search for: xtens

...ed 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, but whenever I dial another SIP or zap channel, I can a "max retries exceeded" on my * box. Can somebody post a working config with * behind firewall w/ports forwarded and xten behind firewall w/ports forwarded? Thanks, -- Aaron

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

hi there! I have a couple phones connected to a sipura ata and if I go into *- IVR, I press options on the regular phones and it all works fine and dandy. then I connect an xten softphone, a new extension in my dialplan, I dial the ivr, * asks me to dial something to go through it, I press keys on xten, but nothing happens, * just times out through as if I did not press anything! is there some sort of configuration out there to tell the xten softphone to work as expected? thanks! Then another p...

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

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....

..., 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. This meant that when you dialed an extension which played back these files, the video wouldn't start unless you sent video first. Now, if you don't click send video in Xten until after you hear the beep for recording, it works. It will play you the video just by dialing the extension. I'm using the latest beta of eyebeam...

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

Does anyone have any views on the best software base SIP client to use that normal users could use with Asterisk without being too techie ? I have tried the X-Lite client with varying success. The first version worked OK but music on hold broke the voice paths and the slightly newer version initiated the call but failed to make the voice connect in both directions. The SJphone client works but

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 (Pro) how I need to configure the user into the Asterisk....

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

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.

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

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 start Xten and it says Logged in. I wanted to start with the easiest thing, so I just wanted to SIP to my loc...

[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 got the XTEN Lite soft phone mostly working with * but it's dropping out like a very bad cell phone call. The GSM codec is worst (unusable), G711u and G711a are best but not good enough to use. I don't think it's a lack of bandwidth. What tuning options or approaches should I be investigating to make this work. Also, what's the best soft phone(s) for Windows XP?

...e 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 to write some extensions to asterisk and need to debug the code but need to phone in to do it. I could route in from the net if it wasn't for the fact that I am behind a NAT and I've decided to give up on getting it to work as I have just got stuck. So this is another option to attempt to get debugging. So t...

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

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!

...phone connected to a local FXS port: SIP/st2030, SIP/xten and Zap/1. Relevant part of dialplan is: exten => 200,1,Dial(Zap/1) exten => 202,1,Dial(SIP/st2030) exten => 203,1,Dial(SIP/xten) exten => 300,1,Queue(testQueue) exten => 502,1,Dial(SIP/st2030) Notes: - all extensions have a 2nd priority wich does Hangup() - the 2xx extensions also have a hint priority - when testing, the testQueue is configured to have the "other two channels" as static members in a "ring all" strategy (ie: the two SIP channels when testing dialling from...

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.