[root@ast asterisk]# asterisk -cvvvdg DEBUG[-1085173632]: File config.c, Line 712 (__ast_load): Parsing /etc/asterisk/asterisk.conf Asterisk CVS-01/11/04-00:00:30, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer ========================================================================= DEBUG[-1085173632]: File config.c, Line 712 (__ast_load): Parsing /etc/asterisk/logger.conf Asterisk Event Logger Started /var/log/asterisk/event_log == Manager registered action Ping == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Redirect == Manager registered action Originate == Manager registered action MailboxStatus == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxCount == RTP Allocating from port range 10000 -> 20000 Asterisk PBX Core Initializing Registering builtin applications: .... Asterisk Dynamic Loader Starting: [chan_modem.so] => (Generic Voice Modem Driver) == Loading modem driver chan_modem_aopen.so => (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver) [res_musiconhold.so] => (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' [res_adsi.so] => (ADSI Resource) [res_parking.so] => (Call Parking Resource) == Registered application 'ParkedCall' [res_crypto.so] => (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'iaxtel' [res_indications.so] => (Indications Configuration) -- Registered indication country 'us' ... -- Setting default indication country to 'us' == Registered application 'Playtones' == Registered application 'StopPlaytones' [res_monitor.so] => (Call Monitoring Resource) == Registered application 'Monitor' == Registered application 'StopMonitor' == Registered application 'ChangeMonitor' == Manager registered action Monitor == Manager registered action StopMonitor == Manager registered action ChangeMonitor [chan_iax.so] => (Inter Asterisk eXchange) == Manager registered action IAXpeers == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 5036 [chan_sip.so] => (Session Initiation Protocol (SIP)) == SIP Listening on xx.xx.33.19:5060 == Using TOS bits 0 == Registered application 'SIPDtmfMode' Warning, flexibel rate not heavily tested! [chan_modem_bestdata.so] => (BestData (Conexant V.90 Chipset) VoiceModem Driver) [chan_modem_i4l.so] => (ISDN4Linux Emulated Modem Driver) [chan_agent.so] => (Agent Proxy Channel) == Registered application 'AgentLogin' == Registered application 'AgentCallbackLogin' [chan_mgcp.so] => (Media Gateway Control Protocol (MGCP)) == MGCP Listening on 0.0.0.0:2427 == Using TOS bits 0 [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers WARNING[-1085173632]: File chan_iax2.c, Line 5470 (set_config): Ignoring port for now == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 4569 [chan_local.so] => (Local Proxy Channel) [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) -- Setting mailbox '500' on cisco7910@5500 -- Allocating Skinny subchannel '0' on 5500@cisco7910 -- Allocating Skinny subchannel '1' on 5500@cisco7910 -- Added device 'cisco7910' == Skinny listening on xx.xx.33.19:2000 [chan_oss.so] => (OSS Console Channel Driver) WARNING[-1085173632]: File chan_oss.c, Line 352 (setformat): Requested 8000 Hz, got 8184 Hz -- sound may be choppy WARNING[-1085173632]: File chan_oss.c, Line 980 (load_module): XXX I don't work right with non-full duplex sound cards XXX WARNING[-1190081616]: File chan_oss.c, Line 238 (sound_thread): Read