[Apr 14 17:03:30] VERBOSE[7943] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log [Apr 14 17:03:30] VERBOSE[7943] loader.c: Asterisk Dynamic Loader Starting: [Apr 14 17:03:30] VERBOSE[7943] pbx.c: Asterisk PBX Core Initializing [Apr 14 17:03:30] VERBOSE[7943] pbx.c: Registering builtin applications: [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [Answer] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [BackGround] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [Busy] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [Congestion] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [ExecIfTime] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [Goto] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [GotoIf] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [GotoIfTime] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [ImportVar] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [Hangup] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [Incomplete] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [NoOp] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [Proceeding] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [Progress] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [RaiseException] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [ResetCDR] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [Ringing] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [SayAlpha] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [SayDigits] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [SayNumber] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [SayPhonetic] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [Set] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [MSet] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [SetAMAFlags] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [Wait] [Apr 14 17:03:30] VERBOSE[7943] pbx.c: [WaitExten] [Apr 14 17:03:30] VERBOSE[7943] loader.c: Asterisk Dynamic Loader Starting: [Apr 14 17:03:30] VERBOSE[7943] loader.c: res_indications.so => (Region-specific tones) [Apr 14 17:03:30] VERBOSE[7943] loader.c: res_monitor.so => (Call Monitoring Resource) [Apr 14 17:03:30] VERBOSE[7943] loader.c: res_agi.so => (Asterisk Gateway Interface (AGI)) [Apr 14 17:03:30] VERBOSE[7943] loader.c: pbx_config.so => (Text Extension Configuration) [Apr 14 17:03:30] VERBOSE[7943] loader.c: res_adsi.so => (ADSI Resource) [Apr 14 17:03:30] VERBOSE[7943] loader.c: res_config_mysql.so => (MySQL RealTime Configuration Driver) [Apr 14 17:03:30] VERBOSE[7943] loader.c: res_musiconhold.so => (Music On Hold Resource) [Apr 14 17:03:30] VERBOSE[7943] loader.c: res_clioriginate.so => (Call origination from the CLI) [Apr 14 17:03:30] VERBOSE[7943] loader.c: res_realtime.so => (Realtime Data Lookup/Rewrite) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_addon_sql_mysql.so => (Simple Mysql Interface) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_authenticate.so => (Authentication Application) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_channelredirect.so => (Redirects a given channel to a dialplan target) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_cdr.so => (Tell Asterisk to not maintain a CDR for the current call) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_controlplayback.so => (Control Playback Application) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_db.so => (Database Access Functions) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_dial.so => (Dialing Application) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_directed_pickup.so => (Directed Call Pickup Application) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_echo.so => (Simple Echo Application) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_externalivr.so => (External IVR Interface Application) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_forkcdr.so => (Fork The CDR into 2 separate entities) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_macro.so => (Extension Macros) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_mp3.so => (Silly MP3 