rendered paste body[13]
username=13
type=friend
secret=123
qualify=yes
port=5060
nat=1
mailbox=13@device
host=dynamic
dtmfmode=rfc2833
context=outgoing
canreinvite=yes
Debug output ------------------
Retransmitting #1 (no NAT) to 66.112.182.11:5060:
OPTIONS sip:13@192.168.1.143:5060 SIP/2.0
Via: SIP/2.0/UDP 66.112.182.171:5060;branch=z9hG4bK24e141f0;rport
From: "Unknown" <sip:Unknown@66.112.182.171>;tag=as7592b36c
To: <sip:13@192.168.1.143:5060>
Contact: <sip:Unknown@66.112.182.171>
Call-ID: 028138744c6fb5c458ba23ea34aa8a0e@66.112.182.171
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 09 Jul 2006 20:20:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.1.143:5060:
REGISTER sip:hmsinet.aleph-com.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.143:5060;branch=z9hG4bKe764022c71791914
From: "Darren Wiebe" <sip:13@hmsinet.aleph-com.net>;tag=b4b5e160041ac66b
To: <sip:13@hmsinet.aleph-com.net>
Contact: *
Call-ID: 8d63a2838d7a5c51@192.168.1.143
CSeq: 10001 REGISTER
Expires: 0
User-Agent: Grandstream GXP2000 1.1.0.13
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
--- (12 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.1.143 : 5060 (NAT)
Transmitting (NAT) to 192.168.1.143:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.143:5060;branch=z9hG4bKe764022c71791914;received=192.168.1.143
From: "Darren Wiebe" <sip:13@hmsinet.aleph-com.net>;tag=b4b5e160041ac66b
To: <sip:13@hmsinet.aleph-com.net>
Call-ID: 8d63a2838d7a5c51@192.168.1.143
CSeq: 10001 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:13@66.112.182.171>
Content-Length: 0
---
Transmitting (NAT) to 192.168.1.143:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.143:5060;branch=z9hG4bKe764022c71791914;received=192.168.1.143
From: "Darren Wiebe" <sip:13@hmsinet.aleph-com.net>;tag=b4b5e160041ac66b
To: <sip:13@hmsinet.aleph-com.net>;tag=as1f7008a7
Call-ID: 8d63a2838d7a5c51@192.168.1.143
CSeq: 10001 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:13@66.112.182.171>
WWW-Authenticate: Digest realm="asterisk", nonce="5a870435"
Content-Length: 0
---
Scheduling destruction of call '8d63a2838d7a5c51@192.168.1.143' in 15000 ms
Retransmitting #2 (no NAT) to 66.112.182.11:5060:
OPTIONS sip:13@192.168.1.143:5060 SIP/2.0
Via: SIP/2.0/UDP 66.112.182.171:5060;branch=z9hG4bK24e141f0;rport
From: "Unknown" <sip:Unknown@66.112.182.171>;tag=as7592b36c
To: <sip:13@192.168.1.143:5060>
Contact: <sip:Unknown@66.112.182.171>
Call-ID: 028138744c6fb5c458ba23ea34aa8a0e@66.112.182.171
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 09 Jul 2006 20:20:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Retransmitting #3 (no NAT) to 66.112.182.11:5060:
OPTIONS sip:13@192.168.1.143:5060 SIP/2.0
Via: SIP/2.0/UDP 66.112.182.171:5060;branch=z9hG4bK24e141f0;rport
From: "Unknown" <sip:Unknown@66.112.182.171>;tag=as7592b36c
To: <sip:13@192.168.1.143:5060>
Contact: <sip:Unknown@66.112.182.171>
Call-ID: 028138744c6fb5c458ba23ea34aa8a0e@66.112.182.171
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 09 Jul 2006 20:20:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Retransmitting #4 (no NAT) to 66.112.182.11:5060:
OPTIONS sip:13@192.168.1.143:5060 SIP/2.0
Via: SIP/2.0/UDP 66.112.182.171:5060;branch=z9hG4bK24e141f0;rport
