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- [Dec 7 17:13:38] VERBOSE[8075] config.c: == Parsing '/etc/asterisk/logger.conf': [Dec 7 17:13:38] VERBOSE[8075] config.c: == Found
- [Dec 7 17:13:38] VERBOSE[8075] logger.c: Asterisk Event Logger restarted
- [Dec 7 17:13:38] VERBOSE[8075] logger.c: Asterisk Queue Logger restarted
- [Dec 7 17:13:42] NOTICE[2984] chan_sip.c: -- Re-registration for mlsmith@inbound24.vitelity.net
- [Dec 7 17:13:42] VERBOSE[2984] dnsmgr.c: > doing dnsmgr_lookup for 'inbound24.vitelity.net'
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: REGISTER 11 headers, 0 lines
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Reliably Transmitting (no NAT) to 66.241.96.164:5060:
- REGISTER sip:inbound24.vitelity.net SIP/2.0
- Via: SIP/2.0/UDP 192.168.230.252:5060;branch=z9hG4bK4d8deaf7;rport
- Max-Forwards: 70
- From: <sip:mlsmith@inbound24.vitelity.net>;tag=as755488b2
- To: <sip:mlsmith@inbound24.vitelity.net>
- Call-ID: 763abf1775ce87cd1faff787433f7218@127.0.0.1
- CSeq: 1078 REGISTER
- User-Agent: Asterisk PBX 1.6.2.11
- Authorization: Digest username="mlsmith", realm="asterisk", algorithm=MD5, uri="sip:inbound24.vitelity.net", nonce="09d2aff0", response="e5bd28812d71291b263e2feb64de4b37"
- Expires: 120
- Contact: <sip:s@192.168.230.252>
- Content-Length: 0
- ---
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c:
- <--- SIP read from UDP:192.168.230.250:5060 --->
- INVITE sip:7794184@192.168.230.252 SIP/2.0
- Via: SIP/2.0/UDP 192.168.230.250:5060;branch=z9hG4bK-6bcf9061
- From: "Office Shannon" <sip:601@192.168.230.252>;tag=ad987cf4205f8be9o0
- To: <sip:7794184@192.168.230.252>
- Call-ID: 65c63e10-42695d7d@192.168.230.250
- CSeq: 101 INVITE
- Max-Forwards: 70
- Contact: "Office Shannon" <sip:601@192.168.230.250:5060>
- Expires: 240
- User-Agent: Cisco/SPA504G-7.4.3a
- Content-Length: 403
- Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
- Supported: replaces
- Content-Type: application/sdp
- v=0
- o=- 7066062 7066062 IN IP4 192.168.230.250
- s=-
- c=IN IP4 192.168.230.250
- t=0 0
- m=audio 16466 RTP/AVP 0 2 8 9 18 96 97 98 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:2 G726-32/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:9 G722/8000
- a=rtpmap:18 G729a/8000
- a=rtpmap:96 G726-40/8000
- a=rtpmap:97 G726-24/8000
- a=rtpmap:98 G726-16/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-15
- a=ptime:30
- a=sendrecv
- <------------->
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: --- (14 headers 18 lines) ---
- [Dec 7 17:13:42] VERBOSE[2984] netsock.c: == Using SIP RTP TOS bits 184
- [Dec 7 17:13:42] VERBOSE[2984] netsock.c: == Using SIP RTP CoS mark 5
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Sending to 192.168.230.250 : 5060 (no NAT)
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Using INVITE request as basis request - 65c63e10-42695d7d@192.168.230.250
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found peer '601' for '601' from 192.168.230.250:5060
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c:
- <--- Reliably Transmitting (NAT) to 192.168.230.250:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/UDP 192.168.230.250:5060;branch=z9hG4bK-6bcf9061;received=192.168.230.250
- From: "Office Shannon" <sip:601@192.168.230.252>;tag=ad987cf4205f8be9o0
- To: <sip:7794184@192.168.230.252>;tag=as3b2cd2bd
- Call-ID: 65c63e10-42695d7d@192.168.230.250
- CSeq: 101 INVITE
- Server: Asterisk PBX 1.6.2.11
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="27563de7"
- Content-Length: 0
- <------------>
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Scheduling destruction of SIP dialog '65c63e10-42695d7d@192.168.230.250' in 6400 ms (Method: INVITE)
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c:
- <--- SIP read from UDP:192.168.230.250:5060 --->
- ACK sip:7794184@192.168.230.252 SIP/2.0
- Via: SIP/2.0/UDP 192.168.230.250:5060;branch=z9hG4bK-6bcf9061
- From: "Office Shannon" <sip:601@192.168.230.252>;tag=ad987cf4205f8be9o0
- To: <sip:7794184@192.168.230.252>;tag=as3b2cd2bd
- Call-ID: 65c63e10-42695d7d@192.168.230.250
- CSeq: 101 ACK
- Max-Forwards: 70
- Contact: "Office Shannon" <sip:601@192.168.230.250:5060>
- User-Agent: Cisco/SPA504G-7.4.3a
- Content-Length: 0
- <------------->
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: --- (10 headers 0 lines) ---
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c:
- <--- SIP read from UDP:192.168.230.250:5060 --->
- INVITE sip:7794184@192.168.230.252 SIP/2.0
- Via: SIP/2.0/UDP 192.168.230.250:5060;branch=z9hG4bK-23822c49
- From: "Office Shannon" <sip:601@192.168.230.252>;tag=ad987cf4205f8be9o0
- To: <sip:7794184@192.168.230.252>
- Call-ID: 65c63e10-42695d7d@192.168.230.250
- CSeq: 102 INVITE
- Max-Forwards: 70
