> Pierre-Luc Bacon pierre-luc.bacon at savoirfairelinux.com Thu Jan 28 21:50:00
> EST 2010
>
> If you don't hate it too much, open up a terminal window, and launch the
> "server" backend of SFLPhone with :
This is the sflphoned (version 0.9.6) output from my colleague Curtis
Gibson. The platform is Debian 5.0 ("lenny") i386. The SIP server is
OpenSER 1.3.2-notls (i386/linux).
Both SIP user accounts don't appear to have an issue while not on a call
with registration, which I assume would happen with the timeout problem
John Ervin mentioned.
I will look at testing 0.9.7 as soon as I can.
>From hostname3/~user4
[sfl-debug] 112
[sfl-debug] Placing new call:
To: sip:[email protected]
From: sip:[email protected]
[sfl-debug] call_on_state_changed to state PJSIP_INV_STATE_CALLING
[sfl-debug] call_on_tsx_changed to state PJSIP_TSX_STATE_CALLING
[sfl-debug] call_on_state_changed to state PJSIP_INV_STATE_DISCONNECTED
[sfl-debug] Invitation falled in state "disconnected".
[sfl-debug] State: 6, Disconnection cause: 404
[sfl-debug] Server error!
[sfl-debug] Manager received getNotify
[sfl-debug] CALL ID = 1421770287
[sfl-debug] SIPCALL::Destructor for this class is called
[sfl-debug] AudioRecord::isOpenFIle(): file not openend
[sfl-debug] AudioRtp::Stopping rtp session
[sfl-debug] Delete AudioRtpRTX instance
[sfl-debug] AudioRtpRTX instance deleted
[sfl-debug] AudioRtp::Audio rtp stopped
[sfl-debug] call_on_tsx_changed to state PJSIP_TSX_STATE_COMPLETED
[sfl-debug] CallManager::hangUp received
[sfl-debug] ManagerImpl::hangupCall()
[sfl-debug] Stop audio stream
[sfl-debug] hangupCall: callList is of size 0 call(s)
[sfl-debug] AlsaLayer::stopStream :: Stop ALSA streams
[sfl-debug] ! Manager Hangup Call: Call doesn't exists
[sfl-debug] call_on_tsx_changed to state PJSIP_TSX_STATE_TERMINATED
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