ourcompany*CLI> <--- SIP read from UDP:192.168.0.102:5060 ---> INVITE sip:99192130@192.168.0.211;user=phone SIP/2.0 Route: Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK0d0o8lo4gdhc6hd91796mnu;rport From: Anonymous ;tag=mq1qahtgulhc7m5n1793 To: Contact: Supported: 100rel,sec-agree CSeq: 1080 INVITE Call-ID: XOa6nARcoIdOPMDisoc1fD_Zeju3w2 Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK Expires: 120 Privacy: id User-Agent: Nokia RM-179 V 21.0.010 P-Preferred-Identity: sip:603@192.168.0.211 Max-Forwards: 70 Content-Type: application/sdp Accept: application/sdp Content-Length: 447 v=0 o=Nokia-SIPUA 63426131115188875 63426131115188875 IN IP4 192.168.0.102 s=- c=IN IP4 192.168.0.102 t=0 0 m=audio 49152 RTP/AVP 96 0 8 97 18 98 13 a=sendrecv a=ptime:20 a=maxptime:200 a=fmtp:96 mode-change-neighbor=1 a=fmtp:18 annexb=no a=fmtp:98 0-15 a=rtpmap:96 AMR/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:97 iLBC/8000/1 a=rtpmap:18 G729/8000/1 a=rtpmap:98 telephone-event/8000/1 a=rtpmap:13 CN/8000/1 <-------------> --- (18 headers 19 lines) --- == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL CoS mark 5 Sending to 192.168.0.102 : 5060 (NAT) Using INVITE request as basis request - XOa6nARcoIdOPMDisoc1fD_Zeju3w2 Found peer '603' for 'anonymous' from 192.168.0.102:5060 Found RTP audio format 96 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 18 Found RTP audio format 98 Found RTP audio format 13 Found audio description format AMR for ID 96 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format iLBC for ID 97 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 98 Capabilities: us - 0x18000e (gsm|ulaw|alaw|h263|h263p), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x c (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.0.102:49152 Peer doesn't provide video Looking for 99192130 in DLPN_DialPlan1 (domain 192.168.0.211) list_route: hop: <--- Transmitting (NAT) to 192.168.0.102:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK0d0o8lo4gdhc6hd91796mnu;received=192.168.0.102;rport=5060 From: Anonymous ;tag=mq1qahtgulhc7m5n1793 To: Call-ID: XOa6nARcoIdOPMDisoc1fD_Zeju3w2 CSeq: 1080 INVITE Server: Asterisk_Eut Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [99192130@DLPN_DialPlan1:1] Macro("SIP/603-00000023", "trunkdial-failover-0.3,SIP/Eutelia2/192130,SIP/Eutelia1/192130,Eutelia2,Euteli a1") in new stack -- Executing [s@macro-trunkdial-failover-0.3:1] GotoIf("SIP/603-00000023", "0?1-fmsetcid,1") in new stack -- Executing [s@macro-trunkdial-failover-0.3:2] GotoIf("SIP/603-00000023", "1?1-setgbobname,1") in new stack -- Goto (macro-trunkdial-failover-0.3,1-setgbobname,1) -- Executing [1-setgbobname@macro-trunkdial-failover-0.3:1] Set("SIP/603-00000023", "CALLERID(name)=ourcompany") in new stack -- Executing [1-setgbobname@macro-trunkdial-failover-0.3:2] Goto("SIP/603-00000023", "s,3") in new stack -- Goto (macro-trunkdial-failover-0.3,s,3) -- Executing [s@macro-trunkdial-failover-0.3:3] Set("SIP/603-00000023", "CALLERID(num)=") in new stack -- Executing [s@macro-trunkdial-failover-0.3:4] GotoIf("SIP/603-00000023", "0?1-dial,1") in new stack -- Executing [s@macro-trunkdial-failover-0.3:5] Set("SIP/603-00000023", "CALLERID(all)=035123456") in new stack -- Executing [s@macro-trunkdial-failover-0.3:6] Goto("SIP/603-00000023", "1-dial,1") in new stack -- Goto (macro-trunkdial-failover-0.3,1-dial,1) -- Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial("SIP/603-00000023", "SIP/Eutelia2/192130") in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL CoS mark 5 Audio is at 12.34.56.78 port 8700 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x800 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 83.211.227.21:5060: INVITE sip:192130@voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK3798c43c;rport