How to Install and setup ViciDialNow(w/SIP Service Provider)

[sunnymannava]

i have followed each and every instruction of yours and i am not unable to make calls and i am getting this message


May 21 10:43:24 NOTICE[5472]: chan_local.c:526 local_alloc: No such extension/context 17802810013@default creating local channel
May 21 10:43:24 NOTICE[5472]: channel.c:2514 __ast_request_and_dial: Unable to request channel Local/17802810013@default
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
May 21 10:43:24 NOTICE[5476]: chan_local.c:526 local_alloc: No such extension/context 17802810014@default creating local channel
May 21 10:43:24 NOTICE[5476]: channel.c:2514 __ast_request_and_dial: Unable to request channel Local/17802810014@default
== 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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on 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
== Manager 'sendcron' logged off from 127.0.0.1
== 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

my sip.conf

; register => 100002:password@206.51.234.171:5060
;
; setup account for SIP trunking:
; [trunk_1]
; disallow=all
; allow=ulaw
; allow=alaw
; type=friend
; username=100002
; secret=password
; host=206.51.234.171
; dtmfmode=inband
; qualify=1000
; nat=yes
; fromuser=100002

and when i am using manual dialing i m getting a voice message "wrong extension"

my extensions.conf

; dial a long distance outbound number through a SIP provider
; exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91NXXNXXXXXX,2,Dial(${SIPtrunk}/${EXTEN:1},55,o)
; exten => _91NXXNXXXXXX,3,Hangup

[caicedo2040]

hi sunnymannava,
You have to check the way you have to dial the destination number.

It seems you are uploading the destination numbers with format 1+ten digits (i, e, 17802810013), but on your dial plan you have 9+1+ten digits as default.
try the following:
; dial a long distance outbound number through a SIP provider
exten => _1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1NXXNXXXXXX,2,Dial(${SIPtrunk}/${EXTEN:0},55,o)
exten => _1NXXNXXXXXX,3,Hangup

also, on campaign settings, you need to replace prefix from "9" to "X".

finally, reload asterisk and all will work as .

[sunnymannava]

hello thank u for your reply and i have done the thing which you wanted me to do.but still i am getting the same message "invalid extension"

asterisk CLI

> Channel SIP/cc100-09a11d78 was answered.
-- Executing MeetMe("SIP/cc100-09a11d78", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-2bba,2", "8600051|F") in new stack
> Channel Local/8600051@default-2bba,1 was answered.
== Starting Local/8600051@default-2bba,1 at default,17802810022,1 failed so falling back to exten 's'
== Starting Local/8600051@default-2bba,1 at default,s,1 still failed so falling back to context 'default'
-- Sent into invalid extension 's' in context 'default' on Local/8600051@default-2bba,1
-- Executing Playback("Local/8600051@default-2bba,1", "invalid") in new stack
-- Playing 'invalid' (language 'en')
May 21 16:20:44 WARNING[27840]: file.c:1045 ast_waitstream: Unexpected control subclass '-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
-- Executing MeetMe("Local/58600051@default-9146,2", "8600051|Fmq") in new stack
> Channel Local/58600051@default-9146,1 was answered.
-- Executing Answer("Local/58600051@default-9146,1", "") in new stack
-- Executing Monitor("Local/58600051@default-9146,1", "wav|20100522-015048_7802810022") in new stack
-- Executing Wait("Local/58600051@default-9146,1", "3600") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
-- Timeout on Local/8600051@default-2bba,1
== CDR updated on Local/8600051@default-2bba,1
-- Executing Goto("Local/8600051@default-2bba,1", "#|1") in new stack
-- Goto (default,#,1)
-- Executing Playback("Local/8600051@default-2bba,1", "invalid") in new stack
-- Playing 'invalid' (language 'en')
== 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
-- Executing Hangup("Local/8600051@default-2bba,1", "") in new stack
== Spawn extension (default, #, 2) exited non-zero on 'Local/8600051@default-2bba,1'
-- Executing DeadAGI("Local/8600051@default-2bba,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-2bba,2'
-- Executing DeadAGI("Local/8600051@default-2bba,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== 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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on 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
== Manager 'sendcron' logged off from 127.0.0.1

