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sipxezphone won't produce RTP stream

Well, I got back to Issue tracker again searching for something that could help me and I think I finally found it.

http://track.sipfoundry.org/browse/XMR-43 Reports a problem and I have confirmed that after undefining the directive DOING_ECHO_SUPPRESSION in file MpCallFlowGraph.cpp I could start sending my audio.

I just hope that not having DOING_ECHO_SUPPRESSION defined doesn't create other problems.

Regards,
	Ricardo

-----Original Message-----
From: sipx-dev-bounces (at) list.sipfoundry.org [mailto:sipx-dev-bounces (at) list.sipfoundry.org] On Behalf Of Ricardo Monteiro
Sent: segunda-feira, 23 de Janeiro de 2006 12:33
To: Michael Cohen; Clyde Rogers; sipx-dev (at) list.sipfoundry.org
Subject: RE: [sipX-dev] sipxezphone won't produce RTP stream

Hi,

	I'm having the same problem (I have posted an email about this but got no help yet - Subject:"[sipX-dev] Problems - voice is not sent" ).

	Have you reached any conclusion Clyde?

Thanks for any clue, or for some tip where to start looking/debugging.

Ricardo

-----Original Message-----
From: Michael Cohen [mailto:mcohen (at) pingtel.com] 
Sent: terça-feira, 3 de Janeiro de 2006 13:46
To: 'Clyde Rogers'; sipx-dev (at) list.sipfoundry.org
Subject: RE: [sipX-dev] sipxezphone won't produce RTP stream

Clyde,

If you can post an Ethereal trace of the network activity, that would be
helpful for debugging the problem.  Also, giving us the sipxezphone-config
settings would be helpful.

-Mike

-----Original Message-----
From: sipx-dev-bounces (at) list.sipfoundry.org
[mailto:sipx-dev-bounces (at) list.sipfoundry.org] On Behalf Of Clyde Rogers
Sent: Thursday, December 29, 2005 1:08 PM
To: sipx-dev (at) list.sipfoundry.org
Subject: [sipX-dev] sipxezphone won't produce RTP stream

Hello, all.

I'm a VoIP newbie.  I've compiled sipxezphone (for windows, off the  
3.0.0 sources, using vs .net 2003) in preparation for a project.  I  
updated the include file that disabled connections based on date, and  
have been attempting to use the resulting executable before modifying  
anything.

I have an asterisk pbx.  Sipxezphone has registered successfully with  
asterisk, and I have been able to make calls between sipxezphone  
instances, hear a ring tone, pick up and even connect.  But then no  
voice flows---I don't even see packets on the network.

If I use an x-lite phone on both ends, everything works great---call  
is made, I pick up, and can talk between phones (although the RTP  
packets all flow though the pbx system---I expected them to go direct  
from phone-to-phone---am I missing something here?).  If I use x-lite  
on one end and sipxezphone on the other, I see RTP from the x-lite  
(through the pbx) to the sipxezphone, and can hear the voice just  
fine on the sipxezphone (hooray!).  But I get nothing when speaking  
into sipxezphone---no outbound RTP, no sound, no nothing.

I also can hang up from the sipxezphone just fine.  Everything seems  
great, except that sipxezphone won't send the RTP stream (not even to  
the wrong destination).

Any ideas what I've done wrong?  Is there some known configuration  
thing I'm missing?  Is there any other information I can provide that  
would be illuminating?  Is there any documentation on the flow of  
voice from microphone to RTP in sipxezphone?  That would at least  
give me a starting point for debugging.

Thanks,

Clyde Rogers

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