call of elastix toward vicidial

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call of elastix toward vicidial

Postby tresor » Fri Nov 22, 2013 7:50 am

Hi, i want to do a call center for the inbound call in my local network i have two servers elastix and vicidial. The agent uses vicidial interface for receive a call from elastix.
My configuration:
Information of vicidial: server IP is: 192.168.1.2 and the elastix server is: 192.168.1.4
Vicibox5.i686-5.0.3.iso|vicidial 2.8b0.5|build opensuse v.12.3 32 bit|Asterisk v1.8.23.0-vici|single server| No Digium/Sangoma Hardware | No Extra Software After Installation |Intel(R) Pentium|DUAL CPU E2220
• For the first I configure the extension number that the agent uses for connected

MODIFY A PHONE RECORD: 103
Phone Extension: 103
Dial Plan Number: 103 (digits only)
Voicemail Box: 103 (digits only)
Outbound CallerID:7275551214 (digits only)
Admin User Group: ALL
Phone IP address: (optional)
Computer IP address: 192.168.1.102 (optional)
Server IP: 192.168.1.2
Agent Screen Login: 103
Login Password: test
Registration Password: test Strength:
Set As Webphone: N
Webphone Dialpad: Y
Webphone Auto-Answer: Y
Use External Server IP: N
Status: ACTIVE
Active Account: Y
Phone Type: SIP
Full Name: sip103
Email:
Delete Voicemail After Email: N
Voicemail Zone: eastern
Voicemail Options:
Company:
Picture:
New Messages:
Old Messages:
Client Protocol: SIP
Local GMT: 0.00 (Do NOT Adjust for DST)
Phone Ring Timeout: 60
On-Hook Agent: N
Manager Login: cron
Manager Secret: 1234
Agent Default User:
Agent Default Pass:
Agent Default Campaign:
Park Exten:8301
Conf Exten: 8302
Agent Park Exten:8301
Agent Park File: park
Monitor Prefix: 8612
Recording Exten:8309
VMailMain Exten: 8501
VMailDump Exten: 85026666666666
Exten Context:default
Phone Context: default
Allowed Codecs:
Allowed Codecs With Template: 0
DTMFSend Channel: local/8500998@default
Outbound Call Group: Zap/g2/
CallerID URL: http://astguiclient.sf.net/test_callerid_output.php
Agent Default URL:
Call Logging: 1
User Switching: 1
Conferencing: 1
Admin Hang Up: 0
Admin Hijack: 0
Admin Monitor: 1
Call Park: 1
Updater Check: 1
AF Logging: 1
Queue Enabled: 1
CallerID Popup: 1
VMail Button: 1
Fast Refresh: 0
Fast Refresh Rate: 1000 (in ms)
Persistant MySQL: 0
Auto Dial Next Number: 1
Stop Rec after each call: 1
Enable SIPSAK Messages: 0
DBX Server: (Primary DB Server)
DBX Database: asterisk (Primary Server Database)
DBX User: cron (Primary DB Login)
DBX Pass: 1234 (Primary DB Secret)
DBX Port: 3306 (Primary DB Port)
DBY Server: (Secondary DB Server)
DBY Database: asterisk (Secondary Server Database)
DBY User:cron (Secondary DB Login)
DBY Pass: 1234 (Secondary DB Secret)
DBY Port: 3306 (Secondary DB Port)
Template ID:
Conf Override:

• then I configure the number of user and password
• MODIFY A USERS RECORD: 3333
User Number: 3333
Password:3333 Strength:
Force Change Password:
Last Login Info: 2013-11-21 18:44:19 - 0 - 192.168.1.65
Full Name: sergi
User Level: 1
User Group: STANDARD
Phone Login:
Phone Pass:
Active: Y
Voicemail ID: voicemail chooser
Email:
User Code:
Main Territory:
AGENT INTERFACE OPTIONS:
Agent Choose Ingroups: 1
Agent Choose Blended: 1
Hot Keys Active: 0
Scheduled Callbacks: 1
Agent-Only Callbacks: 1
Agent Call Manual: 1
Agent Recording: 1
Agent Transfers: 1
Closer Default Blended: 0
Agent Recording Override: DISABLE
Agent Alter Customer Data Override: NOT_ ACTIVE
Agent Alter Customer Phone Override: NOT_ACTIVE
Agent Shift Enforcement Override: DISABLE
Agent Call Log View Override: DISABLE
Agent Lead Search Override: NOT_ACTIVE
Alert Enabled: 0
Allow Alerts: 0
Preset Contact Search: NOT_ACTIVE
Campaign Ranks:
CAMPAIGN RANK GRADE CALLS WEB VARS
TEST_IT - Call inbound campaign 0 1 1

