similar to: G729A

I seem to be having a problem with transcoding and/or agreeing on a valid codec. I am running a new image pulled from CVS at 1:30 PM CST. The issue occurs when I try to make a call to a toll-free number over sipphone.com. Here's what I see in the console: NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1259545280]: File

Hello, When I'm trying to play standard sound files from Asterisk using G729A codec with OH323 channel I get this message: channel.c:1650 ast_set_write_format: Unable to find a path from GSM to G729A It seems that this files must be in G729 format? How can I convert this files to G729? ... or am I wrong? -- wbr, Oleg

Hello! I have a sip device that is sending in the SDP: rtpmap:98 g729a It does not seem like Asterisk is negotiating the codec properly, because while the call rings, the rtp lines fail. However, on other sip devices that have "rtpmap:18 g729" in their SDP, things work fine with Digium's commercial g729 license. How do I get "98 g729a" recognized by Asterisk? Thanks,

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

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the

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

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

Hi. There is not much info on the FWD site about this. What codecs do they use? When I try to connect with X-Lite, it works with GSM. When I try to call out with *, it wants G729. I have disallow=all and allow=gsm in the sip.conf. I end up getting errors: Unable to find a path from G729A to GSM Unable to find a path from GSM to G729A What's up with that? I was able to make a call once

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

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

We've bought the G729 codec for lowering SIP bandwidth usage (we're using grandstream phones) and we're quite happy with it up until I tried using IAXPhone 0.2.0 build 116 with my asterisk 1.0.0 installations. Weirdly enough, calls from IAXphone to the GS phone work just fine. So are calls from both phones to voicemail. And from both phones to analog phones connected to FXS ports.

Hello, I am wondering about the ability of a server that is simply passing G729 through it to have the ability to record the calls. I know for voicemail, meetme, and things like that to work, a G729 license must be installed on the machine since there is transcoding going on. Is this also true for recording of calls? Will I require licensing for each recorded call? Will the server see a

-----Original Message----- I'm crazy here trying to make X-Pro use ONLY g729, and you're struggling to make it not to use it :)... Can you please indicate what's your config for X-Pro and sip.conf? sip.conf: [12345] type=user username=12345 secret=12345 nat=no host=dynamic reinvite=no canreinvite=no disallow=all allow=g729 allow=g729a allow=g723.1 allow=g726 allow=ulaw allow=alaw

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 come to realize that I don't have to have a g729a license in order to make use of an ATA-186 or 7460 connecting to another 7460. I just need to allow the codec in sip.conf. Now what ramification does that have when I dial out over one of my analog line (connected to * by a channelbank and a T100P) using my 7460 or ATA-186. The only benefit I am looking for is reduced bandwidth

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 noticed that on

Hi. I just tried updating asterisk and I guess I broke something. Here's the log: Unable to find a path from G729A to SLINR Unable to find a path from ULAW to G729A Any ideas on what I should try? I tried nuking all the zaptel stuff in the system and the source and started over agian. Also nuked the asterisk config files.... I saw this asked once before but there was no reply :-/

Hello all, I am running Asterisk 1.4.17. I have 2 Linksys SPA3102's and one PAP2-NA (I have a second on order). They have G729a built into them. This is supposed to be compatable with G729. I was trying to have them use that codec when they talk to each other, but it seems they always switch to alaw or ulaw (depending on my sip.conf file). Shouldn't they be able to use G729a in

Hi list! Again Wengo has made changes to their servers that require modifications to * configs. Is there anyone that has the 'new' wengo working with asterisk that could post their configs? Also they switched codecs, now G720a is required to connect. I can only find an (open) G729 codec, is this the same as G729a? Thanks!