error on sound device: Resource temporarily unavailable [chan_phone.so] => (Linux Telephony API Support) [chan_zap.so] => (Zapata Telephony w/PRI) WARNING[-1085173632]: File chan_zap.c, Line 7345 (setup_zap): Ignoring rxwink == Registered application 'CallingPres' == Manager registered action ZapTransfer == Manager registered action ZapHangup == Manager registered action ZapDialOffhook [pbx_config.so] => (Text Extension Configuration) -- Including context 'demo' in context 'default' -- Including context 'siproute' in context 'default' -- Including context 'parkedcalls' in context 'default' [pbx_wilcalu.so] => (Wil Cal U (Auto Dialer)) [pbx_spool.so] => (Outgoing Spool Support) /var/spool/asterisk/outgoing [app_dial.so] => (Dialing Application) == Registered application 'Dial' [app_playback.so] => (Trivial Playback Application) == Registered application 'Playback' [app_voicemail.so] => (Comedian Mail (Voicemail System)) == Registered application 'VoiceMail' == Registered application 'VoiceMail2' == Registered application 'VoiceMailMain' == Registered application 'VoiceMailMain2' [app_directory.so] => (Extension Directory) == Registered application 'Directory' [app_mp3.so] => (Silly MP3 Application) == Registered application 'MP3Player' [app_system.so] => (Generic System() application) == Registered application 'System' [app_echo.so] => (Simple Echo Application) == Registered application 'Echo' [app_record.so] => (Trivial Record Application) == Registered application 'Record' [app_image.so] => (Image Transmission Application) == Registered application 'SendImage' [app_url.so] => (Send URL Applications) == Registered application 'SendURL' [app_disa.so] => (DISA (Direct Inward System Access) Application) == Registered application 'DISA' [app_agi.so] => (Asterisk Gateway Interface (AGI)) == Registered application 'EAGI' == Registered application 'AGI' [app_qcall.so] => (Call from Queue) [app_adsiprog.so] => (Asterisk ADSI Programming Application) == Registered application 'ADSIProg' [app_getcpeid.so] => (Get ADSI CPE ID) == Registered application 'GetCPEID' [app_milliwatt.so] => (Digital Milliwatt (mu-law) Test Application) == Registered application 'Milliwatt' [app_zapateller.so] => (Block Telemarketers with Special Information Tone) == Registered application 'Zapateller' [app_datetime.so] => (Date and Time) == Registered application 'DateTime' [app_setcallerid.so] => (Set CallerID Application) == Registered application 'SetCallerID' [app_festival.so] => (Simple Festival Interface) == Registered application 'Festival' [app_queue.so] => (True Call Queueing) == Registered application 'Queue' == Manager registered action Queues == Manager registered action QueueStatus == Registered application 'AddQueueMember' == Registered application 'RemoveQueueMember' [app_senddtmf.so] => (Send DTMF digits Application) == Registered application 'SendDTMF' [app_parkandannounce.so] => (Call Parking and Announce Application) == Registered application 'ParkAndAnnounce' [app_striplsd.so] => (Strip trailing digits) == Registered application 'StripLSD' [app_setcidname.so] => (Set CallerID Name) == Registered application 'SetCIDName' [app_lookupcidname.so] => (Look up CallerID Name from local database) == Registered application 'LookupCIDName' [app_substring.so] => (Save substring digits in a given variable) == Registered