rendered paste body12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.1.156:5060:
OPTIONS sip:214@192.168.1.156:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK06a549b1;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as56768067
To: <sip:214@192.168.1.156:5060>
Contact: <sip:Unknown@192.168.1.101>
Call-ID: 52a9b7e102aa18f767905c4c77d25fa6@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 27 Jan 2007 01:36:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.1.156:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK06a549b1;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as56768067
To: <sip:214@192.168.1.156:5060>;tag=4bc0f2e39ec3632e
Call-ID: 52a9b7e102aa18f767905c4c77d25fa6@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:214@192.168.1.156:5060>
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 '52a9b7e102aa18f767905c4c77d25fa6@192.168.1.101'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.155:5060:
OPTIONS sip:206@192.168.1.155:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK60d6c435;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as24d7dfe1
To: <sip:206@192.168.1.155:5060>
Contact: <sip:Unknown@192.168.1.101>
Call-ID: 682dc4717cb963461b204a6e33c3ae49@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 27 Jan 2007 01:36:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.1.155:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK60d6c435;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as24d7dfe1
To: <sip:206@192.168.1.155:5060>;tag=c2f0315a81cc52a5
Call-ID: 682dc4717cb963461b204a6e33c3ae49@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Grandstream GXP2000 1.1.1.14
Contact: <sip:206@192.168.1.155:5060>
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 '682dc4717cb963461b204a6e33c3ae49@192.168.1.101'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 204.147.183.18:5060:
OPTIONS sip:sip.stanaphone.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK57bade22;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as4786b685
To: <sip:sip.stanaphone.com>
Contact: <sip:Unknown@192.168.1.101>
Call-ID: 713ec5bc3e2208795ecfb1ee4429f52c@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 27 Jan 2007 01:36:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 204.147.183.18:5060:
OPTIONS sip:sip.stanaphone.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK6c4019f6;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as67b56fe4
To: <sip:sip.stanaphone.com>
Contact: <sip:Unknown@192.168.1.101>
Call-ID: 314f2cf1375dc16640413b842302b89c@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 27 Jan 2007 01:36:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 204.147.183.18:5060:
OPTIONS sip:sip.stanaphone.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK6cb96aa0;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as0eed64e1
To: <sip:sip.stanaphone.com>
Contact: <sip:Unknown@192.168.1.101>
Call-ID: 63628d2a7e69a73e590fb7382f6b3ec0@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 27 Jan 2007 01:36:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 204.147.183.18:5060:
OPTIONS sip:sip.stanaphone.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK3cf0d63d;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as326072c1
To: <sip:sip.stanaphone.com>
Contact: <sip:Unknown@192.168.1.101>
Call-ID: 3894b134638dd83a7a7957ee50efd855@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 27 Jan 2007 01:36:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 204.147.183.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK57bade22;rport=5060;received=66.114.77.37
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as4786b685
To: <sip:sip.stanaphone.com>;tag=4e487ad2f2fae333606267674b3a79b1.9c79
Call-ID: 713ec5bc3e2208795ecfb1ee4429f52c@192.168.1.101
CSeq: 102 OPTIONS
Server: Sip EXpress router (0.9.5-pre1 (i386/linux))
Content-Length: 0
Warning: 392 204.147.183.18:5060 "Noisy feedback tells: pid=7028 req_src_ip=66.114.77.37 req_src_port=5060 in_uri=sip:sip.stanaphone.com out_uri=sip:sip.stanaphone.com via_cnt==1"
--- (9 headers 0 lines) ---
