pbx*CLI> <--- SIP read from UDP://XX.XX.XX.XX:58370 ---> INVITE sip:0018005551212@pbx.mycompany.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:58370;branch=z9hG4bK-d8754z-28200568a64aff46-1---d8754z-;rport Max-Forwards: 70 Contact: To: "0018005551212" From: "Wifi";tag=385de215 Call-ID: NmUyMDA5MWViYTk5MGI4NTViODU1NDQ1YjNhYjJjN2M. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO ontent-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 280 v=0 o=- 7 2 IN IP4 192.168.1.10 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.1.10 t=0 0 m=audio 43632 RTP/AVP 18 101 a=alt:1 1 : nZG5/Fdq uhdwCduk 192.168.1.10 43632 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (12 headers 12 lines) --- == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL CoS mark 5 Sending to XX.XX.XX.XX : 58370 (NAT) Using INVITE request as basis request - NmUyMDA5MWViYTk5MGI4NTViODU1NDQ1YjNhYjJjN2M. Found peer 'Wifi' for 'Wifi' from XX.XX.XX.XX:58370 pbx*CLI> <--- Reliably Transmitting (NAT) to XX.XX.XX.XX:58370 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.10:58370;branch=z9hG4bK-d8754z-28200568a64aff46-1---d8754z-;received=XX.XX.XX.XX;rport=58370 From: "Wifi";tag=385de215 To: "0018005551212";tag=as25319949 Call-ID: NmUyMDA5MWViYTk5MGI4NTViODU1NDQ1YjNhYjJjN2M. CSeq: 1 INVITE Server: MyPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2216bc19" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'NmUyMDA5MWViYTk5MGI4NTViODU1NDQ1YjNhYjJjN2M.' in 32000 ms (Method: INVITE) pbx*CLI> <--- SIP read from UDP://XX.XX.XX.XX:58370 ---> ACK sip:0018005551212@pbx.mycompany.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:58370;branch=z9hG4bK-d8754z-28200568a64aff46-1---d8754z-;rport To: "0018005551212";tag=as25319949 From: "Wifi";tag=385de215 Call-ID: NmUyMDA5MWViYTk5MGI4NTViODU1NDQ1YjNhYjJjN2M. CSeq: 1 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- pbx*CLI> <--- SIP read from UDP://XX.XX.XX.XX:58370 ---> INVITE sip:0018005551212@pbx.mycompany.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:58370;branch=z9hG4bK-d8754z-f81b8020634c0771-1---d8754z-;rport Max-Forwards: 70 Contact: To: "0018005551212" From: "Wifi";tag=385de215 Call-ID: NmUyMDA5MWViYTk5MGI4NTViODU1NDQ1YjNhYjJjN2M. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO ontent-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Authorization: Digest username="Wifi",realm="asterisk",nonce="2216bc19",uri="sip:0018005551212@pbx.mycompany.com",response="1912db3038124484c5869efe89009a95",algorithm=MD5 Content-Length: 280 v=0 o=- 7 2 IN IP4 192.168.1.10 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.1.10 t=0 0 m=audio 43632 RTP/AVP 18 101 a=alt:1 1 : nZG5/Fdq uhdwCduk 192.168.1.10 43632 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (13 headers 12 lines) --- Sending to XX.XX.XX.XX : 58370 (NAT) Using INVITE request as basis request - NmUyMDA5MWViYTk5MGI4NTViODU1NDQ1YjNhYjJjN2M. Found peer 'Wifi' for 'Wifi' from XX.XX.XX.XX:58370 Found RTP audio format 18 Found RTP audio format 101 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.1.10:43632 Looking for 0018005551212 in Dovid (domain pbx.mycompany.com) list_route: hop: <--- Transmitting (NAT) to XX.XX.XX.XX:58370 