similar to: G729 licenses installed - voicemail has no audio...

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

I am trying to create a %100 g729 (with no transcoding) system (using a Soekris, of course). I am running AstLinux with the native sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) - I think I am covering all of my bases. I have only "format=g729" in voicemail.conf. On an incoming call to a mailbox, everything goes well until recording the message. When the

Hello Everyone, We are looking to transition our 23 channels from testing/lab into production. During testing we used the free open source g729 license using the instructions found here: http://blog.manhag.org/2010/05/installing-the-free-g729-codec-for-asterisk/ A little more about our setup. All recordings have been converted to G729, all voicemail messages are also in G729, finally

Hi All; I need a help as the voicemail need GSM codec while I am using G729 for the call, why Asterisk does not do codec translation from G729 to GSM, it does not support? Any need for settings, what I am missing? Regards Bilal ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search.

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 guys, I got 2 licenses of g.729 and while running the asterisk with Monitor (for recording a channel) and using one channel for the call... I receive this error: WARNING[23826]: codec_g729.c:180 g729tolin_framein: Out of G.729 Decoder Licenses! many times.... it starts only when the call through the Zap channel takes place. while this error is being running on my screen I ran the cli command:

How does the g729 encoder/decoder count in regards to the total number of licenses and how does it count an encoder/decoder? I looked on the wiki and don't really see anything that explains it. In other words, how do the calls below count (assume no reinvite)? g729 phone calls into voicemail g729 phone calls g711 phone g729 phone calls other g729 phone

Hi list! My problem: I have three extensions in my Asterisk 1.8.30.0 and they have a voicemail. On two of these numbers the voicemail works without any problem, on the other it doesn't... I get this error: [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm) [Oct 17 17:01:29] WARNING[14700]: file.c:957

List, I have some cisco phones (7940) and asterisk 1.4 running nicely.. Communication between the phones is G.729, and my sip.conf looks like this: disallow=all ; First disallow all codecs allow=g729 ; allow=gsm allow=ulaw allow=alaw However, I cannot call voicemail - I get the following error: [Apr 20 14:58:31] WARNING[87184]: channel.c:2816

Hi, all!!! What will Asterisk do in the following case: For example, we have 4 licenses, and have 4 simultaneous calls, using G729. Will asterisk allow incoming calls from peer, that can talk G729 and ulaw, and will it force it somehow to use ulaw in this case? All phones there in LAN behind Asterisk prefer GSM codec, so it does transcoding. So, what I mean is will Asterisk fall back to use

hello all, i have problem in using g729 codec. my asterisk version is 1.8.22. when i run "core show codecs" in asterisk, there is a g729 codec in the list so i assume that i can use it for my channels. but connection can not be set when i use it for my h323 channel. i read somewhere that codec g729 is a commercial codec and i should buy its license in order to use it. is it true? if

Hi all, I have purchased 2 licenses of G729 from digium and has done the registration. It works well and is quite fine with my Asterisk@Home. Just want to clarify some licensing issues regarding them. If i had to do a full reformat of my PC and reload a@h again will i be able to use the licenses again without re-registration? If no. .Is there are limits for this? Please anyone clarify. Thanks

I haven't been able to find the answer online, and am not currently able to conduct an experiment to find the answer... I understand that in a SIP call where G729 has been negotiated as the preferred codec, a G.729 licence is not consumed until there is a need to perform transcoding, e.g. play a non-g729 sound, or do voicemail, or enter a Meetme, etc. What happens when a SIP call in progress

Asterisk isn't playing my voicemail greetings even though they are defined. Below are the relevant configs(from show dialplan) as well as the level 3 verbose messages asterisk is giving. Also a listing of the directory. Asterisk just plays the "The person at extension..." message, not the greetings I have recorded. Thanks -- asterisk*CLI> show dialplan macro-stdexten [

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 am trying to post this again as I am getting no answers and the support@digium.com bounces... (I have searched the whole list and can't find the answer either) I have installed a 5 user license for G.729 and want to route calls through Asterisk from my G.729 phone to Cisco AS5300 also using G729. Both Cisco and the phone connect using this codec if I do not force the call to go

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

Hey guys, I'm having a really huge problem, it seems like Asterisk is locking my licenses of g729 after being used. For example, 10 people make calls using this codec. Then I can see the channels and the codecs being used, cool. But then when they hang up the call the codes are still there, as being used... I don't know if its a bug or a miss configuration, but the fact is that I've

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

Hi. I've searched the wiki and the newsgroup to no avail. We have a couple of grandstream phones and purchased seceral copies of the G729 codec for use with them. We registered the codec (first time) on a testing machine and it worked. We registered it on the production machine (second time) and it worked. Then the production system crashed. After reinstallation we had to