Inbound Groups:
INBOUND GROUP RANK GRADE CALLS WEB VARS
AGENTDIRECT - Single Agent Direct Queue 0 10 0
i check PRIME - Conpensation 0 1 0

• then I configure the campaign inbound that user selected to enter to the agent screen.
Campaign ID: TEST_IT
Campaign Name: call inbound campaign
Campaign Description:
Campaign Change Date: 2013-11-21 10:08:28
Campaign Login Date: 2013-11-21 18:44:19
Active: Y
Admin User Group: ---ALL---
Park Music-on-Hold:
Web Form:
Allow Closers: Y
Default Transfer Group: ---NONE---
Allow Inbound and Blended: Y
Dial Status 1: QUEUE - Lead To Be Called REMOVE
Dial Status 2: NEW - New Lead REMOVE
Add A Dial Status to Call: NONE
List Order: DOWN
List Mix: DISABLED-DISABLED
Lead Filter: NONE
Minimum Hopper Level: 5
Force Reset of Hopper:
Dial Method: RATIO
Auto Dial Level: 1 (0 = off)
Adapt Intensity Modifier: 0-balanced
Script:
Get Call Launch: NONE
Next Agent Call: Oldest_call_finish
Local Call Time: 24hours
State rules defined for this call time: -1
Dial Timeout: in seconds
Dial Prefix: for 91NXXNXXXXXX value would be 9, for no dial prefix use X
Manual Dial List ID : 998
Allowed Inbound Groups:
AGENTDIRECT - Single Agent Direct Queue - 99
i check this PRIME - Conpensation - 0
LISTS WITHIN THIS CAMPAIGN:
LIST ID LIST NAME DESCRIPTION LEADS COUNT ACTIVE LAST CALL DATE MODIFY

This campaign has 0 active lists and 0 inactive lists

Then I configure the inbound group
Friday November 22, 2013 10:05:02 AM
MODIFY A GROUPS RECORD: PRIME
Group ID: PRIME
Group Name: conpensation
Group Color: yellow
Active: Y
In-Group Calldate:
Admin User Group: ALL
Web Form:
Web Form Two:
Next Agent Call: Oldest_call_finish
Queue Priority: 0-Even
On-Hook Ring Time: 15
On-Hook CID: GENERIC
Fronter Display: Y
Script:
Ignore List Script Override: N
Get Call Launch:
Transfer-Conf DTMF 1:
Transfer-Conf Number 1:
Transfer-Conf DTMF 2:
Transfer-Conf Number 2:
Transfer-Conf Number 3:
Transfer-Conf Number 4:
Transfer-Conf Number 5:
Timer Action:
Timer Action Message:
Timer Action Seconds: 1
Timer Action Destination:
Drop Call Seconds: 360
Drop Action: MESSAGE
Drop Exten: 8307
Voicemail: voicemail chooser
Drop Transfer Group: NONE
Drop Call Menu:
Call Time: 24hours
Holidays defined for this call time: 0
Action Transfer CID: customer
After Hours Action: MESSAGE
After Hours Message Filename: vm-goodbye audio chooser
After Hours Extension: 8300
After Hours Voicemail: voicemail chooser
After Hours Transfer Group:
After Hours Call Menu:
No Agents No Queueing:
No Agent No Queue Action:
Audio File: nbdy-avail-to-take-call|vm-goodbye audio chooser
Max Calls Method:
Max Calls Count: 0
Max Calls Action: NO_AGENT_NO_QUEUE
Welcome Message Filename: --NONE-- audio chooser

Play Welcome Message: ALWAYS
Music On Hold Context: default moh chooser
On Hold Prompt Filename:generic_hold audio chooser
On Hold Prompt Interval: 60
On Hold Prompt No Block: N
On Hold Prompt Seconds: 10
Play Place in Line: N
Play Estimated Hold Time: N
Calculate Estimated Hold Seconds: 0
Estimated Hold Time Minimum Filename: audio chooser

