>>replaced-name*CLI> <-- SIP read from X.X.X.2:5060: INVITE sip:5555550290@X.X.X.19 SIP/2.0 Via: SIP/2.0/UDP X.X.X.2:5060;branch=z9hG4bK-c90530fd From: ;tag=bf094c222ae884b5o0 To: Call-ID: 5fb6ee66-49dba741@X.X.X.2 CSeq: 101 INVITE Max-Forwards: 70 Contact: Expires: 240 User-Agent: Linksys/RT31P2-2.0.10(LIc) Content-Length: 420 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 82968560 82968560 IN IP4 X.X.X.2 s=- c=IN IP4 X.X.X.2 t=0 0 m=audio 16474 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv >>replaced-name*CLI> --- (14 headers 19 lines)--- >>replaced-name*CLI> Using INVITE request as basis request - 5fb6ee66-49dba741@X.X.X.2 >>replaced-name*CLI> Sending to X.X.X.2 : 5060 (non-NAT) >>replaced-name*CLI> Reliably Transmitting (no NAT) to X.X.X.2:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP X.X.X.2:5060;branch=z9hG4bK-c90530fd From: ;tag=bf094c222ae884b5o0 To: ;tag=as0c85a37d Call-ID: 5fb6ee66-49dba741@X.X.X.2 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="62e57e84" Content-Length: 0 --- >>replaced-name*CLI> Scheduling destruction of call '5fb6ee66-49dba741@X.X.X.2' in 15000 ms >>replaced-name*CLI> Found user '5555557444' >>replaced-name*CLI> <-- SIP read from X.X.X.2:5060: ACK sip:5555550290@X.X.X.19 SIP/2.0 Via: SIP/2.0/UDP X.X.X.2:5060;branch=z9hG4bK-c90530fd From: ;tag=bf094c222ae884b5o0 To: ;tag=as0c85a37d Call-ID: 5fb6ee66-49dba741@X.X.X.2 CSeq: 101 ACK Max-Forwards: 70 Contact: User-Agent: Linksys/RT31P2-2.0.10(LIc) Content-Length: 0 >>replaced-name*CLI> --- (10 headers 0 lines)--- >>replaced-name*CLI> <-- SIP read from X.X.X.2:5060: INVITE sip:5555550290@X.X.X.19 SIP/2.0 Via: SIP/2.0/UDP X.X.X.2:5060;branch=z9hG4bK-f3ba2159 From: ;tag=bf094c222ae884b5o0 To: Call-ID: 5fb6ee66-49dba741@X.X.X.2 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="5555557444",realm="asterisk",nonce="62e57e84",uri="sip:5555550290@X.X.X.19",algorithm=MD5,response="3a9b5dfadfda3547d25c5c655e0eb0d1" Contact: Expires: 240 User-Agent: Linksys/RT31P2-2.0.10(LIc) Content-Length: 420 Content-Type: application/sdp v=0 o=- 82968560 82968560 IN IP4 X.X.X.2 s=- c=IN IP4 X.X.X.2 t=0 0 m=audio 16474 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv >>replaced-name*CLI> --- (13 headers 19 lines)--- >>replaced-name*CLI> Using INVITE request as basis request - 5fb6ee66-49dba741@X.X.X.2 >>replaced-name*CLI> Sending to X.X.X.2 : 5060 (non-NAT) >>replaced-name*CLI> Found user '5555557444' >>replaced-name*CLI> Found RTP audio format 0 >>replaced-name*CLI> Found RTP audio format 2 >>replaced-name*CLI> Found RTP audio format 4 >>replaced-name*CLI> Found RTP audio format 8 >>replaced-name*CLI> Found RTP audio format 18 >>replaced-name*CLI> Found RTP audio format 96 >>replaced-name*CLI> Found RTP audio format 97 >>replaced-name*CLI> Found RTP audio format 98 >>replaced-name*CLI> Found RTP audio format 100 >>replaced-name*CLI> Found RTP audio format 101 >>replaced-name*CLI> Peer audio RTP is at port X.X.X.2:16474 >>replaced-name*CLI> Found description format PCMU >>replaced-name*CLI> Found description format G726-32 >>replaced-name*CLI> Found description format G723 >>replaced-name*CLI> Found description format PCMA >>replaced-name*CLI> Found description format G729a >>replaced-name*CLI> Found