asterisk1*CLI> --- (8 headers 0 lines)--- asterisk1*CLI> Destroying call '137c617a01c314bc7b4080c31bbe4792@10.0.10.199' asterisk1*CLI> <-- SIP read from 10.0.10.10:5060: INVITE sip:1000@10.0.10.199 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.10:5060;branch=z9hG4bKcff41fe3d454d5dd From: "600" ;tag=91401244155e7d64 To: Contact: Supported: replaces, timer Call-ID: d7bf395a9552f2c9@10.0.10.10 CSeq: 41457 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 247 v=0 o=600 8000 8000 IN IP4 10.0.10.10 s=SIP Call c=IN IP4 10.0.10.10 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 3 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=ptime:20 --- (13 headers 13 lines)--- Using INVITE request as basis request - d7bf395a9552f2c9@10.0.10.10 Sending to 10.0.10.10 : 5060 (non-NAT) Found user '600' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 3 Peer audio RTP is at port 10.0.10.10:5004 Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format GSM Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 1000 in from-internal (domain 10.0.10.199) list_route: hop: Transmitting (no NAT) to 10.0.10.10:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.10.10:5060;branch=z9hG4bKcff41fe3d454d5dd;received=10.0.10.10 From: "600" ;tag=91401244155e7d64 To: Call-ID: d7bf395a9552f2c9@10.0.10.10 CSeq: 41457 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- asterisk1*CLI> dialparties.agi: Starting New Dialparties.agi asterisk1*CLI> We're at 10.0.10.199 port 11910 asterisk1*CLI> Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 10.0.10.12:5060: INVITE sip:1000@10.0.10.12;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK23aa775b From: "600" ;tag=as047d2384 To: Contact: Call-ID: 408cb0fb2a7c5e5225457e7d7149b84c@10.0.10.199 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 06 Dec 2006 15:59:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 32510 32510 IN IP4 10.0.10.199 s=session c=IN IP4 10.0.10.199 t=0 0 m=audio 11910 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Transmitting (no NAT) to 10.0.10.10:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.10.10:5060;branch=z9hG4bKcff41fe3d454d5dd;received=10.0.10.10 From: "600" ;tag=91401244155e7d64 To: ;tag=as579bc047 Call-ID: d7bf395a9552f2c9@10.0.10.10 CSeq: 41457 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 10.0.10.12:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK23aa775b From: "600" ;tag=as047d2384 To: Call-ID: 408cb0fb2a7c5e5225457e7d7149b84c@10.0.10.199 CSeq: 102 INVITE User-Agent: Grandstream BT110 1.0.8.32 Content-Length: 0 --- (8 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 10.0.10.12:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK23aa775b From: "600" ;tag=as047d2384 To: ;tag=f33403aae779798a Call-ID: 408cb0fb2a7c5e5225457e7d7149b84c@10.0.10.199 CSeq: 102 INVITE User-Agent: Grandstream BT110 1.0.8.32 Content-Length: 0 asterisk1*CLI> --- (8 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 10.0.10.12:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK23aa775b From: "600" ;tag=as047d2384 To: ;tag=f33403aae779798a Call-ID: 408cb0fb2a7c5e5225457e7d7149b84c@10.0.10.199 CSeq: 102 INVITE User-Agent: Grandstream BT110 1.0.8.32 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Supported: replaces Content-Length: 151 v=0 o=1000 8000 8000 IN IP4 10.0.10.12 s=SIP Call c=IN IP4 10.0.10.12 t=0 0 m=audio 5004 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 --- (12 headers 9 lines)--- Found RTP audio format 0 Peer audio RTP is at port 10.0.10.12:5004 Found description format PCMU Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.10.12, port 5060 Transmitting (no NAT) to 10.0.10.12:5060: ACK sip:1000@10.0.10.12;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK414b9055 From: "600" ;tag=as047d2384 To: ;tag=f33403aae779798a Contact: Call-ID: 408cb0fb2a7c5e5225457e7d7149b84c@10.0.10.199 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterisk1*CLI> We're at 10.0.10.199 port 14748 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 10.0.10.10:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.10:5060;branch=z9hG4bKcff41fe3d454d5dd;received=10.0.10.10 From: "600" ;tag=91401244155e7d64 To: ;tag=as579bc047 Call-ID: d7bf395a9552f2c9@10.0.10.10 CSeq: 41457 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 182 v=0 o=root 32510 32510 IN IP4 10.0.10.199 s=session c=IN IP4 10.0.10.199 t=0 0 m=audio 14748 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- asterisk1*CLI> <-- SIP