AsteriskNOW new Phone outgoing no incoming Detailed LOGS help

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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK709c8d03;rport
From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as6385a1b6
To: sip:[email protected]{publicPhoneRouterIP}:1024
Contact: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]{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:[email protected]{publicAsteriskIP};tag=as6385a1b6
To: sip:[email protected]{publicPhoneRouterIP}:1024;tag=0003e311eac4022c6898960c-083ad84c
Call-ID: [email protected]{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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK709c8d03;rport
From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as6385a1b6
To: sip:[email protected]{publicPhoneRouterIP}:1024
Contact: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]{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:[email protected]{publicAsteriskIP};tag=as6385a1b6
To: sip:[email protected]{publicPhoneRouterIP}:1024;tag=0003e311eac4022d36ac771e-1785250b
Call-ID: [email protected]{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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK709c8d03;rport
From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as6385a1b6
To: sip:[email protected]{publicPhoneRouterIP}:1024
Contact: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]{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:[email protected]{publicAsteriskIP};tag=as6385a1b6
To: sip:[email protected]{publicPhoneRouterIP}:1024;tag=0003e311eac4022e771e122e-627abcb0
Call-ID: [email protected]{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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK709c8d03;rport
From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as6385a1b6
To: sip:[email protected]{publicPhoneRouterIP}:1024
Contact: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]{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:[email protected]{publicAsteriskIP};tag=as6385a1b6
To: sip:[email protected]{publicPhoneRouterIP}:1024;tag=0003e311eac4022f74e5b736-0dad3cc3
Call-ID: [email protected]{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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK709c8d03;rport
From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as6385a1b6
To: sip:[email protected]{publicPhoneRouterIP}:1024
Contact: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]{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:[email protected]{publicAsteriskIP};tag=as6385a1b6
To: sip:[email protected]{publicPhoneRouterIP}:1024;tag=0003e311eac4023078a06d03-57092a37
Call-ID: [email protected]{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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK535288a1;rport
From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as4cf6e26e
To: sip:[email protected]{publicPhoneRouterIP}:1024
Contact: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]{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:[email protected]{publicAsteriskIP};tag=as4cf6e26e
To: sip:[email protected]{publicPhoneRouterIP}:1024;tag=0003e311eac4023154f4d16f-39904b22
Call-ID: [email protected]{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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK535288a1;rport
From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as4cf6e26e
To: sip:[email protected]{publicPhoneRouterIP}:1024
Contact: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]{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:[email protected]{publicAsteriskIP};tag=as4cf6e26e
To: sip:[email protected]{publicPhoneRouterIP}:1024;tag=0003e311eac4023212b349c0-0f00a011
Call-ID: [email protected]{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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK535288a1;rport
From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as4cf6e26e
To: sip:[email protected]{publicPhoneRouterIP}:1024
Contact: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]{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:[email protected]{publicAsteriskIP};tag=as4cf6e26e
To: sip:[email protected]{publicPhoneRouterIP}:1024;tag=0003e311eac402332532bcaa-1ed339ba
Call-ID: [email protected]{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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK535288a1;rport
From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as4cf6e26e
To: sip:[email protected]{publicPhoneRouterIP}:1024
Contact: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]{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:[email protected]{publicAsteriskIP};tag=as4cf6e26e
To: sip:[email protected]{publicPhoneRouterIP}:1024;tag=0003e311eac40234740250bd-0330d32f
Call-ID: [email protected]{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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK535288a1;rport
From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as4cf6e26e
To: sip:[email protected]{publicPhoneRouterIP}:1024
Contact: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]{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:[email protected]{publicAsteriskIP};tag=as4cf6e26e
To: sip:[email protected]{publicPhoneRouterIP}:1024;tag=0003e311eac40235337f9f72-0fc18928
Call-ID: [email protected]{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:[email protected]{publicAsteriskIP}
To: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]
Date: Fri, 11 Sep 2009 01:07:53 GMT
CSeq: 160 REGISTER
User-Agent: CSCO/7
Contact: sip:[email protected]{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:[email protected]{publicAsteriskIP}
To: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]
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: [email protected], 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:[email protected]{publicAsteriskIP}
To: sip:[email protected]{publicAsteriskIP};tag=as124ade6b
Call-ID: 0003[email protected]
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: [email protected], 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:[email protected]{publicAsteriskIP}
To: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]
Date: Fri, 11 Sep 2009 01:07:54 GMT
CSeq: 161 REGISTER
User-Agent: CSCO/7
Contact: sip:[email protected]{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:[email protected]{publicAsteriskIP}
To: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]
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: [email protected], cseq=161, cseq_method=REGISTER

[20:07:54] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK2e1b28c2;rport
From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as47234428
To: sip:[email protected]{publicPhoneRouterIP}:1024
Contact: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]{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:[email protected]{publicAsteriskIP};tag=as47234428
To: sip:[email protected]{publicPhoneRouterIP}:1024;tag=0003e311eac402366f8c6065-14dbbb9d
Call-ID: [email protected]{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:[email protected]{publicAsteriskIP}
To: sip:[email protected]{publicAsteriskIP};tag=as124ade6b
Call-ID: [email protected]
CSeq: 161 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: sip:[email protected]{publicPhoneRouterIP}:1024;expires=60
Date: Fri, 11 Sep 2009 01:07:54 GMT
Content-Length: 0