- Authorization: Digest username="601",realm="asterisk",nonce="27563de7",uri="sip:7794184@192.168.230.252",algorithm=MD5,response="fc48f65a3908664c8c04567c071484fa"
- Contact: "Office Shannon" <sip:601@192.168.230.250:5060>
- Expires: 240
- User-Agent: Cisco/SPA504G-7.4.3a
- Content-Length: 403
- Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
- Supported: replaces
- Content-Type: application/sdp
- v=0
- o=- 7066062 7066062 IN IP4 192.168.230.250
- s=-
- c=IN IP4 192.168.230.250
- t=0 0
- m=audio 16466 RTP/AVP 0 2 8 9 18 96 97 98 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:2 G726-32/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:9 G722/8000
- a=rtpmap:18 G729a/8000
- a=rtpmap:96 G726-40/8000
- a=rtpmap:97 G726-24/8000
- a=rtpmap:98 G726-16/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-15
- a=ptime:30
- a=sendrecv
- <------------->
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: --- (15 headers 18 lines) ---
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Sending to 192.168.230.250 : 5060 (NAT)
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Using INVITE request as basis request - 65c63e10-42695d7d@192.168.230.250
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found peer '601' for '601' from 192.168.230.250:5060
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found RTP audio format 0
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found RTP audio format 2
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found RTP audio format 8
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found RTP audio format 9
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found RTP audio format 18
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found RTP audio format 96
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found RTP audio format 97
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found RTP audio format 98
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found RTP audio format 101
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found audio description format PCMU for ID 0
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found audio description format G726-32 for ID 2
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found audio description format PCMA for ID 8
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found audio description format G722 for ID 9
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found audio description format G729a for ID 18
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found audio description format G726-40 for ID 96
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found audio description format G726-24 for ID 97
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found audio description format G726-16 for ID 98
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Found audio description format telephone-event for ID 101
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x101d0c (ulaw|alaw|g726|g729|ilbc|g722|h263p)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Peer audio RTP is at port 192.168.230.250:16466
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Looking for 7794184 in from-internal (domain 192.168.230.252)
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: list_route: hop: <sip:601@192.168.230.250:5060>
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c:
- <--- Transmitting (NAT) to 192.168.230.250:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 192.168.230.250:5060;branch=z9hG4bK-23822c49;received=192.168.230.250
- From: "Office Shannon" <sip:601@192.168.230.252>;tag=ad987cf4205f8be9o0
- To: <sip:7794184@192.168.230.252>
- Call-ID: 65c63e10-42695d7d@192.168.230.250
- CSeq: 102 INVITE
- Server: Asterisk PBX 1.6.2.11
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Contact: <sip:7794184@192.168.230.252>
- Content-Length: 0
- <------------>
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [7794184@from-internal:1] Macro("SIP/601-000001c6", "user-callerid,SKIPTTL,") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-user-callerid:1] Set("SIP/601-000001c6", "AMPUSER=601") in new stack
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c:
- <--- SIP read from UDP:66.241.96.164:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 192.168.230.252:5060;branch=z9hG4bK4d8deaf7;received=192.168.230.252;rport=5060
- From: <sip:mlsmith@inbound24.vitelity.net>;tag=as755488b2
- To: <sip:mlsmith@inbound24.vitelity.net>
- Call-ID: 763abf1775ce87cd1faff787433f7218@127.0.0.1
- CSeq: 1078 REGISTER
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- <------------->
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: --- (10 headers 0 lines) ---
