New asterisknow cisco 7960 out but not in detailed logs w/in

Ok, I’m using a newly downloaded AsteriskNOW. It is on a public IP of {publicAsteriskIP}.
I have a cable modem connection at home with a putlic IP of {publicPhoneRouterIP}

I have a CISCO 7960 which has the internal ip of 192.168.2.3
The router of course has the gateway address of 192.168.2.1.

Below is a telnet debug from the CISCO phone and the debug from the Asterisk CLI.

It shows as registered on the FreePBX however it shows as unreachable.

The phone can make outgoing calls but not receive incoming calls.

From pouring through the Asterisk CLI i believe it’s because of the unreachable status. Asterisk sees this as a failure and makes no attempt to connect with the phone.

As you can see from the debug below, the phone is getting keepalives as 102 OPTIONS messages from the Asterisk server and sending 200 OK responses.
I never see these OK responses on the asterisk CLI in verbose mode.

You can also see that the phone sends re-register requests periodically. I DO SEE THESE requests on the Asterisk CLI as is evident from the responses received below.

Why do the phones register requests get to the asterisk server but the 200 OK responses do not. The message headers look identical in their IP:PORT configurations.

I also find it curious that every other REGISTER command from the phone is successful and then every other one is returned a 401 unauthorized response from asterisk.

Please help. I have added some line feeds to make reading the message packets easier.

replace real.public.asterisk.ip with {publicAsteriskIP}
replace real.public.phonerouter.ip with {publicPhoneRouterIP}


--------- logs from cisco phone ----------------------------------------

[20:07:26] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<OPTIONS sip:301@192.168.2.3:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK709c8d03;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as6385a1b6
To: sip:301@{publicPhoneRouterIP}:1024
Contact: sip:Unknown@{publicAsteriskIP}
Call-ID: 4a9bf9644e4bf9a61fa28d687d0e51a7@{publicAsteriskIP}
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 11 Sep 2009 01:07:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

, length=510

[20:07:26] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=OPTIONS, to_tag=<0003e311eac4022c6898960c-083ad84c>)

[20:07:26] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <{publicAsteriskIP}>:<5060>, handle = 8

[20:07:26] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <{publicAsteriskIP}>:<5060>, handle=<8>:
message=
<SIP/2.0 200 OK
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK709c8d03;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as6385a1b6
To: sip:301@{publicPhoneRouterIP}:1024;tag=0003e311eac4022c6898960c-083ad84c
Call-ID: 4a9bf9644e4bf9a61fa28d687d0e51a7@{publicAsteriskIP}
Date: Fri, 11 Sep 2009 01:07:26 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 243
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.2.3
s=SIP Call
c=IN IP4 192.168.2.3
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

, length=<702>

[20:07:26] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8

[20:07:27] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<OPTIONS sip:301@192.168.2.3:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK709c8d03;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as6385a1b6
To: sip:301@{publicPhoneRouterIP}:1024
Contact: sip:Unknown@{publicAsteriskIP}
Call-ID: 4a9bf9644e4bf9a61fa28d687d0e51a7@{publicAsteriskIP}
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 11 Sep 2009 01:07:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

, length=510

[20:07:27] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=OPTIONS, to_tag=<0003e311eac4022d36ac771e-1785250b>)

[20:07:27] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <{publicAsteriskIP}>:<5060>, handle = 8

[20:07:27] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <{publicAsteriskIP}>:<5060>, handle=<8>:
message=
<SIP/2.0 200 OK
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK709c8d03;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as6385a1b6
To: sip:301@{publicPhoneRouterIP}:1024;tag=0003e311eac4022d36ac771e-1785250b
Call-ID: 4a9bf9644e4bf9a61fa28d687d0e51a7@{publicAsteriskIP}
Date: Fri, 11 Sep 2009 01:07:27 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 243
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.2.3
s=SIP Call
c=IN IP4 192.168.2.3
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

, length=<702>

[20:07:27] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8

[20:07:28] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<OPTIONS sip:301@192.168.2.3:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK709c8d03;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as6385a1b6
To: sip:301@{publicPhoneRouterIP}:1024
Contact: sip:Unknown@{publicAsteriskIP}
Call-ID: 4a9bf9644e4bf9a61fa28d687d0e51a7@{publicAsteriskIP}
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 11 Sep 2009 01:07:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

