--- pbx*CLI> <-- SIP read from 83.125.8.46:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK7ff7d6bb;received=81.214.188.91;rport=5060 From: "1008" ;tag=as09059a05 To: ;tag=as0af265a6 Call-ID: 40f97cd50b4b182b5ed9d76438b60c5a@192.168.2.15 CSeq: 104 BYE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- (11 headers 0 lines) --- Destroying call '40f97cd50b4b182b5ed9d76438b60c5a@192.168.2.15' Destroying call '21969694911200041956@10.0.0.2' pbx*CLI> <-- SIP read from 10.0.0.2:5060: INVITE sip:071311298862@10.0.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236762393 Max-Forwards: 70 From: ;tag=1c236760662 To: Call-ID: 23676042311200042010@10.0.0.2 CSeq: 1 INVITE Contact: Supported: em,100rel,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Content-Type: application/sdp Content-Length: 272 v=0 o=AudiocodesGW 236753870 236753787 IN IP4 10.0.0.2 s=Phone-Call c=IN IP4 10.0.0.2 t=0 0 m=audio 6230 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv a=rtcp:6231 IN IP4 10.0.0.2 --- (13 headers 13 lines) --- Using INVITE request as basis request - 23676042311200042010@10.0.0.2 Sending to 10.0.0.2 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236762393;received=10.0.0.2 From: ;tag=1c236760662 To: ;tag=as50a3ddcd Call-ID: 23676042311200042010@10.0.0.2 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3797ccf7" Content-Length: 0 --- Scheduling destruction of call '23676042311200042010@10.0.0.2' in 15000 ms Found user '1024' pbx*CLI> <-- SIP read from 10.0.0.2:5060: ACK sip:071311298862@10.0.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236762393 Max-Forwards: 70 From: ;tag=1c236760662 To: ;tag=as50a3ddcd Call-ID: 23676042311200042010@10.0.0.2 CSeq: 1 ACK Contact: Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Content-Length: 0 --- (12 headers 0 lines) --- pbx*CLI> <-- SIP read from 10.0.0.2:5060: INVITE sip:071311298862@10.0.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236789509 Max-Forwards: 70 From: ;tag=1c236760662 To: Call-ID: 23676042311200042010@10.0.0.2 CSeq: 2 INVITE Proxy-Authorization: Digest username="1024",realm="asterisk",nonce="3797ccf7",uri="sip:071311298862@10.0.0.1",algorithm=MD5,response="2c03f960b4d9ebe940c111c74d0bf7d0" Contact: Supported: em,100rel,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Content-Type: application/sdp Content-Length: 272 v=0 o=AudiocodesGW 236753870 236753787 IN IP4 10.0.0.2 s=Phone-Call c=IN IP4 10.0.0.2 t=0 0 m=audio 6230 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv a=rtcp:6231 IN IP4 10.0.0.2 --- (14 headers 13 lines) --- Using INVITE request as basis request - 23676042311200042010@10.0.0.2 Sending to 10.0.0.2 : 5060 (non-NAT) Found user '1024' Found RTP audio format 18 Found RTP audio format 96 Peer audio RTP is at port 10.0.0.2:6230 Found description format G729 Found description format telephone-event Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 071311298862 in masus (domain 10.0.0.1;user=phone) list_route: hop: Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236789509;received=10.0.0.2 From: ;tag=1c236760662 To: Call-ID: 23676042311200042010@10.0.0.2 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- We're at 192.168.2.15 port 29314 Adding codec 0x100 (g729) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 15 lines Reliably Transmitting (NAT) to 83.125.8.46:5060: INVITE sip:004971311298862@voip.dus.net SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK01dcd9f7;rport From: "1024" ;tag=as20b9cbea To: Contact: Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 22 Apr 2008 15:39:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 334 v=0 o=root 8992 8992 IN IP4 192.168.2.15 s=session c=IN IP4 192.168.2.15 t=0 0 m=audio 29314 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> <-- SIP read from 83.125.8.46:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK01dcd9f7;received=81.214.188.91;rport=5060 From: "1024" ;tag=as20b9cbea To: ;tag=as71a26820 Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15 CSeq: 102 INVITE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="dus.net", nonce="10993390" Content-Length: 0 --- (11 headers 0 lines) --- Transmitting (NAT) to 83.125.8.46:5060: ACK sip:004971311298862@voip.dus.net SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK01dcd9f7;rport From: "1024" ;tag=as20b9cbea To: ;tag=as71a26820 Contact: Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- We're at 192.168.2.15 port 29314 Adding codec 0x100 (g729) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 83.125.8.46:5060: INVITE sip:004971311298862@voip.dus.net SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK501ab53e;rport From: "1024" ;tag=as20b9cbea To: Contact: Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="000387233745", realm="dus.net", algorithm=MD5, uri="sip:004971311298862@voip.dus.net", nonce="10993390", response="11835a648761c7ce8d3c4c6a2670fb05", opaque="" Date: Tue, 22 Apr 2008 15:39:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 334 v=0 o=root 8992 8993 IN IP4 192.168.2.15 