[caicedo2040]

Hi,

can you send the output for manual dialing? it will be more easy to see final status.
By other hand, on your sip.conf, just to verify.... make sure trunk configuration settings are not commented. I see on your first post you still have ";" character on sip trunk lines.
It has to be like this

;trunk configuration
[trunk_1]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=100002
secret=password
host=206.51.234.171
dtmfmode=inband
qualify=1000
nat=yes
fromuser=100002

[sunnymannava]

i am new to this and really sorry to ask you how to show you Output for manual dialing?

[mubeen.mohd1]

HI ALL
I HAVE INSTALLED VICIDIAL NOW. I CAN SEE CALLS ARE RINGING. BUT NO ONE IS ATTENDING IT OR SOME OTHER PROBLEM. I HAVE CONFIGURED MY sip ACCOUNT ALSO. I'M TRYING TO CALL CANADA

PLZ HELP ME OUT

[carsonrose]

So, I have to edit the sip.conf file... no problem.

Does that mean that I dont have to set it up in the account entry?

I should be registering, why am I not? Do I just have to set it up twice? Once in the GUI and once in sip.conf??


EDIT:

I guess thats what i had to do, now I register:

goautodial*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
vitel-outbound 64.2.142.93 N 5060 OK (48 ms)
vitel-inbound 140.239.143.5 N 5060 OK (49 ms)


Is there some reason that the gui isnt making changes??

[carsonrose]

Now the same thing is happening with extensions.conf...

Am I missing something? I thought the whole reason you have a GUI is so that you dont have to modify the files manually.

Im comfortable editing files manually, but whats the point of the GUI doing it this way?? It doesnt save ANY of my changes.

Any one?

[gardo]

Have you read the GoAutoDial Getting Started Guide?

[hunter2009]

Hi
Am using Goautodial For Domestic Call center with 2 E1/T1 PRI-lines with Digium 2 Port E1 card With Dhadi Configured my self working good . we planning to add 4Port Gsm Gateway .Our Gsm gateway is Giving the out ip Like 192.168.xxx.xxx How to config In sip.conf and Out Going Dialplans We need make Some GSM Calls via this GSM GATEWAY waiting for the reply...

[williamconley]

externip=

[Inteleserv]

Hi,

I'm a newbie to setting up a GoAutodial Server, but I have worked on them. First of all, Kudos to all of you for such a wonderful software. The new interface looks great!

I'm using GoAutoDial 2.1. Here's my issue.

I'm dialling Australia. My carrier wants the calls authenticated via static IP instead of username/pin. The codec preferred is g729. I also need to put in a CLI, say 61333444555. The calls have to be sent in the format 61XXXXXXXXX, where 61 is the country code for Australia, and the rest of it is a nine digit phone number with area code.

I am able to log in to the system as an agent. I'm getting a call on xlite, and when i answer, i get "you are currently the only person in..."

When I dial manually, with 61 as the code and the nine digit phone number, I get the message "That's not a valid extension".

Here is what I added to various files

sip.conf, under [general] section

externip=x.x.x.x (my static IP)
localnet=192.168.1.0/255.255.255.0

in [truck1] section
nat=yes

Under Admin->Carriers, a new carrier was created, and configured as:

Account Entry:

[provider]
disallow=all
allow=ulaw
allow=alaw
type=friend
host=y.y.y.y (provider ip)
dtmfmode=rfc2833
qualify=1000

Globals String

SIPtrunk=SIP/provider

Dialplan Entry

exten => _9XXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXXX,2,Dial(${SIPtrunk}/${EXTEN:1},55,o)
exten => _9XXXXXXXXXXX,3,Hangup

