vicidial.org

Hello,

I am trying to call from vicidial to freeswitch(billing platform) to provider(gateway)
so call flow would be like this

My vicidial
60.80.70.250

my freeswitch
92.42.132.242

my provider(gateway)
72.2.242.42

I am dialing
18177779695

When i try using dialtype 'MANUAL' the call is routing correctly without any issue.
But when i try using dialtype 'RATIO', i am facing issue when freeswitch is resending the provider(gateway)'s response to vicidial the asterisk of vicidial server responding with
X-Asterisk-HangupCause: Unknown.
X-Asterisk-HangupCauseCode: 0.

I am attaching the sip log of both working(MANUAL) and and not working(RATIO) scenario here.
Please note that the attached sip log is from freeswitch server(billing platform).

Working sip log is here
http://pastebin.com/ma4e3xzk

Not working sip log is here
http://pastebin.com/6PAnyDBc

Please advice, what am I missing here?

plz post log ratio, in the asterisk server.

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method (7.X.X?) and vicidial version with build (VERSION: 2.X-XXXx ... BUILD: #####-####).

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "manual/from scratch" you must post your operating system with version (and the .iso version from which you installed your original operating system) plus a link to the installation instructions you used. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) use loopback call method to allow freeswitch to complete autodialed calls.

Loopback dialing to get callerid name to STAY in a freeswitch passthrough.

Code: Select all

exten => _97111NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _97111NXXNXXXXXX,n,Dial(${TRUNKloop}/9${EXTEN:4},55,o)
exten => _97111NXXNXXXXXX,n,Hangup

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,n,Dial(SIP/FreeSwitch/${EXTEN:1},,To)
exten => _91NXXNXXXXXX,n,Hangup()

* Dial Prefix: 9711
* Manual Dial Prefix: 9

4) In the file: usr/local/freeswitch/scripts/astpp/scripts/astpp.xml.lua

Which basically is the Freeswitch XML file for the dial plan, but being dynamic, this script auto populates the dial plan, all that should be needed to change is : “effective_caller_id_name” in two spots change to “origination_caller_id_name”

https://wiki.freeswitch.org/wiki/Variab ... er_id_name

Hi

I had same problem as you forget all other modifications and add this to the /etc/asterisk/extensions.conf somewhere down or up in the file

Code: Select all

; VICIDIAL_auto_dialer transfer script Load Balanced:
exten => 8368,1,Playback(sip-silence)
exten => 8368,n,AGI(agi://127.0.0.1:4577/call_log)
exten => 8368,n,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,n,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,n,Hangup()

Let me know if its working for you, manual calls were okay but calls with Ratio never received to agent.