Application) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_mixmonitor.so => (Mixed Audio Monitoring Application) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_meetme.so => (MeetMe conference bridge) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_playback.so => (Sound File Playback Application) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_read.so => (Read Variable Application) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_sayunixtime.so => (Say time) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_stack.so => (Dialplan subroutines (Gosub, Return, etc)) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_system.so => (Generic System() application) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_transfer.so => (Transfers a caller to another extension) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_userevent.so => (Custom User Event Application) [Apr 14 17:03:30] VERBOSE[7943] loader.c: app_verbose.so => (Send verbose output) [Apr 14 17:03:30] VERBOSE[7943] loader.c: cdr_custom.so => (Customizable Comma Separated Values CDR Backend) [Apr 14 17:03:30] VERBOSE[7943] loader.c: cdr_addon_mysql.so => (MySQL CDR Backend) [Apr 14 17:03:30] VERBOSE[7943] chan_sip.c: SIP channel loading... [Apr 14 17:03:30] VERBOSE[7943] loader.c: chan_sip.so => (Session Initiation Protocol (SIP)) [Apr 14 17:03:30] VERBOSE[7943] loader.c: chan_dahdi.so => (DAHDI Telephony) [Apr 14 17:03:30] VERBOSE[7943] loader.c: chan_local.so => (Local Proxy Channel (Note: used internally by other modules)) [Apr 14 17:03:30] VERBOSE[7943] loader.c: codec_adpcm.so => (Adaptive Differential PCM Coder/Decoder) [Apr 14 17:03:30] VERBOSE[7943] loader.c: codec_alaw.so => (A-law Coder/Decoder) [Apr 14 17:03:30] VERBOSE[7943] loader.c: codec_a_mu.so => (A-law and Mulaw direct Coder/Decoder) [Apr 14 17:03:30] VERBOSE[7943] loader.c: codec_g726.so => (ITU G.726-32kbps G726 Transcoder) [Apr 14 17:03:30] VERBOSE[7943] loader.c: codec_gsm.so => (GSM Coder/Decoder) [Apr 14 17:03:30] VERBOSE[7943] loader.c: codec_lpc10.so => (LPC10 2.4kbps Coder/Decoder) [Apr 14 17:03:30] VERBOSE[7943] loader.c: codec_speex.so => (Speex Coder/Decoder) [Apr 14 17:03:30] VERBOSE[7943] loader.c: codec_ulaw.so => (mu-Law Coder/Decoder) [Apr 14 17:03:30] VERBOSE[7943] loader.c: format_g723.so => (G.723.1 Simple Timestamp File Format) [Apr 14 17:03:30] VERBOSE[7943] loader.c: format_g726.so => (Raw G.726 (16/24/32/40kbps) data) [Apr 14 17:03:30] VERBOSE[7943] loader.c: format_g729.so => (Raw G729 data) [Apr 14 17:03:30] VERBOSE[7943] loader.c: format_gsm.so => (Raw GSM data) [Apr 14 17:03:30] VERBOSE[7943] loader.c: format_h263.so => (Raw H.263 data) [Apr 14 17:03:30] VERBOSE[7943] loader.c: format_h264.so => (Raw H.264 data) [Apr 14 17:03:30] VERBOSE[7943] loader.c: format_ilbc.so => (Raw iLBC data) [Apr 14 17:03:30] VERBOSE[7943] loader.c: format_mp3.so => (MP3 format [Any rate but 8000hz mono is optimal]) [Apr 14 17:03:30] VERBOSE[7943] loader.c: format_pcm.so => (Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G.722 16Khz) [Apr 14 17:03:30] VERBOSE[7943] loader.c: format_sln.so => (Raw Signed Linear Audio support (SLN)) [Apr 14 17:03:30] VERBOSE[7943] loader.c: format_vox.so => (Dialogic VOX (ADPCM) File Format) [Apr 14 17:03:30] VERBOSE[7943] loader.c: format_wav_gsm.so => (Microsoft WAV format (Proprietary GSM)) [Apr 14 17:03:30] VERBOSE[7943] loader.c: format_wav.so => (Microsoft WAV format (8000Hz Signed Linear)) [Apr 14 17:03:30] VERBOSE[7943] loader.c: func_callerid.so => (Caller ID related dialplan functions) [Apr 14 17:03:30] VERBOSE[7943] loader.c: func_cdr.so => (Call Detail Record (CDR) dialplan function) [Apr 14 17:03:30] VERBOSE[7943] loader.c: func_channel.so => (Channel information dialplan functions) [Apr 14 17:03:30] VERBOSE[7943] loader.c: func_cut.so => (Cut out information from a string) [Apr 14 17:03:30] VERBOSE[7943] loader.c: func_db.so => (Database (astdb) related dialplan functions) [Apr 14 17:03:30] VERBOSE[7943] loader.c: func_logic.so => (Logical dialplan functions) [Apr 14 17:03:30] VERBOSE[7943] loader.c: func_math.so => (Mathematical dialplan function) [Apr 14 17:03:30] VERBOSE[7943] loader.c: func_enum.so => (ENUM related dialplan functions) [Apr 14 17:03:30] VERBOSE[7943] loader.c: func_strings.so => (String handling dialplan functions) [Apr 14 17:03:30] VERBOSE[7943] loader.c: func_timeout.so => (Channel timeout dialplan functions) [Apr 14 17:03:30] VERBOSE[7943] loader.c: func_uri.so => (URI encode/decode dialplan functions) [Apr 14 17:03:30] VERBOSE[7943] asterisk.c: Asterisk Ready. [Apr 14 17:03:49] VERBOSE[7950] chan_sip.c: <--- SIP read from UDP://78.105.1.127:5060 ---> INVITE sip:448704711057@dev-sip.wima.co.uk SIP/2.0 Record-Route: Via: SIP/2.0/UDP 78.105.1.127;branch=z9hG4bKb165.2d2af636.0 Via: SIP/2.0/UDP 78.105.1.124:5060;received=78.105.1.124;branch=z9hG4bK57e2e12c;rport=5060 Max-Forwards: 69 From: "447854740947" ;tag=as26710683 To: Contact: Call-ID: 3bca0d1926aec9e51d611a0f0bd5870b@78.105.1.124 CSeq: 102 INVITE User-Agent: From PSTN Date: Tue, 14 Apr 2009 16:04:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 280 v=0 o=root 1482953366 1482953366 IN IP4 78.105.1.124 s=Asterisk PBX SVN-branch-1.6.1-r182123 c=IN IP4 78.105.1.124 t=0 0 m=audio 13438 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Apr 14 17:03:49] VERBOSE[7950] chan_sip.c: --- (16 headers 12 lines) --- [Apr 14 17:03:49] VERBOSE[7950] netsock.c: == Using SIP RTP CoS mark 5 [Apr 14 17:03:49] VERBOSE[7950] chan_sip.c: Sending to 78.105.1.127 : 5060 (no NAT) [Apr 14 17:03:49] VERBOSE[7950] chan_sip.c: Using INVITE request as basis request - 3bca0d1926aec9e51d611a0f0bd5870b@78.105.1.124 [Apr 14 17:03:49] VERBOSE[7950] chan_sip.c: No matching peer for '447854740947' from '78.105.1.127:5060' [Apr 14 17:03:49] VERBOSE[7950] chan_sip.c: Found RTP audio format 8 [Apr 14 17:03:49] VERBOSE[7950] chan_sip.c: Found RTP audio format 101 [Apr 14 17:03:49] VERBOSE[7950] chan_sip.c: Peer audio RTP is at port 78.105.1.124:13438 [Apr 14 17:03:49] VERBOSE[7950] chan_sip.c: Found audio description format PCMA for ID 8 [Apr 14 17:03:49] VERBOSE[7950] chan_sip.c: Found audio description format telephone-event for ID 101 [Apr 14 17:03:49] VERBOSE[7950] chan_sip.c: Got unsupported a:fmtp in SDP offer [Apr 14 17:03:49] VERBOSE[7950] chan_sip.c: Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Apr 14 17:03:49] VERBOSE[7950] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Apr 14 17:03:49] VERBOSE[7950] chan_sip.c: Peer audio RTP is at port 78.105.1.124:13438 [Apr 14 17:03:49] VERBOSE[7950] chan_sip.c: Looking for 448704711057 in common (domain dev-sip.wima.co.uk) [Apr 14 17:03:49] VERBOSE[7950] chan_sip.c: list_route: hop: [Apr 14 17:03:49] VERBOSE[7950] chan_sip.c: <--- Transmitting (no NAT) to 78.105.1.127:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 78.105.1.127;branch=z9hG4bKb165.2d2af636.0;received=78.105.1.127 Via: SIP/2.0/UDP 78.105.1.124:5060;received=78.105.1.124;branch=z9hG4bK57e2e12c;rport=5060 Record-Route: From: "447854740947" ;tag=as26710683 To: Call-ID: 3bca0d1926aec9e51d611a0f0bd5870b@78.105.1.124 CSeq: 102 INVITE Server: Media GW 1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> [Apr 14 17:03:49] VERBOSE[7954] pbx.c: -- Executing [448704711057@common:1] Set("SIP/78.105.1.124-08222338", "DB(ringing/channel)=SIP/78.105.1.124-08222338") in new stack [Apr 14 17:03:50] VERBOSE[7954] pbx.c: -- Executing [448704711057@common:2] Dial("SIP/78.105.1.124-08222338", "SIP/10000@dev-sip.wima.co.uk,30") in new stack [Apr 14 17:03:50] VERBOSE[7954] netsock.c: == Using SIP RTP CoS mark 5 [Apr 14 17:03:50] VERBOSE[7954] chan_sip.c: Audio is at 78.105.1.128 port 11898 [Apr 14 17:03:50] VERBOSE[7954] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Apr 14 17:03:50] VERBOSE[7954] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Apr 14 17:03:50] VERBOSE[7954] chan_sip.c: Adding codec 0x2 (gsm) to SDP [Apr 14 17:03:50] VERBOSE[7954] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Apr 14 17:03:50] VERBOSE[7954] chan_sip.c: Reliably Transmitting (no NAT) to 78.105.1.127:5060: INVITE sip:10000@dev-sip.wima.co.uk SIP/2.0 Via: SIP/2.0/UDP 78.105.1.128:5060;branch=z9hG4bK3c16212c;rport Max-Forwards: 70 From: "447854740947" ;tag=as22834c50 To: Contact: Call-ID: 