[kphone-users] kphone audio from (alsa) mic not getting to called party

[Paul <kp...@ni...>]

Hi all,
I've just setup kphone 4.2. I'm using gradwell.net as my voip provider (ther 
run asterix I think). using xten-lite on windows everything works fine,

Installed kphone, copied settings, all seems OK, I dial 160, hear the voice 
telling me it's an echo test but nothing gets echoed back from the mic. I can 
record from the mic with Krec, so I'm assuming that the sound setup us OK. 
Using the DTMF buttons seems to do something.. it beeps while the 
announcement is being read ( is the call in "receive only mode at this 
point? ), but once the announcement stops, pressing the DTMF keys does 
nothing.

I'm behind a standard NAT/firewall ( a d-link router ), and use gradwells 
outbound proxy (nat.gradwell.net:5082) 

A portion of my log is pasted below 
Note that the "dtmfsenderTimeout" happens immediately after the CallAudio log 
line, at the start of the call,  even if I don't hit the DTMF buttons

Any help gratefully received!

Paul



SipClient: Sending: 15:52:53.639
--------------------------------
ACK sip:GQjTu-bCFMCV5qOLEmPbCbdGe3vD5qv@193.111.200.12:5082 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4;branch=z9hG4bK5AD2EBBC
CSeq: 5261 ACK
To: <sip:16...@si...>;tag=f5177054
From: "Paul Nicklin" <sip:XX...@si...>;tag=71B124F4
Call-ID: 291536689@192.168.1.4
Content-Length: 0
User-Agent: kphone/4.2
Contact: "Paul Nicklin" <sip:XXXXXXX@192.168.1.4;transport=udp>


SipClient: Sending to 'nat.gradwell.net:5082'
SipCallMember: localStatusUpdated: 200
CallAudio: Using G711u for output
CallAudio: Sending to remote site 193.111.200.12:10238
CallAudio: Opening ALSA device for Input



----------<170 - 2730>--------------


CallAudio: Creating ALSA->RTP Diverter
dtmfsenderTimeout