search for: codec_g729b

Hello, I have that problem with codec G729. Please can somebody help me! WARNING[16384]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 == Detected 4 licensed G.729 transcoders WARNING[16384]: File translate.c, Line 219 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR,...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, After upgrading the kernel on an Asterisk box, asterisk segfaults on startup. It seems like it's the g729 codec that causes this: #0 0x4015acad in memset () from /lib/libc.so.6 #1 0x4022686a in load_module () at codec_g729b.c:416 #2 0x08054794 in ast_load_resource (resource_name=0x80d1068 "codec_g729b.so") at loader.c:298 #3 0x08054de3 in load_modules () at loader.c:344 #4 0x0807de5e in main (argc=-1073748912, argv=0x1) at asterisk.c:1398 #5 0x400f4bb4 in __libc_start_main () from /lib/libc.so.6 What d...

I want use iLBC and have following in mind, please help me is it possible ? ISDN <-----(ALAW)-----> * <-----(ALAW)-----> SNOM SIP??<-----(iLBC)----->?*?<-----(ALAW)----->?SNOM 1. ISDN incoming codec is ALAW SNOM codec should be ALAW (can't be iLBC because a lack of codec). 2. SIP incoming codec should be iLBC (snom is ALAW). 3. SIP outgoing codec should be iLBC /snom

...installation. Registration was successful ============== NOW, PLEASE ANSWER THE FOLLOWING QUESTION: Do you accept the terms of this agreement? yes(y) or no(n)y ...Please wait a few seconds Registration successful! ============== But, Now I cant start *, it comes up with the following error; [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) WARNING[1082825920]: File asterisk.c, Line 234 (listener): Select retured error: Interrupted system call WARNING[1082825920]: File asterisk.c, Line 234 (listener): Select retured error: Interrupted system call shmget: Invalid argumen...

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 S...

...nt ok but when I run askterisk via asterisk -vvvgc it gives me this error anyone know? I make sure I entered in the correct reg number. I followed the steps correctly. Too Registration error! Please try again... Cannot allocate channels... Process Stopped! Error -10 WARNING[1074420448]: File codec_g729b.c, Line 511 (load_module): Unable to initialize va stuff: -1 Segmentation fault (core dumped) Here is /var/log/asterisk/messages for this Nov 22 12:25:36 WARNING[1167272000]: File chan_oss.c, Line 238 (sound_thread): Read error on sound device: Resource temporarily unavailable Nov 22 12:25:36...

...et the following warnings and error when I try to place a call via SIP to my X100P. The only way to get out of this is through a restart of *. When the error ocurrs there are no other calls in place. Any ideas? Error Opening channel:2 not available, see va_g729_init_global(..) WARNING[71694]:File codec_g729b.c line 102 (g729lin_new): No available g729b resource for channel 2 WARNING:[71694] File translate.c Line 111 (ast_translator_build_path):Failed to build translator path from 8 to 6 Zap1-1 answered SIP/105-ce3c WARNING[71694]: File chan_zap.c Line 3367 (zt_write):Cannot handle frames in 256 format...

Rumor on the IRC channel has a bug being fixed in the g729.so codec that was causing some hangups under certain circumstances. I see that codec_g729b.so in the ftp://ftp.digium.com/pub/asterisk/g729/ directory has a timestamp of today. Can anyone comment on the advisability of upgrading all of my g729-licensed sites? JT

Hi there, im having some problems with my asterisk box, it seems codec is the principal cause of it. Reading in some forums i found that i can get the new codec_g729 from ftp://ftp.digium.com/pub/telephony/asterisk/g729/new_codec_binary/codec_g729b.so i checked it but the directory new_codec_binary doesnt exist. Anybody knows where can i found it?? Thanks for your help. Carlos Andres Medina --------------------------------- Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! -------------- next part -------------- An H...

What is this? chan_iax2.c line 4695 (build_peer): Unable to support trunking on peer 'lamas-tigris' without zaptel timing codec_g729b.c Line 413 (load_module): Unable to initialize va stuff: -1 This is why I can't start asterisk in the background -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030520/86f0681f/attachment.htm

Hi, I've been trying to get the G.729a beta codec running with my remote Asterisk box that talks IAX2 to my local Asterisk box. Digium fixed the problem I was having in registering the beta codec, so that now works fine. I've removed the old codec_g729b.so from /usr/lib/asterisk/modules and put in place the codec_g729a.so beta from digium FTP. My CVS build of Asterisk is about a day old now. Everytime I try to execute /usr/sbin/safe_astersik with codec_g729a.so in place, it crashes and core dumps, not giving much indication of what's happe...

I'm getting the following message when I start Asterisk: WARNING[1024]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 Did I mess up the registration key or is something else wrong? --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

hi All, We are using Asterisk server with sip phones (SJPhone). On the local LAN, when we use the SJPhone as the SIP client, communication works fine with no disturbances and noices. But when it comes to dialup connection we harldy hear anything except a rough noice. We have included G.729 Codec (Annex B) with the Asterisk server, and we added the G.729 Codec to the SJPhone too. But it seems

Hello, Can somebody tell me what does it means ? I just installed my codec g729 with two channels. [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 2 licensed G.729 transcoders WARNING[16384]: File translate.c, Line 219 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to S...

Hi guys, Just installed G.729 (from digium) codec and after starting asterisk getting the following warning: [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) WARNING[1082809536]: File asterisk.c, Line 234 (listener): Select retured error: Interrupted system call WARNING[1082809536]: File asterisk.c, Line 234 (listener): Select retured error: Interrupted system call == Detected 1 license...

Starting up the asterisk using asterisk -vvvc i get this error is this normal and i purchased license for g729 today? [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) Jan 21 17:31:58 WARNING[1082805040]: asterisk.c:255 listener: Select retured error: Interrupted system call Jan 21 17:31:58 WARNING[1082805040]: asterisk.c:255 listener: Select retured error: Interrupted system call == Detected 4 l...

...;Asterisk -vvvcng" i notice this warning and if i made call the CLI say "No compatible codec!" How can i solve this problem? Thanks in advance Dimitri ------------------------------------------ [app_datetime.so] => (Date and Time) == Registered application 'DateTime' [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 2 licensed G.729 transcoders Apr 21 20:52:15 WARNING[16384]: translate.c:213 calc_cost: Translator 'g729tolinb' does n t produce sample frames. == Registered translator 'g729tolinb' from format G729A...

Hi there, i had installed on all my servers the codec_g729b which is the old voiceage, so a month ago i updated the codec to codec_g729a. After that i started to get this message on my asterisk console: Dec 16 09:19:26 NOTICE[1288699200]: rtp.c:293 process_rfc3389: Don't know how to handle RFC3389 for receive codec 256 Dec 16 09:19:26 NOTICE[128869920...

Trying to configure voicemail with H.323 all I get is the following errors when I call 123, 666, 665, 664 or 031. I'm a newbie at this so, I think it might be a simple fix. [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/oh323.conf': Found 0:00.004 OpenH323 Wrapper OpenH323 Wrapper Version 0.0alpha0 by inAccess Networks

Hello, I've successfully installed Asterisk from last CVS and configured it for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip server. All are work fine at G711 codecs, but then I disable all codecs except g729 some calls failed (Not all calls. Some calls passed at g729 succesfully). All my devices configred to use only g729 and I don't see other codecs at mgcp or sip