vicidial.org

hello

first, i'm just learning and implementing goautodial as beginner and I have configured basic functions of vicidial which is predictive auto dialing and trannsfer to agent direct for closer for my call center with 10 agents.

now, I have external call centers working with me and they're using elastix (freePBX) most of all.
I have to configure this vicidial to receive transfers from external call centers

1. external callcenters call customers (outbound)
2. they have to transfer the customer to my vicidial closer (recording calls obviously)

is this configuration possible?

I have been looking in manager manual but I couldn't find the exact solution for this

could anyone explain step by step how to do this ? (consider that I'm so newbis on this)

thanks

my version info

goautodial-ce-2.1-final
vicidial VERSION: 2.4-309a
BUILD: 110430-1642

Hello

ok, Surfing the web, I think I have one solution what is with creating in-group and a DID extension
so the elastix server and vici server just connect with the IP.

but still I can get the call transfered into one of my closer agent (agent logged to the web)

I have configured my vicidial like this

1. one campaign "outbound" for predictive
2. one campaign "closers" for closers
3. one in-group "AGENTDIRECT" directing campaign closers and selected as transfer group for campaign "outbound" (this in-group comes as default when I installed)

with those configurations, outbound function with predictive, and call transfers from agent of "outbound" to "closers" agent work great.

now for external agents using elastix server (not predictive)
I have

1. new in-group "external" selected as inbound group for campaign "closers"
2. did with number "5000" and routing to in-group "external"
3. my vicidial server and external elastix server are connected with IP (ext to ext working great both)

when from elastix extension dial "5000" I get "the person at extension 5000 is unavailable" and goes to voicemail
and the CLI shows this

-- Executing [5000@default:1] Dial("SIP/xxx.xxx.xxx.xxx-0000786c", "SIP/5000|60|") in new stack
[Apr 15 21:35:54] WARNING[25245] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [5000@default:2] Goto("SIP/xxx.xxx.xxx.xxx-0000786c", "default|850266666666665000|1") in new stack
-- Goto (default,850266666666665000,1)
-- Executing [850266666666665000@default:1] Wait("SIP/xxx.xxx.xxx.xxx-0000786c", "1") in new stack
-- Executing [850266666666665000@default:2] VoiceMail("SIP/xxx.xxx.xxx.xxx-0000786c", "5000|u") in new stack
-- <SIP/xxx.xxx.xxx.xxx-0000786c> Playing 'vm-theperson' (language 'en')
-- <SIP/xxx.xxx.xxx.xxx-0000786c> Playing 'digits/5' (language 'en')
-- <SIP/xxx.xxx.xxx.xxx-0000786c> Playing 'digits/0' (language 'en')
-- <SIP/xxx.xxx.xxx.xxx-0000786c> Playing 'digits/0' (language 'en')
-- <SIP/xxx.xxx.xxx.xxx-0000786c> Playing 'digits/0' (language 'en')
-- <SIP/xxx.xxx.xxx.xxx-0000786c> Playing 'vm-isunavail' (language 'en')
-- <SIP/xxx.xxx.xxx.xxx-0000786c> Playing 'vm-intro' (language 'en')

I know I'm doing something wrong but , can someone please explain

thanks

when the external calls come to YOUR asterisk from THEIR asterisk, the "dialplan" they will use (in extensions.conf or an incuded conf file) upon arrival is determined by the "context=" entry in their sip account. If they are set up as a carrier for the connection that would be in the "Account Entry". If they are set up as a phone, this would be the Context: field of the phone in admin->phones.

ANY inbound calls from ANY source (carrier, another call center, whatever) should always have the account set to "context=trunkinbound" in the sip account entry. This setting causes the "trunkinbound" in "extensions.conf" to be used to route the call, which will immediately invoke the inbound DID agi script ... thus turning the call over to the scripting system that is "Vicidial" and allowing Vicidial to route the call through the settings you have configured in the Vicidial User Interface.

Note that the "context=" entry in the account entry of a carrier is never used for anything other than inbound calls, so there is never a reason to have "context=" anything other than trunkinbound unless your goal is to block inbound calls from this source (ie: if this sip account will only be used for outbound calls, you would then have no need of inbound configuration).