Dear Asterisk experts, I've just downloaded Asterisk CVS version (since I'd like to manage its configuration from RealTime).
Next, I have kphone on the same Linux host, and VmWare virtual machine with Windows and X-Lite IP phone inside. I successfully tested the demo's with X-Lite, but failed to hear something with kphone (v4.1.1). There were NO problem with this kphone and stable 1.0.7 and 1.0.9 asterisk versions. Asterisk does not claim that something wrong, it logs on its condole that it just "-- Playing 'demo-congrats' (language 'en')", nothing else. On the other hand, kphone finishes their log with that: ===================================================================== ... res_search: NO result ! res_search: NO result ! SipClient: Sending to '127.0.0.1:5060' SipCallMember: localStatusUpdated: 200 CallAudio: Using GSM for output CallAudio: Sending to remote site 127.0.0.1:13998 ERROR: Open Failed ** audioIn: openDevice Failed. CallAudio: Creating OSS->RTP Diverter dtmfsenderTimeout DspAudio: Broken pipe (b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b) ... ===================================================================== (The complete log are attached to the e-mail.) So instead of audio I see the repeated "(b)" sequence dumped to the terminal from which kphone was launched. I'd blame kphone for that, but again, why I didn't experience that with the stable asterisks? Thank you a lot for any help! -- Best regards, Timur Elzhov
$ kphone & [1] 29730 $ Found 1 interfaces. SipClient: Listening UDP on port: 5062 SipClient: Our address: 127.0.0.1 KCallWidget: Switching calls... CallAudio: listening for incomming RTP UDPMessageSocket: Listening on 32809 UDPMessageSocket: Retrying... UDPMessageSocket: Listening on 32810 CallAudio: Opening OSS device /dev/dsp for Input and Output ERROR: Open Failed ** audioOut: openDevice Failed. CallAudio: Creating RTP->OSS Diverter SipClient: Sending: 11:22:24.494 -------------------------------- INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK5F746FD3 CSeq: 7312 INVITE To: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp From: "Timur Elzhov" <sip:[EMAIL PROTECTED]>;tag=6873C9D3 Call-ID: [EMAIL PROTECTED] Subject: sip:[EMAIL PROTECTED] Content-Length: 222 User-Agent: kphone/4.1.1 Contact: "Timur Elzhov" <sip:[EMAIL PROTECTED]:5062;transport=udp> v=0 o=username 0 0 IN IP4 127.0.0.1 s=The Funky Flow c=IN IP4 127.0.0.1 t=0 0 m=audio 32810 RTP/AVP 0 97 8 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 res_search: NO result ! res_search: NO result ! SipClient: Sending to '127.0.0.1:5060' SipClient: Receiving message... SipClient: Received: 11:22:24.556 --------------------------------- SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK5F746FD3 From: "Timur Elzhov" <sip:[EMAIL PROTECTED]>;tag=6873C9D3 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 7312 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 SipCall: Incoming response SipTransaction: Incoming Response SipCallMember: localStatusUpdated: 100 SipClient: Receiving message... SipClient: Received: 11:22:25.516 --------------------------------- SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK5F746FD3 From: "Timur Elzhov" <sip:[EMAIL PROTECTED]>;tag=6873C9D3 To: <sip:[EMAIL PROTECTED]>;tag=as4ee16e14 Call-ID: [EMAIL PROTECTED] CSeq: 7312 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 201 v=0 o=root 29731 29731 IN IP4 127.0.0.1 s=session c=IN IP4 127.0.0.1 t=0 0 m=audio 13998 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - SipCall: Incoming response SipCall: Checking for Contact and Record-Route SipCall: Setting Contact for this Call Member SipTransaction: Incoming Response SipClient: Sending: 11:22:25.523 -------------------------------- ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK5F746FD3 CSeq: 7312 ACK To: <sip:[EMAIL PROTECTED]>;tag=as4ee16e14 From: "Timur Elzhov" <sip:[EMAIL PROTECTED]>;tag=6873C9D3 Call-ID: [EMAIL PROTECTED] Content-Length: 0 User-Agent: kphone/4.1.1 Contact: "Timur Elzhov" <sip:[EMAIL PROTECTED]:5062;transport=udp> res_search: NO result ! res_search: NO result ! SipClient: Sending to '127.0.0.1:5060' SipCallMember: localStatusUpdated: 200 CallAudio: Using GSM for output CallAudio: Sending to remote site 127.0.0.1:13998 ERROR: Open Failed ** audioIn: openDevice Failed. CallAudio: Creating OSS->RTP Diverter dtmfsenderTimeout DspAudio: Broken pipe (b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b) ...
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