[Autodial] Agent hear nothing Outbound Calls

[lark]

Hi Good Day,

This is My first Post here so I'm a Newbie just learned vicidial for awhile right now I'm experiencing problem(s)
when agent do autodial some of the calls Agent Hear Nothing sometimes agent hear bing sound but sometimes hear nothing at all even it says live call
so i tried digging up with our friendly neighborhood google(dot)com and see alot of user encounter same problem but it seems case were not resolve

Things I did:

max call per second set to 20
Max VICIDIAL Trunks: 96
Drop Calls Second: 5
Dial timeout: 30
Lock Timeout: 0
Dial method: Ratio
Autodial Level: 5:1

tried making the settings default in campaign and system setting/server/Telephony as much as possible

here is the CLI of Agent hear nothing:

-- Executing [712127773456@default:1] AGI("SIP/201211-0000810b", "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 [712127773456@default:2] Dial("SIP/201211-0000810b", "sip/Voip/12127773456||tToR") in new stack
-- Called Voip/12127773456
-- SIP/Voip-0000810c is making progress passing it to SIP/201211-0000810b
-- SIP/Voip-0000810c is ringing
-- SIP/Voip-0000810c is making progress passing it to SIP/201211-0000810b
-- SIP/Voip-0000810c answered SIP/201211-0000810b
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing [h@default:1] DeadAGI("SIP/201211-0000810b", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----27-----24") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -27-----24 completed, returning 0
== Spawn extension (default, 712127773456, 2) exited non-zero on 'SIP/201211-0000810b'

dial plan was:

exten => _71.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _71.,2,Dial(sip/Voip/${EXTEN:1},,tToR)
exten => _71.,3,Hangup

another CLI using different dial plan:

-- Executing [712127773456@default:1] AGI("SIP/201211-000080f1", "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 [712127773456@default:2] Dial("SIP/201211-000080f1", "SIP/Voip/12127773456||tToR") in new stack
-- Called Voip/12127773456
-- SIP/Voip-000080f2 is making progress passing it to SIP/201211-000080f1
-- SIP/Voip-000080f2 is ringing
-- SIP/Voip-000080f2 is making progress passing it to SIP/201211-000080f1
-- SIP/Voip-000080f2 answered SIP/201211-000080f1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing [h@default:1] DeadAGI("SIP/201211-000080f1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----34-----30") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -34-----30 completed, returning 0
== Spawn extension (default, 712127773456, 2) exited non-zero on 'SIP/201211-000080f1'

dial plan was:

exten => _71NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _71NXXNXXXXXX,n,Dial(SIP/Voip/${EXTEN:1},,tToR)
exten => _71NXXNXXXXXX,n,Hangup

Good Call CLI:

-- Executing [921001112127773456@default:1] AGI("SIP/201211-00008109", "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 [921001112127773456@default:2] Dial("SIP/201211-00008109", "sip/voip2/1001112127773456||tToR") in new stack
-- Called voip2/1001112127773456
== Refreshing DNS lookups.
-- SIP/voip2-0000810a is making progress passing it to SIP/201211-00008109
-- SIP/voip2-0000810a answered SIP/201211-00008109
-- Executing [h@default:1] DeadAGI("SIP/201211-00008109", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----10-----6") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --10-----6 completed, returning 0

does it have to do with these lines :

bad Call:

-- Called Voip/12127773456
-- SIP/Voip-000080f2 is making progress passing it to SIP/201211-000080f1
-- SIP/Voip-000080f2 is ringing
-- SIP/Voip-000080f2 is making progress passing it to SIP/201211-000080f1
-- SIP/Voip-000080f2 answered SIP/201211-000080f1

I noticed it did -- SIP/Voip-000080f2 is making progress passing it to SIP/201211-000080f1 2times

while the

good call:

-- Called voip2/1001112127773456
== Refreshing DNS lookups.
-- SIP/voip2-0000810a is making progress passing it to SIP/201211-00008109
-- SIP/voip2-0000810a answered SIP/201211-00008109

Does my problem have to do with those line i noticed??
Pls Advise thanks much!!



