similar to: EVRC support

Hi all, I realise that asterisk's codec negotiation has been discussed in the past multiple times. What I haven't been able to understand is how asterisk decides which video codecs to advertise to the other end when canreinvite=no in sip.conf and the initial caller doesn't support video. My tests are quite simple, I use an asterisk with 4 peers all on the same LAN. My sip.conf

Does anybody knows if it's posible or if there is some develoment in course to be able to use longer transmit packet sizes (as long as I know this is fixed in 20ms now) with the compressed voip codecs in asterisk (g729, g726, gsm, etc). I need to use asterisk to connect remote sip clients with 24kb bandwidth lines and I'm using a licences g729 codec but because I can't increase

Dear all, I am doing an interop testing with asterisk-1.6.0.5 now, and I have a question about the G.726 codec on asterisk. While my IAD supportes G.726-16,24,32 and 40 codecs, when doing a testing about G.726-40, I found that asterisk removed the G.72-40 sdp attrib when transmitting the INVITE with SDP. I modified sip.conf in order to solve the problem, G.726-32 is ok when allow=g726, but

Hi all! Quite a mystery. The following happened when I was on holiday, and no one else has changed any configs of either Asterisk or the Cisco's in the building... The situation: Incoming works fine on all phones. Outgoing only works from non-Cisco phones. When calling from a Cisco phone to an external phone, all the Cisco-user hears is a ticking crackle and after about a minute the phone

Hi, When ever I put g729 in allow for trunk the other two codecs (ulaw and alaw) stop working and I get the frame type error for them, but g729 works fine. I've cleared general part of sip.conf of codec info to be on safe side. If ulaw and alaw are the only ones allowed they work fine. Asterisk shouldn't be doing any encoding or decoding, all codecs should be passing through. Any

Hi everyone, I have an issue which is kind of a catch 22 situation. I had outgoing calls to my new PSTN provider working perfectly. Then I started focussing on incoming calls. It seems that I can solve an error which gets my incoming calls working but that in turns means my outgoing calls don't work. - Strange. Anyhow I was getting an error: Process_sdp: No compatible codecs! And from

Hi, I have just signed up with Voxee.com and have attached my Asterisk server to dial them via IAX2. Below is the start of the log which dials the number and promply hangs up when the call is answered, with the logs saying that the channel is not compatiable. I have traced this down to the g.729 codec which I don't have installed. Any ideas on how to force that the codec not be used?

> On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote: >> I receive an INVITE/SDP containing: >> >> m=audio 11310 RTP/AVP 3 0 101 >> >> which I interpret as gsm, ulaw, rfc2833. >> >> and I reply with an OK/SDP containing: >> >> m=audio 15884 RTP/AVP 0 3 101 >> >> which I interpret as ulaw, gsm, rfc2833. >>

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,

Hi all, I have encountered this problem: if the caller is connected to the callee using Dial() command called from extensions in extensions.conf, there is no problem. But if the same caller and callee are connected using an AGI->exec('Dial'...), the line is disconnected when asnwer. There's a problem bridging. If the codecs are the same on both ends then there is no problem.

is there a trick to get asterisk (1.6.2.13) to propagate codec-changing sip reinvites when directrtpsetup=yes? i'm trying to route calls to a gateway without keeping asterisk in the rtp stream. the gateway is first routing the call to a media server. when connecting the call to the downstream carrier a different codec is selected. the reinvite makes it to asterisk but asterisk isn't

Hi, I'm a bit confused about how Asterisk decides in which order of preference it should list the different codecs in its SDP message during SIP call setup. In my sip.conf [general] section I've got disallow=all allow=gsm allow=ulaw allow=alaw But when Asterisk bridges a call from an E1 to VoIP it sends out an INVITE with the codecs listed in the following order of preference

Hello People, I am a newbie asterisk and happy user, i have configured a x100p card and everything works nice, i can forward incoming connections to a x-lite software client and works out of the box, However when i try to make a connection between two x-lite clients then no audio is transmited, i have followed the instructions on voip-info.org, the tutorials on onlamp and i have read some

I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711). But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message: -- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack --

Banging my head against a brick wall trying to get a working H.323 implementation for CVS-HEAD. (The ONLY H.323 I have had working is OH323 v0.6.5 with CVS-STABLE - see my other post regarding compile problems on OH323 for HEAD) So, I thought, lets try this wonderful chan_woomera (dubbed "H.323 for Asterisk that works!"). I get exactly the same kind of problem as I have previously had

Hi Guys, I have a the following issue when I ma trying to place a call through my voip provider, I am using an asterisk 1.2.21.1, do you have an idea what could fix this issue (as you can see when the other party answered, the call get dropped because of probably sip incompatibility) Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit frame type 256, while native formats

Hello, I am running asterisk 1.4. For argument's sake I have 4 telephones. 2 support video, 2 do not. Calls between phones work fine and codecs are properly negociated. I have videosupport=yes in sip.conf and when the two video phones communicate I have video. I call meet me with this command EXEC MEETME 1234|d SIP looks like this : -- AGI Script Executing Application: (MeetMe)

Hi, I have an issue which is kind of a catch 22 situation. I had outgoing calls to my new PSTN provider working perfectly. Then I started focussing on incoming calls. It seems that I can solve an error which gets my incoming calls working but that in turns means my outgoing calls don't work. - Strange Anyhow I was getting an error: Process_sdp: No compatible codecs! And from the SIP

Hello, I'm wondering why asterisk is not transferring the callerid to the sip device. Scenario as follows: sangoma <---> zaptel <---> asterisk <---> sip <---> SIP-Device zaptel is reporting the callerid, but in the sip packages the sip-address shows unknown as user part, as this sip debug package shows: Executing Dial("Zap/62-1",

Salve, Imagine ogg vorbis is used to produce radio with free software. A journalist would produce a report end send it with 24kBit/s out from a cricis place somewhere in the world. DRM is going to use MP4 - so his report has to be reconverted with loosing quality :-( Can you imagine to have an free codec someday that would work in embedded radio-reciver like MP4? If yes, should DRM not be open