asterisk*CLI> Verbosity is at least 100 Core debug is at least 100 asterisk*CLI>  <--- SIP read from UDP:192.168.1.151:5060 ---> <-------------> asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> INVITE sip:dialed-number@asterisk-public-ip;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-e27a94c14a764896-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: "153";tag=e9c4fc35 Call-ID: NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.6739 Content-Length: 304 v=0 o=Z 0 0 IN IP4 192.168.1.44 s=Z c=IN IP4 192.168.1.44 t=0 0 m=audio 8000 RTP/AVP 98 0 8 3 110 101 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (12 headers 15 lines) --- asterisk*CLI>  == Using UDPTL CoS mark 5 asterisk*CLI> Sending to 192.168.1.44:5060 (no NAT) Using INVITE request as basis request - NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E. Found peer '153' for '153' from 192.168.1.44:5060 asterisk*CLI>  <--- Reliably Transmitting (no NAT) to 192.168.1.44:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-e27a94c14a764896-1---d8754z-;received=192.168.1.44;rport=5060 From: "153";tag=e9c4fc35 To: ;tag=as2c4d52ed Call-ID: NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E. CSeq: 1 INVITE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="781b612d" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E.' in 6912 ms (Method: INVITE) asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> ACK sip:dialed-number@asterisk-public-ip;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-e27a94c14a764896-1---d8754z-;rport Max-Forwards: 70 To: ;tag=as2c4d52ed From: "153";tag=e9c4fc35 Call-ID: NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E. CSeq: 1 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> INVITE sip:dialed-number@asterisk-public-ip;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-8abdc99fba791c2f-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: "153";tag=e9c4fc35 Call-ID: NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.6739 Authorization: Digest username="153",realm="asterisk",nonce="781b612d",uri="sip:dialed-number@asterisk-public-ip;transport=UDP",response="9257eba0a861cbc944fe20239afcacbd",algorithm=MD5 Content-Length: 304 v=0 o=Z 0 0 IN IP4 192.168.1.44 s=Z c=IN IP4 192.168.1.44 t=0 0 m=audio 8000 RTP/AVP 98 0 8 3 110 101 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (13 headers 15 lines) --- asterisk*CLI> Sending to 192.168.1.44:5060 (no NAT) Using INVITE request as basis request - NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E. Found peer '153' for '153' from 192.168.1.44:5060 asterisk*CLI>  == Using SIP RTP CoS mark 5 asterisk*CLI> Found RTP audio format 98 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 110 Found RTP audio format 101 Found audio description format iLBC for ID 98 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format GSM for ID 3 Found audio description format speex for ID 110 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.1.44:8000 Looking for dialed-number in phones-sip (domain asterisk-public-ip) asterisk*CLI> list_route: hop: <--- Transmitting (no NAT) to 192.168.1.44:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-8abdc99fba791c2f-1---d8754z-;received=192.168.1.44;rport=5060 From: "153";tag=e9c4fc35 To: Call-ID: NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E. CSeq: 2 INVITE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [dialed-number@phones-sip:1] Dial("SIP/153-00000036", "SIP/eutelia/dialed-number,60") in new stack asterisk*CLI>  == Using UDPTL CoS mark 5 asterisk*CLI>  == Using SIP RTP CoS mark 5 asterisk*CLI> Audio is at 5060 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 83.211.227.21:5060: INVITE sip:dialed-number@voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK3ffe5b88 Max-Forwards: 70 From: "153" ;tag=as5159f324 To: Contact: Call-ID: 004c464f69d022e202c355f72d91976f@asterisk-public-ip:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 