similar to: Problems with g729

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

-----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

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

Hi, I just applied four new g729 license to my * 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)

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

Any ideas on this warning? If I call this number, sometimes I get this error and sometimes the call goes thru fine. Why would it work sometimes? -- Executing Goto("SIP/3044-8d49", "cytel-outgoing|915124512424|1") in new stack -- Goto (cytel-outgoing,915124512424,1) -- Executing SetCIDNum("SIP/3044-8d49", "2814494000") in new stack --

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

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

Hey all, Does anyone know what this means? I was running asterisk fine. Installed it on a new pc and I am using the g729b. codec that is optional. I ran the install for the codec it went 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

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

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:

Help-Rs   As someone who is newer to R and trying to make the transition from Access into R, there is a frequetnly used function that I'd like to try and duplicate in the R world.  It involved creating an aggregate table of the top (n)  orders for an item by sum of cost over a select period of time.   So, take the following example :   group <- c(rep(1,10), rep(2,10), rep(3,10)) product

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

Hi, I'm running the latest Asterisk (built last Saturday) and can't get mp3's to playback on my handsets (this includes music on hold). I setup a couple of extensions, 901 and 902 to playback an mp3 I loaded on, and the sample moh that is included with Asterisk. When I attempt to call either extension I don't hear any sound, and the following displays on the console:

Hi, I am compiling a tiny linux kernel 2.6.10 to be used together with linuxbios. The boot phase will go as follows: linuxbios->tiny linux kernel->initramfs The initramfs shall use dhcp to get it's address and then tftp a full featured kernel. This kernel must be started through kexec. I use a tweaked networkdriver in the kernel and cannot use a normal PXE boot. Which files should I

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 I seem to have installed the G729 codec properly but can't seem to get it to work .. in the Asterisk startup I see .. [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 1 licensed G.729 transcoders WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator

Hi i am trying to do the same thing: receive a call from a cisco callmanager and forward it to a SIP user. Asterisk is compiled with h323 support, and is configured as a gateway in the cisco callmanager. h323.conf: [general] port = 1720 bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP address for this machine allow=all extension.conf: exten = 3298,1,Answer exten =

== Parsing '/etc/asterisk/adsi.conf': Found -- Accepting call from '890003' to '185' on channel 27, span 1 -- Executing Answer("Zap/27-1", "") in new stack -- Executing Record("Zap/27-1", "soundexampless:mp3") in new stack -- Playing 'beep' WARNING[360468]: File translate.c, Line 128

Hi users, I have Asterisk 1.6.2.20 in Ubuntu 10.04. I am trying to convert a gsm file to G729 using file convert, but I am facing error as follows, file convert /tmp/welcome.gsm /tmp/welcome.g729 Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729! Command 'file convert /tmp/welcome.gsm /tmp/welcome.g729' failed. [Dec 20 17:24:18] WARNING[2221]: translate.c:256