Destroying call '713ec5bc3e2208795ecfb1ee4429f52c@192.168.1.101'
asterisk1*CLI>
<-- SIP read from 204.147.183.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK6c4019f6;rport=5060;received=66.114.77.37
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as67b56fe4
To: <sip:sip.stanaphone.com>;tag=4e487ad2f2fae333606267674b3a79b1.fa94
Call-ID: 314f2cf1375dc16640413b842302b89c@192.168.1.101
CSeq: 102 OPTIONS
Server: Sip EXpress router (0.9.5-pre1 (i386/linux))
Content-Length: 0
Warning: 392 204.147.183.18:5060 "Noisy feedback tells: pid=7033 req_src_ip=66.114.77.37 req_src_port=5060 in_uri=sip:sip.stanaphone.com out_uri=sip:sip.stanaphone.com via_cnt==1"
--- (9 headers 0 lines) ---
Destroying call '314f2cf1375dc16640413b842302b89c@192.168.1.101'
asterisk1*CLI>
<-- SIP read from 204.147.183.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK6cb96aa0;rport=5060;received=66.114.77.37
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as0eed64e1
To: <sip:sip.stanaphone.com>;tag=4e487ad2f2fae333606267674b3a79b1.e6a7
Call-ID: 63628d2a7e69a73e590fb7382f6b3ec0@192.168.1.101
CSeq: 102 OPTIONS
Server: Sip EXpress router (0.9.5-pre1 (i386/linux))
Content-Length: 0
Warning: 392 204.147.183.18:5060 "Noisy feedback tells: pid=7038 req_src_ip=66.114.77.37 req_src_port=5060 in_uri=sip:sip.stanaphone.com out_uri=sip:sip.stanaphone.com via_cnt==1"
--- (9 headers 0 lines) ---
Destroying call '63628d2a7e69a73e590fb7382f6b3ec0@192.168.1.101'
asterisk1*CLI>
<-- SIP read from 204.147.183.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK3cf0d63d;rport=5060;received=66.114.77.37
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as326072c1
To: <sip:sip.stanaphone.com>;tag=4e487ad2f2fae333606267674b3a79b1.7abb
Call-ID: 3894b134638dd83a7a7957ee50efd855@192.168.1.101
CSeq: 102 OPTIONS
Server: Sip EXpress router (0.9.5-pre1 (i386/linux))
Content-Length: 0
Warning: 392 204.147.183.18:5060 "Noisy feedback tells: pid=7033 req_src_ip=66.114.77.37 req_src_port=5060 in_uri=sip:sip.stanaphone.com out_uri=sip:sip.stanaphone.com via_cnt==1"
--- (9 headers 0 lines) ---
Destroying call '3894b134638dd83a7a7957ee50efd855@192.168.1.101'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.151:5060:
OPTIONS sip:2202@192.168.1.151:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK7da54d1a;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as626ea5eb
To: <sip:2202@192.168.1.151:5060>
Contact: <sip:Unknown@192.168.1.101>
Call-ID: 1c3782162c85b81b748a9f0519627ee6@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 27 Jan 2007 01:36:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.152:5060:
OPTIONS sip:2203@192.168.1.152:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK550d6422;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as0835e077
To: <sip:2203@192.168.1.152:5060>
Contact: <sip:Unknown@192.168.1.101>
Call-ID: 0ec0db843792fe630fa8e2ab572f9930@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 27 Jan 2007 01:36:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.1.150:5060:
OPTIONS sip:2311@192.168.1.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK78ab5137;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as5a3017f6
To: <sip:2311@192.168.1.150:5060>
Contact: <sip:Unknown@192.168.1.101>
Call-ID: 562644e165d57f4c2ad4a182128a1166@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 27 Jan 2007 01:36:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.1.151:50372:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK7da54d1a;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as626ea5eb
To: <sip:2202@192.168.1.151:5060>;tag=0012d9a2268241b4469e9a58-59fe95bb
Call-ID: 1c3782162c85b81b748a9f0519627ee6@192.168.1.101
Date: Sat, 27 Jan 2007 01:36:23 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 247
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.1.151
s=SIP Call
c=IN IP4 192.168.1.151
t=0 0
m=audio 1 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (11 headers 11 lines) ---
Destroying call '1c3782162c85b81b748a9f0519627ee6@192.168.1.101'
asterisk1*CLI>
