vicidial.org

Greetings people

I am having an issue. I installed Vicidial on AWS and after configuring correctly, now when I put a campaign on none-manual method, calls never pass to agent. The output in the logs is:

Code: Select all

[Mar 23 19:16:04] VERBOSE[27948][C-00000009] pbx.c: Executing [138368@default:1] AGI("Local/12345678@default-00000009;1", "agi-VDAD_local_optimize.agi,V3231915390000000009") in new stack
[Mar 23 19:16:04] VERBOSE[27948][C-00000009] res_agi.c: Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_local_optimize.agi
[Mar 23 19:16:04] VERBOSE[27947] manager.c: Manager 'sendcron' logged off from 172.31.16.228
[Mar 23 19:16:04] VERBOSE[27992][C-00000008] bridge_channel.c: Channel SIP/sometrunk1-00000006 joined 'simple_bridge' basic-bridge <dfda09e5-3bf6-4e68-8603-fe9dcc6c93e2>
[Mar 23 19:16:04] VERBOSE[27949][C-00000008] bridge_channel.c: Channel Local/12345678@default-00000009;2 joined 'simple_bridge' basic-bridge <dfda09e5-3bf6-4e68-8603-fe9dcc6c93e2>
[Mar 23 19:16:04] VERBOSE[27948][C-00000009] res_agi.c: <Local/12345678@default-00000009;1>AGI Script agi-VDAD_local_optimize.agi completed, returning 0
[Mar 23 19:16:04] VERBOSE[27948][C-00000009] pbx.c: Executing [138368@default:2] Wait("Local/12345678@default-00000009;1", "2") in new stack
[Mar 23 19:16:06] VERBOSE[27948][C-00000009] pbx.c: Executing [138368@default:3] Hangup("Local/12345678@default-00000009;1", "") in new stack


I had this issue before and decided to use asterisk 11, and fixed but now we need to use 13 version


I am using :
Asterisk 13.34.0-vici
Vicidial: VERSION: 2.14-798a BUILD: 210315-1730

What should I do?

How did you install this system? If there are instructions, please post a link to them (so others with the same instruction set might benefit ... and so someone may be able to spot the flaw in the instructions to help you).

Often with Vicidial not passing calls to agents, the problem is sound. Vicidial will not pass a call to an agent if two-way sound doesn't cause the channel to "initialize" and switch from Local to an actual trunk type (which happens when sound begins flowing). Sometimes this failure to allow sound is due to firewall blockage. Test: Turn off the firewall during an attempt. If that fixes the problem, you need to open whatever IP was being blocked that contained the audio of the call. Note that SIP trunking uses two connections: Port 5060 for control and a random port between 10k and 25k for audio. The random port connection may be from the same IP address, but then again it may NOT. If it's not, you need to find all the possible IPs for the carrier and open them all up.

Another problem can be that the sip.conf base file in use is the wrong one. If the asterisk version in /etc/astguiclient.conf is not 13.X (note the ".X", this is a requirement), the install.pl script will NOT match up and will install the Asterisk 1.4 compatible sip.conf file intead of the asterisk 13 compatible file. This happens during initial installation, and further installations do not copy the sample .conf file unless you specify this option during the running of the install.pl file. Remember that any changes to sip.conf (such as the externip value) will need to be made again if you re-run the installer and get the sample configuration files reloaded.

It's also best to be sure that the admin->servers setting for the asterisk version matches the asterisk version when using "asterisk -V".

Happy Hunting.