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:58370;branch=z9hG4bK-d8754z-f81b8020634c0771-1---d8754z-;received=XX.XX.XX.XX;rport=58370 From: "Wifi";tag=385de215 To: "0018005551212" Call-ID: NmUyMDA5MWViYTk5MGI4NTViODU1NDQ1YjNhYjJjN2M. CSeq: 2 INVITE Server: MyPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [0018005551212@Dovid:1] Set("SIP/Wifi-00000007", "CALLERID(name)=") in new stack -- Executing [0018005551212@Dovid:2] Set("SIP/Wifi-00000007", "CALLERPRES()=prohib_passed_screen") in new stack -- Executing [0018005551212@Dovid:3] Dial("SIP/Wifi-00000007", "SIP/18005551212@some_peer") in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL CoS mark 5 Audio is at XX.XX.XX.XX.XX port 19870 Adding codec 0x100 (g729) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 69.1.229.107:5060: INVITE sip:18005551212@69.1.229.107 SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX.XX:5060;branch=z9hG4bK5871f790;rport Max-Forwards: 70 From: "Anonymous" ;tag=as74b4f474 To: Contact: Call-ID: 19c930af1bd49d5234c2ea000ae7eb98@XX.XX.XX.XX.XX CSeq: 102 INVITE User-Agent: MyPBX Date: Wed, 14 Apr 2010 09:21:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 313 v=0 o=root 1898270944 1898270944 IN IP4 XX.XX.XX.XX.XX s=Asterisk PBX 1.6.1.18 c=IN IP4 XX.XX.XX.XX.XX t=0 0 m=audio 19870 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 18005551212@some_peer pbx*CLI> <--- SIP read from UDP://69.1.229.107:5060 ---> SIP/2.0 302 Redirect Via: SIP/2.0/UDP XX.XX.XX.XX.XX:5060;rport;branch=z9hG4bK5871f790 From: "Anonymous" ;tag=as74b4f474 To: Call-ID: 19c930af1bd49d5234c2ea000ae7eb98@XX.XX.XX.XX.XX CSeq: 102 INVITE Contact: Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- Got SIP response 302 "Redirect" back from 69.1.229.107 Transmitting (no NAT) to 69.1.229.107:5060: ACK sip:18005551212@69.1.229.107 SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX.XX:5060;branch=z9hG4bK5871f790;rport Max-Forwards: 70 From: "Anonymous" ;tag=as74b4f474 To: Contact: Call-ID: 19c930af1bd49d5234c2ea000ae7eb98@XX.XX.XX.XX.XX CSeq: 102 ACK User-Agent: MyPBX Content-Length: 0 --- -- Now forwarding SIP/Wifi-00000007 to 'SIP/18005551212::::UDP@XX.XX.XX.XX:11060' (thanks to SIP/some_peer-00000008) == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL CoS mark 5 Audio is at XX.XX.XX.XX.XX port 15818 Adding codec 0x100 (g729) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (NAT) to XX.XX.XX.XX:11060: INVITE sip:18005551212@XX.XX.XX.XX:11060 SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX.XX:5060;branch=z9hG4bK0c663739;rport Max-Forwards: 70 From: "7182223333" ;tag=as2b4246ec To: Contact: Call-ID: 4bfbc67e3aa249705767ad8355e24f8e@XX.XX.XX.XX.XX CSeq: 102 INVITE User-Agent: MyPBX Date: Wed, 14 Apr 2010 09:21:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 281 v=0 o=root 1854362014 1854362014 IN IP4 XX.XX.XX.XX.XX s=Asterisk PBX 1.6.1.18 c=IN IP4 XX.XX.XX.XX.XX t=0 0 m=audio 15818 RTP/AVP 18 0 8 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx*CLI> <--- SIP read from UDP://XX.XX.XX.XX:11060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP XX.XX.XX.XX.XX:5060;rport=5060;received=XX.XX.XX.XX.XX;branch=z9hG4bK0c663739 Call-ID: 