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/601-000001c6", "0?report") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/601-000001c6", "1?Set(REALCALLERIDNUM=601)") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-user-callerid:4] Set("SIP/601-000001c6", "AMPUSER=601") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-user-callerid:5] Set("SIP/601-000001c6", "AMPUSERCIDNAME=Shannon Office 601") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/601-000001c6", "0?report") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-user-callerid:7] Set("SIP/601-000001c6", "AMPUSERCID=601") in new stack
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c:
- <--- SIP read from UDP:66.241.96.164:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/UDP 192.168.230.252:5060;branch=z9hG4bK4d8deaf7;received=192.168.230.252;rport=5060
- From: <sip:mlsmith@inbound24.vitelity.net>;tag=as755488b2
- To: <sip:mlsmith@inbound24.vitelity.net>;tag=as453ab499
- Call-ID: 763abf1775ce87cd1faff787433f7218@127.0.0.1
- CSeq: 1078 REGISTER
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="60867419"
- Content-Length: 0
- <------------->
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: --- (11 headers 0 lines) ---
- [Dec 7 17:13:42] VERBOSE[2984] chan_sip.c: Responding to challenge, registration to domain/host name inbound24.vitelity.net
- [Dec 7 17:13:42] VERBOSE[2984] dnsmgr.c: > doing dnsmgr_lookup for 'inbound24.vitelity.net'
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-user-callerid:8] Set("SIP/601-000001c6", "CALLERID(all)="Shannon Office 601" <601>") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-user-callerid:9] ExecIf("SIP/601-000001c6", "0?Set(CHANNEL(language)=)") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-user-callerid:10] GotoIf("SIP/601-000001c6", "1?continue") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Goto (macro-user-callerid,s,19)
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-user-callerid:19] NoOp("SIP/601-000001c6", "Using CallerID "Shannon Office 601" <601>") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [7794184@from-internal:2] Set("SIP/601-000001c6", "EMERGENCYROUTE=YES") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [7794184@from-internal:3] Set("SIP/601-000001c6", "_NODEST=") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [7794184@from-internal:4] Macro("SIP/601-000001c6", "record-enable,601,OUT,") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/601-000001c6", "1?check") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Goto (macro-record-enable,s,4)
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-record-enable:4] ExecIf("SIP/601-000001c6", "0?MacroExit()") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-record-enable:5] GotoIf("SIP/601-000001c6", "0?Group:OUT") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Goto (macro-record-enable,s,15)
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-record-enable:15] GotoIf("SIP/601-000001c6", "0?IN") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-record-enable:16] ExecIf("SIP/601-000001c6", "1?MacroExit()") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [7794184@from-internal:5] Macro("SIP/601-000001c6", "dialout-trunk,2,7794184,,") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:1] Set("SIP/601-000001c6", "DIAL_TRUNK=2") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/601-000001c6", "0?sub-pincheck,s,1") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/601-000001c6", "0?disabletrunk,1") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:4] Set("SIP/601-000001c6", "DIAL_NUMBER=7794184") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:5] Set("SIP/601-000001c6", "DIAL_TRUNK_OPTIONS=tr") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:6] Set("SIP/601-000001c6", "OUTBOUND_GROUP=OUT_2") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/601-000001c6", "0?nomax") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/601-000001c6", "0?chanfull") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/601-000001c6", "0?skipoutcid") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:10] Set("SIP/601-000001c6", "DIAL_TRUNK_OPTIONS=") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:11] Macro("SIP/601-000001c6", "outbound-callerid,2") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/601-000001c6", "0?Set(CALLERPRES()=)") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/601-000001c6", "0?Set(REALCALLERIDNUM=601)") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/601-000001c6", "1?normcid") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Goto (macro-outbound-callerid,s,6)
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:6] Set("SIP/601-000001c6", "USEROUTCID=") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:7] Set("SIP/601-000001c6", "EMERGENCYCID=") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:8] Set("SIP/601-000001c6", "TRUNKOUTCID=5012468887") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/601-000001c6", "1?trunkcid") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Goto (macro-outbound-callerid,s,12)
