similar to: Problem with DIAX and Asterisk and Vonage

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 =

Does anyone currently use Vonage with Asterisk? I've tried to set it up but it looks like Asterisk (at least the version that I have) does not handle well the SIP call dialog, sending a BYE with the wrong tag. As a result, when I hang up, Vonage sends back a 400 Bad Request and the call on the PSTN side does not hang up. I know that Vonage does a lot of nasty stuff which impacts UA's

Hello everyone. I've seen postings for connecting asterisk to vonage but I'm still having trouble achieving that. I have a vonage softphone and I'm trying to register to vonage using asterisk. I have not had any luck. I am behind a firewall. I've successfully gotten xlite to connect and work from the same network. When I change the port setting in [general] to 5061, I am able to

My previous post hasn't even made it to the list yet (am I being moderated?), but I got Vonage's Softphone service working for inbound and outbound calls. Keep in mind that there's currently no perceived limit on simultaneous inbound calls, which makes this a wonderful solution for Asterisk (at least for my use). Below is a sanitized snippet from my working sip.conf; your mileage may

Could anyone help me set up Asterisk in such a way that it makes SIP-SIP transfers using the REFER / NOTIFY method according to RFC-3515 ? SCANARIO: - Asterisk registers with PSTN<->SIP VoIP provider "V" (Vonage) as a friend - Asterisk is located in Europe, Vonage in located US. - Asterisk acts as an autoattendant located in Europe. - Asterisk answers and incoming call from

Hi, I'm trying to get Asterisk to connect to Vonage (softphone acct) to allow me to place and receive calls. I have successfully configured Asterisk to route inbound calls and send them to the correct extension, but I can't get outbound calls to work. I have Asterisk successfully registering with Vonage, but when an INVITE is sent out, I get a "404 Not Found" back from Vonage

Hi All, After few problems I have installed Asterisk and changed my iax.conf. I have defined a user in iax.conf and when I try to connect that user from DIAX phone I get the following error Jan 17 23:48:16 NOTICE[16448]: chan_iax2.c:3910 register_verify: No registration for peer 'manoj' (from 59.93.66.12) In DIAX phone I gave the below to connect username = manoj password=manoj

Hi Nitesh, check out my config that I have for the Faktortel config in the asterisk@home sourceforge forum, you'll probably be able to work out how to set it up from there. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Wednesday, February 23, 2005 4:12 PM To:

I've yet to successfully register and receive calls from my Vonage softphone. I've tried what few examples are given in former post this list and some other forums, but nobody seems to be stepping forward saying it works recently. Either they broke something to where you simply cannot use *, or the config examples need updated. If anyone can show a still working config,

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, 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)

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