similar to: Enhancing performance and utility of an Asterisk machine

Hello, I have transcoding card TC400P installed in server running Debian with Asterisk 1.4.23. Everything seams to be fine and after I boot up server I see in dmesg: 7.590966] Zapata Telephony Interface Registered on major 196 [ 7.590966] Zaptel Version: 1.4.12.1 [ 7.590966] Zaptel Echo Canceller: MG2 [ 7.610963] zttranscode: Loaded. [ 7.618969] wctc4xxp: tc400b0: Attached to

Obviously, there are no DS3 TDM cards that are currently compatible with Zap channels. (or are there?) Does anyone know of an inexpensive DS3 card that could perhaps be used with Asterisk if one were to try to port the Zap drivers to such a card? PCI, of course, would be the bus of choice. I think there are quite a few discouraging comments to be made on that question. Firstly, most

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

Hi, Given that the Digium transcoding card has no external connections (AFAIK), it strikes me that it would suit a mini-PCI slot very well. Does such a beast exist, or is it likely to? Am I correct in assuming that this is a Digium-only product, and there is no OEM equivalent "generic" board out there that I could be investigating? It would be such a shame to waste a PCI slot that

Thanks Alex, some great ideas. I think, however, I'm leaning towards Asterisk at this point- since I have quite a bit of experience there, and very little with SER. At this point, I'm wondering from a dimensioning standpoint, what kind of capacity my machine will have (Dual Core Xeon 2.4GHz 4GB RAM). As I said, I don't plan to do any transcoding. I read the voip-info page on

Hi there, up till now I had this two-box setup in mind: * no.1: public IP * no.2: private IP, registers with no.1, serves a small office with clients behind NAT See we'd get something like this: SIP client (GSM) --> *1 --> IAX2 (iLBC) --> *2 --> G.711 --> MGCP UA The codec of the SIP client (on the Internet) I don't have full control over, that depends on the

Hello all, I have a TC400B Digium card in order to deal with transcoding and I have some trouble using it, I have a timer synchronisation problem! I would be very grateful if you have any idea to help me? It seems that the card is not correctly synchronised to the system because when I speak to one side, the sound takes 5 seconds to go to the other side, and increasing, after 30 seconds of call,

Hi, Thanks for quick reply. https://en.wikipedia.org/wiki/Multimedia_framework There are many popular types of multimedia framework, such as gstreamer and directshow. I want to make transcoding program in centos 6.7 which is installed in a server. ffmpeg is used as the core of transcoding program. And I should choose one multimedia framework to do software development. Which framework is

In case there are any Debian developers around here, I wanted to point out message <20020828154322.GA15114@chulak.naquadah.org> on the debian-devel list this morning. Another Debian developer is proposing to submit an mp3->vorbis transcoding program for inclusion in Debian. I have objected to this on the grounds that the resulting vorbis files will sound like crap, and I have also pointed

Hello I use asterisk realtime, and I can set the order of codec preference on my realtime allow column. If I could disable transcoding, then I can always ensure a passthrough of the common codec from origin to destination without transcoding (expensive on CPU) - and more or less, force the codec to use by setting the codec preference So, can I disable transcoding?

As far as I can see, transcoding could be usefull for people who do not primarly care about quality but about filesizes. One could assume that such a user would have a collection of mp3's at 128kbps or higher bitrates, and uses an encoder like BladeEnc or Xing. He wants to take uses of ogg's supposed quality and transcode his 128-or-higher files into 96 or 112kbps oggs to save diskspace.

Hi, We've been running g.722 in asterisk 1.6.09 for awhile now,

Hello, I have a T100P card on a Red Hat 8 machine with Asterisk installed on it and I'm trying to get the data routing properly through it. I purchased an Integrated Data/Voice T1 Line with channels 1-6 voice and 13-24 data. The line is plugged directly into the Asterisk machine's T100P card. Previous to the asterisk set up we had an Adtran router that connected everything to what the

Situation:? * remote virtual server with very little storage (estimate: I can spare about 40G for music) * local music collection of ~80G in all sorts of formats - lossy in varying quality, some lossless too Vision: * stream my whole music collection randomized so I can listen to it anywhere Plan/Idea: * Locally transcode everything to one format that results in files that are?

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

Good afternoon, I'm trying to use the CLI command file convert on an Asterisk 1.4.26 server with a TC400B transcoding card. The transcoding card is working well for calls but I have some trouble converting sound files from alaw to g729. The command creates empty file as you can see below... CLI> file convert /var/lib/asterisk/sounds/fr/service_notactivated.alaw

Opus or AAC will give you comparable results at reasonable bitrates (~128k). Though, I would suggest finding a way to get more storage. You could upload to Backblaze B2 or AWS S3 for pennies, if your current host won't let you upgrade. On Sat, Apr 15, 2023 at 3:36?PM D.T. <ohnonot-github at posteo.de> wrote: > Situation: > > - remote virtual server with very little

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 Everyone, We are looking for solutions where the transcoding is abstracted away from our * box (i.e., to the network layer) using some carrier grade gateway, or router. The reason for my post is to know about solutions people have used in the past, and how it fits into their overall architecture. Our transcoding needs consists mainly of u/alaw <-> g729, and gsm would also be good....

I created some test samples and transcoded to FDK AAC and libopus at fairly low bitrates - I cannot recreate what bothered me about Opus & noisy music previously. It also seems I cannot tease ffmpeg into encoding FDK's AAC with VBR. As it stands, Opus clearly wins in this scenario.* Q: Is it possible to stream in variable bitrate? * ffmpeg -i "$track" -vn -ac 2 -c:a libfdk_aac