similar to: Really Wierd softphone problem ... must read

Im looking for an Intercom solution thats interoperable wit Asterisk. Ive read several posts about people using the 2nd lines on some SIP phones w/speaker phone. Unfortunatley I dont that is going to cut it in a large warehouse enviroment. Does anyone have a solution that uses a "loudspeaker" ? Thank you, Steve Maroney

Dear all expert, (i posted this question one time, but i couldn't reach the answer -so allow me to post here) 1)I download asterisk realse version 1.2 beta1. After that i compile it successfully and run it with: asterisk -vvvc 2)I follow the instruction in http://www.asteriskguru.com/tutorials/asterisk_voip_ipphone.html in sip.conf: i add two account: [ivan] type=friend username=ivan

Hey guys, Im interested in hearing about servers (and thier hardware specs) that successfully run both asterisk and samba for an office of maybe about 12 extensions (SIP) and about 12 workstations. Im hopeing to not only replace a traditional PBXs with Asterisk/Linux but to provide a solution to needs such as a file serving, email serving, etc. Ive read the Success stories form voip-info.org but

Hey guys, Im about to sign up for VoicePulse Connect. Of course, I plan on using my asterisk server to "register =>" with the service. I would rather sign up with VoicePulse via SIP instead of VoicePulse Connect. My asterisk box is behind another Linux box serving as my nat/firewall. Does anybody think I will have a problem ? Should I stick to IAX and VoicePulse Connect or can I use

Hey guys, How much disk space is used by Asterisk to store voice mail for about 10 - 20 users/mailboxes Thank you, Steve Maroney

Anyone know who that good looking female is thats on the Digium.com website ? Ok, my Real question is I noticed that Digium has relesed a new T1 card with an echo canceller. I also noticed that its supports E&M Circuits. Im I have very little knowledge on T1 circuits and traditional PBX's so what Im asking is can I use Digiums T1 card to connect to another PBX via a tie line ? Or does

Im trying to get IAX to work between my * and FWD. I activated my iax2 account on iax.fwdnet.net and I get the output: "Registered to '65.39.205.121', who sees us as 68.14.203.254:4569" when I start asterisk. I tried used the Call Me tool on fwdnet.net but I dont get any calls even though the Call Me tool says everything looks ok. Can someone call my FWD number and just leave

Now here is strange problem i experience. Setup is easy, IAX line out with SIP softphone registered to Asterisk. All work fine except for one client. When using Sjphone the other end can not hear a thing. When using X-pro the opposite happens, local user can not hear a thing. These softphones work fine on other clients on same network. I've also tested several headsets but same outcome. Also

Hi folks last weekend, I tried Windows Messenger first time and was stunned by the little latency it gives. Until now, I've been using softphones on Linux exclusively, like iaxcomm, linphone and sjphone, and they all give me about 1, at times even 2 secs delay. Whereas Messenger really seems to be in the millisec range. Of course, I'm now curious why there is that difference. Clearly,

Googling arround I found a number of pocket pc softphones. Of those I was only able to install SJ-something (removed it). Is there one (pocket pc softphone) that works? Thanks, Cosmin Prund

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

Is there a soft phone for PocketPC or iPaq? If not, is someone working on it? I will be more than willing to contribute my mite if needed. Thanks, -- sudhir

Hi group I'm newbie on Asterisk so I followed the Linux Networking CookBook by Carla Schroder to make my first call. My asterisk box is on a Debian box with an public static IP. The clients (2) are with dynamic private IP's I'm using SJphone on a PC and a Linksys PAP2-NA to make calls between them. Both of them register well on my Asterisk server but when I call from the SJPhone to

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

I am running Asterisk on our LAN with Cisco IP phones and softphone clients, SJPhone and Firefly. Everything on our LAN works fine and the quality is good. We have recently registered other callers not on our LAN who are using SJPhone and Firefly as well. The audio quality is fine, but I consistently hear a buzzing sound in the background. We have experimented with the line in and microphone

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

Hi, I have two softphones connected to an Asterisk "stable". I have two extensions, say 1000 and 2000. When 1000 calls 2000, the call cannot be completed; the softphone (either Diax97a , SJphone, Firefly 1.8) on extension 2000 will ring, but as soon as the call is picked up, extension 2000 will hang up the call. The softphone on 1000 (SIP, SJphone, e.g.) will give a "403

I'm using the SJphone softphone, and I've got a nice little SIP-only setup, using (amongst others) stanaphone, VOIPtalk and FWD. But I'd like to be able to use my SJphones to dial directly to folks who provide a SIP URI, eg: 100@calluk.com, without either having to switch profiles in SJphone (to direct SIP) or having to define calluk.com (in this example) as a destination in

Hi All, I'm currently using SJPhone as a sip client with Asterisk. It works perfectly with a USB Yap Phone but there is was slight problem! If I setup SJphone to use the USB device for audio the ringing tone is also over the USB device. If I'm away from my desk I can't hear the phone ring! Is there an option to use one audio device for calls but another for the ringing sound?

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