search for: mulaw

Hello I have serious problem with zhone z-plex 10 the device sometimes get red alarm on some channel and then the device suddenly shutdown. I try to unplugged the power chord and plug again, but the device still didn't up. I have to wait for long time to get this device up again. How can i fix this problem, so i can get this device immediately? Eris Riswanto

Are there much performance differences when using XEON or not? In my case, I will go with muLaw both in and out of Asterisk. Are there really any processing at all if it's using same codec all the way?

...hat I didn't have to do anything to get slin support. when I do: module show like codec_ I get: Module Description Use codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0 codec_gsm.so GSM Coder/Decoder 0 codec_ulaw.so mu-Law Coder/Decoder 0...

...ed up directly without csu to a TN464 DS1 interface card. With alot of experimentation it connects (With some timing sync issues) with the following settings on the DS1: Bit rate: 1.544 line coding: ami-basic line compensation: 1 framing mode: d4 signalling mode: robbed bit Interface companding: mulaw idle code: 11111111 slip detect: n near-end CSU type: other We then set up a trunk group, but I believe because of the limitations of the card/setup we can only use group type: co which limits us to outbound on the trunk, or inbound to 1 mapped extension, no DID or more advanced switching. The tem...

...va - jcraft, a copy of the JCraft JOgg/Jorbis code - jheora, an implementation of Theora in Java - codecs (currently only containing the Smoke codec, a variant on Jpeg) - JST plugins for: - HTTP source element - Ogg and Multipart demuxers - Theora, JPEG and Smoke video decoders - Vorbis and MuLaw audio decoders - Java 1.1 sun.audio API audio sink - Java 1.4 javax.sound.sampled API audio sink - examples - applets -------------- next part -------------- Release notes for Cortado 0.2.2 "Really Tested Verily Exceptionally" Cortado applet Features of this releas...

I have been having periodic trouble with mpg123. I have tried .59r & .59s and perhaps others a while back and still get the 'broken pipe' and zombie mpg123s (although I think I saw something about a fix in the changelog) once and a while. Is it currently possible to configure moh to run directly on the wav files? Now-a-days hard drives are so big, why use compression at all (at least

...fine, but the Grandstream seems to suffer from one-way voice (remote end can't hear me). The only workaround I found so far (have not spoken with VOIPJet supoprt about this), was to configure the GS entry in sip.conf to disallow=all and allow=ilbc, while iax.conf has disallow=all, allow=ulaw mulaw and ilbc (seems the call is established in this case, with the GS leg in ILBC and the voipjet in ulaw). Should I assume this would force the Asterisk to do the transcoding to any other end-point that doesn't handle/prefer ilbc ? What about DTMF (I've changed to 'SIP INFO' in sip....

...ator 'lintoulaw' from format SLINR to ULAW, cost 1 [codec_alaw.so] => (A-law Coder/Decoder) == Registered translator 'alawtolin' from format ALAW to SLINR, cost 1 == Registered translator 'lintoalaw' from format SLINR to ALAW, cost 1 [codec_a_mu.so] => (A-law and Mulaw direct Coder/Decoder) == Registered translator 'alawtoulaw' from format ALAW to ULAW, cost 1 == Registered translator 'ulawtoalaw' from format ULAW to ALAW, cost 1 [format_g723.so] => (G.723.1 Simple Timestamp File Format) == Registered file format g723sf, extension(s) g72...

...==================== [devices] wanpipe1 = WAN_AFT_ANALOG, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 6 PCIBUS = 1 FE_MEDIA = FXO/FXS TDMV_LAW = MULAW TDMV_OPERMODE = FCC MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = NO

...ator 'lintoulaw' from format SLINR to ULAW, cost 1 [codec_alaw.so] => (A-law Coder/Decoder) == Registered translator 'alawtolin' from format ALAW to SLINR, cost 1 == Registered translator 'lintoalaw' from format SLINR to ALAW, cost 1 [codec_a_mu.so] => (A-law and Mulaw direct Coder/Decoder) == Registered translator 'alawtoulaw' from format ALAW to ULAW, cost 1 == Registered translator 'ulawtoalaw' from format ULAW to ALAW, cost 1 [format_g723.so] => (G.723.1 Simple Timestamp File Format) == Registered file format g723sf, extension(s) g72...

...0 WARNING[17550] chan_zap.c: Unable to set audio mode on channel 1 to 1 Feb 25 11:28:10 WARNING[17550] chan_zap.c: Unable to set law on channel 1 Feb 25 11:28:10 WARNING[17550] chan_zap.c: Unable to set gains on channel 1 Feb 25 11:28:10 WARNING[17550] chan_zap.c: Unable to get parameters, assuming MULAW Feb 25 11:28:10 WARNING[17550] chan_zap.c: Unable to set audio mode on '1' Feb 25 11:28:10 WARNING[17550] chan_zap.c: zt confmute(0) failed on channel 1: Inappropriate ioctl for device Feb 25 11:28:10 WARNING[17550] chan_zap.c: Unable to enable echo cancellation on channel 1 Feb 25 11:28:10...