4bfbc67e3aa249705767ad8355e24f8e@XX.XX.XX.XX.XX From: 7182223333 ;tag=as2b4246ec To: CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '19c930af1bd49d5234c2ea000ae7eb98@XX.XX.XX.XX.XX' Method: INVITE pbx*CLI> <--- SIP read from UDP://XX.XX.XX.XX:11060 ---> SIP/2.0 183 Session Progress Contact: To: ;tag=MDk6MjE6NTUuMDA From: 7182223333 ;tag=as2b4246ec Call-ID: 4bfbc67e3aa249705767ad8355e24f8e@XX.XX.XX.XX.XX Record-Route: Via: SIP/2.0/UDP XX.XX.XX.XX.XX;rport=5060;received=XX.XX.XX.XX.XX;branch=z9hG4bK0c663739 CSeq: 102 INVITE Content-Disposition: session; handling=required Content-Type: application/sdp Content-Length: 169 v=0 o=- 15543 14756 IN IP4 69.1.229.40 s=- c=IN IP4 69.1.229.40 t=0 0 m=audio 51220 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=sendrecv a=ptime:20 <-------------> --- (11 headers 10 lines) --- Found RTP audio format 18 Found audio description format G729 for ID 18 Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 69.1.229.40:51220 Peer doesn't provide video -- SIP/XX.XX.XX.XX:11060-00000009 is making progress passing it to SIP/Wifi-00000007 Audio is at XX.XX.XX.XX.XX port 12720 Adding codec 0x100 (g729) to SDP pbx*CLI> <--- Transmitting (NAT) to XX.XX.XX.XX:58370 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.10:58370;branch=z9hG4bK-d8754z-f81b8020634c0771-1---d8754z-;received=XX.XX.XX.XX;rport=58370 From: "Wifi";tag=385de215 To: "0018005551212";tag=as711faa0b Call-ID: NmUyMDA5MWViYTk5MGI4NTViODU1NDQ1YjNhYjJjN2M. CSeq: 2 INVITE Server: MyPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 233 v=0 o=root 1049280487 1049280487 IN IP4 XX.XX.XX.XX.XX s=Asterisk PBX 1.6.1.18 c=IN IP4 XX.XX.XX.XX.XX t=0 0 m=audio 12720 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> pbx*CLI> <--- SIP read from UDP://XX.XX.XX.XX:11060 ---> SIP/2.0 200 OK Contact: To: ;tag=MDk6MjE6NTUuMDA From: 7182223333 ;tag=as2b4246ec Call-ID: 4bfbc67e3aa249705767ad8355e24f8e@XX.XX.XX.XX.XX Record-Route: Via: SIP/2.0/UDP XX.XX.XX.XX.XX;rport=5060;received=XX.XX.XX.XX.XX;branch=z9hG4bK0c663739 CSeq: 102 INVITE Session-Expires: 10800;refresher=uas Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Content-Disposition: session; handling=required Supported: timer Content-Type: application/sdp Content-Length: 169 v=0 o=- 15543 14756 IN IP4 69.1.229.40 s=- c=IN IP4 69.1.229.40 t=0 0 m=audio 51220 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=sendrecv a=ptime:20 <-------------> --- (14 headers 10 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 11060 Transmitting (NAT) to XX.XX.XX.XX:11060: ACK sip:18005551212@69.1.229.137:11080 SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX.XX:5060;branch=z9hG4bK0d2cb6fd;rport Route: Max-Forwards: 70 From: "7182223333" ;tag=as2b4246ec To: ;tag=MDk6MjE6NTUuMDA Contact: Call-ID: 4bfbc67e3aa249705767ad8355e24f8e@XX.XX.XX.XX.XX CSeq: 102 ACK User-Agent: MyPBX Content-Length: 0 --- -- SIP/XX.XX.XX.XX:11060-00000009 answered SIP/Wifi-00000007 Audio is at XX.XX.XX.XX.XX port 12720 Adding codec 0x100 (g729) to SDP <--- Reliably Transmitting (NAT) to XX.XX.XX.XX:58370 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.10:58370;branch=z9hG4bK-d8754z-f81b8020634c0771-1---d8754z-;received=XX.XX.XX.XX;rport=58370 