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 <[email protected]>: > On an entirely unrelated note, do you have the gsm asterisk sounds > installed? Maybe that vm-*.slin files don’t exist. > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] 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 <[email protected]>: >> 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 <[email protected]> > 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 >>> >>> [10000] >>> 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
