similar to: G729, NAT and Transcoding (all-in-one)

Anyone used the combination above? We are and it sounds like crap. The audio drops out in regular intervals which suggests that someone's g729 isn't doing its job correctly. I'm blameing XTen cause when I make a ulaw call that gets converted to 729 using digium's 729, calls sound fine. Anyone else? Similar experiences? -Matthew --

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

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 -

Folks! We have purchased G729 and have been testing the codec on mUltiple Gateways. Here is what we have found. Here is the config I have used: ------------------------------- Asterisk Server On Dual Pentium Xeons with 6GB of RAM, running on Fedora Core 2 User1 is in USA on Broadband Cable User2 is in India on 64Kbps ISDN Line User1 using SIPURA SPA 2000 user2 using Xten professsional(X-pro)

Just checking the transcoding on our Asterisk boxes and I get the following results. I have the g726, ilbc and lpc10 formats and codecs enabled in 'make menuselect' so I dont understand why its showing as no translation path. Any ideas? I am running certified-asterisk-11.2-cert2 Thanks Gareth > core show translation paths alaw --- Translation paths SRC Codec "alaw"

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

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.

Hello guys, i have a question about a voip gateway we use. I saw those values typing in cli: core show translation g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - - - - - - - - - - - - gsm - - 2001 2001 6000 2001 2000 16000 - 34002 - 6000

Hey gang, I'm trying to setup the ability to dial a SIP user via their email address. I'm using SJPhone as my tester UA, but most clients will be using XTen Pro. I added an SRV DNS entry into our zone, and it returns: ; <<>> DiG 9.2.1 <<>> SRV _sip._udp.cytelcom.com ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status:

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,

All, I have my * box NAT'd with all ports forwarded that are SIP related (based on Wiki). I also have nat=yes, externalip=WAN address of firewall, internalip=LAN network of *. I have my Xten soft phone on a PC which is NAT'd behind firewall with ports forwarded. I have also followed instructions on Wiki for Xten. I can authenticate fine, and sip show peers shows my extension is OK,

I have recently started experimenting with Asterisk. I am running the system the other side of the a NAT router and trying to connect to FWD. I have opened UDP ports and have configured sip.conf to handle NAT. The problem: I can call from the FWD phone and the extension on Asterisk rings and there is two way sound so no problem. Now if in the extension.conf file I have, exten =>

Hi All, We are experiencing a a problem when running calls over IAX with g.729. The call flow is as follows: Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier The first Asterisk system is running 1.2 and the second is running 1.0. When using g726 from the handset all the way thru to Asterisk2(then 729 for the carrier leg) calls go thru fine, but when using g729, there

Hi, I've posted a simular message little over a week ago so sorry for reposting. I need to register to freeworld dial up from behind a nat. Using the xten software sip client works fine but with asterisk I don't know how to do it. Last time I posted I got different responses. Some saying I can't register with an outbound proxy from asterisk others said they have done it. If it is

Since nobody answer my previous question (It looks like g.726 is a bad word). I have this scenario: One box with Asterisk 1.4.0 beta 2 IAX to anothers Asterisk working properly. As an ATA I have only one Grandstream HT496. Two lines on the ATA 727 & 726. >From outside I can call any of those two extensions if: I defined both as ulaw (G.711) One as ulaw and the other as G.729

Guys. Im writing this because Ive checked the wiki, Xten website and read a lot of docs and still cant figure out a way around the NAT issues. Maybe somebody else can give me some ideas from a fresh perpective. My test setup is this: Asterisk -> 2wire homeportal Firewall -> internet Computer with Xten eyebeam The asterisk box and the computer with xten beam are behind the same

Dear Folks, I'm running Asterisk 1.4.31 server, on an Ubuntu 9.10 system. My scenario is simple: connection to the PSTN directly via SIP, using g729 codec, and connection to the softphones (X-lite 3.0 build 56125) trought local network, using ulaw codec. Sometimes, I got messages like: [Jul 1 15:26:16] WARNING[26483]: chan_sip.c:5514 process_sdp: Unsupported SDP media type in offer: image

Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP: ___________ HOME _______________ ____OFFICE ____ SPA2000 <---> Linux Box <--> Asterisk Box 192.168.0.253 192.168.0.1 eth1 200.93.xxx.a 200.93.xxx.b eth0 My problem is when I try to call to any trunk or extention

Yep I did the same. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Francesco Peeters (Asterisk) Sent: Saturday, 21 January 2006 5:34 PM To: fbraeuer@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

I am using Teliax to terminate my calls, and I have 3 licenses' for g729 from Digium. "show translations" verifies that the registration took place. When I place a call, having "allow=g729" as the only allow option in iax.conf, I get the following error: WARNING[361]: chan_iax2.c:6017 socket_read: Call rejected by 208.139.204.228: Unable to negotiate codec If I place a