=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2006.05.10 13:41:33 =~=~=~=~=~=~=~=~=~=~=~= <-- SIP read from 192.168.10.218:5060: INVITE sip:235@voipbackup.testdomain.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.218:5060;branch=z9hG4bK9b3ae99714CEFE6C From: "Test User 1" ;tag=BBDD557D-F7FBB430 To: CSeq: 1 INVITE Call-ID: 6457d0d9-c4a4b97b-6986221e@192.168.10.218 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 253 v=0 o=- 1147293694 1147293694 IN IP4 192.168.10.218 s=Polycom IP Phone c=IN IP4 192.168.10.218 t=0 0 a=sendrecv m=audio 2228 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 voipbackup*CLI> --- (14 headers 11 lines)--- voipbackup*CLI> Sending to 192.168.10.218 : 5060 (no NAT) voipbackup*CLI> Using INVITE request as basis request - 6457d0d9-c4a4b97b-6986221e@192.168.10.218 voipbackup*CLI> Reliably Transmitting (no NAT) to 192.168.10.218:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.10.218:5060;branch=z9hG4bK9b3ae99714CEFE6C;received=192.168.10.218 From: "Test User 1" ;tag=BBDD557D-F7FBB430 To: ;tag=as3d9c1790 Call-ID: 6457d0d9-c4a4b97b-6986221e@192.168.10.218 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="469f4afb" Content-Length: 0 --- voipbackup*CLI> Scheduling destruction of SIP dialog '6457d0d9-c4a4b97b-6986221e@192.168.10.218' in 32000 ms (Method: INVITE) voipbackup*CLI> Found user '547' voipbackup*CLI> <-- SIP read from 192.168.10.218:5060: ACK sip:235@voipbackup.testdomain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.218:5060;branch=z9hG4bK9b3ae99714CEFE6C From: "Test User 1" ;tag=BBDD557D-F7FBB430 To: ;tag=as3d9c1790 CSeq: 1 ACK Call-ID: 6457d0d9-c4a4b97b-6986221e@192.168.10.218 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER voipbackup*CLI> User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067 Max-Forwards: 70 Content-Length: 0 voipbackup*CLI> --- (11 headers 0 lines)--- voipbackup*CLI> <-- SIP read from 192.168.10.218:5060: INVITE sip:235@voipbackup.testdomain.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.218:5060;branch=z9hG4bK7ef056a220ABB6DF From: "Test User 1" ;tag=BBDD557D-F7FBB430 To: CSeq: 2 INVITE Call-ID: 6457d0d9-c4a4b97b-6986221e@192.168.10.218 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067 Supported: 100rel,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="547", realm="asterisk", nonce="469f4afb", uri="sip:235@voipbackup.testdomain.com:5060;user=phone", response="43085feeddb013726e7b30edc3b12339", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 253 v=0 o=- 1147293694 1147293694 IN IP4 192.168.10.218 s=Polycom IP Phone c=IN IP4 192.168.10.218 t=0 0 a=sendrecv m=audio 2228 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 voipbackup*CLI> --- (15 headers 11 lines)--- voipbackup*CLI> Sending to 192.168.10.218 : 5060 (no NAT) voipbackup*CLI> Using INVITE request as basis request - 6457d0d9-c4a4b97b-6986221e@192.168.10.218 Found user '547' voipbackup*CLI> Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 voipbackup*CLI> Found RTP audio format 101 Peer audio RTP is at port 192.168.10.218:2228 voipbackup*CLI> Found description format PCMU Found description format PCMA Found description format G729 voipbackup*CLI> Found description format telephone-event Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) voipbackup*CLI> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 235 in default (domain voipbackup.testdomain.com) voipbackup*CLI> list_route: hop: voipbackup*CLI> Transmitting (no NAT) to 192.168.10.218:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.218:5060;branch=z9hG4bK7ef056a220ABB6DF;received=192.168.10.218 