Re: [asterisk-users] MeetMe not working with GSM codec?
can you look on this from your debug 1. app_meetme.c:3030 find_conf: The requested confno is '12'? 2. == Parsing '/etc/asterisk/meetme.conf': [May 21 09:33:23] DEBUG[6872]: config.c:1306 config_text_file_load: Parsing /etc/asterisk/meetme.conf 3. == Found 4. [May 21 09:33:23] DEBUG[6872]: app_meetme.c:3082 find_conf: 12 isn't a valid conference its on line number 318 it seems that you doesent specify valid conference number can you post meetme.conf regards Dhaval On Thu, May 21, 2009 at 2:26 PM, Chris Maciejewski ch...@wima.co.uk wrote: Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: sip.conf: [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk [1] type=friend secret=test host=dynamic nat=yes -- - extensions.conf: - [common] exten = 501,1,MeetMe(12,MI) exten = 501,n,Hangup() exten = i,1,Hangup() exten = h,1,Hangup() exten = t,1,Hangup() Everything works OK when ALAW is used - see http://pastebin.com/f7222a6d3 but with GSM Asterisk hangs up just after starting MeetMe application - see http://pastebin.com/f78d04c95 line 327. Is there a problem with MeetMe app or I need to adjust my configuration? Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe not working with GSM codec?
Hi Martin, Yes, I do have GSM compiled for sure. $asterisk -r -x core show codecs audio Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARYHEX TYPE NAME DESC 1 (1 0) (0x1) audio g723 (G.723.1) 2 (1 1) (0x2) audiogsm (GSM) 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 16 (1 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8)(0x100) audio g729 (G.729A) 512 (1 9)(0x200) audio speex (SpeeX) 1024 (1 10)(0x400) audio ilbc (iLBC) 2048 (1 11)(0x800) audio g726 (G.726 RFC3551) 4096 (1 12) (0x1000) audio g722 (G722) I will open a bug report. Regards, Chris 2009/5/22 Martin asteriskl...@callthem.info: it should work just fine; do you have the GSM codec compiled/loaded core show modules like codec_gsm ... ? OR that particular version has a BUG... Martin On Thu, May 21, 2009 at 3:56 AM, Chris Maciejewski ch...@wima.co.uk wrote: Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: sip.conf: [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk [1] type=friend secret=test host=dynamic nat=yes -- - extensions.conf: - [common] exten = 501,1,MeetMe(12,MI) exten = 501,n,Hangup() exten = i,1,Hangup() exten = h,1,Hangup() exten = t,1,Hangup() Everything works OK when ALAW is used - see http://pastebin.com/f7222a6d3 but with GSM Asterisk hangs up just after starting MeetMe application - see http://pastebin.com/f78d04c95 line 327. Is there a problem with MeetMe app or I need to adjust my configuration? Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe not working with GSM codec?
Hi Dhaval, The reason confno '12' is not found in meetme.conf is because I am using MySQL as realtime config backend. See few lines below there is: [May 21 09:33:23] DEBUG[6872]: res_config_mysql.c:1478 mysql_reconnect: MySQL RealTime: Connection okay. [May 21 09:33:23] DEBUG[6872]: res_config_mysql.c:365 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM conference WHERE confno = '12' My meetme.conf: [general] audiobuffers=32 logmembercount=yes schedule=no 2009/5/22 DHAVAL INDRODIYA dhaval.it01...@gmail.com: can you look on this from your debug app_meetme.c:3030 find_conf: The requested confno is '12'? == Parsing '/etc/asterisk/meetme.conf': [May 21 09:33:23] DEBUG[6872]: config.c:1306 config_text_file_load: Parsing /etc/asterisk/meetme.conf == Found [May 21 09:33:23] DEBUG[6872]: app_meetme.c:3082 find_conf: 12 isn't a valid conference its on line number 318 it seems that you doesent specify valid conference number can you post meetme.conf regards Dhaval On Thu, May 21, 2009 at 2:26 PM, Chris Maciejewski ch...@wima.co.uk wrote: Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: sip.conf: [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk [1] type=friend secret=test host=dynamic nat=yes -- - extensions.conf: - [common] exten = 501,1,MeetMe(12,MI) exten = 501,n,Hangup() exten = i,1,Hangup() exten = h,1,Hangup() exten = t,1,Hangup() Everything works OK when ALAW is used - see http://pastebin.com/f7222a6d3 but with GSM Asterisk hangs up just after starting MeetMe application - see http://pastebin.com/f78d04c95 line 327. Is there a problem with MeetMe app or I need to adjust my configuration? Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe not working with GSM codec?
