<--- SIP read from 192.168.64.62:5060 --->
INVITE sip:306@192.168.64.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.64.62:5060;branch=z9hG4bK-9db9fbce
From: AAA <sip:305@192.168.64.3>;tag=bbce5ba63a574c7co0
To: <sip:306@192.168.64.3>
Call-ID: 32c1a26e-419b7454@192.168.64.62
CSeq: 101 INVITE
Max-Forwards: 70
Contact: AAA <sip:305@192.168.64.62:5060>
Expires: 240
User-Agent: Sipura/SPA2000-2.0.10(e)
Content-Length: 422
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 12651 12651 IN IP4 192.168.64.62
s=-
c=IN IP4 192.168.64.62
t=0 0
m=audio 16456 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (14 headers 19 lines) ---
Sending to 192.168.64.62 : 5060 (no NAT)
Using INVITE request as basis request - 32c1a26e-419b7454@192.168.64.62
Found peer '306'

<--- Reliably Transmitting (no NAT) to 192.168.64.62:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.64.62:5060;branch=z9hG4bK-9db9fbce;received=192.168.64.62
From: AAA <sip:305@192.168.64.3>;tag=bbce5ba63a574c7co0
To: <sip:306@192.168.64.3>;tag=as7b9560ab
Call-ID: 32c1a26e-419b7454@192.168.64.62
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5ad97d05"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '32c1a26e-419b7454@192.168.64.62' in 32000 ms (Method: INVITE)
localhost*CLI> 
<--- SIP read from 192.168.64.62:5060 --->
ACK sip:306@192.168.64.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.64.62:5060;branch=z9hG4bK-9db9fbce
From: AAA <sip:305@192.168.64.3>;tag=bbce5ba63a574c7co0
To: <sip:306@192.168.64.3>;tag=as7b9560ab
Call-ID: 32c1a26e-419b7454@192.168.64.62
CSeq: 101 ACK
Max-Forwards: 70
Contact: AAA <sip:305@192.168.64.62:5060>
User-Agent: Sipura/SPA2000-2.0.10(e)
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
localhost*CLI> 
<--- SIP read from 192.168.64.62:5060 --->
INVITE sip:306@192.168.64.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.64.62:5060;branch=z9hG4bK-a7ca23e2
From: AAA <sip:305@192.168.64.3>;tag=bbce5ba63a574c7co0
To: <sip:306@192.168.64.3>
Call-ID: 32c1a26e-419b7454@192.168.64.62
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="305",realm="asterisk",nonce="5ad97d05",uri="sip:306@192.168.64.3",algorithm=MD5,response="b8fb556e696c496b5a5e7ba0547ad180"
Contact: AAA <sip:305@192.168.64.62:5060>
Expires: 240
User-Agent: Sipura/SPA2000-2.0.10(e)
Content-Length: 422
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 12651 12651 IN IP4 192.168.64.62
s=-
c=IN IP4 192.168.64.62
t=0 0
m=audio 16456 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (15 headers 19 lines) ---
Sending to 192.168.64.62 : 5060 (no NAT)
Using INVITE request as basis request - 32c1a26e-419b7454@192.168.64.62
Found peer '306'
[Feb 12 19:35:49] WARNING[1103]: chan_sip.c:8023 check_auth: username mismatch, have <306>, digest has <305>
[Feb 12 19:35:49] NOTICE[1103]: chan_sip.c:13200 handle_request_invite: Failed to authenticate user AAA <sip:305@192.168.64.3>;tag=bbce5ba63a574c7co0

<--- Reliably Transmitting (no NAT) to 192.168.64.62:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.64.62:5060;branch=z9hG4bK-a7ca23e2;received=192.168.64.62
From: AAA <sip:305@192.168.64.3>;tag=bbce5ba63a574c7co0
To: <sip:306@192.168.64.3>;tag=as7b9560ab
Call-ID: 32c1a26e-419b7454@192.168.64.62
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '32c1a26e-419b7454@192.168.64.62' in 32000 ms (Method: INVITE)
localhost*CLI> 
<--- SIP read from 192.168.64.62:5060 --->
ACK sip:306@192.168.64.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.64.62:5060;branch=z9hG4bK-a7ca23e2
From: AAA <sip:305@192.168.64.3>;tag=bbce5ba63a574c7co0
To: <sip:306@192.168.64.3>;tag=as7b9560ab
Call-ID: 32c1a26e-419b7454@192.168.64.62
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="305",realm="asterisk",nonce="5ad97d05",uri="sip:306@192.168.64.3",algorithm=MD5,response="231cbd0be50c95030580eef86d08679a"
Contact: AAA <sip:305@192.168.64.62:5060>
User-Agent: Sipura/SPA2000-2.0.10(e)
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---