search for: g279

Hello, Here (http://downloads.digium.com/pub/telephony/codec_g729/README) are instructions to install G729 software. (I think I followed instructions step by step but g729 license doesn't seem to show up). My question is : Is the command bellow still up to date ? >g729 show Regards PS: Here are latest steps: # ./benchg729-1.0.6 Recommended flavor for this system is

2009/5/8 Olivier <oza-4h07 at myamail.com> > Hello, > > Here (http://downloads.digium.com/pub/telephony/codec_g729/README) are > instructions to install G729 software. > (I think I followed instructions step by step but g729 license doesn't seem > to show up). > > My question is : > Is the command bellow still up to date ? > > >g729 show I suddenly

Google has given me too many responses, so I'll ask the list: Where in the Asterisk rtp source code can I find the default millisecond frame size? I've looked around for obvious pointers, but it's not clear. I'd like to "force" my Asterisk server to use a certain frame size all the time. (Of course, ideally I'd like to prefer or even force that frame size in a

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

HI ALL; I found in the following page: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing 1-If I could record all IVR promts in G729 format 2-If I could record voicemail in g279 format with """format_g729.c""""" then I donot need any g729 license (I suppose all my clients have g729 ip phones) My question is, how can I record voicemail in g729 with "''"format_g729.c"""??????????//// warmest rega...

...rces that I downloaded from Digium, my Polycom 300 works just fine. If I build with either various CVS builds, or the 1.0.6 sources from Digium, I get "No compatible codecs!". WTF? I'm using -the exact same- config files for both; I've tried enabling/disabling ULAW, ALAW and G279 on the Soundpoint, to no avail. Plug 1.0.0 back in -- works like a champ. To say that I'm confused would be understating things rather severely. Thanks much, Ken D'Ambrosio

...r: Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 10.9.9.9 202 11a6323d2ff 00102/00000 0x100 (g729) No Tx: ACK 10.9.9.10 201 a8749-c0a80 00101/00001 0x100 (g729) No Rx: ACK Installing the open source binary of g279 (codec_g729.so in modules dir) and restarting asterisk, the same call require transcoding: Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 10.9.9.9 202 1992c2d149e 00102/00000 0x8 (alaw) No Tx: ACK 10.9.9.10 201 140a6d-c0...

Depends what you want to do! Do you want to do VoIP over that T1 to a provider or IP telephones? Do you want to hook up to the PSTN through that T1 as 24 voice channels, through a T1 card on your asterisk? If you want to use the T1 as 24 voice channels, the Telco is going to have to re-provision the T1 as a voice T1, because currently, presumably it is one big channel of data. You could have

...ed with debug info, the best stack trace I can get is: #0 0x4d5544a0 in pipe () from /lib/libc.so.6 #1 0xb69384ce in __cxa_finalize () from /usr/lib/asterisk/modules/codec_g729a.so #2 0xae7fdae4 in ?? () #3 0xae7fcae4 in ?? () #4 0x00001000 in ?? () #5 0x00000000 in ?? () The version of the g279 module is: Digium G.729A Module Version 1.6.2.0_3.1.4 (optimized for generic_32) Just now, on a very low-volume asterisk server I am monitoring, two calls just got processed. The g729a codec did a pair of pipe() calls, and voila! I have one more open file descriptor as reported by lsof. Some of...

I have a whole bunch of remote devices connected to my Asterisk box, including SPA-3000s, PAP2-NAs and Cisco 7960s. The PAP2-NAs I have only rolled out recently and I am having a problem that is intermittent and inconsistent. It happens to some users but not other users on the same ISP. It happens to users in 2 different countries where the Internet setup (NAT issues) are completely different. It

...able to everything and all, if I'm on a 33.6 modem link and you attempt to send me 24.6 kbps with a ptime of 20 ms, it won't work, no matter what and the client might as well try something else (even if that something else is LPC10!). > The other way would be to make it transparent like g279. Not sure what kind of transparence you mean? The Speex decoder (unless you remove some tables) is able to decode anything without even knowing how it was encoded. >> I'm just trying to allow that while still taking into account the fact >> that some clients just don't have en...