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/601-000001c6", "1?Set(CALLERID(all)=5012468887)") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/601-000001c6", "0?Set(CALLERID(all)=)") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/601-000001c6", "0?Set(CALLERID(all)=)") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/601-000001c6", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/601-000001c6", "0?AGI(fixlocalprefix)") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:13] Set("SIP/601-000001c6", "OUTNUM=7794184") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:14] Set("SIP/601-000001c6", "custom=SIP/VelocityTrunk") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/601-000001c6", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:16] Macro("SIP/601-000001c6", "dialout-trunk-predial-hook,") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/601-000001c6", "") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/601-000001c6", "0?bypass,1") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/601-000001c6", "0?customtrunk") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:19] Dial("SIP/601-000001c6", "SIP/VelocityTrunk/7794184,300,") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] netsock.c: == Using SIP RTP TOS bits 184
- [Dec 7 17:13:42] VERBOSE[9156] netsock.c: == Using SIP RTP CoS mark 5
- [Dec 7 17:13:42] VERBOSE[9156] chan_sip.c: Really destroying SIP dialog '436b21e54794f121381d5c962829c452@127.0.0.1' Method: INVITE
- [Dec 7 17:13:42] WARNING[9156] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
- [Dec 7 17:13:42] VERBOSE[9156] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/601-000001c6", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:21] Goto("SIP/601-000001c6", "s-CHANUNAVAIL,1") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/601-000001c6", "RC=20") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/601-000001c6", "20,1") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Goto (macro-dialout-trunk,20,1)
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [20@macro-dialout-trunk:1] Goto("SIP/601-000001c6", "continue,1") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Goto (macro-dialout-trunk,continue,1)
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/601-000001c6", "1?noreport") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Goto (macro-dialout-trunk,continue,3)
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/601-000001c6", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 20 - failing through to other trunks") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [continue@macro-dialout-trunk:4] Set("SIP/601-000001c6", "CALLERID(number)=601") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [7794184@from-internal:6] Macro("SIP/601-000001c6", "dialout-trunk,3,7794184,,") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:1] Set("SIP/601-000001c6", "DIAL_TRUNK=3") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/601-000001c6", "0?sub-pincheck,s,1") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/601-000001c6", "0?disabletrunk,1") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:4] Set("SIP/601-000001c6", "DIAL_NUMBER=7794184") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:5] Set("SIP/601-000001c6", "DIAL_TRUNK_OPTIONS=tr") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:6] Set("SIP/601-000001c6", "OUTBOUND_GROUP=OUT_3") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/601-000001c6", "1?nomax") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Goto (macro-dialout-trunk,s,9)
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/601-000001c6", "0?skipoutcid") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:10] Set("SIP/601-000001c6", "DIAL_TRUNK_OPTIONS=") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:11] Macro("SIP/601-000001c6", "outbound-callerid,3") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/601-000001c6", "0?Set(CALLERPRES()=)") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/601-000001c6", "0?Set(REALCALLERIDNUM=601)") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/601-000001c6", "1?normcid") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Goto (macro-outbound-callerid,s,6)
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:6] Set("SIP/601-000001c6", "USEROUTCID=") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:7] Set("SIP/601-000001c6", "EMERGENCYCID=") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:8] Set("SIP/601-000001c6", "TRUNKOUTCID=NorthSide Aquatics<5018033434>") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/601-000001c6", "1?trunkcid") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Goto (macro-outbound-callerid,s,12)