From: "Wifi";tag=385de215 To: "0018005551212";tag=as711faa0b Call-ID: NmUyMDA5MWViYTk5MGI4NTViODU1NDQ1YjNhYjJjN2M. CSeq: 2 INVITE Server: MyPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 233 v=0 o=root 1049280487 1049280488 IN IP4 XX.XX.XX.XX.XX s=Asterisk PBX 1.6.1.18 c=IN IP4 XX.XX.XX.XX.XX t=0 0 m=audio 12720 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Native bridging SIP/Wifi-00000007 and SIP/XX.XX.XX.XX:11060-00000009 set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 11060 Audio is at XX.XX.XX.XX.XX port 15818 Adding codec 0x100 (g729) to SDP Reliably Transmitting (NAT) to XX.XX.XX.XX:11060: INVITE sip:18005551212@69.1.229.137:11080 SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX.XX:5060;branch=z9hG4bK5cdbaf03;rport Route: Max-Forwards: 70 From: "7182223333" ;tag=as2b4246ec To: ;tag=MDk6MjE6NTUuMDA Contact: Call-ID: 4bfbc67e3aa249705767ad8355e24f8e@XX.XX.XX.XX.XX CSeq: 103 INVITE User-Agent: MyPBX Require: timer Session-Expires: 10800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 231 v=0 o=root 1854362014 1854362015 IN IP4 192.168.1.10 s=Asterisk PBX 1.6.1.18 c=IN IP4 192.168.1.10 t=0 0 m=audio 43632 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx*CLI> <--- SIP read from UDP://XX.XX.XX.XX:11060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP XX.XX.XX.XX.XX:5060;rport=5060;received=XX.XX.XX.XX.XX;branch=z9hG4bK5cdbaf03 Call-ID: 4bfbc67e3aa249705767ad8355e24f8e@XX.XX.XX.XX.XX From: 7182223333 ;tag=as2b4246ec To: ;tag=MDk6MjE6NTUuMDA CSeq: 103 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- pbx*CLI> <--- SIP read from UDP://XX.XX.XX.XX:11060 ---> SIP/2.0 200 OK Contact: To: ;tag=MDk6MjE6NTUuMDA From: 7182223333 ;tag=as2b4246ec Call-ID: 4bfbc67e3aa249705767ad8355e24f8e@XX.XX.XX.XX.XX Record-Route: Via: SIP/2.0/UDP XX.XX.XX.XX.XX;rport=5060;received=XX.XX.XX.XX.XX;branch=z9hG4bK5cdbaf03 CSeq: 103 INVITE Session-Expires: 10800;refresher=uas Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Content-Disposition: session; handling=required Supported: timer Content-Type: application/sdp Content-Length: 198 v=0 o=Sonus_UAC 15543 14757 IN IP4 69.1.229.40 s=SIP Media Capabilities c=IN IP4 69.1.229.40 t=0 0 m=audio 51220 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=sendrecv a=ptime:20 <-------------> --- (14 headers 10 lines) --- Found RTP audio format 18 Found audio description format G729 for ID 18 Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 69.1.229.40:51220 Peer doesn't provide video set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 11060 Transmitting (NAT) to XX.XX.XX.XX:11060: ACK sip:18005551212@69.1.229.137:11080 SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX.XX:5060;branch=z9hG4bK5b0ce949;rport Route: Max-Forwards: 70 From: "7182223333" ;tag=as2b4246ec To: ;tag=MDk6MjE6NTUuMDA Contact: Call-ID: 4bfbc67e3aa249705767ad8355e24f8e@XX.XX.XX.XX.XX CSeq: 103 ACK User-Agent: MyPBX Content-Length: 0 --x*CLI> pbx*CLI> <--- SIP read from UDP://XX.XX.XX.XX:58370 ---> ACK sip:0018005551212@XX.XX.XX.XX.XX SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:58370;branch=z9hG4bK-d8754z-714aeb7b9d0f8d76-1---d8754z-;rport Max-Forwards: 