*CLI> sip list peers Name/username Host Dyn Nat ACL Port Status 202 10.0.6.150 D 5060 Unmonitored 201 10.0.6.198 D 5060 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] *CLI> sip show peer 201 * Name : 201 Secret : MD5Secret : Context : from-sip-201 Subscr.Cont. : Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : -1 Call limit : 0 Dynamic : Yes Callerid : "" <> MaxCallBR : 384 kbps Expire : 57 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No T38 pt RTP : No T38 pt TCP : No CanReinvite : Yes PromiscRedir : No User=Phone : No Video Support: Yes Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 10.0.6.198 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Def. Username: SIP Options : (none) Codecs : 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264) Codec Order : (none) Status : Unmonitored Useragent : Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24 Reg. Contact : sip:10.0.6.198 *CLI> set verbose 9 Verbosity was 3 and is now 9 The 'set verbose' command is deprecated and will be removed in a future release. Please use 'core verbose' instead. *CLI> core verbose 9 Verbosity is at least 9 *CLI> core debug 9 Core debug was 0 and is now 9 *CLI> sip debug SIP Debugging enabled *CLI> <--- SIP read from 10.0.6.198:5060 ---> INVITE sip:202@10.0.6.198 SIP/2.0 Via: SIP/2.0/UDP 10.0.6.198;branch=z9hG4bKa052b6d5 Max-Forwards: 70 From: ;epid=82042503F512B1;tag=ec34af13 To: Call-ID: 209bd28f@10.0.6.198 CSeq: 1 INVITE User-Agent: Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24 Contact: Content-Type: application/sdp Content-Length: 888 v=0 o=Ursys2 1468211940 0 IN IP4 10.0.6.198 s=- c=IN IP4 10.0.6.198 b=AS:128 t=0 0 m=audio 49168 RTP/AVP 99 98 97 102 101 103 9 15 0 8 18 a=rtpmap:99 SIREN14/16000 a=fmtp:99 bitrate=48000 a=rtpmap:98 SIREN14/16000 a=fmtp:98 bitrate=32000 a=rtpmap:97 SIREN14/16000 a=fmtp:97 bitrate=24000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:101 G7221/16000 a=fmtp:101 bitrate=24000 a=rtpmap:103 G7221/16000 a=fmtp:103 bitrate=16000 a=rtpmap:9 G722/8000 a=rtpmap:15 G728/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729A/8000 m=video 49170 RTP/AVP 109 34 96 31 b=TIAS:128000 a=rtpmap:109 H264/90000 a=fmtp:109 profile-level-id=42800c max-mbps=10000 a=rtpmap:34 H263/90000 a=rtpmap:96 H263-1998/90000 a=fmtp:96 SQCIF=1 QCIF=1 CIF=1 CIF4=2 F J T a=rtpmap:31 H261/90000 a=fmtp:31 CIF=1 QCIF=1 m=data 49172 RTP/AVP 100 a=rtpmap:100 H224 <-------------> --- (11 headers 35 lines) --- Sending to 10.0.6.198 : 5060 (no NAT) Using INVITE request as basis request - 209bd28f@10.0.6.198 Found user '201' Found RTP audio format 99 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 102 Found RTP audio format 101 Found RTP audio format 103 Found RTP audio format 9 Found RTP audio format 15 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP video format 109 Found RTP video format 34 Found RTP video format 96 Found RTP video format 31 [Nov 14 09:22:37] WARNING[11736]: chan_sip.c:4691 process_sdp: Unsupported SDP media type in offer: data 49172 RTP/AVP 100 Peer audio RTP is at port 10.0.6.198:49168 Found description format SIREN14 for ID 99 Got unsupported a:fmtp in SDP offer Found description format SIREN14 for ID 98 Got unsupported a:fmtp in SDP offer Found description format SIREN14 for ID 97 Got unsupported a:fmtp in SDP offer Found description format G7221 for ID 102 Got unsupported a:fmtp in SDP offer Found description format G7221 for ID 101 Got unsupported a:fmtp in SDP offer Found description format G7221 for ID 103 Got unsupported a:fmtp in SDP offer Found description format G722 for ID 9 Found description format G728 for ID 15 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G729A for ID 18 Found description format H264 for ID 109 Got unsupported a:fmtp in SDP offer Found description format H263 for ID 34 Found description format H263-1998 for ID 96 Got unsupported a:fmtp in SDP offer Found description format H261 