similar to: Only 8kHz recorded after disallowing all but G722 codec on inbound

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:

Hello, Anyone? I have hard time to believe this is not possible with chan_pjsip. Anyway, may I ask how people handle the following scenario which I imagine should be quite common: - I have internal extensions talk to each other using g722. so their codec setting (with chan_sip now) is "allow=g722,ulaw" - I have carriers trunks that handle ulaw only (allow=ulaw) - calls between

Hello, I finally got to look at chan_sip to chan_pjsip migration again. This time I’m having problems with influencing codec selection on originating (calling) channel. It looks like PJSIP_MEDIA_OFFER only works on outbound (called) channel and has no affect on calling channel. My experiments and function documentation (which says “Media and codec offerings to be set on an outbound SIP

Hello Michael, you are referring to the following behavior - did I get it correctly?: outbound broken: asterisk offers g722 / g711 to provider (callee), callee answers g711. Asterisk now transcodes between caller and callee (g722 <-> g711). inbound works: call from provider: g711 -> asterisk drops g722 and passes g711 to internal callee -> no transcoding. As far as I know,

Let's say I have two devices configured and the follow call scenarios occur. [100] disallow=all allow=g722&ulaw Polycom phone with g722,ulaw,alaw,g729 [101] disallow=all allow=ulaw Polycom phone with g722,ulaw,alaw,g729 101 dials 100 -> ulaw to ulaw is chosen 100 dials 101 -> g722 to ulaw is chosen Ideally when 100 dials 101 ulaw would be chosen since it is the common format.

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,

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 there, I've been playing with a SNOM 360 and 190 trying to get them talk to each other using g722 with 16 kHz. However all I see in the SIP log codec negotiation is "g722/8000" which makes me believe that this is only a 8 kHz link (and that's what it sounds like). Anyone every managed to establish a 16 kHz wideband call between SNOM phones? Cheers, Philipp

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:

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

Well, I'm trying to migrate to chan_pjsip so that I don't have to do that. It's so surprising that the issue so seemingly obvious and trivial hasn't been addressed yet that I wanted to query the collective wisdom of this list to verify my observations. Thanks for github pointer. Michael On 7/5/23 16:46, asterisk at phreaknet.org wrote: > On 7/5/2023 4:19 PM, Michael

I've got the text files created -- thanks to Russell Bryant -- for re-building the core and extra sounds using another voice but I'm not sure which formats to actually build. This will be a small/personal system using Vitelity.net so will only have SIP connections. The /var/lib/asterisk/sounds/ directory contains .alaw, .g722, .g729, .gsm, .ulaw, and .wav. What are the minimal

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

Hello everyone, I am using the Asterisk REST API in order to establish a call to an endpoint and to send over a remote file (HTTP). The issue is that I am experiencing an audio quality issue. I have tried encoding the file differently, but everytime Asterisk is cutting the audio frequencies above 4Khz. The call is established with G.722 and the audio file is mono 16Khz 16 bit sln16 extension.

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

Hello, After I have re-read the "PJSIP Advanced Codec negotiation" document, it occurred to me that the desired behavior should actually happen automatically, just due to the codec negotiation logic, but it looks like asterisk doesn't actually follow the described logic which is likely a bug. Can you please follow with me through a simple sip call and see if I'm missing

We tried to use the CURL fn to POST json, but it's sent as form data and there seems no support for changing the Content-Type header. We switched to invoking curl in the shell. All the documentation I could find says there is just one parameter for the url and an optional second for POST body. Is there an undocumented way to set Content-Type? -------------- next part -------------- An 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

Has anyone found a way to enable the g722 codec as a prefered codec in the Polycom provisioning files for the 550's? I couldn't find a pref for voice.codecPref.IP_550. What needs to be put into the allow field (sip.conf) for asterisk to allow the codec? -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck at gmail.com http://www.shift8.biz

Response below... On Fri, Dec 27, 2019 at 12:02 PM David P <davidswalkabout at gmail.com> wrote: > > > > > I'm looking for a way of detecting in my dialplan when a peer becomes > > non-responsive after answering. [deleted] Is there a way to configure > > a handler for this state? > > > > We use v14.7.6 and we dial the peer this way: > > >