Cannot Route inbound Call to destination problem.

[andy]

Dear all,

We are the new comer on vicidial. We just started lab test on them.
We already installed and config by follow manual V.2.4. Our team face a problem about making inbound call.
Call cannot route to destination Agent Extension

- We registered SIP gateway Server as it own server(Registered on itself IPAddress:192.168.1.64).
- We used extesion to make inbound call to DID of carrier trunk SIP (Extension:131).
When we making a inbound call, call doesnot route to agent group.

Our expected scenario :
extesion:132 make inbound call dial to -->DID:131(Sip CarrierTrunk)-->Agent-Extesion 201

but the real situation we faced :
extesion:132 make inbound call dial to --//-->DID:131(Sip CarrierTrunk)--//-->Agent-Extesion 201


These following are our steps setting:
1. added Admin --- > Phones
- add extension 201,131,132

2. config Carrier as Newsip Server Ipaddress 192.168.1.64:5060

Code: Select all

Carrier ID:  NEWSIP
Carrier Name:      ViciDial SIP carrier
Carrier Description:     
Admin User Group:   ---ALL---   
Registration String:      register => 131:1234@192.168.1.64:5060
Template ID:   --NONE--
Account Entry:     [131]
      disallow=all
      allow=ulaw
      allow=speex
      type=friend
      username=131
      secret=1234
      host=dynamic
      dtmfmode=rfc2833
      context=trunkinbound   
Protocol:  SIP
Globals String:     
Dialplan Entry:  exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(131:1234@192.168.1.64:5060/${EXTEN:2},,tTor)
exten => _91NXXNXXXXXX,3,Hangup   
Server IP:   192.168.1.64 - Server linux-ssk - 192.168.1.64   
Active:  Y


3. add new in group as SALESLINE

Code: Select all

Group ID:  SALESLINE   
Group Name:      Primary Sales Line
Group Color:      red
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:  NONE
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:  NONE
Timer Action Message:     
Timer Action Seconds:      -1
Timer Action Destination:     
Drop Call Seconds:      360
Drop Action:  MESSAGE
Drop Exten:      8307
Voicemail:   
Drop Transfer Group:  ---NONE---   
Drop Call Menu:
Call Time:   24hours - default 24 hours calling
Action Transfer CID:   CUSTOMER
After Hours Action:  MESSAGE
After Hours Message Filename: VM-GOODBYE
After Hours Extension:      8300
After Hours Voicemail:   voicemail chooser   
After Hours Transfer Group:  ---NONE---   
After Hours Call Menu:
No Agents No Queueing:  NO
No Agent No Queue Action:  MESSAGE
Audio File:  nbdy-avail-to-take-call|vm-goodbye
Max Calls Method:  DISABLED
Max Calls Count:      0
Max Calls Action:  NO_AGENT_NO_QUEUE
Welcome Message Filename:   --NONE--
Play Welcome Message:  ALWAYS
Music On Hold Context:   default
On Hold Prompt Filename: generic_hold
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:   
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:   ---NONE---   
Wait Time Option Voicemail: 
Wait Time Option Transfer In-Group:  ---NONE---   
Wait Time Option Press Filename:   to-be-called-back|digits/1
Wait Time Option Press No Block:  N
Wait Time Option Press Filename Seconds:      10
Wait Time Option After Press Filename:   vm-hungup
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:   ---NONE---   
Hold Time Option Voicemail:   
Hold Time Option Transfer In-Group:  ---NONE---   
Hold Time Option Press Filename:   to-be-called-back|digits/1
Hold Time Option Press No Block:  N
Hold Time Option Press Filename Seconds:      10
Hold Time Option After Press Filename:   vm-hungup
Hold Time Option Callback List ID:      999
Agent Alert Filename:   ding
Agent Alert Delay:      1000
Default Transfer Group:  ---NONE---   
Default Group Alias:  NONE   
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:  N   
Uniqueid Status Display:  DISABLED
Uniqueid Status Prefix:     


