| Summary: | ASTERISK-19714: No hungup after BYE. | ||||
| Reporter: | Grzegorz Głowacki (g.glowacki) | Labels: | |||
| Date Opened: | 2012-04-12 06:42:45 | Date Closed: | 2012-10-17 09:31:11 | ||
| Priority: | Major | Regression? | |||
| Status: | Closed/Complete | Components: | Channels/chan_sip/General | ||
| Versions: | 1.8.11.0 | Frequency of Occurrence | |||
| Related Issues: |
| ||||
| Environment: | #cat /etc/issue: AsteriskNOW 2.0.0 Asterisk #env HOSTNAME=pc3861 TERM=xterm SHELL=/bin/bash HISTSIZE=1000 SSH_CLIENT=10.4.1.156 51409 22 SSH_TTY=/dev/pts/1 USER=root LS_COLORS=no=00:fi=00:di=00;34:ln=00;36:pi=40;33:so=00;35:bd=40;33;01:cd=40;33;01:or=01;05;37;41:mi=01;05;37;41:ex=00;32:*.cmd=00;32:*.exe=00;32:*.com=00;32:*.btm=00;32:*.bat=00;32:*.sh=00;32:*.csh=00;32:*.tar=00;31:*.tgz=00;31:*.arj=00;31:*.taz=00;31:*.lzh=00;31:*.zip=00;31:*.z=00;31:*.Z=00;31:*.gz=00;31:*.bz2=00;31:*.bz=00;31:*.tz=00;31:*.rpm=00;31:*.cpio=00;31:*.jpg=00;35:*.gif=00;35:*.bmp=00;35:*.xbm=00;35:*.xpm=00;35:*.png=00;35:*.tif=00;35: MAIL=/var/spool/mail/root PATH=/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin INPUTRC=/etc/inputrc PWD=/etc/asterisk LANG=en_US.UTF-8 SSH_ASKPASS=/usr/libexec/openssh/gnome-ssh-askpass SHLVL=1 HOME=/root LOGNAME=root SSH_CONNECTION=10.4.1.156 51409 10.1.5.111 22 LESSOPEN=|/usr/bin/lesspipe.sh %s G_BROKEN_FILENAMES=1 _=/bin/env OLDPWD=/usr/lib/asterisk/modules pc3861*CLI> module show Module Description Use Count res_adsi ADSI Resource 0 res_agi.so Asterisk Gateway Interface (AGI) 1 res_speech.so Generic Speech Recognition API 0 res_smdi.so Simplified Message Desk Interface (SMDI) 0 res_crypto.so Cryptographic Digital Signatures 0 res_calendar.so Asterisk Calendar integration 0 res_ael_share.so share-able code for AEL 0 res_monitor.so Call Monitoring Resource 0 res_stun_monitor.so STUN Network Monitor 0 res_fax.so Generic FAX Applications 0 format_jpeg.so jpeg (joint picture experts group) image 0 app_sayunixtime.so Say time 0 format_sln16.so Raw Signed Linear 16KHz Audio support (S 0 app_cdr.so Tell Asterisk to not maintain a CDR for 0 app_meetme.so MeetMe conference bridge 0 app_channelredirect.so Redirects a given channel to a dialplan 0 func_strings.so String handling dialplan functions 0 format_wav_gsm.so Microsoft WAV format (Proprietary GSM) 0 app_senddtmf.so Send DTMF digits Application 0 format_g723.so G.723.1 Simple Timestamp File Format 0 app_flash.so Flash channel application 0 format_ilbc.so Raw iLBC data 0 app_chanspy.so Listen to the audio of an active channel 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 func_db.so Database (astdb) related dialplan functi 0 cdr_syslog.so Customizable syslog CDR Backend 0 app_system.so Generic System() application 0 app_chanisavail.so Check channel availability 0 codec_dahdi.so Generic DAHDI Transcoder Codec Translato 0 func_config.so Asterisk configuration file variable acc 0 chan_skinny.so Skinny Client Control Protocol (Skinny) 0 app_followme.so Find-Me/Follow-Me Application 0 func_iconv.so Charset conversions 0 app_speech_utils.so Dialplan Speech Applications 0 codec_adpcm.so Adaptive Differential PCM Coder/Decoder 0 format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0 app_read.so Read Variable Application 0 res_realtime.so Realtime Data Lookup/Rewrite 0 chan_multicast_rtp.so Multicast RTP Paging Channel 0 app_macro.so Extension Macros 0 app_forkcdr.so Fork The CDR into 2 separate entities 0 app_page.so Page Multiple Phones 0 func_enum.so ENUM related dialplan functions 0 app_dahdibarge.so Barge in on DAHDI channel application 0 format_gsm.so Raw GSM data 0 res_rtp_multicast.so Multicast RTP Engine 0 app_getcpeid.so Get ADSI CPE ID 0 res_phoneprov.so HTTP Phone Provisioning 0 func_extstate.so Gets an extension's state in the dialpla 0 func_module.so Checks if Asterisk module is loaded in m 0 app_sms.so SMS/PSTN handler 0 codec_lpc10.so LPC10 2.4kbps Coder/Decoder 0 app_minivm.so Mini VoiceMail (A minimal Voicemail e-ma 0 app_disa.so DISA (Direct Inward System Access) Appli 0 bridge_softmix.so Multi-party software based channel mixin 0 func_srv.so SRV related dialplan functions 0 chan_iax2.so Inter Asterisk eXchange (Ver 2) 0 app_playback.so Sound File Playback Application 0 func_cdr.so Call Detail Record (CDR) dialplan functi 0 format_siren7.so ITU G.722.1 (Siren7, licensed from Polyc 0 format_g729.so Raw G.729 data 0 codec_ilbc.so iLBC Coder/Decoder 0 res_convert.so File format conversion CLI command 0 app_nbscat.so Silly NBS Stream Application 0 func_devstate.so Gets or sets a device state in the dialp 0 app_verbose.so Send verbose output 0 app_while.so While Loops and Conditional Execution 0 app_setcallerid.so Set CallerID Presentation Application 0 app_mixmonitor.so Mixed Audio Monitoring Application 0 func_lock.so Dialplan mutexes 0 func_channel.so Channel information dialplan functions 0 app_directed_pickup.so Directed Call Pickup Application 0 format_h264.so Raw H.264 data 0 app_db.so Database Access Functions 0 func_dialgroup.so Dialgroup dialplan function 0 app_ices.so Encode and Stream via icecast and ices 0 cdr_csv.so Comma Separated Values CDR Backend 0 app_record.so Trivial Record Application 0 bridge_simple.so Simple two channel bridging module 0 func_env.so Environment/filesystem dialplan function 0 format_g719.so ITU G.719 0 chan_local.so Local Proxy Channel (Note: used internal 0 chan_sip.so Session Initiation Protocol (SIP) 0 app_readfile.so Stores output of file into a variable 0 app_morsecode.so Morse code 0 func_md5.so MD5 digest dialplan functions 0 func_rand.so Random number dialplan function 0 func_frame_trace.so Frame Trace for internal ast_frame debug 0 func_aes.so AES dialplan functions 0 app_confbridge.so Conference Bridge Application 0 res_timing_dahdi.so DAHDI Timing Interface 1 func_callerid.so Party ID related dialplan functions (Cal 0 func_sysinfo.so System information related functions 0 pbx_spool.so Outgoing Spool Support 0 bridge_builtin_features.so Built in bridging features 1 app_queue.so True Call Queueing 0 func_base64.so base64 encode/decode dialplan functions 0 app_voicemail.so Comedian Mail (Voicemail System) 0 codec_ulaw.so mu-Law Coder/Decoder 0 cel_manager.so Asterisk Manager Interface CEL Backend 0 app_waitforsilence.so Wait For Silence 0 func_global.so Variable dialplan functions 0 func_shell.so Returns the output of a shell command 0 format_h263.so Raw H.263 data 0 func_version.so Get Asterisk Version/Build Info 0 format_wav.so Microsoft WAV/WAV16 format (8kHz/16kHz S 0 pbx_config.so Text Extension Configuration 0 codec_g726.so ITU G.726-32kbps G726 Transcoder 0 app_dial.so Dialing Application 0 chan_bridge.so Bridge Interaction Channel 0 app_talkdetect.so Playback with Talk Detection 0 cdr_custom.so Customizable Comma Separated Values CDR 0 codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0 cel_custom.so Customizable Comma Separated Values CEL 0 func_groupcount.so Channel group dialplan functions 0 app_softhangup.so Hangs up the requested channel 0 res_security_log.so