14:07:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 251 v=0 o=root 1473574214 1473574214 IN IP4 asterisk-public-ip s=Asterisk PBX 1.8.3-1digium1~squeeze c=IN IP4 asterisk-public-ip t=0 0 m=audio 18200 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called eutelia/dialed-number asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK3ffe5b88 From: "153" ;tag=as5159f324 To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.9a15 Call-ID: 004c464f69d022e202c355f72d91976f@asterisk-public-ip:5060 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="voip.eutelia.it", nonce="4da45ce33ccae1bf338ca7a3fed129344ac517fd", qop="auth" Server: SPS EUT RM GW 02 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Transmitting (no NAT) to 83.211.227.21:5060: ACK sip:dialed-number@voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK3ffe5b88 Max-Forwards: 70 From: "153" ;tag=as5159f324 To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.9a15 Contact: Call-ID: 004c464f69d022e202c355f72d91976f@asterisk-public-ip:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Content-Length: 0 --- Audio is at 5060 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 83.211.227.21:5060: INVITE sip:dialed-number@voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK35c3b8a3 Max-Forwards: 70 From: "153" ;tag=as5159f324 To: Contact: Call-ID: 004c464f69d022e202c355f72d91976f@asterisk-public-ip:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Proxy-Authorization: Digest username="my-eutelia-number", realm="voip.eutelia.it", algorithm=MD5, uri="sip:dialed-number@voip.eutelia.it", nonce="4da45ce33ccae1bf338ca7a3fed129344ac517fd", response="f95e264743b6f26100dfe8c58833d4a1", qop=auth, cnonce="68436bf3", nc=00000001 Date: Tue, 12 Apr 2011 14:07:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 251 v=0 o=root 1473574214 1473574215 IN IP4 asterisk-public-ip s=Asterisk PBX 1.8.3-1digium1~squeeze c=IN IP4 asterisk-public-ip t=0 0 m=audio 18200 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK35c3b8a3 From: "153" ;tag=as5159f324 To: Call-ID: 004c464f69d022e202c355f72d91976f@asterisk-public-ip:5060 CSeq: 103 INVITE Server: SPS EUT RM GW 02 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk*CLI> Reliably Transmitting (no NAT) to 192.168.1.151:5060: OPTIONS sip:152@192.168.1.151:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK308d844a Max-Forwards: 70 From: "asterisk" ;tag=as5931eaa7 To: Contact: Call-ID: 0a26bba30a9eb0391c64c6eb6e2dd9c0@asterisk-public-ip:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 14:07:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI>  <--- SIP read from UDP:192.168.1.151:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK308d844a From: "asterisk" ;tag=as5931eaa7 To: ;tag=3144949534 Call-ID: 0a26bba30a9eb0391c64c6eb6e2dd9c0@asterisk-public-ip:5060 CSeq: 102 OPTIONS Contact: "152" Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Accept: application/sdp,application/dtmf-relay Accept-Encoding: identity Accept-Language: en Content-Length: 0 asterisk*CLI> <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '0a26bba30a9eb0391c64c6eb6e2dd9c0@asterisk-public-ip:5060' Method: OPTIONS asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK35c3b8a3 From: "153" ;tag=as5159f324 To: ;tag=14461840-D8F Date: Tue, 12 Apr 2011 14:07:35 GMT Call-ID: 004c464f69d022e202c355f72d91976f@asterisk-public-ip:5060 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: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 4250 820 IN IP4 83.211.2.218 s=SIP Call c=IN IP4 83.211.223.194 t=0 0 m=audio 50640 RTP/AVP 8 101 c=IN IP4 83.211.223.194 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (15 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 83.211.223.194:50640 -- SIP/eutelia-00000037 is making progress passing it to SIP/153-00000036 Audio is at 5060 