search for: codec_mp3_d

...gned linear) Voice Coder) == Registered translator 'lpc10tolin' from format 7 to 6, cost 50 == Registered translator 'lintolpc10' from format 6 to 7, cost 10 [app_setcidname.so] => (Set CallerID Name) == Registered application 'SetCIDName' [skipping pbx_gtkconsole.so] [codec_mp3_d.so] => (MP3/PCM16 (signed linear) Translator (Decoder only)) == Registered translator 'mp3tolin' from format 4 to 6, cost 8 [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/oh323.conf': Found ERROR[1024]: File chan_oh323.c, Line 1304 (load_module): CAP_...

...ot;-march=*" with "-mv8" 2. Comment out lines containing "k6opt" from asterisk/codecs/gsm/Makefile. 3. Copy libgsm.a to /lib 4. ln -s $ASTERISK_HOME_DIR/include/asterisk /usr/include/asterisk Not everything works on linux on Sparc ... You will have to specify "noload codec_mp3_d.so" in modules.conf chan_h323.so will not load too. It gives undefined symbol error. However chan_oh323.so will work fine. Have not tested everything but basic MGCP and SIP calls work fine ... Voicemail works too. Still struggling to get H323 to work ... Had a codec issue to begin with ......

hi! i did install asterisk on a freebsd box. after removing the codec_mp3_d.so module it starts without SIGSEGV. -> i didn't care about this issue. however while running asterisk without any clients online my system-load is about 1 -> and asterisks seems to run all the time. is this the default behavior? is there any interface or command line command to find...

I have had * installed for a couple of weeks now and am very impressed. I have got Zap, SIP and MGCP working and can call freely between them with just things like transfer still to sort out etc. I then though I would add H.323 support to my working system, having read the previous threads on the subject before I installed I installed the pre-reqs pwlib openh323 gnugk for h.323 gatekeeper

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in