...directly without csu to a TN464 DS1 interface card. With alot of experimentation it connects (With some timing sync issues) with the following settings on the DS1: Bit rate: 1.544 line coding: ami-basic line compensation: 1 framing mode: d4 signalling mode: robbed bit Interface companding: mulaw idle code: 11111111 slip detect: n near-end CSU type: other We then set up a trunk group, but I believe because of the limitations of the card/setup we can only use group type: co which limits us to outbound on the trunk, or inbound to 1 mapped extension, no DID or more advanced switching. The...

...nslator 'lintoulaw' from format slin to ulaw, cost 1 [codec_alaw.so] => (A-law Coder/Decoder) == Registered translator 'alawtolin' from format alaw to slin, cost 1 == Registered translator 'lintoalaw' from format slin to alaw, cost 1 [codec_a_mu.so] => (A-law and Mulaw direct Coder/Decoder) == Registered translator 'alawtoulaw' from format alaw to ulaw, cost 1 == Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1 [codec_g726.so] => (ITU G.726-32kbps G726 Transcoder) == Registered translator 'g726tolin' from forma...

...so Raw G729 data 0 app_userevent.so Custom User Event Application 0 codec_g729a.so Annex A/B (floating point) G.729/PCM16 C 0 codec_gsm.so GSM/PCM16 (signed linear) Codec Translat 0 codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0 app_system.so Generic System() application 0 app_record.so Trivial Record Application 0 app_sayunixtime.so Say time 0 pbx_spool.so Outgoing Spool Support...

...) Creating User (Station/Channel) Extensions 6) Creating Application Extensions f. Sample Configurations [LOTS OF SAMPLES HERE] 4. Common Issues a. Music On Hold/MP3 Playback 1) Proper Version of MPG123 2) Zaptel Timer or Ztdummy b. DTMF Over SIP 1) Inband Only Works On G.711 MuLaw/ALaw 2) SIP-INFO 3) RFC___? 4) The "Flash" c. Internationalization of Asterisk 1) Tones and Ringback 2) Call Supervision d. SIP and NAT e. Optional/Added Codecs 1) G.729 2) G.723 f. Message Waiting Indication g. SIP Service Provider Issues 1) Free World Dial...

Flash player encodes speex at 16 kHz, mono, 16 bit. Fields in the format byte should be ignored if the format is speex. You can set the quality by Microphone.encodeQuality (default 6). You can also set the number of speex frames per tc message using Microphone.framesPerPacket. Flash player can only decode speex at 16 kHz, so make sure you have the proper sample rate. Jozsef > > Message:

...7x ca0106 cmipci emu10k1 emu10k1x ens1370 ens1371 es1938 es1968 fm801 hda-intel intel8x0 intel8x0m maestro3 trident usb-audio via82xx via82xx-modem ymfpci" ALSA_PCM_PLUGINS="adpcm alaw asym copy dmix dshare dsnoop empty extplug file hooks iec958 ioplug ladspa lfloat linear meter mmap_emul mulaw multi null plug rate route share shm softvol" APACHE2_MODULES="actions alias auth_basic authn_alias authn_anon authn_dbm authn_default authn_file authz_dbm authz_default authz_groupfile authz_host authz_owner authz_user autoindex cache dav dav_fs dav_lock deflate dir disk_cache env expire...

Yes, I can make a call on that extension from other definity phone, if you mean it. -----Original Message----- From: Ken Godee [mailto:ken@perfect-image.com] Sent: 30 ÉÀÌÑ 2004 Ç. 19:14 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1. Incoming work, but not outgoing Roman Bessyadovskii wrote: > Hi All. > > I connect

...Dialogic VOX (ADPCM) File Format 0 format_wav_gsm.so Microsoft WAV format (Proprietary GSM) 0 format_wav.so Microsoft WAV format (8000hz Signed Line 0 format_gsm.so Raw GSM data 0 codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0 codec_alaw.so A-law Coder/Decoder 0 codec_ulaw.so Mu-law Coder/Decoder 0 codec_adpcm.so Adaptive Differential PCM Coder/Decoder 0 codec_lpc10.so LPC10 2.4kbps (signed linear) Voice...

...wmp wnck wv wxwindows x264 x86 xattr xcb xinerama xinetd xml xml2 xorg xosd xprint xulrunner xv xvid zeroconf zlib" ALSA_CARDS="emu10k1 intel8x0 usb-audio" ALSA_PCM_PLUGINS="adpcm alaw asym copy dmix dshare dsnoop empty extplug file hooks iec958 ioplug ladspa lfloat linear meter mulaw multi null plug rate route share shm softvol" APACHE2_MODULES="actions alias auth_basic authn_alias authn_anon authn_dbm authn_default authn_file authz_dbm authz_default authz_groupfile authz_host authz_owner authz_user autoindex cache dav dav_fs dav_lock deflate dir disk_cache env expire...