Max-Forwards: 70 From: "035123456" ;tag=as55bb8856 To: Contact: Call-ID: 7e29b8b81b762ba7203d7f8674ddfa76@voip.eutelia.it CSeq: 102 INVITE User-Agent: Asterisk_Eut Date: Tue, 10 Nov 2009 16:43:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 328 v=0 o=root 1533520756 1533520756 IN IP4 12.34.56.78 s=Asterisk PBX SVN-branch-1.6.2-r229101 c=IN IP4 12.34.56.78 t=0 0 m=audio 8700 RTP/AVP 0 8 3 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called Eutelia2/192130 ourcompany*CLI> <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK3798c43c;rport=5060 From: "035123456" ;tag=as55bb8856 To: ;tag=d7cbdeb4f107ce82ed834cadd3d6dbb2.c51a Call-ID: 7e29b8b81b762ba7203d7f8674ddfa76@voip.eutelia.it CSeq: 102 INVITE Proxy-Authenticate: Digest realm="voip.eutelia.it", nonce="4af999b040bdd4376279fd07d011c6ae2b7f8325", qop="auth" Server: SPS EUT RM GW 04 (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 83.211.227.21:5060 "Noisy feedback tells: pid=25032 req_src_ip=12.34.56.78 req_src_port=5060 in_uri=sip:192130@voip.eutelia.it out_uri=s ip:192130@voip.eutelia.it via_cnt==1" <-------------> --- (10 headers 0 lines) --- Transmitting (NAT) to 83.211.227.21:5060: ACK sip:192130@voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK3798c43c;rport Max-Forwards: 70 From: "035123456" ;tag=as55bb8856 To: ;tag=d7cbdeb4f107ce82ed834cadd3d6dbb2.c51a Contact: Call-ID: 7e29b8b81b762ba7203d7f8674ddfa76@voip.eutelia.it CSeq: 102 ACK User-Agent: Asterisk_Eut Content-Length: 0 --- Audio is at 12.34.56.78 port 8700 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x800 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 83.211.227.21:5060: INVITE sip:192130@voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK2bb548ab;rport Max-Forwards: 70 From: "035123456" ;tag=as55bb8856 To: Contact: Call-ID: 7e29b8b81b762ba7203d7f8674ddfa76@voip.eutelia.it CSeq: 103 INVITE User-Agent: Asterisk_Eut Proxy-Authorization: Digest username="03519912345", realm="voip.eutelia.it", algorithm=MD5, uri="sip:192130@voip.eutelia.it", nonce="4af999b040bdd4376 279fd07d011c6ae2b7f8325", response="4f69e108c674a0cd74d4e73877e3b7bf", qop=auth, cnonce="62b6def0", nc=00000001 Date: Tue, 10 Nov 2009 16:43:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 328 v=0 o=root 1533520756 1533520757 IN IP4 12.34.56.78 s=Asterisk PBX SVN-branch-1.6.2-r229101 c=IN IP4 12.34.56.78 t=0 0 m=audio 8700 RTP/AVP 0 8 3 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- ourcompany*CLI> <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK2bb548ab;rport=5060 From: "035123456" ;tag=as55bb8856 To: Call-ID: 7e29b8b81b762ba7203d7f8674ddfa76@voip.eutelia.it CSeq: 103 INVITE Server: SPS EUT RM GW 04 (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 83.211.227.21:5060 "Noisy feedback tells: pid=25037 req_src_ip=12.34.56.78 req_src_port=5060 in_uri=sip:192130@voip.eutelia.it out_uri=s ip:491192130@62.94.71.98:5060 via_cnt==1" <-------------> --- (9 headers 0 lines) --- ourcompany*CLI> <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK2bb548ab;rport=5060 From: "035123456" ;tag=as55bb8856 To: ;tag=62F96714-B89 Date: Tue, 10 Nov 2009 16:44:52 GMT Call-ID: 7e29b8b81b762ba7203d7f8674ddfa76@voip.eutelia.it Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Record-Route: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 246 v=0 o=CiscoSystemsSIP-GW-UserAgent 9308 4128 IN IP4 62.94.71.98 s=SIP Call c=IN IP4 62.94.199.36 t=0 0 m=audio 62486 RTP/AVP 0 101 c=IN IP4 62.94.199.36 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (15 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0x4 (ulaw)/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 62.94.199.36:62486 -- SIP/Eutelia2-00000024 is making progress passing it to SIP/603-00000023 Audio