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP asterisk*CLI>  <--- Transmitting (no NAT) to 192.168.1.44:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-8abdc99fba791c2f-1---d8754z-;received=192.168.1.44;rport=5060 From: "153";tag=e9c4fc35 To: ;tag=as4de0b0df Call-ID: NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E. CSeq: 2 INVITE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 249 v=0 o=root 157384482 157384482 IN IP4 asterisk-public-ip s=Asterisk PBX 1.8.3-1digium1~squeeze c=IN IP4 asterisk-public-ip t=0 0 m=audio 19164 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> asterisk*CLI> Really destroying SIP dialog '04a1c3ee184640a22c3d58216f64f537@192.168.3.2' Method: REGISTER asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK35c3b8a3 From: "153" ;tag=as5159f324 To: ;tag=14461840-D8F Date: Tue, 12 Apr 2011 14:07:35 GMT Call-ID: 004c464f69d022e202c355f72d91976f@asterisk-public-ip:5060 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: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 4250 820 IN IP4 83.211.2.218 s=SIP Call c=IN IP4 83.211.223.194 t=0 0 m=audio 50640 RTP/AVP 8 101 c=IN IP4 83.211.223.194 a=rtpmap:8 PCMA/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 83.211.227.21:5060 Transmitting (no NAT) to 83.211.227.21:5060: ACK sip:494dialed-number@83.211.2.218:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK065c9a80 Route: Max-Forwards: 70 From: "153" ;tag=as5159f324 To: ;tag=14461840-D8F Contact: Call-ID: 004c464f69d022e202c355f72d91976f@asterisk-public-ip:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Content-Length: 0 --- -- SIP/eutelia-00000037 answered SIP/153-00000036 Audio is at 5060 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.1.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-8abdc99fba791c2f-1---d8754z-;received=192.168.1.44;rport=5060 From: "153";tag=e9c4fc35 To: ;tag=as4de0b0df Call-ID: NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E. CSeq: 2 INVITE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 249 v=0 o=root 157384482 157384483 IN IP4 asterisk-public-ip s=Asterisk PBX 1.8.3-1digium1~squeeze c=IN IP4 asterisk-public-ip t=0 0 m=audio 19164 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> -- Locally bridging SIP/153-00000036 and SIP/eutelia-00000037 asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> ACK sip:dialed-number@asterisk-public-ip:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-e68df287d85de4fe-1---d8754z-;rport Max-Forwards: 70 Contact: To: ;tag=as4de0b0df From: "153";tag=e9c4fc35 Call-ID: NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E. CSeq: 2 ACK User-Agent: Zoiper rev.6739 Authorization: Digest username="153",realm="asterisk",nonce="781b612d",uri="sip:dialed-number@asterisk-public-ip;transport=UDP",response="9257eba0a861cbc944fe20239afcacbd",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> INVITE sip:dialed-number@asterisk-public-ip:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-7bce31723d21281a-1---d8754z-;rport Max-Forwards: 70 Contact: To: ;tag=as4de0b0df From: "153";tag=e9c4fc35 Call-ID: NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E. CSeq: 3 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.6739 Authorization: Digest username="153",realm="asterisk",nonce="781b612d",uri="sip:dialed-number@asterisk-public-ip:5060",response="fcfbcd37d00fa65924148c6a11a4786b",algorithm=MD5 Content-Length: 333 v=0 o=Z 0 1 IN IP4 192.168.1.44 s=Z c=IN IP4 192.168.1.44 t=0 0 m=image 8000 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (13 headers 15 lines) --- Sending to 192.168.1.44:5060 (no NAT) Got T.38 offer in SDP in dialog NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E. Capabilities: us - 0x8 (alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. <--- Transmitting (no NAT) to 192.168.1.44:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-7bce31723d21281a-1---d8754z-;received=192.168.1.44;rport=5060 