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[A\]@H=QEtvgH=8EHx]H=1EDtH=EHLf.H=D@WH=DH'H:@UHdL LH H HPHsPH PH PݞH HHGenerates DTMF tonesSource/AudioDTMF tone generatorInterval between tone packetsintervalgst_dtmf_src_change_stategst_dtmf_src_send_eventgst_dtmf_src_unlockgst_dtmf_src_unlock_stopgst_dtmf_src_handle_eventgst_dtmf_src_creategst_dtmf_src_negotiategst_dtmf_src_queryrtpdtmfdepay elementrtpdtmfdepayCodec/Depayloader/Network/RTPRTP DTMF packet depayloaderDuration unittimeunit-timeMaximum durationmax-durationgst_rtp_dtmf_depay_processgst_rtp_dtmf_depay_setcaps../gst/dtmf/gstdtmfsrc.cCalled unlockUnlock stopped../gst/dtmf/gstrtpdtmfsrc.cGstDTMFSrcGstRTPDTMFSrcinit donestartmethodlast-stopvolumeReceived stop eventdtmf-event../gst/dtmf/gstrtpdtmfdepay.cdtmf-event-droppedFlushing event queueparent failed state changeNo clock set for element %sdtmf-event-processedpushing pause_task...Creating buffer for tone %sGenerating silenceGenerating toneCreated buffer of size %luStopping current eventCould not get rateGST_IS_PAD (pad)rtpdtmfsrc elementrtpdtmfsrcGenerates RTP DTMF packetsSource/Network/RTPRTP DTMF packet generatorgst_rtp_dtmf_src_finalizegst_rtp_dtmf_src_set_propertygst_rtp_dtmf_src_get_propertyTimestamptimestampSequence numberseqnumTimestamp Offsettimestamp-offsetSequence number Offsetseqnum-offsetclockrateclock-rateSSRCssrcpayload typeptPacket Redundancypacket-redundancygst_rtp_dtmf_src_change_stategst_rtp_dtmf_src_unlockgst_rtp_dtmf_src_unlock_stopgst_rtp_dtmf_src_handle_eventgst_rtp_dtmf_src_creategst_rtp_dtmf_src_negotiategst_rtp_dtmf_src_queryencoding-nameaudiomediaapplication/x-rtpTELEPHONE-EVENTpayloadno peer caps: %pAusing peer pt %dusing internal pt %dusing clock-rate from caps %dusing existing clock-rate %dusing peer ssrc %08xusing internal ssrc %08xusing peer seqnum-offset %uusing peer ptime %umaxptimeusing internal ptime %uwith peer caps: %pAReceived event: %pAtry poppingtry popped %dNo available clockerror: %sPacket did not validatewarning: %sGstRtpDTMFDepay1.28.2DTMF pluginsdtmfgst-plugins-goodLGPL2026-04-07sinkDTMF_KEY_EVENT_0DTMF_KEY_EVENT_1DTMF_KEY_EVENT_2DTMF_KEY_EVENT_3DTMF_KEY_EVENT_4DTMF_KEY_EVENT_5DTMF_KEY_EVENT_6DTMF_KEY_EVENT_7DTMF_KEY_EVENT_8DTMF_KEY_EVENT_9DTMF_KEY_EVENT_SDTMF_KEY_EVENT_PDTMF_KEY_EVENT_ADTMF_KEY_EVENT_BDTMF_KEY_EVENT_CDTMF_KEY_EVENT_DYouness Alaoui Interval in ms between two tone packetsGenerates DTMF Sound from telephone-event RTP packetsgst_rtp_dtmf_depay_set_propertygst_rtp_dtmf_depay_get_propertyThe smallest unit (ms) the duration must be a multiple of (0 disables it)The maxumimum duration (ms) of the outgoing soundpacket. (0 = no limit)Pushing the PAUSE_TASK event on unlock requestdtmf-event, but not in PLAYING stateReceived start event %d with volume %dReceived an %s event via send_eventReceived an %s event on the src pad%s:%d: invalid %s id %u for "%s" of type '%s' in '%s'Received a DTMF stop event when already stoppedReceived two consecutive DTMF start eventsend event check, now wait for the proper timeCreating new buffer with event %u duration gst: %u:%02u:%02u.%09u at %u:%02u:%02u.%09ucaps != NULL && gst_caps_is_fixed (caps)Reporting latency of %u:%02u:%02u.%09uZeeshan Ali The RTP timestamp of the last processed packetThe RTP sequence number of the last processed packetOffset to add to all outgoing timestamps (-1 = random)Offset to add to all outgoing seqnum (-1 = random)The clock-rate at which to generate the dtmf packetsThe SSRC of the packets (-1 == random)The payload type of the packetsNumber of packets to send to indicate start and stop dtmf eventsCould not get intersection with peer capsIntersection with peer caps is emptyusing peer timestamp-offset %uusing internal timestamp-offset %uusing internal seqnum-offset %uusing peer maxptime as ptime %uReceived %s event on the src padReceived event while not in PLAYING statenew_bit_size && ((new_bit_size & __GST_BITS_WRITER_ALIGNMENT_MASK) == 0)/usr/include/gstreamer-1.0/gst/base/gstbitwriter.hbitwriter->bit_size <= bitwriter->bit_capacitybit_offset < 8 && bitwriter->bit_size <= bitwriter->bit_capacitycur_byte <= (bitwriter->data + (bitwriter->bit_capacity >> 3))Creating new buffer with event %u duration gst: %u:%02u:%02u.