similar to: Asterisk SIP cannot restrict call from softphone before registration

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?

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

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 everybody, I'm new to this matter and I spent three days in trying to connect one SIP Softphone to an Asterisk Box. I always get error 401 or 403... I don't understand very well settings in Softphone program: con anybody show me how to set up a minimal running system with no public lines or external Proxies? Thank you for your kind help... Ciao Mauro

>I went back to the example system direct from CVS with small >additions to sip.conf and extnsion.conf needed to make one >xten X-Lite phone work. I can dail in and hear the anouncements, >call the demo server at Digium. The audio quality I hear >comming from Asterisk back to X-Lite is good (9 on a 10 scale) >but the sound volume comming from the X-Lite extension is very low

Dear all, Mayybe this is a little off-topic but I don't know of any other place to ask for it... my apologies in advance! I'm looking for a softphone (SIP) with multiple profiles support. Right now I use SJPhone on SuSE 9.0 Pro, which allows to create several profiles but, AFAIK, it's not possible to use them all at the same time. I need this feature because I use different VoIP

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

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! I decided to apply Chris's patch for the rtp problem, it is working just fine now. Thanks Chris!. I think that Mark should submit it to the CVS. Ildefonso. icamarg@unet.edu.ve >Pete, > >Try this patch below... I noticed that eStara's softphone has the same >problem as xten's softphone when it comes to DTMF. Seems as though = >Asterisk >is not looking for

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

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.

Can anyone _recommend_ a downloadable OSS softphone that _works_ under Linux and is compatible with Asterisk. So far I have tried kphone and linphone and had problems with both, and I am still waiting to hear back from the X-Lite beta folks. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just

Hi everyone, Has anyone experience with xten.net's X-Lite SIP softphone and asterisk? I have a problem and I think X-Lite is not even trying to contact SIP proxy while dialing. Tomica ---- This mail was sent thru ZGWireless free network - www.zgwireless.net, Internet connection sponsored by Iskon Internet d.o.o. - www.iskon.hr

Sorry if you see this message repeated twice. I would like to set up hard phones and softphones with the same extension so that when anybody in the company dials an extension, each user's hardphone and softphone will ring at the same time. I've tried setting this up before, but I noticed that the last sip device to register with the same extension is the only one that rings when the

Hi U can visit the http://iaxclient.sf.net for some opensource underdevelopment softphones. Take Care Obaid Amin Syed >From: Chris Albertson <chrisalbertson90278@yahoo.com> >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] Good W2K softphone >Date: Fri, 3 Oct 2003 23:00:13 -0700 (PDT) > > >I haven't

Hi guies, I know I know this subject have been The most written subject about VoIP Right... but I just want to make clear, just one time ! If Asterisk is on a Public IP Address and a softphone behind the nat, sip.conf must contains for this phone: nat=yes .... Now if I want to configure my sipphone (X-Lite) placing behing the NAT, it must have in "Domain/Realm" the external IP

are there any open source sip softphone (Window OS version )?

Hi, everybody: I have looked at the Polycom entries on www.voip-info.org, and they're outdated and convoluted and full of errors. All I want to do is get my Polycom 501 to register with a working Asterisk server. I want to do the configuration locally on the phone through the console. (The server works with an Xten X-lite softphone.) Has anyone done this? What do I need to do? Thanks,

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?

I know this has been covered more times than to mention and this is where I got most of my info from... But I am having issues with this. I can't seem to get the phone to register with *. This is being tested on a internal network right now. Here is the setup - sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context