From: "153";tag=e9c4fc35 To: ;tag=as4de0b0df Call-ID: NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E. CSeq: 3 INVITE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 83.211.227.21:5060 Reliably Transmitting (no NAT) to 83.211.227.21:5060: INVITE sip:494dialed-number@83.211.2.218:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK41b7a170 Route: Max-Forwards: 70 From: "153" ;tag=as5159f324 To: ;tag=14461840-D8F Contact: Call-ID: 004c464f69d022e202c355f72d91976f@asterisk-public-ip:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 279 v=0 o=root 1473574214 1473574216 IN IP4 asterisk-public-ip s=Asterisk PBX 1.8.3-1digium1~squeeze c=IN IP4 asterisk-public-ip t=0 0 m=image 4844 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:204 a=T38FaxUdpEC:t38UDPFEC --- asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK41b7a170 From: "153" ;tag=as5159f324 To: ;tag=14461840-D8F Call-ID: 004c464f69d022e202c355f72d91976f@asterisk-public-ip:5060 CSeq: 104 INVITE Server: SPS EUT RM GW 02 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK41b7a170 From: "153" ;tag=as5159f324 To: ;tag=14461840-D8F Date: Tue, 12 Apr 2011 14:07:44 GMT Call-ID: 004c464f69d022e202c355f72d91976f@asterisk-public-ip:5060 Server: Cisco-SIPGateway/IOS-12.x CSeq: 104 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: 265 v=0 o=CiscoSystemsSIP-GW-UserAgent 4250 821 IN IP4 83.211.2.218 s=SIP Call c=IN IP4 83.211.223.194 t=0 0 m=image 51712 udptl t38 c=IN IP4 83.211.223.194 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:72 <-------------> --- (15 headers 11 lines) --- Got T.38 offer in SDP in dialog 004c464f69d022e202c355f72d91976f@asterisk-public-ip:5060 Capabilities: us - 0x8 (alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. set_destination: Parsing for address/port to send to set_destination: set destination to 83.211.227.21:5060 T asterisk*CLI> ransmitting (no NAT) to 83.211.227.21:5060: ACK sip:494dialed-number@83.211.2.218:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK4582e4fb Route: Max-Forwards: 70 From: "153" ;tag=as5159f324 To: ;tag=14461840-D8F Contact: Call-ID: 004c464f69d022e202c355f72d91976f@asterisk-public-ip:5060 CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Content-Length: 0 --- <--- Reliably Transmitting (no NAT) to 192.168.1.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-7bce31723d21281a-1---d8754z-;received=192.168.1.44;rport=5060 From: "153";tag=e9c4fc35 To: ;tag=as4de0b0df Call-ID: NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E. CSeq: 3 INVITE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 285 v=0 o=root 157384482 157384484 IN IP4 asterisk-public-ip s=Asterisk PBX 1.8.3-1digium1~squeeze c=IN IP4 asterisk-public-ip t=0 0 m=image 4558 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy <------------> asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> ACK sip:dialed-number@asterisk-public-ip:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-50da1a8cec363434-1---d8754z-;rport Max-Forwards: 70 Contact: To: ;tag=as4de0b0df From: "153";tag=e9c4fc35 Call-ID: NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E. CSeq: 3 ACK User-Agent: Zoiper rev.6739 Authorization: Digest username="153",realm="asterisk",nonce="781b612d",uri="sip:dialed-number@asterisk-public-ip:5060",response="fcfbcd37d00fa65924148c6a11a4786b",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk*CLI>  <--- SIP read from UDP:192.168.3.4:5060 ---> <-------------> asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> <-------------> asterisk*CLI>  <--- SIP read from UDP:192.168.3.4:5060 ---> REGISTER sip:192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.4:5060;branch=z9hG4bKb987c066a32ac26289d13f3fabd5e8c8;rport From: "102" ;tag=3007678749 To: "102" Call-ID: 2999504275@192_168_3_4 CSeq: 966 REGISTER Contact: "102" Max-Forwards: 70 User-Agent: C450 