=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.10.09 10:55:04 =~=~=~=~=~=~=~=~=~=~=~= <--- SIP read from UDP://10.0.11.1:42486 ---> REGISTER sip:192.168.51.15:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.19.7:5060;branch=z9hG4bK09531c926Max-Forwards: 70Content-Length: 0To: citywok SVR1 From: citywok SVR1 ;tag=2cf485394725105Call-ID: b6dfed9d0f016f5162a575db8a22765a@192.168.19.7CSeq: 462345470 REGISTERContact: citywok SVR1 ;expires=30Allow-Events: talk,hold,conferenceAllow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFOAuthorization:Digest response="ca1182855edb81e33780a91c61a8fc2d",username="1594",realm="asterisk",nonce="3943149a",algorithm=MD5,uri="sip:192.168.51.15:5060"User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (13 headers 0 lines) --- csgtacsip1*CLI> Sending to 192.168.19.7 : 5060 (no NAT) <--- Transmitting (NAT) to 10.0.11.1:42486 ---> SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 192.168.19.7:5060;branch=z9hG4bK09531c926;received=10.0.11.1From: citywok SVR1 ;tag=2cf485394725105To: citywok SVR1 ;tag=as1301ab84Call-ID: b6dfed9d0f016f5162a575db8a22765a@192.168.19.7CSeq: 462345470 REGISTERUser-Agent: Asterisk PBX 1.6.0.14Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerWWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="21166849"Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'b6dfed9d0f016f5162a575db8a22765a@192.168.19.7' in 32000 ms (Method: REGISTER) csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:42486 ---> REGISTER sip:192.168.51.15:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.19.7:5060;branch=z9hG4bK7fc996fe1Max-Forwards: 70Content-Length: 0To: citywok SVR1 From: citywok SVR1 ;tag=2cf485394725105Call-ID: b6dfed9d0f016f5162a575db8a22765a@192.168.19.7CSeq: 462345471 REGISTERContact: citywok SVR1 ;expires=30Allow-Events: talk,hold,conferenceAllow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFOAuthorization:Digest response="e7fb6c8906734a03eefe81af2eabc360",username="1594",realm="asterisk",nonce="21166849",algorithm=MD5,uri="sip:192.168.51.15:5060"User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (13 headers 0 lines) --- csgtacsip1*CLI> Sending to 10.0.11.1 : 42486 (NAT) csgtacsip1*CLI> Reliably Transmitting (NAT) to 10.0.11.1:42486: OPTIONS sip:1594@192.168.19.7:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 192.168.51.15:5060;branch=z9hG4bK457015b2;rportMax-Forwards: 70From: "asterisk" ;tag=as45e7870fTo: Contact: Call-ID: 55e680e600b425697f2dc47f335760aa@192.168.51.15CSeq: 102 OPTIONSUser-Agent: Asterisk PBX 1.6.0.14Date: Fri, 09 Oct 2009 17:52:52 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContent-Length: 0 --- <--- Transmitting (NAT) to 10.0.11.1:42486 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.19.7:5060;branch=z9hG4bK7fc996fe1;received=10.0.11.1From: citywok SVR1 ;tag=2cf485394725105To: citywok SVR1 ;tag=as1301ab84Call-ID: b6dfed9d0f016f5162a575db8a22765a@192.168.19.7CSeq: 462345471 REGISTERUser-Agent: Asterisk PBX 1.6.0.14Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerExpires: 60Contact: ;expires=60Date: Fri, 09 Oct 2009 17:52:52 GMTContent-Length: 0 <------------> Scheduling destruction of SIP dialog 'b6dfed9d0f016f5162a575db8a22765a@192.168.19.7' in 32000 ms (Method: REGISTER) csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:42486 ---> SIP/2.0 200 OKCall-ID: 55e680e600b425697f2dc47f335760aa@192.168.51.15CSeq: 102 OPTIONSFrom: "asterisk" ;tag=as45e7870fTo: ;tag=4aeb0c0030943a7Via: SIP/2.0/UDP 192.168.51.15:5060;branch=z9hG4bK457015b2;rportContent-Length: 0Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFOContact: Supported: replacesUser-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '55e680e600b425697f2dc47f335760aa@192.168.51.15' Method: OPTIONS csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:1218 ---> REGISTER sip:192.168.51.15:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.16.224:5060;branch=z9hG4bKe2f3f471fd99a3a9c.114e703105f7b55c2Max-Forwards: 70From: ;tag=83a78f3cdfTo: Call-ID: 04740a8a939ef6b6CSeq: 9601 REGISTERAllow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFOAllow-Events: talk, hold, conference, LocalModeStatusAuthorization: Digest username="1512",realm="asterisk",nonce="229b63d6",uri="sip:192.168.51.15:5060",response="5d8506be1a0b2ed56e7194012f54e5ce",algorithm=MD5Contact: "Operator" ;expires=60;+sip.instance=""Supported: gruu, pathUser-Agent: Aastra 57iCT/2.5.2.1010Content-Length: 0 <-------------> --- (14 headers 0 lines) --- csgtacsip1*CLI> Sending to 192.168.16.224 : 5060 (no NAT) <--- Transmitting (NAT) to 10.0.11.1:1218 ---> SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 192.168.16.224:5060;branch=z9hG4bKe2f3f471fd99a3a9c.114e703105f7b55c2;received=10.0.11.1From: ;tag=83a78f3cdfTo: ;tag=as4c57deb4Call-ID: 04740a8a939ef6b6CSeq: 9601 REGISTERUser-Agent: Asterisk PBX 1.6.0.14Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerWWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2d2036ed"Content-Length: 0 <------------> Scheduling destruction of SIP dialog '04740a8a939ef6b6' in 32000 ms (Method: REGISTER) csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:1218 ---> REGISTER sip:192.168.51.15:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.16.224:5060;branch=z9hG4bK779fc8b8f9683dd20.870119e636449a082Max-Forwards: 70From: ;tag=83a78f3cdfTo: Call-ID: 04740a8a939ef6b6CSeq: 9602 REGISTERAllow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFOAllow-Events: talk, hold, conference, LocalModeStatusAuthorization: Digest username="1512",realm="asterisk",nonce="2d2036ed",uri="sip:192.168.51.15:5060",response="58e47bbdc221ede55106a004d43826c8",algorithm=MD5Contact: "Operator" ;expires=60;+sip.instance=""Supported: gruu, pathUser-Agent: Aastra 57iCT/2.5.2.1010Content-Length: 0 <-------------> --- (14 headers 0 lines) --- csgtacsip1*CLI> Sending to 10.0.11.1 : 1218 (NAT) csgtacsip1*CLI> Reliably Transmitting (NAT) to 10.0.11.1:1218: OPTIONS sip:1512@192.168.16.224:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 192.168.51.15:5060;branch=z9hG4bK68d221fb;rportMax-Forwards: 70From: "asterisk" ;tag=as15a7b21bTo: Contact: Call-ID: 650000191fa8a52471598447596f8b7d@192.168.51.15CSeq: 102 OPTIONSUser-Agent: Asterisk PBX 1.6.0.14Date: Fri, 09 Oct 2009 17:52:58 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContent-Length: 0 --- <--- Transmitting (NAT) to 10.0.11.1:1218 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.16.224:5060;branch=z9hG4bK779fc8b8f9683dd20.870119e636449a082;received=10.0.11.1From: ;tag=83a78f3cdfTo: ;tag=as4c57deb4Call-ID: 04740a8a939ef6b6CSeq: 9602 REGISTERUser-Agent: Asterisk PBX 1.6.0.14Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerExpires: 60Contact: ;expires=60Date: Fri, 09 Oct 2009 17:52:58 GMTContent-Length: 0 <------------> csgtacsip1*CLI> Scheduling destruction of SIP dialog '04740a8a939ef6b6' in 32000 ms (Method: REGISTER) csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:1218 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.51.15:5060;branch=z9hG4bK68d221fb;rport=5060;received=192.168.51.15From: "asterisk" ;tag=as15a7b21bTo: ;tag=2826610898Call-ID: 650000191fa8a52471598447596f8b7d@192.168.51.15CSeq: 102 OPTIONSAllow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFOServer: Aastra 57iCT/2.5.2.1010Supported: gruu, timer, 100rel, replaces, pathContent-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '650000191fa8a52471598447596f8b7d@192.168.51.15' Method: OPTIONS csgtacsip1*CLI> Really destroying SIP dialog '2c1d14b472c9cb0e9fc8d7bb4f91e033@192.168.16.221' Method: REGISTER csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:1221 ---> INVITE sip:1593@192.168.51.15:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bK4e316abd5Max-Forwards: 70Content-Length: 564To: citywok SVR1 From: citywok SVR1 ;tag=e71029057342561Call-ID: ccc6b59b7b5ffa1fc6191b84331f52cd@192.168.16.221CSeq: 1347600149 INVITESupported: timerAllow-Events: talk,hold,conferenceAllow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFOContent-Type: application/sdpContact: citywok SVR1 Supported: replacesUser-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45v=0o=MxSIP 0 958506772 IN IP4 192.168.16.221s=SIP Callc=IN IP4 192.168.16.221t=0 0m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 97 98 2 99 8 101a=rtpmap:0 PCMU/8000a=rtpmap:18 G729/8000a=rtpmap:96 BV16/8000a=rtpmap:102 BV32/16000a=rtpmap:107 L16/16000a=rtpmap:104 PCMU/16000a=rtpmap:105 PCMA/16000a=rtpmap:106 L16/8000a=rtpmap:97 G726-16/8000a=rtpmap:98 G726-24/8000a=rtpmap:2 G726-32/8000a=rtpmap:99 G726-40/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=ptime:30a=silenceSupp:on - - - - <-------------> csgtacsip1*CLI> --- (15 headers 23 lines) --- == Using SIP RTP CoS mark 5 Sending to 192.168.16.221 : 5060 (no NAT) Using INVITE request as basis request - ccc6b59b7b5ffa1fc6191b84331f52cd@192.168.16.221 No user '1593' in SIP users list Found peer '1593' for '1593' from 10.0.11.1:1221 <--- Reliably Transmitting (NAT) to 10.0.11.1:1221 ---> SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bK4e316abd5;received=10.0.11.1From: citywok SVR1 ;tag=e71029057342561To: citywok SVR1 ;tag=as4da10fc6Call-ID: ccc6b59b7b5ffa1fc6191b84331f52cd@192.168.16.221CSeq: 1347600149 INVITEUser-Agent: Asterisk PBX 1.6.0.14Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerWWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3cfd42a6"Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'ccc6b59b7b5ffa1fc6191b84331f52cd@192.168.16.221' in 11712 ms (Method: INVITE) csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:1221 ---> ACK sip:1593@192.168.51.15:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bK4e316abd5Max-Forwards: 70Content-Length: 0To: citywok SVR1 ;tag=as4da10fc6From: citywok SVR1 ;tag=e71029057342561Call-ID: ccc6b59b7b5ffa1fc6191b84331f52cd@192.168.16.221CSeq: 1347600149 ACKUser-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (9 headers 0 lines) --- csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:1221 ---> INVITE sip:1593@192.168.51.15:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bKc8792b9b9Max-Forwards: 70Content-Length: 564To: citywok SVR1 From: citywok SVR1 ;tag=e71029057342561Call-ID: ccc6b59b7b5ffa1fc6191b84331f52cd@192.168.16.221CSeq: 1347600150 INVITESupported: timerAllow-Events: talk,hold,conferenceAllow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFOContact: citywok SVR1 Content-Type: application/sdpSupported: replacesAuthorization:Digest response="bb91b43b5ae237529eb003e6e4dd9afb",username="1593",realm="asterisk",nonce="3cfd42a6",algorithm=MD5,uri="sip:1593@192.168.51.15:5060"User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45v=0o=MxSIP 0 958506772 IN IP4 192.168.16.221s=SIP Callc=IN IP4 192.168.16.221t=0 0m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 97 98 2 99 8 101a=rtpmap:0 PCMU/8000a=rtpmap:18 G729/8000a=rtpmap:96 BV16/8000a=rtpmap:102 BV32/16000a=rtpmap:107 L16/16000a=rtpmap:104 PCMU/16000a=rtpmap:105 PCMA/16000a=rtpmap:106 L16/8000a=rtpmap:97 G726-16/8000a=rtpmap:98 G726-24/8000a=rtpmap:2 G726-32/8000a=rtpmap:99 G726-40/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=ptime:30a=silenceSupp:on - - - - <-------------> csgtacsip1*CLI> --- (16 headers 23 lines) --- Sending to 10.0.11.1 : 1221 (NAT) Using INVITE request as basis request - ccc6b59b7b5ffa1fc6191b84331f52cd@192.168.16.221 No user '1593' in SIP users list Found peer '1593' for '1593' from 10.0.11.1:1221 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 102 Found RTP audio format 107 Found RTP audio format 104 Found RTP audio format 105 Found RTP audio format 106 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 2 Found RTP audio format 99 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.16.221:3000 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found unknown media description format BV16 for ID 96 Found unknown media description format BV32 for ID 102 Found audio description format L16 for ID 107 Found audio description format PCMU for ID 104 Found audio description format PCMA for ID 105 Found audio description format L16 for ID 106 Found unknown media description format G726-16 for ID 97 Found unknown media description format G726-24 for ID 98 Found audio description format G726-32 for ID 2 Found unknown media description format G726-40 for ID 99 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0x94c (ulaw|alaw|g726|slin|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.16.221:3000 Looking for 1593 in from-staff (domain 192.168.51.15) list_route: hop: <--- Transmitting (NAT) to 10.0.11.1:1221 ---> SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bKc8792b9b9;received=10.0.11.1From: citywok SVR1 ;tag=e71029057342561To: citywok SVR1 Call-ID: ccc6b59b7b5ffa1fc6191b84331f52cd@192.168.16.221CSeq: 1347600150 INVITEUser-Agent: Asterisk PBX 1.6.0.14Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerRequire: timerSession-Expires: -1;refresher=uasContact: Content-Length: 0 <------------> -- Executing [1593@from-staff:1] Dial("SIP/1593-f795bcc0", "SIP/OCS_TRUNK/+2593") in new stack set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.16.224, port 5060 == Using SIP RTP CoS mark 5 Reliably Transmitting (NAT) to 10.0.11.1:1218: NOTIFY sip:1512@192.168.16.224:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 192.168.51.15:5060;branch=z9hG4bK51aa9d97;rportMax-Forwards: 70From: "" ;tag=as0dd7b765To: "Operator" ;tag=e03550dbfeContact: Call-ID: 2eb15443c9dfb81bCSeq: 103 NOTIFYUser-Agent: Asterisk PBX 1.6.0.14Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 210 confirmed --- == Extension Changed 1593[subscribecontext] new state InUse for Notify User 1512 Audio is at 192.168.51.15 port 19748 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.51.16:5060: INVITE sip:+2593@192.168.51.16 SIP/2.0Via: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK4bc47361;rportMax-Forwards: 70From: "citywok" ;tag=as04f7a591To: Contact: Call-ID: 6c5985216192f0cd650770680154a481@192.168.51.15CSeq: 102 INVITEUser-Agent: Asterisk PBX 1.6.0.14Date: Fri, 09 Oct 2009 17:52:59 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContent-Type: application/sdpContent-Length: 264v=0o=root 171629208 171629208 IN IP4 192.168.51.15s=Asterisk PBX 1.6.0.14c=IN IP4 192.168.51.15t=0 0m=audio 19748 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --- -- Called OCS_TRUNK/+2593 csgtacsip1*CLI> <--- SIP read from TCP://192.168.51.16:5060 ---> SIP/2.0 100 TryingFROM: "citywok";tag=as04f7a591TO: CSEQ: 102 INVITECALL-ID: 6c5985216192f0cd650770680154a481@192.168.51.15VIA: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK4bc47361;rportCONTENT-LENGTH: 0 <-------------> --- (7 headers 0 lines) --- csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:1218 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.51.15:5060;branch=z9hG4bK51aa9d97;rport=5060;received=192.168.51.15From: "" ;tag=as0dd7b765To: "Operator" ;tag=e03550dbfeCall-ID: 2eb15443c9dfb81bCSeq: 103 NOTIFYContact: "Operator" ;+sip.instance=""Server: Aastra 57iCT/2.5.2.1010Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived csgtacsip1*CLI> <--- SIP read from TCP://192.168.51.16:5060 ---> SIP/2.0 183 Session ProgressFROM: "citywok";tag=as04f7a591TO: ;epid=FCE3547589;tag=c4d9905f5CSEQ: 102 INVITECALL-ID: 6c5985216192f0cd650770680154a481@192.168.51.15VIA: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK4bc47361;rportCONTENT-LENGTH: 0SERVER: RTCC/3.0.0.0 MediationServer <-------------> --- (8 headers 0 lines) --- csgtacsip1*CLI> <--- SIP read from TCP://192.168.51.16:5060 ---> SIP/2.0 180 RingingFROM: "citywok";tag=as04f7a591TO: ;epid=FCE3547589;tag=c4d9905f5CSEQ: 102 INVITECALL-ID: 6c5985216192f0cd650770680154a481@192.168.51.15VIA: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK4bc47361;rportCONTENT-LENGTH: 0SERVER: RTCC/3.0.0.0 MediationServer <-------------> --- (8 headers 0 lines) --- csgtacsip1*CLI> <--- SIP read from TCP://192.168.51.16:5060 ---> SIP/2.0 180 RingingFROM: "citywok";tag=as04f7a591TO: ;epid=FCE3547589;tag=c4d9905f5CSEQ: 102 INVITECALL-ID: 6c5985216192f0cd650770680154a481@192.168.51.15VIA: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK4bc47361;rportCONTENT-LENGTH: 0SERVER: RTCC/3.0.0.0 MediationServer <-------------> --- (8 headers 0 lines) --- csgtacsip1*CLI> <--- SIP read from TCP://192.168.51.16:5060 ---> SIP/2.0 180 RingingFROM: "citywok";tag=as04f7a591TO: ;epid=FCE3547589;tag=c4d9905f5CSEQ: 102 INVITECALL-ID: 6c5985216192f0cd650770680154a481@192.168.51.15VIA: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK4bc47361;rportCONTENT-LENGTH: 0SERVER: RTCC/3.0.0.0 MediationServer <-------------> --- (8 headers 0 lines) --- csgtacsip1*CLI> <--- SIP read from TCP://192.168.51.16:5060 ---> SIP/2.0 180 RingingFROM: "citywok";tag=as04f7a591TO: ;epid=FCE3547589;tag=c4d9905f5CSEQ: 102 INVITECALL-ID: 6c5985216192f0cd650770680154a481@192.168.51.15VIA: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK4bc47361;rportCONTENT-LENGTH: 0SERVER: RTCC/3.0.0.0 MediationServer <-------------> --- (8 headers 0 lines) --- csgtacsip1*CLI> <--- SIP read from TCP://192.168.51.16:5060 ---> SIP/2.0 200 OKFROM: "citywok";tag=as04f7a591TO: ;epid=FCE3547589;tag=c4d9905f5CSEQ: 102 INVITECALL-ID: 6c5985216192f0cd650770680154a481@192.168.51.15VIA: SIP/2.0/TCP 192.168.51.15:5060;branch=z9hG4bK4bc47361;rportCONTACT: CONTENT-LENGTH: 256SUPPORTED: 100relCONTENT-TYPE: application/sdp; charset=utf-8ALLOW: UPDATESERVER: RTCC/3.0.0.0 MediationServerALLOW: Ack, Cancel, Bye,Invitev=0o=- 0 0 IN IP4 192.168.51.16s=sessionc=IN IP4 192.168.51.16b=CT:1000t=0 0m=audio 63332 RTP/AVP 0 101c=IN IP4 192.168.51.16a=rtcp:63333a=label:Audioa=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20 <-------------> --- (13 headers 14 lines) --- csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:1221 ---> REGISTER sip:192.168.51.15:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bK88dce5d24Max-Forwards: 70Content-Length: 0To: citywok SVR1 From: citywok SVR1 ;tag=b8e0fce3e0d104fCall-ID: 2c1d14b472c9cb0e9fc8d7bb4f91e033@192.168.16.221CSeq: 1585909592 REGISTERContact: citywok SVR1 ;expires=30Allow-Events: talk,hold,conferenceAllow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFOAuthorization:Digest response="2e0be80b2f3b3c4c654c2375a96b2002",username="1593",realm="asterisk",nonce="6e372201",algorithm=MD5,uri="sip:192.168.51.15:5060"User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (13 headers 0 lines) --- csgtacsip1*CLI> Sending to 192.168.16.221 : 5060 (no NAT) <--- Transmitting (NAT) to 10.0.11.1:1221 ---> SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bK88dce5d24;received=10.0.11.1From: citywok SVR1 ;tag=b8e0fce3e0d104fTo: citywok SVR1 ;tag=as0093b46eCall-ID: 2c1d14b472c9cb0e9fc8d7bb4f91e033@192.168.16.221CSeq: 1585909592 REGISTERUser-Agent: Asterisk PBX 