search for: g711

...t by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 <----------------------> g711 | g729 <---------------------------> g729 rtp rtp After a while, we have the reinvite sent by the SIP provider with g711 in the SDP. So asterisk need to change audio codec f...

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 invite requesting G729. However, they proceed to send us a G711 encoded audio str...

Hello, all. I have some problem when testing my IP phone with Windows Messenger. My IP phone supports such codecs as g729, g711. And Windows Messenger supports red, g711, g723 as you know. The problem comes up when testing with this sip.conf file. ([general] context displayed only) =================================================================================== [general] port=5060 bindaddr=0.0.0.0 disallow=all allow=al...

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, there is no working solution as of n...

Hello list, I have many IP hardphones like Siemens 300 basic ( old ) , cisco ata.. etc I need: G711 from old phones must be convert to G729 via asterisk and send to provider ( G729 from digium ) I have this problems: oh323 (last version): ------------- asterisk work with this driver ok for old phones, if I only faststart=no . But problem with codec , asterisk can speak with provider ( G729...

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...

Anyone have a customer sending/receiving multi-page faxes over Verizon Business SIP trunk/g711 ? Verizon Business indicates they don't support it, and I have 2 recent customers that it doesn't work for, and 1 current large customer telling me he's going to make it work <grin>. The issues is the latency/jitter on fax/g711 over Verizon Business seems to spit out only...

The scenario we have is fax send/recieve software that ONLY talks T38 and an asterisk box. We have ITSP providers that do NOT talk T38 but G711 only. Does asterisk have the capability to take the T38 call from an ATA or T38 software then bridge/transcode it and do G711 out to the PSTN providers? If not is there another product PAID or FREE software or hardware that can do this easily and reliably? Thanks very much! Steve Gladden --...

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 mach...

Dears; I do not know if any had experience in using speex or ilbc with IAX and got good results, because I am facing a problem with GSM. I am facing a noise problem when I am using GSM with IAX trunk as following: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Remote Asterisk Box ---> Digium Card (FXO) to terminate the call to the destination While no noise happen if that senario is happening: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Rem...

Hi, We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip encapsulation. Most calls surprisingly work, presumably by th...

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 conference extention the call gets dropped because it wants to us...

All, Has anybody else experienced garbled voice between a phone using alaw/ulaw and one using iLBC? I have a Nokia E series phone with a preference to use iLBC and this works fine in Asterisk 1.2. However, since moving to 1.4 - I get garbled voice on Inbound (g711->iLBC). Outbound voice seems fine (iLBC->g711) though. It's not a 20/30ms framing issue as the phone uses 30ms frames the same as Asterisk. Interestingly, my Grandstream seems to work well with iLBC though. X-Lite on the other hand often starts fine but sometimes becomes garbled...

Can anyone recommend a codec pack with G723 for use under Vista? I have G723 prompts (about 70 prompts totaling 1MB) needing to be converted to G711 uLaw. I tried Audacity but it doesn't have G723 codecs. I tired some google found adware free tools and websites with no success in converting G723. It does appear the old Cool Edit (now Adobe Audition 3.0 for $349USD) can do it -jason ----------------------------------------- Di...

Hi, Does anyone know if Asterisk is a good tool to be used for a large quantity of g711 and g729 transcoding? What is the best alternative for that? -- Woody Dickson woodydickson at gmail.com <woody.dickson at gmail.com> US and Worldwide Termination ============ Contact me for the following offering ============ USA Onnet - 0.0049/min USA Offnet - 0.011/min USA Mobile...

I have multiple voiceage g729 licenses installed on a RH9 box, and have a remote C7960 configured to use it (low bandwidth). In calls like: Remote C7960 -> g729 -> asterisk -> g711 -> C7960 the audio is oftentimes rather choppy. Changing the remote 7960 to use g711 seems to eliminate/reduce the choppyness. Any ideas on what might be behind this?

...s the best codec to use with customers using primarily DSL as internet connectivity? I know that g729 is the king-all, but I want to know what the rest of the professional are using out there. g729 has a cost involved, so does the cost really offset the performance? Or is it better to go with g711 to start off? We plan on using Linksys SPA921 as the primary phone and asterisk open source as the softswitch. Any information you can pass would be appreciated.

Hi All, I want to configure fax with Asterisk and I found that we can do this reliably using G711 codec only. Currently my provider is supporting G729 and G711. During the call initiation the call starts with G729 (1'st priority) and somehow if the receiver is unable to receive call then we are providing the Caller to send a fax, but at that point they are using G729 codec. At this point ho...

...r dialer on machine1 will send to machine2-asterisk (1.4) instructions to dial 400 calls, play the message and hang up. This will be done for about 1 million phones. The asterisk box will communicate via SIP to a voice carrier. the voice carrier will then place the calls on pstn. The codec will be g711. So we will never do any transcoding. I have been calculating the CPU power required to do the calls and in previous posting the usual calculation is about 40MHZ per leg when no transcoding is involved. So if we use the 40MHZ rule, we are talking about 40*400=16000MHZ or 1.6Ghz. Comments? -- --...

...,30,t) exten => 1266,2,Congestion exten => 1266,1,Dial(SIP/[number],30) exten => 1266,2,Congestion (The same results using both of the above dialplans...) The environment... PSTN -> Gateway -> Asterisk -> Phone What I'm seeing works... With the gateway setup to send both G711 and G729, it sends an INVITE which includes both G711 and G729 codecs. Asterisk sends an INVITE to my phone with only G729. The call is made and there's a conversation in G711 with the gateway and G729 with the phone. I assume this means Asterisk is transcoding. What I"m seeing fails......