INVITE sip:s@192.168.10.126 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.175:5060;branch=z9hG4bK10d2ed72;rport
From: "Asterisk User 6000" <sip:jaredsmith@192.168.10.175>;tag=as321b8852
To: <sip:s@192.168.10.126>
Contact: <sip:jaredsmith@192.168.10.175>
Call-ID: 53a61faf626da7b502ed076829be7bf1@192.168.10.175
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 11 Dec 2006 15:25:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 17940 17940 IN IP4 192.168.10.175
s=session
c=IN IP4 192.168.10.175
t=0 0
m=audio 14518 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (14 headers 13 lines) ---
Sending to 192.168.10.175 : 5060 (NAT)
Using INVITE request as basis request - 53a61faf626da7b502ed076829be7bf1@192.168.10.175

<--- Reliably Transmitting (no NAT) to 192.168.10.175:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.10.175:5060;branch=z9hG4bK10d2ed72;received=192.168.10.175;rport=5060
From: "Asterisk User 6000" <sip:jaredsmith@192.168.10.175>;tag=as321b8852
To: <sip:s@192.168.10.126>;tag=as505a833b
Call-ID: 53a61faf626da7b502ed076829be7bf1@192.168.10.175
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2c6d8877"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '53a61faf626da7b502ed076829be7bf1@192.168.10.175' in 32000 ms (Method: INVITE)
Found user 'jaredsmith'
localhost*CLI> 
<--- SIP read from 192.168.10.175:5060 --->
ACK sip:s@192.168.10.126 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.175:5060;branch=z9hG4bK10d2ed72;rport
From: "Asterisk User 6000" <sip:jaredsmith@192.168.10.175>;tag=as321b8852
To: <sip:s@192.168.10.126>;tag=as505a833b
Contact: <sip:jaredsmith@192.168.10.175>
Call-ID: 53a61faf626da7b502ed076829be7bf1@192.168.10.175
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
localhost*CLI> 
<--- SIP read from 192.168.10.175:5060 --->
INVITE sip:s@192.168.10.126 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.175:5060;branch=z9hG4bK2cbe5d4d;rport
From: "Asterisk User 6000" <sip:jaredsmith@192.168.10.175>;tag=as321b8852
To: <sip:s@192.168.10.126>
Contact: <sip:jaredsmith@192.168.10.175>
Call-ID: 53a61faf626da7b502ed076829be7bf1@192.168.10.175
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="s", realm="asterisk", algorithm=MD5, uri="sip:s@192.168.10.126", nonce="2c6d8877", response="da9ffe708f6889d4ea8eebfc0ba6b6a4", opaque=""
Date: Mon, 11 Dec 2006 15:25:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 17940 17941 IN IP4 192.168.10.175
s=session
c=IN IP4 192.168.10.175
t=0 0
m=audio 14518 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (15 headers 13 lines) ---
Sending to 192.168.10.175 : 5060 (NAT)
Using INVITE request as basis request - 53a61faf626da7b502ed076829be7bf1@192.168.10.175
[Dec 11 10:25:53] WARNING[15474]: chan_sip.c:8021 check_auth: username mismatch, have <jaredsmith>, digest has <s>
Found user 'jaredsmith'
[Dec 11 10:25:53] NOTICE[15474]: chan_sip.c:13196 handle_request_invite: Failed to authenticate user "Asterisk User 6000" <sip:jaredsmith@192.168.10.175>;tag=as321b8852

<--- Reliably Transmitting (no NAT) to 192.168.10.175:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.10.175:5060;branch=z9hG4bK2cbe5d4d;received=192.168.10.175;rport=5060
From: "Asterisk User 6000" <sip:jaredsmith@192.168.10.175>;tag=as321b8852
To: <sip:s@192.168.10.126>;tag=as505a833b
Call-ID: 53a61faf626da7b502ed076829be7bf1@192.168.10.175
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '53a61faf626da7b502ed076829be7bf1@192.168.10.175' in 32000 ms (Method: INVITE)
localhost*CLI> 
<--- SIP read from 192.168.10.175:5060 --->
ACK sip:s@192.168.10.126 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.175:5060;branch=z9hG4bK2cbe5d4d;rport
From: "Asterisk User 6000" <sip:jaredsmith@192.168.10.175>;tag=as321b8852
To: <sip:s@192.168.10.126>;tag=as505a833b
Contact: <sip:jaredsmith@192.168.10.175>
Call-ID: 53a61faf626da7b502ed076829be7bf1@192.168.10.175
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0