GOautodial Open Source Omni-channel Contact Center Suite

Hi Jurdan,

You can try to run the command "update_server_ip" and try to login again. If you still encounter the issue, please send us the asterisk error logs from your dialer.

Thank you!

I already did the update server ip command, but i still got the same error. more info: Im using a digitalocean vps. I followed the tutorial to install the goautodial from: http://asteriskonvps.com/how-to-install-vicidial-on-goautodial-in-centos-5/. it's almost perfect, except this issue im facing now.

btw here's the asterisk log:>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

SERVER IP ADDRESS CHANGE FOR VICIDIAL FINISHED! =========================================================================
Parsing '/etc/asterisk/asterisk.conf': Found
Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.39.1-vici.go RPM by demian@goautodial.com currently running on awivicidial (pid = 1854)
Verbosity was 4 and is now 7
[Feb 8 01:44:56] -- Executing [55558600051@default:1] MeetMeAdmin("Local/55558600051@default-b620,2", "8600051|K") in new stack
[Feb 8 01:44:56] WARNING⁸¹⁷¹: app_meetme.c:3134 admin_exec: Conference number '8600051' not found!
[Feb 8 01:44:56] -- Executing [55558600051@default:2] Hangup("Local/55558600051@default-b620,2", "") in new stack
[Feb 8 01:44:56] Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-b620,2'
[Feb 8 01:44:56] -- Executing [h@default:1] DeadAGI("Local/55558600051@default-b620,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Feb 8 01:44:56] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0

- process runtime      (11 sec) (0.183333333333333 minutes)
[root@awivicidial ~]# asterisk -vvvvvvvr
Asterisk 1.4.39.1-vici.go RPM by demian@goautodial.com, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.

[Feb 8 01:45:13] > Channel SIP/8001-00000005 was answered.
[Feb 8 01:45:13] Manager 'sendcron' logged off from 127.0.0.1
[Feb 8 01:45:13] -- Executing [8600051@default:1] MeetMe("SIP/8001-00000005", "8600051|F") in new stack
[Feb 8 01:45:13] == Parsing '/etc/asterisk/meetme.conf': [Feb 8 01:45:13] Found
[Feb 8 01:45:13] WARNING⁸²³²: config.c:812 process_text_line: No '=' (equal sign) in line 5 of /etc/asterisk/meetme.conf
[Feb 8 01:45:13] -- <SIP/8001-00000005> Playing 'conf-invalid' (language 'en')
[Feb 8 01:45:17] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/8001-00000005'
[Feb 8 01:45:17] -- Executing [h@default:1] DeadAGI("SIP/8001-00000005", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Feb 8 01:45:17] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
awivicidial*CLI>

//end of asterisk cli >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

Hi Jurdan,

is this a single server or multi server setup? have you change your server ip if yes please post your /etc/goautodial.conf and /etc/astguiclient.conf
Also post the screen shot of the exact error you have encounter during login

Thanks.

as per ur log seems some wrong confing in meetme.conf

WARNING8232: config.c:812 process_text_line: No '=' (equal sign) in line 5 of /etc/asterisk/meetme.conf

post vi +5 /etc/asterisk/meetme.conf

br
striker
www.striker24x7.blogspot.com

thanks for your help, i finally figured out and gave up already on vps, the dahdi modules wont compile on the current centos image of digitalocean. I tried this setup on a regular pc and it worked. The only problem Im facing rignt now is that after logging in to the campaign, when dialing manually or autodial, i can see the dialer is dialing and on a live call, the other end can hear me, but i can't here anything. I've put the server's local ip to dmz alreay to that no more problem with NAT, but still i have this problem, no audio for me, but the other end can hear me... please help.

One-way audio or no audio is usually related to NAT/firewall issues. Try putting your server on a public IP address and test if you still get the one-way audio issue.

Thanks for your reply, I finally was able to successfully install this on a digitalocean vps based on this solution: https://www.digitalocean.com/community/questions/asterisk-with-kernel-devel, dahdi is the only missing module to make it work to any vps i guess. So This issue has been resolved, thanks.