On an entirely unrelated note, do you have the gsm asterisk sounds installed? Maybe that vm-*.slin files dont exist. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Maciejewski Sent: Friday, May 22, 2009 12:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MeetMe not working with GSM codec? Hi Dhaval, The reason confno '12' is not found in meetme.conf is because I am using MySQL as realtime config backend. See few lines below there is: [May 21 09:33:23] DEBUG[6872]: res_config_mysql.c:1478 mysql_reconnect: MySQL RealTime: Connection okay. [May 21 09:33:23] DEBUG[6872]: res_config_mysql.c:365 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM conference WHERE confno = '12' My meetme.conf: [general] audiobuffers=32 logmembercount=yes schedule=no 2009/5/22 DHAVAL INDRODIYA dhaval.it01...@gmail.com: can you look on this from your debug app_meetme.c:3030 find_conf: The requested confno is '12'? == Parsing '/etc/asterisk/meetme.conf': [May 21 09:33:23] DEBUG[6872]: config.c:1306 config_text_file_load: Parsing /etc/asterisk/meetme.conf == Found [May 21 09:33:23] DEBUG[6872]: app_meetme.c:3082 find_conf: 12 isn't a valid conference its on line number 318 it seems that you doesent specify valid conference number can you post meetme.conf regards Dhaval On Thu, May 21, 2009 at 2:26 PM, Chris Maciejewski ch...@wima.co.uk wrote: Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: sip.conf: [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk [1] type=friend secret=test host=dynamic nat=yes -- - extensions.conf: - [common] exten = 501,1,MeetMe(12,MI) exten = 501,n,Hangup() exten = i,1,Hangup() exten = h,1,Hangup() exten = t,1,Hangup() Everything works OK when ALAW is used - see http://pastebin.com/f7222a6d3 but with GSM Asterisk hangs up just after starting MeetMe application - see http://pastebin.com/f78d04c95 line 327. Is there a problem with MeetMe app or I need to adjust my configuration? Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe not working with GSM codec?
this command doesn't show the codecs present in the system do you have g723 compiled too ? try core show translations or something like that Martin On Fri, May 22, 2009 at 2:25 AM, Chris Maciejewski ch...@wima.co.uk wrote: Hi Martin, Yes, I do have GSM compiled for sure. $asterisk -r -x core show codecs audio Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME DESC 1 (1 0) (0x1) audio g723 (G.723.1) 2 (1 1) (0x2) audio gsm (GSM) 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 16 (1 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8) (0x100) audio g729 (G.729A) 512 (1 9) (0x200) audio speex (SpeeX) 1024 (1 10) (0x400) audio ilbc (iLBC) 2048 (1 11) (0x800) audio g726 (G.726 RFC3551) 4096 (1 12) (0x1000) audio g722 (G722) I will open a bug report. Regards, Chris 2009/5/22 Martin asteriskl...@callthem.info: it should work just fine; do you have the GSM codec compiled/loaded core show modules like codec_gsm ... ? OR that particular version has a BUG... Martin On Thu, May 21, 2009 at 3:56 AM, Chris Maciejewski ch...@wima.co.uk wrote: Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: sip.conf: [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk [1] type=friend secret=test host=dynamic nat=yes -- - extensions.conf: - [common] exten = 501,1,MeetMe(12,MI) exten = 501,n,Hangup() exten = i,1,Hangup() exten = h,1,Hangup() exten = t,1,Hangup() Everything works OK when ALAW is used - see http://pastebin.com/f7222a6d3 but with GSM Asterisk hangs up just after starting MeetMe application - see http://pastebin.com/f78d04c95 line 327. Is there a problem with MeetMe app or I need to adjust my configuration? Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe not working with GSM codec?