, length=510

[20:07:28] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=OPTIONS, to_tag=<0003e311eac4022e771e122e-627abcb0>)

[20:07:28] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <{publicAsteriskIP}>:<5060>, handle = 8

[20:07:28] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <{publicAsteriskIP}>:<5060>, handle=<8>:
message=
<SIP/2.0 200 OK
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK709c8d03;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as6385a1b6
To: sip:301@{publicPhoneRouterIP}:1024;tag=0003e311eac4022e771e122e-627abcb0
Call-ID: 4a9bf9644e4bf9a61fa28d687d0e51a7@{publicAsteriskIP}
Date: Fri, 11 Sep 2009 01:07:28 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 243
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.2.3
s=SIP Call
c=IN IP4 192.168.2.3
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

, length=<702>

[20:07:28] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8

[20:07:29] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<OPTIONS sip:301@192.168.2.3:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK709c8d03;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as6385a1b6
To: sip:301@{publicPhoneRouterIP}:1024
Contact: sip:Unknown@{publicAsteriskIP}
Call-ID: 4a9bf9644e4bf9a61fa28d687d0e51a7@{publicAsteriskIP}
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 11 Sep 2009 01:07:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

, length=510

[20:07:29] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=OPTIONS, to_tag=<0003e311eac4022f74e5b736-0dad3cc3>)

[20:07:29] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <{publicAsteriskIP}>:<5060>, handle = 8

[20:07:29] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <{publicAsteriskIP}>:<5060>, handle=<8>:
message=
<SIP/2.0 200 OK
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK709c8d03;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as6385a1b6
To: sip:301@{publicPhoneRouterIP}:1024;tag=0003e311eac4022f74e5b736-0dad3cc3
Call-ID: 4a9bf9644e4bf9a61fa28d687d0e51a7@{publicAsteriskIP}
Date: Fri, 11 Sep 2009 01:07:29 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 243
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.2.3
s=SIP Call
c=IN IP4 192.168.2.3
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

, length=<702>

[20:07:29] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8

[20:07:30] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<OPTIONS sip:301@192.168.2.3:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK709c8d03;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as6385a1b6
To: sip:301@{publicPhoneRouterIP}:1024
Contact: sip:Unknown@{publicAsteriskIP}
Call-ID: 4a9bf9644e4bf9a61fa28d687d0e51a7@{publicAsteriskIP}
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 11 Sep 2009 01:07:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

, length=510

[20:07:30] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=OPTIONS, to_tag=<0003e311eac4023078a06d03-57092a37>)

[20:07:30] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <{publicAsteriskIP}>:<5060>, handle = 8

[20:07:30] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <{publicAsteriskIP}>:<5060>, handle=<8>:
message=
<SIP/2.0 200 OK
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK709c8d03;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as6385a1b6
To: sip:301@{publicPhoneRouterIP}:1024;tag=0003e311eac4023078a06d03-57092a37
Call-ID: 4a9bf9644e4bf9a61fa28d687d0e51a7@{publicAsteriskIP}
Date: Fri, 11 Sep 2009 01:07:30 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 243
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.2.3
s=SIP Call
c=IN IP4 192.168.2.3
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

, length=<702>

[20:07:30] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8

[20:07:40] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<OPTIONS sip:301@192.168.2.3:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK535288a1;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as4cf6e26e
To: sip:301@{publicPhoneRouterIP}:1024
Contact: sip:Unknown@{publicAsteriskIP}
Call-ID: 4ffa6fe7089a7a8c1b3b31b0380b24ee@{publicAsteriskIP}
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 11 Sep 2009 01:07:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

, length=510

[20:07:40] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=OPTIONS, to_tag=<0003e311eac4023154f4d16f-39904b22>)

[20:07:40] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <{publicAsteriskIP}>:<5060>, handle = 8

[20:07:40] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <{publicAsteriskIP}>:<5060>, handle=<8>:
message=
<SIP/2.0 200 OK
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK535288a1;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as4cf6e26e
To: sip:301@{publicPhoneRouterIP}:1024;tag=0003e311eac4023154f4d16f-39904b22
Call-ID: 4ffa6fe7089a7a8c1b3b31b0380b24ee@{publicAsteriskIP}
Date: Fri, 11 Sep 2009 01:07:40 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 243
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.2.3
s=SIP Call
c=IN IP4 192.168.2.3
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

