=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2006.05.10 13:38:49 =~=~=~=~=~=~=~=~=~=~=~= <-- 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=z9hG4bK66dab9d190BF6956 From: "Test User 1" ;tag=87231757-D037E2DA To: CSeq: 1 INVITE Call-ID: 986fbcf3-3fd7ab75-6e6a61e8@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=- 1147293532 1147293532 IN IP4 192.168.10.218 s=Polycom IP Phone c=IN IP4 192.168.10.218 t=0 0 a=sendrecv m=audio 2226 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)--- Sending to 192.168.10.218 : 5060 (no NAT) Using INVITE request as basis request - 986fbcf3-3fd7ab75-6e6a61e8@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=z9hG4bK66dab9d190BF6956;received=192.168.10.218 From: "Test User 1" ;tag=87231757-D037E2DA To: ;tag=as65e66127 Call-ID: 986fbcf3-3fd7ab75-6e6a61e8@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="4186d326" Content-Length: 0 --- voipbackup*CLI> Scheduling destruction of SIP dialog '986fbcf3-3fd7ab75-6e6a61e8@192.168.10.218' in 32000 ms (Method: INVITE) 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=z9hG4bK66dab9d190BF6956 From: "Test User 1" ;tag=87231757-D037E2DA To: ;tag=as65e66127 CSeq: 1 ACK Call-ID: 986fbcf3-3fd7ab75-6e6a61e8@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 --- (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=z9hG4bK326e6e2c4B6AF299 From: "Test User 1" ;tag=87231757-D037E2DA To: CSeq: 2 INVITE Call-ID: 986fbcf3-3fd7ab75-6e6a61e8@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="4186d326", uri="sip:235@voipbackup.testdomain.com:5060;user=phone", response="e6a5e42dd256ccc5774148b3c139839d", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 253 v=0 o=- 1147293532 1147293532 IN IP4 192.168.10.218 s=Polycom IP Phone c=IN IP4 192.168.10.218 t=0 0 a=sendrecv m=audio 2226 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 --- (15 headers 11 lines)--- Sending to 192.168.10.218 : 5060 (no NAT) Using INVITE request as basis request - 986fbcf3-3fd7ab75-6e6a61e8@192.168.10.218 Found user '547' 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 192.168.10.218:2226 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event voipbackup*CLI> Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) voipbackup*CLI> Looking for 235 in default (domain voipbackup.testdomain.com) voipbackup*CLI> list_route: hop: 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=z9hG4bK326e6e2c4B6AF299;received=192.168.10.218 From: "Test User 1" ;tag=87231757-D037E2DA To: Call-ID: 986fbcf3-3fd7ab75-6e6a61e8@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 --- -- Executing NoOp("SIP/547-2c0f", ""Step One"") in new stack voipbackup*CLI> -- Executing Dial("SIP/547-2c0f", "SIP/235|5") in new stack We're at 192.168.10.24 port 19996 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP 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=z9hG4bK7fb72f6a;rport From: "Test User 1" ;tag=as3e2c93e1 To: Contact: Call-ID: 4e1a2d99019a82203eedde5039d83e72@192.168.10.24 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 10 May 2006 20:36:57 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 19996 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 - - - - --- -- 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=z9hG4bK7fb72f6a;rport From: "Test User 1" ;tag=as3e2c93e1 To: ;tag=C6AC3E23-B3FEEA7C CSeq: 102 INVITE Call-ID: 4e1a2d99019a82203eedde5039d83e72@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> <-- SIP read from 192.168.10.186:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK7fb72f6a;rport From: "Test User 1" ;tag=as3e2c93e1 To: ;tag=C6AC3E23-B3FEEA7C CSeq: 102 INVITE Call-ID: 4e1a2d99019a82203eedde5039d83e72@192.168.10.24 Contact: User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.3.0067 Allow-Events: talk,hold,conference Content-Length: 0 voipbackup*CLI> --- (10 headers 0 lines)--- voipbackup*CLI> -- SIP/235-9ce1 