similar to: X-Lite from Home

Hello People, I am a newbie asterisk and happy user, i have configured a x100p card and everything works nice, i can forward incoming connections to a x-lite software client and works out of the box, However when i try to make a connection between two x-lite clients then no audio is transmited, i have followed the instructions on voip-info.org, the tutorials on onlamp and i have read some

In an attempt to eliminate audio echo I upgraded one side of a working x-lite to x-lite connection to eyebeam. No joy, and what was worse is the audio was even worse now - just noise. Ok, I upgraded the other side to eyebeam and same thing. I'm not even using video (will enable it in sip.conf later, one change at a time). The connection looks something like: eyebeam client

Hi I have the following setup to make outgoing calls: X-Lite (build 34025) at home behind NAT -> Internet -> Asterisk at work behind NAT -> Internet -> VoIP provider -> GSM gateway -> cellphone. I just tried calling my own cellphone, but there is no sound either way. Here's what I did on the X-Lite at home in the Topology section: IP address : Discover global address

Hello, I have a one-way audio problem. If any one can give me a clue on how to solve it, I'd highly appreciate. My configuration is: Both Asterisk server and a SIP phone run within a LAN. Asterisk: CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build stamp 14262. The Linux box that running Asterisk server is RedHat 2.4.18-14. Asterisk server runs on IP: 192.168.1.102. X-Lite

Hello, I have installed sacesfully Asterisk. I connected to that two ATA'a 186. I can call between them wihout a problem. Everything works fine. I' ve installed X-Lite on couple PC's and registered with Asterisk SIP proxy. Problem is now that I cannot call between ATA 186 and X-Lite. I can call from X-Lite to X-Lite and also ATA to ATA. If i try to call X-Lite to ATA or ATA to X-Lite,

Try the snom soft phone! http://snom.com CS > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Dave Chase > Sent: Saturday, February 26, 2005 12:31 PM > To: ich@mateo.ch; Asterisk Users Mailing List - > Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Anybody using

I'm new to Asterisk and have installed it on OS X. I am trying to use an X-Lite softphone to connect through Asterisk to FreeWorld Dialup and am not having any luck. I have looked at several settings pages at voip-info.com but have had no luck. I am on a school network which I assume has some security but the particulars are unknown as they are not very helpful. It is necessary to

Mateo, Dialing the extension to your softphone is the same as any hardware extension. Exten => 1000,1,Dial,(SIP/1000,20,trf) pretty exten => 1000,2,Macro(vmessage,1000) exten => 1000,3,Hangup Change [mateo01] to [1000] in your sip and you will be saying that ext. 1000 is registered with the specifics you are using. Update the settings in your softphone to register the name and

Dear All, I installed an Asterisk on a linux PC, and X-Lite on two Windows PCs, all in a LAN. But, when I make phone call from one X-Lite to another, I always get Call Failed: 404 not found. Here is my sip.conf: [Phone1] type=friend host=dynamic ;defaultip=192.168.1.103

Hi List, Additional with the latest tries from the below I get a nice random seg fault when I hangup on PSTN. (With obviously no sound on x-lite, still!) asterisk -vvvvgc results after hanging up the pstn line in: -- Executing Hangup("SIP/1087997-d79f", "") in new stack == Spawn extension (sip-phone-out, h, 2) exited non-zero on 'SIP/phonenumber-d79f' Segmentation

Hello: I have the linux version 2.0 of x-lite downloaded. Does anyone know exactly what settings needed to reach the asterisk server on my home network? Internet ->DSL transparent bridge ->router ->asterisk ->softphone x-lite attempts to login and register, but times out. There must be some setting I'm

Hi all, Still having the one way sound problem. Any suggestions how to hunt the problem down ? Regards, Thorsten --------------------------------------------------------------- Hi all, We have a very basic * installation for testing purposes. The * is connected to PSTN with BRI and setup with X-Lite over plain lan. (local IP's) OS: Linux/Debian unstable. Asterisk CVS-10/29/03-23:46:26

Hi all, We have a very basic * installation for testing purposes. The * is connected to PSTN with BRI and setup with X-Lite over plain lan. (local IP's) OS: Linux/Debian unstable. Asterisk CVS-10/29/03-23:46:26 chan_capi On the IP side: X-lite (build: 1084) Calling and get calls on PSTN from X-Lite is no problem. We only get sound from PSTN to X-lite. Never from X.-lite to PSTN. The

Hi all, I'm working on an implementation of VoIP en Linux. I have a Debian Suse (*.*.*.173) with an * and a X-lite client and a Red Hat 9.0 (*.*.*.172) with another softphone X-lite. Both of the softphones are registering and appear in the peers (sip show peers) with the good parameters of address and port. If I try to make a call, * receive the INVITE request and send a 404 NOT FOUND answer.

Hello, I'm trying to place a call to asterisk using X-Lite. Asterisk is setup with some Grandstream phones. I can call from one grandstream extension to another. When I try to an extension with X-Lite, it comes back with Status of SIP/2.0 404 Not Found. X-Lite is not registered as asterisk extension. It is just sending a sip invite to extension@IP. Does the X-Lite need to connect to

Hi, I am playing around with SIP extensions on my local lan using X-Lite but I am having a bit of difficulty, I have set up X-Lite and my sip.conf accordingly, but when I start it I get the following message: "Login failed! Contact Network Admin" I am still able to dial local extensions on my * with x-lite even though it is in this state, although trying to dial my sip extension from

I'm having a problem with using X-Lite to initiate a call via Asterisk out an X100P analog port, using the Speex codec. I've put in the registry fix for X-Lite and Speex so that works OK, and calling the echo test extension works. However, if I call out the analog port it appears that audio being initiated by X-Lite is being dropped, but audio being initiated from the analog line is

I'm playing with codecs at the moment and have found some notices errors when x-lite/pro connects to asterisk with Speex or ILBC. Initially I was getting garbled sound, but after changing magic number for both codecs to 97 (as per http://www.voip-info.org/wiki-Asterisk%20phone%20xten%20xlite and http://bugs.digium.com/bug_view_page.php?bug_id=0000918) I was able to get normal voice. BUT,

Hi guys, I do have some weird situation. I do have an * box, and I want to connect to that box from my Windows box by SIP via X-Ten Lite. I made configuration of that soft phone as it was suggested by lots of tutorials I found by Google. But... it doesn't work! I don't know what is wrong there, but I have unobstructed access to my asterisk box, created user in sip.conf, enabled

XLite does not support transfer... You have to buy their XPro -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mateo Meier Sent: Tuesday, February 22, 2005 3:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Anybody using X-Lite Softphone ? tryed to forwarda call to X-Lite.... Hey Guys Im