, length=<702>

[20:07:40] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8

[20:07:41] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<OPTIONS sip:301@192.168.2.3:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK535288a1;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as4cf6e26e
To: sip:301@{publicPhoneRouterIP}:1024
Contact: sip:Unknown@{publicAsteriskIP}
Call-ID: 4ffa6fe7089a7a8c1b3b31b0380b24ee@{publicAsteriskIP}
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 11 Sep 2009 01:07:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

, length=510

[20:07:41] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=OPTIONS, to_tag=<0003e311eac4023212b349c0-0f00a011>)

[20:07:41] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <{publicAsteriskIP}>:<5060>, handle = 8

[20:07:41] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <{publicAsteriskIP}>:<5060>, handle=<8>:
message=
<SIP/2.0 200 OK
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK535288a1;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as4cf6e26e
To: sip:301@{publicPhoneRouterIP}:1024;tag=0003e311eac4023212b349c0-0f00a011
Call-ID: 4ffa6fe7089a7a8c1b3b31b0380b24ee@{publicAsteriskIP}
Date: Fri, 11 Sep 2009 01:07:41 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 243
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.2.3
s=SIP Call
c=IN IP4 192.168.2.3
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

, length=<702>

[20:07:41] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8

[20:07:42] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<OPTIONS sip:301@192.168.2.3:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK535288a1;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as4cf6e26e
To: sip:301@{publicPhoneRouterIP}:1024
Contact: sip:Unknown@{publicAsteriskIP}
Call-ID: 4ffa6fe7089a7a8c1b3b31b0380b24ee@{publicAsteriskIP}
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 11 Sep 2009 01:07:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

, length=510

[20:07:42] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=OPTIONS, to_tag=<0003e311eac402332532bcaa-1ed339ba>)

[20:07:42] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <{publicAsteriskIP}>:<5060>, handle = 8

[20:07:42] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <{publicAsteriskIP}>:<5060>, handle=<8>:
message=
<SIP/2.0 200 OK
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK535288a1;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as4cf6e26e
To: sip:301@{publicPhoneRouterIP}:1024;tag=0003e311eac402332532bcaa-1ed339ba
Call-ID: 4ffa6fe7089a7a8c1b3b31b0380b24ee@{publicAsteriskIP}
Date: Fri, 11 Sep 2009 01:07:42 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 243
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.2.3
s=SIP Call
c=IN IP4 192.168.2.3
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

, length=<702>

[20:07:42] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8

[20:07:43] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<OPTIONS sip:301@192.168.2.3:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK535288a1;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as4cf6e26e
To: sip:301@{publicPhoneRouterIP}:1024
Contact: sip:Unknown@{publicAsteriskIP}
Call-ID: 4ffa6fe7089a7a8c1b3b31b0380b24ee@{publicAsteriskIP}
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 11 Sep 2009 01:07:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

, length=510

[20:07:43] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=OPTIONS, to_tag=<0003e311eac40234740250bd-0330d32f>)

[20:07:43] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <{publicAsteriskIP}>:<5060>, handle = 8

[20:07:43] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <{publicAsteriskIP}>:<5060>, handle=<8>:
message=
<SIP/2.0 200 OK
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK535288a1;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as4cf6e26e
To: sip:301@{publicPhoneRouterIP}:1024;tag=0003e311eac40234740250bd-0330d32f
Call-ID: 4ffa6fe7089a7a8c1b3b31b0380b24ee@{publicAsteriskIP}
Date: Fri, 11 Sep 2009 01:07:43 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 243
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.2.3
s=SIP Call
c=IN IP4 192.168.2.3
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

, length=<702>

[20:07:43] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8

[20:07:44] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<OPTIONS sip:301@192.168.2.3:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK535288a1;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as4cf6e26e
To: sip:301@{publicPhoneRouterIP}:1024
Contact: sip:Unknown@{publicAsteriskIP}
Call-ID: 4ffa6fe7089a7a8c1b3b31b0380b24ee@{publicAsteriskIP}
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 11 Sep 2009 01:07:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

, length=510

[20:07:44] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=OPTIONS, to_tag=<0003e311eac40235337f9f72-0fc18928>)

[20:07:44] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <{publicAsteriskIP}>:<5060>, handle = 8