s=session c=IN IP4 192.168.2.15 t=0 0 m=audio 29314 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> <-- SIP read from 83.125.8.46:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK501ab53e;received=81.214.188.91;rport=5060 From: "1024" ;tag=as20b9cbea To: Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15 CSeq: 103 INVITE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- (11 headers 0 lines) --- pbx*CLI> <-- SIP read from 83.125.8.46:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK48238559;received=81.214.188.91;rport=5060 From: "1001" ;tag=as1387477f To: ;tag=as3c1e61a9 Call-ID: 0b760e162c18bdc90283a21f38126586@192.168.2.15 CSeq: 103 INVITE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 351 v=0 o=root 1387 1388 IN IP4 83.125.8.46 s=session c=IN IP4 83.125.8.46 t=0 0 m=audio 58704 RTP/AVP 8 3 97 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (12 headers 17 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 83.125.8.46:58704 Found description format PCMA Found description format GSM Found description format iLBC Found description format G729 Found description format telephone-event Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x50a (gsm|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50a (gsm|alaw|g729|ilbc) 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 83.125.8.46, port 5060 Transmitting (NAT) to 83.125.8.46:5060: ACK sip:004950519159871@83.125.8.46 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK62715351;rport From: "1001" ;tag=as1387477f To: ;tag=as3c1e61a9 Contact: Call-ID: 0b760e162c18bdc90283a21f38126586@192.168.2.15 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- We're at 10.0.0.1 port 24560 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac224598576;received=10.0.0.2 From: ;tag=1c224543595 To: ;tag=as3b3a07ab Call-ID: 2245433591120004200@10.0.0.2 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 226 v=0 o=root 8992 8993 IN IP4 10.0.0.1 s=session c=IN IP4 10.0.0.1 t=0 0 m=audio 24560 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - --- pbx*CLI> <-- SIP read from 10.0.0.2:5060: ACK sip:050519159871@10.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac237196027 Max-Forwards: 70 From: ;tag=1c224543595 To: ;tag=as3b3a07ab Call-ID: 2245433591120004200@10.0.0.2 CSeq: 2 ACK Contact: Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Content-Length: 0 --- (12 headers 0 lines) --- pbx*CLI> <-- SIP read from 83.125.8.46:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK501ab53e;received=81.214.188.91;rport=5060 From: "1024" ;tag=as20b9cbea To: ;tag=as73d777d3 Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15 CSeq: 103 INVITE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 351 v=0 o=root 1387 1387 IN IP4 83.125.8.46 s=session c=IN IP4 83.125.8.46 t=0 0 m=audio 20182 RTP/AVP 8 3 97 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (12 headers 17 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 83.125.8.46:20182 Found description format PCMA Found description format GSM Found description format iLBC Found description format G729 Found description format telephone-event Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x50a (gsm|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50a (gsm|alaw|g729|ilbc) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) We're at 10.0.0.1 port 12932 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236789509;received=10.0.0.2 From: ;tag=1c236760662 To: ;tag=as35f994c9 Call-ID: 23676042311200042010@10.0.0.2 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 226 v=0 o=root 8992 8992 IN IP4 10.0.0.1 s=session c=IN IP4 10.0.0.1 t=0 0 m=audio 12932 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - --- pbx*CLI> <-- SIP read from 10.0.0.2:5060: INVITE sip:02452155048@10.0.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac239637731 Max-Forwards: 70 From: ;tag=1c239635997 To: Call-ID: 23963575711200042012@10.0.0.2 CSeq: 1 INVITE Contact: Supported: em,100rel,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Content-Type: application/sdp Content-Length: 272 v=0 o=AudiocodesGW 239629226 239629145 IN IP4 10.0.0.2 s=Phone-Call c=IN IP4 10.0.0.2 t=0 0 m=audio 6070 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv a=rtcp:6071 IN IP4 10.0.0.2 --- (13 headers 13 lines) --- Using INVITE request as basis request - 23963575711200042012@10.0.0.2 Sending to 10.0.0.2 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac239637731;received=10.0.0.2 From: ;tag=1c239635997 To: ;tag=as752da0c8 Call-ID: 23963575711200042012@10.0.0.2 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="05560f2b" Content-Length: 0 --- Scheduling destruction of call '23963575711200042012@10.0.0.2' in 15000 ms Found user '1008' pbx*CLI> <-- SIP read from 10.0.0.2:5060: ACK sip:02452155048@10.0.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac239637731 Max-Forwards: 70 From: ;tag=1c239635997 To: ;tag=as752da0c8 Call-ID: 23963575711200042012@10.0.0.2 CSeq: 1 ACK Contact: Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Content-Length: 0 --- (12 headers 0 lines) --- pbx*CLI> <-- SIP read from 10.0.0.2:5060: INVITE sip:02452155048@10.0.