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/601-000001c6", "1?Set(CALLERID(all)=NorthSide Aquatics<5018033434>)") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/601-000001c6", "0?Set(CALLERID(all)=)") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/601-000001c6", "0?Set(CALLERID(all)=)") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/601-000001c6", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/601-000001c6", "1?AGI(fixlocalprefix)") in new stack
- [Dec 7 17:13:42] VERBOSE[9156] res_agi.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
- [Dec 7 17:13:43] VERBOSE[2984] chan_sip.c: REGISTER 11 headers, 0 lines
- [Dec 7 17:13:43] VERBOSE[2984] chan_sip.c: Reliably Transmitting (no NAT) to 66.241.96.164:5060:
- REGISTER sip:inbound24.vitelity.net SIP/2.0
- Via: SIP/2.0/UDP 192.168.230.252:5060;branch=z9hG4bK6ec3efd9;rport
- Max-Forwards: 70
- From: <sip:mlsmith@inbound24.vitelity.net>;tag=as183ffa31
- To: <sip:mlsmith@inbound24.vitelity.net>
- Call-ID: 763abf1775ce87cd1faff787433f7218@127.0.0.1
- CSeq: 1079 REGISTER
- User-Agent: Asterisk PBX 1.6.2.11
- Authorization: Digest username="mlsmith", realm="asterisk", algorithm=MD5, uri="sip:inbound24.vitelity.net", nonce="60867419", response="dde6f189692023dc0d16184ddd75af7c"
- Expires: 120
- Contact: <sip:s@192.168.230.252>
- Content-Length: 0
- ---
- [Dec 7 17:13:43] VERBOSE[9156] res_agi.c: > fixlocalprefix: Using pattern 1+NXXNXXXXXX
- [Dec 7 17:13:43] VERBOSE[9156] res_agi.c: > fixlocalprefix: Using pattern 1501+NXXXXXX
- [Dec 7 17:13:43] VERBOSE[9156] res_agi.c: == fixlocalprefix: Dialpattern 1501+NXXXXXX matched. 7794184 -> 15017794184
- [Dec 7 17:13:43] VERBOSE[9156] res_agi.c: -- <SIP/601-000001c6>AGI Script fixlocalprefix completed, returning 0
- [Dec 7 17:13:43] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:13] Set("SIP/601-000001c6", "OUTNUM=15017794184") in new stack
- [Dec 7 17:13:43] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:14] Set("SIP/601-000001c6", "custom=SIP/vitel-outbound") in new stack
- [Dec 7 17:13:43] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/601-000001c6", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack
- [Dec 7 17:13:43] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:16] Macro("SIP/601-000001c6", "dialout-trunk-predial-hook,") in new stack
- [Dec 7 17:13:43] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/601-000001c6", "") in new stack
- [Dec 7 17:13:43] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/601-000001c6", "0?bypass,1") in new stack
- [Dec 7 17:13:43] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/601-000001c6", "0?customtrunk") in new stack
- [Dec 7 17:13:43] VERBOSE[9156] pbx.c: -- Executing [s@macro-dialout-trunk:19] Dial("SIP/601-000001c6", "SIP/vitel-outbound/15017794184,300,") in new stack
- [Dec 7 17:13:43] VERBOSE[9156] netsock.c: == Using SIP RTP TOS bits 184
- [Dec 7 17:13:43] VERBOSE[9156] netsock.c: == Using SIP RTP CoS mark 5
- [Dec 7 17:13:43] VERBOSE[9156] chan_sip.c: Audio is at 192.168.230.252 port 12114
- [Dec 7 17:13:43] VERBOSE[9156] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
- [Dec 7 17:13:43] VERBOSE[9156] chan_sip.c: Adding codec 0x8 (alaw) to SDP
- [Dec 7 17:13:43] VERBOSE[9156] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
- [Dec 7 17:13:43] VERBOSE[9156] chan_sip.c: Reliably Transmitting (no NAT) to 64.2.142.216:5060:
- INVITE sip:15017794184@outbound.vitelity.net SIP/2.0
- Via: SIP/2.0/UDP 192.168.230.252:5060;branch=z9hG4bK1bc67f57;rport
- Max-Forwards: 70
- From: "NorthSide Aquatics" <sip:mlsmith@192.168.230.252>;tag=as3db90e44
- To: <sip:15017794184@outbound.vitelity.net>
- Contact: <sip:mlsmith@192.168.230.252>
- Call-ID: 5036b9a02aed3dea333adc1940e24adc@192.168.230.252
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 1.6.2.11
- Remote-Party-ID: "NorthSide Aquatics" <sip:5018033434@192.168.230.252>;privacy=off;screen=no
- Date: Tue, 07 Dec 2010 23:13:43 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 267
- v=0
- o=root 1965089195 1965089195 IN IP4 192.168.230.252
- s=Asterisk PBX 1.6.2.11
- c=IN IP4 192.168.230.252
- t=0 0
- m=audio 12114 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
- ---
- [Dec 7 17:13:43] VERBOSE[9156] app_dial.c: -- Called vitel-outbound/15017794184
- [Dec 7 17:13:43] VERBOSE[2984] chan_sip.c:
- <--- SIP read from UDP:66.241.96.164:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 192.168.230.252:5060;branch=z9hG4bK6ec3efd9;received=192.168.230.252;rport=5060
- From: <sip:mlsmith@inbound24.vitelity.net>;tag=as183ffa31
- To: <sip:mlsmith@inbound24.vitelity.net>