Estimated Hold Time Minimum Prompt No Block: N
Estimated Hold Time Minimum Prompt Seconds: 10
Wait Time Option: NONE
Wait Time Second Option: NONE
Wait Time Third Option: NONE
Wait Time Option Seconds: 120
Wait Time Option Extension: 8300
Wait Time Option Callmenu:
Wait Time Option Voicemail: voicemail chooser
Wait Time Option Transfer In-Group: NONE
Wait Time Option Press Filename: to-be-called-back|digits/1 audio chooser
Wait Time Option Press No Block:
Wait Time Option Press Filename Seconds: 10
Wait Time Option After Press Filename: vm-hangup audio chooser
Wait Time Option Callback List ID:999
Wait Hold Option Priority: WAIT
Estimated Hold Time Option: NONE
Hold Time Second Option: NONE
Hold Time Third Option: NONE
Hold Time Option Seconds: 360
Hold Time Option Minimum: 0
Hold Time Option Extension: 8300
Hold Time Option Callmenu:
Hold Time Option Voicemail: voicemail chooser
Hold Time Option Transfer In-Group:
Hold Time Option Press Filename: to-be-called-back|digits/1 audio chooser
Hold Time Option Press No Block:
Hold Time Option Press Filename Seconds: 10
Hold Time Option After Press Filename: vm-hangup audio chooser
Hold Time Option Callback List ID: 999
Agent Alert Filename: ding audio chooser
Agent Alert Delay: 1000
Default Transfer Group: NONE
Default Group Alias: NONE
Dial In-Group CID:
Hold Recall Transfer In-Group: NONE
No Delay Call Route: N
In-Group Recording Override: DISABLED
In-Group Recording Filename: NONE
Stats Percent of Calls Answered Within X seconds 1: 20
Stats Percent of Calls Answered Within X seconds 2: 30
Start Call URL:
Dispo Call URL:
Add Lead URL:
No Agent Call URL:
Extension Append CID:
Uniqueid Status Display:
Uniqueid Status Prefix:

AGENT RANKS FOR THIS INBOUND GROUP:
USER GROUP SELECTED RANK GRADE CALLS TODAY
select all
3333 - sergi STANDARD 0 1 0


Then I configure my DID number
DID Extension: 8002277655
DID Description: inbound 800 numbers
Active: Y
Admin User Group: ALL
DID Route: IN_GROUP
Record Call: N
Extension: 9998811112
Extension Context: default
Voicemail Box: voicemail chooser
Phone Extension: 103
Server IP: 192.168.1.2
Call Menu:
User Agent:
User Unavailable Action:
User Route Settings In-Group: PRIME-Conpensation
In-Group ID: PRIME-Conpensation
In-Group Call Handle Method: CIB
In-Group Agent Search Method: LB
In-Group List ID: 999
In-Group Campaign ID: TEST_IT
In-Group Phone Code: 1
Clean CID Number:
Filter Inbound Number: DISABLED
Filter Phone Group ID:
Filter URL:
Filter Action: IN_GROUP
Filter Extension: 9998811112
Filter Extension Context: default
Filter Voicemail Box: voicemail chooser
Filter Phone Extension:
Filter Server IP:
Filter Call Menu:
Filter User Agent:
Filter User Unavailable Action:
Filter User Route Settings In-Group:
Filter In-Group ID: NONE
Filter In-Group Call Handle Method: CID
Filter In-Group Agent Search Method: LB
Filter In-Group List ID: 999
Filter In-Group Campaign ID:
Filter In-Group Phone Code: 1
Custom 1:
Custom 2:
Custom 3:
Custom 4:
Custom 5:



And I go in vicibox server on /etc/asterisk/extension.conf file under the default section I add this line:
exten => 9998811112,1,Ringing
exten => 9998811112,2,Wait(1)
exten => 9998811112,3,AGI(agi://127.0.0.1:4577/call_log--fullCID--${EXTEN}-----${CALLERID}-----${CALLERIDNUM}-----${CALLERIDNAME})
exten => 9998811112,4,Answer
exten =>9998811112,5,Dial(sip/103,30,To)
Therefore when I use sip 104 configured in vicidial and call this enxtension the sip phone 103 of agent ringing by other line phone the call is not mounted to the agent screen and the agent is in the conference and wait the incoming call.

Then I configure the groups
Group: STANDARD (no spaces or punctuation)
Description: reception and calling (description of group)
Force Timeclock Login:
Shift Enforcement:
Allowed Campaigns:
i check this: ALL-CAMPAIGNS - USERS CAN VIEW ANY CAMPAIGN
TEST_IT - Call inbound campaign


Group Shifts:
24HRMIDNIGHT - 24 hours 7 days a week


Agent Status Viewable Groups:
ALL-GROUPS - All user groups in the system
CAMPAIGN-AGENTS - All users logged into the same campaign as the agent
NOT-LOGGED-IN-AGENTS - All users in the system, even not logged-in users
ADMIN - VICIDIAL ADMINISTRATORS
STANDARD - reception and calling


