asteriskpbx*CLI> <--- SIP read from 192.168.69.51:5060 ---> INVITE sip:4814900@192.168.69.1:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.69.51;branch=z9hG4bK749b7fb7D72F3196 From: "Shawn Byrne" ;tag=81117709-3AC186A2 To: CSeq: 1 INVITE Call-ID: bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/2.1.2.0078 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 978314410 978314410 IN IP4 192.168.69.51 s=Polycom IP Phone c=IN IP4 192.168.69.51 t=0 0 m=audio 2236 RTP/AVP 0 8 18 101 a=sendrecv 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 192.168.69.51 : 5060 (no NAT) Using INVITE request as basis request - bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51 Found peer '201' <--- Reliably Transmitting (NAT) to 192.168.69.51:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.69.51;branch=z9hG4bK749b7fb7D72F3196;received=192.168.69.51 From: "Shawn Byrne" ;tag=81117709-3AC186A2 To: ;tag=as721870ea Call-ID: bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Access-URL: Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3d40a7f2" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51' in 32000 ms (Method: INVITE) asteriskpbx*CLI> <--- SIP read from 192.168.69.51:5060 ---> ACK sip:4814900@192.168.69.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.69.51;branch=z9hG4bK749b7fb7D72F3196 From: "Shawn Byrne" ;tag=81117709-3AC186A2 To: ;tag=as721870ea CSeq: 1 ACK Call-ID: bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/2.1.2.0078 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- asteriskpbx*CLI> <--- SIP read from 192.168.69.51:5060 ---> INVITE sip:4814900@192.168.69.1:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.69.51;branch=z9hG4bKfe8f23985460AC0F From: "Shawn Byrne" ;tag=81117709-3AC186A2 To: CSeq: 2 INVITE Call-ID: bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/2.1.2.0078 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="201", realm="asterisk", nonce="3d40a7f2", uri="sip:4814900@192.168.69.1:5060;user=phone", response="208e082dbd129c41e9e27c14f73e2e19", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 978314410 978314410 IN IP4 192.168.69.51 s=Polycom IP Phone c=IN IP4 192.168.69.51 t=0 0 m=audio 2236 RTP/AVP 0 8 18 101 a=sendrecv 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 192.168.69.51 : 5060 (NAT) Using INVITE request as basis request - bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51 Found peer '201' 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 192.168.69.51:2236 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 - 0x80e (gsm|ulaw|alaw|g726), 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 192.168.69.51:2236 Looking for 4814900 in numberplan-custom-1 (domain 192.168.69.1) <--- Reliably Transmitting (NAT) to 192.168.69.51:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.69.51;branch=z9hG4bKfe8f23985460AC0F;received=192.168.69.51 From: "Shawn Byrne" ;tag=81117709-3AC186A2 To: ;tag=as721870ea Call-ID: bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Access-URL: Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51' in 32000 ms (Method: INVITE) asteriskpbx*CLI> <--- SIP read from 192.168.69.51:5060 ---> ACK sip:4814900@192.168.69.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.69.51;branch=z9hG4bKfe8f23985460AC0F From: "Shawn Byrne" ;tag=81117709-3AC186A2 To: ;tag=as721870ea CSeq: 2 ACK Call-ID: bb3bb6dd-cb9adca3-e5163d4c@192.168.69.51 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/2.1.2.0078 Proxy-Authorization: Digest username="201", realm="asterisk", nonce="3d40a7f2", uri="sip:4814900@192.168.69.1:5060;user=phone", response="208e082dbd129c41e9e27c14f73e2e19", algorithm=MD5 Max-Forwards: 70 Content-Length: 0