- Call-ID: 763abf1775ce87cd1faff787433f7218@127.0.0.1
- CSeq: 1079 REGISTER
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- <------------->
- [Dec 7 17:13:43] VERBOSE[2984] chan_sip.c: --- (10 headers 0 lines) ---
- [Dec 7 17:13:43] VERBOSE[2984] chan_sip.c:
- <--- SIP read from UDP:66.241.96.164:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 192.168.230.252:5060;branch=z9hG4bK6ec3efd9;received=192.168.230.252;rport=5060
- From: <sip:mlsmith@inbound24.vitelity.net>;tag=as183ffa31
- To: <sip:mlsmith@inbound24.vitelity.net>;tag=as453ab499
- Call-ID: 763abf1775ce87cd1faff787433f7218@127.0.0.1
- CSeq: 1079 REGISTER
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Expires: 60
- Contact: <sip:s@192.168.230.252:5060>;expires=60
- Date: Tue, 07 Dec 2010 23:15:03 GMT
- Content-Length: 0
- <------------->
- [Dec 7 17:13:43] VERBOSE[2984] chan_sip.c: --- (13 headers 0 lines) ---
- [Dec 7 17:13:43] VERBOSE[2984] chan_sip.c: Scheduling destruction of SIP dialog '763abf1775ce87cd1faff787433f7218@127.0.0.1' in 32000 ms (Method: REGISTER)
- [Dec 7 17:13:43] NOTICE[2984] chan_sip.c: Outbound Registration: Expiry for inbound24.vitelity.net is 60 sec (Scheduling reregistration in 45 s)
- [Dec 7 17:13:43] VERBOSE[2984] chan_sip.c:
- <--- SIP read from UDP:64.2.142.216:5060 --->
- SIP/2.0 407 Proxy Authentication Required
- Via: SIP/2.0/UDP 192.168.230.252:5060;branch=z9hG4bK1bc67f57;received=192.168.230.252;rport=5060
- From: "NorthSide Aquatics" <sip:mlsmith@192.168.230.252:5060>;tag=as3db90e44
- To: <sip:15017794184@outbound.vitelity.net>;tag=as6224c08a
- Call-ID: 5036b9a02aed3dea333adc1940e24adc@192.168.230.252
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3bd5c9bf"
- Content-Length: 0
- <------------->
- [Dec 7 17:13:43] VERBOSE[2984] chan_sip.c: --- (11 headers 0 lines) ---
- [Dec 7 17:13:43] VERBOSE[2984] chan_sip.c: Transmitting (no NAT) to 64.2.142.216:5060:
- ACK sip:15017794184@outbound.vitelity.net SIP/2.0
- Via: SIP/2.0/UDP 192.168.230.252:5060;branch=z9hG4bK1bc67f57;rport
- Max-Forwards: 70
- From: "NorthSide Aquatics" <sip:mlsmith@192.168.230.252>;tag=as3db90e44
- To: <sip:15017794184@outbound.vitelity.net>;tag=as6224c08a
- Contact: <sip:mlsmith@192.168.230.252>
- Call-ID: 5036b9a02aed3dea333adc1940e24adc@192.168.230.252
- CSeq: 102 ACK
- User-Agent: Asterisk PBX 1.6.2.11
- Remote-Party-ID: "NorthSide Aquatics" <sip:5018033434@192.168.230.252>;privacy=off;screen=no
- Content-Length: 0
- ---
- [Dec 7 17:13:43] VERBOSE[2984] chan_sip.c: Audio is at 192.168.230.252 port 12114
- [Dec 7 17:13:43] VERBOSE[2984] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
- [Dec 7 17:13:43] VERBOSE[2984] chan_sip.c: Adding codec 0x8 (alaw) to SDP
- [Dec 7 17:13:43] VERBOSE[2984] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
- [Dec 7 17:13:43] VERBOSE[2984] chan_sip.c: Reliably Transmitting (no NAT) to 64.2.142.216:5060:
- INVITE sip:15017794184@outbound.vitelity.net SIP/2.0
- Via: SIP/2.0/UDP 192.168.230.252:5060;branch=z9hG4bK3bce8626;rport
- Max-Forwards: 70
- From: "NorthSide Aquatics" <sip:mlsmith@192.168.230.252>;tag=as3db90e44
- To: <sip:15017794184@outbound.vitelity.net>
- Contact: <sip:mlsmith@192.168.230.252>
- Call-ID: 5036b9a02aed3dea333adc1940e24adc@192.168.230.252
- CSeq: 103 INVITE
- User-Agent: Asterisk PBX 1.6.2.11
- Remote-Party-ID: "NorthSide Aquatics" <sip:5018033434@192.168.230.252>;privacy=off;screen=no
- Proxy-Authorization: Digest username="mlsmith", realm="asterisk", algorithm=MD5, uri="sip:15017794184@outbound.vitelity.net", nonce="3bd5c9bf", response="d37d658da3c41a5ec69e778040108ebf"
- Date: Tue, 07 Dec 2010 23:13:43 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 267
- v=0
- o=root 1965089195 1965089196 IN IP4 192.168.230.252
- s=Asterisk PBX 1.6.2.11
- c=IN IP4 192.168.230.252
- t=0 0
- m=audio 12114 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
- ---
- [Dec 7 17:13:43] VERBOSE[2984] chan_sip.c:
- <--- SIP read from UDP:64.2.142.216:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 192.168.230.252:5060;branch=z9hG4bK3bce8626;received=192.168.230.252;rport=5060
- From: "NorthSide Aquatics" <sip:mlsmith@192.168.230.252:5060>;tag=as3db90e44
- To: <sip:15017794184@outbound.vitelity.net>
- Call-ID: 5036b9a02aed3dea333adc1940e24adc@192.168.230.252
- CSeq: 103 INVITE
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Contact: <sip:15017794184@64.2.142.216>
- Content-Length: 0
- <------------->
- [Dec 7 17:13:43] VERBOSE[2984] chan_sip.c: --- (11 headers 0 lines) ---
- [Dec 7 17:13:44] VERBOSE[2984] chan_sip.c:
- <--- SIP read from UDP:64.2.142.216:5060 --->
- SIP/2.0 183 Session Progress
- Via: SIP/2.0/UDP 192.168.230.252:5060;branch=z9hG4bK3bce8626;received=192.168.230.252;rport=5060
- From: "NorthSide Aquatics" <sip:mlsmith@192.168.230.252:5060>;tag=as3db90e44