application 'SubString' [app_macro.so] => (Extension Macros) == Registered application 'Macro' [app_authenticate.so] => (Authentication Application) == Registered application 'Authenticate' [app_softhangup.so] => (Hangs up the requested channel) == Registered application 'SoftHangup' [app_lookupblacklist.so] => (Look up Caller*ID name/number from blacklist database) == Registered application 'LookupBlacklist' [app_waitforring.so] => (Waits until first ring after time) == Registered application 'WaitForRing' [app_privacy.so] => (Require phone number to be entered, if no CallerID sent) == Registered application 'PrivacyManager' [app_db.so] => (Database access functions for Asterisk extension logic) == Registered application 'DBget' == Registered application 'DBput' == Registered application 'DBdel' == Registered application 'DBdeltree' [app_chanisavail.so] => (Check if channel is available) == Registered application 'ChanIsAvail' [app_enumlookup.so] => (ENUM Lookup) == Registered application 'EnumLookup' [app_transfer.so] => (Transfer) == Registered application 'Transfer' [app_setcidnum.so] => (Set CallerID Number) == Registered application 'SetCIDNum' [app_cdr.so] => (Make sure asterisk doesn't save CDR for a certain call) == Registered application 'NoCDR' [app_hasnewvoicemail.so] => (Indicator for whether a voice mailbox has new messages.) == Registered application 'HasNewVoicemail' [app_sayunixtime.so] => (Say time) == Registered application 'SayUnixTime' [app_cut.so] => (Cuts up variables) == Registered application 'Cut' [app_read.so] => (Read Variable Application) == Registered application 'Read' [skipping app_intercom.so] [app_zapras.so] => (Zap RAS Application) == Registered application 'ZapRAS' [app_meetme.so] => (Simple MeetMe conference bridge) == Registered application 'MeetMeCount' == Registered application 'MeetMe' [app_flash.so] => (Flash zap trunk application) == Registered application 'Flash' [app_zapbarge.so] => (Barge in on Zap channel application) == Registered application 'ZapBarge' [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) == Registered translator 'ilbctolin' from format ILBC to SLINR, cost 6 == Registered translator 'lintoilbc' from format SLINR to ILBC, cost 38 [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator) == Registered translator 'gsmtolin' from format GSM to SLINR, cost 2 == Registered translator 'lintogsm' from format SLINR to GSM, cost 4 [codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder) == Registered translator 'lpc10tolin' from format LPC10 to SLINR, cost 5 == Registered translator 'lintolpc10' from format SLINR to LPC10, cost 7 [codec_adpcm.so] => (Adaptive Differential PCM Coder/Decoder) == Registered translator 'adpcmtolin' from format ADPCM to SLINR, cost 1 == Registered translator 'lintoadpcm' from format SLINR to ADPCM, cost 1 [codec_ulaw.so] => (Mu-law Coder/Decoder) == Registered translator 'ulawtolin' from format ULAW to SLINR, cost 1 == Registered translator 'lintoulaw' from format SLINR to ULAW, cost 1 [codec_alaw.so] => (A-law Coder/Decoder) == Registered translator 'alawtolin' from format ALAW to SLINR, cost 1 == Registered translator 'lintoalaw' from format SLINR to ALAW, cost 1 [codec_a_mu.so] => (A-law and Mulaw direct Coder/Decoder) == Registered translator 'alawtoulaw' from format ALAW to ULAW, cost 1 == Registered