similar to: OT: Looking for Dan Toma, author of Diax

Displaying 20 results from an estimated 5000 matches similar to: "OT: Looking for Dan Toma, author of Diax"

2005 Jan 17
0
DIAX 0.9.9g more features and higher stabili ty
I have had the same problem when calling across Asterisk from Diax to a SIP phone. If Asterisk "Answers" the call before the "Dial" to the SIP phone there is no delay. Otherwise there is a 10-20 second delay in the Voice path! Peter -----Original Message----- From: Dan [mailto:danto@rdslink.ro] Sent: 14 January 2005 15:57 To: Denis Galv?o - iSolve; Asterisk Users Mailing
2003 Dec 07
0
Diconnectiong after 15s when calling DIAX to DIAX (Tony?)
Hi, There is someone (Tony?) with disconnection problems (after about 15s) when calling between two DIAX phones? I have a voicemessage regarding this issue, without any contact address. If yes, please send me more details about configuration (iax.conf and extensions.conf files, IAX mode, etc.). As another DIAX user requested that, I'll put on my site some sample configurations files to be
2006 Jan 18
0
Problem with DIAX and Asterisk and Vonage
Hi All, I have installed Asterisk and able to create Users and get them connected to Asterisk after authentication. My question is how can I make calls to different DIAX clients through my Asterisk server. I also have vonage softphone account, using that I tried calling 18882255322 -- Registered 'manoj' (AUTHENTICATED) at 59.93.73.0:4569 -- Registered 'diax'
2004 Dec 30
0
New Diax version 0.9.9f
Hi all, Diax version 0.9.9f is ready to be tested by the interested people. You can download it for the moment from the following location only: http://www.geocities.com/tdanro/diax/diax099f.zip Please do not use older config files with 0.9.9f !!! You have some command line options now for diax.exe: /d - start with debug mode enabled /u - start with ATCOM USB phone support enabled
2009 Aug 29
0
asterisk-users Digest, Vol 61, Issue 84
On Sat, Aug 29, 2009 at 10:30 PM, <asterisk-users-request at lists.digium.com>wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help'
2003 Dec 17
0
DIAX 0.9.6d with IAX2 debug support
Hi all, A new version of DIAX (0.9.6.d) can be downloaded from the following address: http://www.laser.com/dante/diax/diax096d.zip or http://www.geocities.com/tdanro/diax/diax096d.zip Take care that the link is not available from the web page. It has debug in a file support for IAX2 only. In order to activate debugging functionality, right click on the first box from the third row (in the right
2005 Jun 22
0
DIAX 0.9.15a with GSM gateway functionality
Hi all, A new version of DIAX is available for download: 0.9.15a. For the moment you can find it only at the following location: http://www.cosmica.ro/dante and http://www.geocities.com/tdanro Whats new in this version comparing with 0.9.10f (the latest official version): - GSM/PSTN Gateway functionality; - starts with XP Styles even the first time; - can connect to the asterisk server to a
2004 Jan 21
2
Diax IAX2
I've downloaded diax-0.9.6b and configured for IAX2. Calls from Diax to * are perfect. However, when calling from * to Diax, I get the following: channel.c:1097 ast_read: Dropping incompatible voice frame on IAX2[mike]/3 of format GSM since our native format has changed to ULAW In iax.conf I have: allow=all disallow=g723.1 disallow=lpc10 allow gsm Has anyone else seen this? Thanks,
2003 Dec 19
3
DIAX phone busy
I've configured the DIAX phone. It registers with the * server, and I am able to make calls from DIAX. However, when I try to call the DIAX phone from another phone, I get a busy signal. My extensions.conf: exten => 70,1,Dial(IAX/mike/mike,30,tr) # IAX Mike exten => 70,2,Voicemail(u70) exten => 70,102,Voicemail(b70) and my iax.conf: [mike] type=friend username=mike host=dynamic
2003 Nov 18
1
DIAX - Can place a call, but can't be called?!
Greetings, DIAX seems to work well placing calls, but I can't actually receive a call . Here, DIAX (x305) "registers", then I use a sip phone to place a call to DIAX (which definitely is not in use by me at debug time, but it is idle on my desktop.I think), and then * goes to vmail. Here's the debug output: affinity*CLI> iax debug IAX Debugging Enabled Rx-Frame Retry[N/A]
2011 Sep 02
0
No subject
is typing his number, though there is a 15 seconds timeout, and even if = I type the number very fast it still may happen to me.<o:p></o:p></p><p = class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><p class=3DMsoNormal>It has = been my casual observation that the speed at which I enter digits on my = phone is unrelated to the speed at which my
2004 Jan 19
6
IAX2 bug in DIAX solved - Great Thanks to Steven!
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
2005 Feb 08
0
DIAX version 0.9.10a available for download
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)
2003 Nov 21
2
DIAX, IAX2 and latency
Hello, today I tried a DIAX -> * -> DIAX connection over the internet (768/128 ADSL connection on both sides). The sound quality was great. However, we had some latency problems, and also, if both sides where not talking the first words had some problems getting thru. Is this expected, is there anything that can be done on our setup, any magical iax.conf entry? Thanks and best regards
2003 Dec 16
2
DIAX-SJPHONE REGISTRATION PROBLEM
I am having a problem with softphone registration, having read the list and watched it for a while for similar problems I just cant seem to figure out the problem. Using SJPHONE or DIAX , I can make outgoing calls but I can't get them to register with asterisk, I have other sip devices registering OK-7940's. From the list and the digium web site this seems to be a straight forward set up
2004 Jul 12
1
Can I hear voice messages from diax phone button directly ?
Hi, I'm testind Diax. I have flashing note about 1 new voice message. Can I hear it somehow from Diax gui, or must I call pbx to get message ? Thanks, Robert.
2005 Jan 10
1
I need your feedback related to the DIAX 0.9.9f stability
Hi all, I kindly ask DIAX users to send me a feedback related to the stability of the new version (0.9.9f), comparing with the older versions (especially 0.9.8). I ask this because I have DIAX runing for one week now without any crash. It is used mainly to control some X10 devices through a regular phone. Thank you and best regards, Dan
2003 Nov 19
3
inter diax connection
hi i am trying DIAX and *. i cannot make calls from ne DIAX to another. whats the config? exten=>44,1,Dial(IAX/username,20,tr) ??? just a guess. any help will be appreciated. cm ===== Designs __________________________________ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree
2004 Jun 08
8
New version of DIAX (0.9.8a) available now for free download
Hi all, A new version of DIAX (0.9.8a) is ready to be downloaded from the following locations: http://www.laser.com/dante or http://www.geocities.com/tdanro What's new in 0.9.8a: - unconditional autoanswer or based on CallerID (user configurable); - use any Ericsson/SonyEricsson GSM/PCS to control DIAX (feedback on the phone display) through Bluetooth (or serial cable). You do not even
2004 Dec 27
3
Diax echo problem
Good morning, I just got asterisk up and going and set it up to communicate to test calling out to a PSTN phone number. My configuration is as follows: My Diax client --> IAX2 on my asterisk server --> IAX2 (voice pulse server) --> PSTN -->my PSTN phone. I don't have any ZAP hardware. I'm only communicating using IP. For timing, I installed the ztdummy module successfully