<-- SIP read from 192.168.1.1:5060: INVITE sip:%231@192.168.1.16 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bK3a2f9b2aB8CC5E37 From: "5001" ;tag=6AFD29E0-6F43A9DB To: CSeq: 1 INVITE Call-ID: e7e8415c-4cdd9aee-b2a9bfd9@192.168.1.54 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1155028526 1155028526 IN IP4 192.168.1.54 s=Polycom IP Phone c=IN IP4 192.168.1.54 t=0 0 a=sendrecv m=audio 2240 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 --- (14 headers 11 lines)--- Sending to 192.168.1.54 : 5060 (no NAT) Using INVITE request as basis request - e7e8415c-4cdd9aee-b2a9bfd9@192.168.1.54 Reliably Transmitting (no NAT) to 192.168.1.54:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bK3a2f9b2aB8CC5E37;received=192.168.1.1 From: "5001" ;tag=6AFD29E0-6F43A9DB To: ;tag=as4e3a4877 Call-ID: e7e8415c-4cdd9aee-b2a9bfd9@192.168.1.54 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="33d61b8a" Content-Length: 0 --- Scheduling destruction of SIP dialog 'e7e8415c-4cdd9aee-b2a9bfd9@192.168.1.54' in 32000 ms (Method: INVITE) Found user '5001' NCMS-IVR2*CLI> <-- SIP read from 192.168.1.1:5060: ACK sip:%231@192.168.1.16 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bK3a2f9b2aB8CC5E37 From: "5001" ;tag=6AFD29E0-6F43A9DB To: ;tag=as4e3a4877 CSeq: 1 ACK Call-ID: e7e8415c-4cdd9aee-b2a9bfd9@192.168.1.54 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- NCMS-IVR2*CLI> <-- SIP read from 192.168.1.1:5060: INVITE sip:%231@192.168.1.16 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bK22d08e3dA556D032 From: "5001" ;tag=6AFD29E0-6F43A9DB To: CSeq: 2 INVITE Call-ID: e7e8415c-4cdd9aee-b2a9bfd9@192.168.1.54 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="5001", realm="asterisk", nonce="33d61b8a", uri="sip:%231@192.168.1.16", response="e2ff0e11c18ffc8071470adf4b3925a1", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1155028526 1155028526 IN IP4 192.168.1.54 s=Polycom IP Phone c=IN IP4 192.168.1.54 t=0 0 a=sendrecv m=audio 2240 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 192.168.1.54 : 5060 (no NAT) Using INVITE request as basis request - e7e8415c-4cdd9aee-b2a9bfd9@192.168.1.54 Found user '5001' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.54:2240 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/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 192.168.1.54:2240 Looking for %231 in from-internal (domain 192.168.1.16) Reliably Transmitting (no NAT) to 192.168.1.54:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bK22d08e3dA556D032;received=192.168.1.1 From: "5001" ;tag=6AFD29E0-6F43A9DB To: ;tag=as4e3a4877 Call-ID: e7e8415c-4cdd9aee-b2a9bfd9@192.168.1.54 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- NCMS-IVR2*CLI> <-- SIP read from 192.168.1.1:5060: ACK sip:%231@192.168.1.16 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bK22d08e3dA556D032 From: "5001" ;tag=6AFD29E0-6F43A9DB To: ;tag=as4e3a4877 CSeq: 2 ACK Call-ID: e7e8415c-4cdd9aee-b2a9bfd9@192.168.1.54 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Proxy-Authorization: Digest username="5001", realm="asterisk", nonce="33d61b8a", uri="sip:%231@192.168.1.16", response="e2ff0e11c18ffc8071470adf4b3925a1", algorithm=MD5 Max-Forwards: 70 Content-Length: 0