GoAutoDial and FreePBX Integration

[shafique]

Hi All,

I have been in a situation where i need Goautodial to dialout and FreePBX to respond to connected customer. detailed scenario is as below.

1- GoAutodial dials out in a broadcast campaign and voice message on answered calls, as soon as someone presses 1 it should log the call and then forward it to sip URI or registered peer in other FreePBX server.

2- FreePBX will receive the call according to its predefined Inbound Route and so on.

as per my assumption there may be 2 popular options for doing this,,

1- In Survey Opt-in option we set it to drop on a extension and that extension will dial the sip:extension@freepbxserver:5060

2- we create IAX2 Peering in both servers and then dial through peer-2-peer.


If i am going wrong then please guide me..

i personally like to go for option 1 (Dialing SIP Uri) as it is simple,, but i am not strong enough in dialplan configuration, can someone help me how should i implement this..

* sample dial plan extention in goautodial (which will dial the sip uri)
* sample trunk configuration in freepbx (if required)
* sample trunk configuration in GoAutodial (as freepbx as carrier)
* any other configuration that may be needed.

what i have gathered by wondering around is

The freepbx admin has asked me to send calls to his server with 1002 as prefix,

DIALPLAN EXTENSION FOR GOAUTODIAL
extensions.conf

SIPtrunk=SIP/freepbx

exten => 777,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 777,2,Dial(sip/1002${EXTEN:3}@freepbx,55,tTo)
exten => 777,3,Hangup

(1002 is an extension created on freepbx for taking these calls)

sip.conf

[freepbx]
type=peer
canreinvite=no
callerid=1002
disallow=all
allow=g729
allow=gsm
host=xxx.xxx.xxx.xxx
port=5060
nat=yes
canreinvite=no
context=tunkinbound

and when i dial 777 from a local phone 1001 registered on my goautodial below is the asterisk cli output.

Code: Select all

 -- Executing [777@default:1] AGI("SIP/1001-00000001", "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 [777@default:2] Dial("SIP/1001-00000001", "sip/1002@freepbx|55|tTo") in new stack
    -- Called 1002@freepbx
  == Manager 'sendcron' logged off from 127.0.0.1
[Jun 29 09:10:02] WARNING[11015]: chan_sip.c:13482 handle_response_invite: Received response: "Forbidden" from '"1001" <sip:1001@85.13.222.24>;tag=as0248fbff'
    -- SIP/freepbx-00000002 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [777@default:3] Hangup("SIP/1001-00000001", "") in new stack
  == Spawn extension (default, 777, 3) exited non-zero on 'SIP/1001-00000001'
    -- Executing [h@default:1] DeadAGI("SIP/1001-00000001", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CONGESTION----------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CONGESTION---------- completed, returning 0


(William my version build and installation method is in my signature )

[williamconley]

Probably simpler to use Remote Agent for this feature. This feature allows any phone in the world to be used as a Remote Agent as long as your Vicidial system can call it. Then you set the FreePBX server up as a "Carrier" and call the extension of your user/queue/whatever through that carrier as the remote agent extension (using a specific number of digits or dial prefix to route the call extension to that FreePBX carrier).

But you will need to verify the ability to call from Vicidial to FreePBX on that carrier using standard SIP techniques. Carrier in Vicidial -> Extension in FreePBX (or trunk, if you prefer, there are several ways ...). Often registering to the FreePBX server as an extension will help to resolve security/connection issues.