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac239664674 Max-Forwards: 70 From: ;tag=1c239635997 To: Call-ID: 23963575711200042012@10.0.0.2 CSeq: 2 INVITE Proxy-Authorization: Digest username="1008",realm="asterisk",nonce="05560f2b",uri="sip:02452155048@10.0.0.1",algorithm=MD5,response="b26489cd71bf709ba926cb989a6584cd" Contact: Supported: em,100rel,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Content-Type: application/sdp Content-Length: 272 v=0 o=AudiocodesGW 239629226 239629145 IN IP4 10.0.0.2 s=Phone-Call c=IN IP4 10.0.0.2 t=0 0 m=audio 6070 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv a=rtcp:6071 IN IP4 10.0.0.2 --- (14 headers 13 lines) --- Using INVITE request as basis request - 23963575711200042012@10.0.0.2 Sending to 10.0.0.2 : 5060 (non-NAT) Found user '1008' Found RTP audio format 18 Found RTP audio format 96 Peer audio RTP is at port 10.0.0.2:6070 Found description format G729 Found description format telephone-event Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 02452155048 in masus (domain 10.0.0.1;user=phone) list_route: hop: Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac239664674;received=10.0.0.2 From: ;tag=1c239635997 To: Call-ID: 23963575711200042012@10.0.0.2 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- We're at 192.168.2.15 port 5638 Adding codec 0x100 (g729) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 15 lines Reliably Transmitting (NAT) to 83.125.8.46:5060: INVITE sip:00492452155048@voip.dus.net SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK22c4e836;rport From: "1008" ;tag=as26adf52c To: Contact: Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 22 Apr 2008 15:39:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 333 v=0 o=root 8992 8992 IN IP4 192.168.2.15 s=session c=IN IP4 192.168.2.15 t=0 0 m=audio 5638 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> <-- SIP read from 83.125.8.46:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK22c4e836;received=81.214.188.91;rport=5060 From: "1008" ;tag=as26adf52c To: ;tag=as2760b903 Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15 CSeq: 102 INVITE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="dus.net", nonce="7cff320c" Content-Length: 0 --- (11 headers 0 lines) --- Transmitting (NAT) to 83.125.8.46:5060: ACK sip:00492452155048@voip.dus.net SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK22c4e836;rport From: "1008" ;tag=as26adf52c To: ;tag=as2760b903 Contact: Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- We're at 192.168.2.15 port 5638 Adding codec 0x100 (g729) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 83.125.8.46:5060: INVITE sip:00492452155048@voip.dus.net SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK2ff645d4;rport From: "1008" ;tag=as26adf52c To: Contact: Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="000387233745", realm="dus.net", algorithm=MD5, uri="sip:00492452155048@voip.dus.net", nonce="7cff320c", response="1d30e905cb4367b250e3119ea85849e3", opaque="" Date: Tue, 22 Apr 2008 15:39:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY ontent-Type: application/sdp Content-Length: 333 v=0 o=root 8992 8993 IN IP4 192.168.2.15 s=session c=IN IP4 192.168.2.15 t=0 0 m=audio 5638 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> <-- SIP read from 83.125.8.46:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK2ff645d4;received=81.214.188.91;rport=5060 From: "1008" ;tag=as26adf52c To: Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15 CSeq: 103 INVITE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- (11 headers 0 lines) --- pbx*CLI> <-- SIP read from 10.0.0.2:5060: BYE sip:050519159871@10.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac239998965 Max-Forwards: 70 From: ;tag=1c224543595 To: ;tag=as3b3a07ab Call-ID: 2245433591120004200@10.0.0.2 CSeq: 3 BYE Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Reason: Q.850 ;cause=16 Content-Length: 0 --- (12 headers 0 lines) --- Sending to 10.0.0.2 : 5060 (non-NAT) Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac239998965;received=10.0.0.2 From: ;tag=1c224543595 To: ;tag=as3b3a07ab Call-ID: 2245433591120004200@10.0.0.2 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- Scheduling destruction of call '0b760e162c18bdc90283a21f38126586@192.168.2.15' in 32000 ms set_destination: Parsing for address/port to send to set_destination: set destination to 83.125.8.46, port 5060 Reliably Transmitting (NAT) to 83.125.8.46:5060: BYE sip:004950519159871@83.125.8.46 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK0e08051e;rport From: "1001" ;tag=as1387477f To: ;tag=as3c1e61a9 Call-ID: 0b760e162c18bdc90283a21f38126586@192.168.2.15 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="000387233745", realm="dus.net", algorithm=MD5, uri="sip:004950519159871@83.125.8.46", nonce="389ba3cb", response="4065d10f74da0932830075057affa1a8", opaque="" Content-Length: 