read from 10.0.10.10:5060: ACK sip:1000@10.0.10.199 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.10:5060;branch=z9hG4bKd62ae2f4f68e9f66 From: "600" ;tag=91401244155e7d64 To: ;tag=as579bc047 Contact: Call-ID: d7bf395a9552f2c9@10.0.10.10 CSeq: 41457 ACK User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 --- (11 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 192.168.2.204:5060: REGISTER sip:192.168.2.109 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.204;branch=z9hG4bKac415480366 Max-Forwards: 70 From: ;tag=1c415476486 To: Call-ID: 143877930311200004719@192.168.2.204 CSeq: 11348 REGISTER Contact: ;expires=180 Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 180 User-Agent: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Length: 0 --- (13 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.2.204 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.2.204:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.204;branch=z9hG4bKac415480366;received=192.168.2.204 From: ;tag=1c415476486 To: Call-ID: 143877930311200004719@192.168.2.204 CSeq: 11348 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- asterisk1*CLI> Transmitting (no NAT) to 192.168.2.204:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.204;branch=z9hG4bKac415480366;received=192.168.2.204 From: ;tag=1c415476486 To: ;tag=as3959a6b9 Call-ID: 143877930311200004719@192.168.2.204 CSeq: 11348 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 180 Contact: ;expires=180 Date: Wed, 06 Dec 2006 15:59:20 GMT Content-Length: 0 asterisk1*CLI> --- Scheduling destruction of call '143877930311200004719@192.168.2.204' in 15000 ms asterisk1*CLI> <-- SIP read from 192.168.2.204:5060: REGISTER sip:192.168.2.109 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.204;branch=z9hG4bKac415511645 Max-Forwards: 70 From: ;tag=1c415507095 To: Call-ID: 143877952111200004719@192.168.2.204 CSeq: 11346 REGISTER Contact: ;expires=180 Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 180 User-Agent: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Length: 0 --- (13 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.2.204 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.2.204:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.204;branch=z9hG4bKac415511645;received=192.168.2.204 From: ;tag=1c415507095 To: Call-ID: 143877952111200004719@192.168.2.204 CSeq: 11346 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- asterisk1*CLI> Transmitting (no NAT) to 192.168.2.204:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.204;branch=z9hG4bKac415511645;received=192.168.2.204 From: ;tag=1c415507095 To: ;tag=as442d3d0e Call-ID: 143877952111200004719@192.168.2.204 CSeq: 11346 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 180 Contact: ;expires=180 Date: Wed, 06 Dec 2006 15:59:20 GMT Content-Length: 0 asterisk1*CLI> --- Scheduling destruction of call '143877952111200004719@192.168.2.204' in 15000 ms asterisk1*CLI> Destroying call 'A6414240-C665-4FFE-9D44-EBEE13527EE1@10.0.10.16' asterisk1*CLI> <-- SIP read from 10.0.10.10:5060: INVITE sip:1000@10.0.10.199 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.10:5060;branch=z9hG4bK570861d0a723e075 From: "600" ;tag=91401244155e7d64 To: ;tag=as579bc047 Contact: Supported: replaces, timer Call-ID: d7bf395a9552f2c9@10.0.10.10 CSeq: 41458 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 247 v=0 o=600 8000 8001 IN IP4 10.0.10.10 s=SIP Call c=IN IP4 10.0.10.10 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 3 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=ptime:20 --- (13 headers 13 lines)--- Using INVITE request as basis request - d7bf395a9552f2c9@10.0.10.10 Sending to 10.0.10.10 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 3 Peer audio RTP is at port 10.0.10.10:5004 Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format GSM Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) We're at 10.0.10.199 port 14748 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 10.0.10.10:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.10:5060;branch=z9hG4bK570861d0a723e075;received=10.0.10.10 From: "600" ;tag=91401244155e7d64 To: ;tag=as579bc047 Call-ID: d7bf395a9552f2c9@10.0.10.10 CSeq: 41458 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 182 v=0 o=root 32510 32511 IN IP4 10.0.10.199 s=session c=IN IP4 10.0.10.199 t=0 0 m=audio 14748 