[asterisk-users] Can't create audio conversation betweensoftphonesthrough Asterisk

[Kutman.DK at forces.gc.ca]

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.
 
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 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't create audio conversation betweensoftphonesthrough Asterisk


Hi,


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


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. 


The reason I asked is because IAX works better through firewalls and is easier to troubleshoot. It's not as widely deployed as SIP, but it does work around some major things that SIP makes harder. 
I'm not sure of the quality or lineage of the  JAIN application code, so can't comment if it's a good jumping off point. 



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. 


Subscription is used for presence. It can be used in an IM type app, or to "light up" a button on a  phone when someone is busy. 
It shouldn't be needed to exchange a call though, and if you can do it without the subscription piece then it could help to pin down
the issue you are having. (It might be _just_ the subscribe that is having an issue). 

I should have time later this afternoon to check your traces, and I'll try and give Jain a kick.

Thanks,
Gerald.


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