aphrodite*CLI> sip debug SIP Debugging enabled The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. aphrodite*CLI> <--- SIP read from 192.168.0.101:5060 ---> INVITE sip:3344439@192.168.0.8:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bKab540309B5DA95CA From: "110" ;tag=F944654F-A24DF74E To: CSeq: 1 INVITE Call-ID: 8cc09acb-d4533b8d-2697220c@192.168.0.101 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1183752770 1183752770 IN IP4 192.168.0.101 s=Polycom IP Phone c=IN IP4 192.168.0.101 t=0 0 a=sendrecv m=audio 2234 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 192.168.0.101 : 5060 (no NAT) Using INVITE request as basis request - 8cc09acb-d4533b8d-2697220c@192.168.0.101 <--- Reliably Transmitting (NAT) to 192.168.0.101:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bKab540309B5DA95CA;received=192.168.0.101 From: "110" ;tag=F944654F-A24DF74E To: ;tag=as253fd41a Call-ID: 8cc09acb-d4533b8d-2697220c@192.168.0.101 Seq: 1 INVITE> User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1d83f855" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '8cc09acb-d4533b8d-2697220c@192.168.0.101' in 32000 ms (Method: INVITE) Found user '110' aphrodite*CLI> <--- SIP read from 192.168.0.101:5060 ---> ACK sip:3344439@192.168.0.8:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bKab540309B5DA95CA From: "110" ;tag=F944654F-A24DF74E To: ;tag=as253fd41a CSeq: 1 ACK Call-ID: 8cc09acb-d4533b8d-2697220c@192.168.0.101 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- aphrodite*CLI> <--- SIP read from 192.168.0.101:5060 ---> INVITE sip:3344439@192.168.0.8:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bK7f579590390C9811 From: "110" ;tag=F944654F-A24DF74E To: CSeq: 2 INVITE Call-ID: 8cc09acb-d4533b8d-2697220c@192.168.0.101 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 Supported: 100rel,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="110", realm="asterisk", nonce="1d83f855", uri="sip:3344439@192.168.0.8:5060;user=phone", response="8aecf0c1671a5f150d1d54867121a04b", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1183752770 1183752770 IN IP4 192.168.0.101 s=Polycom IP Phone c=IN IP4 192.168.0.101 t=0 0 a=sendrecv m=audio 2234 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 192.168.0.101 : 5060 (NAT) Using INVITE request as basis request - 8cc09acb-d4533b8d-2697220c@192.168.0.101 Found user '110' 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 192.168.0.101:2234 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 - 0x8000e (gsm|ulaw|alaw|h263), 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 192.168.0.101:2234 Looking for 3344439 in downstairs (domain 192.168.0.8) <--- Reliably Transmitting (NAT) to 192.168.0.101:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bK7f579590390C9811;received=192.168.0.101 From: "110" ;tag=F944654F-A24DF74E To: ;tag=as253fd41a Call-ID: 8cc09acb-d4533b8d-2697220c@192.168.0.101 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Scheduling destruction of SIP dialog '8cc09acb-d4533b8d-2697220c@192.168.0.101' in 32000 ms (Method: INVITE) aphrodite*CLI> <--- SIP read from 192.168.0.101:5060 ---> ACK sip:3344439@192.168.0.8:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bK7f579590390C9811 From: "110" ;tag=F944654F-A24DF74E To: ;tag=as253fd41a CSeq: 2 ACK Call-ID: 8cc09acb-d4533b8d-2697220c@192.168.0.101 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- aphrodite*CLI>