%09u at %u:%02u:%02u.%09u(rtp ts:%u dur:%u)Processed events, now lets wait on the clockReceived new RTP DTMF packet : marker=%d - timestamp=%u - event=%d - duration=%dPrevious information : timestamp=%u - duration=%dUnable to send dtmf-event message to busUnable to create dtmf-event messageUnable to create dtmf-event structurenew previous duration : %d - new duration : %d - diff : %d - clock rate : %d - timestamp : %lutimestamp : %lu - time %u:%02u:%02u.%09uhttps://launchpad.net/ubuntu/+source/gst-plugins-good1.0GStreamer Good Plugins (Ubuntu)application/x-rtp, media = (string) "audio", payload = (int) [96, 127], clock-rate = (int) [ 0, MAX ], encoding-name = (string) "TELEPHONE-EVENT"audio/x-raw, format = (string) S16LE, rate = (int) [ 1, max ], channels = (int) 1, layout = (string) interleavedapplication/x-rtp, media = (string) "audio", payload = (int) [ 96, 127 ], clock-rate = (int) [ 0, MAX ], encoding-name = (string) "TELEPHONE-EVENT"audio/x-raw, format = (string) "S16LE", rate = (int) [ 1, max ], channels = (int) 1, layout = (string)interleaved(@Xx+̴ܴ @kDD@.D D@.DD@.DD@D D@DD@DDUD DUDDUDD@kD D@kDD@.D D@D DUD D@kD Dgst_rtp_dtmf_depay_processgst_pad_set_capsgst_rtp_dtmf_src_unlock_stopgst_rtp_dtmf_src_unlockgst_rtp_dtmf_src_change_stategst_rtp_dtmf_src_querygst_pad_set_capsgst_rtp_dtmf_src_negotiategst_rtp_dtmf_src_creategst_bit_writer_put_bits_uint8_unchecked_gst_bit_writer_check_remaining?gst_bit_writer_put_bits_uint16_uncheckedgst_rtp_dtmf_src_create_next_rtp_packetgst_rtp_dtmf_src_handle_dtmf_eventgst_rtp_dtmf_src_handle_custom_upstreamgst_rtp_dtmf_src_handle_eventgst_rtp_dtmf_src_initgst_dtmf_src_change_stategst_dtmf_src_querygst_pad_set_capsgst_dtmf_src_negotiategst_dtmf_src_unlock_stopgst_dtmf_src_unlockgst_dtmf_prepare_timestampsgst_dtmf_src_create_next_tone_packetgst_dtmf_src_creategst_dtmf_src_send_eventgst_dtmf_src_handle_eventgst_dtmf_src_handle_dtmf_eventgst_dtmf_src_init`$@-DT!@?@?zDA>;/t{{40XTxԈ$$DDldT$DT|D(TĘt,T|<ԺhTT L|4TdzRx $r@FJ w?9*3$"Dz\z0t$.EC A K $EC Ai F XTC k ԅXTC k , AVC R $LPEJ Cm A $t(EJ Cq E $EJ Cm A $hEJ Cq E ЈMAJ E|  MAJ E| $,0EH A^ D } $TEH A} E } $|`AG CT A ,(EJ BDB I q O $EF Av F y ,EC BJd H Z (,JC DD` K,XdEC Ah G c E S SC h B(tAC Aj Q E K @HEC FDI E v J  B ^ B ,$AC AZ Q G I f ,H EC DO F 0xEC BJ D u C $@aC BHH J $8aC BHH J $0EC A H H$3EC Ax G c E P H P H Q G Z F Z pC h (rEC I E (P< Ej K C $dEC CN G %A_ ,,з_ AL I G ,\cEG BN H <@IEC EM F q G  G \ D ,P9 EH DNR  G `MAJ E| $EC BD} E D XYEp c GNUp~FDO{"type":"deb","os":"ubuntu","name":"gst-plugins-good1.0","version":"1.28.2-2ubuntu0.1","architecture":"amd64"}p?0?ާק`@kDD'@.D D8@.DDI@.DDZ@D Dk@DD|@DDUD DUDD UDD @kD DѨ @kDD @.D D @D DUD D@kD DNk        0 Lo0     o oooo%00@0P0`0p00000000011 101@1P1`1p11111111122 202@2P2`2p22222222233 303@3P3`3p33333333344 404@4P4`4p44444444455 505@5P5`5p55555555566 606@6P6`6p66666666677 707@7P7Xдh/usr/lib/debug/.dwz/x86_64-linux-gnu/gstreamer1.0-plugins-good.debugb `>OkvHi.2f7fd922f48002f7dd71fad968eb575e02504d.debugd8.shstrtab.note.gnu.build-id.gnu.hash.dynsym.dynstr.gnu.version.gnu.version_r.rela.dyn.rela.plt.init.plt.got.plt.sec.text.fini.rodata.eh_frame_hdr.eh_frame.note.gnu.property.note.package.init_array.fini_array.data.rel.ro.dynamic.data.bss.gnu_debugaltlink.gnu_debuglink $o(( 00 0 8oEo`T  ^B h00c 0 0@n`7`7wp7p70>>^@LL  x  Pr `Y40'