<-- SIP read from 192.168.1.152:52430:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK550d6422;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as0835e077
To: <sip:2203@192.168.1.152:5060>;tag=0012d9bb1f0041980ec2d6c8-0377fa9a
Call-ID: 0ec0db843792fe630fa8e2ab572f9930@192.168.1.101
Date: Sat, 27 Jan 2007 01:36:23 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 247
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.1.152
s=SIP Call
c=IN IP4 192.168.1.152
t=0 0
m=audio 1 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (11 headers 11 lines) ---
Destroying call '0ec0db843792fe630fa8e2ab572f9930@192.168.1.101'
12 headers, 0 lines
Reliably Transmitting (NAT) to 67.55.159.160:5060:
OPTIONS sip:outbound.freedigits.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK32be2762;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as44a724b7
To: <sip:outbound.freedigits.net>
Contact: <sip:Unknown@192.168.1.101>
Call-ID: 54a8e0db7d681c9b71d2b3de00c018c0@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 27 Jan 2007 01:36:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
12 headers, 0 lines
Reliably Transmitting (NAT) to 67.55.159.160:5060:
OPTIONS sip:outbound.freedigits.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK6c060369;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as6bfc24bd
To: <sip:outbound.freedigits.net>
Contact: <sip:Unknown@192.168.1.101>
Call-ID: 60e85d4079bc5b2320f9efa4535a6265@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 27 Jan 2007 01:36:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.1.150:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK78ab5137;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as5a3017f6
To: <sip:2311@192.168.1.150:5060>;tag=0012d9bb1da84a0d11085be5-0bba89b7
Call-ID: 562644e165d57f4c2ad4a182128a1166@192.168.1.101
Date: Sat, 27 Jan 2007 01:36:24 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 247
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.1.150
s=SIP Call
c=IN IP4 192.168.1.150
t=0 0
m=audio 1 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (11 headers 11 lines) ---
Destroying call '562644e165d57f4c2ad4a182128a1166@192.168.1.101'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.151:5060:
OPTIONS sip:202@192.168.1.151:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK41bf3a2c;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as1fc24813
To: <sip:202@192.168.1.151:5060>
Contact: <sip:Unknown@192.168.1.101>
Call-ID: 57b04c60737d8f1c09005b856b6f0170@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 27 Jan 2007 01:36:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.152:5060:
OPTIONS sip:203@192.168.1.152:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK6161435b;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as49ef6a0b
To: <sip:203@192.168.1.152:5060>
Contact: <sip:Unknown@192.168.1.101>
Call-ID: 44fee4972d26660909504530118625a7@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 27 Jan 2007 01:36:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.150:5060:
OPTIONS sip:2211@192.168.1.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK5d9ee270;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as6f7c430f
To: <sip:2211@192.168.1.150:5060>
Contact: <sip:Unknown@192.168.1.101>
Call-ID: 77a5c60c00ca811014059aaa6e35b2a1@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 27 Jan 2007 01:36:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 67.55.159.160:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK32be2762;rport=5060;received=66.114.77.37
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as44a724b7
To: <sip:outbound.freedigits.net>;tag=9eacff2ee73e6595c6126247c2fa2a6e.c789
Call-ID: 54a8e0db7d681c9b71d2b3de00c018c0@192.168.1.101
CSeq: 102 OPTIONS
Server: Sip EXpress router (0.9.3 (i386/linux))
Content-Length: 0
Warning: 392 67.55.159.160:5060 "Noisy feedback tells: pid=9064 req_src_ip=66.114.77.37 req_src_port=5060 in_uri=sip:outbound.freedigits.net out_uri=sip:outbound.freedigits.net via_cnt==1"
--- (9 headers 0 lines) ---
Destroying call '54a8e0db7d681c9b71d2b3de00c018c0@192.168.1.101'
asterisk1*CLI>
