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 >>> >>> [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