IP010720000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.3.4:5060 (no NAT) <--- Transmitting (no NAT) to 192.168.3.4:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.3.4:5060;branch=z9hG4bKb987c066a32ac26289d13f3fabd5e8c8;received=192.168.3.4;rport=5060 From: "102" ;tag=3007678749 To: "102" ;tag=as30763706 Call-ID: 2999504275@192_168_3_4 CSeq: 966 REGISTER Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2e02edce" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2999504275@192_168_3_4' in 32000 ms (Method: REGISTER) asterisk*CLI>  <--- SIP read from UDP:192.168.3.4:5060 ---> REGISTER sip:192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.4:5060;branch=z9hG4bK64a33d006fc8081210bdc6f488177270;rport From: "102" ;tag=3007678749 To: "102" Call-ID: 2999504275@192_168_3_4 CSeq: 967 REGISTER Contact: "102" Authorization: Digest username="102", realm="asterisk", algorithm=MD5, uri="sip:192.168.3.1", nonce="2e02edce", response="bde35e1bba692d5c6fedd8d1a1b1708f" Max-Forwards: 70 User-Agent: C450 IP010720000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.3.4:5060 (no NAT) Reliably Transmitting (no NAT) to 192.168.3.4:5060: OPTIONS sip:102@192.168.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK548bed21 Max-Forwards: 70 From: "asterisk" ;tag=as755b7afa To: Contact: Call-ID: 6fec403f2c32405f17f7838e3e772ad2@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 14:07:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (no NAT) to 192.168.3.4:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.4:5060;branch=z9hG4bK64a33d006fc8081210bdc6f488177270;received=192.168.3.4;rport=5060 From: "102" ;tag=3007678749 To: "102" ;tag=as30763706 Call-ID: 2999504275@192_168_3_4 CSeq: 967 REGISTER Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 180 Contact: ;expires=180 Date: Tue, 12 Apr 2011 14:07:51 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2999504275@192_168_3_4' in 32000 ms (Method: REGISTER) asterisk*CLI>  <--- SIP read from UDP:192.168.3.4:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK548bed21 From: "asterisk" ;tag=as755b7afa To: ;tag=534853020 Call-ID: 6fec403f2c32405f17f7838e3e772ad2@192.168.3.1:5060 CSeq: 102 OPTIONS Contact: "102" Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO Accept: application/sdp,application/dtmf-relay Accept-Encoding: identity Accept-Language: en Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '6fec403f2c32405f17f7838e3e772ad2@192.168.3.1:5060' Method: OPTIONS asterisk*CLI>  <--- SIP read from UDP:192.168.1.151:5060 ---> <-------------> asterisk*CLI>  <--- SIP read from UDP:192.168.3.2:5060 ---> REGISTER sip:192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.2:5060;rport;branch=z9hG4bKb85647e833 From: "wind8" ;tag=036ebd8d To: "wind8" Call-ID: 04a1c3ee184640a22c3d58216f64f537@192.168.3.2 Contact: CSeq: 15810 REGISTER Max-Forwards: 70 Expires: 60 Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE Authorization: Digest username="wind8",realm="asterisk",nonce="2ba448ed",response="5fae36ebdfbe99cab1c7cb32b7c57c00",uri="sip:192.168.3.1",algorithm=MD5 User-Agent: Mv-37x (904290) Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.3.2:5060 (no NAT) <--- Transmitting (no NAT) to 192.168.3.2:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.3.2:5060;branch=z9hG4bKb85647e833;received=192.168.3.2;rport=5060 From: "wind8" ;tag=036ebd8d To: "wind8" ;tag=as19b13661 Call-ID: 04a1c3ee184640a22c3d58216f64f537@192.168.3.2 CSeq: 15810 REGISTER Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f49b6d5" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '04a1c3ee184640a22c3d58216f64f537@192.168.3.2' in 32000 ms (Method: REGISTER) asterisk*CLI>  <--- SIP read from UDP:192.168.3.2:5060 ---> REGISTER sip:192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.2:5060;rport;branch=z9hG4bKcf71eb0801 From: "wind8" ;tag=036ebd8d To: "wind8" Call-ID: 04a1c3ee184640a22c3d58216f64f537@192.168.3.2 Contact: CSeq: 15811 REGISTER Max-Forwards: 70 Expires: 60 Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE Authorization: Digest username="wind8",realm="asterisk",nonce="2f49b6d5",response="f78dac485d47eddd7843d84fdf1d016a",uri="sip:192.168.3.1",algorithm=MD5 User-Agent: Mv-37x (904290) Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.3.2:5060 (no NAT) asterisk*CLI> Reliably Transmitting (no NAT) to 192.168.3.2:5060: OPTIONS sip:wind8@192.168.3.