similar to: iax2 sound problem

Has anyone on here used kiax before? I am asking because I have it installed on several computers and have been able to get it to connect and register to my Asterisk box. I can even call between them and my SIP softphones. The problem I am having is this: when I use kiax to call someone else, they get some kind of background music playing while I am talking to them. I have called from kiax to

Hi all, Can some one suggest me an IAX client for Linux and Windows? I used KIAX once, but know it seems complicated to have it working on Ubuntu. Thanks. Ronaldo.

Hello I need to hook up someone's remote PC onto our Asterisk server over the Net. There are firewalls on each side, so I figured it's time to give IAX a try, and see if it's less of a pain to use than SIP. And since IAX hardphones are pretty are, I guess I'll go softphone. Apparently, the two most well-known IAX and SIP clients for Windows are ZoIPer and X-Lite, respectively.

Hi List; I need to configure a softphone to be client and use it with Asterisk, which is the recommended one? Is it iax2? Regards Bilal ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.

Hey, all. I'm looking -- if possible -- for a decent, multi-platform soft-phone. Specifically, Linux and Windows; that way, I'll go through the same issues my end users do. I've noticed a couple (e.g., minisip, which seems abandoned, and sip-communicator, which, honestly, is probably a great IM client, but has a confusing interface for actual phone calls). So I'm wondering if

Hi, I'm searching for GPLed softphones. I found WengoPhone but actually not available for Asterisk PBX, only for Wengo network. I found Kiax but only for IAX protocol. Did you know a good GPLed softphones which works on Windows ? Thanks Greg -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hi all, I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension on the other * I get a "Failed to

I am investigating an issue with IAX2 extensions in Asterisk 1.8.x. My application connects to Asterisk via AMI and attempts to run an Originate command between an extension (such as SIP/5555 or IAX2/8888) and an application (in my case it is AgentLogin). This works correctly for SIP extensions, in all Asterisk versions. With IAX2 extensions, this worked correctly in Asterisk 1.6.2.20, but

Hi, does anybody know if there's a iax phone running on ubuntu 5.10 which can be used with asterisk? Seems like kiax has got many compiling and libraries problems. TIA Giorgio Incantalupo

A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having a conversation. Call quality is reported as good except for an echo with a 3 second delay. Most of my searches are saying echo happens only on the PSTN piece, but there isn't one here. Can someone point me in the right direction? Asterisk 1.4.21.2 Under 40 users Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM

Using Wireshark I can see that ZoIPer always send a 160 byte payload. First few payloads contain 20 bytes of data (what I believe to be a mode 3 frame, ie first byte in 0x18 - 0x1F range), followed by 0x7B (21st byte), ie. 5 bit 0x0F terminator padded with 011. ... and then zeroes all the way up to 160 bytes. ... but then after a few payloads there are more following these 21 bytes ...

I just installed an Asterisknow server can someone suggest a software to be used for a PC - PC voice comunication to test in easy way the functionalities of my server. Thanks in advance for the help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091027/bb5e7b1a/attachment.htm

Problem 1 : Incoming conversations from the SIP-provider come into the [default]-context and to the 's'-extension. I am unable to change this, even if I have : sip.conf [general] ;context=default ; Default context for incoming calls register => 092779077:XXXX at 85.119.188.3 ; incoming [092779077] type=user host=85.119.188.3 context=from3starsnet So I define no

We have zoiper connected over iax to asterisk in Sydney. The call is to asterisk in New York. The caller in NZ can hear clearly. Nothing in NY. Here's the sydney server: -- Accepting AUTHENTICATED call from <zoiperipaddr>: > requested format = speex, > requested prefs = (), > actual format = ulaw, > host prefs = (silk16|ulaw|gsm|g722),

Using zoiper on a nexus 4, asterisk 11.5.1, sometimes we see failed authentication. The secret seems correct, so we can't figure out why we're getting failed authentication. But at the same time the device shows as registered: [Oct 8 18:14:14] NOTICE[510]: chan_iax2.c:11071 socket_process_helper: Peer 'n4' is now REACHABLE! Time: 441 [Oct 8 18:15:58] NOTICE[519]:

Hello mailinglist, I want to connect Asterisk with OpenBTS and make a call with a mobile phone. I use: Ubuntu 11.10 + Kernel 3.0.22 GnuRadio 3.3.0 Asterisk 1.8.13 OpenBTS 2.8 Nokia Mobile Phone OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also

After switching to Asterisk 1.2.14 from 1.0.x, I encountered a very strange problem. There is no sound with Playback() or Background() commands. Even though, Asterisk console shows the file is being played when I call the extension (i.e. echo test), I can't hear anything. My echo test extension looks like this: exten => 600,1,Answer exten => 600,2,Playback(demo-echotest) exten

Hi I have a cisco 7960 Phone that connects to my Asterisk server without a problem. But when I call the echotest it just hangs up, echotests from other VoIP providers works just fine. I have tried a softphone and it works just fine. The error I get when the 7960 calls is this: -- Executing Playback("SIP/2000-180c", "demo-echotest") in new stack -- Playing

Hi Is it possible to use only the speex echo cancellation module w/o using the speex codec? Here's the scenario: 1. I have my voice recorded in PCM audio file format 2. I want to cleanup the recorded voice by removing any echo included in the audio file/buffer 3. can I just use the ff APIs? - SpeexEchoState* speex_echo_state_init() - int speex_echo_ctl() - void speex_echo_capture()