vicidial.org

Hello,

Below is my Vicidial Config.
OS : Centos 7
Vicidial Version : VERSION: 2.13-576a
Vicidial Build : BUILD: 161126-2138
Asterisk Version : 11.22.0-Vici

I have clustered Vicidial
1 Web
2 Asterisk
1 DB

The problem that i am receiving is when i login as an Agent the phone rings fine but i dont get the IVR " You are the only person in this conference" and i also dont get the beep sound when the call has been connected from the agent panel. The Sound server is active and Audio store is active. Can anyone please let me know what i am doing wrong ?

externip= in sip.conf?

nat=yes in each phone's configuration?

Firewall.

When asterisk initiates a phone call, there are always TWO ports in use if the call is a SIP phone call.

UDP Port 5060 for control and a UDP rport (random port) between 10k and 25k, chosen at the moment of the call, for the audio.

If the 5060 packet gets through, you get a ring and even a "live call" notice. But until the rport traffic succeeds, you won't get any audio.

So: UDP5060 is succeeding, but the rport is being blocked from the vicidial serve to the agent, so no audio.

Test: Try an IAX2 phone instead. IAX2 could be accurately described to a lay-person as "SIP In a Wrapper that gets through the firewall". NO it is not necessary to make any other changes such as carriers. The agent phone being IAX instead of SIP has no other effect. All you need to do is download and install an IAX2 phone (zoiper free is nice for this) and create an IAX2 phone under amin->phones.

We usually use the sql script that adds 50 SIP and 50 IAX2 phones when we build a server. It's necessary to update the IP on all the added phones, but otherwise it's very fast and efficient.

We also like to use an older copy of zoiper, since newer versions kinda suck. Turn off "check for updates" immediately after installing.

http://download.poundteam.com/Zoiper_Fr ... aller.exe2 (must be renamed from .exe2 to .exe after download, this gets past many security protocols that would otherwise block the executable download). If you're not comfortable downloading from our site (reasonable if you don't know us), just find Zoiper Free 2.28 online from zoiper directly ... if you can still find it. This is the executable we put on all our servers at buildout for clients.

i have mentioned the external IP of the asterisk system on all the asterisk servers.
nate=yes on all the phone configs
Firewall has already been disabled and i have allowed all incoming and outgoing traffic.
have tried with iax2 protocol and it is the same, there are no agent sounds or any other IVR when logged in.

Is the agent using a soft phone or a VoIP phone?

If soft phone, USB or 3MM audio jack attached to a sound card?

has this been tested on more than one workstation with more than one audio equipment?

After registering: Does the agent get a dial tone when "off hooking" the phone?

The Agen is on Softphone

The softphone is on a laptop and the audio is on the speaker

Yes, this has been tested on 4 laptops

Yes, the agent gets the dial tone, i can call and get the audio of the calls i can even talk. but i do not receive the beep sound and the ivr that says you are the only person in this conference. other than that everything is working. I guess this is because of the clustering.

You may want to include a link to your installation instructions at this point.

And asterisk CLI example from a single login. (not 3000 lines of unrelated code, just from before the login until after the hangup so nothing in this one test call is missed).

asterisk log

> Channel IAX2/9001-781 was answered
-- Executing [8600003@default:1] MeetMe("IAX2/9001-781", "8600003,q") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
== Parsing '/etc/asterisk/meetme-vicidial.conf': Found
-- Created MeetMe conference 1023 for conference '8600003'




installation link : https://swanand18.blogspot.com/2017/06/ ... tos-7.html

you may have made a mistake in the install, or the instructions may be off.

the Vicidial Conferences and the Conferences are NOT interchangeable.

Conferences are 8600001 to 8600049. Vicidial conferences are 8600051 through 8600299.

i did not understand, what did i do wrong ?

Your agent logged in to conference 8600003. That's impossible. He should have been logging in to 8600053 instead.

So you have loaded "conferences" into the "vicidial_conferences" table by mistake. Thus the "Vicidial Conferences" have the wrong numbers. Delete them and load the proper conferences. They are not interchangable just because they both have the word "conferences" in them. One is Vicidial and the other is not.

Now the calls are dropping as soon as i answer.

> Channel SIP/1012-00000004 was answered
-- Executing [8600051@default:1] MeetMe("SIP/1012-00000004", "8600051,F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
== Parsing '/etc/asterisk/meetme-vicidial.conf': Found
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/1012-00000004'
-- Executing [h@default:1] AGI("SIP/1012-00000004", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- <SIP/1012-00000004>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

turn on sip debug. you'll probably find a reason for the call to fail instead of beginning to transfer sound.

If, for instance, your phone insists on g729, but the server doesn't have g729 installed ... well, that ain't workin'.

process:

Server "invites" phone to a call
Phone receives invite and rings the user interface
Phone sends "ringing!" back to the server

this is how it sits while ringing ... eventually, you pick up the phone.

Phone sends Answer! and here's the codecs I can use to server.

Server sends "here's the codecs I can use for audio" to the phone.

Phone picks a codec ... IF they match. If there is no overlap in the phone vs server codecs, they can't begin audio. Then they both agree this ain't workin' and terminate the call.

But this all happens in SIP, unrelated to the "extension dialplan". All the asterisk CLI shows is the extenson dialplan logging. So if you show SIP DEBUGging, you'll see that background handshake, too.

But I warn you: It's confusing and there's A FREAKIN LOT of packets back and forth. Do not have anthing else happening on the server. It would be best if you even turned off all other workstations and shut down the carriers (change yes to no) for a couple minutes before your attempt. That may make it easier for you to see and reason these packets. Note, of course, that all you need is the "termination/rejection" reason. The rest is just babble.