Security Event Logging 0 app_dumpchan.so Dump Info About The Calling Channel 0 format_vox.so Dialogic VOX (ADPCM) File Format 0 cdr_manager.so Asterisk Manager Interface CDR Backend 0 app_adsiprog.so Asterisk ADSI Programming Application 0 res_clialiases.so CLI Aliases 0 func_realtime.so Read/Write/Store/Destroy values from a R 0 app_mp3.so Silly MP3 Application 0 res_clioriginate.so Call origination and redirection from th 0 app_amd.so Answering Machine Detection Application 0 pbx_ael.so Asterisk Extension Language Compiler 0 app_sendtext.so Send Text Applications 0 func_sprintf.so SPRINTF dialplan function 0 chan_agent.so Agent Proxy Channel 0 app_dahdiras.so DAHDI ISDN Remote Access Server 0 app_zapateller.so Block Telemarketers with Special Informa 0 func_pitchshift.so Audio Effects Dialplan Functions 0 app_dictate.so Virtual Dictation Machine 0 app_test.so Interface Test Application 0 app_externalivr.so External IVR Interface Application 0 bridge_multiplexed.so Multiplexed two channel bridging module 0 app_readexten.so Read and evaluate extension validity 0 codec_gsm.so GSM Coder/Decoder 0 res_rtp_asterisk.so Asterisk RTP Stack 0 app_url.so Send URL Applications 0 func_vmcount.so Indicator for whether a voice mailbox ha 0 format_siren14.so ITU G.722.1 Annex C (Siren14, licensed f 0 app_parkandannounce.so Call Parking and Announce Application 0 res_musiconhold.so Music On Hold Resource 0 app_celgenuserevent.so Generate an User-Defined CEL event 0 pbx_realtime.so Realtime Switch 0 func_uri.so URI encode/decode dialplan functions 0 func_volume.so Technology independent volume control 0 func_dialplan.so Dialplan Context/Extension/Priority Chec 0 func_timeout.so Channel timeout dialplan functions 0 app_festival.so Simple Festival Interface 0 app_originate.so Originate call 0 app_waitforring.so Waits until first ring after time 0 func_callcompletion.so Call Control Configuration Function 0 res_mutestream.so Mute audio stream resources 0 format_sln.so Raw Signed Linear Audio support (SLN) 0 app_controlplayback.so Control Playback Application 0 app_playtones.so Playtones Application 0 func_logic.so Logical dialplan functions 0 func_sha1.so SHA-1 computation dialplan function 0 func_blacklist.so Look up Caller*ID name/number from black 0 app_transfer.so Transfers a caller to another extension 0 app_userevent.so Custom User Event Application 0 func_math.so Mathematical dialplan function 0 codec_alaw.so A-law Coder/Decoder 0 chan_oss.so OSS Console Channel Driver 0 app_authenticate.so Authentication Application 0 app_stack.so Dialplan subroutines (Gosub, Return, etc 0 chan_phone.so Linux Telephony API Support 0 func_cut.so Cut out information from a string 0 chan_dahdi.so DAHDI Telephony Driver w/PRI & SS7 & MFC 0 app_waituntil.so Wait until specified time 0 app_privacy.so Require phone number to be entered, if n 0 app_image.so Image Transmission Application 0 chan_unistim.so UNISTIM Protocol (USTM) 0 res_limit.so Resource limits 0 pbx_loopback.so Loopback Switch 0 app_echo.so Simple Echo Application 0 app_alarmreceiver.so Alarm Receiver for Asterisk 0 res_timing_pthread.so pthread Timing Interface 0 pbx_dundi.so Distributed Universal Number Discovery ( 0 app_milliwatt.so Digital Milliwatt (mu-law) Test Applicat 0 func_audiohookinherit.so Audiohook inheritance function 0 app_exec.so Executes dialplan applications 0 codec_g722.so ITU G.722-64kbps G722 Transcoder 0 app_directory.so Extension Directory 0 pc3861*CLI> sip show settings Global Settings: ---------------- UDP Bindaddress: 10.1.5.111:5060 TCP SIP Bindaddress: Disabled TLS SIP Bindaddress: Disabled Videosupport: Yes Textsupport: No Ignore SDP sess. ver.: No AutoCreate Peer: No Match Auth Username: No Allow unknown access: Yes Allow subscriptions: Yes Allow overlap dialing: Yes Allow promisc. redir: No Enable call counters: Yes SIP domain support: No Realm. auth: No Our auth realm asterisk Use domains as realms: No Call to non-local dom.: Yes URI user is phone no: No Always auth rejects: Yes Direct RTP setup: No User Agent: Asterisk PBX 1.8.11.0 SDP Session Name: Asterisk PBX 1.8.11.0 SDP Owner Name: root Reg. context: (not set) Regexten on Qualify: No Legacy userfield parse: No Caller ID: asterisk From: Domain: Record SIP history: Off Call Events: On Auth. Failure Events: Off T.38 support: No T.38 EC mode: Unknown T.38 MaxDtgrm: -1 SIP realtime: Disabled Qualify Freq : 60000 ms Q.850 Reason header: No Store SIP_CAUSE: No Network QoS Settings: --------------------------- IP ToS SIP: CS0 IP ToS RTP audio: CS0 IP ToS RTP video: CS0 IP ToS RTP text: CS0 802.1p CoS SIP: 4 802.1p CoS RTP audio: 5 802.1p CoS RTP video: 6 802.1p CoS RTP text: 5 Jitterbuffer enabled: No Network Settings: --------------------------- SIP address remapping: Disabled, no localnet list Externhost: <none> Externaddr: (null) Externrefresh: 10 Global Signalling Settings: --------------------------- Codecs: 0x6c040e (gsm|ulaw|alaw|ilbc|h261|h263|h264|mpeg4) Codec Order: ulaw:20,alaw:20,ilbc:30,gsm:20 Relax DTMF: No RFC2833 Compensation: No Symmetric RTP: Yes Compact SIP headers: No RTP Keepalive: 0 (Disabled) RTP Timeout: 2 RTP Hold Timeout: 0 (Disabled) MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: Yes Pedantic SIP support: Yes Reg. min duration 60 secs Reg. max duration: 3600 secs Reg. default duration: 120 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 Notify ringing state: Yes Include CID: No Notify hold state: No SIP Transfer mode: open Max Call Bitrate: 384 kbps Auto-Framing: No Outb. proxy: <not set> Session Timers: Originate Session Refresher: uas Session Expires: 90 secs Session Min-SE: 90 secs Timer T1: 500 Timer T1 minimum: 100 Timer B: 32000 No premature media: Yes Max forwards: 70 Default Settings: ----------------- Allowed transports: UDP Outbound transport: UDP Context: default Force rport: Yes DTMF: rfc2833 Qualify: 0 Use ClientCode: No Progress inband: Never Language: MOH Interpret: default MOH Suggest: Voice Mail Extension: asterisk ---- <--- SIP read from UDP:10.4.1.7:23632 ---> | Attachments: | ( 0) bye.pcap ( 1) conf.tgz | ||