for ID 31 Got unsupported a:fmtp in SDP offer Found description format H224 for ID 100 Capabilities: us - 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264), peer - audio=0x3c050c (ulaw|alaw|g729|ilbc|h261|h263|h263p|h264)/video=0x3c0000 (h261|h263|h263p|h264), combined - 0x3c050c (ulaw|alaw|g729|ilbc|h261|h263|h263p|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.0.6.198:49168 Peer video RTP is at port 10.0.6.198:49170 Looking for 202 in from-sip-201 (domain 10.0.6.198) list_route: hop: <--- Transmitting (no NAT) to 10.0.6.198:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.6.198;branch=z9hG4bKa052b6d5;received=10.0.6.198 From: ;epid=82042503F512B1;tag=ec34af13 To: Call-ID: 209bd28f@10.0.6.198 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [202@from-sip-201:1] AGI("SIP/201-081e1340", "/etc/asterisk/setcid.agi|201") in new stack -- Launched AGI Script /etc/asterisk/setcid.agi == /etc/asterisk/setcid.agi|201: Failed to execute '/etc/asterisk/setcid.agi': No such file or directory -- AGI Script /etc/asterisk/setcid.agi completed, returning 0 -- Executing [202@from-sip-201:2] Goto("SIP/201-081e1340", "from-sip-cid-201|202|1") in new stack -- Goto (from-sip-cid-201,202,1) -- Executing [202@from-sip-cid-201:1] Macro("SIP/201-081e1340", "general-dial|SIP/202") in new stack -- Executing [s@macro-general-dial:1] Dial("SIP/201-081e1340", "SIP/202|20") in new stack Video is at 10.0.6.11 port 12394 Audio is at 10.0.6.11 port 11736 Adding codec 0x4 (ulaw) to SDP Adding codec 0x1 (g723) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x10 (g726aal2) to SDP Adding codec 0x20 (adpcm) to SDP Adding codec 0x40 (slin) to SDP Adding codec 0x80 (lpc10) to SDP Adding codec 0x100 (g729) to SDP Adding codec 0x200 (speex) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x800 (g726) to SDP Adding codec 0x40000 (h261) to SDP Adding codec 0x80000 (h263) to SDP Adding codec 0x100000 (h263p) to SDP Adding codec 0x200000 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.6.150:5060: INVITE sip:10.0.6.150 SIP/2.0 Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK2abc3e4a;rport From: "201" ;tag=as79c5ae7b To: Contact: Call-ID: 7019698d5791958b1c1812be36c3dfcc@10.0.6.11 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 13 Nov 2006 22:22:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 715 v=0 o=root 11716 11716 IN IP4 10.0.6.198 s=session c=IN IP4 10.0.6.198 b=CT:384 t=0 0 m=audio 49168 RTP/AVP 0 4 3 8 112 5 10 7 18 110 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv m=video 49170 RTP/AVP 31 34 103 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:103 h263-1998/90000 a=rtpmap:99 H264/90000 a=sendrecv --- -- Called 202 <--- SIP read from 10.0.6.150:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK2abc3e4a;rport From: "201";tag=as79c5ae7b To: ;tag=8548d49a Call-ID: 7019698d5791958b1c1812be36c3dfcc@10.0.6.11 CSeq: 102 INVITE Contact: User-Agent: Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/202-081e7ca0 is ringing <--- Transmitting (no NAT) to 10.0.6.198:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.6.198;branch=z9hG4bKa052b6d5;received=10.0.6.198 From: ;epid=82042503F512B1;tag=ec34af13 To: ;tag=as2593d0f5 Call-ID: 209bd28f@10.0.6.198 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- SIP read from 10.0.6.150:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK2abc3e4a;rport From: "201";tag=as79c5ae7b To: ;tag=8548d49a Call-ID: 7019698d5791958b1c1812be36c3dfcc@10.0.6.11 CSeq: 102 INVITE Contact: User-Agent: Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24 Content-Length: 425 Content-Type: application/sdp v=0 o=URSYS 701409664 0 IN IP4 10.0.6.150 s=- c=IN IP4 10.0.6.150 b=AS:128 t=0 0 m=audio 49160 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 m=video 49162 RTP/AVP 31 34 96 109 b=TIAS:128000 a=rtpmap:31 H261/90000 a=fmtp:31 CIF=1 QCIF=1 a=rtpmap:34 H263/90000 a=rtpmap:96 H263-1998/90000 a=fmtp:96 SQCIF=1 