Version: 2.6b0.5
DB Schema Version: 1324
DB Schema Update Date: 2012-08-14 07:24:59
Auto User-add Value: 101
Install Date: 2009-04-19
Phone Codes: 1056 - 42456 - 0
Today System Stats: 30117 - 0 - 56 - 245 - 75 - 21
asterisk version: 1.4.39.1-vici

installation Method Auto (.iso)
first installed GoAutodial
then Installed Vicidial with version 2.4 then Upgraded to version 2.6b0.5

Did I miss the requirements on posting?
please advise and thanks much all

[williamconley]

You posted "goautodial", but not the version. Goautodial is the installer, and listing the version is helpful.

You posted Version 2.6b0.5 but not the build. This is the Vicidial Version and is in the bottom left corner of most admin pages and listing the full version with build is a requirement. Vicidial is the software actually being installed (can be installed with Vicibox or GoAutoDial).

In either case, however, you are not likely experiencing a dial plan challenge but a firewall or installation challenge.

When your agent logs in, do they hear "You are the only person in this conference"? How is your network set up? Are your agents on the same physical network as your Vicidial machine? How does your vicidial machine get to the internet? Through a router or does it have an internet IP of its own instead of a private IP on a local subnet?

What type of firewall do you have? Are there two firewalls between your local subnet and the internet (ie: You have a router in your local office hooked up to another router in your landlord's office which is then connected to the internet ...? this would be a bad setup ...)

[lark]

Hi Sir William thanks much for the reply I thought i was ignored anyway your post strike me ill go back to basics i know i can solve this thanks much again your post made me realize what i lack

[lark]

installation Method Auto (.iso)
first installed

GoAutodial
VERSION: 2.4-309a
BUILD: 110430-1642

then Installed
Vicidial
with version 2.4
then Upgraded to
VERSION: 2.6-373a
BUILD: 120810-1018
© 2012 ViciDial Group

Were using Ploycom as our Phones not (zoiper or eyebeam or xlite)


*When My agents logs in we can hear "You are the only person in this conference"

*Were using PFsense Scenario is:

Assigned Public ip 123.123.123.23
Pfsense IP as our gateway
port forward 10000 to 65535 and 5060 in our local Vicidial server
server IP

i.e

Pfsense 192.168.0.10
Vicidial IP: 192.168.0.11 gateway 192.168.0.10 netmask /24
DNS google dns as example

edited sip.conf
externip: (isp's IP)
rtp.conf
1000 to 65535


what happen now is we have 2 different telco

lets say

Telco A: Were ok with it but sometimes there are calls Agent hear nothing only himself even it its status is in LIVE CALLS
Telco B: worts happens Agent hear nothing Outbound Calls but sometimes calls went Ok


Please Advise anyone Give me a clue were i can start troubleshooting this
do i need to reinstall?? thanks much!!

[williamconley]

pfsense is then likely your issue. i've had many clients try to use it and none successfully with a Vicidial system. this is likely because at this point *I* say "try it without pfsense!" and they do ... and it works. so they don't go back.

the question is whether you can acquire an external IP address for your Vicidial system and bypass the pfsense firewall completely, thus removing it from the equation? If so ... try it (be sure you are prepared for a fully locked down whitelist firewall inside your vicidial with iptables if it works ...).

if you cannot, then you'll need to configure your pfsense and your asterisk to "go through it". some carriers require "externip=xxx.xxx.xxx.xxx" in your sip.conf (where xxx.xxx.xxx.xxx is the outside of your pfsense firewall).

do your problems happen "any time" or only during heavy load? pfsense may not be keeping up or may be interfering (security protocols meant to protect you often interfere instead!)

[zasdala]

hi ! im new here and my concern was. when i open the vicidial i van connect with my username and password that the company gave but i cant hear any sounds from the "you are curerently the only person to the live call" it says in the developer tool that theres an identified speakers. how can i fix it?

[williamconley]

hi ! im new here and my concern was. when i open the vicidial i van connect with my username and password that the company gave but i cant hear any sounds from the "you are curerently the only person to the live call" it says in the developer tool that theres an identified speakers. how can i fix it?

"developer tool"? In the web browser? But aren't you using Polycom phones? If so, sound is not being sent through or to the browser at all and as such the browser error related to speakers is irrelevant.

So ... what phone are you using? If it's Polycom, then the polycom phone must ring. Does this happen? If it rings and you answer and you do not hear "You are the only person in this conference ..." you have a firewall issue. (Or a busted phone, of course)

To PROVE that it's a firewall issue: Configure an IAX phone instead of a SIP phone. There are several free IAX based phones. They do not have firewall issues.