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<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=681241019-27082007>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.</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN
class=681241019-27082007></SPAN></FONT> </DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN
class=681241019-27082007>Denis</SPAN></FONT></DIV>
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<DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]<B>On Behalf Of </B>Gerald
A<BR><B>Sent:</B> Monday, August 27, 2007 2:59 PM<BR><B>To:</B> Asterisk Users
Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> Re:
[asterisk-users] Can't create audio conversation betweensoftphonesthrough
Asterisk<BR><BR></FONT></DIV>Hi,<BR><BR>
<DIV><SPAN class=gmail_quote>On 8/27/07, <B class=gmail_sendername><A
href="mailto:Kutman.DK@forces.gc.ca">Kutman.DK@forces.gc.ca</A></B> <<A
href="mailto:Kutman.DK@forces.gc.ca">Kutman.DK@forces.gc.ca</A> >
wrote:</SPAN>
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<DIV>
<DIV>
<DIV><BR><FONT face=Arial size=2>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.
</FONT></DIV></DIV></DIV></BLOCKQUOTE>
<DIV><BR>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. <BR>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. <BR></DIV><BR>
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<DIV>
<DIV>
<DIV><FONT face=Arial size=2>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. </FONT></DIV></DIV></DIV></BLOCKQUOTE>
<DIV><BR>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. <BR>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<BR>the issue you are
having. (It might be _just_ the subscribe that is having an issue). <BR><BR>I
should have time later this afternoon to check your traces, and I'll try and
give Jain a
kick.<BR><BR>Thanks,<BR>Gerald.<BR></DIV><BR></DIV></BLOCKQUOTE></BODY></HTML>