similar to: SNOM, g722 and 16 kHz audio

Hello, I have some problems to use G722, when my client sent an invite request to asterisk using G722/16000 codec asterisk respond with G722/8000 codec. I dont know exactly if Asterisk supports G722/16000 codec?? If yes how can I activate It?? Thanks. Rachid. Below wireshak trace:

Hi, I wanted to follow up on this thread about WB support on the MeetMe bridge that is in 1.6.2. Does it only work for G722 or any WB codec ? I am working with another 16k WB codec that I can transcode to 722 and vice versa. I was curious if the 1.6.2 MeetMe bridge can also mix 722 with any other WB codec natively(without downscaling). Thanks, Serhad Doken ------------------> Razza wrote:

We are getting the following on 1.8.3 and 1.8.4-rc2, HELP! Why is Asterisk unable to transcode to/from ulaw and g722? [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw,

We want to record inbound channels at 16kHz, but send only 8kHz to our peers. I've set our default profile in sip.conf to disallow all but g722, and the peers disallow all but ulaw. We have a proxy in front of Asterisk that is configured to disallow all but G722 also. My test calls show inbound to the proxy is recorded at 16kHz, inbound in Asterisk is only 8kHz, and the peers receive 8kHz. So

Hi all, I?ve read that meetme works at G711 (ulaw), so asterisk would down-mix a number of parties using G722, is that still correct? If so, i?ve also read that Joshua Colp was/is working on a replacement (conf_bridge?) that works with G722. If this is this still in active development are there any planned timelines? If it?s in 1.6.0.6, and i?ve just missed it or it?s been renamed please be nice

Hello, I am working with several SIP projects that use g722, or are trying to do so, with pjsip library. According to pjsip team's interpretation of g722, it works with 14bits PCM for input/output, so pjsip basically 'converts' the audio sample from 16 bits to 14 when encoding and vice-versa. Some implementations don't do 16<->14 bits conversion, so when pjmedia talks to

Hi, Does anyone know of a technical reason why FLAC cannot support 88.2 Khz files? I have a reason to uses this rate since it is easy to perform quality conversions from 24 bit 88.1 Khz master files (stored as flac files) to 16 bit 44.1 khz files for CD mastering purposes. I suppose I could Kludge the wav files so that they were half speed wav files at 44.1 khz and then hand the over to Flac, but

You asked in the Vorbis list, but your text only mentions OGG. The codec commonly used in OGG containers that is limited to 48 khz is Opus. Maybe you are trying to use the wrong codec (i.e. Opus instead of Vorbis)? Using a 44.1 khz wav file, I was able to encode a 192 khz ogg-vorbis file with the following command: $ oggenc --resample 192000 input.wav Of course, if your original material is

Hello, congratulations to the Ogg Vorbis team - RC2 sounds good. But... RC2 in 128 kbps mode seems to clip off all frequencys beyond 16 KHz. On the tracks I tested Beta 4 gave response even beyond 18 KHz. Some testings on a randomly chosen track: (other tracks gave similar results) Artist: Judas Priest Album: Jugulator Title: Bullet Train Beta4: 127 kbps, ~ 18 KHz (!) RC2: 132 kbps (!), ~ 16

I found some posts regarding this issue dating of September 2003, but no real answer. The ADPCM format supported by Asterisk (the .vox files) is 4-bit, 8 kHz. I need 4-bit, 6 kHz, which is also a widespread Dialogic format, to help migration. Is there an existing format/codec for this? If not, can I make myself a shared object in /usr/lib/asterisk/modules? Is this easy??? :-( Thanks, Yves

hi, is it possible to encode 16 bit, 22 kHz, stereo/mono WAV files to FLAC files or could there be a problem with the low frequency 22 kHz (lower then CD quality)? PS: I'm a FLAC beginner thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/flac/attachments/20070512/63ed58f6/attachment.html

In a recent interview someone from Digum indicated that the G722 wideband codec was being worked into Asterisk. This will make Asterisk compatible with Polycom's new HDVoice products like the IP650 phone. This is very interesting, potentially exciting, but it brings up certain questions. Who will benefit as long as calls must typically pass into existing PSTN infrstructure, and so be

One thing that has always bothered me about the ogg format is the distortion of high frequency sounds - even at data rates as high 128 and 160 kbps. I find the best way around this is to resample the wav file to 48 kHz (using SoundForge 6.0) before encoding (using CDex) to ogg. It takes a while, and adds a lot of extra wear and tear on my drive, but what a difference! The result is an 80k ogg file

Hi, I'd like to know the reasons why CELT supports only signals with sampling frequency in the range of 32-96 kHz. In effect, it can clearly outperform speex at high bitrates, and has potential to be used in high quality voice communications even for 11, 16 and 22 kHz speech signals. It could also compete with SILK codec (to be soon released by Skype). See this page for more specifications

Hi list, I have an 44100 Hz file with human voice, stereo with 16Bit. When convertig this to 8 kHz, mono I loose a lot of quality and have some ground noise. I tried several sox options but without success. Can somebody help.... best regards Thomas

________________________________ Hi David, Thank you very much for your reply. Since I need to resample the audio in the program itself, I decided to try out the resampling API in speex. But now, I have another problem. The resampled sound is very much distorted and clicks appear quite often. (I have attached the source code I used for testing it below). The test data I had was a file sampled

So I've got a bit of a head scratcher. Wanted to get some insight. I've got a PBX running 12.3.0 We're a ULAW shop from end to end. But I've been playing with G722 just for fun. I've got a Yealink T46G on my desk, And my colleague, A Polycom IP650 (Same office). Basically, Whenever I make an outbound call to a destination to something not G722 ready, I get no

Anyone got any subjective (!) views on the merits of these two ranges , using asterisk 1.4 ? I need to supply approx 30 handsets to a new client, with the senior managers (6) having some slightly more "managerial" phones than the base phones which will be used for one line only. TIA Julian -------------- next part -------------- An HTML attachment was scrubbed... URL:

Under 1.4.35 I get this message printed MANY times [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096) [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000

Greetings, Sounds like a simple thing to do, but I was not able to do it on these particular phones. Snom wiki was not helpful. My client wants to keep his phones pointed to UTC time, no DST, no change in timezone, i.e. to stay at 0 hours difference. The phones are provisioned from a tftp server. If I remove 'dst' value from the provisioning file, on bootup phones force users to pickup