similar to: OT, I'm looking for SIP/register-enabled softphone

hello there... our telecom sold us VoIP-numbering, managed by Nortel MCS I successfully registered Ekiga to it ( http://sol.chel.skbkontur.ru/ekiga.png) What exactly do I have to write in sip.conf to make Asterisk register on this SIP ? Cheers, Kate -------------- next part -------------- An HTML attachment was scrubbed... URL:

I used to use IDEFISK, but since it was taken over/renamed into Zoiper it's been really hard work - now I'm told that they won't support my chosen distribution - Debian Etch - the current stable version of Debian I prefer. So, looking to dump Zoiper and go with something else - I want something light-weigh (So that rules out Ekiga - and Zoiper was going down the bloatware route

Dear Sirs, out asterisk server has multiple network cards. I want some outgoing calls (from several extensions) to use one IP address, and others to go through another address. is there a way to achive that using asterisk ? Cheers, Kate -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm curious if anyone knows of any possibility to use video VOIP client (like Ekiga or Linphone or...) under Linux that could be operated by touchscreen friendly GUI (bigger buttons, large keypad, etc...) ? I like Ekiga, but GUI is small and cannot be operated via touchscreen... But maybe there are some skins for existing clients that are more touchscreen friendly ? Thanks in

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:

Dear Sirs, we ... 1) buy minutes from other providers 2) sell minutes to out clients some calls terminate to our equipment, others - to h323 proxies. we want calls to be routed according to costs (a route is chosen from many by lowest cost). at the end of it, we'd like to bill our clients and see how much have we earned (money we receive from client on one side, money we pay to proxies on

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to connect to ekiga.net through a client connected to my Asterisk server. For it I am being based on this [1] document. Next I put the configurations that I am using. /etc/asterisk/sip.conf: ; Outgoing to ekiga.net [ekiga] type=friend username=MyUser secret=MyPass host=ekiga.net canreinvite=no qualify=300 nat = yes stunaddr =

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I have ekiga registering to a voip provider (skypho) and receiving external call through the stun server. I want to redirect inconditionally all these calls to my asterisk server, but I can't understand how and what should I configure in asterisk in order to accept the redirected call. In asterisk console I can't see nothing when ekiga passes the call. If I turn asterisk's sip

Dear Sirs, let me repost my question again, probably the last one was lost in a huge amount of messages during weekend. I'm actually looking for web-based tool which can do two level of administration: 1) high level, Administrators, can create "domains" 2) lower level, Users, can manage extensions within certain domain. much like asterisk2billing. so, I want users to manage

I am presently configuring a test Asterisk 11 server based on CentOS-6 and I need to employ a softphone for testing. The base repo has ekiga. The EPEL repo has twinkle. I lack the knowledge of whether other packages exist or might be better suited. Which of the two do you recommend? Or, alternatively, what other package might be a better choice? -- *** E-Mail is NOT a SECURE

I tried numerous but most of them just don't work. They fail on the quality of the microphone. I tried two that support alsa. Kphone does, but dtmf (the numbers) are not recognized by Asterisk :-( I tried X-lite, but that has really terrible voice quality (mic). I tried Sflphone (has alsa support) and that worked on FC4 (not on FC3) and it does not work on Centos44 either: [root at raaf

Hello, I get some problem installing asterisk + ekiga on my debian etch: ii asterisk 1.2.13~dfsg-2 Open Source Private Branch Exchange (PBX) ii ekiga 2.0.3-4 H.323 and SIP compatible VOIP client $: asterisk -U asterisk -vgc give me some WARNING like : ,---- | WARNING[21806]: res_musiconhold.c:852 moh_register: Unable to open | pseudo channel for timing... Sound

Dear Sirs, I'm looking for a tool which can do the following: 1) higher level of administration, only one person, it can create "domain"s and per-domain administration accounts 2) lower level of administration, many persons, each can add new extensions and change passwords with their domains. somewhat similar to asterisk2billing, but with privilege separated into

Hi, I am on amd64 Linux and not really too happy with twinkle, linphone and ekiga. Unfortunately, X-Lite and Zoiper, even though they provide Linux versions (w00t!) have only x86 versions for download. Do you guys know of amd64 versions of those, or can you recommend other softphones that will run on amd64, or which come with source code? Thanks, -- martin | http://madduck.net/ |

Hi, I am using Twinkle to call mobile phone but there is too much noise on the mobile user's end. Mobile user's voice is echoed back to user. While on twinkle end everything is fine. Using Asterisk 11. Please suggest some way to mitigate the problem. Thanks. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in

This Friday's edition of the weekly VoIP Users Conference call is all about wideband audio (aka HD Voice) and conferencing. The guest for this call is David Frankel, CEO of ZipDX a commercial service that specializes in wideband conferencing. We expect an interesting call touching on many aspects of VoIP going beyond the traditional phone service, conference bridges, technical standards,

When I was looking for a Java FastAGI interface for Asterisk I came across asterisk-java first and didn't realize there was more than one out there. It seems to work fine and I've got my first project working with it, but I was wondering which Java FastAGI implementation is the most popular and how they compare against each other. So I'm aware of: asterisk-java JastAGI

On 02/01/16 14:20, Yamaban wrote: > On Mon, 1 Feb 2016 19:22, H <agents at ...> wrote: > >> I have installed the kate editor on Centos 6.7 but it seems to be a >> very old version, 3.3.4, installed as part of kdesdk. On Centos 7 I >> can simply run 'yum install kate' but, alas, not on Centos 6. >> >> What is the recommended way of updating kate on

I have several hardware and software phones connected to Asterisk 1.2.1 from Debian via SIP or IAX2 and I have defined call transfer codes in features.conf. Everything works with the only exception: When I call a _SIP_ _software_ phone (namely Ekiga or Kphone), I can't transfer the call from the _callee_ via the configured DTMF codes. It seems Asterisk completely ignores the sent DTMF codes