[asterisk-users] Can't create audioconversationbetweensoftphonesthrough Asterisk

[Kutman.DK at forces.gc.ca]

Hello,
 
Looks like I have been able to get the jain-sip-phone to work.  The problem seemed to have been an sdpFactory.createconnection call.  It was passing one parameter, which was the IP Address.  I had to change this to the call with three parameters (ie: sdpFactory.createconnection("IN", "IP4", etc).  This is because by default eclipse was setting the address type to "IPV4", which didn't seem to work.  Such a minor issue, but it wasn't easy to find.
 
Thanks,
 
Denis

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Kutman.DK at forces.gc.ca
Sent: Tuesday, August 28, 2007 9:45 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Can't create audioconversationbetweensoftphonesthrough Asterisk


Hello,
 
I do not think that the presence bit will be crucial to our application.  Thanks for your help.  I will keep you posted if I get any progress.
 
Thanks,
 
Denis 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Gerald A
Sent: Monday, August 27, 2007 5:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't create audio conversationbetweensoftphonesthrough Asterisk


Hi,


On 8/27/07, Kutman.DK at forces.gc.ca < Kutman.DK at forces.gc.ca > wrote: 

Thanks very much for the help, I appreciate it.  Recently, one of my co-workers and I have altered the code to just register with the Asterisk server and place an audio call.  This gets rid of the subscription part of the application, so I do not get the "489 Bad Event" error anymore.  I believe the "488 Not Acceptable Here" error occurs when the invite is being sent.  After the sdp body and header information are created, they are sent as an invite for the audio call.  The problem seems to be some part of the invite that we are sending.  I have a hunch that it may have to do with the codecs that the Jain-phone chooses.  I will continue looking into this.


Glad to hear you were able to get some traction with the voice calling.

Is the presence bit something that is critical to your custom app? I'm going to be fiddling with some soft phone stuff soon, so I am still planning on taking a peek at Jain just for the heck of it. 

Keep me updated on your progress, and if you need any assistance, give me a shout.

Thanks,
Gerald.

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