similar to: Dimensioning

Where can I find alaw, ulaw, gsm, g729 formats for native music on hold? I have some mp3 files and I have tried to transcode them to above, but it seams that SOX can't do that. Please, tell me where to download some MOH files (in above formats) or how to transcode mp3? Thank you for your time! -- Tomislav Parcina tparcina#lama.hr

Hi To continue the interesting streamTranscode thread but in a slightly different direction, I have a user station that would like to send us a 64kbps AAC stream and have us transcode that to 64kbps MP3 and 32kbps AAC - is this possible? Points to consider: a) we would be happy to go with another permutation such as: she sends us 64kbps MP3 and we convert that to 64kbps AAC and 32kbps ACC. b)

Greetings everyone, I was wondering if there was a way to change the codec that Asterisk uses when streaming via MulticastRTP. Or perhaps a way to transcode the multicast stream. In the CLI, when I have a multicast stream in progress, I am typing 'core show channel MulticastRTP/0x7f7........' to get lots of helpful information. I have noticed that when I do a MULTICAST page and send data

I will be using a dell R320 Xeon E5-2420 2G and 4G RAM. also using a SIP trunk with ulaw/alaw codec. How many calls could I expect to make at the same time? no transcoding or anything. Just call a number and play a gsm file. Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm new to Asterisk and have a couple of basic questions. We're interested in using * simply as a SIP <-> PSTN gateway using a T400P connected to one or more ISDN PRI lines (instead of using a Cisco box which would cost more and come with no hackable source code :-) First, is Asterisk's SIP stack up to date and fully functional with respect to the SIP protocol? Are there

Hi, This seems to be a fairly common question, but I have Googled for this quite a bit and looked at the Asterisk documentation/book and haven't been able to find an answer. My question is: Can I get my IP phone to select a different codec depending on the final destination of each call? I've got these things connected to my Asterisk box: - Snom 300 phone (supports g729 and

Zonbu.com sells a little box I was planning to use as a router, but I couldn't resist putting Asterisk on it just for fun. It may never see its intended purpose. The box costs US$249 (and was delivered 40 hours after being orderd!), but you can get it for less if you subscribe to their service. I didn't. The box is an Intel compatible processor (VIA Esther processor 1200MHz) with

Dear Thomas I Really Appreciate your enlightment Thomas B. Ruecker wrote: > let's stop RIGHT HERE! > Isn't it obvious? > Something is wrong with your pipeline here. There is NOTHING coming through. > Which brings us back to the previous point. > First make sure ffmpeg2theora has all the right options set, then verify > that it produces an valid ogg file/stream and only

Hello everyone, As I promised at eTel last week, I have finished up work on my "Asterisk Native Sounds" project. Here's a little diddy from astlinux.org: ----------------------------------- Asterisk Native Sounds are a collection of audio prompts for Asterisk. They will improve quality, reduce CPU usage, reduce latency, and (in some cases) eliminate the need for G729

As stated here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe A Meetme room uses Ulaw as the audio codec, so if the other channels use different codecs, then * will transcode. Does the app_conference application works the same way? Or if i have SIP/g729 users and i create a conference with other users also at g729 asterisk will not transcode (when using app_conference)?

Hi, Some help with dimensioning the server will be gladly accepted. -50 sip phones (g729) or g711(to avoid transcoding) in LAN -an asterisk server (1.4) doing normal pbx functions + voicemail in the same LAN -Some sporadic conferencing with no more than 2 sip phones and maybe 2 or 3 calls coming from the E1 for a total of 5 people in a conference. The asterisk server will get an E1(pri) via one

Hi there, I had a problem, basically, I have 4 different types of end users (gsm, ilbc, g729, ulaw). However, I only have one user with my DID provider. My provider supports all 4 codecs. The issue is then: When an incoming call comes in, a codec is negotiated (usually ULAW), later on, when the extension is dialed, we'll see we're doing GSM, and thus transcode. Here's an example

But they row.names() cannot give me the IDs On Wednesday, November 1, 2017 9:45 AM, David Wolfskill <r at catwhisker.org> wrote: On Wed, Nov 01, 2017 at 04:13:42PM +0000, Elahe chalabi via R-help wrote: > Hi all, > I have two data frames that one of them does not have the column ID: > > > str(data) > 'data.frame': 499 obs. of 608 variables:

Hello, Does asterisk do transcoding when the call goes through the system, codecs are the same but signaling protocol is changed. example: SIP with GSM ---> IAX with GSM What quality destruction happen when I use transcoding? I know this is not a concrete/precise question, but I would like to know how is it in general. What CPU performance is needed for transcoding 30 channels e.g. from

Hi All, I have recently installed a TC400B card into a system and am trying to get it to work. As far as I ca tell from the docco on Digiums website, there is no config as such unless you want to enable / disable only 1 codec, otherwise by default it runs as 92 channels of either. I have tried asterisk 1.4.9, 1.4.10 and 1.4.10.1 along with zaptel 1.4.4 and addons 1.4.2. The zaptel modules

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

I have been trying to get a feel for scaling or dimensioning using asterisk 11. if I desire to use something like a dell r320, hardware RAID, 2G E5-2420, 4G RAM and only SIP trunking using gsm (least bandwidth and no transcoding) how many calls "out" can I expect to make at one time and asterisk still be OK and responsive? Thanks, Jerry -------------- next part -------------- An HTML

Hello list, I have a file to be played in wav-format. I thought Asterisk would automatically take the wav-file and translate it to the codec used, but I see this : [Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File /var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any format [Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to open

When I call inbound with a cell phone (via SIP PSTN trunk) some of my prompts the first word is cut off. I'm assuming the prompt is needing to be transcoded on the fly and it's not getting transcoded fast enough. I did a file convert to create gsm versions (currently they are referenced in my dial plan with no extension Seem to have same problem. How do I determine which file

Hi, is there any good calculator/table/reference about proper dimensioning? I read the wiki and they basically say "xx users run fine in yy hardware" http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning. SO far I read that: -Run up to 4 E1s per CPU (which one? an i386 or a dual core? -it is very CPU intensive to do transcoding. Try to minimize it. -you can help the CPU