From: "Test User 1" ;tag=BBDD557D-F7FBB430 To: Call-ID: 6457d0d9-c4a4b97b-6986221e@192.168.10.218 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- voipbackup*CLI> -- Executing Set("SIP/547-f958", "TIMEOUT(absolute)=32400") in new stack -- Channel will hangup at 2006-05-11 05:39:39 UTC. -- Executing Dial("SIP/547-f958", "SIP/235|5") in new stack voipbackup*CLI> We're at 192.168.10.24 port 15548 voipbackup*CLI> Adding codec 0x4 (ulaw) to SDP voipbackup*CLI> Adding codec 0x100 (g729) to SDP voipbackup*CLI> Adding non-codec 0x1 (telephone-event) to SDP voipbackup*CLI> Reliably Transmitting (no NAT) to 192.168.10.186:5060: INVITE sip:235@192.168.10.186:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK33946e87;rport From: "Test User 1" ;tag=as2df4c19e To: Contact: Call-ID: 0234b31f3f0190c54b9951063c53f9ca@192.168.10.24 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 10 May 2006 20:39:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 18105 18105 IN IP4 192.168.10.24 s=session c=IN IP4 192.168.10.24 t=0 0 m=audio 15548 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- voipbackup*CLI> -- Called 235 voipbackup*CLI> <-- SIP read from 192.168.10.186:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK33946e87;rport From: "Test User 1" ;tag=as2df4c19e To: ;tag=45ECE2D6-A190372F CSeq: 102 INVITE Call-ID: 0234b31f3f0190c54b9951063c53f9ca@192.168.10.24 Contact: User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.3.0067 Content-Length: 0 --- (9 headers 0 lines)--- voipbackup*CLI> <-- SIP read from 192.168.10.186:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK33946e87;rport From: "Test User 1" ;tag=as2df4c19e To: ;tag=45ECE2D6-A190372F CSeq: 102 INVITE Call-ID: 0234b31f3f0190c54b9951063c53f9ca@192.168.10.24 Contact: User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.3.0067 Allow-Events: talk,hold,conference Content-Length: 0 --- (10 headers 0 lines)--- -- SIP/235-ccf0 is ringing voipbackup*CLI> Transmitting (no NAT) to 192.168.10.218:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.218:5060;branch=z9hG4bK7ef056a220ABB6DF;received=192.168.10.218 From: "Test User 1" ;tag=BBDD557D-F7FBB430 To: ;tag=as389a696a Call-ID: 6457d0d9-c4a4b97b-6986221e@192.168.10.218 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- voipbackup*CLI> <-- SIP read from 192.168.10.186:5060: SUBSCRIBE sip:8000@voipbackup.testdomain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.186:5060;branch=z9hG4bK3079867d66F62224 From: "Test User 2" ;tag=B3B4B467-AB4840C0 To: CSeq: 1 SUBSCRIBE Call-ID: ef22b5cb-b8bc519-cebf4172@192.168.10.186 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.3.0067 Accept: application/simple-message-summary Max-Forwards: 70 Expires: 3600 Content-Length: 0 voipbackup*CLI> --- (14 headers 0 lines)--- voipbackup*CLI> Creating new subscription voipbackup*CLI> Sending to 192.168.10.186 : 5060 (no NAT) voipbackup*CLI> Found peer '235' voipbackup*CLI> Transmitting (no NAT) to 192.168.10.186:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.186:5060;branch=z9hG4bK3079867d66F62224;received=192.168.10.186 From: "Test User 2" ;tag=B3B4B467-AB4840C0 To: ;tag=as3d32e852 Call-ID: ef22b5cb-b8bc519-cebf4172@192.168.10.186 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: WWW-Authenticate: Digest realm="asterisk", nonce="7b5f7c34" Content-Length: 0 --- voipbackup*CLI> Scheduling destruction of SIP dialog 'ef22b5cb-b8bc519-cebf4172@192.168.10.186' in 32000 ms (Method: SUBSCRIBE) voipbackup*CLI> <-- SIP read from 