<-- SIP read from 67.55.159.160:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK6c060369;rport=5060;received=66.114.77.37
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as6bfc24bd
To: <sip:outbound.freedigits.net>;tag=9eacff2ee73e6595c6126247c2fa2a6e.a562
Call-ID: 60e85d4079bc5b2320f9efa4535a6265@192.168.1.101
CSeq: 102 OPTIONS
Server: Sip EXpress router (0.9.3 (i386/linux))
Content-Length: 0
Warning: 392 67.55.159.160:5060 "Noisy feedback tells: pid=9062 req_src_ip=66.114.77.37 req_src_port=5060 in_uri=sip:outbound.freedigits.net out_uri=sip:outbound.freedigits.net via_cnt==1"
--- (9 headers 0 lines) ---
Destroying call '60e85d4079bc5b2320f9efa4535a6265@192.168.1.101'
asterisk1*CLI>
<-- SIP read from 192.168.1.151:50373:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK41bf3a2c;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as1fc24813
To: <sip:202@192.168.1.151:5060>;tag=0012d9a2268241b501d97d9f-0865de42
Call-ID: 57b04c60737d8f1c09005b856b6f0170@192.168.1.101
Date: Sat, 27 Jan 2007 01:36:23 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 247
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.1.151
s=SIP Call
c=IN IP4 192.168.1.151
t=0 0
m=audio 1 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (11 headers 11 lines) ---
Destroying call '57b04c60737d8f1c09005b856b6f0170@192.168.1.101'
asterisk1*CLI>
<-- SIP read from 192.168.1.152:52431:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK6161435b;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as49ef6a0b
To: <sip:203@192.168.1.152:5060>;tag=0012d9bb1f00419902fc048c-7b65b164
Call-ID: 44fee4972d26660909504530118625a7@192.168.1.101
Date: Sat, 27 Jan 2007 01:36:23 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 247
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.1.152
s=SIP Call
c=IN IP4 192.168.1.152
t=0 0
m=audio 1 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (11 headers 11 lines) ---
Destroying call '44fee4972d26660909504530118625a7@192.168.1.101'
asterisk1*CLI>
<-- SIP read from 192.168.1.150:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK5d9ee270;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as6f7c430f
To: <sip:2211@192.168.1.150:5060>;tag=0012d9bb1da84a0e6210061f-5eaf8cd8
Call-ID: 77a5c60c00ca811014059aaa6e35b2a1@192.168.1.101
Date: Sat, 27 Jan 2007 01:36:24 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 247
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.1.150
s=SIP Call
c=IN IP4 192.168.1.150
t=0 0
m=audio 1 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (11 headers 11 lines) ---
Destroying call '77a5c60c00ca811014059aaa6e35b2a1@192.168.1.101'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.150:5060:
OPTIONS sip:211@192.168.1.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK1454a78d;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as1a38b547
To: <sip:211@192.168.1.150:5060>
Contact: <sip:Unknown@192.168.1.101>
Call-ID: 258f7c11111fc4bf25f98cf26aa0b1a9@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 27 Jan 2007 01:36:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.1.150:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK1454a78d;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as1a38b547
To: <sip:211@192.168.1.150:5060>;tag=0012d9bb1da84a0f3b6ba7d6-63127682
Call-ID: 258f7c11111fc4bf25f98cf26aa0b1a9@192.168.1.101
Date: Sat, 27 Jan 2007 01:36:24 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 247
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.1.150
s=SIP Call
c=IN IP4 192.168.1.150
t=0 0
m=audio 1 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (11 headers 11 lines) ---
Destroying call '258f7c11111fc4bf25f98cf26aa0b1a9@192.168.1.101'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 64.34.176.212:5060:
OPTIONS sip:did.voip.les.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK5aa14293;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as1ec4c81e
To: <sip:did.voip.les.net>
Contact: <sip:Unknown@192.168.1.101>