Agent Status View Time: N
Agent Call Log View: N
Agent Allow Consultative Xfer: Y
Agent Allow Dial Override Xfer: Y
Agent Allow Voicemail Message Xfer: Y
Agent Allow Blind Xfer: Y
Agent Allow Dial With Customer Xfer: Y
Agent Allow Park Customer Dial Xfer: Y
Agent Fullscreen: N
Allowed Reports: ALL REPORTS
Allowed User Groups:
ALL-GROUPS - All user groups in the system
ADMIN - VICIDIAL ADMINISTRATORS
STANDARD - reception and calling
Allowed Call Times:
ALL-CALLTIMES - All call times in the system
i checkthis: 12pm-5pm - default 12pm to 5pm calling
12pm-9pm - default 12pm to 9pm calling
i check this: 24hours - default 24 hours calling
5pm-9pm - default 5pm to 9pm calling
9am-5pm - default 9am to 5pm calling
9am-9pm - default 9am to 9pm calling

USERS WITHIN THIS USER GROUP: 1
USER FULL NAME LEVEL ACTIVE
3333 sergi 1 Y

And the of the part of vicidial I configure the trunk towards elastix server:
carrierID: elastix
carrier name: elastix
admin user group: all
registration string: register=>vicidial1:vicidial1@192.168.1.4:4569
account entry:
[elastix1]
host=192.168.1.4
username=vicidial1
secret=vicidial1
type=friend
disallow=all
allow=ulaw
allow=alaw
qualify=yes
context=trunkinbound
insecure=very
canreinvite=no

protocol: IAX2
dialplan entry:
exten => _9XXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXX.,2,Dial(${elastix1}/${EXTEN:1},,To)
exten => _9XXX.,3,Hangup

server IP: 192.168.1.2
active: Y

the configuration trunk with elastix:
trunk name: vicidial1
CID options: allow all cid
Dialed number manipulation: 91|XXXXXXXXXX
Outgoing settings
Trunk name: vicidial1
Peer details:
host=192.168.1.2
type=friend
username=vicidial1
secret=vicidial1
disallow=all
allow=ulaw&alaw
qualify=yes
context=trunkinbound
insecure=very canreinvite=no
register string: vicidial1:vicidial1@192.168.1.4
:cry:
the configuration of the outbound route with elastix
route name: vicidial
dial patterns that will use this route:
91|XXXXXXXXXX
Trunk sequence for matched routes: 0 vicidial1.

In my elastix server I configure sip 101 when I try to call the agent of vicidial who is in the conference by this number 919998811112 or this number 918002277655 my x-lite tel me the line is busy try to call later and when I go to the asterisk CLI> of vicidial I type” iax2 show registry “ the vicibox answer host: 192.168.1.4 username: vicidial1 perceived: <Unregistred> state : rejected .
That’s the same of the elastix server .
Please I need your help for any suggestions that permit the call who came from elastix to mount of the interface agent vicidial :(
tresor
 
Posts: 21
Joined: Thu Oct 31, 2013 5:40 am

Re: call of elastix toward vicidial

Postby tresor » Mon Nov 25, 2013 2:54 am

in my elastix I'm not use free pbx
tresor
 
Posts: 21
Joined: Thu Oct 31, 2013 5:40 am

Re: call of elastix toward vicidial

Postby tresor » Mon Nov 25, 2013 2:56 am

please help
tresor
 
Posts: 21
Joined: Thu Oct 31, 2013 5:40 am

Re: call of elastix toward vicidial

Postby striker » Wed Nov 27, 2013 3:08 am

1. make sure you have proper trunking between asterisk server check the below link for the same
http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html

2. post the Asterisk cli output while dialling ( bothfrom vicidial as well elastix)
www.striker24x7.com www.youtube.com/c/striker24x7 Telegram/skype id : striker24x7
striker
 
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Re: call of elastix toward vicidial

Postby tresor » Wed Nov 27, 2013 12:26 pm

thanks for all i listen geofers he tel me that to configure two phone on vicidial and take one for the agent and the second for the caller and add this line under the default context in /etc/asterisk/extensions.conf file:
exten => DIDnumberAGI(agi_did_route.agi)

please help me how can i do to direct the inbound call from IVR
thanks for answer good night
tresor
 
Posts: 21
Joined: Thu Oct 31, 2013 5:40 am

Re: call of elastix toward vicidial

Postby williamconley » Wed Nov 27, 2013 3:48 pm

You seem to be discussing different aspects of the problem and jumping all over.