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- asterisk1*CLI> <-- SIP read from 10.0.10.10:5060: ACK sip:1000@10.0.10.199 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.10:5060;branch=z9hG4bKba4a1680373b5682 From: "600" ;tag=91401244155e7d64 To: ;tag=as579bc047 Contact: Call-ID: d7bf395a9552f2c9@10.0.10.10 CSeq: 41458 ACK User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 --- (11 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 10.0.10.16:1000: OPTIONS sip:10.0.10.199 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.16:1000;rport;branch=z9hG4bK0a000a10000000104576e94c00004dab0000324a Content-Length: 0 Call-ID: EF31813A-583F-43BC-80AF-F7B5E01C37EE@10.0.10.16 CSeq: 4759 OPTIONS From: ;tag=2030494211040 Max-Forwards: 70 To: --- (8 headers 0 lines)--- Looking for s in from-sip-external (domain 10.0.10.199) Transmitting (no NAT) to 10.0.10.16:1000: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.16:1000;rport;branch=z9hG4bK0a000a10000000104576e94c00004dab0000324a;received=10.0.10.16 From: ;tag=2030494211040 To: ;tag=as7e34949c Call-ID: EF31813A-583F-43BC-80AF-F7B5E01C37EE@10.0.10.16 CSeq: 4759 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call 'EF31813A-583F-43BC-80AF-F7B5E01C37EE@10.0.10.16' asterisk1*CLI> <-- SIP read from 10.0.10.10:5060: REFER sip:1000@10.0.10.199 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.10:5060;branch=z9hG4bKd51a57d3ec87fc10 From: "600" ;tag=91401244155e7d64 To: ;tag=as579bc047 Contact: Supported: replaces Refer-To: Referred-By: Call-ID: d7bf395a9552f2c9@10.0.10.10 CSeq: 41459 REFER User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 --- (14 headers 0 lines)--- Transfer to 555 in from-internal Transfer from 600 in from-internal Transmitting (no NAT) to 10.0.10.10:5060: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.0.10.10:5060;branch=z9hG4bKd51a57d3ec87fc10;received=10.0.10.10 From: "600" ;tag=91401244155e7d64 To: ;tag=as579bc047 Call-ID: d7bf395a9552f2c9@10.0.10.10 CSeq: 41459 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.10.10, port 5060 Reliably Transmitting (no NAT) to 10.0.10.10:5060: NOTIFY sip:600@10.0.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK5a9a14fb From: ;tag=as579bc047 To: "600" ;tag=91401244155e7d64 Contact: Call-ID: d7bf395a9552f2c9@10.0.10.10 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=41459 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.10.10, port 5060 Reliably Transmitting (no NAT) to 10.0.10.10:5060: BYE sip:600@10.0.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK7de8142c From: ;tag=as579bc047 To: "600" ;tag=91401244155e7d64 Contact: Call-ID: d7bf395a9552f2c9@10.0.10.10 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 10.0.10.10:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK5a9a14fb From: ;tag=as579bc047 To: "600" ;tag=91401244155e7d64 Call-ID: d7bf395a9552f2c9@10.0.10.10 CSeq: 102 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 --- (11 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 10.0.10.10:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK7de8142c From: ;tag=as579bc047 To: "600" ;tag=91401244155e7d64 Call-ID: d7bf395a9552f2c9@10.0.10.10 CSeq: 103 BYE User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 --- (11 headers 0 lines)--- 12 headers, 3 lines Reliably Transmitting (no NAT) to 192.168.2.204:5060: NOTIFY sip:502@192.168.2.204 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.109:5060;branch=z9hG4bK257c096b From: "Unknown" ;tag=as11347da9 To: Contact: Call-ID: 4066dced46e4901a7c75907965dd4088@192.168.2.109 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 93 asterisk1*CLI> Messages-Waiting: no Message-Account: sip:asterisk@192.168.2.109 Voice-Message: 0/0 (0/0) --- Scheduling destruction of call '4066dced46e4901a7c75907965dd4088@192.168.2.109' in 15000 ms asterisk1*CLI> 12 headers, 3 lines Reliably Transmitting (no NAT) to 192.168.2.204:5060: NOTIFY sip:501@192.168.2.204 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.109:5060;branch=z9hG4bK431bf4bc From: "Unknown" ;tag=as5ab61ee9 To: Contact: Call-ID: 669308043310a776282169c4657efcce@192.168.2.109 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 93 Messages-Waiting: no Message-Account: sip:asterisk@192.168.2.109 Voice-Message: 0/0 (0/0) --- Scheduling destruction of call '669308043310a776282169c4657efcce@192.168.2.109' in 15000 ms asterisk1*CLI> Destroying call 'd7bf395a9552f2c9@10.0.10.10' asterisk1*CLI> <-- SIP read from 192.168.2.204:5060: SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.2.109:5060;branch=z9hG4bK257c096b From: "Unknown" ;tag=as11347da9 To: ;tag=1c419490996 Call-ID: 4066dced46e4901a7c75907965dd4088@192.168.2.109 CSeq: 102 NOTIFY Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '4066dced46e4901a7c75907965dd4088@192.168.2.109' asterisk1*CLI> <-- SIP read from 192.168.2.204:5060: SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.2.109:5060;branch=z9hG4bK431bf4bc From: "Unknown" ;tag=as5ab61ee9 To: ;tag=1c419514073 Call-ID: 669308043310a776282169c4657efcce@192.168.2.109 CSeq: 102 NOTIFY Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Content-Length: 0 --- (10 headers 0 lines)--- asterisk1*CLI> Destroying call '669308043310a776282169c4657efcce@192.168.2.109' asterisk1*CLI> dialparties.agi: Starting New Dialparties.agi asterisk1*CLI> We're at 10.0.10.199 port 14482 asterisk1*CLI> Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP asterisk1*CLI> 13 headers, 11 lines Reliably Transmitting (no NAT) to 10.0.10.11:5060: INVITE sip:555@10.0.10.11:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK4ab3c499 From: "1000" ;tag=as31e5c26c To: Contact: Call-ID: 5f70a1da72ed035574b018116ff098d5@10.0.10.199 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 06 Dec 2006 15:59:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 32510 32510 IN IP4 10.0.10.199 s=session c=IN IP4 10.0.10.199 t=0 0 m=audio 14482 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterisk1*CLI> <-- SIP read from 10.0.10.11:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK4ab3c499 From: "1000" ;tag=as31e5c26c To: Call-ID: 5f70a1da72ed035574b018116ff098d5@10.0.10.199 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Content-Length: 0 --- (8 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 10.0.10.11:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK4ab3c499 From: "1000" ;tag=as31e5c26c To: ;tag=512c306e719ad8e1 Call-ID: 5f70a1da72ed035574b018116ff098d5@10.0.10.199 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 --- (10 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 10.0.10.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK4ab3c499 From: "1000" ;tag=as31e5c26c To: ;tag=512c306e719ad8e1 Call-ID: 5f70a1da72ed035574b018116ff098d5@10.0.10.199 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 206 v=0 o=555 8000 8000 IN IP4 10.0.10.11 s=SIP Call c=IN IP4 10.0.10.11 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 --- (12 headers 11 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.0.10.11:5004 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.10.11, port 5060 Transmitting (no NAT) to 10.0.10.11:5060: ACK sip:555@10.0.10.11:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK3ec8f94f From: "1000" ;tag=as31e5c26c To: ;tag=512c306e719ad8e1 Contact: Call-ID: 5f70a1da72ed035574b018116ff098d5@10.0.10.199 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 192.168.2.204:5060: REGISTER sip:192.168.2.109 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.204;branch=z9hG4bKac425760995 Max-Forwards: 70 From: ;tag=1c425757117 To: Call-ID: 143877972511200004719@192.168.2.204 CSeq: 11345 REGISTER Contact: ;expires=180 Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 180 User-Agent: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Length: 0 --- (13 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.2.204 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.2.204:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.204;branch=z9hG4bKac425760995;received=192.168.2.204 From: ;tag=1c425757117 To: Call-ID: 143877972511200004719@192.168.2.204 CSeq: 11345 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- asterisk1*CLI> Transmitting (no NAT) to 192.168.2.204:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.204;branch=z9hG4bKac425760995;received=192.168.2.204 From: ;tag=1c425757117 To: ;tag=as709322ac Call-ID: 143877972511200004719@192.168.2.204 CSeq: 11345 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 180 Contact: ;expires=180 Date: Wed, 06 Dec 2006 15:59:28 GMT Content-Length: 0 asterisk1*CLI> --- Scheduling destruction of call '143877972511200004719@192.168.2.204' in 15000 ms asterisk1*CLI> <-- SIP read from 10.0.10.11:5060: INVITE sip:1000@10.0.10.199 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bKd04de911a7c375cc From: ;tag=512c306e719ad8e1 To: "1000" ;tag=as31e5c26c Contact: Supported: replaces, timer Call-ID: 