description format G726-40 >>replaced-name*CLI> Found description format G726-24 >>replaced-name*CLI> Found description format G726-16 >>replaced-name*CLI> Found description format NSE >>replaced-name*CLI> Found description format telephone-event >>replaced-name*CLI> Capabilities: us - 0x4 (ulaw), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw) >>replaced-name*CLI> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) >>replaced-name*CLI> Looking for 5555550290 in subscriber >>replaced-name*CLI> list_route: hop: >>replaced-name*CLI> Transmitting (no NAT) to X.X.X.2:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.2:5060;branch=z9hG4bK-f3ba2159 From: ;tag=bf094c222ae884b5o0 To: Call-ID: 5fb6ee66-49dba741@X.X.X.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- >>replaced-name*CLI> -- Executing [1;36;40mMacro[0;37;40m("[1;35;40mSIP/5555557444-faaf[0;37;40m", "[1;35;40mdial-local|1[0;37;40m") in new stack >>replaced-name*CLI> -- Executing [1;36;40mGotoIf[0;37;40m("[1;35;40mSIP/5555557444-faaf[0;37;40m", "[1;35;40m0?2:3[0;37;40m") in new stack >>replaced-name*CLI> -- Goto (macro-dial-local,s,3) >>replaced-name*CLI> -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/5555557444-faaf[0;37;40m", "[1;35;40mSIP/5555550290@xo|60[0;37;40m") in new stack >>replaced-name*CLI> We're at X.X.X.19 port 12566 >>replaced-name*CLI> Answering/Requesting with root capability 0x4 (ulaw) >>replaced-name*CLI> 12 headers, 8 lines >>replaced-name*CLI> Reliably Transmitting (no NAT) to X.X.X.188:5060: INVITE sip:5555550290@X.X.X.188 SIP/2.0 Via: SIP/2.0/UDP X.X.X.19:5060;branch=z9hG4bK4d7c55d2 From: "5555557444" ;tag=as6778cefb To: Contact: Call-ID: 769d78d77883e0cb39c8286b25e07799@X.X.X.19 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 06 Sep 2005 16:07:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 154 v=0 o=root 16881 16881 IN IP4 X.X.X.19 s=session c=IN IP4 X.X.X.19 t=0 0 m=audio 12566 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - --- >>replaced-name*CLI> -- Called 5555550290@xo >>replaced-name*CLI> <-- SIP read from X.X.X.188:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.19:5060;branch=z9hG4bK4d7c55d2 From: "5555557444" ;tag=as6778cefb To: Call-ID: 769d78d77883e0cb39c8286b25e07799@X.X.X.19 CSeq: 102 INVITE >>replaced-name*CLI> --- (6 headers 0 lines)--- >>replaced-name*CLI> <-- SIP read from X.X.X.188:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP X.X.X.19:5060;branch=z9hG4bK4d7c55d2 From: "5555557444" ;tag=as6778cefb To: ;tag=SD785e199-09a2ce01 Call-ID: 769d78d77883e0cb39c8286b25e07799@X.X.X.19 CSeq: 102 INVITE Content-Length: 170 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 8824 20772 IN IP4 X.X.X.188 s=SIP Media Capabilities c=IN IP4 X.X.X.188 t=0 0 m=audio 25098 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv >>replaced-name*CLI> --- (9 headers 8 lines)--- >>replaced-name*CLI> Found RTP audio format 0 >>replaced-name*CLI> Peer audio RTP is at port X.X.X.188:25098 >>replaced-name*CLI> Found description format PCMU >>replaced-name*CLI> Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) >>replaced-name*CLI> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) -- SIP/xo-bf71 is making progress passing it to SIP/5555557444-faaf We're at X.X.X.19 port 17286 Answering with preferred capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) Transmitting (no NAT) to X.X.X.2:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP X.X.X.2:5060;branch=z9hG4bK-f3ba2159 From: ;tag=bf094c222ae884b5o0 To: ;tag=as469c1f45 Call-ID: 5fb6ee66-49dba741@X.X.X.