translator 'ulawtoalaw' from format ULAW to ALAW, cost 1 [format_gsm.so] => (Raw GSM data) == Registered file format gsm, extension(s) gsm [format_wav.so] => (Microsoft WAV format (8000hz Signed Linear)) == Registered file format wav, extension(s) wav [format_wav_gsm.so] => (Microsoft WAV format (Proprietary GSM)) == Registered file format wav49, extension(s) WAV [format_vox.so] => (Dialogic VOX (ADPCM) File Format) == Registered file format vox, extension(s) vox [format_pcm.so] => (Raw uLaw 8khz Audio support (PCM)) == Registered file format pcm, extension(s) pcm|ulaw|ul|mu [format_g729.so] => (Raw G729 data) == Registered file format g729, extension(s) g729 [format_pcm_alaw.so] => (Raw aLaw 8khz PCM Audio support) == Registered file format alaw, extension(s) alaw|al [format_h263.so] => (Raw h263 data) == Registered file format h263, extension(s) h263 [format_jpeg.so] => (JPEG (Joint Picture Experts Group) Image Format) == Registered format 'jpg' (JPEG (Joint Picture Experts Group)) [cdr_csv.so] => (Comma Separated Values CDR Backend) [chan_oh323.so] => (OpenH323 Channel Driver) 0:00.005 OpenH323 Wrapper OpenH323 Wrapper Version 0.0alpha0 by inAccess Networks (www.inaccessnetworks.com) on Unix Linux (2.4.22-1.2115.nptl-i686) at 2004/1/11 0:24:22.353 WrapMutex::WrapMutex: Created mutex answerMutex WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323 v1.12.2, PWlib v1.5.2 Wrapper API::h323_end_point_create: Endpoint created. Wrapper API::h323_set_options: Setting endpoint options. Wrapper API::h323_set_options: FastStart option. Wrapper API::h323_set_options: H245Tunnelling option. Wrapper API::h323_set_options: H245InSetup option. Wrapper API::h323_set_options: Jitter options. Wrapper API::h323_set_options: RTP IP TOS option. Wrapper API::h323_set_ports: Setting endpoint port ranges. Wrapper API::h323_start_listener: Started listener Listener[ip$xx.xx.33.19:1720] Wrapper API::h323_set_gk: Configuring gatekeeper. WrapH323EndPoint::SetGatekeeperTimeToLive: Gatekeeper registration TTL set at 600 sec 0:00.007 H323 Listener:85961f0 H323 Awaiting TCP connections on port 1720 Wrapper API::h323_set_gk: Setting gatekeeper... 0:00.010 OpenH323 Wrapper RAS Authenticator H235AnnexD_Procedure1 not active during GRQ SetCapability negotiation 0:00.010 OpenH323 Wrapper RAS Authenticator MD5 not active during GRQ SetCapability negotiation 0:00.010 OpenH323 Wrapper RAS Authenticator CAT not active during GRQ SetCapability negotiation 0:00.018 OpenH323 Wrapper RAS Gatekeeper discovery found ip$xx.xx.33.7:1719 0:00.018 OpenH323 Wrapper RAS Gatekeeper discovered at: xx.xx.33.7:1719 (if=xx.xx.33.19:10001) 0:00.021 Transactor:859c980 Trans Starting listener thread on Transport[remote=ip$xx.xx.33.7:1719 if=ip$xx.xx.33.19:10001] Wrapper API::h323_set_gk: Gatekeeper found. Wrapper API::h323_set_capability: Inserted capability G.711-ALaw-64k{hw} Wrapper API::h323_set_capability: Inserted capability GSM-06.10{hw} Wrapper API::h323_set_senduimode: User-input mode set. Wrapper API::h323_callback_register: Callback functions installed. == OpenH323 Channel Ready (v0.5.6) [cdr_odbc.so] => (ODBC CDR Backend) [cdr_pgsql.so] => (PostgreSQL CDR Backend) Asterisk Ready. *CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-ccard.agi Urgent handler -- Playing 'ru_enter_card_num' (language 'en') Urgent handler agi-ccard.agi: Check amount for pin: 12345 Urgent handler agi-ccard.agi: Setting call allowable time to: 4980 -- AGI Script Executing Application: (SetAccount) Options: (12345) -- AGI Script Executing Application: (SayDigits) Options: (83) -- Playing 'digits/8' (language 'en') Urgent handler -- Playing 'digits/3' (language 'en') Urgent handler -- AGI Script Executing Application: (Wait) Options: (1) Urgent handler (null): Entered destination - target: 4444 and len equal to: length(4444) Urgent handler -- AGI Script Executing Application: (AbsoluteTimeout) Options: (4980) -- Set Absolute Timeout to 4980 Urgent handler -- AGI Script Executing Application: (Dial) Options: (OH323/380442304444|60) Wrapper API::h323_make_call: Making call. WrapMutex::Wait: Requesting mutex callMutex [wrapendpoint.cxx, 250, MakeCall] WrapMutex::Wait: Got mutex callMutex [wrapendpoint.cxx, 250, MakeCall] WrapH323EndPoint::MakeCall: Making call to 380442304444 0:27.541 Urgent handler ThreadID=0xb2301bb0 H323 Making call to: 380442304444 WrapH323EndPoint::CreateConnection: Creating a H323Connection [5210] WrapH323Connection::WrapH323Connection: Outgoing call with capability GSM-06.10{hw} WrapH323Connection::WrapH323Connection: Caller ID name on outgoing call sip-proxy WrapH323Connection::WrapH323Connection: LocalPartyName sip-proxy WrapH323Connection::WrapH323Connection: DestExtraCallInfo WrapH323Connection::WrapH323Connection: Caller ID on outgoing call alex WrapH323Connection::WrapH323Connection: WrapH323Connection created. WrapH323EndPoint::MakeCall: Call token is ip$localhost/5210 WrapH323EndPoint::MakeCall: Call reference is 5210 WrapMutex::Signal: Released mutex callMutex [wrapendpoint.cxx, 301, MakeCall] 0:27.573 H225 Caller:859d280 H323TCP Started connection: host=xx.xx.33.12:1720, if=xx.xx.33.19:10000, handle=52 0:27.580 H225 Caller:859d280 H245 Default OnSelectLogicalChannels, FastStartInitiate 0:27.588 H225 Caller:859d280 RTP_UDP Session 1 created: xx.xx.33.19:10002-10003 ssrc=1868486511 0:27.593 H225 Caller:859d280 RTP Adding session RTP_UDP WrapH323Connection::OnSendSignalSetup: Sending SETUP message... WrapH323Connection::OnSendSignalSetup: Setting display name sip-proxy WrapH323Connection::OnSendSignalSetup: Setting calling party number alex 0:27.602 H225 Caller:859d280 H225 Reading PDUs: callRef=5210 0:27.617 H225 Caller:859d280 H225 Set remote party name: "xx.xx.33.12" 0:27.620 H225 Caller:859d280 H225 Set remote application name: "Cisco IOS 12.2 181/18" WrapMutex::Wait: Requesting mutex channelMutex [wrapendpoint.cxx, 615, OpenAudioChannel] WrapMutex::Wait: Got mutex channelMutex [wrapendpoint.cxx, 615, OpenAudioChannel] WrapH323EndPoint::OpenAudioChannel: Direction => RECODER, Buffer => 320 WrapH323EndPoint::OpenAudioChannel: Media format: FrameSize 33, FrameTime 160, TimeUnits 8 WrapH323EndPoint::OpenAudioChannel: Codec info: FrameRate 160 WrapH323EndPoint::OpenAudioChannel: Packet size: 33 WrapH323EndPoint::OpenAudioChannel: Frames per packet: 1 WrapH323EndPoint::OpenAudioChannel: LID Codec GSM-06.10 WrapH323EndPoint::OpenAudioChannel: The sound channel is audiosocket:in0(fd=46) WrapH323EndPoint::OpenAudioChannel: The audio device name is audiosocket:in0 PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. PAsteriskSoundChannel::Open: os_handle 46, mediaFormat 3, frameTime 20 ms, frameNum 1, packetSize 33 