5f70a1da72ed035574b018116ff098d5@10.0.10.199 CSeq: 35166 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 303 v=0 o=555 8000 8001 IN IP4 10.0.10.11 s=SIP Call c=IN IP4 10.0.10.11 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 3 101 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 --- (13 headers 15 lines)--- Using INVITE request as basis request - 5f70a1da72ed035574b018116ff098d5@10.0.10.199 Sending to 10.0.10.11 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 10.0.10.11:5004 Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format GSM Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) We're at 10.0.10.199 port 14482 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.10.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bKd04de911a7c375cc;received=10.0.10.11 From: ;tag=512c306e719ad8e1 To: "1000" ;tag=as31e5c26c Call-ID: 5f70a1da72ed035574b018116ff098d5@10.0.10.199 CSeq: 35166 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 32510 32511 IN IP4 10.0.10.199 s=session c=IN IP4 10.0.10.199 t=0 0 m=audio 14482 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterisk1*CLI> <-- SIP read from 10.0.10.11:5060: ACK sip:1000@10.0.10.199 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK77d284a454fd5917 From: ;tag=512c306e719ad8e1 To: "1000" ;tag=as31e5c26c Contact: Call-ID: 5f70a1da72ed035574b018116ff098d5@10.0.10.199 CSeq: 35166 ACK User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 --- (11 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 10.0.10.11:5060: REFER sip:1000@10.0.10.199 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bKbed3cd2b9acbf7ae From: ;tag=512c306e719ad8e1 To: "1000" ;tag=as31e5c26c Contact: Supported: replaces Refer-To: Referred-By: Call-ID: 5f70a1da72ed035574b018116ff098d5@10.0.10.199 CSeq: 35167 REFER User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 --- (14 headers 0 lines)--- Transfer to 600 in from-internal Transfer from 555 in from-internal Transmitting (no NAT) to 10.0.10.11:5060: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bKbed3cd2b9acbf7ae;received=10.0.10.11 From: ;tag=512c306e719ad8e1 To: "1000" ;tag=as31e5c26c Call-ID: 5f70a1da72ed035574b018116ff098d5@10.0.10.199 CSeq: 35167 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.10.11, port 5060 Reliably Transmitting (no NAT) to 10.0.10.11:5060: NOTIFY sip:555@10.0.10.11:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK0f0488f6 From: "1000" ;tag=as31e5c26c To: ;tag=512c306e719ad8e1 Contact: Call-ID: 5f70a1da72ed035574b018116ff098d5@10.0.10.199 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=35167 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.10.11, port 5060 Reliably Transmitting (no NAT) to 10.0.10.11:5060: BYE sip:555@10.0.10.11:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK2e931053 From: "1000" ;tag=as31e5c26c To: ;tag=512c306e719ad8e1 Contact: Call-ID: 5f70a1da72ed035574b018116ff098d5@10.0.10.199 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 10.0.10.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK0f0488f6 From: "1000" ;tag=as31e5c26c To: ;tag=512c306e719ad8e1 Call-ID: 5f70a1da72ed035574b018116ff098d5@10.0.10.199 CSeq: 103 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 --- (11 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 10.0.10.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK2e931053 From: "1000" ;tag=as31e5c26c To: ;tag=512c306e719ad8e1 Call-ID: 5f70a1da72ed035574b018116ff098d5@10.0.10.199 CSeq: 104 BYE User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '5f70a1da72ed035574b018116ff098d5@10.0.10.199' asterisk1*CLI> dialparties.agi: Starting New Dialparties.agi asterisk1*CLI> We're at 10.0.10.199 port 15694 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 10.0.10.10:5060: INVITE sip:600@10.0.