similar to: Voicemail format - no transcode?

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

Kai, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice versa. I

Hi all, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

Dears Any one knows a standalone voip transcoder software name,not an ip pbx. What I want is to transcode the incoming sip calls from g711 to g723 or ilbc or g729 ..... and forward it to a media gateway .. Regards Khaled chehab ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by

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 -

What I am asking for is something to take an incoming SIP INVITE, change the codecs listed in the SDP, forward the (new) INVITE to a media gateway, perform the reverse codec handling for the 200 OK and perform RTP transcoding on the resulting 2 legs of the call. -How can asterisk do that ! -do any one know a distribution contain asterisk have solution like that ? Regards -----Original

Hello All, Anyone here has experience of accepting a ilbc call and sending it on g711 or g729 I am having problem in VOICE , call goes though but there is no voice. Senario: Call is coming in from Machine A to Machine B, sending to Machine C Machine B is an asterisk box, transcoding it from IBLC to G711 and g729. Problem: Voice is not appearing on the sip user sitting on machine A Already

Hello, I'd like to use g729 pass-thru when I dial out to a sip provider from my IP phone but because I have no license for g729 I'd like to use g711 ulaw for asterisk voicemail, conference bridge and other services. When I set in [general] section of sip.conf the following: disalow=all allow=g729 allow=ulaw the g279 pass-thru works fine with my SIP provider but when I call the

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

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

Ok, With everything restore on rtp.c, still I have no audio however the call is not destroyed immediately as before. I'm going to put a second Granstream box, and findout if between two boxes this happen too. I cannot believe that we cannot do 2 g726 on the same box at one time. Carlos -----Original Message----- From: Carlos Alperin [mailto:calperin@senecacom.net] Sent: Wednesday,

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my asterisk doesn't need to "transcode" (I don't have licences for g729) - sip.conf

Hello, We have an Asterisk server receiving calls using G711 (ulaw). This server has rerouters de calls to other server using G729 (we bought the codecs, installed, sip show channels shows the codec properly, etc.) Using G729, there is a "click" while talking. Well, more than a click it seems that voice is missing during some ms (maybe 100 ms?) Using G711 we don't have any click.

We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second

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While we're on the subject of G.729... I can end to end use it with no transcoding, but voicemail is the main sticking point for me - I'd need to transcode. So why can't voicemail store the audio in the format it's being streamed in on? Is there a technical reason for no voicemail storage in G.729? We have prompts in G.729, so why not the messages? It doesn't have to mix

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

How does the g729 encoder/decoder count in regards to the total number of licenses and how does it count an encoder/decoder? I looked on the wiki and don't really see anything that explains it. In other words, how do the calls below count (assume no reinvite)? g729 phone calls into voicemail g729 phone calls g711 phone g729 phone calls other g729 phone