similar to: xten will not send tones to * and i from sip phone

Hello Everyone I just bough my Xten Eyebeam but i don figure out how to make the video works i only see a black screen where de remote video suposse to appear, Any help regarding this matter will be very preciated Thank You

All, I have my * box NAT'd with all ports forwarded that are SIP related (based on Wiki). I also have nat=yes, externalip=WAN address of firewall, internalip=LAN network of *. I have my Xten soft phone on a PC which is NAT'd behind firewall with ports forwarded. I have also followed instructions on Wiki for Xten. I can authenticate fine, and sip show peers shows my extension is OK,

Which rpm has libstdc++.so.5? xten-xlite for linux says it needs this. Of course, there will be something else it will need after I get this....

Sorry, I don't remember who was asking about this, but it seems that if you record a video message that contains the send video start, it will actually fire up the remote receive window. I.E. Previously I was using the recording section of voicemail to create my video IVR's. This meant that when I arrived at the section to record the message, I had already clicked send video in Xten.

Dear Guys, I?ve test Xten Lite softphone to connect to my Asterisk Box but it registers all the three lines at the same time and if I try to dial an extension it tries to reach 3 Ext. at the same time, can somebody haved this trouble? and how can I fix it. Also, I ?ll like to have the Xten LITE or PRO Softphone (Lite is free and PRO about $50.00 USD) it can hanle 3 lines (lite) and 6 lines

hi xten.de produced an activex for web phone. but I can not find any link for download. can u help me ? best Mani ____________________________________________________________________________________ Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit. http://farechase.yahoo.com/promo-generic-14795097

Guys. Im writing this because Ive checked the wiki, Xten website and read a lot of docs and still cant figure out a way around the NAT issues. Maybe somebody else can give me some ideas from a fresh perpective. My test setup is this: Asterisk -> 2wire homeportal Firewall -> internet Computer with Xten eyebeam The asterisk box and the computer with xten beam are behind the same

I have tried loads of configurations but I cannot get it to work. I basically have three computers and want to use soft phones on two of them to connect to asterisk so I can use them instead of dialling into my X100P card as the phone bill is getting bigger. I assume that I can configure * to accept calls from the soft phones and then route them according to my dial plan. I am basically trying

I have d/l the iso (*@home 0.9) , built the * box and followed the directions in the * handbook and http://www.geekgazette.com/index.php?option=com_content&task=view&id=2&Itemi d=26. I created extension 200 and verified that * was running fine. Loaded Xten lite, setup the proxy for local ip (10.0.0.201) per the handbook. After turning off the Norton Firewall protection, I am able to

Ok I have a question. I have Xten-lite working with our Asterisk system and I am able to make and get calls. But the main problem is the sound is very choppy and sometimes it cuts off words. I have tested it with ulaw and alaw as well as GSM. They all do the same. ulaw seems to work better. I also have an ATA-186 which works great without this problem. Here is my Sip.conf settings.

I have seen a couple of messages on the Xten and the work done by William Walsh (Kudos). It is not clear in my mind the advantages of SIP phones versus using GnoPhones (once we complete the work for the Windows version). Since I lack the experience with IP SIP phones, can someone, high level, tell me when it makes sense to use them. Is it complicated to set up on the Asterisk side? Thank you.

from: http://www.xten.com/index.php?menu=download Seems to be working fine. just untar and run. But now I have to do some real voice testing. Not to my desk phone 2 feet away and get blasted with feedback!

I've just down loaded Xten Lite and it is now build 1079. It now finds the NAT firewall type and has loads more to configure. But it doesn't work on my poor W95 tablet PC. -- Dave Cotton Directeur Linux Autrement 193 rue Marcel Cerdan 84270 Vedene 04 90 23 30 81 "Internet Sheriff Technology" revendeur en France <http://www.linuxautrement.com> IAX 17004902330

I have a tdm 3xfxs and 1xfxo, aslo I have a setting with 1 snom 190 and 2 xten line. I can call from the snom to the ptsn line at the fxo port ok. I can call from the ptsn to the xten lite phone. I can call from the xten lite to snom but what I CAN`T do is; Call from xten to ptsn. When I dial from the xten, I can hear the dialed party, but he cannot hear me... Tips? Help? What I'm

Same here Michael and the PocketPC version seems unaudible with any codec; early days trying that though. Simon -----Original Message----- >From: "Michael Van Donselaar"<mvand@neb.rr.com> >Sent: 22/04/03 04:10:24 >To: "asterisk-users@lists.digium.com"<asterisk-users@lists.digium.com> >Subject: Re: [Asterisk-Users] Xten - Free windows SIP

[this post is just as much for the archives as it is for immediate consumption] Somehow I had completely missed the announcement that XTen Networks had released a SIP client for Mac OSX. Well, they have. It seems to mostly work, even! Very cool. The User interface leaves something to be desired, with it's menus trying to force me to use cell-phone sized windows (!!!!) to manipulate

I've tried every softphone available for linux and I've found that Xten X-pro for Windows seems to load fine under the WINE API, the problem is - it wont work the sound... how do i get the sound to be emulated so I can place and receive calls using this software? the program's options seem to list my audio devices. keep in mind i have to use a microphone and sound output. is there a

Can I use the Xten-Lite with Asterisk? I am experimenting with a total soft solution for a carrier that uses the Vocal Data platform (Hosted PBX). I would also like to know how I could use asterisk to connect the carrier without using any gateways. I would like to do something that Avaya has already done, IP trunking. What I want to accomplish is having the carriers gateway connecting

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

It's me again. I'm using XTen X-Pro softphones which have the ability to handle multiple conversations simultaneously. When my agents login and calls come in, everything works fine, as expected. Calls are directed to available agents, and when no agent is available, the caller sits in queue. However, since the X-Pro has the ability to handle multiple calls, how can I make the