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK4be6b3a4 From: "1000" ;tag=as31f794d8 To: Contact: Call-ID: 313da53171ca562c1c46b47f418c32fc@10.0.10.199 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 06 Dec 2006 15:59:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 32510 32510 IN IP4 10.0.10.199 s=session c=IN IP4 10.0.10.199 t=0 0 m=audio 15694 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterisk1*CLI> <-- SIP read from 10.0.10.10:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK4be6b3a4 From: "1000" ;tag=as31f794d8 To: Call-ID: 313da53171ca562c1c46b47f418c32fc@10.0.10.199 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Content-Length: 0 --- (8 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 10.0.10.10:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK4be6b3a4 From: "1000" ;tag=as31f794d8 To: ;tag=1c3f6aec8f609021 Call-ID: 313da53171ca562c1c46b47f418c32fc@10.0.10.199 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 --- (10 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 10.0.10.10:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK4be6b3a4 From: "1000" ;tag=as31f794d8 To: ;tag=1c3f6aec8f609021 Call-ID: 313da53171ca562c1c46b47f418c32fc@10.0.10.199 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 150 v=0 o=600 8000 8000 IN IP4 10.0.10.10 s=SIP Call c=IN IP4 10.0.10.10 t=0 0 m=audio 5004 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 --- (12 headers 9 lines)--- Found RTP audio format 0 Peer audio RTP is at port 10.0.10.10:5004 Found description format PCMU Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.10.10, port 5060 Transmitting (no NAT) to 10.0.10.10:5060: ACK sip:600@10.0.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK05a2845b From: "1000" ;tag=as31f794d8 To: ;tag=1c3f6aec8f609021 Contact: Call-ID: 313da53171ca562c1c46b47f418c32fc@10.0.10.199 asterisk1*CLI> CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterisk1*CLI> 12 headers, 3 lines Reliably Transmitting (no NAT) to 192.168.2.204:5060: NOTIFY sip:503@192.168.2.204 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.109:5060;branch=z9hG4bK31261f0d From: "Unknown" ;tag=as66fc159c To: Contact: Call-ID: 60d433ef132a7d833a9c7e5a601887c6@192.168.2.109 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 93 Messages-Waiting: no Message-Account: sip:asterisk@192.168.2.109 Voice-Message: 0/0 (0/0) --- Scheduling destruction of call '60d433ef132a7d833a9c7e5a601887c6@192.168.2.109' in 15000 ms asterisk1*CLI> <-- SIP read from 192.168.2.204:5060: SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.2.109:5060;branch=z9hG4bK31261f0d From: "Unknown" ;tag=as66fc159c To: ;tag=1c433303887 Call-ID: 60d433ef132a7d833a9c7e5a601887c6@192.168.2.109 CSeq: 102 NOTIFY Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Content-Length: 0 asterisk1*CLI> --- (10 headers 0 lines)--- Destroying call '60d433ef132a7d833a9c7e5a601887c6@192.168.2.109' asterisk1*CLI> Destroying call '143877930311200004719@192.168.2.204' asterisk1*CLI> Destroying call '143877952111200004719@192.168.2.204' asterisk1*CLI> <-- SIP read from 10.0.10.10:5060: INVITE sip:1000@10.0.10.199 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.10:5060;branch=z9hG4bKa73c030654a256c1 From: ;tag=1c3f6aec8f609021 To: "1000" ;tag=as31f794d8 Contact: Supported: replaces, timer Call-ID: 313da53171ca562c1c46b47f418c32fc@10.0.10.199 CSeq: 38616 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 247 v=0 o=600 8000 8001 IN IP4 10.0.10.10 s=SIP Call c=IN IP4 10.0.10.10 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 3 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=ptime:20 --- (13 headers 13 lines)--- Using INVITE request as basis request - 313da53171ca562c1c46b47f418c32fc@10.0.10.199 Sending to 10.0.10.10 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 3 Peer audio RTP is at port 10.0.10.10:5004 Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format GSM Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) We're at 10.0.10.199 port 15694 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 10.0.10.10:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.10:5060;branch=z9hG4bKa73c030654a256c1;received=10.0.10.10 From: ;tag=1c3f6aec8f609021 To: "1000" ;tag=as31f794d8 Call-ID: 313da53171ca562c1c46b47f418c32fc@10.0.10.199 CSeq: 38616 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 182 v=0 o=root 32510 32511 IN IP4 10.0.10.199 s=session c=IN IP4 10.0.10.199 t=0 0 m=audio 15694 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- asterisk1*CLI> <-- SIP read from 10.0.10.10:5060: ACK sip:1000@10.0.10.199 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.10:5060;branch=z9hG4bK613360a5e7c0414b From: ;tag=1c3f6aec8f609021 To: "1000" ;tag=as31f794d8 Contact: Call-ID: 313da53171ca562c1c46b47f418c32fc@10.0.10.199 CSeq: 38616 ACK User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 --- (11 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 10.0.10.10:5060: REFER sip:1000@10.0.10.199 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.10:5060;branch=z9hG4bK6bcbbc6867b5b679 From: ;tag=1c3f6aec8f609021 