1.6.0.14Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerWWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1d69e013"Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2c1d14b472c9cb0e9fc8d7bb4f91e033@192.168.16.221' in 32000 ms (Method: REGISTER) csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:1221 ---> REGISTER sip:192.168.51.15:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bKfc5d09d11Max-Forwards: 70Content-Length: 0To: citywok SVR1 From: citywok SVR1 ;tag=b8e0fce3e0d104fCall-ID: 2c1d14b472c9cb0e9fc8d7bb4f91e033@192.168.16.221CSeq: 1585909593 REGISTERContact: citywok SVR1 ;expires=30Allow-Events: talk,hold,conferenceAllow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFOAuthorization:Digest response="f8c39e04bfcb948b3b0a3ff9e7ad7ffc",username="1593",realm="asterisk",nonce="1d69e013",algorithm=MD5,uri="sip:192.168.51.15:5060"User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (13 headers 0 lines) --- csgtacsip1*CLI> Sending to 10.0.11.1 : 1221 (NAT) csgtacsip1*CLI> Reliably Transmitting (NAT) to 10.0.11.1:1221: OPTIONS sip:1593@192.168.16.221:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 192.168.51.15:5060;branch=z9hG4bK6a635dd7;rportMax-Forwards: 70From: "asterisk" ;tag=as317cb3c1To: Contact: Call-ID: 23ffe5f67d0b1a262b7970aa7a035f85@192.168.51.15CSeq: 102 OPTIONSUser-Agent: Asterisk PBX 1.6.0.14Date: Fri, 09 Oct 2009 17:53:12 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContent-Length: 0 --- <--- Transmitting (NAT) to 10.0.11.1:1221 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bKfc5d09d11;received=10.0.11.1From: citywok SVR1 ;tag=b8e0fce3e0d104fTo: citywok SVR1 ;tag=as0093b46eCall-ID: 2c1d14b472c9cb0e9fc8d7bb4f91e033@192.168.16.221CSeq: 1585909593 REGISTERUser-Agent: Asterisk PBX 1.6.0.14Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerExpires: 60Contact: ;expires=60Date: Fri, 09 Oct 2009 17:53:12 GMTContent-Length: 0 <------------> Scheduling destruction of SIP dialog '2c1d14b472c9cb0e9fc8d7bb4f91e033@192.168.16.221' in 32000 ms (Method: REGISTER) csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:1221 ---> SIP/2.0 200 OKCall-ID: 23ffe5f67d0b1a262b7970aa7a035f85@192.168.51.15CSeq: 102 OPTIONSFrom: "asterisk" ;tag=as317cb3c1To: ;tag=18e66fc5262fb7eVia: SIP/2.0/UDP 192.168.51.15:5060;branch=z9hG4bK6a635dd7;rportContent-Length: 0Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFOContact: Supported: replacesUser-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (11 headers 0 lines) --- csgtacsip1*CLI> Really destroying SIP dialog '23ffe5f67d0b1a262b7970aa7a035f85@192.168.51.15' Method: OPTIONS csgtacsip1*CLI> [Oct 9 10:53:15] NOTICE[9003]: chan_sip.c:9705 sip_reregister: -- Re-registration for YxD93jsh61@sjc-primary.voicepulse.com csgtacsip1*CLI> REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 67.108.9.165:5060: REGISTER sip:sjc-primary.voicepulse.com SIP/2.0Via: SIP/2.0/UDP 70.35.113.86:5060;branch=z9hG4bK5e91b448;rportMax-Forwards: 70From: ;tag=as5e63cb98To: Call-ID: 442cc8df3818c91a3edb029f59f9c0e2@127.0.1.1CSeq: 109 REGISTERUser-Agent: Asterisk PBX 1.6.0.14Authorization: Digest username="YxD93jsh61", realm="sjc-primary.voicepulse.com", algorithm=MD5, uri="sip:sjc-primary.voicepulse.com", nonce="4acf77fe0000138c5588de6bc2839d988b7d56e31d498b36", response="906ec6dd37242f8e6e0b89af86f35743", qop=auth, cnonce="5028ff91", nc=00000002Expires: 120Contact: Event: registrationContent-Length: 0 --- csgtacsip1*CLI> <--- SIP read from UDP://67.108.9.165:5060 ---> SIP/2.0 100 Trying RegistrationVia: SIP/2.0/UDP 70.35.113.86:5060;branch=z9hG4bK5e91b448;rport=5060From: ;tag=as5e63cb98To: Call-ID: 442cc8df3818c91a3edb029f59f9c0e2@127.0.1.1CSeq: 109 REGISTERServer: OpenSER (1.3.2-notls (i386/linux))Content-Length: 0 <-------------> --- (8 headers 0 lines) --- csgtacsip1*CLI> <--- SIP read from