QCIF=1 CIF=1 CIF4=2 a=rtpmap:109 H264/90000 a=fmtp:109 profile-level-id=42800c max-mbps=10000 <-------------> --- (10 headers 18 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP video format 31 Found RTP video format 34 Found RTP video format 96 Found RTP video format 109 Peer audio RTP is at port 10.0.6.150:49160 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format H261 for ID 31 Got unsupported a:fmtp in SDP offer Found description format H263 for ID 34 Found description format H263-1998 for ID 96 Got unsupported a:fmtp in SDP offer Found description format H264 for ID 109 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264), peer - audio=0x3c000c (ulaw|alaw|h261|h263|h263p|h264)/video=0x3c0000 (h261|h263|h263p|h264), combined - 0x3c000c (ulaw|alaw|h261|h263|h263p|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.0.6.150:49160 Peer video RTP is at port 10.0.6.150:49162 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.6.150, port 5060 Transmitting (no NAT) to 10.0.6.150:5060: ACK sip:10.0.6.150 SIP/2.0 Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK66b06791;rport From: "201" ;tag=as79c5ae7b To: ;tag=8548d49a Contact: Call-ID: 7019698d5791958b1c1812be36c3dfcc@10.0.6.11 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/202-081e7ca0 answered SIP/201-081e1340 Video is at 10.0.6.11 port 14306 Audio is at 10.0.6.11 port 11746 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x100 (g729) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x40000 (h261) to SDP Adding codec 0x80000 (h263) to SDP Adding codec 0x100000 (h263p) to SDP Adding codec 0x200000 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.0.6.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.6.198;branch=z9hG4bKa052b6d5;received=10.0.6.198 From: ;epid=82042503F512B1;tag=ec34af13 To: ;tag=as2593d0f5 Call-ID: 209bd28f@10.0.6.198 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 500 v=0 o=root 11716 11716 IN IP4 10.0.6.150 s=session c=IN IP4 10.0.6.150 b=CT:384 t=0 0 m=audio 49160 RTP/AVP 0 8 18 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv m=video 49162 RTP/AVP 31 34 103 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:103 h263-1998/90000 a=rtpmap:99 H264/90000 a=sendrecv <------------> -- Native bridging SIP/201-081e1340 and SIP/202-081e7ca0 Retransmitting #1 (no NAT) to 10.0.6.198:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.6.198;branch=z9hG4bKa052b6d5;received=10.0.6.198 From: ;epid=82042503F512B1;tag=ec34af13 To: ;tag=as2593d0f5 Call-ID: 209bd28f@10.0.6.198 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 500 v=0 o=root 11716 11716 IN IP4 10.0.6.150 s=session c=IN IP4 10.0.6.150 b=CT:384 t=0 0 m=audio 49160 RTP/AVP 0 8 18 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv m=video 49162 RTP/AVP 31 34 103 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:103 h263-1998/90000 a=rtpmap:99 H264/90000 a=sendrecv --- <--- SIP read from 10.0.6.198:5060 ---> ACK sip:10.0.6.198 SIP/2.0 Via: SIP/2.0/UDP 10.0.6.198;branch=z9hG4bKa052b6d5 Max-Forwards: 70 From: ;epid=82042503F512B1;tag=ec34af13 To: ;tag=as2593d0f5 Call-ID: 209bd28f@10.0.6.198 CSeq: 1 ACK Contact: Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 10.0.6.198:5060 ---> REGISTER sip:10.0.6.11 SIP/2.0 Via: SIP/2.0/UDP 10.0.6.198 Max-Forwards: 70 From: ;epid=82042503F512B1 To: Call-ID: 448690996@10.0.6.198 CSeq: 482 REGISTER User-Agent: Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24 Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 10.0.6.198 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.0.6.198:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.6.198;received=10.0.6.198 From: ;epid=82042503F512B1 To: Call-ID: 448690996@10.0.6.198 CSeq: 482 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 10.0.6.