similar to: DIAX 0.9.10d with Eutectics USB phone suport

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 0.9.9g is available for download (including the updated help file and web page) from the following locations: http://www.laser.com/dante or http://www.geocities.com/tdanro What's new in 0.9.9g (from 0.9.9f): - during a call, accept DTMF tones as monitored events to trigger output commands - call timer on the phone display - Swedish language added - can run a command from the

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

Hi all! Somebody knows something to do with a high delay using Asterisk + DIAX!? When I used IAXComm(Linux) in both sides(peer and me) no problems. Whan I used DIAX099f(WinXP) in both sides(peer and me) I have a delay in the voice coming from the person that I called. I don't have delay in my voice to the peer phone. CODEC: u-law (I tried with all available codecs) Thanks for your help!

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)

Hi all. Is this possible to use an include parameter on zaptel.conf file? I mean, I want to have a bunch of files with zaptel configurations, each one with the configuration of one kind of board(TDM, analog, and so on). Thanks, Denis Galv?o

Hi all. When I place a call Im getting this error: Jul 25 09:50:07 WARNING[3200] channel.c: Exception flag set on 'UniCall/13-1', but no exception handler Lots of this messages appeared on my Asterisk full log and the CPU got 100%. Topology: Analog Phone - Hicom300H E1 - E1 Asterisk - IP Trunk Problem: 1. Calls from Analog Phone through Asterisk is ok, but the messages appeared.

Does anybody have contact info for Dan Toma, the author of Diax? I've tried danto at clicknet.ro and danto at rdslink.ro without success. Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

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:

Hi, I'm having some problems regarding the packages lme4 and nlme, more specifically in the denominator degrees of freedom. I used data Orthodont for the two packages. The commands used are below. require(nlme) data(Orthodont) fm1<-lme(distance~age+ Sex, data=Orthodont,random=~1|Subject, method="REML") anova(fm1) numDF DenDF F-value p-value (Intercept) 1

Hi, There is any other user of DIAX with this problem? Thanks, Dan

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

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

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,

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

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'

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

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

Hi all, The multilingual version of DIAX with both IAX and IAX2 support will be available for download later today or tommorrow. If someone interested to help me traslate from English to one of the following languages (ant not oly) please drop me a direct mail. - italian - spanish - french - portuguese In this moment the following are done: - english - romanian - german - dutch If this can be

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.