70 Contact: To: "0018005551212";tag=as711faa0b From: "Wifi";tag=385de215 Call-ID: NmUyMDA5MWViYTk5MGI4NTViODU1NDQ1YjNhYjJjN2M. CSeq: 2 ACK User-Agent: eyeBeam release 1102q stamp 51814 Authorization: Digest username="Wifi",realm="asterisk",nonce="2216bc19",uri="sip:0018005551212@pbx.mycompany.com",response="1912db3038124484c5869efe89009a95",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 58370 Audio is at XX.XX.XX.XX.XX port 12720 Adding codec 0x100 (g729) to SDP Reliably Transmitting (NAT) to XX.XX.XX.XX:58370: INVITE sip:Wifi@XX.XX.XX.XX:58370 SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX.XX:5060;branch=z9hG4bK429a5f06;rport Max-Forwards: 70 From: "0018005551212";tag=as711faa0b To: "Wifi";tag=385de215 Contact: Call-ID: NmUyMDA5MWViYTk5MGI4NTViODU1NDQ1YjNhYjJjN2M. CSeq: 102 INVITE User-Agent: MyPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 229 v=0 o=root 1049280487 1049280489 IN IP4 69.1.229.40 s=Asterisk PBX 1.6.1.18 c=IN IP4 69.1.229.40 t=0 0 m=audio 51220 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx*CLI> <--- SIP read from UDP://XX.XX.XX.XX:58370 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP XX.XX.XX.XX.XX:5060;branch=z9hG4bK429a5f06;rport=5060 Contact: To: "Wifi";tag=385de215 From: "0018005551212";tag=as711faa0b Call-ID: NmUyMDA5MWViYTk5MGI4NTViODU1NDQ1YjNhYjJjN2M. CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 174 v=0 o=- 7 3 IN IP4 192.168.1.10 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.1.10 t=0 0 m=audio 43632 RTP/AVP 18 a=fmtp:18 annexb=yes a=rtpmap:18 G729/8000 a=sendrecv <-------------> --- (11 headers 9 lines) --- Found RTP audio format 18 Found audio description format G729 for ID 18 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.1.10:43632 set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 58370 Transmitting (NAT) to XX.XX.XX.XX:58370: ACK sip:Wifi@XX.XX.XX.XX:58370 SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX.XX:5060;branch=z9hG4bK24f76a5d;rport Max-Forwards: 70 From: "0018005551212";tag=as711faa0b To: "Wifi";tag=385de215 Contact: Call-ID: NmUyMDA5MWViYTk5MGI4NTViODU1NDQ1YjNhYjJjN2M. CSeq: 102 ACK User-Agent: MyPBX Content-Length: 0 --- pbx*CLI> <--- SIP read from UDP://XX.XX.XX.XX:58370 ---> <-------------> pbx*CLI> <--- SIP read from UDP://XX.XX.XX.XX:43694 ---> <-------------> pbx*CLI> <--- SIP read from UDP://XX.XX.XX.XX:58370 ---> BYE sip:0018005551212@XX.XX.XX.XX.XX SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:58370;branch=z9hG4bK-d8754z-490a2c3acf157634-1---d8754z-;rport Max-Forwards: 70 Contact: To: "0018005551212";tag=as711faa0b From: "Wifi";tag=385de215 Call-ID: NmUyMDA5MWViYTk5MGI4NTViODU1NDQ1YjNhYjJjN2M. CSeq: 3 BYE User-Agent: eyeBeam release 1102q stamp 51814 Authorization: Digest username="Wifi",realm="asterisk",nonce="2216bc19",uri="sip:0018005551212@XX.XX.XX.XX.XX",response="f94eebf57573e1de4785758597b85035",algorithm=MD5 Reason: SIP;description="User Hung Up" Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to XX.XX.XX.XX : 58370 (NAT) <--- Transmitting (NAT) to XX.XX.XX.XX:58370 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.10:58370;branch=z9hG4bK-d8754z-490a2c3acf157634-1---d8754z-;received=XX.XX.XX.XX;rport=58370 From: "Wifi";tag=385de215 To: "0018005551212";tag=as711faa0b Call-ID: NmUyMDA5MWViYTk5MGI4NTViODU1NDQ1YjNhYjJjN2M. CSeq: 3 BYE Server: MyPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 11060 Audio is at XX.XX.XX.XX.XX port 15818 Adding codec 0x100 (g729) to SDP Reliably Transmitting (NAT) to XX.XX.XX.XX:11060: INVITE sip:18005551212@69.1.229.137:11080 SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX.XX:5060;branch=z9hG4bK70a1fb91;rport Route: Max-Forwards: 70 From: "7182223333" ;tag=as2b4246ec To: ;tag=MDk6MjE6NTUuMDA Contact: Call-ID: 4bfbc67e3aa249705767ad8355e24f8e@XX.XX.XX.XX.XX CSeq: 104 INVITE User-Agent: MyPBX Require: timer Session-Expires: 10800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 233 v=0 o=root 1854362014 1854362016 IN IP4 XX.XX.XX.XX.XX s=Asterisk PBX 1.6.1.18 c=IN IP4 XX.XX.XX.XX.XX t=0 0 m=audio 15818 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Executing [h@Dovid:1] NoOp("SIP/Wifi-00000007", "") in new stack Scheduling destruction of SIP dialog '4bfbc67e3aa249705767ad8355e24f8e@XX.XX.XX.XX.XX' in 32000 ms (Method: INVITE) == Spawn extension (Dovid, 0018005551212, 3) exited non-zero on 'SIP/Wifi-00000007' pbx*CLI> <--- SIP read from UDP://XX.XX.XX.XX:11060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP XX.XX.XX.XX.XX:5060;rport=5060;received=XX.XX.XX.XX.XX;branch=z9hG4bK70a1fb91 Call-ID: 4bfbc67e3aa249705767ad8355e24f8e@XX.XX.XX.XX.XX From: 7182223333 ;tag=as2b4246ec To: ;tag=MDk6MjE6NTUuMDA CSeq: 104 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog 'NmUyMDA5MWViYTk5MGI4NTViODU1NDQ1YjNhYjJjN2M.' Method: BYE pbx*CLI> <--- SIP read from UDP://XX.XX.XX.XX:11060 ---> SIP/2.0 200 OK Contact: To: ;tag=MDk6MjE6NTUuMDA From: 7182223333 ;tag=as2b4246ec Call-ID: 4bfbc67e3aa249705767ad8355e24f8e@XX.XX.XX.XX.XX Record-Route: Via: SIP/2.0/UDP XX.XX.XX.XX.XX;rport=5060;received=XX.XX.XX.XX.XX;branch=z9hG4bK70a1fb91 CSeq: 104 INVITE Session-Expires: 10800;refresher=uas Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Content-Disposition: session; handling=required Supported: timer Content-Type: application/sdp Content-Length: 198 v=0 o=Sonus_UAC 15543 14758 IN IP4 69.1.229.40 s=SIP Media Capabilities c=IN IP4 69.1.229.40 t=0 0 m=audio 51220 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=sendrecv a=ptime:20 <-------------> --- (14 headers 10 lines) --- Found RTP audio format 18 Found audio description format G729 for ID 18 Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 69.1.229.40:51220 Peer doesn't provide video set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 11060 Transmitting (NAT) to XX.XX.XX.XX:11060: ACK sip:18005551212@69.1.229.137:11080 SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX.XX:5060;branch=z9hG4bK47c8e42f;rport Route: Max-Forwards: 70 From: "7182223333" ;tag=as2b4246ec To: ;tag=MDk6MjE6NTUuMDA Contact: Call-ID: 4bfbc67e3aa249705767ad8355e24f8e@XX.XX.XX.XX.XX CSeq: 104 ACK User-Agent: MyPBX Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 11060 Reliably Transmitting (NAT) to XX.XX.XX.XX:11060: BYE sip:18005551212@69.1.229.137:11080 SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX.XX:5060;branch=z9hG4bK491d2745;rport Route: Max-Forwards: 70 From: "7182223333" ;tag=as2b4246ec To: ;tag=MDk6MjE6NTUuMDA Call-ID: 4bfbc67e3aa249705767ad8355e24f8e@XX.XX.XX.XX.XX CSeq: 105 BYE User-Agent: MyPBX X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Scheduling destruction of SIP dialog '4bfbc67e3aa249705767ad8355e24f8e@XX.XX.XX.XX.XX' in 32000 ms (Method: INVITE) pbx*CLI> <--- SIP read from UDP://XX.XX.XX.XX:11060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP XX.XX.XX.XX.XX;rport=5060;received=XX.XX.XX.XX.XX;branch=z9hG4bK491d2745 From: 7182223333 ;tag=as2b4246ec To: ;tag=MDk6MjE6NTUuMDA CSeq: 105 BYE Call-ID: 4bfbc67e3aa249705767ad8355e24f8e@XX.XX.XX.XX.XX Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '4bfbc67e3aa249705767ad8355e24f8e@XX.XX.XX.XX.XX' Method: INVITE pbx*CLI> <--- SIP read from UDP://XX.XX.XX.XX:43694 ---> SUBSCRIBE sip:scanner@pbx.mycompany.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:43694;branch=z9hG4bK-d8754z-e971f53b1558a562-1---d8754z-;rport Max-Forwards: 70 Contact: To: "scanner" From: "scanner";tag=04274949 Call-ID: MmU0ODFlZjdkZGE3MjE2YzhmMmQ1ODg4ZDlhZDY2ZmY. CSeq: 1 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: eyeBeam release 1102q stamp 51814 Event: message-summary Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Creating new subscription Sending to XX.XX.XX.XX : 43694 (NAT) list_route: hop: Found peer 'scanner' for 'scanner' from XX.XX.XX.XX:43694 <--- Transmitting (NAT) to XX.XX.XX.XX:43694 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.10:43694;branch=z9hG4bK-d8754z-e971f53b1558a562-1---d8754z-;received=XX.XX.XX.XX;rport=43694 From: "scanner";tag=04274949 To: "scanner";tag=as10231b00 Call-ID: MmU0ODFlZjdkZGE3MjE2YzhmMmQ1ODg4ZDlhZDY2ZmY. CSeq: 1 SUBSCRIBE Server: MyPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56cc9a24" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'MmU0ODFlZjdkZGE3MjE2YzhmMmQ1ODg4ZDlhZDY2ZmY.' in 32000 ms (Method: SUBSCRIBE) pbx*CLI> <--- SIP read from UDP://XX.XX.XX.XX:43694 ---> SUBSCRIBE sip:scanner@pbx.mycompany.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:43694;branch=z9hG4bK-d8754z-bd60c92aaa55b93b-1---d8754z-;rport Max-Forwards: 70 Contact: To: "scanner" From: "scanner";tag=04274949 Call-ID: MmU0ODFlZjdkZGE3MjE2YzhmMmQ1ODg4ZDlhZDY2ZmY. CSeq: 2 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: eyeBeam release 1102q stamp 51814 Authorization: Digest username="scanner",realm="asterisk",nonce="56cc9a24",uri="sip:scanner@pbx.mycompany.com",response="23b82e3fb1ec4dfa6fdfbacabe043f24",algorithm=MD5 Event: message-summary Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Creating new subscription Sending to XX.XX.XX.XX : 43694 (NAT) Found peer 'scanner' for 'scanner' from XX.XX.XX.XX:43694 <--- Transmitting (NAT) to XX.XX.XX.XX:43694 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 192.168.1.10:43694;branch=z9hG4bK-d8754z-bd60c92aaa55b93b-1---d8754z-;received=XX.XX.XX.XX;rport=43694 From: "scanner";tag=04274949 To: "scanner";tag=as10231b00 Call-ID: MmU0ODFlZjdkZGE3MjE2YzhmMmQ1ODg4ZDlhZDY2ZmY. CSeq: 2 SUBSCRIBE Server: MyPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog 'MmU0ODFlZjdkZGE3MjE2YzhmMmQ1ODg4ZDlhZDY2ZmY.' Method: SUBSCRIBE