- To: <sip:15017794184@outbound.vitelity.net>;tag=as586c36b8
- Call-ID: 5036b9a02aed3dea333adc1940e24adc@192.168.230.252
- CSeq: 103 INVITE
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Contact: <sip:15017794184@64.2.142.216>
- Content-Type: application/sdp
- Content-Length: 264
- v=0
- o=root 13387 13387 IN IP4 64.2.142.216
- s=session
- c=IN IP4 64.2.142.216
- t=0 0
- m=audio 16192 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=silenceSupp:off - - - -
- a=ptime:20
- a=sendrecv
- <------------->
- [Dec 7 17:13:44] VERBOSE[2984] chan_sip.c: --- (12 headers 13 lines) ---
- [Dec 7 17:13:44] VERBOSE[2984] chan_sip.c: Found RTP audio format 0
- [Dec 7 17:13:44] VERBOSE[2984] chan_sip.c: Found RTP audio format 8
- [Dec 7 17:13:44] VERBOSE[2984] chan_sip.c: Found RTP audio format 101
- [Dec 7 17:13:44] VERBOSE[2984] chan_sip.c: Found audio description format PCMU for ID 0
- [Dec 7 17:13:44] VERBOSE[2984] chan_sip.c: Found audio description format PCMA for ID 8
- [Dec 7 17:13:44] VERBOSE[2984] chan_sip.c: Found audio description format telephone-event for ID 101
- [Dec 7 17:13:44] VERBOSE[2984] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
- [Dec 7 17:13:44] VERBOSE[2984] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
- [Dec 7 17:13:44] VERBOSE[2984] chan_sip.c: Peer audio RTP is at port 64.2.142.216:16192
- [Dec 7 17:13:44] VERBOSE[9156] app_dial.c: -- SIP/vitel-outbound-000001c7 is making progress passing it to SIP/601-000001c6
- [Dec 7 17:13:44] VERBOSE[9156] chan_sip.c: Audio is at 192.168.230.252 port 19368
- [Dec 7 17:13:44] VERBOSE[9156] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
- [Dec 7 17:13:44] VERBOSE[9156] chan_sip.c: Adding codec 0x8 (alaw) to SDP
- [Dec 7 17:13:44] VERBOSE[9156] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
- [Dec 7 17:13:44] VERBOSE[9156] chan_sip.c:
- <--- Transmitting (NAT) to 192.168.230.250:5060 --->
- SIP/2.0 183 Session Progress
- Via: SIP/2.0/UDP 192.168.230.250:5060;branch=z9hG4bK-23822c49;received=192.168.230.250
- From: "Office Shannon" <sip:601@192.168.230.252>;tag=ad987cf4205f8be9o0
- To: <sip:7794184@192.168.230.252>;tag=as25b40b68
- Call-ID: 65c63e10-42695d7d@192.168.230.250
- CSeq: 102 INVITE
- Server: Asterisk PBX 1.6.2.11
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Contact: <sip:7794184@192.168.230.252>
- Content-Type: application/sdp
- Content-Length: 265
- v=0
- o=root 768564408 768564408 IN IP4 192.168.230.252
- s=Asterisk PBX 1.6.2.11
- c=IN IP4 192.168.230.252
- t=0 0
- m=audio 19368 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
- <------------>
- [Dec 7 17:13:46] VERBOSE[2984] chan_sip.c:
- <--- SIP read from UDP:192.168.230.250:5060 --->
- CANCEL sip:7794184@192.168.230.252 SIP/2.0
- Via: SIP/2.0/UDP 192.168.230.250:5060;branch=z9hG4bK-23822c49
- From: "Office Shannon" <sip:601@192.168.230.252>;tag=ad987cf4205f8be9o0
- To: <sip:7794184@192.168.230.252>
- Call-ID: 65c63e10-42695d7d@192.168.230.250
- CSeq: 102 CANCEL
- Max-Forwards: 70
- Authorization: Digest username="601",realm="asterisk",nonce="27563de7",uri="sip:7794184@192.168.230.252",algorithm=MD5,response="fcb4b828c4ea557f9d993b8c1152ddf3"
- User-Agent: Cisco/SPA504G-7.4.3a
- Content-Length: 0
- <------------->
- [Dec 7 17:13:46] VERBOSE[2984] chan_sip.c: --- (10 headers 0 lines) ---
- [Dec 7 17:13:46] VERBOSE[2984] chan_sip.c: Sending to 192.168.230.250 : 5060 (NAT)
- [Dec 7 17:13:46] VERBOSE[2984] chan_sip.c:
- <--- Reliably Transmitting (NAT) to 192.168.230.250:5060 --->
- SIP/2.0 487 Request Terminated
- Via: SIP/2.0/UDP 192.168.230.250:5060;branch=z9hG4bK-23822c49;received=192.168.230.250
- From: "Office Shannon" <sip:601@192.168.230.252>;tag=ad987cf4205f8be9o0
- To: <sip:7794184@192.168.230.252>;tag=as25b40b68
- Call-ID: 65c63e10-42695d7d@192.168.230.250
- CSeq: 102 INVITE
- Server: Asterisk PBX 1.6.2.11
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- [Dec 7 17:13:46] VERBOSE[2984] chan_sip.c:
- <--- Transmitting (NAT) to 192.168.230.250:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 192.168.230.250:5060;branch=z9hG4bK-23822c49;received=192.168.230.250
- From: "Office Shannon" <sip:601@192.168.230.252>;tag=ad987cf4205f8be9o0
- To: <sip:7794184@192.168.230.252>;tag=as25b40b68
- Call-ID: 65c63e10-42695d7d@192.168.230.250
- CSeq: 102 CANCEL
- Server: Asterisk PBX 1.6.2.11
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- [Dec 7 17:13:46] VERBOSE[9156] chan_sip.c: Scheduling destruction of SIP dialog '5036b9a02aed3dea333adc1940e24adc@192.168.230.252' in 32000 ms (Method: INVITE)
- [Dec 7 17:13:46] VERBOSE[9156] chan_sip.c: Reliably Transmitting (no NAT) to 64.2.142.216:5060:
- CANCEL sip:15017794184@outbound.vitelity.net SIP/2.0
- Via: SIP/2.0/UDP 192.168.230.252:5060;branch=z9hG4bK3bce8626;rport
