search for: libiax

...rom CVS and got the following error when using the command line client (similar to miniphone). Text Based Telephony Client. TeleClient> call 192.168.2.40 TeleClient> Unknown event: 15 dump Dumping call! TeleClient> quit Good bye! Does anyone have an idea what I'm missing here? Does libiax need to be updated? Let me know. Thanks. Shawn Lawrence -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030520/d2c6797b/attachment.htm

Hello, from what I've understood Iax2 should support aes128 encryption. I've found this old info: http://www.voip-info.org/wiki/view/IAX+encryption and this (unanswered?) post http://lists.digium.com/pipermail/asterisk-security/2005-August/000060.h tml Is this the libiax used currently on asterisk http://ftp.digium.com/pub/libiax/ ? I would like to understand if someone is using this in production. Moreover which Iax client do you use to test this? Best Regards, Claudio Internet Email Confidentiality Footer -------------------------------------------------------...

...and can only find old references to this problem that appear to be no longer relevant, so I thought I'd ask again. I am having a problem with periodic breaks in audio over an IAX trunk. The interruption only happens in one direction, and (I think) only with clients built on the open source libiax. Codec is irrelevant, and jitterbuffer on/off seems to make no difference either. The pause happens every few seconds, and is regular. If I disable trunking, audio is perfect. I am running CVS HEAD as of 1st March. Can anyone shed any light on this? -- ======================================...

I have committed my first stab at chan_iax2. IAX2 is a work in progress and is designed to fix some of the original design limitations of the first IAX protocol. This runs on a separate port number (4569) so it will not affect any of your existing IAX connections. So far, here is a list of the features for IAX2: * Separate ACK field to reduce the number of ACK messages required to be sent *

Hi, I try to use Libiax in order to put un transfert button in my iax softphone. Is there a way to make a call transfert ? Best regards rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031210/e1e6acf3/attachment.htm

Hi All, I've been lurking here for a while now, having only made a couple of posts. I am starting a new hardphone project and was wondering if there is some GPL'ed IAX source that I could start with. I've searched and haven't come up with much beyond iaxClient. While iaxClient does give me a little bit to start with, it looks like it is really intended to be more of a

...make some calls both across local networks, as well as over the internet. Some more information, including links to pre-compiled binaries, are available at http://iaxclient.sourceforge.net Source code and build directions are available from the sourceforge CVS servers. Iaxclient uses digium's libiax, the portaudio audio abstraction libraries, the gsm codec, and other components. Feedback is, of course, welcome. People interested in making a cross-platform GUI for a more full-featured client would also be welcomed. Features upcoming would include: 1) Audio processing code: Audio level...

Andres sounds as if this is Andres's own development. He mentioned IAX, not IAX2. My guess is that he might have used one of the IAX GPL Libraries and source trees, based on iaxClient and not libiax2. It is possible that Andres is not aware of the GPL terms that he has to adhere to, if he wants to commercialize this product. The Source code for IAX Phone is available from Steven Sokol's Web site and from a few other places. Some links are below. http://www.sokol-associates.com/Body.asp?I...

Hi all, I have developed a full featured Windows IAX phone based on LIBIAX library . It is now in a prerelease version (0.9.0) and you can download it for free from my web page: http://www.laser.com/dante or http://www.geocities.com/tdanro Some of the features are: - registering with Asterisk PBX; - can use any audio device as ring device (including PC speaker), indepen...

Hi to all. I've put on line a cvs viewer for asterisk source code. Is based onto the suite horde+chora. The website is http://asterisk.espia-net.net The cvs modules shown are * asterisk * asterisk-addons * zaptel * zapata * libpri * libr2 * libiax * libiax2 * gnophone * phpconfig * gastman all revisions, branch , comments & whatever cvs is has been preserved. this could be a sort of mirror. I've installed such system, since I prefer to have a quick view of the cvs changes via web. and chora is pretty good for that. Note that no cvs...

Is it possible for two PDA's to communicate like telephones via SIP channels on a PC running Asterisk? If that is possible, does there exist any applications that can be installed on a Zaurus 5600, which is a PDA with an Xscale processor running on a Linux OS, that can essentially turn it into a softphone? Thanks in advance for any input, Daniel

DOes anyone have any good suggestions as to good SIP or IAX clients for linux? I have set up and am currently testing asterisk in a controlled environment. I have gnophone running on one of my boxes but the gnophone site has been down. So I can't seem to fing the IAX and Ix-devel rpms or the gsm and gsm-devel rpms. So that prevents me from setting up gnophone on another box. I am

Anyone else having problems connecting to http://downloads.digium.com/pub/ ?? Jonn

Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT.

...col. You could get some lower end systems > > > that > > > each have a dedicated soundcard and ethernet device, then link to a > > > machine that controls the telephony gateway. A old Pentium should > > be > > > more than powerful enough for a soundcard and libiax. Come to think > > > of > > > it, a PC104 board might even have all the hardware you want and be > > > around a few inches square. > > > > > > Then your P500 should be plenty powerful enough and have enough > > > expansion to get your couple of...

...ling list and the voip-info wiki to no avail. Setup is as follows: Linux 2.6.23 (gentoo) / AMD Athlon(tm) 64 Processor 3000+ / 2GB RAM Sangoma A102 card + E1 (30 channels) Asterisk 1.4.17 (custom compile from source, not using gentoo package or any patches) Wanpipe drivers 3.2.3.0 Iaxmodem + libiax 2-0.2.3-SVN-20071223+ I have tried both kphone and zoiper (linux) clients. On kphone the interface's register result is 'bad password', on zoiper registration continues indefinitely but after the first request it is ignored by asterisk due to being duplicate, after a time it fails sil...