similar to: Iaxphone - unreachable if qualify yes ?

Announcing a new Windows-based IAX/IAX2 client. Please download it and give it a try. Let me know about any bugs, and any missing features. I have yet to come up with a catchy name for it, so at this point it calls itself IAX Phone. (Suggestions? Non-derogatory suggestions, preferably). Download: http://www.sokol-associates.com/Downloads/IaxPhone.zip Reference & Support Page:

We've bought the G729 codec for lowering SIP bandwidth usage (we're using grandstream phones) and we're quite happy with it up until I tried using IAXPhone 0.2.0 build 116 with my asterisk 1.0.0 installations. Weirdly enough, calls from IAXphone to the GS phone work just fine. So are calls from both phones to voicemail. And from both phones to analog phones connected to FXS ports.

Hi, I'm trying to get Asterisk working on P4 2.8 server behind NAT and Firewall (all ports we're set according to instructions) on DSL line. When pbx connects to Digium demo server( I'm located in Slovenia, Europe), it gets first few words, then silence and then comes back when enumerating dial possibilities ("4 for accounting ...). Same happens from SIP or IAX local extension.

We have many IAXy devices in the field now. In all cases, in iax.conf, we have "qualify=yes", so that using "iax2 show peers", we can see whether or not the device is currently online. In some cases, the IAXy device and/or Asterisk are not communicating their qualification, because "iax2 show peers" shows the device as status UNKNOWN. However, when a user picks up

Is there a softphone for windows that supports IAX? I can't seem to find anything out there...maybe im looking in the wrong places... Jeromy Grimmett VoipEmpire.com jeromy@voipempire.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/e668136a/attachment.htm

Hello, I have just install Asterisk software on my pc (PIII, 2Ghz, 512Mo, Linux 2.4.20 ) connected to a LAN (10 BaseT). I use on the users Pc (Xp windows) a client IAX Phone ( Sokol & Associates\IaxPhone). In my very good surprise all works correctly, but I have a little problem of sound quality: When I use the Voicemail the quality of sound is affected with a noise like "Scrrr"

Just curious, how does Asterisk pack Speex frames in a packet. AFAIK, Linphone just sends raw packets, as specified in the RTP draft. Jean-Marc Le mardi 31 janvier 2006 ? 10:43 -0500, Steve Kann a ?crit : > jonathan blais wrote: > > I'm using Linphone. I tested with Asterisk and Speex only, I created > > a channel with echo and it worked. It seems to have problem when >

Hi, I'm looking for IAXphone 2.0 (from sokol-associates) source code but the site is unavailable did some one can help me please. ?? thanks _____________________________________________ Erwan Desvergnes - ANDIUM - 82/86 rue Ch?teau Gaillard 69100 Villeurbanne Tel. 04 37 43 44 45 / Fax 04 37 43 44 44 E-mail: erwan.desvergnes@andium.fr

Hi, I seem to be having problems with IAX clients based on the iaxClient library. I have been working on my own client (an augmentation to the Call Manager I released last week) and it seems to regularly miss incoming calls entirely. It also occasionally misses the drop signal when the remote end drops a call. Has anybody else seen this kind of behavior? I have tested with my client, with

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

We've bought the G729 codec for lowering SIP bandwidth usage (we're using grandstream phones) and we're quite happy with it up until I tried using IAXPhone 0.2.0 build 116 with my asterisk 1.0.0 installations. Weirdly enough, calls from IAXphone to the GS phone work just fine. So are calls from both phones to voicemail. And from both phones to analog phones connected to FXS ports.

Hi, does anybody know if there's a iax phone running on ubuntu 5.10 which can be used with asterisk? Seems like kiax has got many compiling and libraries problems. TIA Giorgio Incantalupo

Hi all, Thanks to Steven Sokol great work, the IAX2 bug in DIAX is now solved. For the interested people, you can download the new DLL (just the IAX2 version) from the following location: http://www.laser.com/dante/diax/wiax2.zip Replace the wiax2.dll file in the app directory with the new one and this is all. Please test it and send me your feedback. I intend to release a new DIAX version this

Hi, I've spotted weird crash of Asterisk cvs Stable. I have defined queue in queues.conf : [prodaja] music = default announce = queue-markq strategy = ringall context = from-pstn timeout = 15 retry = 5 maxlen = 0 announce-holdtime = no announce-frequency = 30 announce-holdtime = yes monitor-format = gsm|wav|wav49 monitor-join = yes eventwhencalled = yes member => Agent/1000

I was curious if you had ever peeked at a Teamspeak voice packet? I already have large chunks of the protocol torn apart. I believe the voice packets are the last big hurdle. As I know nothing about speex encoding, it is proving difficult to figure out the start of the actual voice data and/or any voice specific state data contained within the packet. If you have any information and are

Hi I'm browsing around the internet looking for signs that the IAX client library and app_meetme support video. I stumbled across this post by SteveK on the 27th of Feb 2006. "My company is looking to hire a full-time developer, who will be working about 25-50% of the time on iaxclient; in particular to finally integrate, build, polish and enhance video in iaxclient, add video

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

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

On 5/3/06, Jean-Marc Valin <Jean-Marc.Valin@usherbrooke.ca> wrote: > > I must say I really like the generalized jitter buffer though :) It's a > > cleaner and more flexible implementation and can more easily be adjusted > > to contain additional information with each packet. This looks interesting to tie into asterisk's jb and plc code as well.