| Description: | I have two sip peers. One is XLITE 4, and the other one is an IVR application based on dialogic 4.1 HMP software. When i connect from XLITE4(gglowack) to IVR( 1000) through asterisk everything works fine until i hung up call from XLITE - it does disconnect, sending BYE message, and retrieves OK. But then asterisk doesn't send BYE to my IVR, so the device is never released - second call is impossible, because line is busy. On the other hand, when i do hung up from IVR the connection is properly released on both sides. [root@pc3861 asterisk]# asterisk -r Asterisk 1.8.11.0, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 1.8.11.0 currently running on pc3861 (pid = 2866) Verbosity is at least 3 pc3861*CLI> sip set debug on SIP Debugging re-enabled Really destroying SIP dialog 'NGU2MWYyZDg5OWUyZjIzZWRiNGY3MDliMGMzYWM2NzE.' Method: REGISTER <--- SIP read from UDP:10.4.1.7:53398 ---> REGISTER sip:10.1.5.111 SIP/2.0 Via: SIP/2.0/UDP 10.4.1.7:53398;branch=z9hG4bK-d8754z-620854dd554b6b8d-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:gglowack@10.4.1.7:53398;rinstance=ce50807a18c76c4d> To: <sip:gglowack@10.1.5.111> From: <sip:gglowack@10.1.5.111>;tag=d94fde8a Call-ID: MGM5M2RkODdhMzQ3MGZhMDExYzk5Nzk1MTgzODM1OWQ. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.4.1.7:53398 (NAT) <--- Transmitting (NAT) to 10.4.1.7:53398 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.4.1.7:53398;branch=z9hG4bK-d8754z-620854dd554b6b8d-1---d8754z-;received=10.4.1.7;rport=53398 From: <sip:gglowack@10.1.5.111>;tag=d94fde8a To: <sip:gglowack@10.1.5.111>;tag=as395b9762 Call-ID: MGM5M2RkODdhMzQ3MGZhMDExYzk5Nzk1MTgzODM1OWQ. CSeq: 1 REGISTER Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4527a4d9" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'MGM5M2RkODdhMzQ3MGZhMDExYzk5Nzk1MTgzODM1OWQ.' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.4.1.7:53398 ---> REGISTER sip:10.1.5.111 SIP/2.0 Via: SIP/2.0/UDP 10.4.1.7:53398;branch=z9hG4bK-d8754z-2bb6ba4f4d01adad-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:gglowack@10.4.1.7:53398;rinstance=ce50807a18c76c4d> To: <sip:gglowack@10.1.5.111> From: <sip:gglowack@10.1.5.111>;tag=d94fde8a Call-ID: MGM5M2RkODdhMzQ3MGZhMDExYzk5Nzk1MTgzODM1OWQ. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.1 stamp 63214 Authorization: Digest username="gglowack",realm="asterisk",nonce="4527a4d9",uri="sip:10.1.5.111",response="a59b75f4e61ab957bfec9ce2af5dac8f",algorithm=MD5 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 10.4.1.7:53398 (NAT) [Apr 12 13:15:47] WARNING[2881]: db.c:115 dbinit: Unable to open Asterisk database '/var/lib/asterisk/astdb': Permission denied -- Registered SIP 'gglowack' at 10.4.1.7:53398 <--- Transmitting (NAT) to 10.4.1.7:53398 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.1.7:53398;branch=z9hG4bK-d8754z-2bb6ba4f4d01adad-1---d8754z-;received=10.4.1.7;rport=53398 From: <sip:gglowack@10.1.5.111>;tag=d94fde8a To: <sip:gglowack@10.1.5.111>;tag=as395b9762 Call-ID: MGM5M2RkODdhMzQ3MGZhMDExYzk5Nzk1MTgzODM1OWQ. CSeq: 2 REGISTER Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: <sip:gglowack@10.4.1.7:53398;rinstance=ce50807a18c76c4d>;expires=3600 Date: Thu, 12 Apr 2012 11:15:47 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'MGM5M2RkODdhMzQ3MGZhMDExYzk5Nzk1MTgzODM1OWQ.' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.4.1.7:53398 ---> SUBSCRIBE sip:gglowack@10.1.5.111 SIP/2.0 Via: SIP/2.0/UDP 10.4.1.7:53398;branch=z9hG4bK-d8754z-8948575ee3924a79-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:gglowack@10.4.1.7:53398> To: <sip:gglowack@10.1.5.111> From: <sip:gglowack@10.1.5.111>;tag=cbf6f11c Call-ID: NjA3OTUwNjNiMTg0ODU0OGZhYmIzOTAzMTM5MGVlN2Y. CSeq: 1 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.1 stamp 63214 Event: message-summary Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Creating new subscription Sending to 10.4.1.7:53398 (NAT) list_route: hop: <sip:gglowack@10.4.1.7:53398> Found peer 'gglowack' for 'gglowack' from 10.4.1.7:53398 <--- Transmitting (NAT) to 10.4.1.7:53398 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.4.1.7:53398;branch=z9hG4bK-d8754z-8948575ee3924a79-1---d8754z-;received=10.4.1.7;rport=53398 From: <sip:gglowack@10.1.5.111>;tag=cbf6f11c To: <sip:gglowack@10.1.5.111>;tag=as11b7955b Call-ID: NjA3OTUwNjNiMTg0ODU0OGZhYmIzOTAzMTM5MGVlN2Y. CSeq: 1 SUBSCRIBE Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="49009bd6" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'NjA3OTUwNjNiMTg0ODU0OGZhYmIzOTAzMTM5MGVlN2Y.' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from UDP:10.4.1.7:53398 ---> SUBSCRIBE sip:gglowack@10.1.5.111 SIP/2.0 Via: SIP/2.0/UDP 10.4.1.7:53398;branch=z9hG4bK-d8754z-3a99a74371dcad53-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:gglowack@10.4.1.7:53398> To: <sip:gglowack@10.1.5.111> From: <sip:gglowack@10.1.5.111>;tag=cbf6f11c Call-ID: NjA3OTUwNjNiMTg0ODU0OGZhYmIzOTAzMTM5MGVlN2Y. CSeq: 2 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.1 stamp 63214 Authorization: Digest username="gglowack",realm="asterisk",nonce="49009bd6",uri="sip:gglowack@10.1.5.111",response="a89dc7af3279bbb2e16add77276a6b36",algorithm=MD5 Event: message-summary Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Creating new subscription Sending to 10.4.1.7:53398 (NAT) Found peer 'gglowack' for 'gglowack' from 10.4.1.7:53398 <--- Transmitting (NAT) to 10.4.1.7:53398 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 10.4.1.7:53398;branch=z9hG4bK-d8754z-3a99a74371dcad53-1---d8754z-;received=10.4.1.7;rport=53398 From: <sip:gglowack@10.1.5.111>;tag=cbf6f11c To: <sip:gglowack@10.1.5.111>;tag=as11b7955b Call-ID: NjA3OTUwNjNiMTg0ODU0OGZhYmIzOTAzMTM5MGVlN2Y. CSeq: 2 SUBSCRIBE Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Apr 12 13:15:47] NOTICE[2881]: chan_sip.c:24752 handle_request_subscribe: Received SIP subscribe for peer without mailbox: gglowack Really destroying SIP dialog 'NjA3OTUwNjNiMTg0ODU0OGZhYmIzOTAzMTM5MGVlN2Y.' Method: SUBSCRIBE Really destroying SIP dialog 'bd909a8-0-13c6-50022-c7adc-1c7e2e22-c7adc' Method: REGISTER Really destroying SIP dialog '3c37e01445539d961d5012dc4adda05e@10.1.5.111:5060' Method: BYE pc3861*CLI> Disconnected from Asterisk server [root@pc3861 asterisk]# asterisk -r Asterisk 1.8.11.0, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 1.8.11.0 currently running on pc3861 (pid = 2866) Verbosity is at least 3 <--- SIP read from UDP:10.4.1.7:53398 ---> REGISTER sip:10.1.5.111 SIP/2.0 Via: SIP/2.0/UDP 10.4.1.7:53398;branch=z9hG4bK-d8754z-97b9d59b2ae15442-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:gglowack@10.4.1.7:53398;rinstance=ce50807a18c76c4d>;expires=0 To: <sip:gglowack@10.1.5.111> From: <sip:gglowack@10.1.5.111>;tag=d94fde8a Call-ID: MGM5M2RkODdhMzQ3MGZhMDExYzk5Nzk1MTgzODM1OWQ. CSeq: 3 REGISTER Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.1 stamp 63214 Authorization: Digest username="gglowack",realm="asterisk",nonce="4527a4d9",uri="sip:10.1.5.111",response="a59b75f4e61ab957bfec9ce2af5dac8f",algorithm=MD5 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.4.1.7:53398 (NAT) [Apr 12 13:15:58] NOTICE[2881]: chan_sip.c:14441 check_auth: Correct auth, but based on stale nonce received from '<sip:gglowack@10.1.5.111>;tag=d94fde8a' <--- Transmitting (NAT) to 10.4.1.7:53398 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.4.1.7:53398;branch=z9hG4bK-d8754z-97b9d59b2ae15442-1---d8754z-;received=10.4.1.7;rport=53398 From: <sip:gglowack@10.1.5.111>;tag=d94fde8a To: <sip:gglowack@10.1.5.111>;tag=as395b9762 Call-ID: MGM5M2RkODdhMzQ3MGZhMDExYzk5Nzk1MTgzODM1OWQ. CSeq: 3 REGISTER Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2131bb8b", stale=true Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'MGM5M2RkODdhMzQ3MGZhMDExYzk5Nzk1MTgzODM1OWQ.' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.4.1.7:53398 ---> REGISTER sip:10.1.5.111 SIP/2.0 Via: SIP/2.0/UDP 10.4.1.7:53398;branch=z9hG4bK-d8754z-067b73265ab60fa7-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:gglowack@10.4.1.7:53398;rinstance=ce50807a18c76c4d>;expires=0 To: <sip:gglowack@10.1.5.111> From: <sip:gglowack@10.1.5.111>;tag=d94fde8a Call-ID: MGM5M2RkODdhMzQ3MGZhMDExYzk5Nzk1MTgzODM1OWQ. CSeq: 4 REGISTER Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.1 stamp 63214 Authorization: Digest username="gglowack",realm="asterisk",nonce="2131bb8b",uri="sip:10.1.5.111",response="eee9eb53bcfa3e28dfefeb4bac0af8f1",algorithm=MD5 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.4.1.7:53398 (NAT) -- Unregistered SIP 'gglowack' [Apr 12 13:15:58] WARNING[2881]: db.c:115 dbinit: Unable to open Asterisk database '/var/lib/asterisk/astdb': Permission denied [Apr 12 13:15:58] WARNING[2881]: db.c:115 dbinit: Unable to open Asterisk database '/var/lib/asterisk/astdb': Permission denied <--- Transmitting (NAT) to 10.4.1.7:53398 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.1.7:53398;branch=z9hG4bK-d8754z-067b73265ab60fa7-1---d8754z-;received=10.4.1.7;rport=53398 From: <sip:gglowack@10.1.5.111>;tag=d94fde8a To: <sip:gglowack@10.1.5.111>;tag=as395b9762 Call-ID: MGM5M2RkODdhMzQ3MGZhMDExYzk5Nzk1MTgzODM1OWQ. CSeq: 4 REGISTER Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: <sip:gglowack@10.4.1.7:53398;rinstance=ce50807a18c76c4d>;expires=3600 Date: Thu, 12 Apr 2012 11:15:58 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'MGM5M2RkODdhMzQ3MGZhMDExYzk5Nzk1MTgzODM1OWQ.' in 32000 ms (Method: REGISTER) pc3861*CLI> Disconnected from Asterisk server [root@pc3861 asterisk]# asterisk -r Asterisk 1.8.11.0, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 1.8.11.0 currently running on pc3861 (pid = 2866) Verbosity is at least 3 pc3861*CLI> sip set debug on SIP Debugging re-enabled Really destroying SIP dialog 'MGM5M2RkODdhMzQ3MGZhMDExYzk5Nzk1MTgzODM1OWQ.' Method: REGISTER <--- SIP read from UDP:10.4.1.7:23632 ---> REGISTER sip:10.1.5.111 SIP/2.0 Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-4733ccc0ef27360b-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:gglowack@10.4.1.7:23632;rinstance=78f10e94b63e6096> To: <sip:gglowack@10.1.5.111> From: <sip:gglowack@10.1.5.111>;tag=0315a899 Call-ID: M2YyNWEzZTk5M2IyZjllNjIyZDBjZTg1MzViMmMyZmI. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.4.1.7:23632 (NAT) <--- Transmitting (NAT) to 10.4.1.7:23632 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-4733ccc0ef27360b-1---d8754z-;received=10.4.1.7;rport=23632 From: <sip:gglowack@10.1.5.111>;tag=0315a899 To: <sip:gglowack@10.1.5.111>;tag=as3fe521d9 Call-ID: M2YyNWEzZTk5M2IyZjllNjIyZDBjZTg1MzViMmMyZmI. CSeq: 1 REGISTER Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5ef85ed7" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'M2YyNWEzZTk5M2IyZjllNjIyZDBjZTg1MzViMmMyZmI.' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.4.1.7:23632 ---> REGISTER sip:10.1.5.111 SIP/2.0 Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-a8c9d309d4b3d4ce-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:gglowack@10.4.1.7:23632;rinstance=78f10e94b63e6096> To: <sip:gglowack@10.1.5.111> From: <sip:gglowack@10.1.5.111>;tag=0315a899 Call-ID: M2YyNWEzZTk5M2IyZjllNjIyZDBjZTg1MzViMmMyZmI. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.1 stamp 63214 Authorization: Digest username="gglowack",realm="asterisk",nonce="5ef85ed7",uri="sip:10.1.5.111",response="320467fc1bf25b5c861f1ca004bc5671",algorithm=MD5 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 10.4.1.7:23632 (NAT) [Apr 12 13:16:33] WARNING[2881]: db.c:115 dbinit: Unable to open Asterisk database '/var/lib/asterisk/astdb': Permission denied -- Registered SIP 'gglowack' at 10.4.1.7:23632 <--- Transmitting (NAT) to 10.4.1.7:23632 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-a8c9d309d4b3d4ce-1---d8754z-;received=10.4.1.7;rport=23632 From: <sip:gglowack@10.1.5.111>;tag=0315a899 To: <sip:gglowack@10.1.5.111>;tag=as3fe521d9 Call-ID: M2YyNWEzZTk5M2IyZjllNjIyZDBjZTg1MzViMmMyZmI. CSeq: 2 REGISTER Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: <sip:gglowack@10.4.1.7:23632;rinstance=78f10e94b63e6096>;expires=3600 Date: Thu, 12 Apr 2012 11:16:33 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'M2YyNWEzZTk5M2IyZjllNjIyZDBjZTg1MzViMmMyZmI.' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.4.1.7:23632 ---> SUBSCRIBE sip:gglowack@10.1.5.111 SIP/2.0 Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-1c9d2a4940eb43a4-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:gglowack@10.4.1.7:23632> To: <sip:gglowack@10.1.5.111> From: <sip:gglowack@10.1.5.111>;tag=4f813dda Call-ID: ZmI4Yzc4YTYzNjc5NGVjNjkyZGM5ZGI2OTIwYjk3NDk. CSeq: 1 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.1 stamp 63214 Event: message-summary Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Creating new subscription Sending to 10.4.1.7:23632 (NAT) list_route: hop: <sip:gglowack@10.4.1.7:23632> Found peer 'gglowack' for 'gglowack' from 10.4.1.7:23632 <--- Transmitting (NAT) to 10.4.1.7:23632 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-1c9d2a4940eb43a4-1---d8754z-;received=10.4.1.7;rport=23632 From: <sip:gglowack@10.1.5.111>;tag=4f813dda To: <sip:gglowack@10.1.5.111>;tag=as76743a8d Call-ID: ZmI4Yzc4YTYzNjc5NGVjNjkyZGM5ZGI2OTIwYjk3NDk. CSeq: 1 SUBSCRIBE Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56135c05" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'ZmI4Yzc4YTYzNjc5NGVjNjkyZGM5ZGI2OTIwYjk3NDk.' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from UDP:10.4.1.7:23632 ---> SUBSCRIBE sip:gglowack@10.1.5.111 SIP/2.0 Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-fb86204c0fd94d3d-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:gglowack@10.4.1.7:23632> To: <sip:gglowack@10.1.5.111> From: <sip:gglowack@10.1.5.111>;tag=4f813dda Call-ID: ZmI4Yzc4YTYzNjc5NGVjNjkyZGM5ZGI2OTIwYjk3NDk. CSeq: 2 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite 4 release 4.1 stamp 63214 Authorization: Digest username="gglowack",realm="asterisk",nonce="56135c05",uri="sip:gglowack@10.1.5.111",response="413e5f3a6dba51cc3e037eb24455602e",algorithm=MD5 Event: message-summary Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Creating new subscription Sending to 10.4.1.7:23632 (NAT) Found peer 'gglowack' for 'gglowack' from 10.4.1.7:23632 <--- Transmitting (NAT) to 10.4.1.7:23632 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-fb86204c0fd94d3d-1---d8754z-;received=10.4.1.7;rport=23632 From: <sip:gglowack@10.1.5.111>;tag=4f813dda To: <sip:gglowack@10.1.5.111>;tag=as76743a8d Call-ID: ZmI4Yzc4YTYzNjc5NGVjNjkyZGM5ZGI2OTIwYjk3NDk. CSeq: 2 SUBSCRIBE Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Apr 12 13:16:33] NOTICE[2881]: chan_sip.c:24752 handle_request_subscribe: Received SIP subscribe for peer without mailbox: gglowack Really destroying SIP dialog 'ZmI4Yzc4YTYzNjc5NGVjNjkyZGM5ZGI2OTIwYjk3NDk.' Method: SUBSCRIBE Really destroying SIP dialog '3b1c5b4308196cfe2acb450d27437f02@10.1.5.111' Method: REGISTER <--- SIP read from UDP:10.1.5.123:5062 ---> REGISTER sip:10.1.5.111 SIP/2.0 From: <sip:1000@10.1.5.111>;tag=cf4e710-0-13c6-50022-c7cd3-7c4707a0-c7cd3 To: <sip:1000@10.1.5.111> Call-ID: cf5c9a8-0-13c6-50022-c7cd3-69566da-c7cd3 CSeq: 1 REGISTER Via: SIP/2.0/UDP 10.1.5.123:5062;branch=z9hG4bK-c7cd3-30c79bb4-561765a9 Max-Forwards: 70 Expires: 60 Contact: <sip:1000@10.1.5.111:5062>;methods="INVITE, INFO, SUBSCRIBE, BYE, CANCEL, NOTIFY, ACK, REFER" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.1.5.123:5062 (NAT) [Apr 12 13:16:48] WARNING[2881]: db.c:115 dbinit: Unable to open Asterisk database '/var/lib/asterisk/astdb': Permission denied -- Registered SIP '1000' at 10.1.5.123:5062 <--- Transmitting (NAT) to 10.1.5.123:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.5.123:5062;branch=z9hG4bK-c7cd3-30c79bb4-561765a9;received=10.1.5.123;rport=5062 From: <sip:1000@10.1.5.111>;tag=cf4e710-0-13c6-50022-c7cd3-7c4707a0-c7cd3 To: <sip:1000@10.1.5.111>;tag=as130a6fb8 Call-ID: cf5c9a8-0-13c6-50022-c7cd3-69566da-c7cd3 CSeq: 1 REGISTER Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 60 Contact: <sip:1000@10.1.5.111:5062>;expires=60 Date: Thu, 12 Apr 2012 11:16:48 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'cf5c9a8-0-13c6-50022-c7cd3-69566da-c7cd3' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.4.1.7:23632 ---> INVITE sip:1000@10.1.5.111 SIP/2.0 Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-c5df1dbb193c2f8b-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:gglowack@10.4.1.7:23632> To: "1000"<sip:1000@10.1.5.111> From: <sip:gglowack@10.1.5.111>;tag=fad5cbaf Call-ID: YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 390 v=0 o=- 12978707194892505 1 IN IP4 10.4.1.7 s=CounterPath X-Lite 4.1 c=IN IP4 10.4.1.7 t=0 0 a=ice-ufrag:2b1621 a=ice-pwd:60f1508bc138ab3d7d652c0ebf1afaf0 m=audio 60970 RTP/AVP 107 0 8 101 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 10.4.1.7 60970 typ host a=candidate:1 2 UDP 659134 10.4.1.7 60971 typ host <-------------> --- (13 headers 14 lines) --- Sending to 10.4.1.7:23632 (NAT) Using INVITE request as basis request - YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU. Found peer 'gglowack' for 'gglowack' from 10.4.1.7:23632 <--- Reliably Transmitting (NAT) to 10.4.1.7:23632 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-c5df1dbb193c2f8b-1---d8754z-;received=10.4.1.7;rport=23632 From: <sip:gglowack@10.1.5.111>;tag=fad5cbaf To: "1000"<sip:1000@10.1.5.111>;tag=as6f5ae84d Call-ID: YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU. CSeq: 1 INVITE erver: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="29ca45ee" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU.' in 32000 ms (Method: INVITE) <--- SIP read from UDP:10.4.1.7:23632 ---> ACK sip:1000@10.1.5.111 SIP/2.0 Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-c5df1dbb193c2f8b-1---d8754z-;rport Max-Forwards: 70 To: "1000"<sip:1000@10.1.5.111>;tag=as6f5ae84d From: <sip:gglowack@10.1.5.111>;tag=fad5cbaf Call-ID: YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU. CSeq: 1 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:10.4.1.7:23632 ---> INVITE sip:1000@10.1.5.111 SIP/2.0 Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-582411da0254670f-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:gglowack@10.4.1.7:23632> To: "1000"<sip:1000@10.1.5.111> From: <sip:gglowack@10.1.5.111>;tag=fad5cbaf Call-ID: YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.1 stamp 63214 Authorization: Digest username="gglowack",realm="asterisk",nonce="29ca45ee",uri="sip:1000@10.1.5.111",response="65264656e298d908168da695a0fa95ec",algorithm=MD5 Content-Length: 390 v=0 o=- 12978707194892505 1 IN IP4 10.4.1.7 s=CounterPath X-Lite 4.1 c=IN IP4 10.4.1.7 t=0 0 a=ice-ufrag:2b1621 a=ice-pwd:60f1508bc138ab3d7d652c0ebf1afaf0 m=audio 60970 RTP/AVP 107 0 8 101 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 10.4.1.7 60970 typ host a=candidate:1 2 UDP 659134 10.4.1.7 60971 typ host <-------------> --- (14 headers 14 lines) --- Sending to 10.4.1.7:23632 (NAT) Using INVITE request as basis request - YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU. Found peer 'gglowack' for 'gglowack' from 10.4.1.7:23632 == Using SIP RTP CoS mark 5 Found RTP audio format 107 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found unknown media description format BV32 for ID 107 Found audio description format telephone-event for ID 101 Capabilities: us - 0x6c040e (gsm|ulaw|alaw|ilbc|h261|h263|h264|mpeg4), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.4.1.7:60970 Peer doesn't provide video Looking for 1000 in default (domain 10.1.5.111) list_route: hop: <sip:gglowack@10.4.1.7:23632> <--- Transmitting (NAT) to 10.4.1.7:23632 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-582411da0254670f-1---d8754z-;received=10.4.1.7;rport=23632 From: <sip:gglowack@10.1.5.111>;tag=fad5cbaf To: "1000"<sip:1000@10.1.5.111> Call-ID: YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU. CSeq: 2 INVITE Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:1000@10.1.5.111:5060> Content-Length: 0 <------------> -- Executing [1000@default:1] Dial("SIP/gglowack-00000005", "SIP/1000") in new stack == Using SIP RTP CoS mark 5 Audio is at 10766 Video is at 10.1.5.111:12408 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x2 (gsm) to SDP Adding video codec 0x40000 (h261) to SDP Adding video codec 0x80000 (h263) to SDP Adding video codec 0x200000 (h264) to SDP Adding video codec 0x400000 (mpeg4) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.1.5.123:5062: INVITE sip:1000@10.1.5.111:5062 SIP/2.0 Via: SIP/2.0/UDP 10.1.5.111:5060;branch=z9hG4bK250c8fef;rport Max-Forwards: 70 From: "gglowack" <sip:gglowack@10.1.5.111>;tag=as5b5c2f32 To: <sip:1000@10.1.5.111:5062> Contact: <sip:gglowack@10.1.5.111:5060> Call-ID: 7b5ae3db062ce0387f5a07e35660dd14@10.1.5.111:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.11.0 Date: Thu, 12 Apr 2012 11:16:56 GMT Session-Expires: 90 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 481 v=0 o=root 142380546 142380546 IN IP4 10.1.5.111 s=Asterisk PBX 1.8.11.0 c=IN IP4 10.1.5.111 b=CT:384 t=0 0 m=audio 10766 RTP/AVP 0 8 97 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 12408 RTP/AVP 31 34 99 104 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:99 H264/90000 a=rtpmap:104 MP4V-ES/90000 a=sendrecv --- -- Called SIP/1000 <--- SIP read from UDP:10.1.5.123:5062 ---> SIP/2.0 180 Ringing From: "gglowack"<sip:gglowack@10.1.5.111>;tag=as5b5c2f32 To: <sip:1000@10.1.5.111:5062>;tag=cf4e8a8-0-13c6-50022-c7cde-2719b0dd-c7cde Call-ID: 7b5ae3db062ce0387f5a07e35660dd14@10.1.5.111:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.1.5.111:5060;rport=5060;branch=z9hG4bK250c8fef Contact: <sip:1000@10.1.5.111:5062> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO Content-Type: application/sdp Content-Length: 355 v=0 o=IPMediaServer 1334229989 1334229989 IN IP4 10.1.5.123 s=Video Demonstration t=0 0 m=audio 49152 RTP/AVP 8 101 c=IN IP4 10.1.5.123 a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:20 a=rtpmap:101 telephone-event/8000 m=video 57344 RTP/AVP 