is at 192.168.0.211 port 9012 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 192.168.0.102:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK0d0o8lo4gdhc6hd91796mnu;received=192.168.0.102;rport=5060 From: Anonymous ;tag=mq1qahtgulhc7m5n1793 To: ;tag=as746c53d2 Call-ID: XOa6nARcoIdOPMDisoc1fD_Zeju3w2 CSeq: 1080 INVITE Server: Asterisk_Eut Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 273 v=0 o=root 796852349 796852349 IN IP4 192.168.0.211 s=Asterisk PBX SVN-branch-1.6.2-r229101 c=IN IP4 192.168.0.211 t=0 0 m=audio 9012 RTP/AVP 0 8 98 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:98 telephone-event/8000 a=fmtp:98 0-16 a=ptime:20 a=sendrecv <------------> ourcompany*CLI> <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK2bb548ab;rport=5060 From: "035123456" ;tag=as55bb8856 To: ;tag=62F96714-B89 Date: Tue, 10 Nov 2009 16:44:52 GMT Call-ID: 7e29b8b81b762ba7203d7f8674ddfa76@voip.eutelia.it Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Supported: replaces Allow-Events: telephone-event Contact: Record-Route: Content-Type: application/sdp Content-Length: 246 v=0 o=CiscoSystemsSIP-GW-UserAgent 9308 4128 IN IP4 62.94.71.98 s=SIP Call c=IN IP4 62.94.199.36 t=0 0 m=audio 62486 RTP/AVP 0 101 c=IN IP4 62.94.199.36 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (15 headers 11 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 83.211.227.21, port 5060 Transmitting (NAT) to 83.211.227.21:5060: ACK sip:491192130@62.94.71.98:5060 SIP/2.0 Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK1d164065;rport Route: Max-Forwards: 70 From: "035123456" ;tag=as55bb8856 To: ;tag=62F96714-B89 Contact: Call-ID: 7e29b8b81b762ba7203d7f8674ddfa76@voip.eutelia.it CSeq: 103 ACK User-Agent: Asterisk_Eut Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 83.211.227.21, port 5060 Reliably Transmitting (NAT) to 83.211.227.21:5060: BYE sip:491192130@62.94.71.98:5060 SIP/2.0 Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK0ca8f087;rport Route: Max-Forwards: 70 From: "035123456" ;tag=as55bb8856 To: ;tag=62F96714-B89 Call-ID: 7e29b8b81b762ba7203d7f8674ddfa76@voip.eutelia.it CSeq: 104 BYE User-Agent: Asterisk_Eut Proxy-Authorization: Digest username="03519912345", realm="voip.eutelia.it", algorithm=MD5, uri="sip:491192130@62.94.71.98:5060", nonce="4af999b040bdd 4376279fd07d011c6ae2b7f8325", response="a8b4a268e18d024710706bbd4346d1ba", qop=auth, cnonce="4e029ac2", nc=00000002 X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 --- Scheduling destruction of SIP dialog '7e29b8b81b762ba7203d7f8674ddfa76@voip.eutelia.it' in 6400 ms (Method: INVITE) -- SIP/Eutelia2-00000024 answered SIP/603-00000023 Audio is at 192.168.0.211 port 9012 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.0.102:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK0d0o8lo4gdhc6hd91796mnu;received=192.168.0.102;rport=5060 From: Anonymous ;tag=mq1qahtgulhc7m5n1793 To: ;tag=as746c53d2 Call-ID: XOa6nARcoIdOPMDisoc1fD_Zeju3w2 CSeq: 1080 INVITE Server: Asterisk_Eut Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 273 v=0 o=root 796852349 796852350 IN IP4 192.168.0.211 s=Asterisk PBX SVN-branch-1.6.2-r229101 c=IN IP4 192.168.0.211 t=0 0 m=audio 9012 RTP/AVP 0 8 98 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:98 telephone-event/8000 a=fmtp:98 0-16 a=ptime:20 a=sendrecv <------------> ourcompany*CLI> <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK0ca8f087;rport=5060 From: "035123456" ;tag=as55bb8856 To: ;tag=62F96714-B89 Date: Tue, 10 Nov 2009 16:44:57 GMT Call-ID: 7e29b8b81b762ba7203d7f8674ddfa76@voip.eutelia.it Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 104 BYE <-------------> --- (9 headers 0 lines) --- ourcompany*CLI> <--- SIP read from UDP:192.168.0.102:5060 ---> ACK