search for: g729b

...on has 20 bytes of payload. We have verified that VAD/DTX is used by the carrier side endpoint by noting that there exist successive RTP packets that differ by 1 in their sequence number but have a timestamp difference > 160 and MARK bits are set in the RTP header. Our understanding is that for G729B, the SID frame that is transmitted before a period of discontinuous transmission has a size of 2 bytes. However we see that ALL RTP packets sent by the carrier side end point has a length of 20 bytes. Has anybody else seen this behavior from a carrier side endpoint ? Is there an RFC or document th...

Hi all, i have purchased the g.729 codec from digium. The registration was successful. (with the "old" binary) But there're a few questions: - should not the codec listed in the codec list when i enter "show codecs" ? - the codec is named with g729b but if i enter show codecs there is a codec g729a listed also the g729b is not installed. what is the difference between g729a built in * and the puchased g729b codec? Thanks for help. Regards, Thomas.

Hello, I'm trying to find out if Asterisk will support plain G729 or G729b. I've read all over that it supports G729, but I can't seem to find any explicit remarks regarding the specific versions of the codec Asterisk will support. I noticed that Digium allows Asterisk users to register and download G729a, but refers to it as G729 on it's pages. I also n...

I am thinking about using the G729 codecs on my endpoint devices and purchasing some G729 licenses for Asterisk but I have several questions: 1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I? 2. If I have G729A on one end and G729B on the other, are they compatible? I have looked all over the place for question 2, but without buying the ITU docs I cannot seem to find this answer. I know that G729A is low complexity which seems to be what Cisco 7960's use but I have some others that support G729B which has comfort nois...

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

...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...

...the first digit which is 1 and play 'Invalid conference number' right a way. I am using chan_oh323, I am close to get this thing to work (having sorted the correct codec), just the dtmf issue now. I am using digium's g729. By the way how many variation of g729 are there. I know g729a, g729b, but there seem to be others. Please help. Thanks Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030806/7d38e04d/attachment.htm

I own a G729b codec from voiceage which I had from Digium a couple of months ago , I friend of mine had the new Digium G729 codec which registers in the asterisk as a Annex A/B codec, the problem that we saw is that the call goes thru find but we cannot here any sound. Asterisk is showing this :...

I have a fresh 1.4.10.1 installation that appears to have a problem with g729 pass-through. I can see the gateway in question sending an INVITE using g729b. However, the Asterisk is only sending g711 in the INVITE to my Polycom phone. [gateway] disallow=all allow=g729 [phone] disallow=all allow=ulaw allow=alaw allow=g729 There's the codec configs for the gateway and the phone in question. I even attempted to setup the phone with only the allow...

Hello! I am wondering how the differences between G729, G729a, and G729b effect call bridging and server interoperability. For example, can one server use the G729 code with another server that uses the G729A codec? Also, which version is Asterisk set up to use? Thanks! Elliot

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 again... Cannot allocate chan...

Hello, I am trying to see what I need to do SIP to H323 using G.729. I have Oh323 and SIP working with G711 fine. If I have a SIP client configured to use G729 and H323 client also to G729, how many license should I need to buy from Digium. Many thanks SW

Hello.... I have the follow situatuion: < ISDN > | | V E100P |----------------| IAX2 / g729A |----------------| T100P | Asterisk1 |- - - - - - - - - - - - - - > | Asterisk2 | - - - - - -> |--------------| | | | | | Zhone | ----------------- ----------------- --------------- Here's the situation: I receive calls from the PSTN

Per my conversation below with digium, are there any legal alternatives to digium's G729? It is out of date, and doesn't support VAD nor silence detection. Digium has stated that they have no plans to update it anytime soon. VAD/Silence is a big deal with major carriers and we are having to fight a battle to get them to make special arrangements to turn off VAD/Silence in their

...1A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G726 - G.726(32k) ; G72616K - G.726(16k) ; G72624K - G.726(24k) ; G72632K - G.726(32k) ; G72640K - G.726(40k) ; G728 - G.728 ; G729 - G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP packet (if not specified) is 1. ; ;codec=G711A ;frames=20 ;codec=G711U ;frames=20 ;codec=GSM0610 ;frames=4 ;codec=G723...

...1A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G726 - G.726(32k) ; G72616K - G.726(16k) ; G72624K - G.726(24k) ; G72632K - G.726(32k) ; G72640K - G.726(40k) ; G728 - G.728 ; G729 - G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP packet (if not specified) is 1. ; codec=G711U frames=20 codec=GSM0610 frames=4 codec=G7231 frames=2 codec=G729 frames=2 c...

...tiation", which states that G.729b is supported on pretty much everything except for AS5350/5400. Also that G.729b is a "high complexity" codec that will require twice the DSP processing. Now given all of that, has anyone interoperated Digium (E1(4)00P, T1(4)00P) -> Asterisk (g729b) -> CISCO termination -> PSTN. This whole setup is for purposes of VoIP. How inter-operatable is VoiceAge's codec with cisco's version of it? Also in the short description of G729 on the Digium's site, they state that "Dual Intel? Xeon 1.8GHz processors allowed 60 concur...

...onfiguring either oh323.conf and h323.conf use alias=192.168.0.1 if your cisco is 192.168.0.1. Cisco's dial-peers can use either "codec" or "voice-class" command for codec specification both worked for me. Also remember that g729 from digium is in fact g729a and cisco uses g729b or g729a depending on what codec complexity has been used. You cannot change on cisco manually "a" or "b". Medium codec complexity means g729a.

...n't think that it had a G.729 license so when the Cisco phone registered to the main Asterisk server answered, the call failed. Here is the gateway log that I found to support this: Apr 25 17:55:33 NOTICE[2883600]: Got event 2 (Ring/Answered)... Apr 25 17:55:34 WARNING[131081]: No available g729b resources for channel 2 Apr 25 17:55:34 WARNING[131081]: Failed to build translator step from 8 to 6 There were no such errors on the main Asterisk server. The problem is that when the call that generated the above logs came in, there was no activity on the gateway or main asterisk server. (i.e...

...G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3 - G.723.1(6.3k) ; G72315K3 - G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G728 - G.728 ; G729 - G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 codec=G729 codec=G7231 codec=G711A codec=G711U