similar to: g729 test

Hello - I have 8 polycom 501s all setup great using ulaw. We have put them through a pretty rigorous torture over the last 4 months, and they've performed famously. No dropped calls ever. We invested in some g729 licenses. changed my ipmid.cfg so that g729 is priority 1 and ulaw is priority 2. I added allow=g729 to my extension's sip.conf entry, where existed before disallow=all

I am trying to figure out the best way to deal with this. I want all of the calls in the network to be G729 and this is working. I do have hardware that provides me 30 g729 licenses. I am setting each extensions to disallow=all and allow=g729. However when I have this setup, I get no voice mail prompts. I tried setting to disallow=all and allow=g729,alaw and I still have no audio when calling

I am using Teliax to terminate my calls, and I have 3 licenses' for g729 from Digium. "show translations" verifies that the registration took place. When I place a call, having "allow=g729" as the only allow option in iax.conf, I get the following error: WARNING[361]: chan_iax2.c:6017 socket_read: Call rejected by 208.139.204.228: Unable to negotiate codec If I place a

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

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've purchsed 5 g729 licenses from digium, but am a little confused about why things are acting the way they are. When I do show g729 I see: 0/0 encoders/decoders of 5 licensed channels are currently in use My sip.conf starts out: [general] disallow=all allow=ulaw allow=g726 then my hosts look like this: [6016716] username=6016716 accountcode=75415 type=friend secret=obsurified qualify=yes

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

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

We want some of our users to use G729, and some others to use ULAW. Our PSTN gateway provider supports both, so that's not a problem, and if I force him (the PSTN gateway) to allow G729 only, the outgoing call will take place with G729. The problem is that I want to have my PSTN provider configured to allow ULAW as a first priority, then G729. I did it like that: [mypstngate] type=friend

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

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

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

Hi, I use asterisk 1.4.11 version for making outbound calls. Running it on linux(fedora core 7) machine. Recently purchased the g729 codec, got it registered with my asterisk box. I have two queries for you to help me. 1. How do i know when an outbound call is placed that it makes use of the g729 codec. when i use the command "show g729" i get the following: 0/0 encoders/decoders of

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I've done some reviewing of the archives for G729 and H323 experiences. The landscape of that query isn't pretty - lots of pleas for help, and nor do I see too many "answers." I have a pending bid that requires some data before I can implement * on this particular solution. My question is perhaps a slightly differently worded one than has been asked before, but it may be

Hi guys, I've registered some g729 licenses, during register process everything worked fine. astk2*CLI> show g729 0/0 encoders/decoders of 20 licensed channels are currently in use But I'm not able to use this codec. I'm trying to use a linksys PAP2 to talk using g729 but I got this answer from asterisk: Got SIP response 488 "Not Acceptable Here" back from

I'm planning to upgrade my asterisk 1.4.4 to 1.4.6. usually for asterisk upgrade i delete modules directory and include, then compile the new version. Since i have couple of G729 Licenses on this server installed, would i need to call Digium to reactivate these Licenses? Is there any better and faster way of upgrade asterisk? Possibly without losing G729 License? Thanks!

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

I have multiple voiceage g729 licenses installed on a RH9 box, and have a remote C7960 configured to use it (low bandwidth). In calls like: Remote C7960 -> g729 -> asterisk -> g711 -> C7960 the audio is oftentimes rather choppy. Changing the remote 7960 to use g711 seems to eliminate/reduce the choppyness. Any ideas on what might be behind this?