
Active: Y

This is the log when I try to dial out

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Connected to Asterisk 1.2.30.2 currently running on vici (pid = 3410)
Verbosity is at least 21
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
    -- Executing AGI("SIP/cc101-0993a490", "agi://127.0.0.1:4577/call_log") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
    -- Executing Dial("SIP/cc101-0993a490", "/17277568772||To") in new stack
Jul  3 15:36:22 WARNING[5758]: channel.c:2621 ast_request: No channel type registered for ''
Jul  3 15:36:22 NOTICE[5758]: app_dial.c:1076 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Hangup("SIP/cc101-0993a490", "") in new stack
  == Spawn extension (default, 97568772, 3) exited non-zero on 'SIP/cc101-0993a490'
    -- Executing DeadAGI("SIP/cc101-0993a490", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----66-----CHANUNAVAIL----------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----66-----CHANUNAVAIL---------- completed, returning 0


I am trying to resolve this myself as current budget for external consultant is non-existent. Please can someone advise on best course of action.

[williamconley]

when they moved, did they change the ip address of the server itself (and the gsm server, of course may also have changed)?

ie: Stage 1 verify that all your screens are running with "screen -list", stage 2 verify that asterisk is running, stage 3 verify information in /etc/astguiclient.conf is correct (basic "before you begin" verification steps for troubleshooting)

then verify that the carrier information from the GUI is actually being stored in the configuration files and asterisk has been reloaded to include that information. (sip show peers)

then you'll need to verify networking between the two machines in question ... and move onward from there!

[badoudnace]

Apologies for the late response.

Thanks for the advice it is much appreciated.