WrapH323EndPoint::OpenAudioChannel: Opened sound channel "audiosocket:in0" for recording using 1x33 byte buffers. WrapMutex::Signal: Released mutex channelMutex [wrapendpoint.cxx, 729, OpenAudioChannel] 0:27.626 H225 Caller:859d280 H323 Started sending logical channel: GSM-06.10{hw} <1> WrapH323EndPoint::OnStartLogicalChannel: Started logical channel [5210] : sending GSM-06.10{hw} WrapMutex::Wait: Requesting mutex channelMutex [wrapendpoint.cxx, 794, OnStartLogicalChannel] WrapMutex::Wait: Got mutex channelMutex [wrapendpoint.cxx, 794, OnStartLogicalChannel] WrapH323EndPoint::OnStartLogicalChannel: TxFrames = 4 WrapH323EndPoint::OnStartLogicalChannel: channelsOpen = 1 WrapMutex::Signal: Released mutex channelMutex [wrapendpoint.cxx, 810, OnStartLogicalChannel] WrapMutex::Wait: Requesting mutex channelMutex [wrapendpoint.cxx, 615, OpenAudioChannel] WrapMutex::Wait: Got mutex channelMutex [wrapendpoint.cxx, 615, OpenAudioChannel] WrapH323EndPoint::OpenAudioChannel: Direction => PLAYER, Buffer => 320 WrapH323EndPoint::OpenAudioChannel: Media format: FrameSize 33, FrameTime 160, TimeUnits 8 WrapH323EndPoint::OpenAudioChannel: Codec info: FrameRate 160 WrapH323EndPoint::OpenAudioChannel: Packet size: 33 WrapH323EndPoint::OpenAudioChannel: Frames per packet: 1 WrapH323EndPoint::OpenAudioChannel: LID Codec GSM-06.10 WrapH323EndPoint::OpenAudioChannel: The sound channel is audiosocket:out0(fd=43) WrapH323EndPoint::OpenAudioChannel: The audio device name is audiosocket:out0 PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. PAsteriskSoundChannel::Open: os_handle 43, mediaFormat 3, frameTime 20 ms, frameNum 1, packetSize 33 WrapH323EndPoint::OpenAudioChannel: Opened sound channel "audiosocket:out0" for playing using 1x33 byte buffers. 0:27.633 WrapMutex::Signal: Released mutex channelMutex [wrapendpoint.cxx, 729, OpenAudioChannel] LogChanTx:85cb8a0 H323RTP Transmit GSM-06.10 thread started: rate=160 time=20ms size=4*33=132 0:27.636 H225 Caller:859d280 H323 Started receiving logical channel: GSM-06.10{hw} <1> WrapH323EndPoint::OnStartLogicalChannel: Started logical channel [5210] : receiving GSM-06.10{hw} WrapMutex::Wait: Requesting mutex channelMutex [wrapendpoint.cxx, 815, OnStartLogicalChannel] WrapMutex::Wait: Got mutex channelMutex [wrapendpoint.cxx, 815, OnStartLogicalChannel] WrapH323EndPoint::OnStartLogicalChannel: RxFrames = 4 WrapH323EndPoint::OnStartLogicalChannel: channelsOpen = 2 WrapMutex::Signal: Released mutex channelMutex [wrapendpoint.cxx, 831, OnStartLogicalChannel] 0:27.639 LogChanRx:85cbf20 H323RTP Receive GSM-06.10 thread started. 0:27.641 LogChanRx:85cbf20 RTP Jitter buffer created: size=21 delay=160-800/373 (46ms) obj=0x85b56a0 0:27.643 H225 Caller:859d280 H225 Fast starting 2 channels -- Called 380442304444 Urgent handler -- H323:5210 answered SIP/-085ac998 Urgent handler 0:27.897 H225 Caller:859d280 H225 Set remote party name: "xx.xx.33.12" 0:27.898 H225 Caller:859d280 H225 Set remote application name: "Cisco IOS 12.2 181/18" WrapH323Connection::OnAlerting: Ringing phone for "xx.xx.33.12" ... WARNING[-1315963984]: File chan_oh323.c, Line 2674 (alerted_h323_connection): Call with reference 5210 in unexpected state (4). 