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK153cba0b Max-Forwards: 70 From: "asterisk" ;tag=as5bc222ef To: Contact: Call-ID: 305b26845b0f9f0d2adefef05a6b1754@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 14:08:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI>  <--- Transmitting (no NAT) to 192.168.3.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.2:5060;branch=z9hG4bKcf71eb0801;received=192.168.3.2;rport=5060 From: "wind8" ;tag=036ebd8d To: "wind8" ;tag=as19b13661 Call-ID: 04a1c3ee184640a22c3d58216f64f537@192.168.3.2 CSeq: 15811 REGISTER Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 12 Apr 2011 14:08:03 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '04a1c3ee184640a22c3d58216f64f537@192.168.3.2' in 32000 ms (Method: REGISTER) asterisk*CLI>  <--- SIP read from UDP:192.168.3.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK153cba0b From: "asterisk" ;tag=as5bc222ef To: ;tag=718b3f95 Call-ID: 305b26845b0f9f0d2adefef05a6b1754@192.168.3.1:5060 CSeq: 102 OPTIONS Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '305b26845b0f9f0d2adefef05a6b1754@192.168.3.1:5060' Method: OPTIONS asterisk*CLI> Reliably Transmitting (no NAT) to 192.168.3.3:2048: OPTIONS sip:104@192.168.3.3:2048;line=k5l8dvj0 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK7002eaf6 Max-Forwards: 70 From: "asterisk" ;tag=as4d4b5710 To: Contact: Call-ID: 56c93df11f5cc93f4735555a4da9ca57@192.168.3.1:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 14:08:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI>  <--- SIP read from UDP:192.168.3.3:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK7002eaf6 From: "asterisk" ;tag=as4d4b5710 To: Call-ID: 56c93df11f5cc93f4735555a4da9ca57@192.168.3.1:5060 CSeq: 102 OPTIONS Contact: ;reg-id=1 User-Agent: snom300/8.4.31 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '56c93df11f5cc93f4735555a4da9ca57@192.168.3.1:5060' Method: OPTIONS asterisk*CLI> Reliably Transmitting (no NAT) to 192.168.1.44:5060: OPTIONS sip:153@192.168.1.44:5060;rinstance=018c81174ecaa090;transport=UDP SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK33c73503 Max-Forwards: 70 From: "asterisk" ;tag=as6e1e09c8 To: Contact: Call-ID: 3c3df1771b1f5d28636dc501268da45d@asterisk-public-ip:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 14:08:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK33c73503 Contact: To: ;tag=00485d12 From: "asterisk";tag=as6e1e09c8 Call-ID: 3c3df1771b1f5d28636dc501268da45d@asterisk-public-ip:5060 CSeq: 102 OPTIONS Accept: application/sdp, application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Zoiper rev.6739 Allow-Events: presence Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '3c3df1771b1f5d28636dc501268da45d@asterisk-public-ip:5060' Method: OPTIONS asterisk*CLI> Reliably Transmitting (no NAT) to 192.168.1.158:5060: OPTIONS sip:zJZlTpTaQyFc4k0voKjL@192.168.1.158 SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK628fd7ef Max-Forwards: 70 From: "asterisk" ;tag=as1f91d7eb To: Contact: Call-ID: 3a0fc95a7b7a0d4445f0bbeb32129c3b@asterisk-public-ip:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 14:08:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI>  <--- SIP read from UDP:192.168.1.158:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK628fd7ef To: ;tag=lk2n7u20pdhc72np7gln From: "asterisk" ;tag=as1f91d7eb Call-ID: 3a0fc95a7b7a0d4445f0bbeb32129c3b@asterisk-public-ip:5060 CSeq: 102 OPTIONS Accept: application/sdp Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '3a0fc95a7b7a0d4445f0bbeb32129c3b@asterisk-public-ip:5060' Method: OPTIONS asterisk*CLI> Reliably Transmitting (no NAT) to 83.211.227.21:5060: OPTIONS sip:voip.eutelia.it SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK118d8d02 Max-Forwards: 70 From: "asterisk" ;tag=as25c69a6a To: Contact: Call-ID: 1dad65c40d985290165a5a295d080585@asterisk-public-ip:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Date: Tue, 12 Apr 2011 14:08:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK118d8d02 From: "asterisk" ;tag=as25c69a6a To: ;tag=c040a69dfc7733bdec8c921a7a9f2d3a.a51a Call-ID: 1dad65c40d985290165a5a295d080585@asterisk-public-ip:5060 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Supported: Server: SPS EUT RM GW 04 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- asterisk*CLI> Really destroying SIP dialog '1dad65c40d985290165a5a295d080585@asterisk-public-ip:5060' Method: OPTIONS asterisk*CLI>  <--- SIP read from UDP:192.168.3.4:5060 ---> <-------------> asterisk*CLI>  <--- SIP read from UDP:192.168.1.151:5060 ---> <-------------> asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> BYE sip:dialed-number@asterisk-public-ip:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-ccc1ed262157acaa-1---d8754z-;rport Max-Forwards: 70 Contact: To: ;tag=as4de0b0df From: "153";tag=e9c4fc35 Call-ID: NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E. CSeq: 4 BYE User-Agent: Zoiper rev.6739 Authorization: Digest username="153",realm="asterisk",nonce="781b612d",uri="sip:dialed-number@asterisk-public-ip:5060",response="0deaa9d5a171bce395d60d360c0251d6",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- asterisk*CLI> Sending to 192.168.1.44:5060 (no NAT) asterisk*CLI> Scheduling destruction of SIP dialog 'NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E.' in 6912 ms (Method: BYE) <--- Transmitting (no NAT) to 192.168.1.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.44:5060;branch=z9hG4bK-d8754z-ccc1ed262157acaa-1---d8754z-;received=192.168.1.44;rport=5060 From: "153";tag=e9c4fc35 To: ;tag=as4de0b0df Call-ID: NmU5ZTdlOTNjZDlmMGUwMzE1OGQzZWRiMjg4YWZjY2E. CSeq: 4 BYE Server: Asterisk PBX 1.8.3-1digium1~squeeze Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> asterisk*CLI> Scheduling destruction of SIP dialog '004c464f69d022e202c355f72d91976f@asterisk-public-ip:5060' in 6400 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 83.211.227.21:5060 Reliably Transmitting (no NAT) to 83.211.227.21:5060: BYE sip:494dialed-number@83.211.2.218:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK70c778d8 Route: Max-Forwards: 70 From: "153" ;tag=as5159f324 To: ;tag=14461840-D8F Call-ID: 004c464f69d022e202c355f72d91976f@asterisk-public-ip:5060 CSeq: 105 BYE User-Agent: Asterisk PBX 1.8.3-1digium1~squeeze Proxy-Authorization: Digest username="my-eutelia-number", realm="voip.eutelia.it", algorithm=MD5, uri="sip:494dialed-number@83.211.2.218:5060", nonce="4da45ce33ccae1bf338ca7a3fed129344ac517fd", response="d0fe1c2ac8217b9e3390db79014c8e99", qop=auth, cnonce="06138ffa", nc=00000002 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (phones-sip, dialed-number, 1) exited non-zero on 'SIP/153-00000036' asterisk*CLI>  <--- SIP read from UDP:83.211.227.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk-public-ip:5060;branch=z9hG4bK70c778d8 From: "153" ;tag=as5159f324 To: ;tag=14461840-D8F Date: Tue, 12 Apr 2011 14:08:16 GMT Call-ID: 004c464f69d022e202c355f72d91976f@asterisk-public-ip:5060 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 105 BYE <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '004c464f69d022e202c355f72d91976f@asterisk-public-ip:5060' Method: INVITE asterisk*CLI>  <--- SIP read from UDP:192.168.1.44:5060 ---> <-------------> asterisk*CLI> Disconnected from Asterisk server