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.6.198;received=10.0.6.198 From: ;epid=82042503F512B1 To: ;tag=as21d351c8 Call-ID: 448690996@10.0.6.198 CSeq: 482 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 120 Contact: ;expires=120 Date: Mon, 13 Nov 2006 22:22:45 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '448690996@10.0.6.198' in 32000 ms (Method: REGISTER) <--- SIP read from 10.0.6.198:5060 ---> BYE sip:202@10.0.6.198 SIP/2.0 Via: SIP/2.0/UDP 10.0.6.198;branch=z9hG4bK595bd6ab Max-Forwards: 70 From: ;epid=82042503F512B1;tag=ec34af13 To: ;tag=as2593d0f5 Call-ID: 209bd28f@10.0.6.198 CSeq: 2 BYE User-Agent: Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24 Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.0.6.198 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.0.6.198:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.6.198;branch=z9hG4bK595bd6ab;received=10.0.6.198 From: ;epid=82042503F512B1;tag=ec34af13 To: ;tag=as2593d0f5 Call-ID: 209bd28f@10.0.6.198 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.6.150, port 5060 Video is at 10.0.6.11 port 12394 Audio is at 10.0.6.11 port 11736 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x40000 (h261) to SDP Adding codec 0x80000 (h263) to SDP Adding codec 0x100000 (h263p) to SDP Adding codec 0x200000 (h264) to SDP Reliably Transmitting (no NAT) to 10.0.6.150:5060: INVITE sip:10.0.6.150 SIP/2.0 Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK59b84b97;rport From: "201" ;tag=as79c5ae7b To: ;tag=8548d49a Contact: Call-ID: 7019698d5791958b1c1812be36c3dfcc@10.0.6.11 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 350 v=0 o=root 11716 11717 IN IP4 10.0.6.11 s=session c=IN IP4 10.0.6.11 b=CT:384 t=0 0 m=audio 11736 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=sendrecv m=video 12394 RTP/AVP 31 34 103 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:103 h263-1998/90000 a=rtpmap:99 H264/90000 a=sendrecv --- Scheduling destruction of SIP dialog '7019698d5791958b1c1812be36c3dfcc@10.0.6.11' in 32000 ms (Method: INVITE) == Spawn extension (macro-general-dial, s, 1) exited non-zero on 'SIP/201-081e1340' -- Executing [h@macro-general-dial:1] Hangup("SIP/201-081e1340", "") in new stack == Spawn extension (macro-general-dial, h, 1) exited non-zero on 'SIP/201-081e1340' <--- SIP read from 10.0.6.150:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK59b84b97;rport From: "201";tag=as79c5ae7b To: ;tag=8548d49a Call-ID: 7019698d5791958b1c1812be36c3dfcc@10.0.6.11 CSeq: 103 INVITE Contact: User-Agent: Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24 Content-Length: 422 Content-Type: application/sdp v=0 o=URSYS 1279946760 0 IN IP4 10.0.6.150 s=- c=IN IP4 10.0.6.150 b=AS:128 t=0 0 m=audio 49160 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 m=video 0 RTP/AVP 31 34 96 109 b=TIAS:128000 a=rtpmap:31 H261/90000 a=fmtp:31 CIF=1 QCIF=1 a=rtpmap:34 H263/90000 a=rtpmap:96 H263-1998/90000 a=fmtp:96 SQCIF=1 QCIF=1 CIF=1 CIF4=2 a=rtpmap:109 H264/90000 a=fmtp:109 profile-level-id=42800c max-mbps=10000 <-------------> --- (10 headers 18 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP video format 31 Found RTP video format 34 Found RTP video format 96 Found RTP video format 109 Peer audio RTP is at port 10.0.6.150:49160 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format H261 for ID 31 Got unsupported a:fmtp in SDP offer Found description format H263 for ID 34 Found description format H263-1998 for ID 96 Got unsupported a:fmtp in SDP offer Found description format H264 for ID 109 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264), peer - audio=0x3c000c (ulaw|alaw|h261|h263|h263p|h264)/video=0x3c0000 (h261|h263|h263p|h264), combined - 0x3c000c (ulaw|alaw|h261|h263|h263p|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.0.6.150:49160 set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.6.150, port 5060 Transmitting (no NAT) to 10.0.6.150:5060: ACK sip:10.0.6.150 SIP/2.0 Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK06f3c449;rport From: "201" ;tag=as79c5ae7b To: ;tag=8548d49a