Call-ID: 150eba733b76a9c515614254015d6711@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 27 Jan 2007 01:36:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 64.34.176.212:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK5aa14293;received=66.114.77.37;rport=5060
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as1ec4c81e
To: <sip:did.voip.les.net>;tag=as07f28c88
Call-ID: 150eba733b76a9c515614254015d6711@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: LES.NET.VoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:64.34.176.212>
Accept: application/sdp
Content-Length: 0
--- (11 headers 0 lines) ---
Destroying call '150eba733b76a9c515614254015d6711@192.168.1.101'
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 64.34.176.212:5060:
REGISTER sip:did.voip.les.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK09ee8639;rport
From: <sip:1098912229@did.voip.les.net>;tag=as2a6fc141
To: <sip:1098912229@did.voip.les.net>
Call-ID: 6160a1621cbf0e2f3e949d516ce16d81@127.0.0.1
CSeq: 112 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="1098912229", realm="did.voip.les.net", algorithm=MD5, uri="sip:did.voip.les.net", nonce="05d235e6", response="a15f39cfff66f6a13fb49d66d0c29f15", opaque=""
Expires: 120
Contact: <sip:1098912229@192.168.1.101>
Event: registration
Content-Length: 0
---
<-- SIP read from 67.55.159.160:5060:
--- (0 headers 0 lines) Nat keepalive ---
<-- SIP read from 67.55.159.160:5060:
--- (0 headers 0 lines) Nat keepalive ---
asterisk1*CLI>
<-- SIP read from 64.34.176.212:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK09ee8639;received=66.114.77.37;rport=5060
From: <sip:1098912229@did.voip.les.net>;tag=as2a6fc141
To: <sip:1098912229@did.voip.les.net>
Call-ID: 6160a1621cbf0e2f3e949d516ce16d81@127.0.0.1
CSeq: 112 REGISTER
User-Agent: LES.NET.VoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1098912229@64.34.176.212>
Content-Length: 0
--- (10 headers 0 lines) ---
asterisk1*CLI>
<-- SIP read from 64.34.176.212:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK09ee8639;received=66.114.77.37;rport=5060
From: <sip:1098912229@did.voip.les.net>;tag=as2a6fc141
To: <sip:1098912229@did.voip.les.net>;tag=as6f0faa58
Call-ID: 6160a1621cbf0e2f3e949d516ce16d81@127.0.0.1
CSeq: 112 REGISTER
User-Agent: LES.NET.VoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1098912229@64.34.176.212>
WWW-Authenticate: Digest algorithm=MD5, realm="did.voip.les.net", nonce="3b3b1473"
Content-Length: 0
--- (11 headers 0 lines) ---
Responding to challenge, registration to domain/host name did.voip.les.net
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 64.34.176.212:5060:
REGISTER sip:did.voip.les.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK230b6c14;rport
From: <sip:1098912229@did.voip.les.net>;tag=as578b87c8
To: <sip:1098912229@did.voip.les.net>
Call-ID: 6160a1621cbf0e2f3e949d516ce16d81@127.0.0.1
CSeq: 113 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="1098912229", realm="did.voip.les.net", algorithm=MD5, uri="sip:did.voip.les.net", nonce="3b3b1473", response="13aea1856a9329b2c8421a5ea6fa57fe", opaque=""
Expires: 120
Contact: <sip:1098912229@192.168.1.101>
Event: registration
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 64.34.176.212:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK230b6c14;received=66.114.77.37;rport=5060
From: <sip:1098912229@did.voip.les.net>;tag=as578b87c8
To: <sip:1098912229@did.voip.les.net>
Call-ID: 6160a1621cbf0e2f3e949d516ce16d81@127.0.0.1
CSeq: 113 REGISTER
User-Agent: LES.NET.VoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1098912229@64.34.176.212>
Content-Length: 0
--- (10 headers 0 lines) ---
asterisk1*CLI>
<-- SIP read from 64.34.176.212:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK230b6c14;received=66.114.77.37;rport=5060
From: <sip:1098912229@did.voip.les.net>;tag=as578b87c8
To: <sip:1098912229@did.voip.les.net>;tag=as6f0faa58
Call-ID: 6160a1621cbf0e2f3e949d516ce16d81@127.0.0.1
CSeq: 113 REGISTER
User-Agent: LES.NET.VoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 120
Contact: <sip:1098912229@192.168.1.101>;expires=120
Date: Sat, 27 Jan 2007 01:36:31 GMT
Content-Length: 0
--- (12 headers 0 lines) ---