0 --- pbx*CLI> <-- SIP read from 83.125.8.46:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK0e08051e;received=81.214.188.91;rport=5060 From: "1001" ;tag=as1387477f To: ;tag=as3c1e61a9 Call-ID: 0b760e162c18bdc90283a21f38126586@192.168.2.15 CSeq: 104 BYE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- (11 headers 0 lines) --- Destroying call '0b760e162c18bdc90283a21f38126586@192.168.2.15' Destroying call '2245433591120004200@10.0.0.2' pbx*CLI> <-- SIP read from 83.125.8.46:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK6ec06309;received=81.214.188.91;rport=5060 From: "1010" ;tag=as018eec44 To: ;tag=as66795132 Call-ID: 0adee35a7e61c7d10bffba995ec07c88@192.168.2.15 CSeq: 103 INVITE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: ontent-Type: application/sdp Content-Length: 351 v=0 o=root 1387 1388 IN IP4 83.125.8.46 s=session c=IN IP4 83.125.8.46 t=0 0 m=audio 16736 RTP/AVP 8 3 97 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (12 headers 17 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 83.125.8.46:16736 Found description format PCMA Found description format GSM Found description format iLBC Found description format G729 Found description format telephone-event Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x50a (gsm|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50a (gsm|alaw|g729|ilbc) 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 83.125.8.46, port 5060 Transmitting (NAT) to 83.125.8.46:5060: ACK sip:0049226157403@83.125.8.46 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK58f07536;rport From: "1010" ;tag=as018eec44 To: ;tag=as66795132 Contact: Call-ID: 0adee35a7e61c7d10bffba995ec07c88@192.168.2.15 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- We're at 10.0.0.1 port 26290 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac223785312;received=10.0.0.2 From: ;tag=1c223759377 To: ;tag=as57837876 Call-ID: 22375913911200041959@10.0.0.2 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 226 v=0 o=root 8992 8993 IN IP4 10.0.0.1 s=session c=IN IP4 10.0.0.1 t=0 0 m=audio 26290 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - --- pbx*CLI> <-- SIP read from 10.0.0.2:5060: ACK sip:0226157403@10.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac241725686 Max-Forwards: 70 From: ;tag=1c223759377 To: ;tag=as57837876 Call-ID: 22375913911200041959@10.0.0.2 CSeq: 2 ACK Contact: Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Content-Length: 0 --- (12 headers 0 lines) --- pbx*CLI> <-- SIP read from 83.125.8.46:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK2ff645d4;received=81.214.188.91;rport=5060 From: "1008" ;tag=as26adf52c To: ;tag=as53cf50c2 Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15 CSeq: 103 INVITE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 351 v=0 o=root 1387 1387 IN IP4 83.125.8.46 s=session c=IN IP4 83.125.8.46 t=0 0 m=audio 24116 RTP/AVP 8 3 97 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (12 headers 17 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 83.125.8.46:24116 Found description format PCMA Found description format GSM Found description format iLBC Found description format G729 Found description format telephone-event Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x50a (gsm|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50a (gsm|alaw|g729|ilbc) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) We're at 10.0.0.1 port 11754 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac239664674;received=10.0.0.2 From: ;tag=1c239635997 To: ;tag=as07d9b14a Call-ID: 23963575711200042012@10.0.0.2 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 226 v=0 o=root 8992 8992 IN IP4 10.0.0.1 s=session c=IN IP4 10.0.0.1 t=0 0 m=audio 11754 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - --- pbx*CLI> <-- SIP read from 10.0.0.2:5060: INVITE sip:02771819382@10.0.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac244172515 Max-Forwards: 70 From: ;tag=1c244170773 To: Call-ID: 24417053211200042016@10.0.0.2 CSeq: 1 INVITE Contact: Supported: em,100rel,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Content-Type: application/sdp Content-Length: 272 v=0 o=AudiocodesGW 244164123 244164040 IN IP4 10.0.0.2 s=Phone-Call c=IN IP4 10.0.0.2 t=0 0 m=audio 6010 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv a=rtcp:6011 IN IP4 10.0.0.2 --- (13 headers 13 lines) --- Using INVITE request as basis request - 24417053211200042016@10.0.0.2 Sending to 10.0.0.2 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac244172515;received=10.0.0.2 From: ;tag=1c244170773 To: ;tag=as1dc1fe85 Call-ID: 24417053211200042016@10.0.0.2 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4dbc9192" Content-Length: 0 --- Scheduling destruction of call '24417053211200042016@10.0.0.2' in 15000 ms Found user '1002' pbx*CLI> <-- SIP read from 10.0.0.2:5060: ACK sip:02771819382@10.0.