Asterisk Log Entry for agent login and call:

[Jun 5 23:35:19] VERBOSE[7627] logger.c: [Jun 5 23:35:19] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-719d,2", "8600051|F") in new stack
[Jun 5 23:35:20] VERBOSE[7626] logger.c: [Jun 5 23:35:20] > Channel Local/8600051@default-719d,1 was answered.
[Jun 5 23:35:20] VERBOSE[7628] logger.c: [Jun 5 23:35:20] == Starting Local/8600051@default-719d,1 at default,61399399399,1 failed so falling back to exten 's'
[Jun 5 23:35:20] VERBOSE[7628] logger.c: [Jun 5 23:35:20] == Starting Local/8600051@default-719d,1 at default,s,1 still failed so falling back to context 'default'
[Jun 5 23:35:20] VERBOSE[7628] logger.c: [Jun 5 23:35:20] -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-719d,1
[Jun 5 23:35:20] VERBOSE[7628] logger.c: [Jun 5 23:35:20] -- Executing [i@default:1] Playback("Local/8600051@default-719d,1", "invalid") in new stack
[Jun 5 23:35:20] VERBOSE[7628] logger.c: [Jun 5 23:35:20] -- Playing 'invalid' (language 'en')

I logged in as agent001 on xlite configured as 8001, campaign TESTCAMP, accepted the call. From the agent screen, i chose manual dial, put in 61 as country code and 399399399 as the number.

Also, I have X as a prefix in TESTCAMP

I'd be really grateful. if someone could please tell me what I'm doing wrong?

[striker]

as your dialplan starts with 9 , then u need to dial any number with 9 as perfix.

and your dialplan will discard the 9 while dialling through your carrier.

or put 9 as dialprefix in the campaign.

[switch2voip]

Is there a complete guide for dummies? I have many people asking everyday how they can configure Vicidial, generally people with no experience configuring the software, i already found a guide for dummies but it doesn't includes how to setup both the trunk and the inbound DID's and that's what people usually needs.

[williamconley]

Have you tried the Vicidial Manager's Manual? (Free Version and Paid version both available on EFLO.net)

[striker]

goautodial Wiki has very good HowTos below is the link
http://goautodial.org/projects/goautodialce/wiki/HowTos3

[gardo]

And starting with GOautodial version 3.0, setting up trunks and campaigns (including DIDs) are easier using our user friendly "wizards" tools.

[Deepak209e]

whatever leads i load half of them are going in No Answer Goautodial !!!!!
why this is happening and whats the solution ?
reply fast , i need to fix this issue so calling goes better

[Asterisk]

hello all ..


i wannt to install and configure in centos .. i wannt to use this server for server voip 5.11 asterisk-1.4.39.1-vici.go , asterisk-addons-1.4.13-1.go van enyone help me please

[williamconley]

hello all ..


i wannt to install and configure in centos .. i wannt to use this server for server voip 5.11 asterisk-1.4.39.1-vici.go , asterisk-addons-1.4.13-1.go van enyone help me please

1) You should not use CentOS, you should use the installer from Vicibox.com which is an easy and supported installer and uses OpenSuSE. The installation manual is on Vicibox.com, and after installation you switch to the Vicidial Manager's Manual which has both Free and Paid versions on EFLO.net (Free is all you need for a functional dialer, Paid version will help you down the road, though! Highly recommended).

2) If, however, you MUST use CentOS (you should provide your reason, to make people like me stop suggesting otherwise), you should use the installer from Goautodial.org. The site also has a Wiki with instructions for installation. After installation, again you switch to the Vicidial Manager's Manual. Also: If I recall, this site has instructions for installation on an existing CentOS server if you can not install from an .iso.

3) You should not hijack a thread for your own nefarious purposes, but instead create your own, unless your question is actually the same or a continuation of the previous post. For future reference, of course.

Happy Hunting!