extension.conf

[john_usc]

I am getting call declined error on my xlite phone and on cli> I get this

-- Registered IAX2 to '127.0.0.1', who sees us as 127.0.0.1:32768 with no messages waiting

-- Registered IAX2 to '127.0.0.1', who sees us as 127.0.0.1:32769 with no messages waiting

-- Executing AGI("SIP/2000-08242270", "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 Dial("SIP/2000-08242270", "Zap/g1/17201234567||To") in new stack
Jan 1 17:31:51 NOTICE[28525]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("SIP/2000-08242270", "") in new stack
== Spawn extension (default, 17201234567, 3) exited non-zero on 'SIP/2000-08242270'
-- Executing DeadAGI("SIP/2000-08242270", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0


XXXXXXXXXXXXXXXXXXXXX


here is the entry in extension.conf file

; dial a local 720 outbound number with area code
exten => _1720NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1720NXXXXXX,2,Dial(${TRUNK}/1${EXTEN:1},,To)
exten => _1720NXXXXXX,3,Hangup

here is my trunck

[globals]
CONSOLE=Console/dsp ; Console interface for demo

SIPtrunk=SIP/login@voicepulse.com ; SIP trunk



thanks in advance

[newx22]

-- Executing Dial("SIP/2000-08242270", "Zap/g1/17203334762||To") in new stack


your dialer is "Executing Dial over a Zap trunk which is a zapata device which you don't have because you are on sip trunk"

so this :

Code: Select all

exten => _1720NXXXXXX,2,Dial(${TRUNK}/1${EXTEN:1},,To)

should be changed to something like:

Code: Select all

; 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(sip/${EXTEN:1}@SIPtrunk,55,o)
; exten => _91NXXNXXXXXX,3,Hangup


where in your extentions.conf :

Code: Select all

SIPtrunk=SIP/1234:PASSWORD@sip.provider.net

[williamconley]

exten => _1720NXXXXXX,2,Dial(${TRUNK}/1${EXTEN:1},,To)

[globals]
SIPtrunk=SIP/login@voicepulse.com

You have 'TRUNK' in use as your dialout trunk, but you do not show the definition of 'TRUNK' in your [GLOBALS] definition.

You also have 'SIPtrunk' defined, but i do not see it in use in your dial plan.

Perhaps you could bring these together by changing one to match the other.

On the other hand, you could go back to the install and go over the section where you set up and define your trunks and go over it one step at a time and you may resolve your issue cleanly and properly for the next time, and have a good understanding of this step. It is kind of important, and easy after you learn it.

[john_usc]

Thank you guys. I figured that one out. Now the problem is I can not hear anything. I seen your reply in my other post but I thought I would post here too in case you guys check here.
Thanks a millions guys

== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Executing AGI("Local/8600051@default-f8db,1", "agi://127.0.0.1:4577/call_log") in new stack
-- Playing 'conf-onlyperson' (language 'en')
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-f8db,1", "sip/13038003432@voicepulse-primary|55|o") in new stack
-- Called 113038003432@voicepulse-primary
-- SIP/voicepulse-primary-09e7b5c8 is making progress passing it to Local/8600051@default-f8db,1
Jan 3 13:34:22 WARNING[14873]: file.c:1045 ast_waitstream: Unexpected control subclass '14'
Jan 3 13:34:22 NOTICE[2336]: pbx.c:1761 pbx_extension_helper: Cannot find extension context 'banned'
-- SIP/voicepulse-primary-09e7b5c8 is ringing
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/voicepulse-primary-09e7b5c8 answered Local/8600051@default-f8db,1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 917203334828, 2) exited non-zero on 'Local/8600051@default-f8db,1'
-- Executing DeadAGI("Local/8600051@default-f8db,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----26-----6") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --26-----6 completed, returning 0
-- Hungup 'Zap/pseudo-1767787704'
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-f8db,2'
-- Executing DeadAGI("Local/8600051@default-f8db,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
== Manager 'sendcron' logged off from 127.0.0.1

[williamconley]

Do you have the appropriate ports open through your firewall?

