similar to: G729 Codec Error

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

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

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)

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

Dear i have buy two license of G729 codec and i have install/registered as documented but after i start "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

I left these information out sorry = Detected 1 licensed G.729 transcoders WARNING[1024]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format 8 to 6, cost 99999 == Registered translator 'lintog729b' from format 6 to 8, cost 18 -------------- next part -------------- An HTML

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

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

hey, I bought a license for 729 but I can't use it this is the message. == Registered translator 'g729tolinb' from format 8 to 6, cost 99999 == Registered translator 'lintog729b' from format 6 to 8, cost 18 == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI> WARNING[5126]: File chan_sip.c, Line 1601 (process_sdp): No compatible codecs!

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

Hi, I have licensed the digium G.729A codec. But for some reason incoming and outgoing calls will ALWAYS use G.711a. When I force my phone to only accept 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

Hi, I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card. Incoming calls and outgoing calls between my cisco and my SIP phone works fine on G.729. Recording messages in the asterisk voice-mailbox also works fine from both my SIP phone as well as PSTN -> Cisco -> Asterisk. I have licensed the digium G.729A codec. When I connect my ISDN PRI to my Zap card and I call

-----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

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

Hi folks, In an effort to save bandwidth (our 7905s run over a WAN) we've switched from ulaw to g729a. We purchased 4 licenses from Digium (4 SIP clients, low call volume), and they seem to have been accepted: [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 07:53:aa:d3:e2:f2:bd:cc:27:60:9d:5f:da:eb:5d:e9:6e:09:a1:4e == Found license

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was

hi, I'm currently trying to use Speex with Asterisk from my OpenH.323 client. It seems to mismatch the codecs, below is my log from Asterisk. My Openh323 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

Hi, I have a new phone in our IP phone network: Planet VIP-101T. When calling from that Planet phone to anybody, everthing is fine. But when calling from anybody to that Planet phone, I get a mashine gun noise and the following msg in asterisk log: NOTICE[262161]: File channel.c, Line 1091 (ast_read): Dropping incompatible voice frame on H323:0 of format SLINR since our native format has

Where do these strange noises come from? <http://www.tobiasjonsson.se/asterisk/recorded-sound.wav> First sound in the recording above is from a ISDN (EuroISDN) connection thru chan_modem in Asterisk. Second sound is recorded from a SIP soft phone to the same RxFAX(), which now sounds all right. I have talked to Steve Underwood who says I am the first to report this problem and he thinks the