From: Ronald Chan ;tag=ff7a36c716cb548bo0 To: Call-ID: 8b1ce077-fcd9369b@192.168.4.196 CSeq: 101 INVITE Max-Forwards: 70 Contact: Ronald Chan Expires: 240 User-Agent: Linksys/PAP2-3.1.3(LS) Content-Length: 424 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 508455 508455 IN IP4 192.168.4.196 s=- c=IN IP4 192.168.4.196 t=0 0 m=audio 16424 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 <-------------> --- (14 headers 19 lines) --- Sending to 192.168.4.196 : 5060 (no NAT) Using INVITE request as basis request - 8b1ce077-fcd9369b@192.168.4.196 <--- Reliably Transmitting (NAT) to 192.168.4.196:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.4.196:5060;branch=z9hG4bK-d4141ceb;received=192.168.4.196 From: Ronald Chan ;tag=ff7a36c716cb548bo0 To: ;tag=as77a04afd Call-ID: 8b1ce077-fcd9369b@192.168.4.196 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4cf864a9" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '8b1ce077-fcd9369b@192.168.4.196' in 32000 ms (Method: INVITE) Found user '201' stealth*CLI> <--- SIP read from 192.168.4.196:5060 ---> ACK sip:918007654321@192.168.4.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.196:5060;branch=z9hG4bK-d4141ceb From: Ronald Chan ;tag=ff7a36c716cb548bo0 To: ;tag=as77a04afd Call-ID: 8b1ce077-fcd9369b@192.168.4.196 CSeq: 101 ACK Max-Forwards: 70 Contact: Ronald Chan User-Agent: Linksys/PAP2-3.1.3(LS) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- stealth*CLI> <--- SIP read from 192.168.4.196:5060 ---> INVITE sip:918007654321@192.168.4.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.196:5060;branch=z9hG4bK-63318f43 From: Ronald Chan ;tag=ff7a36c716cb548bo0 To: Call-ID: 8b1ce077-fcd9369b@192.168.4.196 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="201",realm="asterisk",nonce="4cf864a9",uri="sip:918007654321@192.168.4.1",algorithm=MD5,response="ee485e474cfee3ab43ac61d2074ef116" Contact: Ronald Chan Expires: 240 User-Agent: Linksys/PAP2-3.1.3(LS) Content-Length: 424 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 508455 508455 IN IP4 192.168.4.196 s=- c=IN IP4 192.168.4.196 t=0 0 m=audio 16424 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 <-------------> --- (15 headers 19 lines) --- Sending to 192.168.4.196 : 5060 (NAT) Using INVITE request as basis request - 8b1ce077-fcd9369b@192.168.4.196 Found user '201' Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.4.196:16424 Found description format PCMU for ID 0 Found description format G726-32 for ID 2 Found description format G723 for ID 4 Found description format PCMA for ID 8 Found description format G729a for ID 18 Found description format G726-40 for ID 96 Found description format G726-24 for ID 97 Found description format G726-16 for ID 98 Found description format NSE for ID 100 Found description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.4.196:16424 Looking for 918007654321 in internal (domain 192.168.4.1) list_route: hop: <--- Transmitting (NAT) to 192.168.4.196:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.4.196:5060;branch=z9hG4bK-63318f43;received=192.168.4.196 From: Ronald Chan ;tag=ff7a36c716cb548bo0 To: Call-ID: 8b1ce077-fcd9369b@192.168.4.196 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [918007654321@internal:1] Macro("SIP/201-0902b608", "extensions|Sip/fwd-pulver|*18007654321") in new stack -- Executing [s@macro-extensions:1] GotoIf("SIP/201-0902b608", "0?s|intercom") in new stack -- Executing [s@macro-extensions:2] GotoIf("SIP/201-0902b608", "0?s|intercom:s|outgoing") in new stack -- Goto (macro-extensions,s,3) -- Executing [s@macro-extensions:3] Set("SIP/201-0902b608", "CALLFILE=OUTGOING-CALL/*18007654321--1161879618.12.wav") in new stack -- Executing [s@macro-extensions:4] Goto("SIP/201-0902b608", "s|mixnow") in new stack -- Goto (macro-extensions,s,6) -- Executing [s@macro-extensions:6] MixMonitor("SIP/201-0902b608", "OUTGOING-CALL/*18007654321--1161879618.12.wav|bW(1)") in new stack -- Executing [s@macro-extensions:7] Dial("SIP/201-0902b608", "Sip/fwd-pulver/*18007654321|120|tH") in new stack [Oct 27 00:20:18] WARNING[10515]: translate.c:86 powerof: No bits set? 0 Audio is at 210.213.198.134 port 16936 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 69.90.155.70:5060: INVITE sip:*18007654321@fwd.pulver.com SIP/2.0 Via: SIP/2.0/UDP 210.213.198.134:5060;branch=z9hG4bK393d6f4b;rport From: "Ronald Chan" ;tag=as07f0cd64 To: Contact: Call-ID: 4a3a23646669936f5e6cf45b7fa97d15@210.213.198.134 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 26 Oct 2006 16:20:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 234 v=0 o=root 10321 10321 IN IP4 210.213.198.134 s=session c=IN IP4 210.213.198.134 t=0 0 m=audio 16936 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- -- Called fwd-pulver/*18007654321 == Begin MixMonitor Recording SIP/201-0902b608 stealth*CLI> <--- SIP read from 69.90.155.70:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 210.213.198.134:5060;branch=z9hG4bK393d6f4b;rport=5060 From: "Ronald Chan" ;tag=as07f0cd64 To: Call-ID: 4a3a23646669936f5e6cf45b7fa97d15@210.213.198.134 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- stealth*CLI> <--- SIP read from 69.90.155.70:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 210.213.198.134:5060;branch=z9hG4bK393d6f4b;rport=5060 From: "Ronald Chan" ;tag=as07f0cd64 To: Call-ID: 4a3a23646669936f5e6cf45b7fa97d15@210.213.198.134 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- stealth*CLI> <--- SIP read from 69.90.155.70:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 210.213.198.134:5060;branch=z9hG4bK393d6f4b;rport=5060 From: "Ronald Chan" ;tag=as07f0cd64 To: ;tag=as7b0a5644 Call-ID: 4a3a23646669936f5e6cf45b7fa97d15@210.213.198.134 CSeq: 102 INVITE User-Agent: astlax02 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 12873 12873 IN IP4 216.168.169.100 s=session c=IN IP4 216.168.169.100 t=0 0 m=audio 14444 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> --- (11 headers 13 lines) --- Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 216.168.169.100:14444 Found description format G729 for ID 18 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=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 216.168.169.100:14444 -- SIP/fwd-pulver-09030500 is making progress passing it to SIP/201-0902b608 [Oct 27 00:20:21] WARNING[10515]: translate.c:86 powerof: No bits set? 0 [Oct 27 00:20:21] WARNING[10515]: translate.c:86 powerof: No bits set? 0 Audio is at 192.168.4.1 port 10570 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 192.168.4.196:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.4.196:5060;branch=z9hG4bK-63318f43;received=192.168.4.196 From: Ronald Chan ;tag=ff7a36c716cb548bo0 To: ;tag=as1c66eefe Call-ID: 8b1ce077-fcd9369b@192.168.4.196 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp ontent-Length: 250 v=0 o=root 10321 10321 IN IP4 192.168.4.1 s=session c=IN IP4 192.168.4.1 t=0 0 m=audio 10570 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv <------------> stealth*CLI> <--- SIP read from 69.90.155.70:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 210.213.198.134:5060;branch=z9hG4bK393d6f4b;rport=5060 Record-Route: From: "Ronald Chan" ;tag=as07f0cd64 To: ;tag=as7b0a5644 Call-ID: 4a3a23646669936f5e6cf45b7fa97d15@210.213.198.134 CSeq: 102 INVITE User-Agent: astlax02 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 12873 12874 IN IP4 216.168.169.100 s=session c=IN IP4 216.168.169.100 t=0 0 m=audio 14444 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> --- (12 headers 13 lines) --- Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 216.168.169.100:14444 Found description format G729 for ID 18 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=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 216.168.169.100:14444 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 69.90.155.70, port 5060 Transmitting (no NAT) to 69.90.155.70:5060: ACK sip:1885018007654321@216.168.169.100 SIP/2.0 Via: SIP/2.0/UDP 210.213.198.134:5060;branch=z9hG4bK2dd07c5a;rport Route: From: "Ronald Chan" ;tag=as07f0cd64 To: ;tag=as7b0a5644 Contact: Call-ID: 4a3a23646669936f5e6cf45b7fa97d15@210.213.198.134 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/fwd-pulver-09030500 answered SIP/201-0902b608 [Oct 27 00:20:22] WARNING[10515]: translate.c:86 powerof: No bits set? 0 [Oct 27 00:20:22] WARNING[10515]: translate.c:86 powerof: No bits set? 0 Audio is at 192.168.4.1 port 10570 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.4.196:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.4.196:5060;branch=z9hG4bK-63318f43;received=192.168.4.196 From: Ronald Chan ;tag=ff7a36c716cb548bo0 To: ;tag=as1c66eefe Call-ID: 8b1ce077-fcd9369b@192.168.4.196 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 250 v=0 o=root 10321 10322 IN IP4 192.168.4.1 s=session c=IN IP4 192.168.4.1 t=0 0 m=audio 10570 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv <------------> stealth*CLI> <--- SIP read from 192.168.4.196:5060 ---> ACK sip:918007654321@192.168.4.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.196:5060;branch=z9hG4bK-71d977e6 From: Ronald Chan ;tag=ff7a36c716cb548bo0 To: ;tag=as1c66eefe Call-ID: 8b1ce077-fcd9369b@192.168.4.196 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="201",realm="asterisk",nonce="4cf864a9",uri="sip:918007654321@192.168.4.1",algorithm=MD5,response="20e3cdbb1ac810d431b7b24a48f0dafd" Contact: Ronald Chan User-Agent: Linksys/PAP2-3.1.3(LS) Content-Length: 0 <-------------> --- (11 headers 0 lines) --- stealth*CLI> <--- SIP read from 192.168.4.196:5060 ---> BYE sip:918007654321@192.168.4.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.196:5060;branch=z9hG4bK-335473cd From: Ronald Chan ;tag=ff7a36c716cb548bo0 To: ;tag=as1c66eefe Call-ID: 8b1ce077-fcd9369b@192.168.4.196 CSeq: 103 BYE Max-Forwards: 70 Proxy-Authorization: Digest username="201",realm="asterisk",nonce="4cf864a9",uri="sip:918007654321@192.168.4.1",algorithm=MD5,response="6cb3f3ae3a8aa2d1d79287a8098a09a3" User-Agent: Linksys/PAP2-3.1.3(LS) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 192.168.4.196 : 5060 (NAT) <--- Transmitting (NAT) to 192.168.4.196:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.4.196:5060;branch=z9hG4bK-335473cd;received=192.168.4.196 From: Ronald Chan ;tag=ff7a36c716cb548bo0 To: ;tag=as1c66eefe Call-ID: 8b1ce077-fcd9369b@192.168.4.196 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '4a3a23646669936f5e6cf45b7fa97d15@210.213.198.134' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 69.90.155.70, port 5060 Reliably Transmitting (no NAT) to 69.90.155.70:5060: BYE sip:1885018007654321@216.168.169.100 SIP/2.0 Via: SIP/2.0/UDP 210.213.198.134:5060;branch=z9hG4bK0c5d0df3;rport Route: From: "Ronald Chan" ;tag=as07f0cd64 To: ;tag=as7b0a5644 Call-ID: 4a3a23646669936f5e6cf45b7fa97d15@210.213.198.134 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (macro-extensions, s, 7) exited non-zero on 'SIP/201-0902b608' in macro 'extensions' == Spawn extension (macro-extensions, s, 7) exited non-zero on 'SIP/201-0902b608' == End MixMonitor Recording SIP/201-0902b608 Really destroying SIP dialog '8b1ce077-fcd9369b@192.168.4.196' Method: BYE stealth*CLI> <--- SIP read from 69.90.155.70:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 210.213.198.134:5060;branch=z9hG4bK0c5d0df3;rport=5060 Record-Route: From: "Ronald Chan" ;tag=as07f0cd64 To: ;tag=as7b0a5644 Call-ID: 4a3a23646669936f5e6cf45b7fa97d15@210.213.198.134 CSeq: 103 BYE User-Agent: astlax02 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '4a3a23646669936f5e6cf45b7fa97d15@210.213.198.134' Method: INVITE Really destroying SIP dialog '0d664fea3cdafb1369079d823a58bbae@210.213.198.134' Method: REGISTER stealth*CLI> core nodebug stealth*CLI>