Thanks Kinjal! Missing sound files was the problem. There were no .gsm files in my sounds directory. Despite console shows .slin, the actual files required are .gsm. Once I copied .gsm into /var/lib/asterisk/sounds everything works OK. Regards, Chris 2009/5/22 Kinjal Dixit kinjal.di...@gmail.com: On an entirely unrelated note, do you have the gsm asterisk sounds installed? Maybe that vm-*.slin files don’t exist. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Maciejewski Sent: Friday, May 22, 2009 12:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MeetMe not working with GSM codec? Hi Dhaval, The reason confno '12' is not found in meetme.conf is because I am using MySQL as realtime config backend. See few lines below there is: [May 21 09:33:23] DEBUG[6872]: res_config_mysql.c:1478 mysql_reconnect: MySQL RealTime: Connection okay. [May 21 09:33:23] DEBUG[6872]: res_config_mysql.c:365 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM conference WHERE confno = '12' My meetme.conf: [general] audiobuffers=32 logmembercount=yes schedule=no 2009/5/22 DHAVAL INDRODIYA dhaval.it01...@gmail.com: can you look on this from your debug app_meetme.c:3030 find_conf: The requested confno is '12'? == Parsing '/etc/asterisk/meetme.conf': [May 21 09:33:23] DEBUG[6872]: config.c:1306 config_text_file_load: Parsing /etc/asterisk/meetme.conf == Found [May 21 09:33:23] DEBUG[6872]: app_meetme.c:3082 find_conf: 12 isn't a valid conference its on line number 318 it seems that you doesent specify valid conference number can you post meetme.conf regards Dhaval On Thu, May 21, 2009 at 2:26 PM, Chris Maciejewski ch...@wima.co.uk wrote: Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: sip.conf: [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk [1] type=friend secret=test host=dynamic nat=yes -- - extensions.conf: - [common] exten = 501,1,MeetMe(12,MI) exten = 501,n,Hangup() exten = i,1,Hangup() exten = h,1,Hangup() exten = t,1,Hangup() Everything works OK when ALAW is used - see http://pastebin.com/f7222a6d3 but with GSM Asterisk hangs up just after starting MeetMe application - see http://pastebin.com/f78d04c95 line 327. Is there a problem with MeetMe app or I need to adjust my configuration? Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe not working with GSM codec?
On Friday 22 May 2009 02:25:26 Chris Maciejewski wrote: Hi Martin, Yes, I do have GSM compiled for sure. $asterisk -r -x core show codecs audio Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARYHEX TYPE NAME DESC --- - 1 (1 0) (0x1) audio g723 (G.723.1) 2 (1 1) (0x2) audiogsm (GSM) 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 16 (1 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8)(0x100) audio g729 (G.729A) 512 (1 9)(0x200) audio speex (SpeeX) 1024 (1 10)(0x400) audio ilbc (iLBC) 2048 (1 11)(0x800) audio g726 (G.726 RFC3551) 4096 (1 12) (0x1000) audio g722 (G722) I will open a bug report. You, sir, fail on reading comprehension: Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe not working with GSM codec?
it should work just fine; do you have the GSM codec compiled/loaded core show modules like codec_gsm ... ? OR that particular version has a BUG... Martin On Thu, May 21, 2009 at 3:56 AM, Chris Maciejewski ch...@wima.co.uk wrote: Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: sip.conf: [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk [1] type=friend secret=test host=dynamic nat=yes -- - extensions.conf: - [common] exten = 501,1,MeetMe(12,MI) exten = 501,n,Hangup() exten = i,1,Hangup() exten = h,1,Hangup() exten = t,1,Hangup() Everything works OK when ALAW is used - see http://pastebin.com/f7222a6d3 but with GSM Asterisk hangs up just after starting MeetMe application - see http://pastebin.com/f78d04c95 line 327. Is there a problem with MeetMe app or I need to adjust my configuration? Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
