Hello all, I'm having a problem here, and don't have a clue what to do next.When calling from 8335 to any other number, I can hear the "ring" tone, but as the other side picks up the sflphoned die.
This happens when I pick up some call too.It was working normally, until last week. I wonder if there's anything that's server related, or can it be some updated library.
I've attached the sflphoned log to see if helps.I'm using version 0.9.6, build from ubuntu source package. I've made changes only to add the "User-Agent" header in every message ( needed by our corporate SIP server ).
Thanks! Thomas
[sfl-debug] Initialize manager in main [sfl-debug] Init config file /home/thomas/.sflphone/sflphonedrc 17:27:24.150 os_core_unix.c pjlib 1.0.3 for POSIX initialized [sfl-debug] UserAgent: Checking network, setting local IP address to: 10.80.76.188 [sfl-debug] UserAgent: UDP server listening on port 27060 [sfl-debug] Transport initialized successfully! [sfl-debug] UserAgent: SIP Init -- listening on port 27060 [sfl-debug] UserAgent: VOIP callbacks initialized [sfl-debug] UserAgent: pjsip version 1.0.3 for linux initialized [sfl-debug] Loading account Account:1256560157 [sfl-debug] nbAccount loaded 1 [sfl-init] Initiate Volume [sfl-init] AudioLayer Creation PID TTY TIME CMD 3867 ? 00:03:49 pulseaudio [sfl-debug] PulseLayer::Pulse audio constructor: Create context [sfl-debug] Audio layer type: 1 [sfl-debug] PulseLayer::openDevice [sfl-debug] PulseLayer::connectPulseAudioServer [sfl-debug] Connect the context to the server [sfl-debug] PulseLayer::context_state_callback ::The state of the context changed [sfl-debug] Waiting.... [sfl-debug] PulseLayer::context_state_callback ::The state of the context changed [sfl-debug] Waiting.... [sfl-debug] PulseLayer::context_state_callback ::The state of the context changed [sfl-debug] PulseLayer::createStreams [sfl-debug] Connection to PulseAudio server established [sfl-debug] Context creation done [sfl-debug] Connection Done!! [sfl-init] Active Codecs List [sfl-debug] Scanning /home/thomas/.sflphone/ to find audio codecs.... [sfl-debug] Scanning /usr/lib/sflphone/codecs/ to find audio codecs.... warning: The VAD has been replaced by a hack pending a complete rewrite warning: The VAD has been replaced by a hack pending a complete rewrite warning: The VAD has been replaced by a hack pending a complete rewrite [sfl-debug] speex [sfl-debug] speex [sfl-debug] PCMA [sfl-debug] speex [sfl-debug] PCMU [sfl-debug] G722 [sfl-debug] GSM [sfl-init] Load Telephone Tone [sfl-init] Loading DTMF key [sfl-init] Initiate VoIP Links Registration [sfl-debug] status : 305419896 [sfl-debug] status : 305419896 [sfl-debug] status : 0 [sfl-debug] 1 servers where obtained from name resolution. [sfl-debug] Account::setRegistrationState [sfl-debug] <sip:[email protected]:27060> [sfl-debug] Starting DBus event loop [sfl-debug] Account::setRegistrationState [sfl-debug] AudioStream::stream_state_callback :: The state of the stream changed [sfl-debug] Stream successfully created, connected to alsa_output.pci_8086_293e_sound_card_0_alsa_playback_0 [sfl-debug] AudioStream::stream_state_callback :: The state of the stream changed [sfl-debug] Stream successfully created, connected to alsa_input.pci_8086_293e_sound_card_0_alsa_capture_0 [sfl-debug] Instance::register received [sfl-debug] ConfigurationManager::getAccountList received [sfl-debug] ConfigurationManager::getAccountDetails [sfl-debug] ConfigurationManager::getCodecList received [sfl-debug] ConfigurationManager::getActiveCodecList received [sfl-debug] ManagerImpl::getActiveCodecList [sfl-debug] Codec with payload number 0 is active [sfl-debug] Codec with payload number 3 is active [sfl-debug] Codec with payload number 8 is active [sfl-debug] Codec with payload number 9 is active [sfl-debug] Codec with payload number 110 is active [sfl-debug] Codec with payload number 111 is active [sfl-debug] Codec with payload number 112 is active [sfl-debug] ConfigurationManager::getCodecDetails