192.168.10.186:5060: SUBSCRIBE sip:8000@voipbackup.testdomain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.186:5060;branch=z9hG4bK2a91200e159D83B5 From: "Test User 2" ;tag=B3B4B467-AB4840C0 To: CSeq: 2 SUBSCRIBE Call-ID: ef22b5cb-b8bc519-cebf4172@192.168.10.186 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.3.0067 Accept: application/simple-message-summary Authorization: Digest username="235", realm="asterisk", nonce="7b5f7c34", uri="sip:8000@voipbackup.testdomain.com:5060", response="ef6b0ae174589ee2b17129466970f726", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 voipbackup*CLI> --- (15 headers 0 lines)--- voipbackup*CLI> Found peer '235' voipbackup*CLI> Looking for 8000 in default (domain voipbackup.testdomain.com) voipbackup*CLI> Transmitting (no NAT) to 192.168.10.186:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.10.186:5060;branch=z9hG4bK2a91200e159D83B5;received=192.168.10.186 From: "Test User 2" ;tag=B3B4B467-AB4840C0 To: ;tag=as3d32e852 Call-ID: ef22b5cb-b8bc519-cebf4172@192.168.10.186 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: voipbackup*CLI> Content-Length: 0 --- Really destroying SIP dialog 'ef22b5cb-b8bc519-cebf4172@192.168.10.186' Method: SUBSCRIBE voipbackup*CLI> <-- SIP read from 192.168.10.218:5060: SUBSCRIBE sip:8000@voipbackup.testdomain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.218:5060;branch=z9hG4bK785f1c31A1D9438 From: "Test User 1" ;tag=8624E469-C1FBDE7C To: CSeq: 1 SUBSCRIBE Call-ID: 5a1fcf45-158f7a27-3de64c2a@192.168.10.218 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067 Accept: application/simple-message-summary Max-Forwards: 70 Expires: 3600 Content-Length: 0 voipbackup*CLI> --- (14 headers 0 lines)--- voipbackup*CLI> Creating new subscription voipbackup*CLI> Sending to 192.168.10.218 : 5060 (no NAT) voipbackup*CLI> Found peer '547' voipbackup*CLI> Transmitting (no NAT) to 192.168.10.218:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.218:5060;branch=z9hG4bK785f1c31A1D9438;received=192.168.10.218 From: "Test User 1" ;tag=8624E469-C1FBDE7C To: ;tag=as32e4a512 Call-ID: 5a1fcf45-158f7a27-3de64c2a@192.168.10.218 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: WWW-Authenticate: Digest realm="asterisk", nonce="46b67f4f" Content-Length: 0 --- Scheduling destruction of SIP dialog '5a1fcf45-158f7a27-3de64c2a@192.168.10.218' in 32000 ms (Method: SUBSCRIBE) voipbackup*CLI> <-- SIP read from 192.168.10.218:5060: SUBSCRIBE sip:8000@voipbackup.testdomain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.218:5060;branch=z9hG4bKaae12d2eED8A600B From: "Test User 1" ;tag=8624E469-C1FBDE7C To: CSeq: 2 SUBSCRIBE Call-ID: 5a1fcf45-158f7a27-3de64c2a@192.168.10.218 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067 Accept: application/simple-message-summary Authorization: Digest username="547", realm="asterisk", nonce="46b67f4f", uri="sip:8000@voipbackup.testdomain.com:5060", response="a0eb9061d12c559fb562674711a112dd", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 voipbackup*CLI> --- (15 headers 0 lines)--- voipbackup*CLI> Found peer '547' voipbackup*CLI> Looking for 8000 in default (domain voipbackup.testdomain.com) voipbackup*CLI> Transmitting (no NAT) to 192.168.10.218:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.10.218:5060;branch=z9hG4bKaae12d2eED8A600B;received=192.168.10.218 From: "Test User 1" ;tag=8624E469-C1FBDE7C To: ;tag=as32e4a512 Call-ID: 5a1fcf45-158f7a27-3de64c2a@192.168.10.218 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- voipbackup*CLI> Really destroying SIP dialog '5a1fcf45-158f7a27-3de64c2a@192.168.10.218' Method: SUBSCRIBE voipbackup*CLI> <-- SIP read from 192.168.10.218:5060: SUBSCRIBE sip:547@voipbackup.testdomain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.218:5060;branch=z9hG4bK27587a40DDA69F0D From: "Test User 1" ;tag=B8D18BB6-DFB80F31 To: CSeq: 1 SUBSCRIBE Call-ID: 52b567b2-b38ef784-52deb36f@192.168.10.218 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: missed-call-summary User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067 Accept: message/sipfrag Max-Forwards: 70 Expires: 3600 Content-Length: 0 voipbackup*CLI> --- (14 headers 0 lines)--- voipbackup*CLI> Creating new subscription voipbackup*CLI> Sending to 192.168.10.218 : 5060 (no NAT) voipbackup*CLI> Found peer '547' voipbackup*CLI> Transmitting (no NAT) to 192.168.10.218:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.218:5060;branch=z9hG4bK27587a40DDA69F0D;received=192.168.10.218 From: "Test User 1" ;tag=B8D18BB6-DFB80F31 To: ;tag=as62787f02 Call-ID: 52b567b2-b38ef784-52deb36f@192.168.10.218 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: WWW-Authenticate: Digest realm="asterisk", nonce="5fc723e8" Content-Length: 0 --- voipbackup*CLI> Scheduling destruction of SIP dialog '52b567b2-b38ef784-52deb36f@192.168.10.218' in 32000 ms (Method: SUBSCRIBE) voipbackup*CLI> <-- SIP read from 192.168.10.218:5060: SUBSCRIBE sip:547@voipbackup.testdomain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.218:5060;branch=z9hG4bK9bda8453D730E648 From: "Test User 1" ;tag=B8D18BB6-DFB80F31 To: CSeq: 2 SUBSCRIBE Call-ID: 52b567b2-b38ef784-52deb36f@192.168.10.218 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: missed-call-summary User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067 Accept: message/sipfrag Authorization: Digest username="547", realm="asterisk", nonce="5fc723e8", uri="sip:547@voipbackup.testdomain.com:5060", response="f80626d62e147debcfbc0e816bf10f9f", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 voipbackup*CLI> --- (15 headers 0 lines)--- voipbackup*CLI> Found peer '547' voipbackup*CLI> Looking for 547 in default (domain voipbackup.testdomain.com) voipbackup*CLI> Transmitting (no NAT) to 192.168.10.218:5060: SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 192.168.10.218:5060;branch=z9hG4bK9bda8453D730E648;received=192.168.10.218 From: "Test User 1" ;tag=B8D18BB6-DFB80F31 To: ;tag=as62787f02 Call-ID: 52b567b2-b38ef784-52deb36f@192.168.10.218 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- voipbackup*CLI> Really destroying SIP dialog '52b567b2-b38ef784-52deb36f@192.168.10.218' Method: SUBSCRIBE voipbackup*CLI> <-- SIP read from 192.168.10.186:5060: SIP/2.0 603 Decline Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK33946e87;rport From: "Test User 1" ;tag=as2df4c19e To: ;tag=45ECE2D6-A190372F CSeq: 102 INVITE Call-ID: 0234b31f3f0190c54b9951063c53f9ca@192.168.10.24 Contact: User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.3.0067 Content-Length: 0 voipbackup*CLI> --- (9 headers 0 lines)--- voipbackup*CLI> -- Got SIP response 603 "Decline" back from 192.168.10.186 voipbackup*CLI> Transmitting (no NAT) to 192.168.10.186:5060: ACK sip:235@192.168.10.186:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK33946e87;rport From: "Test User 1" ;tag=as2df4c19e To: ;tag=45ECE2D6-A190372F Contact: Call-ID: 0234b31f3f0190c54b9951063c53f9ca@192.168.10.24 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- voipbackup*CLI> -- SIP/235-ccf0 is busy voipbackup*CLI> -- Nobody picked up in 5000 ms voipbackup*CLI> -- Executing Voi