[20:07:44] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <{publicAsteriskIP}>:<5060>, handle=<8>:
message=
<SIP/2.0 200 OK
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK535288a1;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as4cf6e26e
To: sip:301@{publicPhoneRouterIP}:1024;tag=0003e311eac40235337f9f72-0fc18928
Call-ID: 4ffa6fe7089a7a8c1b3b31b0380b24ee@{publicAsteriskIP}
Date: Fri, 11 Sep 2009 01:07:44 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 243
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.2.3
s=SIP Call
c=IN IP4 192.168.2.3
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

, length=<702>

[20:07:44] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8

[20:07:53] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <{publicAsteriskIP}>:<5060>, handle = 8

[20:07:53] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <{publicAsteriskIP}>:<5060>, handle=<8>:
message=
<REGISTER sip:{publicAsteriskIP} SIP/2.0
Via: SIP/2.0/UDP {publicPhoneRouterIP}:5060;branch=z9hG4bK06160927
From: sip:301@{publicAsteriskIP}
To: sip:301@{publicAsteriskIP}
Call-ID: 0003e311-eac40002-39b26751-41a71789@192.168.2.1
Date: Fri, 11 Sep 2009 01:07:53 GMT
CSeq: 160 REGISTER
User-Agent: CSCO/7
Contact: sip:301@{publicPhoneRouterIP}:5060
Content-Length: 0
Expires: 60

, length=<361>

[20:07:53] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8

[20:07:53] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK06160927;received={publicPhoneRouterIP}
From: sip:301@{publicAsteriskIP}
To: sip:301@{publicAsteriskIP}
Call-ID: 0003e311-eac40002-39b26751-41a71789@192.168.2.1
CSeq: 160 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

, length=372

[20:07:53] sipSPICheckResponse: Response match: callid=0003e311-eac40002-39b26751-41a71789@192.168.2.1, cseq=160, cseq_method=REGISTER

[20:07:54] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK06160927;received={publicPhoneRouterIP}
From: sip:301@{publicAsteriskIP}
To: sip:301@{publicAsteriskIP};tag=as124ade6b
Call-ID: 0003e311-eac40002-39b26751-41a71789@192.168.2.1
CSeq: 160 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="75d08021"
Content-Length: 0

, length=469

[20:07:54] sipSPICheckResponse: Response match: callid=0003e311-eac40002-39b26751-41a71789@192.168.2.1, cseq=160, cseq_method=REGISTER

[20:07:54] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <{publicAsteriskIP}>:<5060>, handle = 8

[20:07:54] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <{publicAsteriskIP}>:<5060>, handle=<8>:
message=
<REGISTER sip:{publicAsteriskIP} SIP/2.0
Via: SIP/2.0/UDP {publicPhoneRouterIP}:5060;branch=z9hG4bK205af4c4
From: sip:301@{publicAsteriskIP}
To: sip:301@{publicAsteriskIP}
Call-ID: 0003e311-eac40002-39b26751-41a71789@192.168.2.1
Date: Fri, 11 Sep 2009 01:07:54 GMT
CSeq: 161 REGISTER
User-Agent: CSCO/7
Contact: sip:301@{publicPhoneRouterIP}:5060
Authorization: Digest username=“301”,realm=“asterisk”,uri=“sip:{publicAsteriskIP}”,response=“72edcb19fe93f059fbac9053d9220e57”,nonce=“75d08021”,algorithm=MD5
Content-Length: 0
Expires: 60

, length=<516>

[20:07:54] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8

[20:07:54] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK205af4c4;received={publicPhoneRouterIP}
From: sip:301@{publicAsteriskIP}
To: sip:301@{publicAsteriskIP}
Call-ID: 0003e311-eac40002-39b26751-41a71789@192.168.2.1
CSeq: 161 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

, length=372

[20:07:54] sipSPICheckResponse: Response match: callid=0003e311-eac40002-39b26751-41a71789@192.168.2.1, cseq=161, cseq_method=REGISTER

[20:07:54] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<OPTIONS sip:301@192.168.2.3:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK2e1b28c2;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as47234428
To: sip:301@{publicPhoneRouterIP}:1024
Contact: sip:Unknown@{publicAsteriskIP}
Call-ID: 4db78814773e69ee4e63bce31712320a@{publicAsteriskIP}
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 11 Sep 2009 01:07:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

, length=510

[20:07:54] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=OPTIONS, to_tag=<0003e311eac402366f8c6065-14dbbb9d>)

[20:07:54] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <{publicAsteriskIP}>:<5060>, handle = 8

