similar to: Justvoip linux

=20 Andrew Fenn wrote: > You don't need their program to use justvoip, voipdiscount, etc=2E You > can use any sip client to connect to Betamax servers=2E Try Twinkle=2E >=20 > On Mon, Jul 27, 2009 at 11:24 PM, miroa84<wineforum-user at winehq=2Eorg> wrote: >=20 > > I tried to install justvoip several times and I cannot install it=2E Can somebody tell me how to

Hi i'm trying to install this application by wine installation has been successful but software doesn't properly it doesn't start at all or it's giving me critical error msg

I have been encountering a rather hard to debug problem for the last couple of months: * Calls are setup fine. * After a couple of minutes, two way audio becomes one-way and the remote or local party drops out of the call. Setup: * Nokia E71i sip on NAT'd network (multihomed linux box) * Remote asterisk 1.4.21 on Ubuntu on public network * using a Finera/Betamax provider to route calls to

hi i use wine for runnig an application rynga.i success fully install it. but when i try to run an error message appear that it cant be run because of the deficiency of wine. i am running this programme in windows 7 without any problem. i want to run it in ubuntu11.10.this is a software which help to making phone calls from internet

Thanks Martin. I can only use RHEL6 so upgrading libvirt Version is not a choice. RegardsAbhishek From: Martin Kletzander <mkletzan@redhat.com> To: abhishek jain <abhish_jain@yahoo.com> Cc: "libvirt-users@redhat.com" <libvirt-users@redhat.com> Sent: Wednesday, 15 February 2017 2:33 PM Subject: Re: [libvirt-users] Error in libvirt-GUI On Wed, Feb 15, 2017

Hello there I'd like to use sip users and peers realtime. I think I done all I need to get asterisk works fine in realtime: res_odbc.conf configuration. extconfig.conf sippeers => odbc,asterisk,sipclient sipusers => odbc,asterisk,sipclient sip.conf [general] rtcachefriends=yes The sipclient table as suggest in this article: SIP Realtime, MySQL table structure (

Andrew Fenn wrote: > > If you're trying to just phone people using the voip service then you > can use any sip client to do so. Twinkle and Ekiga are two such linux > applications you might want to try instead of the one provided by > Betamax. yes i now, but i use voipcheap because it allow to connect my home phone with my iterlocutor phone directly without headsets or network

Hi all, Does anyone have any experience porting libvorbis to platforms with small total memory sizes? Specifically, I'm trying to figure out if it's feasible to run Ogg Vorbis decoding on one of the PlayStation 3's SPU's -- secondary processors with only 256kb of local memory. It looks like out of the box the memory footprint of libvorbis is roughly: ~130k code ~60k static

Hi, I am using libvirt --> 0.10.2  and virt-manager --> 0.9.0 version (RHEL 6). Very rare I get following error message dialogue box. (Image is attached for the error dialogue box ) Error saysError polling connection:'qemu+ssh......' Internal error client socket is closed TraceBack (most recent calls)engine.py: 440  conn.tickconnection.py: 1433   self.hostinfo() = self.vmm.getinfo()

The Gnu Telephony site: http://wiki.gnutelephony.org Has a Centos repo: http://dist.gnutelephony.org/RPMS/ But I caught some text stating that this is for Centos 4.2. Is it really? Is there a difference; i.e. would it be safe to install these on Centos 4.4? Really I am after Twinkle, and it seems there is a lot you need to actually get Twinkle installed...

Hi , I had some problems in the past with sip trunks, asterisk-users Digest, Vol 68, Issue 4, message 6, and had a reply (message 9) saying that It could be a dns issue. Well today I had a problem again with sip module and it really seams a dns issue. I have an asterisk, version 1.4.26.1, that has 4 bri access and two sip trunks. I'm having internet access problems and when this happens

-- Executing [0473775006 at intern:1] NoOp("SIP/twinkle-088e6ea8", "conversation to GSM") in new stack -- Executing [0473775006 at intern:2] Dial("SIP/twinkle-088e6ea8", "SIP/3starsnet/0473775006") in new stack -- Called 3starsnet/0473775006 -- Got SIP response 482 "Loop Detected" back from 85.119.188.3 -- Now forwarding

Hello, I have an error while try to install twinkle: # yum install twinkle [...] Resolving Dependencies --> Running transaction check ---> Package twinkle.i386 0:1.2-1.el5.rf set to be updated --> Processing Dependency: libresolv.so.2(GLIBC_PRIVATE) for package: twinkle --> Finished Dependency Resolution Error: Missing Dependency: libresolv.so.2(GLIBC_PRIVATE) is needed by package

It won't be long until big labels attempt to eliminate the digital ins and outs of equipment. The problem they face is most people that currently use a rack system featuring digital interconnects will NEVER revert to an analog only system. I know that I won't! Think back to VHS, BetaMax, SVHS and LaserDisc. VHS has incredible market share because its licensing is open. Betamax was

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

hi wich soft phone do you recomend but i need this feature it must ask for user name and password when it start. i know xline and zoipper but they dont have that i can acomplish this whit twinkle but i need it for Windows :-( any ideas? thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next

Hi, I've successfully setup Asterisk on my local PC and can make call using Twinkle to the server. But, I cannot call to my Asterisk server at Rackspace. I have been trying several things to figure it out, no luck. My PC is behind NAT, so I've set that up in sip.conf (nat=yes). I can ping my Rackspace server so it seems to be Public-static IP. Anyway, I tried with setting externip,

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to test with a friend who has an Asterisk in his office with the Asterisk which I have in my house. Then I have an extension that he is trying to register remotely. Trying with the Twinkle client, I see that it is registered: - --------------------------------------------------------------------------- 400/400

Currently running version 3.6.25 on a SPARC Solaris 10 64 bit server. Due to CVE-2016-2118 need to upgrade to version 4.2.11 / 4.3.8 / 4.4.2 No Solaris package available. Configure script fails with “Couldn't determine size of 'bool'” Is it possible to install these versions on Solaris 10 and if so how? Many Thanks Steve. This Email and any attachments contains confidential