similar to: I need your feedback related to the DIAX 0.9.9f stability

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

Im getting the following error when I try to start Asterisk WARNING[16384]: chan_zap.c:765 zt_open: Unable to specify channel 1: No such device or address Nov 29 21:40:43 ERROR[16384]: chan_zap.c:6195 mkintf: Unable to open channel 1: No such device or address I'm using a TDM22B card. ============= # Zaptel Configuration File # loadzone=us defaultzone=us fxoks=1-2 fxsks=3-4 ==============

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

Just on the off chance that Canadian Asterisk users might be interested in a place to discuss topics specific to the "great white north" (sources, services, telcos, etc.), I created the asterisk-canada mailing list: http://lists.syonex.com/mailman/listinfo/asterisk-canada or asterisk-canada-subscribe@lists.syonex.com Cheers! John

Hi every one . There are any out there that have a unicall deploy working without problem ? Can give me some tips or referenece about his config ? Regards

iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X (Panther) systems. Recent Changes: * Improved jitterbuffer code * Steve Underwood's Packet Loss Concealment Code Features Include: * iLBC support * GSM support * speex support * ulaw and alaw support * Blind Transfer. * Custom Ringtones per

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!

Hi all, DIAX 0.9.9f 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 You can see there what's new in 0.9.9f and the online help (for the newcomers). Please send me your feedback. The source file for the wiax.dll is available for download too. Best regards, Dan

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

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

Hi, I've just got a TE110P card and installed at Asterisk. I configured zapata.conf, according to www.digium.com/index.php?menu=configuration, but the following error is happening: ... ... ... [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver)

Hello All, Please forgive the lack of understanding as of yet but I have been trying to follow the mailing list messages over the last few days and would like to know if someone could wither point me into the right direction or possibly give me a brief overview of the complete process. Basically, I see that the Asterisk PBX systems can run on linux and seems to offer the engine base that is

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,

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

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

Hi all, With the great help of some of the DIAX users, 2 bugs where corrected and the files from my site updated. 1. Error sending X10 commands on some systems 2. "Run-time error '13', Type mismatch" crash everytime Config -> Audio is selected This seems to be related with Visual Studio /.NET and the new VB runtime libraries. Please download the files again if you have

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

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]

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