[20:07:54] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <{publicAsteriskIP}>:<5060>, handle=<8>:
message=
<SIP/2.0 200 OK
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK2e1b28c2;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as47234428
To: sip:301@{publicPhoneRouterIP}:1024;tag=0003e311eac402366f8c6065-14dbbb9d
Call-ID: 4db78814773e69ee4e63bce31712320a@{publicAsteriskIP}
Date: Fri, 11 Sep 2009 01:07:54 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 243
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.2.3
s=SIP Call
c=IN IP4 192.168.2.3
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

, length=<702>

[20:07:54] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8

[20:07:54] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK205af4c4;received={publicPhoneRouterIP}
From: sip:301@{publicAsteriskIP}
To: sip:301@{publicAsteriskIP};tag=as124ade6b
Call-ID: 0003e311-eac40002-39b26751-41a71789@192.168.2.1
CSeq: 161 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: sip:301@{publicPhoneRouterIP}:1024;expires=60
Date: Fri, 11 Sep 2009 01:07:54 GMT
Content-Length: 0

, length=482

[20:07:54] sipSPICheckResponse: Response match: callid=0003e311-eac40002-39b26751-41a71789@192.168.2.1, cseq=161, cseq_method=REGISTER

[20:07:55] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<OPTIONS sip:301@192.168.2.3:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK2e1b28c2;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as47234428
To: sip:301@{publicPhoneRouterIP}:1024
Contact: sip:Unknown@{publicAsteriskIP}
Call-ID: 4db78814773e69ee4e63bce31712320a@{publicAsteriskIP}
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 11 Sep 2009 01:07:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

, length=510

[20:07:55] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=OPTIONS, to_tag=<0003e311eac40237000e2b8f-1c6226d0>)

[20:07:55] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <{publicAsteriskIP}>:<5060>, handle = 8

[20:07:55] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <{publicAsteriskIP}>:<5060>, handle=<8>:
message=
<SIP/2.0 200 OK
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK2e1b28c2;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as47234428
To: sip:301@{publicPhoneRouterIP}:1024;tag=0003e311eac40237000e2b8f-1c6226d0
Call-ID: 4db78814773e69ee4e63bce31712320a@{publicAsteriskIP}
Date: Fri, 11 Sep 2009 01:07:55 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 243
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.2.3
s=SIP Call
c=IN IP4 192.168.2.3
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

, length=<702>

[20:07:55] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8

[20:07:56] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<OPTIONS sip:301@192.168.2.3:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK2e1b28c2;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as47234428
To: sip:301@{publicPhoneRouterIP}:1024
Contact: sip:Unknown@{publicAsteriskIP}
Call-ID: 4db78814773e69ee4e63bce31712320a@{publicAsteriskIP}
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 11 Sep 2009 01:07:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

, length=510

[20:07:56] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=OPTIONS, to_tag=<0003e311eac40238265c7ff4-4afe1cda>)

[20:07:56] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <{publicAsteriskIP}>:<5060>, handle = 8

[20:07:56] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <{publicAsteriskIP}>:<5060>, handle=<8>:
message=
<SIP/2.0 200 OK
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK2e1b28c2;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as47234428
To: sip:301@{publicPhoneRouterIP}:1024;tag=0003e311eac40238265c7ff4-4afe1cda
Call-ID: 4db78814773e69ee4e63bce31712320a@{publicAsteriskIP}
Date: Fri, 11 Sep 2009 01:07:56 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 243
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.2.3
s=SIP Call
c=IN IP4 192.168.2.3
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

, length=<702>

[20:07:56] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8

[20:07:57] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<OPTIONS sip:301@192.168.2.3:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK2e1b28c2;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as47234428
To: sip:301@{publicPhoneRouterIP}:1024
Contact: sip:Unknown@{publicAsteriskIP}
Call-ID: 4db78814773e69ee4e63bce31712320a@{publicAsteriskIP}
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 11 Sep 2009 01:07:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

, length=510

[20:07:57] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=OPTIONS, to_tag=<0003e311eac4023937c8b008-5eee0d81>)

[20:07:57] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <{publicAsteriskIP}>:<5060>, handle = 8

[20:07:57] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <{publicAsteriskIP}>:<5060>, handle=<8>:
message=
<SIP/2.0 200 OK
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK2e1b28c2;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as47234428
To: sip:301@{publicPhoneRouterIP}:1024;tag=0003e311eac4023937c8b008-5eee0d81
Call-ID: 4db78814773e69ee4e63bce31712320a@{publicAsteriskIP}
Date: Fri, 11 Sep 2009 01:07:57 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 243
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.2.3
s=SIP Call
c=IN IP4 192.168.2.3
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