received [sfl-debug] ConfigurationManager::getCodecDetails received [sfl-debug] ConfigurationManager::getCodecDetails received [sfl-debug] ConfigurationManager::getCodecDetails received [sfl-debug] ConfigurationManager::getCodecDetails received [sfl-debug] ConfigurationManager::getCodecDetails received [sfl-debug] ConfigurationManager::getCodecDetails received [sfl-debug] ConfigurationManager::getCodecDetails received [sfl-debug] ConfigurationManager::getCodecDetails received [sfl-debug] ConfigurationManager::getCodecDetails received [sfl-debug] ConfigurationManager::getCodecDetails received [sfl-debug] ConfigurationManager::getCodecDetails received [sfl-debug] ConfigurationManager::getCodecDetails received [sfl-debug] ConfigurationManager::getCodecDetails received [sfl-debug] CallManager::getVolume received [sfl-debug] Current speaker = 100 [sfl-debug] CallManager::getVolume received [sfl-debug] Current mic = 100 [sfl-debug] Manager received isStartHidden [sfl-debug] PulseLayer::Start stream [sfl-debug] PulseLayer::Start stream [sfl-debug] PulseLayer::Start stream [sfl-debug] PulseLayer::Start stream [sfl-debug] CallManager::placeCall received [sfl-debug] ManagerImpl::outgoingCall() method [sfl-debug] Associate Call 319810123 with config 0 [sfl-debug] Current call ID = [sfl-debug] - Manager Action: Adding Outgoing Call 319810123 on account Account:1256560157 [sfl-debug] Associate Call 319810123 with Account Account:1256560157 AudioRecord::createFilename::filename for this call 20091109-17:28:00 [sfl-debug] SIPCALL::Constructor for this class is called [sfl-debug] Setting local audio port to: 25006 [sfl-debug] Setting local audio port (external) to: 25006 [sfl-debug] Creating new rtp session in newOutgoingCall [sfl-debug] AudioRtp::Create new rtp session [sfl-debug] 25006 AudioRecord::openFile::concatenate .wav file extension: name : 20091109-17:28:00 [sfl-debug] Try to make a call to: 8226 with call ID: 319810123 [sfl-debug] Create initial offer [sfl-debug] Create local offer [sfl-debug] Fetch local media capabilities. Local extern audio port: 25006 [sfl-debug] 0 [sfl-debug] 3 [sfl-debug] 8 [sfl-debug] 9 [sfl-debug] 110 [sfl-debug] 111 [sfl-debug] 112 [sfl-debug] Placing new call: To: sip:[email protected]:5060 From: sip:[email protected]:5060 [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_CALLING [sfl-debug] call_on_tsx_changed to state PJSIP_TSX_STATE_CALLING [sfl-debug] call_on_tsx_changed to state PJSIP_TSX_STATE_COMPLETED [sfl-debug] call_on_tsx_changed to state PJSIP_TSX_STATE_PROCEEDING [sfl-debug] call_on_state_changed to state PJSIP_INV_STATE_EARLY [sfl-debug] call_on_tsx_changed to state PJSIP_TSX_STATE_PROCEEDING [sfl-debug] call_on_state_changed to state PJSIP_INV_STATE_CONNECTING [sfl-debug] Got SDP answer Response msg 200/INVITE/cseq=26230 (rdata0x9022d3c) [sfl-debug] Negotiation returned with status 0 PJ_SUCCESS being 0 [sfl-debug] sipvoiplink.cpp (2905): on_rx_offer REINVITE [sfl-debug] Receiving initial offer [sfl-debug] Create local offer [sfl-debug] Fetch local media capabilities. Local extern audio port: 25006 [sfl-debug] 0 [sfl-debug] 3 [sfl-debug] 8 [sfl-debug] 9 [sfl-debug] 110 [sfl-debug] 111 [sfl-debug] 112 [sfl-debug] Fetching media from sdp [sfl-debug] Remote Audio Port from fetching SDP: 4342 [sfl-debug] Remote IP from fetching SDP: 10.80.133.15 [sfl-debug] AudioRtp::Stopping rtp session [sfl-debug] Delete AudioRtpRTX instance [sfl-debug] AudioRtpRTX instance deleted [sfl-debug] AudioRtp::Audio rtp stopped [sfl-debug] Create new rtp session from handle_reinvite [sfl-debug] AudioRtp::Create new rtp session [sfl-debug] 25006 [sfl-debug] call_on_media_update sflphoned: ../src/pjmedia/sdp.c:458: pjmedia_sdp_attr_to_rtpmap: Assertion `pool && attr && p_rtpmap' failed.
smime.p7s
Description: S/MIME Cryptographic Signature
_______________________________________________ SFLphone mailing list [email protected] http://lists.savoirfairelinux.net/mailman/listinfo/sflphone