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac244172515 Max-Forwards: 70 From: ;tag=1c244170773 To: ;tag=as1dc1fe85 Call-ID: 24417053211200042016@10.0.0.2 CSeq: 1 ACK Contact: Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Content-Length: 0 --- (12 headers 0 lines) --- pbx*CLI> <-- SIP read from 10.0.0.2:5060: INVITE sip:02771819382@10.0.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac244195942 Max-Forwards: 70 From: ;tag=1c244170773 To: Call-ID: 24417053211200042016@10.0.0.2 CSeq: 2 INVITE Proxy-Authorization: Digest username="1002",realm="asterisk",nonce="4dbc9192",uri="sip:02771819382@10.0.0.1",algorithm=MD5,response="92bc14b5bb273b85e539bebf71a494e5" Contact: Supported: em,100rel,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Content-Type: application/sdp Content-Length: 272 v=0 o=AudiocodesGW 244164123 244164040 IN IP4 10.0.0.2 s=Phone-Call c=IN IP4 10.0.0.2 t=0 0 m=audio 6010 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv a=rtcp:6011 IN IP4 10.0.0.2 --- (14 headers 13 lines) --- Using INVITE request as basis request - 24417053211200042016@10.0.0.2 Sending to 10.0.0.2 : 5060 (non-NAT) Found user '1002' Found RTP audio format 18 Found RTP audio format 96 Peer audio RTP is at port 10.0.0.2:6010 Found description format G729 Found description format telephone-event Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 02771819382 in masus (domain 10.0.0.1;user=phone) list_route: hop: Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac244195942;received=10.0.0.2 From: ;tag=1c244170773 To: Call-ID: 24417053211200042016@10.0.0.2 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- We're at 192.168.2.15 port 24582 Adding codec 0x100 (g729) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 15 lines Reliably Transmitting (NAT) to 83.125.8.46:5060: INVITE sip:00492771819382@voip.dus.net SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK1dffbf90;rport From: "1002" ;tag=as495acd44 To: Contact: Call-ID: 00d9911a36d325e378cc75451986412f@192.168.2.15 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 22 Apr 2008 15:39:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 334 v=0 o=root 8992 8992 IN IP4 192.168.2.15 s=session c=IN IP4 192.168.2.15 t=0 0 m=audio 24582 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> <-- SIP read from 83.125.8.46:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK1dffbf90;received=81.214.188.91;rport=5060 From: "1002" ;tag=as495acd44 To: ;tag=as33330631 Call-ID: 00d9911a36d325e378cc75451986412f@192.168.2.15 CSeq: 102 INVITE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="dus.net", nonce="1e752fb2" Content-Length: 0 --- (11 headers 0 lines) --- Transmitting (NAT) to 83.125.8.46:5060: ACK sip:00492771819382@voip.dus.net SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK1dffbf90;rport From: "1002" ;tag=as495acd44 To: ;tag=as33330631 Contact: Call-ID: 00d9911a36d325e378cc75451986412f@192.168.2.15 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- We're at 192.168.2.15 port 24582 Adding codec 0x100 (g729) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 83.125.8.46:5060: INVITE sip:00492771819382@voip.dus.net SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK5cf7c4c2;rport From: "1002" ;tag=as495acd44 To: Contact: Call-ID: 00d9911a36d325e378cc75451986412f@192.168.2.15 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="000387233745", realm="dus.net", algorithm=MD5, uri="sip:00492771819382@voip.dus.net", nonce="1e752fb2", response="f7059c8df9a5908bf0bd0501757ed16b", opaque="" Date: Tue, 22 Apr 2008 15:39:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 334 v=0 o=root 8992 8993 IN IP4 192.168.2.15 s=session c=IN IP4 192.168.2.15 t=0 0 m=audio 24582 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> <-- SIP read from 83.125.8.46:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK5cf7c4c2;received=81.214.188.91;rport=5060 From: "1002" ;tag=as495acd44 To: Call-ID: 00d9911a36d325e378cc75451986412f@192.168.2.15 CSeq: 103 INVITE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- (11 headers 0 lines) --- pbx*CLI> <-- SIP read from 83.125.8.46:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK5cf7c4c2;received=81.214.188.91;rport=5060 From: "1002" ;tag=as495acd44 To: ;tag=as1a8e074b Call-ID: 00d9911a36d325e378cc75451986412f@192.168.2.15 CSeq: 103 INVITE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 350 v=0 o=root 1387 1387 IN IP4 83.125.8.46 s=session c=IN IP4 83.125.8.46 t=0 0 m=audio 5948 RTP/AVP 8 3 97 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (12 headers 17 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 83.125.8.46:5948 Found description format PCMA Found description format GSM Found description format iLBC Found description format G729 Found description format telephone-event Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x50a (gsm|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50a (gsm|alaw|g729|ilbc) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) We're at 10.0.0.1 port 25312 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac244195942;received=10.0.0.2 From: ;tag=1c244170773 To: ;tag=as3430dbc1 Call-ID: 24417053211200042016@10.0.0.2 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 226 v=0 o=root 8992 8992 IN IP4 10.0.0.1 s=session c=IN IP4 10.0.0.1 t=0 0 m=audio 25312 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - --- pbx*CLI> <-- SIP read from 10.0.0.2:5060: INVITE sip:0545173819@10.0.