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=z9hG4bK326e6e2c4B6AF299;received=192.168.10.218 From: "Test User 1" ;tag=87231757-D037E2DA To: ;tag=as22ade61f Call-ID: 986fbcf3-3fd7ab75-6e6a61e8@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> -- Nobody picked up in 5000 ms Reliably Transmitting (no NAT) to 192.168.10.186:5060: CANCEL sip:235@192.168.10.186:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK7fb72f6a;rport From: "Test User 1" ;tag=as3e2c93e1 To: Contact: Call-ID: 4e1a2d99019a82203eedde5039d83e72@192.168.10.24 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '4e1a2d99019a82203eedde5039d83e72@192.168.10.24' in 32000 ms (Method: INVITE) -- Executing VoiceMail("SIP/547-2c0f", "235|u") in new stack We're at 192.168.10.24 port 11090 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.10.218:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.218:5060;branch=z9hG4bK326e6e2c4B6AF299;received=192.168.10.218 From: "Test User 1" ;tag=87231757-D037E2DA To: ;tag=as22ade61f Call-ID: 986fbcf3-3fd7ab75-6e6a61e8@192.168.10.218 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: 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 11090 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> -- Playing 'vm-theperson' (language 'en') voipbackup*CLI> <-- SIP read from 192.168.10.186:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK7fb72f6a;rport From: "Test User 1" ;tag=as3e2c93e1 To: CSeq: 102 CANCEL Call-ID: 4e1a2d99019a82203eedde5039d83e72@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/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK7fb72f6a;rport From: "Test User 1" ;tag=as3e2c93e1 To: ;tag=C6AC3E23-B3FEEA7C CSeq: 102 INVITE Call-ID: 4e1a2d99019a82203eedde5039d83e72@192.168.10.24 Contact: User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.3.0067 Content-Length: 0 --- (9 headers 0 lines)--- 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=z9hG4bK7fb72f6a;rport From: "Test User 1" ;tag=as3e2c93e1 To: ;tag=C6AC3E23-B3FEEA7C Contact: Call-ID: 4e1a2d99019a82203eedde5039d83e72@192.168.10.24 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '4e1a2d99019a82203eedde5039d83e72@192.168.10.24' Method: INVITE voipbackup*CLI> <-- SIP read from 192.168.10.218:5060: ACK sip:235@192.168.10.24 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.218:5060;branch=z9hG4bK72e99f3dD62E0262 From: "Test User 1" ;tag=87231757-D037E2DA To: ;tag=as22ade61f CSeq: 2 ACK Call-ID: 986fbcf3-3fd7ab75-6e6a61e8@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 Max-Forwards: 70 Content-Length: 0 voipbackup*CLI> --- (11 headers 0 lines)--- voipbackup*CLI> -- Playing 'digits/2' (language 'en') voipbackup*CLI> <-- SIP read from 192.168.10.218:5060: BYE sip:235@192.168.10.24 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.218:5060;branch=z9hG4bK6ee349fD3034B34 From: "Test User 1" ;tag=87231757-D037E2DA To: ;tag=as22ade61f CSeq: 3 BYE Call-ID: 986fbcf3-3fd7ab75-6e6a61e8@192.168.10.218 Contact: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067 Proxy-Authorization: Digest username="547", realm="asterisk", nonce="4186d326", uri="sip:235@voipbackup.testdomain.com:5060;user=phone", response="e6a5e42dd256ccc5774148b3c139839d", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- Sending to 192.168.10.218 : 5060 (no NAT) Transmitting (no NAT) to 192.168.10.218:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.218:5060;branch=z9hG4bK6ee349fD3034B34;received=192.168.10.218 From: "Test User 1" ;tag=87231757-D037E2DA To: ;tag=as22ade61f Call-ID: 986fbcf3-3fd7ab75-6e6a61e8@192.168.10.218 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- voipbackup*CLI> == Spawn extension (default, 235, 3) exited non-zero on 'SIP/547-2c0f' voipbackup*CLI> Really destroying SIP dialog '986fbcf3-3fd7ab75-6e6a61e8@192.168.10.218' Method: BYE voipbackup*CLI> eexit ]0;root@voipbackup:/etc/asterisk[root@voipbackup asterisk]# exit logout