- Max-Forwards: 70
- From: "NorthSide Aquatics" <sip:mlsmith@192.168.230.252>;tag=as3db90e44
- To: <sip:15017794184@outbound.vitelity.net>
- Call-ID: 5036b9a02aed3dea333adc1940e24adc@192.168.230.252
- CSeq: 103 CANCEL
- User-Agent: Asterisk PBX 1.6.2.11
- Remote-Party-ID: "NorthSide Aquatics" <sip:5018033434@192.168.230.252>;privacy=off;screen=no
- Content-Length: 0
- ---
- [Dec 7 17:13:46] VERBOSE[9156] chan_sip.c: Scheduling destruction of SIP dialog '5036b9a02aed3dea333adc1940e24adc@192.168.230.252' in 32000 ms (Method: INVITE)
- [Dec 7 17:13:46] VERBOSE[9156] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/601-000001c6' in macro 'dialout-trunk'
- [Dec 7 17:13:46] VERBOSE[9156] pbx.c: == Spawn extension (from-internal, 7794184, 6) exited non-zero on 'SIP/601-000001c6'
- [Dec 7 17:13:46] VERBOSE[9156] pbx.c: -- Executing [h@from-internal:1] Macro("SIP/601-000001c6", "hangupcall") in new stack
- [Dec 7 17:13:46] VERBOSE[9156] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/601-000001c6", "1?skiprg") in new stack
- [Dec 7 17:13:46] VERBOSE[9156] pbx.c: -- Goto (macro-hangupcall,s,4)
- [Dec 7 17:13:46] VERBOSE[9156] pbx.c: -- Executing [s@macro-hangupcall:4] GotoIf("SIP/601-000001c6", "1?skipblkvm") in new stack
- [Dec 7 17:13:46] VERBOSE[9156] pbx.c: -- Goto (macro-hangupcall,s,7)
- [Dec 7 17:13:46] VERBOSE[9156] pbx.c: -- Executing [s@macro-hangupcall:7] GotoIf("SIP/601-000001c6", "1?theend") in new stack
- [Dec 7 17:13:46] VERBOSE[9156] pbx.c: -- Goto (macro-hangupcall,s,9)
- [Dec 7 17:13:46] VERBOSE[9156] pbx.c: -- Executing [s@macro-hangupcall:9] Hangup("SIP/601-000001c6", "") in new stack
- [Dec 7 17:13:46] VERBOSE[9156] app_macro.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/601-000001c6' in macro 'hangupcall'
- [Dec 7 17:13:46] VERBOSE[9156] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/601-000001c6'
- [Dec 7 17:13:46] VERBOSE[2984] chan_sip.c:
- <--- SIP read from UDP:192.168.230.250:5060 --->
- ACK sip:7794184@192.168.230.252 SIP/2.0
- Via: SIP/2.0/UDP 192.168.230.250:5060;branch=z9hG4bK-23822c49
- From: "Office Shannon" <sip:601@192.168.230.252>;tag=ad987cf4205f8be9o0
- To: <sip:7794184@192.168.230.252>;tag=as25b40b68
- Call-ID: 65c63e10-42695d7d@192.168.230.250
- CSeq: 102 ACK
- Max-Forwards: 70
- Authorization: Digest username="601",realm="asterisk",nonce="27563de7",uri="sip:7794184@192.168.230.252",algorithm=MD5,response="fc48f65a3908664c8c04567c071484fa"
- Contact: "Office Shannon" <sip:601@192.168.230.250:5060>
- User-Agent: Cisco/SPA504G-7.4.3a
- Content-Length: 0
- <------------->
- [Dec 7 17:13:46] VERBOSE[2984] chan_sip.c: --- (11 headers 0 lines) ---
- [Dec 7 17:13:46] VERBOSE[2984] chan_sip.c: Really destroying SIP dialog '65c63e10-42695d7d@192.168.230.250' Method: ACK
- [Dec 7 17:13:46] VERBOSE[2984] chan_sip.c:
- <--- SIP read from UDP:64.2.142.216:5060 --->
- SIP/2.0 487 Request Terminated
- Via: SIP/2.0/UDP 192.168.230.252:5060;branch=z9hG4bK3bce8626;received=192.168.230.252;rport=5060
- From: "NorthSide Aquatics" <sip:mlsmith@192.168.230.252:5060>;tag=as3db90e44
- To: <sip:15017794184@outbound.vitelity.net>;tag=as586c36b8
- Call-ID: 5036b9a02aed3dea333adc1940e24adc@192.168.230.252
- CSeq: 103 INVITE
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- <------------->
- [Dec 7 17:13:46] VERBOSE[2984] chan_sip.c: --- (10 headers 0 lines) ---
- [Dec 7 17:13:46] VERBOSE[2984] chan_sip.c: Transmitting (no NAT) to 64.2.142.216:5060:
- ACK sip:15017794184@outbound.vitelity.net SIP/2.0
- Via: SIP/2.0/UDP 192.168.230.252:5060;branch=z9hG4bK3bce8626;rport
- Max-Forwards: 70
- From: "NorthSide Aquatics" <sip:mlsmith@192.168.230.252>;tag=as3db90e44
- To: <sip:15017794184@outbound.vitelity.net>;tag=as586c36b8
- Contact: <sip:mlsmith@192.168.230.252>
- Call-ID: 5036b9a02aed3dea333adc1940e24adc@192.168.230.252
- CSeq: 103 ACK
- User-Agent: Asterisk PBX 1.6.2.11
- Remote-Party-ID: "NorthSide Aquatics" <sip:5018033434@192.168.230.252>;privacy=off;screen=no
- Content-Length: 0
- ---
- [Dec 7 17:13:46] VERBOSE[2984] chan_sip.c:
- <--- SIP read from UDP:64.2.142.216:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 192.168.230.252:5060;branch=z9hG4bK3bce8626;received=192.168.230.252;rport=5060
- From: "NorthSide Aquatics" <sip:mlsmith@192.168.230.252:5060>;tag=as3db90e44
- To: <sip:15017794184@outbound.vitelity.net>;tag=as586c36b8
- Call-ID: 5036b9a02aed3dea333adc1940e24adc@192.168.230.252
- CSeq: 103 CANCEL
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- <------------->
- [Dec 7 17:13:46] VERBOSE[2984] chan_sip.c: --- (10 headers 0 lines) ---
- [Dec 7 17:13:46] VERBOSE[2984] chan_sip.c: Really destroying SIP dialog '5036b9a02aed3dea333adc1940e24adc@192.168.230.252' Method: INVITE
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