daffy*CLI> sip debug SIP Debugging re-enabled daffy*CLI> <-- SIP read from 10.0.1.242:49912: INVITE sip:999@10.0.1.253 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.242:5060;branch=z9hG4bKeed39952 From: "Brian West" ;tag=0015faeab9540875dbb9056d-667cb534 To: Call-ID: 0015faea-b9540143-90df8cb7-8d5f4b06@10.0.1.242 Max-Forwards: 70 Date: Thu, 28 Sep 2006 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7970G/8.0 Contact: Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: "Brian West" ;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Allow-Events: kpml,dialog Content-Length: 269 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 23056 0 IN IP4 10.0.1.242 s=SIP Call t=0 0 m=audio 19848 RTP/AVP 0 8 18 101 c=IN IP4 10.0.1.242 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/0 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (19 headers 13 lines)--- Using INVITE request as basis request - 0015faea-b9540143-90df8cb7-8d5f4b06@10.0.1.242 Sending to 10.0.1.242 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 10.0.1.242:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.1.242:5060;branch=z9hG4bKeed39952;received=10.0.1.242 From: "Brian West" ;tag=0015faeab9540875dbb9056d-667cb534 To: ;tag=as38e68169 Call-ID: 0015faea-b9540143-90df8cb7-8d5f4b06@10.0.1.242 CSeq: 101 INVITE User-Agent: bkw.org/1.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="7a9b4749" Content-Length: 0 --- Scheduling destruction of call '0015faea-b9540143-90df8cb7-8d5f4b06@10.0.1.242' in 15000 ms Found user '1000' daffy*CLI> <-- SIP read from 10.0.1.242:49913: ACK sip:999@10.0.1.253 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.242:5060;branch=z9hG4bKeed39952 From: "Brian West" ;tag=0015faeab9540875dbb9056d-667cb534 To: ;tag=as38e68169 Call-ID: 0015faea-b9540143-90df8cb7-8d5f4b06@10.0.1.242 Date: Thu, 28 Sep 2006 GMT CSeq: 101 ACK Content-Length: 0 --- (8 headers 0 lines)--- daffy*CLI> <-- SIP read from 10.0.1.242:49914: INVITE sip:999@10.0.1.253 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.242:5060;branch=z9hG4bKef44bd73 From: "Brian West" ;tag=0015faeab9540875dbb9056d-667cb534 To: Call-ID: 0015faea-b9540143-90df8cb7-8d5f4b06@10.0.1.242 Max-Forwards: 70 Date: Thu, 28 Sep 2006 GMT CSeq: 102 INVITE User-Agent: Cisco-CP7970G/8.0 Contact: Proxy-Authorization: Digest username="1000",realm="asterisk",uri="sip:999@10.0.1.253",response="ad38ca9bcb573ab2995aff00db2f00e6",nonce="7a9b4749",algorithm=md5 Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: "Brian West" ;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Allow-Events: kpml,dialog Content-Length: 269 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 23056 0 IN IP4 10.0.1.242 s=SIP Call t=0 0 m=audio 19848 RTP/AVP 0 8 18 101 c=IN IP4 10.0.1.242 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/0 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (20 headers 13 lines)--- Using INVITE request as basis request - 0015faea-b9540143-90df8cb7-8d5f4b06@10.0.1.242 Sending to 10.0.1.242 : 5060 (non-NAT) Found user '1000' 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 10.0.1.242:19848 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 999 in default (domain 10.0.1.253) list_route: hop: Transmitting (no NAT) to 10.0.1.242:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.1.242:5060;branch=z9hG4bKef44bd73;received=10.0.1.242 From: "Brian West" ;tag=0015faeab9540875dbb9056d-667cb534 To: Call-ID: 0015faea-b9540143-90df8cb7-8d5f4b06@10.0.1.242 CSeq: 102 INVITE User-Agent: bkw.org/1.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Answer("SIP/1000-03cf", "") in new stack We're at 10.0.1.253 port 31830 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.1.242:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.1.242:5060;branch=z9hG4bKef44bd73;received=10.0.1.242 From: "Brian West" ;tag=0015faeab9540875dbb9056d-667cb534 To: ;tag=as38ff4b9c Call-ID: 0015faea-b9540143-90df8cb7-8d5f4b06@10.0.1.242 CSeq: 102 INVITE User-Agent: bkw.org/1.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 257 v=0 o=root 2650 2650 IN IP4 10.0.1.253 s=session c=IN IP4 10.0.1.253 t=0 0 m=audio 31830 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing MusicOnHold("SIP/1000-03cf", "default") in new stack -- Started music on hold, class 'default', on channel 'SIP/1000-03cf' daffy*CLI> <-- SIP read from 10.0.1.242:49915: ACK sip:999@10.0.1.253:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.242:5060;branch=z9hG4bKed0aea40 From: "Brian West" ;tag=0015faeab9540875dbb9056d-667cb534 To: ;tag=as38ff4b9c Call-ID: 0015faea-b9540143-90df8cb7-8d5f4b06@10.0.1.242 Max-Forwards: 70 Date: Thu, 28 Sep 2006 GMT CSeq: 102 ACK User-Agent: Cisco-CP7970G/8.0 Proxy-Authorization: Digest username="1000",realm="asterisk",uri="sip:999@10.0.1.253",response="ad38ca9bcb573ab2995aff00db2f00e6",nonce="7a9b4749",algorithm=md5 Remote-Party-ID: "Brian West" ;party=calling;id-type=subscriber;privacy=off;screen=yes Content-Length: 0 --- (12 headers 0 lines)--- daffy*CLI>