similar to: Error Connecting using libiax

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

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

I have looked through the archives, 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

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

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 *

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'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

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), independent of the play device;

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

Hi everybody, Is there a way to bind client ip address that connects to pop3 with the login name? For example if the client ip address is 192.168.2.40 then his login name is 'test' and he can log in with only that account name. I think it is possible to do it with an external authentication program. Thanks.

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.

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

What is the no authority found problem? And how can I register with * on IAX. It keeps rejecting the request telling that XXX not dynamic host. rejected any idea THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030520/a8a5907d/attachment.htm

Hi Chris, I know this is quite an old email, but I was browsing through the archive :) I am currently working on "embedding" asterisk in one of Allwell's STB's. The idea is more or less exactly like yours. The STB will be solid-state and contain OS, Asterisk, Basic configuration and voice files on a flash disk. It will boot up and get a network share via DHCP. This network

Is there a way to flag a bad or noisy PRI channel in Asterisk so it will be skipped over? Shawn

Speex 1.0 - in file sb_celp.c line 218 change speex_decoder_ctl(...) to speex_encoder_ctl(...): void *sb_encoder_init(SpeexMode *m) { . . . --> speex_decoder_ctl(st->st_low, SPEEX_GET_SAMPLING_RATE, &st->sampling_rate); // Replace <-- speex_encoder_ctl(st->st_low, SPEEX_GET_SAMPLING_RATE, &st->sampling_rate); st->sampling_rate*=2; return st; } --

Hi all, After upgrading my fontconfig to 2.2.0, fc-cache -vf gives me an error: /usr/libexec/ld-elf.so.1: /usr/X11R6/lib/libfontconfig.so.1: Undefined symbol "FT_Get_BDF_Property" when it tries to build the cache for non bitmapped fonts. I have: 1. updated my CVS ports tree 2. rebuilt fontconfig, freetype, expat and Xft using portupgrade (portupgrade -frR ) 3. rebuilt my all

Hi, Recently I am encountering an authentication error when making a phone call between Asterisks. That call is intended as follows. (1) SIP_phone2 to Asterisk#2 (2) Asterisk#2 to Asterisk#3 (3) Asterisk#3 to SIP_Phone3 At (2), that is transferring a call such as -- Executing Dial("SIP/211-6da6", "iax/k0.dyndns.org/302") in new stack -- Calling using options