104 c=IN IP4 10.1.5.123 b=AS:128 a=rtpmap:104 MP4V-ES/90000 a=fmtp:104 profile-level-id=1 a=sendrecv <-------------> --- (10 headers 16 lines) --- list_route: hop: <sip:1000@10.1.5.111:5062> Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Found RTP video format 104 Found video description format MP4V-ES for ID 104 Capabilities: us - 0x6c040e (gsm|ulaw|alaw|ilbc|h261|h263|h264|mpeg4), peer - audio=0x8 (alaw)/video=0x400000 (mpeg4)/text=0x0 (nothing), combined - 0x400008 (alaw|mpeg4) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.1.5.123:49152 Peer video RTP is at port 10.1.5.123:57344 <--- SIP read from UDP:10.1.5.123:5062 ---> SIP/2.0 200 OK From: "gglowack"<sip:gglowack@10.1.5.111>;tag=as5b5c2f32 To: <sip:1000@10.1.5.111:5062>;tag=cf4e8a8-0-13c6-50022-c7cde-2719b0dd-c7cde Call-ID: 7b5ae3db062ce0387f5a07e35660dd14@10.1.5.111:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.1.5.111:5060;rport=5060;branch=z9hG4bK250c8fef Contact: <sip:1000@10.1.5.111:5062> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO Content-Type: application/sdp Content-Length: 355 v=0 o=IPMediaServer 1334229989 1334229989 IN IP4 10.1.5.123 s=Video Demonstration t=0 0 m=audio 49152 RTP/AVP 8 101 c=IN IP4 10.1.5.123 a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:20 a=rtpmap:101 telephone-event/8000 m=video 57344 RTP/AVP 104 c=IN IP4 10.1.5.123 b=AS:128 a=rtpmap:104 MP4V-ES/90000 a=fmtp:104 profile-level-id=1 a=sendrecv <-------------> --- (10 headers 16 lines) --- list_route: hop: <sip:1000@10.1.5.111:5062> set_destination: Parsing <sip:1000@10.1.5.111:5062> for address/port to send to set_destination: set destination to 10.1.5.111:5062 Transmitting (NAT) to 10.1.5.123:5062: ACK sip:1000@10.1.5.111:5062 SIP/2.0 Via: SIP/2.0/UDP 10.1.5.111:5060;branch=z9hG4bK00a11107;rport Max-Forwards: 70 From: "gglowack" <sip:gglowack@10.1.5.111>;tag=as5b5c2f32 To: <sip:1000@10.1.5.111:5062>;tag=cf4e8a8-0-13c6-50022-c7cde-2719b0dd-c7cde Contact: <sip:gglowack@10.1.5.111:5060> Call-ID: 7b5ae3db062ce0387f5a07e35660dd14@10.1.5.111:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.11.0 Content-Length: 0 --- -- SIP/1000-00000006 is ringing <--- Transmitting (NAT) to 10.4.1.7:23632 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-582411da0254670f-1---d8754z-;received=10.4.1.7;rport=23632 From: <sip:gglowack@10.1.5.111>;tag=fad5cbaf To: "1000"<sip:1000@10.1.5.111>;tag=as4179037b Call-ID: YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU. CSeq: 2 INVITE Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:1000@10.1.5.111:5060> Content-Length: 0 <------------> -- SIP/1000-00000006 is making progress passing it to SIP/gglowack-00000005 Audio is at 11458 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 10.4.1.7:23632 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-582411da0254670f-1---d8754z-;received=10.4.1.7;rport=23632 From: <sip:gglowack@10.1.5.111>;tag=fad5cbaf To: "1000"<sip:1000@10.1.5.111>;tag=as4179037b Call-ID: YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU. CSeq: 2 INVITE Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:1000@10.1.5.111:5060> Content-Type: application/sdp Content-Length: 257 v=0 o=root 1621452656 1621452656 IN IP4 10.1.5.111 s=Asterisk PBX 1.8.11.0 c=IN IP4 10.1.5.111 t=0 0 m=audio 11458 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> -- SIP/1000-00000006 answered SIP/gglowack-00000005 Audio is at 11458 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 10.4.1.7:23632 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-582411da0254670f-1---d8754z-;received=10.4.1.7;rport=23632 From: <sip:gglowack@10.1.5.111>;tag=fad5cbaf To: "1000"<sip:1000@10.1.5.111>;tag=as4179037b Call-ID: YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU. CSeq: 2 INVITE Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:1000@10.1.5.111:5060> Content-Type: application/sdp Content-Length: 257 v=0 o=root 1621452656 1621452657 IN IP4 10.1.5.111 s=Asterisk PBX 1.8.11.0 c=IN IP4 10.1.5.111 t=0 0 m=audio 11458 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> -- Remotely bridging SIP/gglowack-00000005 and SIP/1000-00000006 set_destination: Parsing <sip:1000@10.1.5.111:5062> for address/port to send to set_destination: set destination to 10.1.5.111:5062 Audio is at 10766 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.1.5.123:5062: INVITE sip:1000@10.1.5.111:5062 SIP/2.0 Via: SIP/2.0/UDP 10.1.5.111:5060;branch=z9hG4bK3e4f491c;rport Max-Forwards: 70 From: "gglowack" <sip:gglowack@10.1.5.111>;tag=as5b5c2f32 To: <sip:1000@10.1.5.111:5062>;tag=cf4e8a8-0-13c6-50022-c7cde-2719b0dd-c7cde Contact: <sip:gglowack@10.1.5.111:5060> Call-ID: 7b5ae3db062ce0387f5a07e35660dd14@10.1.5.111:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 227 v=0 o=root 142380546 142380547 IN IP4 10.4.1.7 s=Asterisk PBX 1.8.11.0 c=IN IP4 10.4.1.7 t=0 0 m=audio 60970 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:10.1.5.123:5062 ---> SIP/2.0 100 Trying From: "gglowack"<sip:gglowack@10.1.5.111>;tag=as5b5c2f32 To: <sip:1000@10.1.5.111:5062>;tag=cf4e8a8-0-13c6-50022-c7cde-2719b0dd-c7cde Call-ID: 7b5ae3db062ce0387f5a07e35660dd14@10.1.5.111:5060 CSeq: 103 INVITE Via: SIP/2.0/UDP 10.1.5.111:5060;rport=5060;branch=z9hG4bK3e4f491c Contact: <sip:1000@10.1.5.111:5062> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Retransmitting #1 (NAT) to 10.4.1.7:23632: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-582411da0254670f-1---d8754z-;received=10.4.1.7;rport=23632 From: <sip:gglowack@10.1.5.111>;tag=fad5cbaf To: "1000"<sip:1000@10.1.5.111>;tag=as4179037b Call-ID: YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU. CSeq: 2 INVITE Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:1000@10.1.5.111:5060> Content-Type: application/sdp Content-Length: 257 v=0 o=root 1621452656 1621452657 IN IP4 10.1.5.111 s=Asterisk PBX 1.8.11.0 c=IN IP4 10.1.5.111 t=0 0 m=audio 11458 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:10.4.1.7:23632 ---> ACK sip:1000@10.1.5.111:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-d59efa0431dd8153-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:gglowack@10.4.1.7:23632> To: "1000"<sip:1000@10.1.5.111>;tag=as4179037b From: <sip:gglowack@10.1.5.111>;tag=fad5cbaf Call-ID: YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU. CSeq: 2 ACK User-Agent: X-Lite 4 release 4.1 stamp 63214 Authorization: Digest username="gglowack",realm="asterisk",nonce="29ca45ee",uri="sip:1000@10.1.5.111",response="65264656e298d908168da695a0fa95ec",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- set_destination: Parsing <sip:gglowack@10.4.1.7:23632> for address/port to send to set_destination: set destination to 10.4.1.7:23632 Audio is at 11458 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.4.1.7:23632: INVITE sip:gglowack@10.4.1.7:23632 SIP/2.0 Via: SIP/2.0/UDP 10.1.5.111:5060;branch=z9hG4bK59e75662;rport Max-Forwards: 70 From: "1000"<sip:1000@10.1.5.111>;tag=as4179037b To: <sip:gglowack@10.1.5.111>;tag=fad5cbaf Contact: <sip:1000@10.1.5.111:5060> Call-ID: YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU. CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 233 v=0 o=root 1621452656 1621452658 IN IP4 10.1.5.123 s=Asterisk PBX 1.8.11.0 c=IN IP4 10.1.5.123 t=0 0 m=audio 49152 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:10.4.1.7:23632 ---> ACK sip:1000@10.1.5.111:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-d59efa0431dd8153-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:gglowack@10.4.1.7:23632> To: "1000"<sip:1000@10.1.5.111>;tag=as4179037b From: <sip:gglowack@10.1.5.111>;tag=fad5cbaf Call-ID: YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU. CSeq: 2 ACK User-Agent: X-Lite 4 release 4.1 stamp 63214 Authorization: Digest username="gglowack",realm="asterisk",nonce="29ca45ee",uri="sip:1000@10.1.5.111",response="65264656e298d908168da695a0fa95ec",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:10.4.1.7:23632 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.5.111:5060;branch=z9hG4bK59e75662;rport=5060 Contact: <sip:gglowack@10.4.1.7:23632> To: <sip:gglowack@10.1.5.111>;tag=fad5cbaf From: "1000"<sip:1000@10.1.5.111>;tag=as4179037b Call-ID: YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU. CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 359 v=0 o=- 12978707194892505 2 IN IP4 10.4.1.7 s=CounterPath X-Lite 4.1 c=IN IP4 10.4.1.7 t=0 0 a=ice-ufrag:2b1621 a=ice-pwd:60f1508bc138ab3d7d652c0ebf1afaf0 m=audio 60970 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 10.4.1.7 60970 typ host a=candidate:1 2 UDP 659134 10.4.1.7 60971 typ host <-------------> --- (12 headers 13 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - 0x6c040e (gsm|ulaw|alaw|ilbc|h261|h263|h264|mpeg4), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.4.1.7:60970 Peer doesn't provide video set_destination: Parsing <sip:gglowack@10.4.1.7:23632> for address/port to send to set_destination: set destination to 10.4.1.7:23632 Transmitting (NAT) to 10.4.1.7:23632: ACK sip:gglowack@10.4.1.7:23632 SIP/2.0 Via: SIP/2.0/UDP 10.1.5.111:5060;branch=z9hG4bK7d53cc73;rport Max-Forwards: 70 From: "1000"<sip:1000@10.1.5.111>;tag=as4179037b To: <sip:gglowack@10.1.5.111>;tag=fad5cbaf Contact: <sip:1000@10.1.5.111:5060> Call-ID: YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU. CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.11.0 Content-Length: 0 --- <--- SIP read from UDP:10.4.1.7:23632 ---> BYE sip:1000@10.1.5.111:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-33c8e107bd04cf6f-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:gglowack@10.4.1.7:23632> To: "1000"<sip:1000@10.1.5.111>;tag=as4179037b From: <sip:gglowack@10.1.5.111>;tag=fad5cbaf Call-ID: YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU. CSeq: 3 BYE User-Agent: X-Lite 4 release 4.1 stamp 63214 Authorization: Digest username="gglowack",realm="asterisk",nonce="29ca45ee",uri="sip:1000@10.1.5.111:5060",response="850d6a2665473a64d8cef4bbff7e754d",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 10.4.1.7:23632 (NAT) Scheduling destruction of SIP dialog 'YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU.' in 32000 ms (Method: BYE) <--- Transmitting (NAT) to 10.4.1.7:23632 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.1.7:23632;branch=z9hG4bK-d8754z-33c8e107bd04cf6f-1---d8754z-;received=10.4.1.7;rport=23632 From: <sip:gglowack@10.1.5.111>;tag=fad5cbaf To: "1000"<sip:1000@10.1.5.111>;tag=as4179037b Call-ID: YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU. CSeq: 3 BYE Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> Scheduling destruction of SIP dialog '7b5ae3db062ce0387f5a07e35660dd14@10.1.5.111:5060' in 32000 ms (Method: INVITE) == Spawn extension (default, 1000, 1) exited non-zero on 'SIP/gglowack-00000005' <--- SIP read from UDP:10.4.1.7:23632 ---> <-------------> Really destroying SIP dialog 'M2YyNWEzZTk5M2IyZjllNjIyZDBjZTg1MzViMmMyZmI.' Method: REGISTER <--- SIP read from UDP:10.1.5.123:5062 ---> REGISTER sip:10.1.5.111 SIP/2.0 From: <sip:1000@10.1.5.111>;tag=cf4e710-0-13c6-50022-c7cd3-7c4707a0-c7cd3 To: <sip:1000@10.1.5.111> Call-ID: cf5c9a8-0-13c6-50022-c7cd3-69566da-c7cd3 CSeq: 2 REGISTER Via: SIP/2.0/UDP 10.1.5.123:5062;branch=z9hG4bK-c7cf3-30c81615-3f251fd3 Max-Forwards: 70 Expires: 60 Contact: <sip:1000@10.1.5.111:5062>;methods="INVITE, INFO, SUBSCRIBE, BYE, CANCEL, NOTIFY, ACK, REFER" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.1.5.123:5062 (NAT) [Apr 12 13:17:14] WARNING[2881]: db.c:115 dbinit: Unable to open Asterisk database '/var/lib/asterisk/astdb': Permission denied <--- Transmitting (NAT) to 10.1.5.123:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.5.123:5062;branch=z9hG4bK-c7cf3-30c81615-3f251fd3;received=10.1.5.123;rport=5062 From: <sip:1000@10.1.5.111>;tag=cf4e710-0-13c6-50022-c7cd3-7c4707a0-c7cd3 To: <sip:1000@10.1.5.111>;tag=as130a6fb8 Call-ID: cf5c9a8-0-13c6-50022-c7cd3-69566da-c7cd3 CSeq: 2 REGISTER Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 60 Contact: <sip:1000@10.1.5.111:5062>;expires=60 Date: Thu, 12 Apr 2012 11:17:14 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'cf5c9a8-0-13c6-50022-c7cd3-69566da-c7cd3' in 32000 ms (Method: REGISTER) pc3861*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 10.1.5.123 (None) cf5c9a8-0-13c6- 0x0 (nothing) No Rx: REGISTER <guest> 10.4.1.7 gglowack YzlhMmUzOGQxMTI 0x0 (nothing) No Rx: BYE gglowack 10.1.5.123 1000 7b5ae3db062ce03 0x0 (nothing) No Tx: INVITE 1000 3 active SIP dialogs Really destroying SIP dialog 'YzlhMmUzOGQxMTIzN2Q1MzNiYWUyOWU4MjA4NDMzZDU.' Method: BYE <--- SIP read from UDP:10.4.1.7:23632 ---> <-------------> <--- SIP read from UDP:10.1.5.123:5062 ---> REGISTER sip:10.1.5.111 SIP/2.0 From: <sip:1000@10.1.5.111>;tag=cf4e710-0-13c6-50022-c7cd3-7c4707a0-c7cd3 To: <sip:1000@10.1.5.111> Call-ID: cf5c9a8-0-13c6-50022-c7cd3-69566da-c7cd3 CSeq: 3 REGISTER Via: SIP/2.0/UDP 10.1.5.123:5062;branch=z9hG4bK-c7d14-30c89749-676897a1 Max-Forwards: 70 Expires: 60 Contact: <sip:1000@10.1.5.111:5062>;methods="INVITE, INFO, SUBSCRIBE, BYE, CANCEL, NOTIFY, ACK, REFER" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.1.5.123:5062 (NAT) [Apr 12 13:17:43] WARNING[2881]: db.c:115 dbinit: Unable to open Asterisk database '/var/lib/asterisk/astdb': Permission denied <--- Transmitting (NAT) to 10.1.5.123:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.5.123:5062;branch=z9hG4bK-c7d14-30c89749-676897a1;received=10.1.5.123;rport=5062 From: <sip:1000@10.1.5.111>;tag=cf4e710-0-13c6-50022-c7cd3-7c4707a0-c7cd3 To: <sip:1000@10.1.5.111>;tag=as130a6fb8 Call-ID: cf5c9a8-0-13c6-50022-c7cd3-69566da-c7cd3 CSeq: 3 REGISTER Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 60 Contact: <sip:1000@10.1.5.111:5062>;expires=60 Date: Thu, 12 Apr 2012 11:17:43 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'cf5c9a8-0-13c6-50022-c7cd3-69566da-c7cd3' in 32000 ms (Method: REGISTER) [Apr 12 13:17:50] NOTICE[2881]: chan_sip.c:13059 sip_reregister: -- Re-registration for fe80a5851fd131487557@myprovider REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 