similar to: Echo after upgrade * 1.05 -> 1.06

Hi list! Again Wengo has made changes to their servers that require modifications to * configs. Is there anyone that has the 'new' wengo working with asterisk that could post their configs? Also they switched codecs, now G720a is required to connect. I can only find an (open) G729 codec, is this the same as G729a? Thanks!

Asterisk must have a reasonably large community here in France judging from the number of people who came out to meet Mark. Either that or we were ALL there :) Something I've been waiting for, a voIP carrier on the models we are used to (low monthly or pay as you go, web account) has just set up their first beta test for 1 euro for the first month, 6euros if you decide to keep it. The basic

Hello there! I need some help to configure two Asterix boxes to route calls using SIP. I followed the instructions present at this site: "http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html", but I couldn't get it working so far. The only difference, besides the names that I've used, is that I'm using realtime to retrieve

Dear friends, Asterisk's voicemail functions work fine for me, but I am having difficulty storing the voice messages inside mysql. My real-time CDR recording works so I assume the odbc connection is fine. The voicemail.conf I have is : [general] format = wav attach = yes dbuser=root dbpass=sqlpass dbhost=localhost dbname=asterisk odbcstorage=asterisk odbctable=voicemessages Asterisk shows

Hi, I've been using nufone DIDs for months with no problem. Now there are codec problems that prevent any kind of calls working. For example, May 5 13:04:12 WARNING[928]: channel.c:2115 ast_channel_make_compatible: No path to translate from IAX2/NuFone@NuFone/25(256) to SIP/wengo-out-968a(4) May 5 13:04:12 WARNING[928]: app_dial.c:1006 dial_exec: Had to drop call because I couldn't

Hi, I'm searching for GPLed softphones. I found WengoPhone but actually not available for Asterisk PBX, only for Wengo network. I found Kiax but only for IAX protocol. Did you know a good GPLed softphones which works on Windows ? Thanks Greg -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear all, I have Asterisk 1.4 and I'm using the voicemail feature. I edit /etc/asterisk/voicemail.conf with "envelope=yes" and after that I left a message in a given mailbox near 11:00 AM. When a dial the voicemail number in order to hear the message, the Astreisk server close the cal and I get this error from te CLI: [Apr 10 14:09:08] WARNING[12955]: file.c:563 ast_openstream_full:

Hi All, I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code on CentOS7. -- https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject The installation is OK. But the connected SIP cilents (both Linphone on Windows7) cannot communicate. I hope your comment such as the testing for resolving the problem. My status is the following(1 and 2). Why 'Everyone

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:

Hi; It is my first time installing an asterisk PBX system . I do have a TDM400 wildcard with 4 FXO moduls on a PC with 3.0GHZ HT CPU and INTEL 915 moatherboard . Fedora C2 Linux as O.S. and I have the latest CVS astreisk , Zaptel and Libpri downloaded the zaptel drivers installation and configuration seems to be fine and the libpri but when I tried to compile and install the asterisk software

Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server connected to Twinkle and X-Lite clients. I have to use the GSM codec for all of my clients, and it was set up in the sip.conf specifically in "allow=gsm" line. Twinkle has GSM codec built in, but when I open X-Lite audio codecs settings I can't see the GSM codec, being that the official web site and the PDF

Hi I use Asterisk with some SIP phone (grandstrea), while with my notebook when I'm out of home/office I use X-lite and all work. Some days ago I try to install wengophone and I decided that I want replace X-lite for use wengophone as client for my Asterisk. One of the reasons is that wengophone support g729 codec. I make some test and I see that is possible to configure other sip server

Hi I am a newbie to * and I am having a problem which appears strange as I did not find any mention of it anywhere in my search. Simply speaking, I have an external SIP proxy server which I am trying to configure for incoming and outgoing calls from my asterisk installation. So here is my configuration in sip.conf [general] register =>

Hello, I am testing speex 1.1.6's echo canceller. I am using testecho.c, with a few modifications to get it to run on Windows. My problem is that I am unable to get the echo cancellation to work correctly. I am working on an audio conferencing software, and one issue we have is sometimes the microphone picks up what is being played through the headset, resulting in an echo of the other

Would just like to warn everybody for Wengo.fr Once you sign up there is no possibility to remove your credit card and even though you send them resignation letters they keep charging your credit card. Now I understand what they mean when they say `unlimited subscription'.

Hi list! I'm using phones that emulate a Cisco 7940 with chan_sccp. When I was using Asterisk 1.0.5 (bristuffed) I never had any such message on the console. The phones do work. Is this a bug in chan_sccp or a feature of asterisk 1.0.6? Thx!

Hi All, Currently I am investigating how interesting OGG is for me. The most important requirements are: - Decoding on embedded hardware, ARM7 or ARM9. - Mixing multiple audio streams/files on embedded hardware First of all how difficult is it to decode OGG on an ARM7/9 processor? I read that OGG decoding needs floating point support? Secondly I would like to know how many MIPS are needed to

Here's the problem: > 2) Encoding with rate control as in single pass "Bitrate > control" will not > lead to better quality than fixed quant (with the right value > of the fixed > quant). Ratecontrol doens't know anything about "quality". It > will try to > reach more-or-less CBR. > > But somehow this is not a fair comparison, because

Here is the output I get from nmake. I am not a guru at getting C programs to build under windows, I never had any troubles I could not solve on OSX :-( I am using the released sources, I have Visual Studio 2005, I have the make files from lemur. I read ALL the readme.txt file before starting. And this is what I get. C:\Documents and Settings\jroberson\My Documents\DOWNLOADS\xapian-

hello list, i need your help please regarding send mail i use astreisk 1.4; i try to send mail when no response like below exten => 5xx,1,Dial(SIP/223, 10) exten => 5xx,n,system(echo test ${DNIS} Email| mail -s 'Call failed' myadresseemail at gmail.com) when i launch the CLI i found : You have new mail in /var/spool/mail/root i check the root and i found : Return-Path: