Hi, In the early stages of deciding how to try and develop this environment, I looked at all the protocols that could be used. SIP was chosen just because it seemed to me that it was the most widely used protocol. I believe IAX is a new protocol with a little less documentation and examples. The good thing about this Jain-sip-phone is that it saves a lot of time since many of the important classes are more or less written already. In short, my goal is to create a custom softphone GUI interface. I am using this Jain-sip-phone as an example, so that I could learn the SIP protocol/RTP transmission better.
I have not really started altering much of the code yet because I was trying to see if it would run as is, so I have not tried dialing the Jain clients without a subscription. I believe Asterisk does accept subscription requests, but for some reason it doesn't like this one. I will soon start to experiment with the source code. Thanks, Denis -----Original Message----- From: Gerald A [mailto:[EMAIL PROTECTED] Sent: Monday, August 27, 2007 9:30 AM To: Kutman [EMAIL PROTECTED](Mat) DAEPM(R&CS)@Ottawa-Hull Subject: Re: [asterisk-users] Can't create audio conversation between softphonesthrough Asterisk Hi, On 8/27/07, [EMAIL PROTECTED] < [EMAIL PROTECTED] > wrote: Thanks for the reply. I have a small LAN network which I have connected with an Asterisk server. My Asterisk box and the user pc's are connected through a LAN switch. This network is not connected to the internet. The "UNREACHABLE" message does seem to point to what you mentioned below (Asterisk not being able to ping the phones), which seems weird to me. When I use X-Lite softphones on those user pc's, I can connect them to the Asterisk server fine and make calls. The subscription occurs when I try to add another contact(In the same LAN network) from one of the user pc's. I am attaching the console results that I get within Eclipse when I run this softphone. Ok, one more silly question -- might it be possible to do this with IAX? (I tend to lean on IAX for things, as it's more versitile and robust, if not so widely deployed). I'm not sure exactly what you are trying to accomplish, so I'm focusing on the questions you are having issues with. A bit of context might show up as another solution, though -- if you are able to provide it. I don't have time right now to dig through the traces, but I have a related question. Have you ever got a call to go through dialling from one Jain client to the other, without the subscription? My gut feeling is that there might be a basic config issue with the Jain client that is causing an issue, as what you want to do doesn't sound too difficult. Thanks, Gerald.
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