[asterisk-users] Line DC

Sun Feb 14 03:33:53 CST 2010

My dialer works perfectly , but whenever I dial a number manually from xlite
and press a Key like 6055 for DTMF , line gets disconnected. Line gets DC as
soon as I press any key from xlite

What could be the issues ?

I tried the SAME VOIP from another center and Its Ok there.

I tried the Same dialer Xlite over Static IP, problem is there.

I tried the same number from other Dialer , it works perfectly.

Normal Hang Up :
-----------------------------

Quote:

vici*CLI>
-- Executing AGI("SIP/cc101-09f44300", "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/cc101-09f44300", "SIP/VOIP74/17274507674||tTor") in
new stack
-- Called VOIP74/17274507674
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/VOIP74-09ecb770 is ringing
-- SIP/VOIP74-09ecb770 is making progress passing it to SIP/cc101-09f44300
-- SIP/VOIP74-09ecb770 is ringing
-- SIP/VOIP74-09ecb770 is making progress passing it to SIP/cc101-09f44300
Feb 14 01:50:21 NOTICE[24692]: rtp.c:331 process_rfc3389: Comfort noise
support incomplete in Asterisk (RFC 3389). Please turn off on client if
possible. Client IP: 74.222.1.92
-- SIP/VOIP74-09ecb770 answered SIP/cc101-09f44300
== Spawn extension (default, 9117274507674, 2) exited non-zero on
'SIP/cc101-09f44300'
-- Executing DeadAGI("SIP/cc101-09f44300", "agi://
127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----21-----11")
in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -21-----11
completed, returning 0
vici*CLI>

Quote:

Hang Up when pressed any key from the soft Phone:
-------------------------------------------------------------------------------

vici*CLI>
-- Executing AGI("SIP/cc101-09f44300", "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/cc101-09f44300", "SIP/VOIP74/17274507674||tTor") in
new stack
-- Called VOIP74/17274507674
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/VOIP74-09ecb770 is ringing
-- SIP/VOIP74-09ecb770 is making progress passing it to SIP/cc101-09f44300
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/VOIP74-09ecb770 is ringing
-- SIP/VOIP74-09ecb770 is making progress passing it to SIP/cc101-09f44300
Feb 14 01:51:16 NOTICE[24845]: rtp.c:331 process_rfc3389: Comfort noise
support incomplete in Asterisk (RFC 3389). Please turn off on client if
possible. Client IP: 74.222.1.92
-- SIP/VOIP74-09ecb770 answered SIP/cc101-09f44300
== Spawn extension (default, 9117274507674, 2) exited non-zero on
'SIP/cc101-09f44300'
-- Executing DeadAGI("SIP/cc101-09f44300", "agi://
127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----22-----10")
in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -22-----10
completed, returning 0
vici*CLI>

Dial Plan :

register =>user:pass123 at 74.222.1.92:5060

[VOIP74_7]
disallow=all
allow=g729
allow=g711
allow=ulaw
type=friend
username=user
secret=password
host=74.222.1.92
dtmfmode=rfc2833

SIP74_7 = SIP/VOIP74_7

exten => _7X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _7X.,2,Dial(${SIP74_7}/${EXTEN:2},,tTor)
exten => _7X.,3,Hangup

Please guide me .

Entry from Master.csv

Quote:

""cc101"
<cc101>","cc101","9117274507674","default","SIP/cc101-09f44300","SIP/VOIP74-09ecb770","DeadAGI","agi://
127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----","2010-02-14
01:47:02","2010-02-14 01:47:14","2010-02-14
01:47:19","17","5","ANSWERED","DOCUMENTATION","","1266130022.0",""
""cc101"
<cc101>","cc101","9119545090201","default","SIP/cc101-09f44300","SIP/VOIP74-09ecb770","DeadAGI","agi://
127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----","2010-02-14
01:47:35","2010-02-14 01:47:38","2010-02-14
01:47:42","7","4","ANSWERED","DOCUMENTATION","","1266130055.2",""
""cc101"
<cc101>","cc101","9119545090201","default","SIP/cc101-09f44300","SIP/VOIP74-09ecb770","DeadAGI","agi://
127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----","2010-02-14
01:48:06","2010-02-14 01:48:09","2010-02-14
01:48:14","8","5","ANSWERED","DOCUMENTATION","","1266130086.4",""
""cc101"
<cc101>","cc101","9117274507674","default","SIP/cc101-09f44300","SIP/VOIP74-09ecb770","Dial","SIP/VOIP74/17274507674||tTor","2010-02-14
01:48:24","2010-02-14 01:48:35","2010-02-14
01:48:38","14","3","ANSWERED","DOCUMENTATION","","1266130104.6",""
""cc101"
<cc101>","cc101","9117274507674","default","SIP/cc101-09f44300","SIP/VOIP74-09ecb770","DeadAGI","agi://
127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----","2010-02-14
01:48:43","2010-02-14 01:48:55","2010-02-14
01:48:57","14","2","ANSWERED","DOCUMENTATION","","1266130123.8",""
""cc101"
<cc101>","cc101","9117274507674","default","SIP/cc101-09f44300","SIP/VOIP74-09ecb770","Dial","SIP/VOIP74/17274507674||tTor","2010-02-14
01:50:12","2010-02-14 01:50:22","2010-02-14
01:50:33","21","11","ANSWERED","DOCUMENTATION","","1266130212.10",""
""cc101"
<cc101>","cc101","9117274507674","default","SIP/cc101-09f44300","SIP/VOIP74-09ecb770","DeadAGI","agi://
127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----","2010-02-14
01:51:05","2010-02-14 01:51:17","2010-02-14
01:51:27","22","10","ANSWERED","DOCUMENTATION","","1266130265.12",""
""cc101"
<cc101>","cc101","7117274507674","default","SIP/cc101-09f44300","SIP/VOIP74_7-09ecb770","DeadAGI","agi://
127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----","2010-02-14
01:57:20","2010-02-14 01:57:32","2010-02-14
01:57:36","16","4","ANSWERED","DOCUMENTATION","","1266130640.14",""
""cc101"
<cc101>","cc101","7117274507674","default","SIP/cc101-09f44300","SIP/VOIP74_7-09ecb770","Dial","SIP/VOIP74_7/17274507674||tTor","2010-02-14
02:00:57","","2010-02-14 02:00:59","2","0","NO
ANSWER","DOCUMENTATION","","1266130857.16",""

Also, I see that my event log file size is 0.
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