sip:99192130@192.168.0.211;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bKb1fe2ktkd3l7b31soprmorj;rport To: ;tag=as746c53d2 From: Anonymous ;tag=mq1qahtgulhc7m5n1793 Call-ID: XOa6nARcoIdOPMDisoc1fD_Zeju3w2 CSeq: 1080 ACK Supported: sec-agree Max-Forwards: 70 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Reliably Transmitting (NAT) to 192.168.0.102:5060: OPTIONS sip:603@192.168.0.102;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK068671f5;rport Max-Forwards: 70 From: "asterisk" ;tag=as49b32f78 To: Contact: Call-ID: 67ef92cf5bb58edd1b6703fe20b831e2@192.168.0.211 CSeq: 102 OPTIONS User-Agent: Asterisk_Eut Date: Tue, 10 Nov 2009 16:43:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- ourcompany*CLI> <--- SIP read from UDP:192.168.0.102:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK068671f5;rport=5060;received=192.168.0.211 To: ;tag=nupuepuo31hc75s208s5 From: "asterisk" ;tag=as49b32f78 Call-ID: 67ef92cf5bb58edd1b6703fe20b831e2@192.168.0.211 CSeq: 102 OPTIONS Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '67ef92cf5bb58edd1b6703fe20b831e2@192.168.0.211' Method: OPTIONS Reliably Transmitting (NAT) to 83.211.227.21:5060: OPTIONS sip:voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK49c9cbf5;rport Max-Forwards: 70 From: "asterisk" ;tag=as17b08caa To: Contact: Call-ID: 7c1e4204570fedc9599ed2d502248039@12.34.56.78 CSeq: 102 OPTIONS User-Agent: Asterisk_Eut Date: Tue, 10 Nov 2009 16:43:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (NAT) to 83.211.227.21:5060: OPTIONS sip:voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK77c346d0;rport Max-Forwards: 70 From: "asterisk" ;tag=as303bcbcf To: Contact: Call-ID: 05343d7d5089657008b2890612e2151f@12.34.56.78 CSeq: 102 OPTIONS User-Agent: Asterisk_Eut Date: Tue, 10 Nov 2009 16:43:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (NAT) to 83.211.227.21:5060: OPTIONS sip:voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK0a95f1f1;rport Max-Forwards: 70 From: "asterisk" ;tag=as08a14c35 To: Contact: Call-ID: 7b74cfb51a120f4555cc31ae6548e589@12.34.56.78 CSeq: 102 OPTIONS User-Agent: Asterisk_Eut Date: Tue, 10 Nov 2009 16:43:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (NAT) to 83.211.227.21:5060: OPTIONS sip:voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK494a71ed;rport Max-Forwards: 70 From: "asterisk" ;tag=as21f43aef To: Contact: Call-ID: 1f5f7f1100707a9b4f08d44c38b721ed@12.34.56.78 CSeq: 102 OPTIONS User-Agent: Asterisk_Eut Date: Tue, 10 Nov 2009 16:43:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- ourcompany*CLI> <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK49c9cbf5;rport=5060 From: "asterisk" ;tag=as17b08caa To: ;tag=d7cbdeb4f107ce82ed834cadd3d6dbb2.e961 Call-ID: 7c1e4204570fedc9599ed2d502248039@12.34.56.78 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Support: Server: SPS EUT RM GW 04 (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 83.211.227.21:5060 "Noisy feedback tells: pid=25035 req_src_ip=12.34.56.78 req_src_port=5060 in_uri=sip:voip.eutelia.it out_uri=sip:voip .eutelia.it via_cnt==1" <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '7c1e4204570fedc9599ed2d502248039@12.34.56.78' Method: OPTIONS ourcompany*CLI> <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK77c346d0;rport=5060 From: "asterisk" ;tag=as303bcbcf To: ;tag=d7cbdeb4f107ce82ed834cadd3d6dbb2.f605 Call-ID: 05343d7d5089657008b2890612e2151f@12.34.56.78 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Support: Server: SPS EUT RM GW 04 (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 83.211.227.21:5060 "Noisy feedback tells: pid=25035 req_src_ip=12.34.56.78 req_src_port=5060 in_uri=sip:voip.eutelia.it out_uri=sip:voip .eutelia.it via_cnt==1" <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '05343d7d5089657008b2890612e2151f@12.34.56.78' Method: OPTIONS ourcompany*CLI> <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK0a95f1f1;rport=5060 From: "asterisk" ;tag=as08a14c35 To: ;tag=d7cbdeb4f107ce82ed834cadd3d6dbb2.8c6f Call-ID: 7b74cfb51a120f4555cc31ae6548e589@12.34.56.78 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Support: Server: SPS EUT RM GW 04 (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 83.211.227.21:5060 "Noisy feedback tells: pid=25033 req_src_ip=12.34.56.78 req_src_port=5060 in_uri=sip:voip.eutelia.it out_uri=sip:voip .eutelia.it via_cnt==1" <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '7b74cfb51a120f4555cc31ae6548e589@12.34.56.78' Method: OPTIONS ourcompany*CLI> <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK494a71ed;rport=5060 From: "asterisk" ;tag=as21f43aef To: ;tag=d7cbdeb4f107ce82ed834cadd3d6dbb2.577c Call-ID: 1f5f7f1100707a9b4f08d44c38b721ed@12.34.56.78 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Support: Server: SPS EUT RM GW 02 (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 83.211.227.21:5060 "Noisy feedback tells: pid=2480 req_src_ip=12.34.56.78 req_src_port=5060 in_uri=sip:voip.eutelia.it out_uri=sip:voip. eutelia.it via_cnt==1" <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '1f5f7f1100707a9b4f08d44c38b721ed@12.34.56.78' Method: OPTIONS [Nov 10 17:43:52] WARNING[32468]: chan_sip.c:3973 __sip_autodestruct: Autodestruct on dialog '7e29b8b81b762ba7203d7f8674ddfa76@voip.eutelia.it' with o wner in place (Method: INVITE) Scheduling destruction of SIP dialog '7e29b8b81b762ba7203d7f8674ddfa76@voip.eutelia.it' in 6400 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 83.211.227.21, port 5060 Reliably Transmitting (NAT) to 83.211.227.21:5060: BYE sip:491192130@62.94.71.98:5060 SIP/2.0 Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK4870e764;rport Route: Max-Forwards: 70 From: "035123456" ;tag=as55bb8856 To: ;tag=62F96714-B89 Call-ID: 7e29b8b81b762ba7203d7f8674ddfa76@voip.eutelia.it CSeq: 105 BYE User-Agent: Asterisk_Eut Proxy-Authorization: Digest username="03519912345", realm="voip.eutelia.it", algorithm=MD5, uri="sip:491192130@62.94.71.98:5060", nonce="4af999b040bdd 4376279fd07d011c6ae2b7f8325", response="437bcd82e51029870a49742078c4b5b3", qop=auth, cnonce="0a446164", nc=00000003 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero on 'SIP/603-00000023' in macro 'trunkdial-failover-0.3' == Spawn extension (DLPN_DialPlan1, 99192130, 1) exited non-zero on 'SIP/603-00000023' Scheduling destruction of SIP dialog 'XOa6nARcoIdOPMDisoc1fD_Zeju3w2' in 7232 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.102, port 5060 Reliably Transmitting (NAT) to 192.168.0.102:5060: BYE sip:603@192.168.0.102;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK33951cef;rport Max-Forwards: 70 From: ;tag=as746c53d2 To: Anonymous ;tag=mq1qahtgulhc7m5n1793 Call-ID: XOa6nARcoIdOPMDisoc1fD_Zeju3w2 CSeq: 102 BYE User-Agent: Asterisk_Eut X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- ourcompany*CLI> <--- SIP read from UDP:192.168.0.102:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.211:5060;branch=z9hG4bK33951cef;rport=5060;received=192.168.0.211 To: Anonymous ;tag=mq1qahtgulhc7m5n1793 From: ;tag=as746c53d2 Call-ID: XOa6nARcoIdOPMDisoc1fD_Zeju3w2 CSeq: 102 BYE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'XOa6nARcoIdOPMDisoc1fD_Zeju3w2' Method: ACK ourcompany*CLI> <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK4870e764;rport=5060 From: "035123456" ;tag=as55bb8856 To: ;tag=62F96714-B89 Call-ID: 7e29b8b81b762ba7203d7f8674ddfa76@voip.eutelia.it CSeq: 105 BYE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '7e29b8b81b762ba7203d7f8674ddfa76@voip.eutelia.it' Method: INVITE ourcompany*CLI> sip set debug off SIP Debugging Disabled