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From: Paul <kp...@ni...> - 2005-10-04 15:02:45
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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 |