4dc300e2602a0225416aa487725328f4@78.105.1.128 CSeq: 102 INVITE User-Agent: Media GW 1 Date: Tue, 14 Apr 2009 16:03:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 298 v=0 o=root 321599475 321599475 IN IP4 78.105.1.128 s=Media GW 1 c=IN IP4 78.105.1.128 t=0 0 m=audio 11898 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Apr 14 17:03:50] VERBOSE[7954] app_dial.c: -- Called 10000@dev-sip.wima.co.uk [Apr 14 17:03:50] VERBOSE[7950] chan_sip.c: <--- SIP read from UDP://78.105.1.127:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 78.105.1.128:5060;branch=z9hG4bK3c16212c;rport=5060 From: "447854740947" ;tag=as22834c50 To: Call-ID: 4dc300e2602a0225416aa487725328f4@78.105.1.128 CSeq: 102 INVITE Server: SIP Proxy 1 Content-Length: 0 <-------------> [Apr 14 17:03:50] VERBOSE[7950] chan_sip.c: --- (8 headers 0 lines) --- [Apr 14 17:03:50] VERBOSE[7950] chan_sip.c: <--- SIP read from UDP://78.105.1.127:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 78.105.1.128:5060;branch=z9hG4bK3c16212c;rport=5060 From: "447854740947" ;tag=as22834c50 To: Call-ID: 4dc300e2602a0225416aa487725328f4@78.105.1.128 CSeq: 102 INVITE Server: SIP Proxy 1 Content-Length: 0 <-------------> [Apr 14 17:03:50] VERBOSE[7950] chan_sip.c: --- (8 headers 0 lines) --- [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: <--- SIP read from UDP://78.105.1.127:5060 ---> INVITE sip:**@dev-sip.wima.co.uk SIP/2.0 Record-Route: Via: SIP/2.0/UDP 78.105.1.127;branch=z9hG4bK6c2b.32a25ad3.0 Via: SIP/2.0/UDP 78.105.1.131:5060;received=78.105.1.131;rport=5060;branch=z9hG4bKjfqujefv Max-Forwards: 69 To: From: "10002" ;tag=mnpxu Call-ID: jpenyjenkpwjdmk@salon CSeq: 925 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.2 Content-Length: 308 X-auth: 10002 X-Error: 501 v=0 o=twinkle 1162642365 292612326 IN IP4 192.168.7.10 s=- c=IN IP4 78.105.1.131 t=0 0 m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: --- (17 headers 14 lines) --- [Apr 14 17:03:52] VERBOSE[7950] netsock.c: == Using SIP RTP CoS mark 5 [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Sending to 78.105.1.127 : 5060 (no NAT) [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Using INVITE request as basis request - jpenyjenkpwjdmk@salon [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: No matching peer for '10002' from '78.105.1.127:5060' [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Found RTP audio format 98 [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Found RTP audio format 97 [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Found RTP audio format 8 [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Found RTP audio format 0 [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Found RTP audio format 3 [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Found RTP audio format 101 [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Peer audio RTP is at port 78.105.1.131:8000 [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Found audio description format speex for ID 98 [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Found audio description format speex for ID 97 [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Found audio description format PCMA for ID 8 [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Found audio description format PCMU for ID 0 [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Found audio description format GSM for ID 3 [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Found audio description format telephone-event for ID 101 [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Got unsupported a:fmtp in SDP offer [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer - audio=0x20e (gsm|ulaw|alaw|speex)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Peer audio RTP is at port 78.105.1.131:8000 [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: Looking for ** in common (domain dev-sip.wima.co.uk) [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: list_route: hop: [Apr 14 17:03:52] VERBOSE[7950] chan_sip.c: <--- Transmitting (no NAT) to 78.105.1.127:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 78.105.1.127;branch=z9hG4bK6c2b.32a25ad3.0;received=78.105.1.127 