[john_usc]

here is a complete output

[root@vici ~]# asterisk -r
Asterisk 1.2.27, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.27 currently running on vici (pid = 2326)
Verbosity is at least 21
-- Remote UNIX connection
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
Jan 3 13:33:43 NOTICE[14760]: channel.c:2514 __ast_request_and_dial: Unable to request channel SIP/cc100
== 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
-- Executing MeetMeAdmin("Local/55558600051@default-6f94,2", "8600051|K") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
Jan 3 13:33:45 NOTICE[14773]: app_meetme.c:2210 admin_exec: Conference Number n ot found
-- Executing Hangup("Local/55558600051@default-6f94,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/5555860 0051@default-6f94,2'
-- Executing DeadAGI("Local/55558600051@default-6f94,2", "agi://127.0.0.1:45 77/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----1 6--------------- completed, returning 0
== Manager 'sendcron' logged off 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
Jan 3 13:33:54 NOTICE[14791]: channel.c:2514 __ast_request_and_dial: Unable to request channel SIP/cc100
== 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
== 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-f8db,2", "8600051|F") in new stack
> Channel Local/8600051@default-f8db,1 was answered.
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Executing AGI("Local/8600051@default-f8db,1", "agi://127.0.0.1:4577/call_log") in new stack
-- Playing 'conf-onlyperson' (language 'en')
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-f8db,1", "sip/13038009876@voicepulse-primary|55|o") in new stack
-- Called 13038009876@voicepulse-primary
-- SIP/voicepulse-primary-09e7b5c8 is making progress passing it to Local/8600051@default-f8db,1
Jan 3 13:34:22 WARNING[14873]: file.c:1045 ast_waitstream: Unexpected control subclass '14'
Jan 3 13:34:22 NOTICE[2336]: pbx.c:1761 pbx_extension_helper: Cannot find extension context 'banned'
-- SIP/voicepulse-primary-09e7b5c8 is ringing
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/voicepulse-primary-09e7b5c8 answered Local/8600051@default-f8db,1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 913038009876, 2) exited non-zero on 'Local/8600051@default-f8db,1'
-- Executing DeadAGI("Local/8600051@default-f8db,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----26-----6") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --26-----6 completed, returning 0
-- Hungup 'Zap/pseudo-1767787704'
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-f8db,2'
-- Executing DeadAGI("Local/8600051@default-f8db,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
== Manager 'sendcron' logged off from 127.0.0.1
== Refreshing DNS lookups.


Thanks

[john_usc]

Do you have the appropriate ports open through your firewall?

How would I check that. I am pretty sure there was a firewall file in centos don't remeber the name now, and I set to allow everything even mysql

not sure how else can I check

Thanks

[williamconley]

Do you have a router?

[john_usc]

Yea I have a Dlink router. But my call gets connected fine and from the agent screen if I send dtmf tones I can hear those too. So I think the ports are not blocked, unless I am missing something.
Thanks

[williamconley]

What ports do you have forwarded through your firewall?

[john_usc]

I am using DMZ so all the ports are forwarded to the server.
Thanks

[williamconley]

Jan 3 13:33:43 NOTICE[14760]: channel.c:2514 __ast_request_and_dial: Unable to request channel SIP/cc100

Jan 3 13:33:45 NOTICE[14773]: app_meetme.c:2210 admin_exec: Conference Number n ot found
-- Executing Hangup("Local/55558600051@default-6f94,2", "") in new stack

Jan 3 13:33:54 NOTICE[14791]: channel.c:2514 __ast_request_and_dial: Unable to request channel SIP/cc100


Jan 3 13:34:22 WARNING[14873]: file.c:1045 ast_waitstream: Unexpected control subclass '14'

Jan 3 13:34:22 NOTICE[2336]: pbx.c:1761 pbx_extension_helper: Cannot find extension context 'banned'

== Spawn extension (default, 913038009876, 2) exited non-zero on 'Local/8600051@default-f8db,1'

== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-f8db,2'

You have a few problems in your system. You need to kill them one at a time. are you using SIP/cc100? are your meetme rooms working?

[john_usc]

But I am not sure why can I not hear the voice. I can even hear the DTMF if entered by the agent on the agent GUI. Why would voice not go. Do we need to set up our headphones or sumthing? Does it have a way to do it from GUI for admin? What on earth can I do. Should I re-install. Would hate to do it if it was a minor issue

[williamconley]

depending on how your dtmf is being transmitted, it may go through on a different port along with your data headers. so sound may be blocked while dtmf transmits. which is why i asked about your ports. make sure 5060 and 10000-20000 are explicitly forwarded to the server through your firewall.