, length=<702>

[20:07:57] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8

[20:07:58] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<OPTIONS sip:301@192.168.2.3:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK2e1b28c2;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as47234428
To: sip:301@{publicPhoneRouterIP}:1024
Contact: sip:Unknown@{publicAsteriskIP}
Call-ID: 4db78814773e69ee4e63bce31712320a@{publicAsteriskIP}
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 11 Sep 2009 01:07:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

, length=510

[20:07:58] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=OPTIONS, to_tag=<0003e311eac4023a57eed7d2-495bc026>)

[20:07:58] sip_sm_cc_channel_send_buf_fcn: Opened a one-time UDP send channel to server <{publicAsteriskIP}>:<5060>, handle = 8

[20:07:58] sip_sm_cc_channel_send_buf_fcn:Sent SIP message to <{publicAsteriskIP}>:<5060>, handle=<8>:
message=
<SIP/2.0 200 OK
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK2e1b28c2;rport
From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as47234428
To: sip:301@{publicPhoneRouterIP}:1024;tag=0003e311eac4023a57eed7d2-495bc026
Call-ID: 4db78814773e69ee4e63bce31712320a@{publicAsteriskIP}
Date: Fri, 11 Sep 2009 01:07:58 GMT
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 243
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER
v=0
o=Cisco-SIPUA (null) (null) IN IP4 192.168.2.3
s=SIP Call
c=IN IP4 192.168.2.3
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

, length=<702>

[20:07:58] sip_sm_cc_channel_send_buf_fcn: Closed a one-time UDP send channel handle = 8


---------- logs from asterisk cli ------------------------------------------

<— SIP read from {publicPhoneRouterIP}:1024 —>
REGISTER sip:{publicAsteriskIP} SIP/2.0 Via: SIP/2.0/UDP {publicPhoneRouterIP}:1024;branch=z9hG4bK0183776f From: sip:301@{publicAsteriskIP} To: sip:301@{publicAsteriskIP} Call-ID: 0003e311-eac40002-39b26751-41a71789@192.168.2.1 Date: Fri, 11 Sep 2009 03:36:28 GMT CSeq: 480 REGISTER User-Agent: CSCO/7 Contact: sip:301@{publicPhoneRouterIP}:1024 Content-Length: 0 Expires: 60
<------------->
— (11 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to {publicPhoneRouterIP} : 1024 (NAT)

<— Transmitting (no NAT) to {publicPhoneRouterIP}:1024 —>
SIP/2.0 100 Trying Via: SIP/2.0/UDP {publicPhoneRouterIP}:1024;branch=z9hG4bK0183776f;received={publicPhoneRouterIP} From: sip:301@{publicAsteriskIP} To: sip:301@{publicAsteriskIP} Call-ID: 0003e311-eac40002-39b26751-41a71789@192.168.2.1 CSeq: 480 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0
<------------>

<— Transmitting (no NAT) to {publicPhoneRouterIP}:1024 —>
SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP {publicPhoneRouterIP}:1024;branch=z9hG4bK0183776f;received={publicPhoneRouterIP} From: sip:301@{publicAsteriskIP} To: sip:301@{publicAsteriskIP};tag=as3207e5db Call-ID: 0003e311-eac40002-39b26751-41a71789@192.168.2.1 CSeq: 480 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“0d214f17” Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘0003e311-eac40002-39b26751-41a71789@192.168.2.1’ in 32000 ms (Method: REGISTER)

<— SIP read from {publicPhoneRouterIP}:1024 —>
REGISTER sip:{publicAsteriskIP} SIP/2.0 Via: SIP/2.0/UDP {publicPhoneRouterIP}:1024;branch=z9hG4bK511e9420 From: sip:301@{publicAsteriskIP} To: sip:301@{publicAsteriskIP} Call-ID: 0003e311-eac40002-39b26751-41a71789@192.168.2.1 Date: Fri, 11 Sep 2009 03:36:28 GMT CSeq: 481 REGISTER User-Agent: CSCO/7 Contact: sip:301@{publicPhoneRouterIP}:1024 Authorization: Digest username=“301”,realm=“asterisk”,uri=“sip:{publicAsteriskIP}”,response=“1b68c5bb9d07650a6c4e3f522b53da4f”,nonce=“0d214f17”,algorithm=MD5 Content-Length: 0 Expires: 60
<------------->
— (12 headers 0 lines) —
Using latest REGISTER request as basis request