Via: SIP/2.0/UDP 78.105.1.131:5060;received=78.105.1.131;rport=5060;branch=z9hG4bKjfqujefv Record-Route: From: "10002" ;tag=mnpxu To: Call-ID: jpenyjenkpwjdmk@salon CSeq: 925 INVITE Server: Media GW 1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> [Apr 14 17:03:52] VERBOSE[7955] pbx.c: -- Executing [**@common:1] Set("SIP/dev-sip.wima.co.uk-0822a1b0", "X-RingingChannel=SIP/78.105.1.124-08222338") in new stack [Apr 14 17:03:52] VERBOSE[7955] pbx.c: -- Executing [**@common:2] NoOp("SIP/dev-sip.wima.co.uk-0822a1b0", "Ringing: SIP/78.105.1.124-08222338") in new stack [Apr 14 17:03:52] VERBOSE[7955] pbx.c: -- Executing [**@common:3] Bridge("SIP/dev-sip.wima.co.uk-0822a1b0", "SIP/78.105.1.124-08222338") in new stack [Apr 14 17:03:52] VERBOSE[7955] chan_sip.c: Audio is at 78.105.1.128 port 14082 [Apr 14 17:03:52] VERBOSE[7955] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Apr 14 17:03:52] VERBOSE[7955] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Apr 14 17:03:52] VERBOSE[7955] chan_sip.c: <--- Reliably Transmitting (no NAT) to 78.105.1.127:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 78.105.1.127;branch=z9hG4bKb165.2d2af636.0;received=78.105.1.127 Via: SIP/2.0/UDP 78.105.1.124:5060;received=78.105.1.124;branch=z9hG4bK57e2e12c;rport=5060 Record-Route: From: "447854740947" ;tag=as26710683 To: ;tag=as7fa7d635 Call-ID: 3bca0d1926aec9e51d611a0f0bd5870b@78.105.1.124 CSeq: 102 INVITE Server: Media GW 1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 253 v=0 o=root 1180464738 1180464738 IN IP4 78.105.1.128 s=Media GW 1 c=IN IP4 78.105.1.128 t=0 0 m=audio 14082 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Apr 14 17:03:53] VERBOSE[7950] chan_sip.c: <--- SIP read from UDP://78.105.1.127:5060 ---> ACK sip:448704711057@78.105.1.128 SIP/2.0 Via: SIP/2.0/UDP 78.105.1.127;branch=z9hG4bKb165.2d2af636.2 Via: SIP/2.0/UDP 78.105.1.124:5060;received=78.105.1.124;branch=z9hG4bK481b5acb;rport=5060 Max-Forwards: 69 From: "447854740947" ;tag=as26710683 To: ;tag=as7fa7d635 Contact: Call-ID: 3bca0d1926aec9e51d611a0f0bd5870b@78.105.1.124 CSeq: 102 ACK User-Agent: From PSTN Content-Length: 0 <-------------> [Apr 14 17:03:53] VERBOSE[7950] chan_sip.c: --- (11 headers 0 lines) --- [Apr 14 17:03:53] VERBOSE[7955] chan_sip.c: Audio is at 78.105.1.128 port 15094 [Apr 14 17:03:53] VERBOSE[7955] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Apr 14 17:03:53] VERBOSE[7955] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Apr 14 17:03:53] VERBOSE[7955] chan_sip.c: Adding codec 0x2 (gsm) to SDP [Apr 14 17:03:53] VERBOSE[7955] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Apr 14 17:03:53] VERBOSE[7955] chan_sip.c: <--- Reliably Transmitting (no NAT) to 78.105.1.127:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 78.105.1.127;branch=z9hG4bK6c2b.32a25ad3.0;received=78.105.1.127 Via: SIP/2.0/UDP 78.105.1.131:5060;received=78.105.1.131;rport=5060;branch=z9hG4bKjfqujefv Record-Route: From: "10002" ;tag=mnpxu To: ;tag=as3fd6811b Call-ID: jpenyjenkpwjdmk@salon CSeq: 925 INVITE Server: Media GW 1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 300 v=0 o=root 1480164571 1480164571 IN IP4 78.105.1.128 s=Media GW 1 c=IN IP4 78.105.1.128 t=0 0 m=audio 15094 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Apr 14 17:03:53] VERBOSE[7955] rtp.c: -- Packet2Packet bridging SIP/dev-sip.wima.co.uk-0822a1b0 and SIP/78.105.1.124-08222338 [Apr 14 17:03:53] VERBOSE[7950] chan_sip.c: <--- SIP read from UDP://78.105.1.127:5060 ---> ACK sip:**@78.105.1.128 SIP/2.0 Via: SIP/2.0/UDP 78.105.1.127;branch=z9hG4bK6c2b.32a25ad3.2 Via: SIP/2.0/UDP 78.105.1.131:5060;received=78.105.1.131;rport=5060;branch=z9hG4bKsnhaoplp Max-Forwards: 69 Proxy-Authorization: Digest username="10002",realm="dev-sip.wima.co.uk",nonce="49e4b4060000003bf3e93e4a3ce4f306504d604f7104b692",uri="sip:**@dev-sip.wima.co.uk",response="905f2fc3e619fb919821b1d2a2528277",algorithm=MD5 To: ;tag=as3fd6811b From: "10002" ;tag=mnpxu Call-ID: jpenyjenkpwjdmk@salon CSeq: 925 ACK User-Agent: Twinkle/1.2 Content-Length: 0 <-------------> [Apr 14 17:03:53] VERBOSE[7950] chan_sip.c: --- (11 headers 0 lines) ---