similar to: Newbie question on soft phones with SIP

I am new to the Telephony world and am trying to get a basic idea of how things work. Can I use Asterisk to connect two soft phones on a LAN to communicate with one another without any additional hardware (besides the sound-card)? If this is possible, I seem to be doing something wrong. I have installed a version of X-Lite and can't seem to get it to ring on my Win PC. Any pointers to docs

Hi, I'm experimenting with the dev kit lite and now past the USB unpleasantness it's working great with standard phones and lines. The priority right now is getting soft phones (under Windows XP) working well. So far, I've only been able to get the XTEN Lite phone working and I really don't understand how I set it up. I used "xten" for every option everywhere (display

After scouring the list archive and not finding the answer I decided to post to the list. I'm sure the answer is glaringly obvious but please bear with me. Using Asterisk, I'm tasked with setting up a SOHO with 5 analogue (FXS?) lines and a number of soft-phones for internal extensions. I'm confused by the telephony hardware needed for this exercise - 1) I need the equivalent of two

Hello! Can anyone recommend a good/handy/nice sip soft phone? I have already done some testing with kphone and gnome meeting (which cant do sip).Can you recommend a open source project? It should mainly be practial and have a address book. I found kphone quite unstable, the address book is designed quite poor, and if you would like to transfer a call with the transfer button you cant access the

hi My stations are behinds a firewall, the system is windows 2000 & 98, i use sjphone asterisk is on the internet gateway where is the firewall Shorewall the system is linux debian (sid) kernel 2.4.20 j do whaton http://www.automated.it/guidetoasterisk.htm (grateful Andy) to write my sip.conf but i can't call an external sip user. (an external user can call me) i try without asterisk with

Hi everyone, I have the following configuration: Calls are coming in into the Asterisk system via an ISDN card (AVM ISDN-Controller C4) When a call is received, it's placed into a hold using the dial-plan: exten => s,1,Wait,2 exten => s,2,Answer exten => s,3,SetMusicOnHold,random exten => s,4,MusicOnHold,random An application is running in the background, using the

Hi, Am 16.02.2017 um 14:19 schrieb Annus Fictus: > And Microsip using PJSIP SIP stack :) Sorry (also, for off-topic), based on my latest experience with PJSIP, I'm not sure if this really is a sign of good quality. Maybe it's a problem with the implementation in Asterisk (I haven't tried PJSIP in other software), but after just five minutes of testing I found several bugs

Microsip (Windows) is free and small. 2.5Mb download, 10Mb RAM usage, does everything I need and configuring TLS is a doddle. http://www.microsip.org/ On 16 February 2017 at 13:04, Max Grobecker <max.grobecker at ml.grobecker.info> wrote: > Hello, > > I'm a big fan of PhonerLite. > It's more poplar in Germany, but also available in English language. > This client

Hi folks, well the easiest soft phone I have found (from this list) is the x-ten. great little piece of software. BUT, on a windows machine I need to plug in a USB device and find the software to drive it which can actually use a normal handset (similar to the s100p) but can be used to actually generate/answer the calls and actually rings when a call is received. Any ideas ?? Gary .

Hello, I have a user that prefers Soft SIP phone install on his laptop, for security reasons I have enable TLS on our Asterisk server to support TLS authentication, It works well with hard phones. Has anybody in this forum use SIP Soft phones with TLS authentication enabled? Any suggestions? Thanks, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, iam a student trying to build an asterisk pbx as a simple configuration only two extention (using Xlite)without outsite telephone line. i already follow the instruction and seem the asterisk work fine because there is no error message. when i configure SIP.conf and extention.conf i hope the phone will ring each other as an extention. but it doesn't work. i follow the instruction at

Hi Everyone After speaking with my local telco service provider, Telkom, I am aware that either we don't have E1 services in South Africa or the technical skills of our the engineers leaves much to be desired. Has/does anyone know if the Digium E1 (E100P) card is certified/works in South Africa? Thanks for the help so far! Tielman /* Tielman Koekemoer Unix and

What I've got... Software: Linux: Slackware 9.1 Asterisk: out of CVS - so its new. isdn4k-utils: to test the ISDN Card Hardware: PII Pentium 400Mhz (Its a test of concept machine) with 320Kb RAM 1 x ISDN BRI Card - DIVA EICON (Installed + working) 2 x Grandstream (Barbie?) BT100 SIP Phones. What Works.. I can call from one phone to the other... get read voicemail... I can

Hi, I'm trying to setup Asterisk on a Linux (redhat 8) machine. There are no analogue phones to be used & use is purely for internet traffic (SIPs). I've followed the setup guide from : http://www.automated.it/guidetoasterisk.htm But cannot get asterisk to run. If I type asterisk at the command prompt I get invalid instruction If I type ./asterisk start I get [FAILED] Can

Good morning all, I'm setting up Asterisk for the first time with no prior PBX experience. I'm following Andy Powell's 'Getting Started with Asterisk' (http://www.automated.it/guidetoasterisk.htm). This is my second time through that document - as I did something weird the first time and really upset it somehow - and I wanted to ask a few general questions of the list.

Does anyone know of a reputable repository where I can obtain a CentOS-6 version of Blink as an rpm/yum package? I have to support three platforms (OSX, MS-Win and CentOS-Linux) with SRTP capable soft-phones. interoperating with an Asterisk-11 PBX; and I would rather have only one client to deal with. -- *** E-Mail is NOT a SECURE channel *** James B. Byrne

Hi, i just installed latest asterisk on fedora rc2 and on the same machine i installed a sip soft phone called kphone. Kphone complains about /dev/dsp being used and can't place/answer calls (/dev/dsp is obviously used by asterisk) . how can "share" my sound card with these two programs? or can i disable the sound card in asterisk so i can use kphone to place/answer calls? BTW kphone

Hello-- I've installed a few X-Lite softphones on windows machines, and am playing with the settings. I've written before about this, and since have discovered that the X-Lite has an option to turn off registration, which I have set, because I bumped into a letter from Mark, saying that Asterisk doesn't do registrations yet. And, without such on the X-Lite, I can still dial out, and

I have a problem with one way audio on Sip and I guess it may be a NAT issue, in the example below 204 is rung by 208 (xlite external) I dial perfectly but when I get to the answering of the Asterisk, I can hear audio from the Asterisk but cannot get audio to the Asterisk, ie If I ring the voice mail , Asterisk answers and then cannot hear my password... I have put the Ports Forward

Hi, before you start throwing stones to me let me tell you that I am a bit new to Linux. I downloaded Asterisk from the cvs server on Wednesday 15 July 2004, as described in Andy Powell's "Getting Started with Asterisk" (http://www.automated.it/guidetoasterisk.htm). Thanks Andy! I read about the Asterisk 1.0 RC1, and I would like to download it and install it. Could someone tell