, length=482

[20:07:54] sipSPICheckResponse: Response match: [email protected], cseq=161, cseq_method=REGISTER

[20:07:55] SIPProcessUDPMessage: recv UDP message from <{publicAsteriskIP}>:<50195>:
<OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK2e1b28c2;rport
From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as47234428
To: sip:[email protected]{publicPhoneRouterIP}:1024
Contact: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]{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:[email protected]{publicAsteriskIP};tag=as47234428
To: sip:[email protected]{publicPhoneRouterIP}:1024;tag=0003e311eac40237000e2b8f-1c6226d0
Call-ID: [email protected]{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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK2e1b28c2;rport
From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as47234428
To: sip:[email protected]{publicPhoneRouterIP}:1024
Contact: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]{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:[email protected]{publicAsteriskIP};tag=as47234428
To: sip:[email protected]{publicPhoneRouterIP}:1024;tag=0003e311eac40238265c7ff4-4afe1cda
Call-ID: [email protected]{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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK2e1b28c2;rport
From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as47234428
To: sip:[email protected]{publicPhoneRouterIP}:1024
Contact: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]{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:[email protected]{publicAsteriskIP};tag=as47234428
To: sip:[email protected]{publicPhoneRouterIP}:1024;tag=0003e311eac4023937c8b008-5eee0d81
Call-ID: [email protected]{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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK2e1b28c2;rport
From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as47234428
To: sip:[email protected]{publicPhoneRouterIP}:1024
Contact: sip:[email protected]{publicAsteriskIP}
Call-ID: [email protected]{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:[email protected]{publicAsteriskIP};tag=as47234428
To: sip:[email protected]{publicPhoneRouterIP}:1024;tag=0003e311eac4023a57eed7d2-495bc026
Call-ID: [email protected]{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:[email protected]{publicAsteriskIP} To: sip:[email protected]{publicAsteriskIP} Call-ID: [email protected] Date: Fri, 11 Sep 2009 03:36:28 GMT CSeq: 480 REGISTER User-Agent: CSCO/7 Contact: sip:[email protected]{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:[email protected]{publicAsteriskIP} To: sip:[email protected]{publicAsteriskIP} Call-ID: [email protected] 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:[email protected]{publicAsteriskIP} To: sip:[email protected]{publicAsteriskIP};tag=as3207e5db Call-ID: [email protected] 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 ‘[email protected]’ 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:[email protected]{publicAsteriskIP} To: sip:[email protected]{publicAsteriskIP} Call-ID: [email protected] Date: Fri, 11 Sep 2009 03:36:28 GMT CSeq: 481 REGISTER User-Agent: CSCO/7 Contact: sip:[email protected]{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:[email protected]{publicAsteriskIP} To: sip:[email protected]{publicAsteriskIP} Call-ID: [email protected] 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:[email protected]{publicPhoneRouterIP}:1024 SIP/2.0 Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK64e9721b From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as1116e461 To: sip:[email protected]{publicPhoneRouterIP}:1024 Contact: sip:[email protected]{publicAsteriskIP} Call-ID: [email protected]{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:[email protected]{publicAsteriskIP} To: sip:[email protected]{publicAsteriskIP};tag=as3207e5db Call-ID: [email protected] CSeq: 481 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: sip:[email protected]{publicPhoneRouterIP}:1024;expires=60 Date: Fri, 11 Sep 2009 03:36:28 GMT Content-Length: 0
<------------>

Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)

Retransmitting #1 (no NAT) to {publicPhoneRouterIP}:1024:
OPTIONS sip:[email protected]{publicPhoneRouterIP}:1024 SIP/2.0 Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK64e9721b From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as1116e461 To: sip:[email protected]{publicPhoneRouterIP}:1024 Contact: sip:[email protected]{publicAsteriskIP} Call-ID: [email protected]{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:[email protected]{publicPhoneRouterIP}:1024 SIP/2.0 Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK64e9721b From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as1116e461 To: sip:[email protected]{publicPhoneRouterIP}:1024 Contact: sip:[email protected]{publicAsteriskIP} Call-ID: [email protected]{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:[email protected]{publicPhoneRouterIP}:1024 SIP/2.0 Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK64e9721b From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as1116e461 To: sip:[email protected]{publicPhoneRouterIP}:1024 Contact: sip:[email protected]{publicAsteriskIP} Call-ID: [email protected]{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 ‘[email protected]’ Method: NOTIFY

Retransmitting #4 (no NAT) to {publicPhoneRouterIP}:1024:
OPTIONS sip:[email protected]{publicPhoneRouterIP}:1024 SIP/2.0 Via: SIP/2.0/UDP {publicAsteriskIP}:5060;branch=z9hG4bK64e9721b From: “Unknown” sip:[email protected]{publicAsteriskIP};tag=as1116e461 To: sip:[email protected]{publicPhoneRouterIP}:1024 Contact: sip:[email protected]{publicAsteriskIP} Call-ID: [email protected]{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 ‘[email protected]{publicAsteriskIP}’ Method: OPTIONS