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac249298169 Max-Forwards: 70 From: ;tag=1c249296417 To: Call-ID: 24929617611200042020@10.0.0.2 CSeq: 1 INVITE Contact: Supported: em,100rel,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Content-Type: application/sdp Content-Length: 272 v=0 o=AudiocodesGW 249289592 249289509 IN IP4 10.0.0.2 s=Phone-Call c=IN IP4 10.0.0.2 t=0 0 m=audio 6030 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv a=rtcp:6031 IN IP4 10.0.0.2 --- (13 headers 13 lines) --- Using INVITE request as basis request - 24929617611200042020@10.0.0.2 Sending to 10.0.0.2 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac249298169;received=10.0.0.2 From: ;tag=1c249296417 To: ;tag=as2f5000a2 Call-ID: 24929617611200042020@10.0.0.2 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c97d623" Content-Length: 0 --- Scheduling destruction of call '24929617611200042020@10.0.0.2' in 15000 ms Found user '1004' pbx*CLI> <-- SIP read from 10.0.0.2:5060: ACK sip:0545173819@10.0.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac249298169 Max-Forwards: 70 From: ;tag=1c249296417 To: ;tag=as2f5000a2 Call-ID: 24929617611200042020@10.0.0.2 CSeq: 1 ACK Contact: Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Content-Length: 0 --- (12 headers 0 lines) --- pbx*CLI> <-- SIP read from 10.0.0.2:5060: INVITE sip:0545173819@10.0.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac249321137 Max-Forwards: 70 From: ;tag=1c249296417 To: Call-ID: 24929617611200042020@10.0.0.2 CSeq: 2 INVITE Proxy-Authorization: Digest username="1004",realm="asterisk",nonce="6c97d623",uri="sip:0545173819@10.0.0.1",algorithm=MD5,response="44a88de12c2cf00002e7ec68784557fe" Contact: Supported: em,100rel,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Content-Type: application/sdp Content-Length: 272 v=0 o=AudiocodesGW 249289592 249289509 IN IP4 10.0.0.2 s=Phone-Call c=IN IP4 10.0.0.2 t=0 0 m=audio 6030 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv a=rtcp:6031 IN IP4 10.0.0.2 --- (14 headers 13 lines) --- Using INVITE request as basis request - 24929617611200042020@10.0.0.2 Sending to 10.0.0.2 : 5060 (non-NAT) Found user '1004' Found RTP audio format 18 Found RTP audio format 96 Peer audio RTP is at port 10.0.0.2:6030 Found description format G729 Found description format telephone-event Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 0545173819 in masus (domain 10.0.0.1;user=phone) list_route: hop: Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac249321137;received=10.0.0.2 From: ;tag=1c249296417 To: Call-ID: 24929617611200042020@10.0.0.2 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- We're at 192.168.2.15 port 19604 Adding codec 0x100 (g729) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 15 lines Reliably Transmitting (NAT) to 83.125.8.46:5060: INVITE sip:0049545173819@voip.dus.net SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK100cd22e;rport From: "1004" ;tag=as284f8fbf To: Contact: Call-ID: 50e49713563aa63e5e1457881e006e58@192.168.2.15 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 22 Apr 2008 15:39:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 334 v=0 o=root 8992 8992 IN IP4 192.168.2.15 s=session c=IN IP4 192.168.2.15 t=0 0 m=audio 19604 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> <-- SIP read from 83.125.8.46:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK100cd22e;received=81.214.188.91;rport=5060 From: "1004" ;tag=as284f8fbf To: ;tag=as65e54448 Call-ID: 50e49713563aa63e5e1457881e006e58@192.168.2.15 CSeq: 102 INVITE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="dus.net", nonce="562cd475" Content-Length: 0 --- (11 headers 0 lines) --- Transmitting (NAT) to 83.125.8.46:5060: ACK sip:0049545173819@voip.dus.net SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK100cd22e;rport From: "1004" ;tag=as284f8fbf To: ;tag=as65e54448 Contact: Call-ID: 50e49713563aa63e5e1457881e006e58@192.168.2.15 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- We're at 192.168.2.15 port 19604 Adding codec 0x100 (g729) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 83.125.8.46:5060: INVITE sip:0049545173819@voip.dus.net SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK727b201f;rport From: "1004" ;tag=as284f8fbf To: Contact: Call-ID: 50e49713563aa63e5e1457881e006e58@192.168.2.15 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="000387233745", realm="dus.net", algorithm=MD5, uri="sip:0049545173819@voip.dus.net", nonce="562cd475", response="6616af67c44205bf87bd1504f6699fc0", opaque="" Date: Tue, 22 Apr 2008 15:39:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 334 v=0 o=root 8992 8993 IN IP4 192.168.2.15 s=session c=IN IP4 192.168.2.15 t=0 0 m=audio 19604 