[Asterisk-Users] kphone & Asterisk CVS HEAD: no audio

[Timur V. Elzhov]

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

-------------- next part --------------
$ 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:1000 at 127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK5F746FD3
CSeq: 7312 INVITE
To: <sip:1000 at 127.0.0.1>
Content-Type: application/sdp
From: "Timur Elzhov" <sip:elzhov at 127.0.0.1>;tag=6873C9D3
Call-ID: 1062457919 at 127.0.0.1
Subject: sip:elzhov at 127.0.0.1
Content-Length: 222
User-Agent: kphone/4.1.1
Contact: "Timur Elzhov" <sip:elzhov at 127.0.0.1:5062;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:elzhov at 127.0.0.1>;tag=6873C9D3
To: <sip:1000 at 127.0.0.1>
Call-ID: 1062457919 at 127.0.0.1
CSeq: 7312 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:1000 at 127.0.0.1>
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:elzhov at 127.0.0.1>;tag=6873C9D3
To: <sip:1000 at 127.0.0.1>;tag=as4ee16e14
Call-ID: 1062457919 at 127.0.0.1
CSeq: 7312 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:1000 at 127.0.0.1>
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:1000 at 127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK5F746FD3
CSeq: 7312 ACK
To: <sip:1000 at 127.0.0.1>;tag=as4ee16e14
From: "Timur Elzhov" <sip:elzhov at 127.0.0.1>;tag=6873C9D3
Call-ID: 1062457919 at 127.0.0.1
Content-Length: 0
User-Agent: kphone/4.1.1
Contact: "Timur Elzhov" <sip:elzhov at 127.0.0.1:5062;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)
...