sip set debug peer sip1 SIP Debugging Enabled for IP: 192.168.1.21 <--- SIP read from UDP:192.168.1.21:5060 ---> INVITE sip:333@example.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.21:5060;branch=z9hG4bK079510e58 Max-Forwards: 70 Content-Length: 277 To: 333 From: sip:sip1@example.com:5060;tag=c696eaa546ef857 Call-ID: b9f227c13acff83f35e71c0b2fde5abc@192.168.1.21 CSeq: 1094661775 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Contact: Supported: replaces User-Agent: Aastra 480i Cordless/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 v=0 o=MxSIP 0 82298883 IN IP4 192.168.1.21 s=SIP Call c=IN IP4 192.168.1.21 t=0 0 m=audio 3000 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 a=fmtp:101 0-15 a=ptime:30 a=silenceSupp:on - - - - <-------------> --- (15 headers 13 lines) --- == Using UDPTL CoS mark 5 Sending to 192.168.1.21:5060 (no NAT) Using INVITE request as basis request - b9f227c13acff83f35e71c0b2fde5abc@192.168.1.21 Found peer 'sip1' for 'sip1' from 192.168.1.21:5060 <--- Reliably Transmitting (no NAT) to 192.168.1.21:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.21:5060;branch=z9hG4bK079510e58;received=192.168.1.21 From: sip:sip1@example.com:5060;tag=c696eaa546ef857 To: 333 ;tag=as3fe9c646 Call-ID: b9f227c13acff83f35e71c0b2fde5abc@192.168.1.21 CSeq: 1094661775 INVITE Server: Asterisk - example.com Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="example.com", nonce="60f848be" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'b9f227c13acff83f35e71c0b2fde5abc@192.168.1.21' in 6400 ms (Method: INVITE) <--- SIP read from UDP:192.168.1.21:5060 ---> ACK sip:333@example.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.21:5060;branch=z9hG4bK079510e58 Max-Forwards: 70 Content-Length: 0 To: 333 ;tag=as3fe9c646 From: sip:sip1@example.com:5060;tag=c696eaa546ef857 Call-ID: b9f227c13acff83f35e71c0b2fde5abc@192.168.1.21 CSeq: 1094661775 ACK User-Agent: Aastra 480i Cordless/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.1.21:5060 ---> INVITE sip:333@example.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.21:5060;branch=z9hG4bKf06e4b872 Max-Forwards: 70 Content-Length: 277 To: 333 From: sip:sip1@example.com:5060;tag=c696eaa546ef857 Call-ID: b9f227c13acff83f35e71c0b2fde5abc@192.168.1.21 CSeq: 1094661776 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: Content-Type: application/sdp Supported: replaces Authorization:Digest response="3df62228bd65f3c456a39b1f402f6554",username="sip1",realm="example.com",nonce="60f848be",algorithm=MD5,uri="sip:333@example.com:5060" User-Agent: Aastra 480i Cordless/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 v=0 o=MxSIP 0 82298883 IN IP4 192.168.1.21 s=SIP Call c=IN IP4 192.168.1.21 t=0 0 m=audio 3000 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 a=fmtp:101 0-15 a=ptime:30 a=silenceSupp:on - - - - <-------------> --- (16 headers 13 lines) --- Sending to 192.168.1.21:5060 (no NAT) Using INVITE request as basis request - b9f227c13acff83f35e71c0b2fde5abc@192.168.1.21 Found peer 'sip1' for 'sip1' from 192.168.1.21:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=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 192.168.1.21:3000 Looking for 333 in mss-chicago (domain example.com:5060) <--- Reliably Transmitting (no NAT) to 192.168.1.21:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.21:5060;branch=z9hG4bKf06e4b872;received=192.168.1.21 From: sip:sip1@example.com:5060;tag=c696eaa546ef857 To: 333 ;tag=as3fe9c646 Call-ID: b9f227c13acff83f35e71c0b2fde5abc@192.168.1.21 CSeq: 1094661776 INVITE Server: Asterisk - example.com Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [May 24 16:14:14] NOTICE[18204]: chan_sip.c:21581 handle_request_invite: Call from 'sip1' to extension '333' rejected because extension not found in context 'mss-chicago'. Scheduling destruction of SIP dialog 'b9f227c13acff83f35e71c0b2fde5abc@192.168.1.21' in 6400 ms (Method: INVITE) <--- SIP read from UDP:192.168.1.21:5060 ---> ACK sip:333@example.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.21:5060;branch=z9hG4bKf06e4b872 Max-Forwards: 70 Content-Length: 0 To: 333 ;tag=as3fe9c646 From: sip:sip1@example.com:5060;tag=c696eaa546ef857 Call-ID: b9f227c13acff83f35e71c0b2fde5abc@192.168.1.21 CSeq: 1094661776 ACK Authorization:Digest response="3df62228bd65f3c456a39b1f402f6554",username="sip1",realm="example.com",nonce="60f848be",algorithm=MD5,uri="sip:333@example.com:5060" User-Agent: Aastra 480i Cordless/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (10 headers 0 lines) ---