To: "1000" ;tag=as31f794d8 Contact: Supported: replaces Refer-To: Referred-By: Call-ID: 313da53171ca562c1c46b47f418c32fc@10.0.10.199 CSeq: 38617 REFER User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 --- (14 headers 0 lines)--- Transfer to 555 in from-internal Transfer from 600 in from-internal Transmitting (no NAT) to 10.0.10.10:5060: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.0.10.10:5060;branch=z9hG4bK6bcbbc6867b5b679;received=10.0.10.10 From: ;tag=1c3f6aec8f609021 To: "1000" ;tag=as31f794d8 Call-ID: 313da53171ca562c1c46b47f418c32fc@10.0.10.199 CSeq: 38617 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.10.10, port 5060 Reliably Transmitting (no NAT) to 10.0.10.10:5060: NOTIFY sip:600@10.0.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK74ef8134 From: "1000" ;tag=as31f794d8 To: ;tag=1c3f6aec8f609021 Contact: Call-ID: 313da53171ca562c1c46b47f418c32fc@10.0.10.199 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=38617 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.10.10, port 5060 Reliably Transmitting (no NAT) to 10.0.10.10:5060: BYE sip:600@10.0.10.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK32403b13 From: "1000" ;tag=as31f794d8 To: ;tag=1c3f6aec8f609021 Contact: Call-ID: 313da53171ca562c1c46b47f418c32fc@10.0.10.199 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing Content-Length: 0 --- <-- SIP read from 10.0.10.10:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK74ef8134 From: "1000" ;tag=as31f794d8 To: ;tag=1c3f6aec8f609021 Call-ID: 313da53171ca562c1c46b47f418c32fc@10.0.10.199 CSeq: 103 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 --- (11 headers 0 lines)--- <-- SIP read from 10.0.10.10:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK32403b13 From: "1000" ;tag=as31f794d8 To: ;tag=1c3f6aec8f609021 Call-ID: 313da53171ca562c1c46b47f418c32fc@10.0.10.199 CSeq: 104 BYE User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 asterisk1*CLI> --- (11 headers 0 lines)--- Destroying call '313da53171ca562c1c46b47f418c32fc@10.0.10.199' asterisk1*CLI> dialparties.agi: Starting New Dialparties.agi asterisk1*CLI> We're at 10.0.10.199 port 14556 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP asterisk1*CLI> 13 headers, 11 lines Reliably Transmitting (no NAT) to 10.0.10.11:5060: INVITE sip:555@10.0.10.11:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK2423673d From: "1000" ;tag=as4cb41ecc To: Contact: Call-ID: 18213d964c7c7d83508724a8488d18d0@10.0.10.199 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 06 Dec 2006 15:59:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 32510 32510 IN IP4 10.0.10.199 s=session c=IN IP4 10.0.10.199 t=0 0 m=audio 14556 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterisk1*CLI> <-- SIP read from 10.0.10.11:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK2423673d From: "1000" ;tag=as4cb41ecc To: Call-ID: 18213d964c7c7d83508724a8488d18d0@10.0.10.199 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Content-Length: 0 --- (8 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 10.0.10.11:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK2423673d From: "1000" ;tag=as4cb41ecc To: ;tag=482a2b2d0fac5215 Call-ID: 18213d964c7c7d83508724a8488d18d0@10.0.10.199 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 asterisk1*CLI> --- (10 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 10.0.10.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK2423673d From: "1000" ;tag=as4cb41ecc To: ;tag=482a2b2d0fac5215 Call-ID: 18213d964c7c7d83508724a8488d18d0@10.0.10.199 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 206 v=0 o=555 8000 8000 IN IP4 10.0.10.11 s=SIP Call c=IN IP4 10.0.10.11 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 --- (12 headers 11 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.0.10.11:5004 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.10.11, port 5060 Transmitting (no NAT) to 10.0.10.11:5060: ACK sip:555@10.0.10.11:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK4613af76 From: "1000" ;tag=as4cb41ecc To: ;tag=482a2b2d0fac5215 Contact: Call-ID: 18213d964c7c7d83508724a8488d18d0@10.0.10.199 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 10.0.10.16:1000: OPTIONS sip:10.0.10.199 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.16:1000;rport;branch=z9hG4bK0a000a10000000104576e960000075730000324c Content-Length: 0 Call-ID: EF31813A-583F-43BC-80AF-F7B5E01C37EE@10.0.10.16 CSeq: 4760 