From: "Unknown" <sip:Unknown@66.112.182.171>;tag=as7592b36c
To: <sip:13@192.168.1.143:5060>
Contact: <sip:Unknown@66.112.182.171>
Call-ID: 028138744c6fb5c458ba23ea34aa8a0e@66.112.182.171
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 09 Jul 2006 20:20:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
------------------- Here's where I try placing a call
---
Transmitting (NAT) to 192.168.1.143:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.143:5060;branch=z9hG4bK64a528a22f9a31b5;received=192.168.1.143
From: "Darren Wiebe" <sip:13@hmsinet.aleph-com.net>;tag=b4b5e160041ac66b
To: <sip:13@hmsinet.aleph-com.net>;tag=as1f7008a7
Call-ID: 8d63a2838d7a5c51@192.168.1.143
CSeq: 10001 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:13@66.112.182.171>
WWW-Authenticate: Digest realm="asterisk", nonce="5a870435"
Content-Length: 0
---
Scheduling destruction of call '8d63a2838d7a5c51@192.168.1.143' in 15000 ms
asterisk1*CLI>
<-- SIP read from 192.168.1.143:5060:
INVITE sip:14037422000@hmsinet.aleph-com.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.143:5060;branch=z9hG4bK81369072f3780e27
From: "Darren Wiebe" <sip:13@hmsinet.aleph-com.net>;tag=c821980168b25116
To: <sip:14037422000@hmsinet.aleph-com.net>
Contact: <sip:13@192.168.1.143:5060>
Supported: replaces, timer
Call-ID: f780d0e67535a7bc@192.168.1.143
CSeq: 55305 INVITE
User-Agent: Grandstream GXP2000 1.1.0.13
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 308
v=0
o=13 8000 8000 IN IP4 192.168.1.143
s=SIP Call
c=IN IP4 192.168.1.143
t=0 0
m=audio 5004 RTP/AVP 0 8 3 18 4 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
--- (13 headers 15 lines)---
Using INVITE request as basis request - f780d0e67535a7bc@192.168.1.143
Sending to 192.168.1.143 : 5060 (non-NAT)
Reliably Transmitting (NAT) to 192.168.1.143:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.143:5060;branch=z9hG4bK81369072f3780e27;received=192.168.1.143
From: "Darren Wiebe" <sip:13@hmsinet.aleph-com.net>;tag=c821980168b25116
To: <sip:14037422000@hmsinet.aleph-com.net>;tag=as0ef478c9
Call-ID: f780d0e67535a7bc@192.168.1.143
CSeq: 55305 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:14037422000@66.112.182.171>
Proxy-Authenticate: Digest realm="asterisk", nonce="5d0052f7"
Content-Length: 0
---
Scheduling destruction of call 'f780d0e67535a7bc@192.168.1.143' in 15000 ms
Found user '13'
asterisk1*CLI>
<-- SIP read from 192.168.1.143:5060:
INVITE sip:14037422000@hmsinet.aleph-com.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.143:5060;branch=z9hG4bK81369072f3780e27
From: "Darren Wiebe" <sip:13@hmsinet.aleph-com.net>;tag=c821980168b25116
To: <sip:14037422000@hmsinet.aleph-com.net>
Contact: <sip:13@192.168.1.143:5060>
Supported: replaces, timer
Call-ID: f780d0e67535a7bc@192.168.1.143
CSeq: 55305 INVITE
User-Agent: Grandstream GXP2000 1.1.0.13
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 308
v=0
o=13 8000 8001 IN IP4 192.168.1.143
s=SIP Call
c=IN IP4 192.168.1.143
t=0 0
m=audio 5004 RTP/AVP 0 8 3 18 4 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
--- (13 headers 15 lines)---
Ignoring this INVITE request
Retransmitting #1 (NAT) to 192.168.1.143:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.143:5060;branch=z9hG4bK81369072f3780e27;received=192.168.1.143
