<-- SIP read from x.x.x.x:5060: INVITE sip:199@y.y.y.y:1030 SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK521f8c6a;rport From: "cftest" ;tag=as17a89309 To: Contact: Call-ID: 710596a15272a4161a3ae91568954319@x.x.x.x CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 08 Sep 2005 23:47:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 241 v=0 o=root 30921 30921 IN IP4 x.x.x.x s=session c=IN IP4 x.x.x.x t=0 0 m=audio 6876 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (12 headers 11 lines)--- Using INVITE request as basis request - 710596a15272a4161a3ae91568954319@x.x.x.x Sending to x.x.x.x : 5060 (non-NAT) Reliably Transmitting (no NAT) to x.x.x.x:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK521f8c6a From: "cftest" ;tag=as17a89309 To: ;tag=as30a3da9f Call-ID: 710596a15272a4161a3ae91568954319@x.x.x.x CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="5f3ba829" Content-Length: 0 --- Scheduling destruction of call '710596a15272a4161a3ae91568954319@x.x.x.x' in 15000 ms Found user '5520' pbx*CLI> <-- SIP read from x.x.x.x:5060: ACK sip:199@y.y.y.y:1030 SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK521f8c6a;rport From: "cftest" ;tag=as17a89309 To: ;tag=as30a3da9f Contact: Call-ID: 710596a15272a4161a3ae91568954319@x.x.x.x CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- (9 headers 0 lines)--- pbx*CLI> <-- SIP read from x.x.x.x:5060: INVITE sip:199@192.168.24.50 SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK25ffcf5d;rport From: "cftest" ;tag=as17a89309 To: Contact: Call-ID: 710596a15272a4161a3ae91568954319@x.x.x.x CSeq: 103 INVITE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="5520", realm="asterisk", algorithm=MD5, uri="sip:199@192.168.24.50", nonce="5f3ba829", response="c6b73a7db1051fc24eaf14c3ab0a67db", opaque="" Date: Thu, 08 Sep 2005 23:47:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 241 v=0 o=root 30921 30922 IN IP4 x.x.x.x s=session c=IN IP4 x.x.x.x t=0 0 m=audio 6876 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (13 headers 11 lines)--- Using INVITE request as basis request - 710596a15272a4161a3ae91568954319@x.x.x.x Sending to x.x.x.x : 5060 (non-NAT) Found user '5520' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port x.x.x.x:6876 Found description format G729 Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 199 in gmm-internal list_route: hop: Transmitting (no NAT) to x.x.x.x:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK25ffcf5d From: "cftest" ;tag=as17a89309 To: Call-ID: 710596a15272a4161a3ae91568954319@x.x.x.x CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 ---