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 210 v=0 o=root 16881 16881 IN IP4 X.X.X.19 s=session c=IN IP4 X.X.X.19 t=0 0 m=audio 17286 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- >>replaced-name*CLI> <-- SIP read from X.X.X.188:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP X.X.X.19:5060;branch=z9hG4bK4d7c55d2 From: "5555557444" ;tag=as6778cefb To: ;tag=SD785e199-09a2ce01 Call-ID: 769d78d77883e0cb39c8286b25e07799@X.X.X.19 CSeq: 102 INVITE Content-Length: 170 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 8824 20772 IN IP4 X.X.X.188 s=SIP Media Capabilities c=IN IP4 X.X.X.188 t=0 0 m=audio 25098 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv >>replaced-name*CLI> --- (9 headers 8 lines)--- >>replaced-name*CLI> -- SIP/xo-bf71 is ringing >>replaced-name*CLI> Transmitting (no NAT) to X.X.X.2:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP X.X.X.2:5060;branch=z9hG4bK-f3ba2159 From: ;tag=bf094c222ae884b5o0 To: ;tag=as469c1f45 Call-ID: 5fb6ee66-49dba741@X.X.X.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- >>replaced-name*CLI> <-- SIP read from X.X.X.188:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.19:5060;branch=z9hG4bK4d7c55d2 From: "5555557444" ;tag=as6778cefb To: ;tag=SD785e199-09a2ce01 Call-ID: 769d78d77883e0cb39c8286b25e07799@X.X.X.19 CSeq: 102 INVITE Contact: Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO Accept: multipart/mixed, application/sdp, application/isup, application/dtmf, application/dtmf-relay Session-Expires: 120;refresher=uas Supported: timer Content-Length: 170 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 8824 20772 IN IP4 X.X.X.188 s=SIP Media Capabilities c=IN IP4 X.X.X.188 t=0 0 m=audio 25098 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv >>replaced-name*CLI> --- (14 headers 8 lines)--- >>replaced-name*CLI> Found RTP audio format 0 >>replaced-name*CLI> Peer audio RTP is at port X.X.X.188:25098 >>replaced-name*CLI> Found description format PCMU >>replaced-name*CLI> Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) >>replaced-name*CLI> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) >>replaced-name*CLI> list_route: hop: >>replaced-name*CLI> set_destination: Parsing for address/port to send to >>replaced-name*CLI> set_destination: set destination to X.X.X.188, port 5060 >>replaced-name*CLI> Transmitting (no NAT) to X.X.X.188:5060: ACK sip:5555550290-xosuburban-odrmtmoog0h4a@X.X.X.188:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP X.X.X.19:5060;branch=z9hG4bK7bee3b39 From: "5555557444" ;tag=as6778cefb To: ;tag=SD785e199-09a2ce01 Contact: Call-ID: 769d78d77883e0cb39c8286b25e07799@X.X.X.19 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- >>replaced-name*CLI> -- SIP/xo-bf71 answered SIP/5555557444-faaf >>replaced-name*CLI> We're at X.X.X.19 port 17286 >>replaced-name*CLI> Answering with preferred capability 0x4 (ulaw) >>replaced-name*CLI> Answering with non-codec capability 0x1 (telephone-event) >>replaced-name*CLI> Reliably Transmitting (no NAT) to X.X.X.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.2:5060;branch=z9hG4bK-f3ba2159 From: ;tag=bf094c222ae884b5o0 To: ;tag=as469c1f45 Call-ID: 5fb6ee66-49dba741@X.X.X.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 210 v=0 o=root 16881 16882 IN IP4 X.X.X.19 s=session c >>replaced-name*CLI> =IN IP4 X.X.X.19 t=0 0 m=audio 17286 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- >>replaced-name*CLI> -- Attempting native bridge of SIP/5555557444-faaf and SIP/xo-bf71 >>replaced-name*CLI> <-- SIP read from X.X.X.2:5060: ACK sip:5555550290@X.X.X.19 SIP/2.0 Via: SIP/2.0/UDP X.X.X.2:5060;branch=z9hG4bK-d436677 From: ;tag=bf094c222ae884b5o0 To: ;tag=as469c1f45 Call-ID: 5fb6ee66-49dba741@X.X.X.2 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="5555557444",realm="asterisk",nonce="62e57e84",uri="sip:5555550290@X.X.X.19",algorithm=MD5,response="a8c89c80ea6e674b8d7ba02aec77af59" Contact: User-Agent: Linksys/RT31P2-2.0.10(LIc) Content-Length: 0 >>replaced-name*CLI> --- (11 headers 0 lines)--- >>replaced-name*CLI> <-- SIP read from X.X.X.2:5060: BYE sip:5555550290@X.X.X.19 SIP/2.0 Via: SIP/2.0/UDP X.X.X.2:5060;branch=z9hG4bK-c75eb6cf From: ;tag=bf094c222ae884b5o0 To: ;tag=as469c1f45 Call-ID: 5fb6ee66-49dba741@X.X.X.2 CSeq: 103 BYE Max-Forwards: 70 Proxy-Authorization: Digest username="5555557444",realm="asterisk",nonce="62e57e84",uri="sip:5555550290@X.X.X.19",algorithm=MD5,response="0b4ff096b84ad3bfd6c37ae6e0a3c6b5" User-Agent: Linksys/RT31P2-2.0.10(LIc) Content-Length: 0 >>replaced-name*CLI> --- (10 headers 0 lines)--- >>replaced-name*CLI> Sending to X.X.X.2 : 5060 (non-NAT) >>replaced-name*CLI> Transmitting (no NAT) to X.X.X.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.2:5060;branch=z9hG4bK-c75eb6cf From: ;tag=bf094c222ae884b5o0 To: ;tag=as469c1f45 Call-ID: 5fb6ee66-49dba741@X.X.X.2 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause:: Normal Clearing --- >>replaced-name*CLI> set_destination: Parsing for address/port to send to >>replaced-name*CLI> set_destination: set destination to X.X.X.188, port 5060 >>replaced-name*CLI> Reliably Transmitting (no NAT) to X.X.X.188:5060: BYE sip:5555550290-xosuburban-odrmtmoog0h4a@X.X.X.188:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP X.X.X.19:5060;branch=z9hG4bK748a6366 From: "5555557444" ;tag=as6778cefb To: ;tag=SD785e199-09a2ce01 Contact: Call-ID: 769d78d77883e0cb39c8286b25e07799@X.X.X.19 CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 --- >>replaced-name*CLI> == Spawn extension (macro-dial-local, s, 3) exited non-zero on 'SIP/5555557444-faaf' in macro 'dial-local' == Spawn extension (subscriber, 5555550290, 1) exited non-zero on 'SIP/5555557444-faaf' >>replaced-name*CLI> <-- SIP read from X.X.X.188:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.19:5060;branch=z9hG4bK748a6366 From: "5555557444" ;tag=as6778cefb To: ;tag=SD785e199-09a2ce01 Call-ID: 769d78d77883e0cb39c8286b25e07799@X.X.X.19 CSeq: 103 BYE Content-Length: 0 >>replaced-name*CLI> --- (7 headers 0 lines)--- >>replaced-name*CLI> Destroying call '769d78d77883e0cb39c8286b25e07799@X.X.X.19' >>replaced-name*CLI> Destroying call '5fb6ee66-49dba741@X.X.X.2' >>replaced-name*CLI>