UDP://67.108.9.165:5060 ---> SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 70.35.113.86:5060;branch=z9hG4bK5e91b448;rport=5060From: ;tag=as5e63cb98To: ;tag=c97b4d1cb1f3d0da549e06a8d482ef63.4adaCall-ID: 442cc8df3818c91a3edb029f59f9c0e2@127.0.1.1CSeq: 109 REGISTERWWW-Authenticate: Digest realm="sjc-primary.voicepulse.com", nonce="4acf786700001fdc24744483a4f620900c64e8678541306a", qop="auth", stale=trueServer: OpenSIPS (1.4.2-notls (i386/linux))Content-Length: 0 <-------------> --- (9 headers 0 lines) --- csgtacsip1*CLI> Responding to challenge, registration to domain/host name sjc-primary.voicepulse.com REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 67.108.9.165:5060: REGISTER sip:sjc-primary.voicepulse.com SIP/2.0Via: SIP/2.0/UDP 70.35.113.86:5060;branch=z9hG4bK2a648dc8;rportMax-Forwards: 70From: ;tag=as2c404039To: Call-ID: 442cc8df3818c91a3edb029f59f9c0e2@127.0.1.1CSeq: 110 REGISTERUser-Agent: Asterisk PBX 1.6.0.14Authorization: Digest username="YxD93jsh61", realm="sjc-primary.voicepulse.com", algorithm=MD5, uri="sip:sjc-primary.voicepulse.com", nonce="4acf786700001fdc24744483a4f620900c64e8678541306a", response="02ae44af8f992089495b577e6c8737e4", qop=auth, cnonce="6309f447", nc=00000001Expires: 120Contact: Event: registrationContent-Length: 0 --- csgtacsip1*CLI> <--- SIP read from UDP://67.108.9.165:5060 ---> SIP/2.0 100 Trying RegistrationVia: SIP/2.0/UDP 70.35.113.86:5060;branch=z9hG4bK2a648dc8;rport=5060From: ;tag=as2c404039To: Call-ID: 442cc8df3818c91a3edb029f59f9c0e2@127.0.1.1CSeq: 110 REGISTERServer: OpenSER (1.3.2-notls (i386/linux))Content-Length: 0 <-------------> --- (8 headers 0 lines) --- csgtacsip1*CLI> <--- SIP read from UDP://67.108.9.165:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 70.35.113.86:5060;branch=z9hG4bK2a648dc8;rport=5060From: ;tag=as2c404039To: ;tag=c97b4d1cb1f3d0da549e06a8d482ef63.28b7Call-ID: 442cc8df3818c91a3edb029f59f9c0e2@127.0.1.1CSeq: 110 REGISTERContact: ;expires=106;received="sip:70.35.113.66:12861", ;expires=79;received="sip:67.148.102.2:1188", ;expires=79;received="sip:67.148.102.2:5060", ;expires=72;received="sip:67.148.102.30:5060", ;expires=120;received="sip:70.35.113.86:5060"Server: OpenSIPS (1.4.2-notls (i386/linux))Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Scheduling destruction of SIP dialog '442cc8df3818c91a3edb029f59f9c0e2@127.0.1.1' in 32000 ms (Method: REGISTER) csgtacsip1*CLI> [Oct 9 10:53:15] NOTICE[9003]: chan_sip.c:15899 handle_response_register: Outbound Registration: Expiry for sjc-primary.voicepulse.com is 120 sec (Scheduling reregistration in 105 s) csgtacsip1*CLI> <--- SIP read from TCP://192.168.51.16:5060 ---> BYE sip:1593@192.168.51.15;transport=TCP SIP/2.0FROM: ;epid=FCE3547589;tag=c4d9905f5TO: ;tag=as04f7a591CSEQ: 1 BYECALL-ID: 6c5985216192f0cd650770680154a481@192.168.51.15MAX-FORWARDS: 70VIA: SIP/2.0/TCP 192.168.51.16:5060;branch=z9hG4bKf86862bdCONTENT-LENGTH: 0USER-AGENT: RTCC/3.0.0.0 MediationServer <-------------> --- (9 headers 0 lines) --- Sending to 192.168.51.16 : 5060 (no NAT) csgtacsip1*CLI> <--- Transmitting (no NAT) to 192.168.51.16:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/TCP 192.168.51.16:5060;branch=z9hG4bKf86862bd;received=192.168.51.16From: ;epid=FCE3547589;tag=c4d9905f5To: ;tag=as04f7a591Call-ID: 6c5985216192f0cd650770680154a481@192.168.51.15CSeq: 1 BYEUser-Agent: Asterisk PBX 1.6.0.14Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContent-Length: 0 <------------> csgtacsip1*CLI> -- No one is available to answer at this time (1:0/0/0) csgtacsip1*CLI> -- Executing [1593@from-staff:2] VoiceMail("SIP/1593-f795bcc0", "1593@default,u") in new stack csgtacsip1*CLI> Audio is at 192.168.51.15 port 13222 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP csgtacsip1*CLI> <--- Reliably