0:28.079 RTP Jitter:85b56a0 RTP First data: ver=2 pt=GSM psz=33 m=1 x=0 seq=35078 ts=1455258334 src=248979724 ccnt=0 0:28.208 H225 Caller:859d280 H225 Set remote party name: "xx.xx.33.12" 0:28.210 H225 Caller:859d280 H225 Set remote application name: "Cisco IOS 12.2 181/18" 0:28.212 H225 Caller:859d280 H225 Received connect PDU. 0:28.213 H225 Caller:859d280 H225 No H245 address provided by remote, starting control channel WrapH323Connection::OnEstablished: WrapH323Connection [ip$localhost/5210] established (FastStartAcknowledged/noH245Tunneling) WrapH323Connection::OnReceivedFacility: Received FACILITY message... 0:28.220 H225 Caller:859d280 H225 Simultaneous start of H.245 channel, using local listener. 0:30.124 RTP Jitter:85b56a0 RTP Receive statistics: packets=103 octets=3399 lost=0 tooLate=0 order=0 avgTime=20 maxTime=23 minTime=14 jitter=1 maxJitter=2 0:32.122 RTP Jitter:85b56a0 RTP Receive statistics: packets=203 octets=6699 lost=0 tooLate=0 order=0 avgTime=19 maxTime=22 minTime=15 jitter=2 maxJitter=2 0:34.119 RTP Jitter:85b56a0 RTP Receive statistics: packets=303 octets=9999 lost=0 tooLate=0 order=0 avgTime=19 maxTime=23 minTime=14 jitter=2 maxJitter=2 0:36.122 RTP Jitter:85b56a0 RTP Receive statistics: packets=403 octets=13299 lost=0 tooLate=0 order=0 avgTime=20 maxTime=26 minTime=14 jitter=2 maxJitter=3 Ouch ... error while writing audio data: : Broken pipe Segmentation fault (core dumped) ---------------------------------------------------------------------------------------------- [root@ast asterisk]# ls core.* core.6102 [root@ast asterisk]# gdb /usr/sbin/asterisk core.6102 GNU gdb Red Hat Linux (5.3.90-0.20030710.41rh) Using host libthread_db library "/lib/tls/libthread_db.so.1". Core was generated by `asterisk -cvvvdg'. .... #0 ast_smoother_feed (s=0xcbbe0380, f=0x85ae5e8) at frame.c:72 72 if (!s->format) { (gdb) bt #0 ast_smoother_feed (s=0xcbbe0380, f=0x85ae5e8) at frame.c:72 #1 0x0065a0f3 in oh323_write (c=0x85b26e0, f=0x85ae5e8) at chan_oh323.c:1504 #2 0x08059724 in ast_write (chan=0x85b26e0, fr=0x85ae5e8) at channel.c:1392 #3 0x0805c1d1 in ast_channel_bridge (c0=0x85b0ed8, c1=0x85b26e0, flags=0, fo=0xb23000c8, rc=0xb23000cc) at channel.c:2312 #4 0x00112daa in ast_bridge_call (chan=0x85b0ed8, peer=0x85b26e0, allowredirect_in=0, allowredirect_out=0, allowdisconnect=0) at res_parking.c:224 #5 0x00bcafad in dial_exec (chan=0x85b0ed8, data=0x0) at app_dial.c:676 #6 0x08063f22 in pbx_exec (c=0x85b0ed8, app=0x8572e98, data=0xb2300b3a, newstack=1) at pbx.c:396 #7 0x0079d731 in handle_exec (chan=0x8551570, agi=0x8551570, argc=3, argv=0x85b0ed8) at app_agi.c:670 #8 0x0079f62d in agi_handle_command (chan=0x8551570, agi=0xb23013a0, buf=0x8551570 "*") at app_agi.c:1210 #9 0x0079f382 in run_agi (chan=0x85b0ed8, request=0xb23013b0 "agi-ccard.agi", agi=0xb23013a0, pid=6128) at app_agi.c:1282 #10 0x0079ea7d in agi_exec_full (chan=0x85b0ed8, data=0x8551570, enhanced=0) at app_agi.c:1424 #11 0x0079e6ae in agi_exec (chan=0x8551570, data=0x8551570) at app_agi.c:1436 #12 0x08063f22 in pbx_exec (c=0x85b0ed8, app=0x857e280, data=0xb2301710, newstack=1) at pbx.c:396 #13 0x0806b676 in pbx_extension_helper (c=0x85b0ed8, context=0x85b1030 "default", exten=0x85b1124 "1113", priority=3, callerid=0x859e520 "\"sip-proxy\" ", action=1163) at pbx.c:1170 #14 0x08065f1a in ast_pbx_run (c=0x85b0ed8) at pbx.c:1654 #15 0x0806bca1 in pbx_thread (data=0x8551570) at pbx.c:1875 #16 0x00b7679c in start_thread () from /lib/tls/libpthread.so.0 #17 0x00ad627a in clone () from /lib/tls/libc.so.6