4. add DID as 131

Code: Select all

DID Extension: 131
DID Description:    Inbound 800 number 
Active:  Y
Admin User Group:   All Admin User Groups ADMIN - VICIDIAL ADMINISTRATORS AGENTS -

ViCiDiAL AGENTS ---ALL---   
DID Route:  IN_GROUP
Record Call:  N   
Extension: 9998811112
Extension Context:      default
Voicemail Box:   
Phone Extension:     
Server IP:
Call Menu: 
User Agent:     
User Unavailable Action:  VOICEMAIL   
User Route Settings In-Group:  AGENTDIRECT - Single Agent Direct Queue
In-Group ID:  SALESLINE - Primary Sales Line
In-Group Call Handle Method:  CID
In-Group Agent Search Method:  LB
In-Group List ID:     999
In-Group Campaign ID:     
In-Group Phone Code:  1   
Clean CID Number:     
Filter Inbound Number:  DISABLED
Filter Phone Group ID:     
Filter URL:     
Filter Action:  EXTEN   
Filter Extension:      9998811112
Filter Extension Context:      default
Filter Voicemail Box:   
Filter Phone Extension:     
Filter Server IP: 
Filter Call Menu: 
Filter User Agent:     
Filter User Unavailable Action:  VOICEMAIL   
Filter User Route Settings In-Group:  AGENTDIRECT - Single Agent Direct Queue
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:     


5. add test_in campaign

Code: Select all

Campaign ID:  TEST_IN   
Campaign Name:      Closer and inbound campaign
Campaign Description:     
Campaign Change Date:  2012-09-11 21:11:07     
Campaign Login Date:  2012-09-11 22:31:35     
Active:  Y
Admin User Group:  ---ALL---   
Park Music-on-Hold:     
Web Form:     
Allow Closers:  Y   
Default Transfer Group:  ---NONE---   
Allow Inbound and Blended:  N   
Dial Status 1:  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:  N   
Dial Method:  MANUAL
Auto Dial Level:  0
Adapt Intensity Modifier:   0 - Balanced
Script:   
Get Call Launch:  NONE


6. edit value in detail view

Allow Inbound and Blended: Y
Dial Method: RATIO
and check box SALESLINE in allowed transfer groups

7. added user 7777

User Number: 7777
Password: test
Force Change Password: N
Full Name:
User Level: 1
User Group: AGENTS
Phone Login:
Phone Pass:
Active: Y

8. We tried to make inbound call from "extension:132" to "DID:131".
Found that call route to "voice mail box" as context default.
Then Changed Context value from "default" to be "Trunkinbound" within Admin-->Phone of context of "131" then retest again

CLI Log

Code: Select all

linux-ssk*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
gs102/gs102                (Unspecified)    D   N      0        UNKNOWN
201/201                    (Unspecified)    D   N      0        UNKNOWN
139/139                    (Unspecified)    D   N      0        UNKNOWN
133/133                    192.168.1.54     D   N      2274     UNREACHABLE
132/132                    192.168.1.53     D   N      5060     OK (5 ms)
131/131                    192.168.1.64     D   N      5060     OK (1 ms)
6 sip peers [Monitored: 2 online, 4 offline Unmonitored: 0 online, 0 offline]

[Sep 15 19:05:58]     -- Executing [131@default:1] Dial("SIP/132-00000004", "SIP/131|60|") in

new stack
[Sep 15 19:05:58]     -- Called 131
[Sep 15 19:05:58]     -- Got SIP response 482 "Loop Detected" back from 192.168.1.64
[Sep 15 19:05:58]     -- Now forwarding SIP/132-00000004 to 'Local/131@default' (thanks to

SIP/131-00000005)
[Sep 15 19:05:58]     -- Executing [131@default:1] Dial("Local/131@default-95c4,2", "SIP/131|