OPTIONS From: ;tag=20306942118588 Max-Forwards: 70 To: --- (8 headers 0 lines)--- Looking for s in from-sip-external (domain 10.0.10.199) Transmitting (no NAT) to 10.0.10.16:1000: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.16:1000;rport;branch=z9hG4bK0a000a10000000104576e960000075730000324c;received=10.0.10.16 From: ;tag=20306942118588 To: ;tag=as6fd12edd Call-ID: EF31813A-583F-43BC-80AF-F7B5E01C37EE@10.0.10.16 CSeq: 4760 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call 'EF31813A-583F-43BC-80AF-F7B5E01C37EE@10.0.10.16' asterisk1*CLI> Destroying call '143877972511200004719@192.168.2.204' asterisk1*CLI> <-- SIP read from 10.0.10.11:5060: INVITE sip:1000@10.0.10.199 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bKf466147d40cba021 From: ;tag=482a2b2d0fac5215 To: "1000" ;tag=as4cb41ecc Contact: Supported: replaces, timer Call-ID: 18213d964c7c7d83508724a8488d18d0@10.0.10.199 CSeq: 60175 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 303 v=0 o=555 8000 8001 IN IP4 10.0.10.11 s=SIP Call c=IN IP4 10.0.10.11 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 3 101 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 --- (13 headers 15 lines)--- Using INVITE request as basis request - 18213d964c7c7d83508724a8488d18d0@10.0.10.199 Sending to 10.0.10.11 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 10.0.10.11:5004 Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format GSM Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) We're at 10.0.10.199 port 14556 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.10.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bKf466147d40cba021;received=10.0.10.11 From: ;tag=482a2b2d0fac5215 To: "1000" ;tag=as4cb41ecc Call-ID: 18213d964c7c7d83508724a8488d18d0@10.0.10.199 CSeq: 60175 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 238 v=0 o=root 32510 32511 IN IP4 10.0.10.199 s=session c=IN IP4 10.0.10.199 t=0 0 m=audio 14556 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterisk1*CLI> <-- SIP read from 10.0.10.11:5060: ACK sip:1000@10.0.10.199 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bKc5791ad0abc9f62b From: ;tag=482a2b2d0fac5215 To: "1000" ;tag=as4cb41ecc Contact: Call-ID: 18213d964c7c7d83508724a8488d18d0@10.0.10.199 CSeq: 60175 ACK User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 --- (11 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 10.0.10.11:5060: REFER sip:1000@10.0.10.199 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK71a7c1fa2b751d6d From: ;tag=482a2b2d0fac5215 To: "1000" ;tag=as4cb41ecc Contact: Supported: replaces Refer-To: Referred-By: Call-ID: 18213d964c7c7d83508724a8488d18d0@10.0.10.199 CSeq: 60176 REFER User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 --- (14 headers 0 lines)--- Transfer to 600 in from-internal Transfer from 555 in from-internal Transmitting (no NAT) to 10.0.10.11:5060: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK71a7c1fa2b751d6d;received=10.0.10.11 From: ;tag=482a2b2d0fac5215 To: "1000" ;tag=as4cb41ecc Call-ID: 18213d964c7c7d83508724a8488d18d0@10.0.10.199 CSeq: 60176 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.10.11, port 5060 Reliably Transmitting (no NAT) to 10.0.10.11:5060: NOTIFY sip:555@10.0.10.11:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK76fdfcad From: "1000" ;tag=as4cb41ecc To: ;tag=482a2b2d0fac5215 Contact: Call-ID: 18213d964c7c7d83508724a8488d18d0@10.0.10.199 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=60176 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.10.11, port 5060 Reliably Transmitting (no NAT) to 10.0.10.11:5060: BYE sip:555@10.0.10.11:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK1429eae5 From: "1000" ;tag=as4cb41ecc To: ;tag=482a2b2d0fac5215 Contact: Call-ID: 18213d964c7c7d83508724a8488d18d0@10.0.10.199 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 10.0.10.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK76fdfcad From: "1000" ;tag=as4cb41ecc To: ;tag=482a2b2d0fac5215 Call-ID: 18213d964c7c7d83508724a8488d18d0@10.0.10.199 CSeq: 103 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 --- (11 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 10.0.10.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.10.199:5060;branch=z9hG4bK1429eae5 From: "1000" ;tag=as4cb41ecc To: ;tag=482a2b2d0fac5215 Call-ID: 18213d964c7c7d83508724a8488d18d0@10.0.10.199 CSeq: 104 BYE User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0