[Asterisk-Users] Trouble with all linux sip softphones.... And asterisk/linphone/kphone SRPMs

[John Morris]

Hi, I've been messing with getting SIP working for days now, with
limited success.  I've got Asterisk set up on a remote server with the
echo test.  Please try it out to verify I've got the server working
right:

    sip:robot at nixon.butchwax.com

Running FC1, ThinkPad T22, headset thru the soundcard.  Asterisk is
asterisk-1.0_RC1.  No NAT.  The phones I've tried so far are as follows.

** Linphone:  Check out my SRPM at http://www.bigu.org/SRPMS/
  Sound is fine.  Doesn't seem to pick up anything from the microphone,
though.

** kphone:  Check out my SRPM at http://www.bigu.org/SRPMS/
  Sound is fine.  Picks up sound from the microphone, but the echo-test
repeats it back after passing it through a Mr-Roboto filter.

** tkPhone:
  Sound is fine.  Doesn't seem to be reading from the mic, no traffic
going over the network after the 'demo-echotest' recording finishes. 
The following errors are continuously repeated from tkPhone:

  sent 63426 (3),received 30228 (3);read 615040 write 293120 need
  608000 jitter 38!!!!!!!!!!!!!!!!!!!!!!!!!! Error:
  !!!!!!!!!!!!!!!!!!!!!!!
  Data won't fit within the current RTP packet size

** SJphone:
  Last night:  sound worked fine.  Actually sent sound from the mic,
which came back after about a 5-second delay, but which sounded quite
good.
  Today:  establishes a connection, but absolutely no sound in or out. 
:P

If anyone's interested in my SRPMs, I'd like to know.  The asterisk RPM
builds the zapata, zaptel, and asterisk sources; you must have your
kernel-source rpm installed for it to build the modules against.

Let me know if there's something obvious I'm missing.  Thanks-

	John