Scheduling destruction of call '6160a1621cbf0e2f3e949d516ce16d81@127.0.0.1' in 32000 ms
asterisk1*CLI>
<-- SIP read from 192.168.1.105:5060:
REGISTER sip:192.168.1.101 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060
From: <sip:204@192.168.1.101;user=phone>;tag=2902186237
To: <sip:204@192.168.1.101;user=phone>
Call-ID: 82145926@192.168.1.105
CSeq: 536550 REGISTER
Contact: <sip:204@192.168.1.105:5060;user=phone;transport=udp>;expires=60
User-Agent: Cisco ATA 186 v2.16 ata18x (030509d)
Authorization: Digest username="204",realm="asterisk",nonce="29ed22aa",uri="sip:192.168.1.101",response="5a1ebf7ecface2819f71a391bd1e1001"
Content-Length: 0
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.105 : 5060 (non-NAT)
Transmitting (NAT) to 192.168.1.105:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.105:5060;received=192.168.1.105
From: <sip:204@192.168.1.101;user=phone>;tag=2902186237
To: <sip:204@192.168.1.101;user=phone>
Call-ID: 82145926@192.168.1.105
CSeq: 536550 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:204@192.168.1.101>
Content-Length: 0
---
Transmitting (NAT) to 192.168.1.105:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.105:5060;received=192.168.1.105
From: <sip:204@192.168.1.101;user=phone>;tag=2902186237
To: <sip:204@192.168.1.101;user=phone>;tag=as742352ab
Call-ID: 82145926@192.168.1.105
CSeq: 536550 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="13e59abe"
Content-Length: 0
---
Scheduling destruction of call '82145926@192.168.1.105' in 15000 ms
asterisk1*CLI>
<-- SIP read from 192.168.1.105:5060:
REGISTER sip:192.168.1.101 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060
From: <sip:204@192.168.1.101;user=phone>;tag=2902186237
To: <sip:204@192.168.1.101;user=phone>
Call-ID: 82145926@192.168.1.105
CSeq: 536551 REGISTER
Contact: <sip:204@192.168.1.105:5060;user=phone;transport=udp>;expires=60
User-Agent: Cisco ATA 186 v2.16 ata18x (030509d)
Authorization: Digest username="204",realm="asterisk",nonce="13e59abe",uri="sip:192.168.1.101",response="be267bccc18b698a3b5de0d95dcb4b7a"
Content-Length: 0
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.105 : 5060 (NAT)
Transmitting (NAT) to 192.168.1.105:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.105:5060;received=192.168.1.105
From: <sip:204@192.168.1.101;user=phone>;tag=2902186237
To: <sip:204@192.168.1.101;user=phone>
Call-ID: 82145926@192.168.1.105
CSeq: 536551 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:204@192.168.1.101>
Content-Length: 0
---
Transmitting (NAT) to 192.168.1.105:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.105:5060;received=192.168.1.105
From: <sip:204@192.168.1.101;user=phone>;tag=2902186237
To: <sip:204@192.168.1.101;user=phone>;tag=as742352ab
Call-ID: 82145926@192.168.1.105
CSeq: 536551 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:204@192.168.1.105:5060;user=phone;transport=udp>;expires=60
Date: Sat, 27 Jan 2007 01:36:35 GMT
Content-Length: 0
---
Scheduling destruction of call '82145926@192.168.1.105' in 15000 ms
Destroying call '1da2add93e32325422751d8519e941be@127.0.0.1'
Destroying call '7e6d45ee4a7232646757920f7b5eaf73@127.0.0.1'
Destroying call '65900910301a7c8455f0f7a3626259a7@127.0.0.1'
asterisk1*CLI>
<-- SIP read from 67.55.159.160:5060:
--- (0 headers 0 lines) Nat keepalive ---
<-- SIP read from 67.55.159.160:5060:
--- (0 headers 0 lines) Nat keepalive ---
Destroying call '3fc5325e11c31f9678283dca23117741@127.0.0.1'
Destroying call '6515eb1d57b9b1c93120f87350e5a37c@127.0.0.1'
asterisk1*CLI>
<-- SIP read from 192.168.1.105:5060:
REGISTER sip:192.168.1.101 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060
From: <sip:201@192.168.1.101;user=phone>;tag=1311799237
To: <sip:201@192.168.1.101;user=phone>
Call-ID: 291072665@192.168.1.105
CSeq: 536550 REGISTER
Contact: <sip:201@192.168.1.105:5060;user=phone;transport=udp>;expires=60
User-Agent: Cisco ATA 186 v2.16 ata18x (030509d)
Authorization: Digest username="201",realm="asterisk",nonce="62a6b93e",uri="sip:192.168.1.101",response="1285415b9bf58d4acd7e854ba4cc212e"
Content-Length: 0
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.105 : 5060 (non-NAT)