Transmitting (NAT) to 10.0.11.1:1221 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bKc8792b9b9;received=10.0.11.1From: citywok SVR1 ;tag=e71029057342561To: citywok SVR1 ;tag=as221bab74Call-ID: ccc6b59b7b5ffa1fc6191b84331f52cd@192.168.16.221CSeq: 1347600150 INVITEUser-Agent: Asterisk PBX 1.6.0.14Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerRequire: timerSession-Expires: -1;refresher=uasContact: Content-Type: application/sdpContent-Length: 264v=0o=root 756916649 756916649 IN IP4 192.168.51.15s=Asterisk PBX 1.6.0.14c=IN IP4 192.168.51.15t=0 0m=audio 13222 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv <------------> csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:1221 ---> ACK sip:1593@192.168.51.15 SIP/2.0Via: SIP/2.0/UDP 192.168.16.221:5060;branch=z9hG4bKd0d667626Max-Forwards: 70Content-Length: 0To: citywok SVR1 ;tag=as221bab74From: citywok SVR1 ;tag=e71029057342561Call-ID: ccc6b59b7b5ffa1fc6191b84331f52cd@192.168.16.221CSeq: 1347600150 ACKContact: citywok SVR1 Authorization:Digest response="bb91b43b5ae237529eb003e6e4dd9afb",username="1593",realm="asterisk",nonce="3cfd42a6",algorithm=MD5,uri="sip:1593@192.168.51.15:5060"User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '6c5985216192f0cd650770680154a481@192.168.51.15' Method: BYE csgtacsip1*CLI> [Oct 9 10:53:15] WARNING[9030]: res_config_mysql.c:159 realtime_mysql: MySQL RealTime: Failed to query database. Check debug for more info. [Oct 9 10:53:15] WARNING[9030]: app_voicemail.c:4325 leave_voicemail: No entry in voicemail config file for '1593' -- Auto fallthrough, channel 'SIP/1593-f795bcc0' status is 'NOANSWER' Scheduling destruction of SIP dialog 'ccc6b59b7b5ffa1fc6191b84331f52cd@192.168.16.221' in 11712 ms (Method: ACK) csgtacsip1*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.16.221, port 5060 Reliably Transmitting (NAT) to 10.0.11.1:1221: BYE sip:1593@192.168.16.221:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 192.168.51.15:5060;branch=z9hG4bK4dca39b2;rportMax-Forwards: 70From: citywok SVR1 ;tag=as221bab74To: citywok SVR1 ;tag=e71029057342561Call-ID: ccc6b59b7b5ffa1fc6191b84331f52cd@192.168.16.221CSeq: 102 BYEUser-Agent: Asterisk PBX 1.6.0.14X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0 --- csgtacsip1*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.16.224, port 5060 Reliably Transmitting (NAT) to 10.0.11.1:1218: NOTIFY sip:1512@192.168.16.224:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 192.168.51.15:5060;branch=z9hG4bK4f797d14;rportMax-Forwards: 70From: "" ;tag=as0dd7b765To: "Operator" ;tag=e03550dbfeContact: Call-ID: 2eb15443c9dfb81bCSeq: 104 NOTIFYUser-Agent: Asterisk PBX 1.6.0.14Event: dialogContent-Type: application/dialog-info+xmlSubscription-State: activeContent-Length: 211 terminated --- == Extension Changed 1593[subscribecontext] new state Idle for Notify User 1512 csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:1218 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.51.15:5060;branch=z9hG4bK4f797d14;rport=5060;received=192.168.51.15From: "" ;tag=as0dd7b765To: "Operator" ;tag=e03550dbfeCall-ID: 2eb15443c9dfb81bCSeq: 104 NOTIFYContact: "Operator" ;+sip.instance=""Server: Aastra 57iCT/2.5.2.1010Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived csgtacsip1*CLI> <--- SIP read from UDP://10.0.11.1:1221 ---> SIP/2.0 200 OKCall-ID: ccc6b59b7b5ffa1fc6191b84331f52cd@192.168.16.221CSeq: 102 BYEFrom: citywok SVR1 ;tag=as221bab74To: citywok SVR1 ;tag=e71029057342561Via: SIP/2.0/UDP 192.168.51.15:5060;branch=z9hG4bK4dca39b2;rportContent-Length: 0Supported: replacesUser-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'ccc6b59b7b5ffa1fc6191b84331f52cd@192.168.16.221' Method: ACK csgtacsip1*CLI> Disconnected from Asterisk server csgtacsip1:~/asterisk-1.6.0.14#