Hi Alexandre, Pierre-Luc, Initiated a call which is ringing on the other side but am not receiving audio. While dialling I do hear DTMF tones.
>From the log: <debug> Manager: Put call 153024602 on hold <debug> * SIP Info: Stopping AudioRTP for onhold action <info> RTP: Stopping audio rtp session <debug> RTP: Delete AudioRtpSession instance <info> Manager: Remove waiting call 153024602 (0 calls) <debug> ----- Switch current call id to ----- <debug> Manager: Selecting account Account:1286263870 <error> Manager: Error: Call id already exists in outgoing call however I did *not* put it on hold at the time the log said the call was being put on hold... this is something SFLPhone did. Once connected however, I put the call on hold, then took it off hold, and incoming audio starts. Hope this sheds some light on the problem. cheers, Jan Full log of session below: info> UserAgent: Transaction REQUEST received using transport: udp0x8393438 udp 10.10.14.6:5062 [published as 10.10.14.6:5062] (refcnt=1) <info> Manager : username = 92 , server = 10.10.14.6 <debug> Manager: Matching account id in request with username 92 <debug> UserAgent: The receiver is: [email protected] <debug> UserAgent: The callee account is Account:1286263870 <debug> UserAgent: Peer number: asterisk <debug> Audio: Start stream <debug> pcmp: default, index 1 <debug> pcmr: default, index 1 <debug> pcmc: default, index 1 <debug> Audio: Open playback device (and ringtone) <debug> Audio: Set audio rate to 8000 <debug> Audio: Open capture device <debug> Audio: Set audio rate to 8000 <debug> Audio: Prepare ALSA capture <debug> Audio: Prepare playback stream <debug> Audio: Start ALSA capture <debug> Audio: Start ALSA playback <debug> Audio: Start Audio Thread <debug> Audio: Start stream <debug> Audio: Start stream <debug> Audio: Start stream <debug> Audio: Start stream <debug> Manager: New outgoing call 153024602 to 6600 <debug> Number: Set phone number prefix <debug> Manager: Associate call 153024602 with config 0 <debug> Manager: Selecting account Account:1286263870 <debug> Manager: Adding Outgoing Call 153024602 on account Account:1286263870 <debug> Associate Call 153024602 with Account Account:1286263870 <debug> UserAgent: New outgoing call 153024602 to 6600 <info> AudioRecord: create filename for this call 20101024-21:38:44 <debug> SIPCall: Create new call 153024602 <debug> UserAgent: Local address for call: 10.10.14.6 <debug> Setting local ip address: 10.10.14.6 <debug> Setting local audio port to: 13258 <debug> Setting local audio port (external) to: 13258 <info> UserAgent: Creating new rtp session <debug> AudioRtpFactory: Init rtp session for account Account:1286263870 <debug> Registered account Account:1286263870 profile selected with key exchange protocol number 2 <debug> AudioRtpFactory: Srtp enable: 0 <info> Rtp: Local audio port 13258 will be used <debug> AudioSymmetricRtpSession initialized <debug> Starting a symmetric unencrypted rtp session <debug> AudioRecord: concatenate .wav file extension: name : 20101024-21:38:44 <debug> UserAgent: Try to make a call to: 6600 with call ID: 153024602 <info> SDP: Create initial offer <info> SDP: Create local offer <debug> SDP: Fetch local media capabilities. Local extern audio port: 13258 <debug> 112 <debug> 111 <debug> 110 <debug> 3 <debug> 9 <debug> 8 <debug> 0 <warning> No hash specified <debug> UserAgent: Start sip call <debug> SIP: Find local address from URI <debug> SIP: Local address discovered from attached transport: 10.10.14.6 <debug> UserAgent: transport ID: udp0x8393438 <debug> UserAgent: local port discovered from attached transport: 5062 <debug> Display Name: <debug> UserAgent: FROM uri: <sip:[email protected]:5060>, TO uri: <sip:[email protected]:5060>, CONTACT uri: <sip:[email protected]:5062> <debug> UserAgent: Call state changed to PJSIP_INV_STATE_CALLING <debug> UserAgent: Transaction changed to state PJSIP_TSX_STATE_CALLING <debug> UserAgent: Sent invite request using transport: udp0x8393438 udp 10.10.14.6:5062 [published as 10.10.14.6:5062] (refcnt=5) <debug> ----- Switch current call id to 153024602 ----- <debug> Manager: New outgoing call 153024602 to 6600 <debug> Number: Set phone number prefix <debug> Manager: Associate call 153024602 with config 0 <debug> Manager: Has current call (153024602) put it onhold <debug> Manager: Put call 153024602 on hold <debug> * SIP Info: Stopping AudioRTP for onhold action <info> RTP: Stopping audio rtp session <debug> RTP: Delete AudioRtpSession instance <info> Manager: Remove waiting call 153024602 (0 calls) <debug> ----- Switch current call id to ----- <debug> Manager: Selecting account Account:1286263870 <error> Manager: Error: Call id already exists in outgoing call <debug> UserAgent: Call state changed to PJSIP_INV_STATE_CALLING <debug> UserAgent: Transaction changed to state PJSIP_TSX_STATE_CALLING <debug> UserAgent: Transaction changed to state PJSIP_TSX_STATE_COMPLETED <debug> UserAgent: Transaction changed to state PJSIP_TSX_STATE_PROCEEDING <debug> UserAgent: Call state changed to PJSIP_INV_STATE_EARLY <debug> Manager: Peer call 153024602 ringing <debug> UserAgent: Transaction changed to state PJSIP_TSX_STATE_PROCEEDING <debug> UserAgent: Call state changed to PJSIP_INV_STATE_CONNECTING <debug> UserAgent: Call media update <info> SDP: Clean session media <info> SDP: Fetching media from sdp <info> SDP: Remote Audio Port from fetching SDP: 10084 <info> SDP: Remote IP from fetching SDP: 10.10.14.1 <info> RTP: Updating IP address <debug> SDP: Executing sdp line 579 - get_session_media() <error> UserAgent: Error: RTP: Error: _rtpSession was null when trying to start audio thread <debug> UserAgent: Call state changed to PJSIP_INV_STATE_CONFIRMED <info> UserAgent: SIP call answered <debug> UserAgent: Update call state , id = 153024602 <debug> Manager: Peer answered call 153024602 <debug> UserAgent: Transaction changed to state PJSIP_TSX_STATE_TERMINATED <debug> CallManager::hold received 153024602 <debug> Manager: Put call 153024602 on hold <debug> * SIP Info: Stopping AudioRTP for onhold action <info> RTP: Stopping audio rtp session <debug> RTP: Error: _rtpSession is null when trying to stop. Returning. <debug> UserAgent: Transaction changed to state PJSIP_TSX_STATE_CALLING <info> Manager: Remove waiting call 153024602 (0 calls) <debug> UserAgent: Call media update <info> SDP: Clean session media <info> SDP: Fetching media from sdp <info> SDP: Remote Audio Port from fetching SDP: 10084 <info> SDP: Remote IP from fetching SDP: 10.10.14.1 <info> RTP: Updating IP address <debug> SDP: Executing sdp line 579 - get_session_media() <error> UserAgent: Error: RTP: Error: _rtpSession was null when trying to start audio thread <debug> UserAgent: Transaction changed to state PJSIP_TSX_STATE_TERMINATED <debug> CallManager::unhold received 153024602 <debug> Manager: Put call 153024602 off hold <debug> Manager: Setting offhold, Account Account:1286263870, callid 153024602 <debug> AudioRtpFactory: Init rtp session for account Account:1286263870 <debug> Registered account Account:1286263870 profile selected with key exchange protocol number 2 <debug> AudioRtpFactory: Srtp enable: 0 <info> Rtp: Local audio port 13258 will be used <debug> AudioSymmetricRtpSession initialized <debug> Starting a symmetric unencrypted rtp session <debug> UserAgent: Transaction changed to state PJSIP_TSX_STATE_CALLING <debug> ----- Switch current call id to 153024602 ----- <debug> SDP: Executing sdp line 579 - get_session_media() <debug> UserAgent: Call media update <info> SDP: Clean session media <info> SDP: Fetching media from sdp <info> SDP: Remote Audio Port from fetching SDP: 10084 <info> SDP: Remote IP from fetching SDP: 10.10.14.1 <info> RTP: Updating IP address <warning> RTP: Could not remove previous destination <info> RTP: Setting IP address for the RTP session <info> RTP: New remote address for session: 10.10.14.1:10084 <debug> SDP: Executing sdp line 579 - get_session_media() <debug> Starting symmetric rtp thread <debug> RTP: Starting main thread <debug> RTP: Init codec payload 3 <debug> RTP: Codec sampling rate: 8000 <debug> RTP: Codec frame size: 160 <debug> RTP: RTP timestamp increment: 160 <debug> RTP: Setting static payload format <debug> Manager: Add audio stream 153024602 <debug> Manager: Add stream to call <debug> BufferManager: Bind calls 153024602, audiolayer_id <debug> BufferManager: Create ring buffer 153024602 <debug> BufferManager: Create ring buffer audiolayer_id <debug> MainBuffer: State info <debug> Call: 153024602 is bound to: audiolayer_id, <debug> Call: audiolayer_id is bound to: 153024602, <debug> Buffer: 153024602 as read pointer: audiolayer_id, <debug> Buffer: audiolayer_id as read pointer: 153024602, <debug> UserAgent: Transaction changed to state PJSIP_TSX_STATE_TERMINATED warning: The VAD has been replaced by a hack pending a complete rewrite <debug> Audio: Start stream <debug> RTP: Entering mainloop for call 153024602 <debug> CallManager: HangUp received 153024602 <info> Manager: Hangup call 153024602 <debug> Manager: Send DBUS call state change (HUNGUP) for id 153024602 <debug> ----- Switch current call id to ----- <debug> UserAgent: Transaction changed to state PJSIP_TSX_STATE_CALLING <debug> UserAgent: Terminate call 153024602 <debug> SIPCall: Delete call <info> RTP: Stopping audio rtp session <debug> RTP: Delete AudioRtpSession instance <debug> BufferManager: Unbind calls 153024602, audiolayer_id <debug> BufferManager: Remove Ringbuffer audiolayer_id <debug> BufferManager: Remove Ringbuffer 153024602 <debug> Manager: stop audio stream, ther is only 0 call(s) remaining <debug> Audio: Stop stream <debug> Audio: Stop audio thread <debug> Audio: Stop ALSA capture <debug> Audio: Close ALSA capture <debug> Audio: Stop ALSA playback <debug> Audio: Close ALSA playback <debug> UserAgent: Call state changed to PJSIP_INV_STATE_DISCONNECTED <error> UserAgent: Error: Call is NULL in call state changed callback <debug> UserAgent: Transaction changed to state PJSIP_TSX_STATE_COMPLETED <debug> UserAgent: Transaction changed to state PJSIP_TSX_STATE_TERMINATED _______________________________________________ SFLphone mailing list [email protected] http://lists.savoirfairelinux.net/mailman/listinfo/sflphone