Transmitting (NAT) to 192.168.1.105:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.105:5060;received=192.168.1.105
From: <sip:201@192.168.1.101;user=phone>;tag=1311799237
To: <sip:201@192.168.1.101;user=phone>
Call-ID: 291072665@192.168.1.105
CSeq: 536550 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:201@192.168.1.101>
Content-Length: 0
---
Transmitting (NAT) to 192.168.1.105:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.105:5060;received=192.168.1.105
From: <sip:201@192.168.1.101;user=phone>;tag=1311799237
To: <sip:201@192.168.1.101;user=phone>;tag=as2cc37150
Call-ID: 291072665@192.168.1.105
CSeq: 536550 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="390a8d6d"
Content-Length: 0
---
Scheduling destruction of call '291072665@192.168.1.105' in 15000 ms
asterisk1*CLI>
<-- SIP read from 192.168.1.105:5060:
REGISTER sip:192.168.1.101 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060
From: <sip:201@192.168.1.101;user=phone>;tag=1311799237
To: <sip:201@192.168.1.101;user=phone>
Call-ID: 291072665@192.168.1.105
CSeq: 536551 REGISTER
Contact: <sip:201@192.168.1.105:5060;user=phone;transport=udp>;expires=60
User-Agent: Cisco ATA 186 v2.16 ata18x (030509d)
Authorization: Digest username="201",realm="asterisk",nonce="390a8d6d",uri="sip:192.168.1.101",response="5cf2f6ef9ca9f39614ab2a992c3ce50f"
Content-Length: 0
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.105 : 5060 (NAT)
Transmitting (NAT) to 192.168.1.105:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.105:5060;received=192.168.1.105
From: <sip:201@192.168.1.101;user=phone>;tag=1311799237
To: <sip:201@192.168.1.101;user=phone>
Call-ID: 291072665@192.168.1.105
CSeq: 536551 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:201@192.168.1.101>
Content-Length: 0
---
Transmitting (NAT) to 192.168.1.105:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.105:5060;received=192.168.1.105
From: <sip:201@192.168.1.101;user=phone>;tag=1311799237
To: <sip:201@192.168.1.101;user=phone>;tag=as2cc37150
Call-ID: 291072665@192.168.1.105
CSeq: 536551 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:201@192.168.1.105:5060;user=phone;transport=udp>;expires=60
Date: Sat, 27 Jan 2007 01:36:50 GMT
Content-Length: 0
---
Scheduling destruction of call '291072665@192.168.1.105' in 15000 ms
Destroying call '82145926@192.168.1.105'
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.1.105:5060:
OPTIONS sip:204@192.168.1.105:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK669fa137;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as177b4dd8
To: <sip:204@192.168.1.105:5060;user=phone;transport=udp>
Contact: <sip:Unknown@192.168.1.101>
Call-ID: 5f72d48d29623e9817b6cb132242d9a3@192.168.1.101
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 27 Jan 2007 01:36:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.1.105:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK669fa137;rport
From: "Unknown" <sip:Unknown@192.168.1.101>;tag=as177b4dd8
To: <sip:204@192.168.1.105:5060;user=phone;transport=udp>;tag=1811405849
Call-ID: 5f72d48d29623e9817b6cb132242d9a3@192.168.1.101
CSeq: 102 OPTIONS
Server: Cisco ATA 186 v2.16 ata18x (030509d)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 250
Content-Type: application/sdp
v=0
o=204 129627 129627 IN IP4 192.168.1.105
s=ATA186 Call
c=IN IP4 192.168.1.105
t=0 0
m=audio 16384 RTP/AVP 0 8 4 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (10 headers 11 lines) ---
Destroying call '5f72d48d29623e9817b6cb132242d9a3@192.168.1.101'
asterisk1*CLI>
<-- SIP read from 67.55.159.160:5060:
--- (0 headers 0 lines) Nat keepalive ---
<-- SIP read from 67.55.159.160:5060:
--- (0 headers 0 lines) Nat keepalive ---
Destroying call '338e09871ea7477735852fd35327ddac@127.0.0.1'
Destroying call '6160a1621cbf0e2f3e949d516ce16d81@127.0.0.1'
Destroying call '291072665@192.168.1.105'
asterisk1*CLI>
<-- SIP read from 67.55.159.160:5060:
--- (0 headers 0 lines) Nat keepalive ---
asterisk1*CLI>
<-- SIP read from 67.55.159.160:5060:
--- (0 headers 0 lines) Nat keepalive ---
asterisk1*CLI>