193.178.240.39:5060: REGISTER sip:myprovider SIP/2.0 Via: SIP/2.0/UDP 10.1.5.111:5060;branch=z9hG4bK6ceb0df8;rport Max-Forwards: 70 From: <sip:fe80a5851fd131487557@193.178.240.39>;tag=as34c576a4 To: <sip:fe80a5851fd131487557@193.178.240.39> Call-ID: 3b1c5b4308196cfe2acb450d27437f02@10.1.5.111 CSeq: 116 REGISTER User-Agent: Asterisk PBX 1.8.11.0 Authorization: Digest username="fe80a5851fd131487557", realm="asterisk", algorithm=MD5, uri="sip:myprovider", nonce="30e841cb", response="076330002427673feb3a3cf62ac7aa84" Expires: 120 Contact: <sip:s@10.1.5.111:5060> Content-Length: 0 --- <--- SIP read from UDP:193.178.240.39:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.1.5.111:5060;branch=z9hG4bK6ceb0df8;received=10.1.5.111;rport=5060 From: <sip:fe80a5851fd131487557@193.178.240.39>;tag=as34c576a4 To: <sip:fe80a5851fd131487557@193.178.240.39>;tag=as07e7f45d Call-ID: 3b1c5b4308196cfe2acb450d27437f02@10.1.5.111 CSeq: 116 REGISTER Server: SIP_PBX_WASKO_Berberckiego_6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0954362b" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Responding to challenge, registration to domain/host name myprovider REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 193.178.240.39:5060: REGISTER sip:myprovider SIP/2.0 Via: SIP/2.0/UDP 10.1.5.111:5060;branch=z9hG4bK2c981c3b;rport Max-Forwards: 70 From: <sip:fe80a5851fd131487557@193.178.240.39>;tag=as789d04c6 To: <sip:fe80a5851fd131487557@193.178.240.39> Call-ID: 3b1c5b4308196cfe2acb450d27437f02@10.1.5.111 CSeq: 117 REGISTER User-Agent: Asterisk PBX 1.8.11.0 Authorization: Digest username="fe80a5851fd131487557", realm="asterisk", algorithm=MD5, uri="sip:myprovider", nonce="0954362b", response="86c4398635e80d04503bb427feb6758b" Expires: 120 Contact: <sip:s@10.1.5.111:5060> Content-Length: 0 --- <--- SIP read from UDP:193.178.240.39:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.5.111:5060;branch=z9hG4bK2c981c3b;received=10.1.5.111;rport=5060 From: <sip:fe80a5851fd131487557@193.178.240.39>;tag=as789d04c6 To: <sip:fe80a5851fd131487557@193.178.240.39>;tag=as07e7f45d Call-ID: 3b1c5b4308196cfe2acb450d27437f02@10.1.5.111 CSeq: 117 REGISTER Server: SIP_PBX_WASKO_Berberckiego_6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 120 Contact: <sip:s@10.1.5.111:5060>;expires=120 Date: Thu, 12 Apr 2012 11:27:23 GMT Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Scheduling destruction of SIP dialog '3b1c5b4308196cfe2acb450d27437f02@10.1.5.111' in 32000 ms (Method: REGISTER) [Apr 12 13:17:50] NOTICE[2881]: chan_sip.c:20738 handle_response_register: Outbound Registration: Expiry for myprovider is 120 sec (Scheduling reregistration in 105 s) <--- SIP read from UDP:10.4.1.7:23632 ---> <-------------> <--- SIP read from UDP:10.1.5.123:5062 ---> REGISTER sip:10.1.5.111 SIP/2.0 From: <sip:1000@10.1.5.111>;tag=cf4e710-0-13c6-50022-c7cd3-7c4707a0-c7cd3 To: <sip:1000@10.1.5.111> Call-ID: cf5c9a8-0-13c6-50022-c7cd3-69566da-c7cd3 CSeq: 4 REGISTER Via: SIP/2.0/UDP 10.1.5.123:5062;branch=z9hG4bK-c7d33-30c91152-5f26be4f Max-Forwards: 70 Expires: 60 Contact: <sip:1000@10.1.5.111:5062>;methods="INVITE, INFO, SUBSCRIBE, BYE, CANCEL, NOTIFY, ACK, REFER" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.1.5.123:5062 (NAT) [Apr 12 13:18:10] WARNING[2881]: db.c:115 dbinit: Unable to open Asterisk database '/var/lib/asterisk/astdb': Permission denied <--- Transmitting (NAT) to 10.1.5.123:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.5.123:5062;branch=z9hG4bK-c7d33-30c91152-5f26be4f;received=10.1.5.123;rport=5062 From: <sip:1000@10.1.5.111>;tag=cf4e710-0-13c6-50022-c7cd3-7c4707a0-c7cd3 To: <sip:1000@10.1.5.111>;tag=as130a6fb8 Call-ID: cf5c9a8-0-13c6-50022-c7cd3-69566da-c7cd3 CSeq: 4 REGISTER Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 60 Contact: <sip:1000@10.1.5.111:5062>;expires=60 Date: Thu, 12 Apr 2012 11:18:10 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'cf5c9a8-0-13c6-50022-c7cd3-69566da-c7cd3' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.1.5.123:5062 ---> BYE sip:gglowack@10.1.5.111:5060 SIP/2.0 From: <sip:1000@10.1.5.111:5062>;tag=cf4e8a8-0-13c6-50022-c7cde-2719b0dd-c7cde To: "gglowack"<sip:gglowack@10.1.5.111>;tag=as5b5c2f32 Call-ID: 7b5ae3db062ce0387f5a07e35660dd14@10.1.5.111:5060 CSeq: 1 BYE Via: SIP/2.0/UDP 10.1.5.123:5062;branch=z9hG4bK-c7d33-30c91153-540ba1dd Max-Forwards: 70 Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 10.1.5.123:5062 (NAT) Scheduling destruction of SIP dialog '7b5ae3db062ce0387f5a07e35660dd14@10.1.5.111:5060' in 32000 ms (Method: BYE) <--- Transmitting (NAT) to 10.1.5.123:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.5.123:5062;branch=z9hG4bK-c7d33-30c91153-540ba1dd;received=10.1.5.123;rport=5062 From: <sip:1000@10.1.5.111:5062>;tag=cf4e8a8-0-13c6-50022-c7cde-2719b0dd-c7cde To: "gglowack"<sip:gglowack@10.1.5.111>;tag=as5b5c2f32 Call-ID: 7b5ae3db062ce0387f5a07e35660dd14@10.1.5.111:5060 CSeq: 1 BYE Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> <--- SIP read from UDP:10.1.5.123:5062 ---> REGISTER sip:10.1.5.111 SIP/2.0 From: <sip:1000@10.1.5.111>;tag=cf4e710-0-13c6-50022-c7cd3-7c4707a0-c7cd3 To: <sip:1000@10.1.5.111> Call-ID: cf5c9a8-0-13c6-50022-c7cd3-69566da-c7cd3 CSeq: 5 REGISTER Via: SIP/2.0/UDP 10.1.5.123:5062;branch=z9hG4bK-c7d33-30c91154-7d76304b Max-Forwards: 70 Expires: 0 Contact: <sip:1000@10.1.5.111:5062>;expires=0;methods="INVITE, INFO, SUBSCRIBE, BYE, CANCEL, NOTIFY, ACK, REFER" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.1.5.123:5062 (NAT) -- Unregistered SIP '1000' [Apr 12 13:18:10] WARNING[2881]: db.c:115 dbinit: Unable to open Asterisk database '/var/lib/asterisk/astdb': Permission denied [Apr 12 13:18:10] WARNING[2881]: db.c:115 dbinit: Unable to open Asterisk database '/var/lib/asterisk/astdb': Permission denied <--- Transmitting (NAT) to 10.1.5.123:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.5.123:5062;branch=z9hG4bK-c7d33-30c91154-7d76304b;received=10.1.5.123;rport=5062 From: <sip:1000@10.1.5.111>;tag=cf4e710-0-13c6-50022-c7cd3-7c4707a0-c7cd3 To: <sip:1000@10.1.5.111>;tag=as130a6fb8 Call-ID: cf5c9a8-0-13c6-50022-c7cd3-69566da-c7cd3 CSeq: 5 REGISTER Server: Asterisk PBX 1.8.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 60 Contact: <sip:1000@10.1.5.111:5062>;expires=60 Date: Thu, 12 Apr 2012 11:18:10 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'cf5c9a8-0-13c6-50022-c7cd3-69566da-c7cd3' in 32000 ms (Method: REGISTER) pc3861*CLI> quit [root@pc3861 asterisk]# | ||||
| Comments: | By: Grzegorz Głowacki (g.glowacki) 2012-04-12 06:45:06.503-0500 Wireshark communication dump. By: Grzegorz Głowacki (g.glowacki) 2012-04-12 06:51:21.359-0500 Asterisk config files. By: Grzegorz Głowacki (g.glowacki) 2012-04-12 07:27:15.403-0500 This ticket might be related to [ASTERISK-19425] By: Matt Jordan (mjordan) 2012-09-24 16:11:03.646-0500 Can you confirm if this is still a problem in versions 1.8.14.1 or later? By: Matt Jordan (mjordan) 2012-10-17 09:11:02.448-0500 Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines | ||||