Contact: Call-ID: 7019698d5791958b1c1812be36c3dfcc@10.0.6.11 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.6.150, port 5060 Reliably Transmitting (no NAT) to 10.0.6.150:5060: BYE sip:10.0.6.150 SIP/2.0 Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK2a2fe0b0;rport From: "201" ;tag=as79c5ae7b To: ;tag=8548d49a Contact: Call-ID: 7019698d5791958b1c1812be36c3dfcc@10.0.6.11 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '7019698d5791958b1c1812be36c3dfcc@10.0.6.11' in 32000 ms (Method: INVITE) Really destroying SIP dialog '209bd28f@10.0.6.198' Method: BYE <--- SIP read from 10.0.6.150:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK59b84b97;rport From: "201";tag=as79c5ae7b To: ;tag=8548d49a Call-ID: 7019698d5791958b1c1812be36c3dfcc@10.0.6.11 CSeq: 103 INVITE Contact: User-Agent: Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24 Content-Length: 422 Content-Type: application/sdp v=0 o=URSYS 1279946760 0 IN IP4 10.0.6.150 s=- c=IN IP4 10.0.6.150 b=AS:128 t=0 0 m=audio 49160 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 m=video 0 RTP/AVP 31 34 96 109 b=TIAS:128000 a=rtpmap:31 H261/90000 a=fmtp:31 CIF=1 QCIF=1 a=rtpmap:34 H263/90000 a=rtpmap:96 H263-1998/90000 a=fmtp:96 SQCIF=1 QCIF=1 CIF=1 CIF4=2 a=rtpmap:109 H264/90000 a=fmtp:109 profile-level-id=42800c max-mbps=10000 <-------------> --- (10 headers 18 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP video format 31 Found RTP video format 34 Found RTP video format 96 Found RTP video format 109 Peer audio RTP is at port 10.0.6.150:49160 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format H261 for ID 31 Got unsupported a:fmtp in SDP offer Found description format H263 for ID 34 Found description format H263-1998 for ID 96 Got unsupported a:fmtp in SDP offer Found description format H264 for ID 109 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264), peer - audio=0x3c000c (ulaw|alaw|h261|h263|h263p|h264)/video=0x3c0000 (h261|h263|h263p|h264), combined - 0x3c000c (ulaw|alaw|h261|h263|h263p|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.0.6.150:49160 set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.6.150, port 5060 Transmitting (no NAT) to 10.0.6.150:5060: ACK sip:10.0.6.150 SIP/2.0 Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK566237bd;rport From: "201" ;tag=as79c5ae7b To: ;tag=8548d49a Contact: Call-ID: 7019698d5791958b1c1812be36c3dfcc@10.0.6.11 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- <--- SIP read from 10.0.6.150:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK2a2fe0b0;rport From: "201";tag=as79c5ae7b To: ;tag=8548d49a Call-ID: 7019698d5791958b1c1812be36c3dfcc@10.0.6.11 CSeq: 104 BYE Contact: User-Agent: Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '7019698d5791958b1c1812be36c3dfcc@10.0.6.11' Method: INVITE Really destroying SIP dialog '448690996@10.0.6.198' Method: REGISTER <--- SIP read from 10.0.6.150:5060 ---> REGISTER sip:10.0.6.11 SIP/2.0 Via: SIP/2.0/UDP 10.0.6.150 Max-Forwards: 70 From: ;epid=82042503F811B1 To: Call-ID: 663860435@10.0.6.150 CSeq: 493 REGISTER User-Agent: Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24 Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 10.0.6.150 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.0.6.150:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.6.150;received=10.0.6.150 From: ;epid=82042503F811B1 To: Call-ID: 663860435@10.0.6.150 CSeq: 493 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 10.0.6.150:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.6.150;received=10.0.6.150 From: ;epid=82042503F811B1 To: ;tag=as6806220b Call-ID: 663860435@10.0.6.150 CSeq: 493 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 120 Contact: ;expires=120 Date: Mon, 13 Nov 2006 22:23:39 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '663860435@10.0.6.150' in 32000 ms (Method: REGISTER) Really destroying SIP dialog '663860435@10.0.6.150' Method: REGISTER