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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100214/b2f9f5d6/attachment.htm
Any help ? On Sun, Feb 14, 2010 at 3:03 PM, Global Meds <gm.cust3 at gmail.com> wrote:> > 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. >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100214/ddef3d22/attachment.htm
Can we debug or get the log file when we press any key in Xlite and which send to asterisk and the output we get ? On Sun, Feb 14, 2010 at 3:03 PM, Global Meds <gm.cust3 at gmail.com> wrote:> > 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. >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100215/dac2fa75/attachment.htm
Interestingly .Note this : Phone Number : 7274507674 Room ID: 6055 When I dial this number through Xlite and asterisk , on pressing any key , line get disconnect. When I dial this number through Skype, its perfect. Phone Number : 2127773456 When I dial this number through Xlite and asterisk , on pressing any key , line DOESNT get disconnect. When I dial this number through Skype, its perfect. On Sun, Feb 14, 2010 at 3:03 PM, Global Meds <gm.cust3 at gmail.com> wrote:> > 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. >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100215/e62657ec/attachment-0001.htm