...from what we have now. I would still keep the thing that says 8 kbps SHOULD be supported. Or maybe we can say "all modes SHOULD be supported and if it's not possible, then at least 8 kbps (mode 3) SHOULD be supported". >>> The other way would be to make it transparent like g279. >> >> Not sure what kind of transparence you mean? The Speex decoder (unless >> you remove some tables) is able to decode anything without even knowing >> how it was encoded. > > Except for limited device which are not capable of decoding the "mode" > th...

Hi, What do I need to do to get Asterisk to allow me to use codec G-726? I've already tried allow=all in my sip.conf config.. didn't work...

...se g729 pass-thru when I dial out to a sip provider from my IP phone but because I have no license for g729 I'd like to use g711 ulaw for asterisk voicemail, conference bridge and other services. When I set in [general] section of sip.conf the following: disalow=all allow=g729 allow=ulaw the g279 pass-thru works fine with my SIP provider but when I call the conference extention the call gets dropped because it wants to use g729 as well but there is no license. Would it be possible to use g711 ulaw in this case somehow? Any ideas? Thanks. Regards, Gyorgy

...as possible. I understand transcoding is a big part of CPU usage on Asterisk, and I have the following situation: - A SIP provider that offers me G729 and ULAW (my choice, both as allowed) - Some of my calls are from G729 enabled phones to outside lines, and I`d like to send those calls using G279 to my provider - Some of my calls are from PSTN (ulaw) and send out to PSTN (to a cell phone let's say, in a find-me-follow-me fashion). I`d like those calls to be sent out in ulaw. Can this be done, or do I have to choose one or the other for all calls? I know I can do that using sip.con...

I have question concerns asterisk 1-What is difference between G.729 and G.729A? 2-How can I know the requirement hardware for 150 extension on asterisk 1.4.4 making 50 simultaneous call? 3-Do asterisk have a codec conversion? Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with

...om Ltd. P.S. Our company have ordered your hardware cards. -------------- next part -------------- [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) tos=0x14 delayreject=yes disallow=all allow=g273.1 allow=g279 allow=ilbc allow=ulaw allow=alaw allow=gsm allow=g726 jitterbuffer=no mailboxdetail=yes notransfer=no ;register => gamcom:2046590@iaxtel.com ;register => 478933:2046590@iax2.fwdnet.net ;[fwd-gw] ;type=peer ;auth=md5 ;secret=2046590 ;username=478933 ;qualify=yes ;host=iax2.fwdnet.net ;disall...

Hi to all in a scenario where: - the bandwith is shared with other traffic (HTTP,VPN,ecc) - the PBX is on a remote VPN peer - due to many reasons Qos is not usable There is a IAX trunk between 2 Asterisk 1.4 i've tried different codecs (ulaw,alaw,gsm) but the main problem still remain the same: too many voice packet get lost. The main problem is surely on the network, but the strange thing

...s 8 kbps SHOULD > be supported. Or maybe we can say "all modes SHOULD be supported and if > it's not possible, then at least 8 kbps (mode 3) SHOULD be supported". I think this is close to be equivalent anyway. >>>> The other way would be to make it transparent like g279. >>> >>> Not sure what kind of transparence you mean? The Speex decoder (unless >>> you remove some tables) is able to decode anything without even knowing >>> how it was encoded. >> >> Except for limited device which are not capable of decoding the &...

...LPC10!). This is one reason why ptime should be round up! Anyway, if a mode is not supported for network reason, the application that supports any more should propose a fmtp line: (acting like a limited device) a=fmtp:97 mode=1;mode=2 >> The other way would be to make it transparent like g279. > > Not sure what kind of transparence you mean? The Speex decoder (unless > you remove some tables) is able to decode anything without even knowing > how it was encoded. Except for limited device which are not capable of decoding the "mode" that were removed from code ;)...