Moderators: gerski, enjay, williamconley, Op3r, Staydog, gardo, mflorell, MJCoate, mcargile, Kumba, Michael_N
mflorell wrote:Did you install the Net::Telnet perl module on this system?
What instructions did you follow to install ViciDial?
proxium wrote:sip show peers
Name/username Host Dyn Nat ACL Port Status
tofreepbx1/400 192.168.1.89 N 5060 OK (1 ms)
from-trunk (Unspecified) 5060 Unmonitored
2000/2000 (Unspecified) D N A 0 UNKNOWN
1111/1111 192.168.1.113 D N A 5060 OK (3 ms)
1000 (Unspecified) D N A 0 UNKNOWN
5 sip peers [Monitored: 2 online, 2 offline Unmonitored: 1 online, 0 offline]
I have a trunk on 192.168.1.89 to simulate a call between two AsteriskServer
proxium wrote:SELECT extension FROM phones where protocol = 'SIP' and server_ip='10.48.10.8'
this query normally return no record in your database, but you need to check the VARserver_ip in /etc/astguiclient.conf and replace with the IP of your Asterisk Server then try to update your "phones "table and assign the same value to "server_ip" field, wich is the server that contains all your phone extensions.
If this return valid value, then perl script will use Telnet module to register the phone in Asterisk side and start calling.
[default]
include => ext-local
; Extension 8600 + 8601 conference rooms
exten => 8600,1,Meetme,8600
exten => 8601,1,Meetme,8601
'; Extension 129
exten => 129,1,Playback,transfer|skip ; "Please hold while..."
exten => 129,2,Dial,sip/129|20|to ; Ring, 20 secs max
exten => 129,3,Voicemail,u129 ; Send to voicemail...
; # timeout invalid rules
exten => #,1,Playback(invalid) ; "Thanks for trying the demo"
exten => #,2,Hangup ; Hang them up.
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
; Give voicemail at extension 8500
exten => 8500,1,VoicemailMain?
exten => 8500,2,Goto(s,6)
; ASTERISK AGENTS LOGINS FOR QUEUES (NOT part of VICIDIAL)
; the following assumes phone agent login and exten are 3 digits and the same
; also assumes that 3-digit login is present in agents.conf and queueus.conf
;Agent Logout then stay onhook, DIAL 54 + 3-digit ID
exten => _54XXX,1,AgentCallbackLogin()
; the following are used to login and logout of Asterisk Queues from phone
;Agent Login then stay offhook on the phone, DIAL 55 + 3-digit ID
exten => _55XXX,1,AgentLogin(${EXTEN:1})
;Agent Login then stay onhook, phones will ring, DIAL 56 + 3-digit ID
exten => _56XXX,1,AgentCallbackLogin(${EXTEN:1}@default)
#include vicidial_extensions.conf
[custom-vicidial]
include => default
[from-internal-custom]
include => default
2009-04-23 16:05:35|SERVER CALLS PER SECOND MAXIMUM SET TO: 20 |50||
2009-04-23 16:05:35|LIVE AGENTS LOGGED IN: 0 ACTIVE CALLS: 0|
2009-04-23 16:05:35| : agents: dial_level: 0|
2009-04-23 16:05:35| : Calls to place: 0 (0 - 0) 0 |
2009-04-23 16:05:35|CAMPAIGN DIFFERENTIAL: 0 0 (0 - 0)|
2009-04-23 16:05:35|LOCAL TRUNK SHORTAGE: 0|0 (0 - 23)|
2009-04-23 16:05:35| : CALLING|
2009-04-23 16:05:35|| lagged call vla agent PAUSED 0E0|20090423160505|20090423160525|20090423160535||
2009-04-23 16:05:35|| lagged call vac agent DELETED 0E0|2009-04-23 16:03:35||
2009-04-23 16:02:02||TESTCAMP|Added to hopper 0||
2009-04-23 16:02:03||TESTCAMP|50|2|24hours|||
2009-04-23 16:02:03||TESTCAMP|Added to hopper 0||
2009-04-23 16:03:02||TESTCAMP|50|2|24hours|||
2009-04-23 16:03:02||TESTCAMP|Added to hopper 0||
2009-04-23 16:03:03||TESTCAMP|50|2|24hours|||
2009-04-23 16:03:03||TESTCAMP|Added to hopper 0||
2009-04-23 16:04:03||TESTCAMP|50|2|24hours|||
2009-04-23 16:04:03||TESTCAMP|Added to hopper 0||
2009-04-23 16:04:03||TESTCAMP|50|2|24hours|||
2009-04-23 16:04:03||TESTCAMP|Added to hopper 0||
2009-04-23 16:05:01||TESTCAMP|50|2|24hours|||
2009-04-23 16:05:01||TESTCAMP|Added to hopper 0||
2009-04-23 16:05:02||TESTCAMP|50|2|24hours|||
2009-04-23 16:05:02||TESTCAMP|Added to hopper 0||
2009-04-23 16:06:02||TESTCAMP|50|2|24hours|||
2009-04-23 16:06:02||TESTCAMP|Added to hopper 0||
2009-04-23 16:06:03||TESTCAMP|50|2|24hours|||
2009-04-23 16:06:03||TESTCAMP|Added to hopper 0||
2009-04-23 16:07:02||TESTCAMP|50|2|24hours|||
2009-04-23 16:07:02||TESTCAMP|Added to hopper 0||
2009-04-23 16:07:02||TESTCAMP|50|2|24hours|||
2009-04-23 16:07:02||TESTCAMP|Added to hopper 0||
I'm happy to work with you on the "all-in-one" with vici built in. But when i add vici, i keep vici / freepbx separate (they merely coexist). I was using freepbx to create trunks and extensions.wizzbangca wrote:Op3r, can't disagree with you. In large centers, that's the way I am going. But gotta have an all in one solution.
mflorell wrote:So, you're using FreePBX...
That could cause all sorts of problems.
In the Asterisk CLI type "sip show peers" and post what the entry for that phone looks like.
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