From: "Darren Wiebe" <sip:13@hmsinet.aleph-com.net>;tag=c821980168b25116
To: <sip:14037422000@hmsinet.aleph-com.net>;tag=as0ef478c9
Call-ID: f780d0e67535a7bc@192.168.1.143
CSeq: 55305 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:14037422000@66.112.182.171>
Proxy-Authenticate: Digest realm="asterisk", nonce="5d0052f7"
Content-Length: 0
---
Retransmitting #2 (NAT) to 192.168.1.143:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.143:5060;branch=z9hG4bK81369072f3780e27;received=192.168.1.143
From: "Darren Wiebe" <sip:13@hmsinet.aleph-com.net>;tag=c821980168b25116
To: <sip:14037422000@hmsinet.aleph-com.net>;tag=as0ef478c9
Call-ID: f780d0e67535a7bc@192.168.1.143
CSeq: 55305 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:14037422000@66.112.182.171>
Proxy-Authenticate: Digest realm="asterisk", nonce="5d0052f7"
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.1.143:5060:
INVITE sip:14037422000@hmsinet.aleph-com.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.143:5060;branch=z9hG4bK81369072f3780e27
From: "Darren Wiebe" <sip:13@hmsinet.aleph-com.net>;tag=c821980168b25116
To: <sip:14037422000@hmsinet.aleph-com.net>
Contact: <sip:13@192.168.1.143:5060>
Supported: replaces, timer
Call-ID: f780d0e67535a7bc@192.168.1.143
CSeq: 55305 INVITE
User-Agent: Grandstream GXP2000 1.1.0.13
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 308
v=0
o=13 8000 8002 IN IP4 192.168.1.143
s=SIP Call
c=IN IP4 192.168.1.143
t=0 0
m=audio 5004 RTP/AVP 0 8 3 18 4 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
--- (13 headers 15 lines)---
Ignoring this INVITE request
asterisk1*CLI>
<-- SIP read from 192.168.1.143:5060:
REGISTER sip:hmsinet.aleph-com.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.143:5060;branch=z9hG4bK64a528a22f9a31b5
From: "Darren Wiebe" <sip:13@hmsinet.aleph-com.net>;tag=b4b5e160041ac66b
To: <sip:13@hmsinet.aleph-com.net>
Contact: *
Call-ID: 8d63a2838d7a5c51@192.168.1.143
CSeq: 10001 REGISTER
Expires: 0
User-Agent: Grandstream GXP2000 1.1.0.13
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
--- (12 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.1.143 : 5060 (NAT)
Transmitting (NAT) to 192.168.1.143:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.143:5060;branch=z9hG4bK64a528a22f9a31b5;received=192.168.1.143
From: "Darren Wiebe" <sip:13@hmsinet.aleph-com.net>;tag=b4b5e160041ac66b
To: <sip:13@hmsinet.aleph-com.net>
Call-ID: 8d63a2838d7a5c51@192.168.1.143
CSeq: 10001 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:13@66.112.182.171>
Content-Length: 0
---
Transmitting (NAT) to 192.168.1.143:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.143:5060;branch=z9hG4bK64a528a22f9a31b5;received=192.168.1.143
From: "Darren Wiebe" <sip:13@hmsinet.aleph-com.net>;tag=b4b5e160041ac66b
To: <sip:13@hmsinet.aleph-com.net>;tag=as1f7008a7
Call-ID: 8d63a2838d7a5c51@192.168.1.143
CSeq: 10001 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:13@66.112.182.171>
WWW-Authenticate: Digest realm="asterisk", nonce="5a870435"
Content-Length: 0
---
Scheduling destruction of call '8d63a2838d7a5c51@192.168.1.143' in 15000 ms
Retransmitting #3 (NAT) to 192.168.1.143:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.143:5060;branch=z9hG4bK81369072f3780e27;received=192.168.1.143