60|") in new stack
[Sep 15 19:05:58] WARNING[4489]: app_dial.c:1277 dial_exec_full: Skipping dialing interface

'SIP/131' again since it has already been dialed
[Sep 15 19:05:58]   == Spawn extension (default, 131, 1) exited non-zero on

'Local/131@default-95c4,2'
[Sep 15 19:05:58]     -- Executing [h@default:1] DeadAGI("Local/131@default-95c4,2",

"agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CHANUNAVAIL----------") in

new stack
[Sep 15 19:05:58]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----

NODEBUG-----0-----CHANUNAVAIL---------- completed, returning 0
[Sep 15 19:05:58]   == Everyone is busy/congested at this time (1:0/0/1)
[Sep 15 19:05:58]     -- Executing [131@default:2] Goto("SIP/132-00000004", "default|

85026666666666131|1") in new stack
[Sep 15 19:05:58]     -- Goto (default,85026666666666131,1)
[Sep 15 19:05:58]     -- Executing [85026666666666131@default:1] Wait("SIP/132-00000004",

"1") in new stack
[Sep 15 19:05:59]     -- Executing [85026666666666131@default:2] VoiceMail("SIP/132-

00000004", "131|u") in new stack
[Sep 15 19:05:59]     -- <SIP/132-00000004> Playing 'vm-theperson' (language 'en')
[Sep 15 19:06:01]     -- <SIP/132-00000004> Playing 'digits/1' (language 'en')
[Sep 15 19:06:01]     -- <SIP/132-00000004> Playing 'digits/3' (language 'en')
[Sep 15 19:06:01]   == Parsing '/etc/asterisk/manager.conf': [Sep 15 19:06:01] Found
[Sep 15 19:06:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 15 19:06:01]   == Parsing '/etc/asterisk/manager.conf': [Sep 15 19:06:01] Found
[Sep 15 19:06:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 15 19:06:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 15 19:06:02]     -- <SIP/132-00000004> Playing 'digits/1' (language 'en')
[Sep 15 19:06:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 15 19:06:03]     -- <SIP/132-00000004> Playing 'vm-isunavail' (language 'en')
[Sep 15 19:06:04]     -- <SIP/132-00000004> Playing 'vm-intro' (language 'en')
[Sep 15 19:06:06]   == Spawn extension (default, 85026666666666131, 2) exited non-zero on

'SIP/132-00000004'
[Sep 15 19:06:06]     -- Executing [h@default:1] DeadAGI("SIP/132-00000004",

"agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CHANUNAVAIL----------") in

new stack
[Sep 15 19:06:06]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----

NODEBUG-----0-----CHANUNAVAIL---------- completed, returning 0


Would you kind give us some hint or suggestion for solving this problem.
Thankz you in advanced.

[williamconley]

Agents do not have "extensions", they have phones and agent IDs. Agent IDs are not phones.

When agents dial out, they do not "dial out" on DIDs, they dial out on Carriers. Those carriers do not route the calls back to the system.

You may be a person familiar with systems such as FreePBX. I think you would do very well to start your install over, download the Vicidial Manager's Manual, and begin at the start of the Manager's Manual and work your way to the end. Do not skip any of the pages/sections/tutorials. And most importantly do not attempt to apply anything you know from FreePBX/Asterisk to this system.

If you do that, you should end up with a working system and then you can ask questions about how to make it do what you want it to do in addition to what it already does. One thing at a time.

Otherwise, if you insist on managing the system the way you are ... I'd have to know why you want one phone to call another? (This is a Dialer ... not a Generic PBX, agents do not call each other on their phones, as a rule. They may transfer calls to one another, but that is done in the Agent Logged In Screen ... in a completely different method, tracked and logged ...)

Thanks for posting your specs! That's helpful, but I'd love to hear why you are trying to use Vicidial and "convert it" to a FreePBX system instead of using Vicidial as designed and finding out what it can do ... (not saying what you're doing is impossible, just that if you want FreePBX ... install FreePBX! LOL).