similar to: G729, voicemail, no codec_g729

Hello everyone, One more little problem with a %100 g729 setup. Native moh: musiconhold.conf: [default] mode=files directory=/mnt/kd/moh/default random=yes ; Play the files in a random order ls /mnt/kd/moh/default fpm-calm-river.g729 fpm-calm-river.ulaw fpm-sunshine.g729 fpm-sunshine.ulaw fpm-world-mix.g729 fpm-world-mix.ulaw Place a call on hold: Jun 1 14:55:30

I am trying to set up a simple Asterisk server. All it's going to do for now is to act as my voicemail box. I've got a DID from Voicepulse, and am using IAX (I'll get to SIP someday when I want to circumvent the phone company for long-distance, but for now I'd be happy to get a trial version of Asterisk running). So far, I've managed to set up voicemail.conf, extensions.conf

Greetings, The software G.729 codec module from Digium has been updated for all platforms. There are x86_32 and x86_64 versions optimized for specific processors available for both Asterisk 1.6 and 1.4 for the following platforms. * Linux * Solaris 10 * FreeBSD 7.0 * FreeBSD 6.1 Changes: * For Asterisk trunk / 1.6, builds have been updated for CLI API changes. * All non-Linux

Greetings, The software G.729 codec module from Digium has been updated for all platforms. There are x86_32 and x86_64 versions optimized for specific processors available for both Asterisk 1.6 and 1.4 for the following platforms. * Linux * Solaris 10 * FreeBSD 7.0 * FreeBSD 6.1 Changes: * For Asterisk trunk / 1.6, builds have been updated for CLI API changes. * All non-Linux

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

Hello everyone can anyonone of you send me the file codec_g729.so this file has to be inserted in /urs/lib/asterisk/modules Thank You

pbx18*CLI> module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]:

Hi All, I have been trying to get the Solaris version of the G729 codec to work with asterisk 1.2.17 and 1.2.22. However, I come up against the very same error every time I try to install it. Has anyone out there seen this error, taken from the asterisk console straight from startup: [codec_g729a.so] => (Annex A/B (floating point) G.729 Codec (optimized for i386)) Jul 19 14:11:23

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

I have noticed that when I try to connect multiple G729 VoIP devices into a MeetMe conference that I can only add up to the number of G729 licenses I have. Now I would think that because all the devices are G729, this wouldn't be the case and the only license that would ever be used would be if a non G729 device or Zap channel was a part of the Meetme conference. This is apparently note the

Humm... should the list would be magic again? I have just intsalled, using the register, benchmark and downloared the correct codec to my asterisk installation, but I don't have the g729 command at my CLI... Any advice... Do I reboot? ;D

List, Anybody have success with Digium's G729 codec and asterisk 1.6.0? Reading http://www.russellbryant.net/blog/index.php/2008/03/05/codec_g729-v34-builds-now-available/ is seems they are build for 1.6 and trunk. But all I could find / use is 1.4 builds from http://downloads.digium.com/pub/telephony/codec_g729/ Thoughts? PB

Hi All, I have this codec problem, I use "gsm" in my iax.conf file and in teliax settings also, but the error is still appearing as below. please help me. Kumara Starting simple switch on 'Zap/1-1' -- Executing Dial("Zap/1-1","IAX2/kumara@teliax/01194777070239|30|tr") in new stack -- Called kumara@teliax/01194777070239 -- Call accepted by

Hello everybody, I was trying to install a web interface to my Voice Mail, Vmail.cgi I can log on it, list messages, but no play with the following error msg; "Hrm, can't seem to open /var/spool/asterisk/voicemail/default/234/INBOX/msg0001.WAV" Remark: playing the message msg0001.WAV directly OK Any smart guy up there could help ? Thanks, --------------------------------- Do

I have a user, running CVS a/o 11/23/03, who has complained about "phantom" messages showing up days or even weeks after she has deleted them. So I asked her to let me know when it happened again, and she called a few minutes ago. The directory listing below shows a listing of the /var/spool/asterisk/voicemail/default/XXXX/Old directory, and to my surprise the messages are indeed

This question is about 1.2.x asterisk. Please no flames, or "you should upgrade to 1.4". Does anyone know what might be the cause for 'stuck voicemail's in 1.2.6asterisk? By stuck, I mean the phones show a voicemail, and if you log in you get "you have 1 new voicemail", and if you delete it it says 'deleted', however it remains. Going into the mail

Is there a way to make the asterisk voicemail app play back messages in NEWEST FIRST order, instead of OLDEST FIRST? I see the situation repeatedly where someone needs to dip into their voicemail archive to get something from a recently saved voicemail message, and they have to slog through lots of irrelevant stuff to get there. I have seen this question come up previously on this list without

Greetings! to all Now, with some hard time and help from many genurous people's in the list, I have come to this point with my TDM20B card & my teliax's IAX2 account. I hope someone may help me with this issue mentioned below. I have already selected my codec as gms in my iax.conf as well as in teliax's "my account page" but still i have the same error when I attempt

Hi all, I am currently running asterisk (CVS HEAD) on a p4 machine with rh7.3 and using nufone's h323 channel driver. I was using the old voiceage g729 and have replaced it with the digiums g729. I used to have the message "Measured length exceeds frame length" but since I changed the g729 codec I have not encountered this problem. However, another peculiar problem has cropped

hi, all i want to use g729 codec for set up a call. so i donwloaded the so file from web site: http://asterisk.hosting.lv/#bin and install it properly. *CLI> *CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin