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Paste #1206: sip debug
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Pasted by chrivers
<-- SIP read from 10.0.0.103:5060: INVITE sip:1234@10.0.0.3 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-869fe26d From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=d4ba946f155b91do0 To: <sip:1234@10.0.0.3> Call-ID: 1e4fcc57-edeba255@10.0.0.103 CSeq: 101 INVITE Max-Forwards: 70 Contact: "Sikkerhed.org" <sip:chrivers@10.0.0.103:5060> Expires: 240 User-Agent: Linksys/SPA942-5.1.15(a) Content-Length: 391 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 206913 206913 IN IP4 10.0.0.103 s=- c=IN IP4 10.0.0.103 t=0 0 m=audio 16472 RTP/AVP 0 2 4 8 18 96 97 98 101 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:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (14 headers 18 lines) --- Using INVITE request as basis request - 1e4fcc57-edeba255@10.0.0.103 Sending to 10.0.0.103 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 10.0.0.103:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-869fe26d;received=10.0.0.103 From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=d4ba946f155b91do0 To: <sip:1234@10.0.0.3>;tag=as4c2b15c7 Call-ID: 1e4fcc57-edeba255@10.0.0.103 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6dd0fcca" Content-Length: 0 --- Scheduling destruction of call '1e4fcc57-edeba255@10.0.0.103' in 15000 ms Found user 'chrivers' <-- SIP read from 10.0.0.103:5060: ACK sip:1234@10.0.0.3 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-869fe26d From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=d4ba946f155b91do0 To: <sip:1234@10.0.0.3>;tag=as4c2b15c7 Call-ID: 1e4fcc57-edeba255@10.0.0.103 CSeq: 101 ACK Max-Forwards: 70 Contact: "Sikkerhed.org" <sip:chrivers@10.0.0.103:5060> User-Agent: Linksys/SPA942-5.1.15(a) Content-Length: 0 --- (10 headers 0 lines) --- <-- SIP read from 10.0.0.103:5060: INVITE sip:1234@10.0.0.3 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-3bbd20e9 From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=d4ba946f155b91do0 To: <sip:1234@10.0.0.3> Call-ID: 1e4fcc57-edeba255@10.0.0.103 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="chrivers",realm="asterisk",nonce="6dd0fcca",uri="sip:1234@10.0.0.3",algorithm=MD5,response="4445e375eb676a3ab8f6b53536add93d" Contact: "Sikkerhed.org" <sip:chrivers@10.0.0.103:5060> Expires: 240 User-Agent: Linksys/SPA942-5.1.15(a) Content-Length: 391 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 206913 206913 IN IP4 10.0.0.103 s=- c=IN IP4 10.0.0.103 t=0 0 m=audio 16472 RTP/AVP 0 2 4 8 18 96 97 98 101 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:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (15 headers 18 lines) --- Using INVITE request as basis request - 1e4fcc57-edeba255@10.0.0.103 Sending to 10.0.0.103 : 5060 (non-NAT) Found user 'chrivers' 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 101 Peer audio RTP is at port 10.0.0.103:16472 Found description format PCMU Found description format G726-32 Found description format G723 Found description format PCMA Found description format G729a Found description format G726-40 Found description format G726-24 Found description format G726-16 Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 1234 in demo (domain 10.0.0.3) list_route: hop: <sip:chrivers@10.0.0.103:5060> Transmitting (no NAT) to 10.0.0.103:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-3bbd20e9;received=10.0.0.103 From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=d4ba946f155b91do0 To: <sip:1234@10.0.0.3> Call-ID: 1e4fcc57-edeba255@10.0.0.103 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:1234@10.0.0.3> Content-Length: 0 --- Nov 6 02:21:42 WARNING[8813]: channel.c:2597 ast_request: No channel type registered for 'Console' Nov 6 02:21:42 NOTICE[8813]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Console' (cause 66 - Channel not implemented) We're at 10.0.0.3 port 17976 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.0.103:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-3bbd20e9;received=10.0.0.103 From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=d4ba946f155b91do0 To: <sip:1234@10.0.0.3>;tag=as5dea0fcc Call-ID: 1e4fcc57-edeba255@10.0.0.103 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:1234@10.0.0.3> Content-Type: application/sdp Content-Length: 230 v=0 o=root 8776 8776 IN IP4 10.0.0.3 s=session c=IN IP4 10.0.0.3 t=0 0 m=audio 17976 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 - - - - --- <-- SIP read from 10.0.0.103:5060: ACK sip:1234@10.0.0.3 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-8d1b06ee From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=d4ba946f155b91do0 To: <sip:1234@10.0.0.3>;tag=as5dea0fcc Call-ID: 1e4fcc57-edeba255@10.0.0.103 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="chrivers",realm="asterisk",nonce="6dd0fcca",uri="sip:1234@10.0.0.3",algorithm=MD5,response="4445e375eb676a3ab8f6b53536add93d" Contact: "Sikkerhed.org" <sip:chrivers@10.0.0.103:5060> User-Agent: Linksys/SPA942-5.1.15(a) Content-Length: 0 --- (11 headers 0 lines) --- <-- SIP read from 10.0.0.103:5060: BYE sip:1234@10.0.0.3 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-e891b871 From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=d4ba946f155b91do0 To: <sip:1234@10.0.0.3>;tag=as5dea0fcc Call-ID: 1e4fcc57-edeba255@10.0.0.103 CSeq: 103 BYE Max-Forwards: 70 Proxy-Authorization: Digest username="chrivers",realm="asterisk",nonce="6dd0fcca",uri="sip:1234@10.0.0.3",algorithm=MD5,response="468cf3766d5f8808f7b69845380de63c" User-Agent: Linksys/SPA942-5.1.15(a) Content-Length: 0 --- (10 headers 0 lines) --- Sending to 10.0.0.103 : 5060 (non-NAT) Transmitting (no NAT) to 10.0.0.103:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-e891b871;received=10.0.0.103 From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=d4ba946f155b91do0 To: <sip:1234@10.0.0.3>;tag=as5dea0fcc Call-ID: 1e4fcc57-edeba255@10.0.0.103 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:1234@10.0.0.3> Content-Length: 0 X-Asterisk-HangupCause: Channel not implemented --- Destroying call '1e4fcc57-edeba255@10.0.0.103'
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