similar to: Sangoma transcoding card bug - drops audio samples

Hello, I try to move our asterisk installation (3 Asterisk servers in different offices connected using IAX and a lot of SIP phones, as well as ISDN connections using CAPI) to use G.722 instead of G.711. Asterisk 1.4.25.1 is used with the G.722 patch (the fixed one, which solves the gain problem). So SIP-to-SIP and to ISDN there is no problem. G.722 itself works and transconding to G.711 for

Could this gap in sequence numbers caused by a codec change generate errors like the one below? [2018-02-13 12:57:43] WARNING[4917][C-0004c2cb] codec_sangoma.c: [526559][g722toulaw] Got Seq 15944 but expecting 10106 (time since last read = 0ms), dropped 5838 packets On 02/13/2018 01:24 PM, Andres wrote: > On 2/13/18 11:55 AM, Michael Maier wrote: >> On 02/13/2018 at 08:41 AM Floimair

On 02/13/2018 at 08:41 AM Floimair Florian wrote: > No you're reading it wrong. > > There are 188K received with no loss, and 16441K transmitted. This doesn't make any sense to me, either. There can't be more packages transmitted than received. It's the same codec in and out and it's been running exactly the same time. > ...........Receive.........

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?

It appears that there are no transcoders from g723 to anything else in Asterisk 10.7.1. Does anybody know how to fix that?

Hi All, I have 2 SIP clients configured and connected to Asterisk. When I place a call from SIP1 to SIP2, if both codecs are the same then everything works as expected. I then allowed one of the clients to use alaw instead of ulaw and there were audio problems (couldn't hear the other end, etc). Same thing happened when I tried to use gsm<->alaw/ulaw. Any ideas? I'm using

Hi, I am trying to configure asterisk to translate between rfc2833 and inband DTMF. I have a cisco gateway which is configured as a trunk, and a cisco IP phone which is registered to asterisk. The gateway does not support rfc2833 and the IP phone does. I tried changing directrtpsetup to "no", and that didn't help. I tried changing "canreinvite" to "no", but that

I'd like to store all of my music in FLAC form; however, I would also like to stream the library as Vorbis. Is there a tool to do on the fly stream encoding to Vorbis from FLAC? Basically, I'm looking for something that is almost like a DJ application that lets you pull in the FLAC files, do whatever DSP algorithms you need, and then code them into Vorbis and shove them out a stream.

Hi, I have written a small app in C# that uses oggdsf to transcode to Ogg Theora on Windows. I thought some of you might find it useful. You can download it from: http://www.a2ii.com/tech/directx/TransTheora.zip Cheers, -daniel ps. it will only handle files with a single audio and/or video stream _________________________________________________________________ Win a trip for four to a

Is there a way for voice media clients (like SIP phones and POTS/PSTN phones) that connect their call legs to Asterisk to negotiate a common codec that they both use at their end, so Asterisk doesn't have to transcode? Asterisk would know which codecs each client can use, and which each prefers, then find the one they each have in common so the fewest legs need Asterisk to transcode to their

Sorry for a possible retransmit: the first was sent from an incorrect email address. I'm trying to use the Polycom SoundStation IP 7000 with Confbridge. But the transcoding from siren14 to slin32 is via slin. First, it seems odd that there's no transcoder directly to slin32 since anything else will lower fidelity. But, more importantly, there is transcoding from siren14 to slin16 and

Can't find an answer to this, but maybe I've not looked hard enough ... Does MixMonitor work without transcoding? ie. if I have a g729 stream passing through and I'm recording it with e.g. MixMonitor(/dump/filename.g729,b) and specify g729 in the filename, does MixMonitor transcode both legs of the stream to a format it can then "mix" then transcode it back to g729 to

Hello, I'd like to ask whether there is some documentation with recommended parameters for transcoding voice codecs such as G722, G711a/u <-> Opus with near-transparency. My Idea is to have something like: HW-Phone <-> Asterisk <---------> Asterisk <-> HW-Phone (G722) (Opus) (G722) in order to lower the bandwidth between the two

Soliciting your help on the following question ... XPL uses the UTF-8 encoding for its identifiers. As such it supports Unicode and many non-ASCII characters. LLVM uses std::string for identifiers which is based on a signed character which only supports 7-bit ASCII. Although the size of the characters in both schemes is the same, the bit encoding is different (UTF-8 is unsigned, ASCII is

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

Before I put myself to the task (next month, maybe) of surveying the CPU costs of transcoding, perhaps someone else has already done this work and would be willing to share it or refer me to a link of previously published data. My reviews of the mailing list with various keywords were unsuccessful in finding adequate references, though I admit I only spent 20 minutes looking. What I seek is

Okay this may be a very noobie question, but I've looked around the WIKI and haven't found anything about this. I was looking at some VO-IP phones, an they list one of the codecs as g.711, this isn't listed on the * page I was looking at, is this also known as another name, or is it not supported by *? Thanks Joel Duffield Near North Business Machines www.NearNorthBusiness.com

Got two phones here. 1 is Cisco 7960 and other is XTen Pro. Both have 729 capabilities and plenty of licenses on Asterisk. The Cisco phone has and registers/talks with asterisk on an internal IP (* = 10.0.3.10, phone = 10.0.3.151). The SIP peer for this phone is set to NAT=No and has this Codec Order: (g729|ulaw|alaw|gsm|g726). The XTen registers to the Asterisk external/public IP, even when it is

I have a connect to * via H.323/g711 from device A and want to connect to B which want for H.323/g729 h323.conf contains disallow=all allow=alaw allow=g729 but outgoing faststart/TCS contains only g711 (from h323_request(format) i think) and so no codec negotiation and no voice. Howto run up g711/H323 -> * -> g729/H323 PS intel's g729 was used. ast 1.0.3-6 PPS stupid -

Hi, my setup is like: phones (g729/g711)--(SER)--> Asterisk <--(oh323)--gateway (supports g729&g711) problem begin when phone supports only g711 and Asterisk doesn't negotiate this codec in full path (from phone to gateway), but tries to do transcoding (and because I haven't g729 codec in asterisk, the call fail). Is there any solution how to tell to Asterisk to negotiate