<--- SIP read from 192.168.1.142:5060 ---> INVITE sip:914805775268@192.168.1.20;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.142;branch=z9hG4bK3502d7e8274D2553 From: "Dan Trainor" ;tag=30F10D6-94944C67 To: CSeq: 1 INVITE Call-ID: 412b3a12-499a898c-88b504d5@192.168.1.142 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.2.0041 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 978308813 978308813 IN IP4 192.168.1.142 s=Polycom IP Phone c=IN IP4 192.168.1.142 t=0 0 a=sendrecv m=audio 2260 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 192.168.1.142 : 5060 (no NAT) Using INVITE request as basis request - 412b3a12-499a898c-88b504d5@192.168.1.142 Found peer '101' <--- Reliably Transmitting (no NAT) to 192.168.1.142:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.142;branch=z9hG4bK3502d7e8274D2553;received=192.168.1.142 From: "Dan Trainor" ;tag=30F10D6-94944C67 To: ;tag=as1018a264 Call-ID: 412b3a12-499a898c-88b504d5@192.168.1.142 CSeq: 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="26463511" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '412b3a12-499a898c-88b504d5@192.168.1.142' in 32000 ms (Method: INVITE) pbx*CLI> <--- SIP read from 192.168.1.142:5060 ---> ACK sip:914805775268@192.168.1.20 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.142;branch=z9hG4bK3502d7e8274D2553 From: "Dan Trainor" ;tag=30F10D6-94944C67 To: ;tag=as1018a264 CSeq: 1 ACK Call-ID: 412b3a12-499a898c-88b504d5@192.168.1.142 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.2.0041 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- pbx*CLI> <--- SIP read from 192.168.1.142:5060 ---> INVITE sip:914805775268@192.168.1.20;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.142;branch=z9hG4bK1319c891FB06E70 From: "Dan Trainor" ;tag=30F10D6-94944C67 To: CSeq: 2 INVITE Call-ID: 412b3a12-499a898c-88b504d5@192.168.1.142 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.2.0041 Supported: 100rel,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="101", realm="asterisk", nonce="26463511", uri="sip:914805775268@192.168.1.20;user=phone", response="08520023e3c026fac24f529a5e46879a", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 978308813 978308813 IN IP4 192.168.1.142 s=Polycom IP Phone c=IN IP4 192.168.1.142 t=0 0 a=sendrecv m=audio 2260 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 192.168.1.142 : 5060 (no NAT) Using INVITE request as basis request - 412b3a12-499a898c-88b504d5@192.168.1.142 Found peer '101' 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 192.168.1.142:2260 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 192.168.1.142:2260 Looking for 914805775268 in numberplan-custom-1 (domain 192.168.1.20) <--- Reliably Transmitting (no NAT) to 192.168.1.142:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.142;branch=z9hG4bK1319c891FB06E70;received=192.168.1.142 From: "Dan Trainor" ;tag=30F10D6-94944C67 To: ;tag=as1018a264 Call-ID: 412b3a12-499a898c-88b504d5@192.168.1.142 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 '412b3a12-499a898c-88b504d5@192.168.1.142' in 32000 ms (Method: INVITE) pbx*CLI> <--- SIP read from 192.168.1.142:5060 ---> ACK sip:914805775268@192.168.1.20 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.142;branch=z9hG4bK1319c891FB06E70 From: "Dan Trainor" ;tag=30F10D6-94944C67 To: ;tag=as1018a264 CSeq: 2 ACK Call-ID: 412b3a12-499a898c-88b504d5@192.168.1.142 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.2.0041 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- pbx*CLI>