From: "Darren Wiebe" <sip:13@hmsinet.aleph-com.net>;tag=c821980168b25116
To: <sip:14037422000@hmsinet.aleph-com.net>;tag=as0ef478c9
Call-ID: f780d0e67535a7bc@192.168.1.143
CSeq: 55305 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:14037422000@66.112.182.171>
Proxy-Authenticate: Digest realm="asterisk", nonce="5d0052f7"
Content-Length: 0
---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 66.112.182.11:5060:
OPTIONS sip:13@192.168.1.143:5060 SIP/2.0
Via: SIP/2.0/UDP 66.112.182.171:5060;branch=z9hG4bK46f3e21b;rport
From: "Unknown" <sip:Unknown@66.112.182.171>;tag=as08fe679e
To: <sip:13@192.168.1.143:5060>
Contact: <sip:Unknown@66.112.182.171>
Call-ID: 4e06bdf0331b9192359cca06020c7fcb@66.112.182.171
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 09 Jul 2006 20:21:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Retransmitting #1 (no NAT) to 66.112.182.11:5060:
OPTIONS sip:13@192.168.1.143:5060 SIP/2.0
Via: SIP/2.0/UDP 66.112.182.171:5060;branch=z9hG4bK46f3e21b;rport
From: "Unknown" <sip:Unknown@66.112.182.171>;tag=as08fe679e
To: <sip:13@192.168.1.143:5060>
Contact: <sip:Unknown@66.112.182.171>
Call-ID: 4e06bdf0331b9192359cca06020c7fcb@66.112.182.171
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 09 Jul 2006 20:21:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.1.143:5060:
REGISTER sip:hmsinet.aleph-com.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.143:5060;branch=z9hG4bK64a528a22f9a31b5
From: "Darren Wiebe" <sip:13@hmsinet.aleph-com.net>;tag=b4b5e160041ac66b
To: <sip:13@hmsinet.aleph-com.net>
Contact: *
Call-ID: 8d63a2838d7a5c51@192.168.1.143
CSeq: 10001 REGISTER
Expires: 0
User-Agent: Grandstream GXP2000 1.1.0.13
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
--- (12 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.1.143 : 5060 (NAT)
Transmitting (NAT) to 192.168.1.143:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.143:5060;branch=z9hG4bK64a528a22f9a31b5;received=192.168.1.143
From: "Darren Wiebe" <sip:13@hmsinet.aleph-com.net>;tag=b4b5e160041ac66b
To: <sip:13@hmsinet.aleph-com.net>
Call-ID: 8d63a2838d7a5c51@192.168.1.143
CSeq: 10001 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:13@66.112.182.171>
Content-Length: 0
---
Transmitting (NAT) to 192.168.1.143:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.143:5060;branch=z9hG4bK64a528a22f9a31b5;received=192.168.1.143
From: "Darren Wiebe" <sip:13@hmsinet.aleph-com.net>;tag=b4b5e160041ac66b
To: <sip:13@hmsinet.aleph-com.net>;tag=as1f7008a7
Call-ID: 8d63a2838d7a5c51@192.168.1.143
CSeq: 10001 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:13@66.112.182.171>
WWW-Authenticate: Digest realm="asterisk", nonce="5a870435"
Content-Length: 0
---
Scheduling destruction of call '8d63a2838d7a5c51@192.168.1.143' in 15000 ms
Retransmitting #4 (NAT) to 192.168.1.143:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.143:5060;branch=z9hG4bK81369072f3780e27;received=192.168.1.143
From: "Darren Wiebe" <sip:13@hmsinet.aleph-com.net>;tag=c821980168b25116
To: <sip:14037422000@hmsinet.aleph-com.net>;tag=as0ef478c9
Call-ID: f780d0e67535a7bc@192.168.1.143
CSeq: 55305 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:14037422000@66.112.182.171>
Proxy-Authenticate