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> <-- SIP read from 83.125.8.46:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK727b201f;received=81.214.188.91;rport=5060 From: "1004" ;tag=as284f8fbf To: Call-ID: 50e49713563aa63e5e1457881e006e58@192.168.2.15 CSeq: 103 INVITE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- (11 headers 0 lines) --- pbx*CLI> <-- SIP read from 83.125.8.46:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK727b201f;received=81.214.188.91;rport=5060 From: "1004" ;tag=as284f8fbf To: ;tag=as68d70d13 Call-ID: 50e49713563aa63e5e1457881e006e58@192.168.2.15 CSeq: 103 INVITE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 351 v=0 o=root 1387 1387 IN IP4 83.125.8.46 s=session c=IN IP4 83.125.8.46 t=0 0 m=audio 10584 RTP/AVP 8 3 97 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (12 headers 17 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 83.125.8.46:10584 Found description format PCMA Found description format GSM Found description format iLBC Found description format G729 Found description format telephone-event Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x50a (gsm|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50a (gsm|alaw|g729|ilbc) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) We're at 10.0.0.1 port 21060 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac249321137;received=10.0.0.2 From: ;tag=1c249296417 To: ;tag=as147d1b15 Call-ID: 24929617611200042020@10.0.0.2 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 226 v=0 o=root 8992 8992 IN IP4 10.0.0.1 s=session c=IN IP4 10.0.0.1 t=0 0 m=audio 21060 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - --- pbx*CLI> <-- SIP read from 83.125.8.46:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK2ff645d4;received=81.214.188.91;rport=5060 From: "1008" ;tag=as26adf52c To: ;tag=as53cf50c2 Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15 CSeq: 103 INVITE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 351 v=0 o=root 1387 1388 IN IP4 83.125.8.46 s=session c=IN IP4 83.125.8.46 t=0 0 m=audio 24116 RTP/AVP 8 3 97 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (12 headers 17 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 83.125.8.46:24116 Found description format PCMA Found description format GSM Found description format iLBC Found description format G729 Found description format telephone-event Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x50a (gsm|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50a (gsm|alaw|g729|ilbc) 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 83.125.8.46, port 5060 Transmitting (NAT) to 83.125.8.46:5060: ACK sip:00492452155048@83.125.8.46 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK6827d41b;rport From: "1008" ;tag=as26adf52c To: ;tag=as53cf50c2 Contact: Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- We're at 10.0.0.1 port 11754 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac239664674;received=10.0.0.2 From: ;tag=1c239635997 To: ;tag=as07d9b14a Call-ID: 23963575711200042012@10.0.0.2 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 226 v=0 o=root 8992 8993 IN IP4 10.0.0.1 s=session c=IN IP4 10.0.0.1 t=0 0 m=audio 11754 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - --- pbx*CLI> s <-- SIP read from 10.0.0.2:5060: ACK sip:02452155048@10.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac256916755 Max-Forwards: 70 From: ;tag=1c239635997 To: ;tag=as07d9b14a Call-ID: 23963575711200042012@10.0.0.2 CSeq: 2 ACK Contact: Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Content-Length: 0 --- (12 headers 0 lines) --- pbx*CLI> sip deb <-- SIP read from 10.0.0.2:5060: CANCEL sip:071311298862@10.0.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236789509 Max-Forwards: 70 From: ;tag=1c236760662 To: Call-ID: 23676042311200042010@10.0.0.2 CSeq: 2 CANCEL Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Reason: Q.850 ;cause=16 Content-Length: 0 --- (12 headers 0 lines) --- Sending to 10.0.0.2 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236789509;received=10.0.0.2 From: ;tag=1c236760662 To: ;tag=as35f994c9 Call-ID: 23676042311200042010@10.0.0.2 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236789509;received=10.0.0.2 From: ;tag=1c236760662 To: ;tag=as35f994c9 Call-ID: 23676042311200042010@10.0.0.2 CSeq: 2 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Scheduling destruction of call '70e04fe371dd539c6cab3d4627e544f2@192.168.2.15' in 32000 ms Reliably Transmitting (NAT) to 83.125.8.46:5060: CANCEL sip:004971311298862@voip.dus.net SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK501ab53e;rport From: "1024" ;tag=as20b9cbea To: Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15 CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbx*CLI> sip deb <-- SIP read from 10.0.0.2:5060: ACK sip:071311298862@10.0.