similar to: Problem with Asterisk and DIAX, Please help me

Hi All, I am a newbie to VOIP and after some problems I was able to install Asterisk. If I start Asterisk I could find "Asterisk Ready" at the end and I am thinking that Asterisk is started successfully. Later after changing my Extensions.conf and ser.conf nothing works, I could still see the message "Asterisk Ready" but when I try using DIAX and connect to Asterisk nothing

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'

mkumar@mantragroup.com wrote: > Hi All, > > I am a newbie and trying to install Asterisk from instructions given > in http://www.voip-info.org/tiki-index.php?page=Asterisk+RPM. We have > Centos 3.3 so > I downloaded rpm's from > ftp://ftp.linuxsys.com/pub/LSE/packages/CentOS-3.4/asterisk-1.0.9/ and > tried installing one by one but I get the following errors

Hi All, I installed Asterisk recently and it was working from 2 weeks without a problem until today. Today it started showing strange error Feb 28 03:14:08 WARNING[31430]: chan_sip.c:4826 check_auth: Stale nonce received from '<sip:18006733555@mantragroup.com>' Whatever number I call it displays this, please tell how can I fix this? I have no idea what is happening and the cause

Hi All, I am a newbie to Asterisk and I was able to install Asterisk and call out. Recently I purchased two DID's, can someone please tell me or point to some links showing how to configure these DID's for SIP based softphones like Express talk? Thanks, Manoj.

Hi All, I was able to install Asterisk and make outgoing calls. Recently I purchased two DID's and I am facing a problem configuring them to my Asterisk, I hope with the help I get from this list I will be able to configure successfully. Mu errors are Feb 28 08:31:58 NOTICE[19133]: pbx.c:1331 pbx_extension_helper: Cannot find extension context 'context_mantra2' Feb 28 08:31:58

Hello *'s, First Of all Marry Christmas, I want to setup asterisk at large means "my main asterisk server placed in my office(in Pakistan), and some offices outside Pakistan and i want to connect these locations to my main * server (in Pakistan) on remote locations i'll used asterisk can i do this or may be i changed my plans kindly guides me. Thanks In Advance. Adnan Ahmed.

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

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

Hi All, I am a newbie and trying to install Asterisk from instructions given in http://www.voip-info.org/tiki-index.php?page=Asterisk+RPM. We have Centos 3.3 so I downloaded rpm's from ftp://ftp.linuxsys.com/pub/LSE/packages/CentOS-3.4/asterisk-1.0.9/ and tried installing one by one but I get the following errors error: Failed dependencies: asterisk = v1.0.9 is needed by

Hi All, I tried with different configurations and referred many articles to configure Asterisk with a Vonage account I have but all my attempts failed. I am a newbie and hope this mailing list will help fixing my problem and configure Asterisk. The error I get after I make a call to outside number like 18007633555 is -- Accepting AUTHENTICATED call from 59.93.69.218, requested format =

Hi, I am very closed to implement the IAX2 version in DIAX, but still some issues which I don't know how to handle, maybe someone from this list can help me. Trying to register with the * server as in version 1, I get the following in the * console: NOTICE[1150495040]: File chan_iax2.c, Line 2919 (register_verify): Inappropriate authentication received and in the client: Registration

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

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, 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

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

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]

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

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)