similar to: is g729 codec free? or under license???

Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI> core show version Asterisk

Hello,all Is the text chat such as LINE not possible in Asterisk?

I hope this isn't considered cross posting, i sent the following email to Digium support but figured someone on the list may also have better insight into my questions. I have purchased 2 g729 licenses from Digium for testing and have the following questions; ** My configuration is a single asterisk box configured with 2 g729 licenses and 2 x Cisco 7960 Phones, I have confirmed the

Hi Users, I was wondering if someone of you have the same thing on CentOS 64x? asterisk01*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 lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723

Hi all - I am using asterisk 13.27.0 with Linphone. I turned off all codes on linphone except the one I want to try. For example: opus and speex (so only one enabled at a time). Then did this same on asterisk for the linphone extension. disallow=all allow=speex (for example). Then I place my call and the call fails. if I enable something like gsm, ulaw, alaw the call works fine. Why does the

Let's say I have two devices configured and the follow call scenarios occur. [100] disallow=all allow=g722&ulaw Polycom phone with g722,ulaw,alaw,g729 [101] disallow=all allow=ulaw Polycom phone with g722,ulaw,alaw,g729 101 dials 100 -> ulaw to ulaw is chosen 100 dials 101 -> g722 to ulaw is chosen Ideally when 100 dials 101 ulaw would be chosen since it is the common format.

Hello, Anyone? I have hard time to believe this is not possible with chan_pjsip. Anyway, may I ask how people handle the following scenario which I imagine should be quite common: - I have internal extensions talk to each other using g722. so their codec setting (with chan_sip now) is "allow=g722,ulaw" - I have carriers trunks that handle ulaw only (allow=ulaw) - calls between

Hello, How many licenses to buy?? : From what we understood from digium website, we must buy as many licenses as the number of maximum simultaneous calls using G729 Codec we wish to make. For example, If we want to be able to make a maximum of 10 simultaneous calls using G729 Codec, we must buy 10 licenses. Is it right? Thanks you

Hi list, we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk 1.4.36 (Elastix). Both 64bits, no hardware involved, dahdi on both machines for meetme timing. Doing core show translation give on the Lenny server Translation times between formats (in microseconds) for one second of data

Hello guys, i have a question about a voip gateway we use. I saw those values typing in cli: core show translation g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - - - - - - - - - - - - gsm - - 2001 2001 6000 2001 2000 16000 - 34002 - 6000

List, I purchased 2 g729 licenses but I can't get it to answer a g729 call from a cisco router with a vwic card. In the debug output below you will see that asterisk thinks it only supports: (gsm|ulaw|alaw|h263) when it should support g729 according to the config also listed below. The real odd thing is I can place g729 calls to the router, just not from the router to *. Anyone have any

I need a refresher course on how many licenses I need to buy. I have an Asterisk server that receives calls by SIP (G729) and then sends them to the PSTN via 32 Zap interfaces on an Astribank. I cannot remember if the license is per channel or per call so I do not know if I need 32 or 64 licenses for this application. Could anyone please remind me? -- Telecomunicaciones Abiertas de M?xico

Hello Everybody, Since I upgraded to 1.4 I always get the difficulties as below, which I have never had in 1.2: [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Call accepted by 202.153.128.34 (format g729) [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Format for call is g729 [Dec 28 21:06:00] VERBOSE[1756] logger.c: -- IAX2/VoIPRakyat-2 is ringing [Dec 28 21:06:00] DEBUG[1756]

I am wondering how the g729 licenses are done during calls. If I have N licenses for g729, and N are in use and an additional call comes in that requests N+1 to be in use, how does asterisk handle that call? Does it dump it? Does it negotiate another codec automagically? Basically what happens to that call, obviously it wont (shouldnt) let you use more licenses than you have available, but

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

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

Hi, I have trouble re-registering a G729 license for Asterisk (bought 6 years ago) My license looks like: 10D2XXXXX-XXXX-XXXX-XXXX-XXXXXXXXXXXXX Tried to re-register the codec according to the http://downloads.digium.com/pub/telephony/codec_g729/README document, but the register failed with this error message: You selected 5, G.729 Codec Please enter your Key-ID:

I've an Asterisk 1.2.4 installation, where I've also installed the G729 codec license. I'd like to upgrade that installation to 1.2.5, but I'm not sure if I'll lost the license in the process (and if I'll be able to recover it later!!!). Is there any special consideration I've to keep in mind in this case, or should I just run the typical "make + make

Hello! I'm facing the following scenario: - Initial call opened to asterisk: SDP g722,alaw,ulaw - Outgoing call to provider started with Invite / SDP alaw, g726 and g729. - Provider sends 183 Session progress SDP: g729, alaw - Provider sends g729 rtp packages But: there is no license to transcode g729. What is asterisk doing? Asterisk decides to stop the call at all: - Sends cancel

Greetings all, First time poster, Sorry if this has been answered here before. We recently replaced a failed 1.4x asterisk PBX at a customer location. Voicemail access was setup when the customer dialed *8, This worked in 1.4. Now, Running 1.6 (I know it's old I had to load it quickly, And that's what I got working first. It'll get upgraded to 1.8 soon). The strange part is *8 no