similar to: WINE and Sound - XTEN X-Pro Softphone

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

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

Hello, To do some testing with Asterisk installed the latest Asterisk @ Home in a Vmware system. All worked fine, I can access the web interface (AMP). I have setup the extention and X-Lite softphone according to the description in the Wike (http://www.voip-info.org/wiki-Asterisk+phone+xten+xlite). I can dial 200 (the softphone extention) and 1234 and they connect (the softphone shows this, as

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

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

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

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.

Anyone used the combination above? We are and it sounds like crap. The audio drops out in regular intervals which suggests that someone's g729 isn't doing its job correctly. I'm blameing XTen cause when I make a ulaw call that gets converted to 729 using digium's 729, calls sound fine. Anyone else? Similar experiences? -Matthew --

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,

Hi, IF i use a sip softphone or a iax softphone with asterisk, i get a lag of about 1 second. The two phones were on 2 different pc's near me. When I speak on one, i hear it on the other after about 1 second. I tried using iaxComm, Xten Xlite, etc. Same. FYI: The codec used was GSM. Using the fxo and fxs interfaces on the digium cards with POTS have no such issues. Any clue where the

Hi all, I have problem with my Asterisk. I'm using the softphone Xten-Lite. I've removed the SIP client information in sip.conf. The softphone can't register to Asterisk, but it can make outgoing calls. I've tried to add back the SIP client information into the sip.conf, but make a wrong password in the softphone. The registration and outgoing calls are failed as expected.

Guys, anybody has seen or is using some kind of softphone on any square screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they do work on Wm5 but they are designed for standard screens, anybody using anything on square ones?

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

Subject says it all. We are looking for an open source IAX or SIP softphone that can handle multiple lines and have G729 abilities. We don't mind paying a few bucks for the 729 license. We need open source so we can co-brand. Right now our only solution is to buy $5000 worth of XTen Pro; then we get co-branding and 729. But if we can save $5000, all the better. -Matthew

To all: I am having an echo problem with X-Lite and iaxComm. I am using the monitor speakers and a desktop microphone. My problem is that the sound from the speakers is repeated by the microphone causing an echo that is annoying. Is this correctable? I have searched throught this archives and played with various settings but have not been able to fix this. I have purchased a couple if

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.

I have tried sjphone - worked well, although I think my 3 year old IPAQ had a bit of a hard time keeping up with the pace as there was quite a delay in the speech. Probably says more about my ancient IPAQ than SJPhone. Sam Lex Lethol <lethol@gmail.com> wrote on 23/09/2004 15:31:39: > I tried the xten one and didn;t like at all.. > > Havent tried to SJPhone, but my guess is

Hello I Installed Asterisk on RedHat 9. I am currently try to configure minimum with two softphone talking to each other over the LAN. I am using X-Lite softphones from xten.com site. I defined 3 phones in sip.conf and also specifies in extensions.conf file. I am able to ring other computer but there is no voice exchange ( i can't hear any think except ring). Here is the portion of sip.conf

Hey guys, I have just developed this problem with my Windows XP box. I think it started since I installed XP SP2. Both SJPhone and Xlite does some kind of bridging with the speaker out port. When ever I make a sip call to where ever, the other party hears a lot of echoing. Well I noticed just now when I was playing mp3's via Winamp, the music was being played through my sip calls that I

Hi All, I've just setup a test Asterisk system that allows incoming/outgoing calls via an ISDN card (l4i) and incoming/outgoing calls via SIP (iconnecthere). I have two SIP Softphones (Xten X-Lite) for making and receiving calls. When receiving an incoming call via the ISDN interface the sound quality is fine for the Softphone user (i can hear the caller perfectly), but the person