similar to: how to receive call on iaxclient

Hello, I'm new to Asterisk and would like to know SIP support status. Are there any testing been done with some widely deployed client (Cisco SIP IP phone, ...)? I was using Vocal but I'm now interested in Asterisk as it seems to offer more features...if it supports SIP. Thanks for your help. Francois.

About 2 weeks ago, I made the first announcement to the list about the first IAXCLIENT cross-platform builds. What we had then, was a command-line application which could make a single outgoing call. We've since done quite a bit of work on the clients, and the client now has: 1) A working cross-platform GUI 2) Volume level controls, and re-tuned audio processing 3) Multiple call apparances

On Thu, 08 Apr 2004 10:14:09 -0400, Steve Kann wrote: >Gary wrote: > >>I have noticed lack of codec negotiation with calls thru a registrated >>asterisk box. >> >>No seen problems with outbound calls, (though I haven't specifically >>tried it), but the problem exists inbound. >> >>Easiest method for testing this was ring in via a sip client set

Asterisk-people, Some of you may have heard that we were working on a simple, cross-platform IAX client library called "iaxclient". We've pretty much been "on vacation" with the project for a while, but recently have made some progress, and now have the library working across platforms, and a simple test client called "testcall", up and running on 3

Hi All, I try to compile the last version of iaxclient with the new build system, i also compiled speex 1.2b2 . When the makefile try to compile the sample code testcall I have the following mistales: gcc -I/d/Lib/iaxclient-2.0.1/portaudio-1.9/src/common -I/d/Lib/iaxclient-2.0.1/speex-1.2/include -o testcall.exe testcall.o ../../lib/.libs/libiaxclient.a

Hi, I had downlaoded iaxclient-2.0.2 and complie project *\iaxclient-2.0.2\contrib\win\vs2005* ** It gives many83 fatal and file missing error of file missing Error 1 fatal error C1083: Cannot open include file: 'portaudio.h': No such file or directory d:\mohit\asterisk\iaxclient-2.0.2\iaxclient-2.0.2\lib\portmixer\px_win_wmme\px_win_wmme.c 40 Error 2 fatal error C1083: Cannot open

David Brazier wrote: > Hi > > The latest SVN trunk for speex has changed the SpeexPreprocessState to > an opaque structure, for jolly good software engineering reasons. > However, the Analogue AGC (AAGC) feature of iaxclient (in audio_enode.c) > relies on some members of this. It uses speech_prob to detect when > there is enough speech to consider AAGC and then loudness2 to

iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X systems. Sources included in the iaxclient library: http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz Precompiled binaries at: http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz Features: * Register with multiple servers (ie enterprise server and iaxtel).

Jean-Marc Valin wrote: >> As you can tell, the AAGC integration with speex was really a classic >> hack. Instead of re-creating the hack, what's probably best here is to >> integrate AAGC back into speex, and have a proper API. >> > > Agreed here. If you can come up with a clean patch to add that feature, > it's something I'd like to see in

Jean-Marc Valin wrote: >> I hate to be a talker and not a do-er, but I won't be able to write this >> myself, probably someone on the iaxclient team could do it. >> > > Anyway, let me know if/when someone's working on that. > > >>> Hmm, or does that mean the analogue AGC is actually completely >>> independent from the "real"

Hello, I can?t compile "iaxclient", because one needs to compile the new version "wiax.dll". I tried to compile it under MinGW/Cygwin, but I had the messages like: cc -I. -Igsm/inc -Iportaudio/pa_common -Iportaudio/pablio -Iportmixer/px_common -Ilibspeex/include -g -O2 -DSPEEX_PREPROCESS=1 -DNEWJB -Ilibiax2/src -IAXC_IAX -DLIBIAX -DSPEEX_EC=1 -DWIN32 -DBUILDING_DLL -c

Hi All; Did anyone used iaxclient? I would like to know how is the voice quality? OS to be used Microsoft. Regards Bilal

Hi all, A new DIAX version 0.9.10f is available for download at one of the followind locations: http://www.laser.com/dante http://www.cosmica.ro/dante http://www.geocities.com/tdanro What's new in 0.9.10f comparing with the old 0.9.10a: - volume and mic levels saved in the config file; - master and wave levels adjusted in the same time for the play device; - better handling registration

Hi all, DIAX 0.9.3 is available for download from the same place: http://www.laser.com/dante or http://www.geocities.com/tdanro The new DLL contain the latest updates made by Steve in the iaxclient library. Still just IAX1 is supported (for the moment). What's new in 0.9.3? - accept blank passwords; - accept for registration/calls host names, not only IP Address; - password no

All, I have a simple question and a complicated reason for asking: Is it possible to change asterisk's source port for outbound IAX2 connections? I've tried using "sourceaddress" to no avail. I can set it to: proper.ip.of.box:4569 or 0.0.0.0:4569 and it works as expected. But if I try to set it to: proper.ip.of.box:5000 or 0.0.0.0:5000 it fails around line 8536 in

> I hate to be a talker and not a do-er, but I won't be able to write this > myself, probably someone on the iaxclient team could do it. Anyway, let me know if/when someone's working on that. >> Hmm, or does that mean the analogue AGC is actually completely >> independent from the "real" AGC. Any thoughts? >> > > It's actually a bit more

Hi The latest SVN trunk for speex has changed the SpeexPreprocessState to an opaque structure, for jolly good software engineering reasons. However, the Analogue AGC (AAGC) feature of iaxclient (in audio_enode.c) relies on some members of this. It uses speech_prob to detect when there is enough speech to consider AAGC and then loudness2 to decide how to adjust the input mixer. We want to use

> As you can tell, the AAGC integration with speex was really a classic > hack. Instead of re-creating the hack, what's probably best here is to > integrate AAGC back into speex, and have a proper API. Agreed here. If you can come up with a clean patch to add that feature, it's something I'd like to see in Speex. > For those of you just tuning in, what I call

Dear all, The new version of DIAX (0.9.10a) is ready to be downloaded. The web site and the help file are updated too. What's new comparing with 0.9.9g: - independent codec configuration for each registration server; - use control chars in the dial string to automatically send some DTMF codes after dialing: - '#' dial separator - 'p' pause 1s (long press on '*' key)

Hello Folks, I noticed that all of the IaxClient based softphones with exception of Firefly only seem to have support for GSM but what about iLBC? The quality is excellent with iLBC even on a dialup connection! Meanwhile while the audio on GSM often sounds scratchy. Is anyone looking to implement iLBC in an IaxClient based softphone soon? Errol Biz4Web Solutions Limited