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236789509 Max-Forwards: 70 From: ;tag=1c236760662 To: ;tag=as35f994c9 Call-ID: 23676042311200042010@10.0.0.2 CSeq: 2 ACK Contact: Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Content-Length: 0 --- (12 headers 0 lines) --- Destroying call '23676042311200042010@10.0.0.2' pbx*CLI> sip debu <-- SIP read from 83.125.8.46:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK501ab53e;received=81.214.188.91;rport=5060 From: "1024" ;tag=as20b9cbea To: ;tag=as73d777d3 Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15 CSeq: 103 INVITE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- (10 headers 0 lines) --- Transmitting (NAT) to 83.125.8.46:5060: ACK sip:004971311298862@voip.dus.net SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK501ab53e;rport From: "1024" ;tag=as20b9cbea To: ;tag=as73d777d3 Contact: Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbx*CLI> sip debu <-- SIP read from 83.125.8.46:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK501ab53e;received=81.214.188.91;rport=5060 From: "1024" ;tag=as20b9cbea To: ;tag=as73d777d3 Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15 CSeq: 103 CANCEL User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- (11 headers 0 lines) --- Destroying call '70e04fe371dd539c6cab3d4627e544f2@192.168.2.15' pbx*CLI> sip debug <-- SIP read from 10.0.0.2:5060: BYE sip:02452155048@10.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac260466823 Max-Forwards: 70 From: ;tag=1c239635997 To: ;tag=as07d9b14a Call-ID: 23963575711200042012@10.0.0.2 CSeq: 3 BYE Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Reason: Q.850 ;cause=16 Content-Length: 0 --- (12 headers 0 lines) --- Sending to 10.0.0.2 : 5060 (non-NAT) Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac260466823;received=10.0.0.2 From: ;tag=1c239635997 To: ;tag=as07d9b14a Call-ID: 23963575711200042012@10.0.0.2 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing pbx*CLI> sip debug --- Scheduling destruction of call '262ccf6115d4bbb4467624dc0262a62f@192.168.2.15' in 32000 ms set_destination: Parsing for address/port to send to set_destination: set destination to 83.125.8.46, port 5060 Reliably Transmitting (NAT) to 83.125.8.46:5060: BYE sip:00492452155048@83.125.8.46 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK20fe8355;rport From: "1008" ;tag=as26adf52c To: ;tag=as53cf50c2 Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="000387233745", realm="dus.net", algorithm=MD5, uri="sip:00492452155048@83.125.8.46", nonce="7cff320c", response="4e64b3fb70a09315f7f7c96b4fb3167c", opaque="" Content-Length: 0 --- pbx*CLI> sip debug <-- SIP read from 83.125.8.46:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK20fe8355;received=81.214.188.91;rport=5060 From: "1008" ;tag=as26adf52c To: ;tag=as53cf50c2 Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15 CSeq: 104 BYE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- (11 headers 0 lines) --- Destroying call '262ccf6115d4bbb4467624dc0262a62f@192.168.2.15' pbx*CLI> sip debug <-- SIP read from 83.125.8.46:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK5cf7c4c2;received=81.214.188.91;rport=5060 From: "1002" ;tag=as495acd44 To: ;tag=as1a8e074b Call-ID: 00d9911a36d325e378cc75451986412f@192.168.2.15 CSeq: 103 INVITE User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 350 v=0 o=root 1387 1388 IN IP4 83.125.8.46 s=session c=IN IP4 83.125.8.46 t=0 0 m=audio 5948 RTP/AVP 8 3 97 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (12 headers 17 lines) --- Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 83.125.8.46:5948 Found description format PCMA Found description format GSM Found description format iLBC Found description format G729 Found description format telephone-event Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x50a (gsm|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50a (gsm|alaw|g729|ilbc) 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 83.125.8.46, port 5060 Transmitting (NAT) to 83.125.8.46:5060: ACK sip:00492771819382@83.125.8.46 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK593d3c6b;rport From: "1002" ;tag=as495acd44 To: ;tag=as1a8e074b Contact: Call-ID: 00d9911a36d325e378cc75451986412f@192.168.2.15 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Destroying call '23963575711200042012@10.0.0.2' We're at 10.0.0.1 port 25312 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.0.2:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac244195942;received=10.0.0.2 From: ;tag=1c244170773 To: ;tag=as3430dbc1 Call-ID: 24417053211200042016@10.0.0.2 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 226 v=0 o=root 8992 8993 IN IP4 10.0.0.1 s=session c=IN IP4 10.0.0.1 t=0 0 m=audio 25312 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - --- pbx*CLI> sip debug <-- SIP read from 10.0.0.2:5060: ACK sip:02771819382@10.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac260694128 Max-Forwards: 70 From: ;tag=1c244170773 To: ;tag=as3430dbc1 Call-ID: 24417053211200042016@10.0.0.2 CSeq: 2 ACK Contact: Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005 Content-Length: 0 --- (12 headers 0 lines) --- pbx*CLI> sip no debug