outbound survey calls hang up exact after 14Sec

[javad-afzal]

Hi
trying to setup a survey outbound campaign every settings looks fine but every outbound call hang up after 14Sec unable to figure out why.


ViciBox.i686-4.0.3.preload.iso | Vicidial 2.8-406a Build 130627-0745 | Asterisk 1.4.44-vici | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

[williamconley]

Post some CLI and/or AGI output and perhaps we can help you. Did you receive any of these calls to see if the outbound message is playing? Perhaps the recording ended ...?

[javad-afzal]

Thanks
that is working fine now. but i have an another issue with that i am running an outbound survey campaign which drops on a call menu than to the ingroup. my survey settings are
Survey First Audio File: None
Survey DTMF Digits:123456789#
Survey Not Interested Digit: 2
Survey Wait Seconds: 2
Survey Opt-in Audio File:
Survey Not Interested Audio File:
Survey Method: Call menu
Survey No-Response Action: Optout
using campaign with the Answering machine detection.( routing Extension 8373)
call menu is simple press 1 for transfer and 2 for hangup. pressed 1 calls drops on an ingroup where a welcome message is played than the call goes to the agents in that group.
problem i am facing is most of my calls gets disconnected within 1 minute duration, and some of the calls are blank(no customer on the other end). i have tried changing my trunk i thought issue was from the carrier but it didnt solved the issue. there is no Lag issue between server and agents. server load average remains less than than 0.80 overall my shift. checked for the codecs also no issues found on that.
Kindly give any advice with the issue, thanks in advance.

ViciBox.i686-4.0.3.preload.iso | Vicidial 2.8-406a Build 130627-0745 | Asterisk 1.4.44-vici | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

[amjohnson]

Are you using a sip provider?

Try this. From a phone connected to vicidial dial 9 and then your cell phone or office phone. When the call completes do you have audio both directions?

Is your vicidial system setup behind a router? If it is read the following. If it is not stop here and ignore my long winded post.

SIP is notorious for having problems behind a firewall. If its behind a firewall you need to setup a few things. Make sure you have nat=yes and canreinvite=no in your carrier config. Usually they are defined globally in the configs but it wont hurt to add it to the carrier definition. Also ports 5060 and 10000-20000 must be forwarded to your vicidial system.

The instructions below involve editing core files in the asterisk system, the heart of vicidial. Setting these options wrong can leave asterisk unable to start and your vicidial system dead in the water. If you are not familiar with Linux find someone who is before proceeding. Always always ALWAYS make a backup of any config file before you attempt changes like these in case things go horribly wrong.

If you have more than one server you will need to edit /etc/asterisk/rtp.conf and set different none overlapping ranges for each server. IE server1 10000-20000 with those forwarded to that server, server2 20001-30000 with those ports forwarded to server2 in your firewall.

The same must be done in /etc/asterisk/sip.conf. The default port setting in asterisk is set via bindport=5060 in /etc/asterisk/sip.conf. If you have more than one server you must forward 5060 to server1 and forward 5061 to server2 and setup the forward in your router for 5061 to server2.

Some carriers do not play nice unless you send the proper ip address in your sip headers. If your sending 192.168.0.xx as your ip address many carriers will drop your call after 15-30 seconds as they many not see the call connect.

If, and only if, your internet provider gave you a static ip address Set either externip= to your ip address in sip.conf and uncomment it if it is commented.

Say your STATIC ip address is 42.24.56.234 your /etc/asteriks/sip.conf should look similar to:

Code: Select all

externip = 42.24.56.234       ; Address that we're going to put in outbound SIP
;externhost=test.test.com     ; Alternatively you can specify a domain
;externrefresh=10               ; How often to refresh externhost if
localnet=192.168.0.0/255.255.0.0 ; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0


If you have a dynamic up address or are unsure if you have a static ip address or not you can still setup asterisk to send the carrier a dynamic ip hostname that will help their server find your Vicidial system. You will need to setup a dynamic ip address via dyndns.com, no-ip.com or another dynamic ip provider and setup your router you automatically update that dynamic ip address when it changes.

In /etc/asterisk/sip.conf, comment out the externip= if its set. Uncomment externhost= and set it to your dynamic ip address. Also uncomment the externrefresh= so that asterisk will periodically check to see if the ip address changed. The default setting is 10. This means if your ip address changes mid day it will take up to 10 minutes for vicidial to get the new ip address and while vicidial has the wrong ip address, it is likely all your calls will drop. Although in most cases the ip address will not change in the middle of the day unless your cable modem/router are reset.

Say you have a dynamic ip address and you setup vicidial.kicks-ass.net as your dynamic ip address name and configured it in your router to update it.

Code: Select all

;externip = 42.24.56.234       ; Address that we're going to put in outbound SIP
externhost=vicidial.kicks-ass.net     ; Alternatively you can specify a domain
externrefresh=10               ; How often to refresh externhost if
localnet=192.168.0.0/255.255.0.0 ; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0


I always make sure these are setup even if the server has a static ip address as I have have just got into the routine of setting vicidial up this way. Setting these also will help if you have a public and private network interface in your vicidial box.

Hope this helps and I didn't just confuse the hell out of you..