What is it that you are trying to accomplish? If you want to make test calls into the Vicidial system for training/testing purposes ... elastix is not necessary. For that you can just register a soft phone to a phone in admin->phones and change that phone's context to "trunkinbound" and then configure vicidial normally. When you want to run the test, this phone would be treated as a carrier and capable of dialing directly into any DID configured in Vicidial, thus simulating an inbound call without any external equipment except that soft phone. No cost, nothing off the network, single server (Just Vicidial) in use.

If that's not what you're trying to do ... what is?
Vicidial Installation and Repair, plus Hosting and Colocation
Newest Product: Vicidial Agent Only Beep - Beta
http://www.PoundTeam.com # 352-269-0000 # +44(203) 769-2294
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Re: call of elastix toward vicidial

Postby tresor » Thu Nov 28, 2013 10:45 am

thanks williamconley
now i don't want to use elastix. and i want that when i call the DID number of in group with my sip phone who configured in vicidial i listen a want call menu who purpose me to press one digits to joint the agent of the campaign inbound.
please help with the step to configure call menu for inbound call because it is not present to free vicidial manager manual .
thanks advance for your all answer.
tresor
 
Posts: 21
Joined: Thu Oct 31, 2013 5:40 am

Re: call of elastix toward vicidial

Postby williamconley » Sat Nov 30, 2013 8:33 pm

Modify a sip phone in the system by changing both the Exten context and Phone context to "trunkinbound". This phone is now your "Simulated Carrier Call" phone. Any call made on this phone will be viewed as an inbound call from a Carrier.

Configure your DID (Inbound -> Show DIDs) to route to a Call Menu. Configure the Call Menu (Inbound -> Show Call Menus) to point callers to the Ingroup in question.

then you dial that DID on your "Simulated Carrier Call" phone and you'll be presented with the menu.

Use the Vicidial Manager's Manual for all configuration except "Simulated Carrier Call" phone setup (which is not in the manual as far as I know).
Vicidial Installation and Repair, plus Hosting and Colocation
Newest Product: Vicidial Agent Only Beep - Beta
http://www.PoundTeam.com # 352-269-0000 # +44(203) 769-2294
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Re: call of elastix toward vicidial

Postby tresor » Sun Dec 01, 2013 7:17 am

thank you williamconley for your answer.
i have the new problem to download my audio file to the vicidial. i use ease audio converter to convert mp3 file to wave that use by vicidial and to settings by ease audio converter i put it in:
frquency(hz): 8000
channels: mono
output format: u_law and 8bit wave pcm
and when i test by call 8365 with my sip phone 103 the system said:

Executing [8365@default :1] playback(‘’sip/103-00000001’’, ‘’name of file’’) in new stack
WARNING[2270]: file.c:391 fn_wrapper: Unable to open format wav
WARNING[2270]:file.c:957 ast_streamfile: Unable to open name of file (format 0x4(ulaw)): No such file or directory
WARNING[2270]: app_playback.c:475 playback_exec: ast_streamfile failed on sip/103-00000001 for name of file. i add this file in extensions.conf file
please help me with the step to use ease audio converter to convert file to 16 bit mono 8k pcm wave audio file
thanks for answer
tresor
 
Posts: 21
Joined: Thu Oct 31, 2013 5:40 am

Re: call of elastix toward vicidial

Postby williamconley » Sun Dec 01, 2013 7:11 pm

1) Do not piggyback another non-related question onto an existing post. Make a new post with an appropriate subject line so Others may Benefit from your solution should one become available. Also: while creating your cool new subject line ... use it as a google search term before posting and you will often find your answer without a post OR find an identical problem and continue on that thread instead. But NOT on an unrelated thread (even if it's yours).

2) Dial 8167 (pass: 4321) to create sounds in Vicidial directly from any registered soft phone (except perhaps that fake carrier phone we just created ... which may not be able to do so!).

3) I note you showed the CLI output, but you skipped posting the configuration information ...

4) DO NOT modify extensions.conf. It's not upgrade proof after you do so and you'll likely bump into other problems created by doing so as well.

5) Whatever you're trying to accomplish, it is likely covered in the Vicidial Manager's Manual. Even the free version has a massive amount of setup information, but the paid version has three times as much and can save you hours of headache. That being said: What are you trying to accomplish this time? LOL (Create a new post with your goal and what you tried and how it failed ... and we'll help as usual!)
Vicidial Installation and Repair, plus Hosting and Colocation
Newest Product: Vicidial Agent Only Beep - Beta
http://www.PoundTeam.com # 352-269-0000 # +44(203) 769-2294
williamconley
 
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Joined: Wed Oct 31, 2007 4:17 pm
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