search for: pcm16

Hi, I'm running asterisk-1.0.2-2mdk. When I tried to start it with /usr/sbin/asterisk -vvvvvvvvvvvvvvvvvvvvgc, I get [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) Ouch ... error while writing audio data: : Broken pipe # ps aux | grep mpg123 root 5237 0.1 0.4 5816 4444 pts/0 S 18:45 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 Could someone please g...

...== 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_NSUP_ER. WARNING[1024...

...d application 'Playback' [app_dumpchan.so] => (Dump Info About The Calling Channel) == Registered application 'DumpChan' [app_zapateller.so] => (Block Telemarketers with Special Information Tone) == Registered application 'Zapateller' [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) == Registered translator 'ilbctolin' from format ilbc to slin, cost 7 == Registered translator 'lintoilbc' from format slin to ilbc, cost 245 [codec_g729a.so]Apr 29 22:25:25 WARNING[16253]: loader.c:325 __load_resource: /usr/lib/asterisk/module...

...the requested channel 0 app_setcidname.so Set CallerID Name 0 format_g729.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...

Hi. I trying to encode PCM16 8000Hz stereo data to speex and put it into the .flv file format. But at the output I can hear only noise. What I doing wrong? Here is the code: void main() { SpeexBits bits; void *enc_state; int frame_size; int quality = 10; char cbits[MAX_FRAME_BYTES]; FILE *fin, *speex;...

...ian Mail (Voicemail System)) == Parsing '/etc/asterisk/voicemail.conf': Found == Registered application 'VoiceMail2' == Registered application 'VoiceMailMain2' [app_transfer.so] => (Transfer) == Registered application 'Transfer' [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) Illegal instruction thanks hallian _________________________________________________________________ The new MSN 8: advanced junk mail protection and 2 months FREE* http://join.msn.com/?page=features/junkmail

...client crashes in responding to a Speex request for bits per frame. I'm guessing it either isn't running the codec correctly or doesn't support the same subset of speex codecs as openh323. (I'm using speex-1.0.1 with asterisk) thanks, Adam Hart [codec_speex.so] => (Speex/PCM16 (signed linear) Codec Translator) == Registered translator 'speextolin' from format SPEEX to SLINR, cost 2 == Registered translator 'lintospeex' from format SLINR to SPEEX, cost 47 == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready.

I have Now a G729 codec license and when i start it comes: [format_g729.so] => (Raw G729 data) == Registered file format g729, extension(s) g729 [app_datetime.so] => (Date and Time) == Registered application 'DateTime' [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) sh: line 1: tmp: Is a directory rm: cannot remove `tmp': Is a directory Cannot allocate channels... Process Stopped! Error -11 May 12 18:40:08 WARNING[16384]: codec_g729b.c:511 load_module: Unable to initialize va stuff: -1 Segmentation fault alberspilnx8:~ # Ouch ... erro...

...I do not listen the dial tone or the busy tone but the call it is completed normally. sip-phone-g729-------------asterisk--------h323-g729--------------nortel-pbx thi is may configuration: RedHat 8 2.4.18-14 Asterisk 1.0.7 The NuFone Network's Open H.323 Channel Driver G.729/PCM16 Codec Translator Raw G729 data It is a problem of codecs compatiblility or wath? Thanks to all.

...cept G.729 then an incoming call from ZAP goes straight to my voicemailbox as the phone doesn't accept the codec Asterisk wants, even if I force it in sip.conf. Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ? The translator is loaded... [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator) == Registered translator 'gsmtolin' from format GSM to SLINR, cost 1 == Registered translator 'lintogsm' from format SLINR to GSM, cost 4 [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 1b:a1...

Hi Jean-Marc, Regarding you points: 1) Is it ok if the audio is encoded (using Nelly Moser ASAO) and sent to the client and decoded when it is recevied so the AEC is always performed on raw PCM16 8KHZ ? 2) The audio is moved in 32ms (512 byte) chunks and the reading and writing to the AEC code will be done by separate threads at regular 32 ms intervals. 3) Occasionaly audio is dropped if it has become delayed but a jitter buffer of 120ms is in use. People at different distances...

...phone, claiming the phone does not support the codec asterisk wants, as I forced it to G.729. For some reason incoming and outgoing calls will ALWAYS use G.711a. Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ? The translator is loaded... [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator) == Registered translator 'gsmtolin' from format GSM to SLINR, cost 1 == Registered translator 'lintogsm' from format SLINR to GSM, cost 4 [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 1b:a1...

...:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with our capability 0xfe02. I do not understand why because my Asterisk box load these codecs properly! Does somebody know why? [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) == Registered translator 'ilbctolin' from format ilbc to slin, cost 16 == Registered translator 'lintoilbc' from format slin to ilbc, cost 90 [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator) == Registered translator 'gsmt...

Dear Speex developers, I am currently experimenting with Speex on Symbian smartphones. I have compiled the Speex library, and I am now using it in the following way: 1. Record 320-byte buffers of data in PCM16 format, 8000 Hz sampling rate. 2. Feed the resulting buffer to an instance of a narrowband Speex encoder. 3. Send the encoded data over RTP. 4. Upon receiving on the other side, feed the payload to a narrowband Speex decoderd. 5. Play the resulting PCM to the reproductor. However, currently I am s...

.../sbin/asterisk -cvvvvvvg to get as much verboseness as possible, and have cut the last few lines (host ID and license intentionally blanked out):- [format_g729.so] => (Raw G729 data) == Registered file format g729, extension(s) g729 [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: **masked** == Found license '**masked**' providing 2 channels == Found total of 2 G.729 licenses Illegal instruction (core dumped) The machine this is running on is rather old - it's a Pentium MMX (166Mhz according to Linux, I thought it wa...

..., CANCEL, OPTIONS, BYE, REFER Contact: <sip:9999@192.168.11.17> Content-Length: 0 in sip.conf: [router] type=friend context=default host=192.168.77.254 dtmfmode=info disallow=all allow=g729 nat=no canreinvite=yes qualify=yes in debug: [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx == Found license 'G729-XXXXXXXX' providing 2 channels == Found total of 2 G.729 licenses == Registered translator 'g729tolin' from format g729 to slin, cost 2 == Registered tran...

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

...conference has mic to close to speakers. Any ideas? Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > 1) Is it ok if the audio is encoded (using Nelly Moser ASAO) and sent > to the client and decoded when it is recevied so the AEC is always > performed on raw PCM16 8KHZ ? No. The entire path from AEC to loudspeaker and from mic back to AEC must be free of any non-linearity, codec, drift, ... > 2) The audio is moved in 32ms (512 byte) chunks and the reading and > writing to the AEC code will be done by separate threads at regular > 32 ms intervals....

...with No compatible codecs! I have bought a 5 user license just to try and fix this, apparently it doesn't work. I want to protect the Cisco gateway from unauthorized use, but still using a cost-effective codec such as g.723 or g.729 ? [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 5f:a1:18:82:47:6f:a8:f7:33:4e:7d:77:e8:1d:60:15:53:ec:49:aa == Found license 'G729-700241AB' providing 5 channels == Found total of 5 G.729 licenses == Registered translator 'g729tolin' from format G729A to SLINR, cost 2 == Register...

I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please shoot me an email. Thanks Cory Andrews ++++++++++++++++++ VoIPSupply.com PBXSelect.com ++++++++++++++++++ 454 Sonwil Drive Buffalo, NY 14225 voice direct - 716.250.3402 fax -