Sending to {publicPhoneRouterIP} : 1024 (no NAT)

<— Transmitting (no NAT) to {publicPhoneRouterIP}:1024 —>
SIP/2.0 100 Trying Via: SIP/2.0/UDP {publicPhoneRouterIP}:1024;branch=z9hG4bK511e9420;received={publicPhoneRouterIP} From: sip:301@{publicAsteriskIP} To: sip:301@{publicAsteriskIP} Call-ID: 0003e311-eac40002-39b26751-41a71789@192.168.2.1 CSeq: 481 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0
<------------>

Reliably Transmitting (no NAT) to {publicPhoneRouterIP}:1024:
OPTIONS sip:301@{publicPhoneRouterIP}:1024 SIP/2.0 Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK64e9721b From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as1116e461 To: sip:301@{publicPhoneRouterIP}:1024 Contact: sip:Unknown@{publicAsteriskIP} Call-ID: 21634c9b2af814c666215a38484bca11@{publicAsteriskIP} CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 11 Sep 2009 03:36:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0

<— Transmitting (no NAT) to {publicPhoneRouterIP}:1024 —>
SIP/2.0 200 OK Via: SIP/2.0/UDP {publicPhoneRouterIP}:1024;branch=z9hG4bK511e9420;received={publicPhoneRouterIP} From: sip:301@{publicAsteriskIP} To: sip:301@{publicAsteriskIP};tag=as3207e5db Call-ID: 0003e311-eac40002-39b26751-41a71789@192.168.2.1 CSeq: 481 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: sip:301@{publicPhoneRouterIP}:1024;expires=60 Date: Fri, 11 Sep 2009 03:36:28 GMT Content-Length: 0
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Scheduling destruction of SIP dialog ‘0003e311-eac40002-39b26751-41a71789@192.168.2.1’ in 32000 ms (Method: REGISTER)

Retransmitting #1 (no NAT) to {publicPhoneRouterIP}:1024:
OPTIONS sip:301@{publicPhoneRouterIP}:1024 SIP/2.0 Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK64e9721b From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as1116e461 To: sip:301@{publicPhoneRouterIP}:1024 Contact: sip:Unknown@{publicAsteriskIP} Call-ID: 21634c9b2af814c666215a38484bca11@{publicAsteriskIP} CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 11 Sep 2009 03:36:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0

Retransmitting #2 (no NAT) to {publicPhoneRouterIP}:1024:
OPTIONS sip:301@{publicPhoneRouterIP}:1024 SIP/2.0 Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK64e9721b From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as1116e461 To: sip:301@{publicPhoneRouterIP}:1024 Contact: sip:Unknown@{publicAsteriskIP} Call-ID: 21634c9b2af814c666215a38484bca11@{publicAsteriskIP} CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 11 Sep 2009 03:36:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0

Retransmitting #3 (no NAT) to {publicPhoneRouterIP}:1024:
OPTIONS sip:301@{publicPhoneRouterIP}:1024 SIP/2.0 Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK64e9721b From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as1116e461 To: sip:301@{publicPhoneRouterIP}:1024 Contact: sip:Unknown@{publicAsteriskIP} Call-ID: 21634c9b2af814c666215a38484bca11@{publicAsteriskIP} CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 11 Sep 2009 03:36:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0

Really destroying SIP dialog ‘1368a1bb216cd1185abbeeab3449f7b3@127.0.0.1’ Method: NOTIFY

Retransmitting #4 (no NAT) to {publicPhoneRouterIP}:1024:
OPTIONS sip:301@{publicPhoneRouterIP}:1024 SIP/2.0 Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK64e9721b From: “Unknown” sip:Unknown@{publicAsteriskIP};tag=as1116e461 To: sip:301@{publicPhoneRouterIP}:1024 Contact: sip:Unknown@{publicAsteriskIP} Call-ID: 21634c9b2af814c666215a38484bca11@{publicAsteriskIP} CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 11 Sep 2009 03:36:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0

Really destroying SIP dialog ‘21634c9b2af814c666215a38484bca11@{publicAsteriskIP}’ Method: OPTIONS