Code:
raspbx*CLI> sip set debug on
SIP Debugging re-enabled
<--- SIP read from UDP:192.168.1.2:57693 --->
INVITE sip:32499411198@192.168.1.250 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:57693;rport;branch=z9hG4bKPjU7rEcXJHWT-WqCDZscnxC8MYKHUZu2Uq
Max-Forwards: 70
From: "Arnaud CRESP" <sip:10@192.168.1.250>;tag=APF0hcfO8eJaYihQBbOHwG8oCxoRuLyY
To: <sip:32499411198@192.168.1.250>
Contact: "Arnaud CRESP" <sip:10@192.168.1.2:57693;ob>
Call-ID: YyTmjEUeIiijiDpVYirRp2CqrNgeJuOa
CSeq: 5716 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Telephone 1.0.4
Content-Type: application/sdp
Content-Length: 456
v=0
o=- 3589213431 3589213431 IN IP4 192.168.1.2
s=pjmedia
c=IN IP4 192.168.1.2
t=0 0
a=X-nat:0
m=audio 4008 RTP/AVP 103 102 104 109 3 0 8 9 101
a=rtcp:4009 IN IP4 192.168.1.2
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (15 headers 20 lines) ---
Sending to 192.168.1.2:57693 (NAT)
Sending to 192.168.1.2:57693 (NAT)
Using INVITE request as basis request - YyTmjEUeIiijiDpVYirRp2CqrNgeJuOa
Found peer '10' for '10' from 192.168.1.2:57693
<--- Reliably Transmitting (no NAT) to 192.168.1.2:57693 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:57693;branch=z9hG4bKPjU7rEcXJHWT-WqCDZscnxC8MYKHUZu2Uq;received=192.168.1.2;rport=57693
From: "Arnaud CRESP" <sip:10@192.168.1.250>;tag=APF0hcfO8eJaYihQBbOHwG8oCxoRuLyY
To: <sip:32499411198@192.168.1.250>;tag=as70387093
Call-ID: YyTmjEUeIiijiDpVYirRp2CqrNgeJuOa
CSeq: 5716 INVITE
Server: FPBX-2.11.0(11.5.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3593fdaf"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'YyTmjEUeIiijiDpVYirRp2CqrNgeJuOa' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:192.168.1.2:57693 --->
ACK sip:32499411198@192.168.1.250 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:57693;rport;branch=z9hG4bKPjU7rEcXJHWT-WqCDZscnxC8MYKHUZu2Uq
Max-Forwards: 70
From: "Arnaud CRESP" <sip:10@192.168.1.250>;tag=APF0hcfO8eJaYihQBbOHwG8oCxoRuLyY
To: <sip:32499411198@192.168.1.250>;tag=as70387093
Call-ID: YyTmjEUeIiijiDpVYirRp2CqrNgeJuOa
CSeq: 5716 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:192.168.1.2:57693 --->
INVITE sip:32499411198@192.168.1.250 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:57693;rport;branch=z9hG4bKPjn9UF-mxGYJ1do9tU.hoTM7UCw7n3gQ.2
Max-Forwards: 70
From: "Arnaud CRESP" <sip:10@192.168.1.250>;tag=APF0hcfO8eJaYihQBbOHwG8oCxoRuLyY
To: <sip:32499411198@192.168.1.250>
Contact: "Arnaud CRESP" <sip:10@192.168.1.2:57693;ob>
Call-ID: YyTmjEUeIiijiDpVYirRp2CqrNgeJuOa
CSeq: 5717 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Telephone 1.0.4
Authorization: Digest username="10", realm="asterisk", nonce="3593fdaf", uri="sip:32499411198@192.168.1.250", response="acb03f0fd2589ba0719ed8a80c4afe5b", algorithm=MD5
Content-Type: application/sdp
Content-Length: 456
v=0
o=- 3589213431 3589213431 IN IP4 192.168.1.2
s=pjmedia
c=IN IP4 192.168.1.2
t=0 0
a=X-nat:0
m=audio 4008 RTP/AVP 103 102 104 109 3 0 8 9 101
a=rtcp:4009 IN IP4 192.168.1.2
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (16 headers 20 lines) ---
Sending to 192.168.1.2:57693 (no NAT)
Using INVITE request as basis request - YyTmjEUeIiijiDpVYirRp2CqrNgeJuOa
Found peer '10' for '10' from 192.168.1.2:57693
Found RTP audio format 103
Found RTP audio format 102
Found RTP audio format 104
Found RTP audio format 109
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 101
Found audio description format speex for ID 103
Found audio description format speex for ID 102
Found audio description format speex for ID 104
Found audio description format iLBC for ID 109
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=(gsm|ulaw|alaw|speex|speex16|ilbc|g722|speex32)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.2:4008
Looking for 32499411198 in from-internal (domain 192.168.1.250)
list_route: hop: <sip:10@192.168.1.2:57693;ob>
<--- Transmitting (no NAT) to 192.168.1.2:57693 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:57693;branch=z9hG4bKPjn9UF-mxGYJ1do9tU.hoTM7UCw7n3gQ.2;received=192.168.1.2;rport=57693
From: "Arnaud CRESP" <sip:10@192.168.1.250>;tag=APF0hcfO8eJaYihQBbOHwG8oCxoRuLyY
To: <sip:32499411198@192.168.1.250>
Call-ID: YyTmjEUeIiijiDpVYirRp2CqrNgeJuOa
CSeq: 5717 INVITE
Server: FPBX-2.11.0(11.5.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:32499411198@192.168.1.250:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:192.168.1.2:57693 --->
<------------->
Reliably Transmitting (no NAT) to 192.168.1.2:57693:
OPTIONS sip:10@192.168.1.2:57693;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK2b289f5e
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.250>;tag=as7c391998
To: <sip:10@192.168.1.2:57693;ob>
Contact: <sip:Unknown@192.168.1.250:5060>
Call-ID: 3267783720f8341c67e0382c0dc75c25@192.168.1.250:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.5.1)
Date: Thu, 26 Sep 2013 19:43:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:192.168.1.2:57693 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.250:5060;received=192.168.1.250;branch=z9hG4bK2b289f5e
Call-ID: 3267783720f8341c67e0382c0dc75c25@192.168.1.250:5060
From: "Unknown" <sip:Unknown@192.168.1.250>;tag=as7c391998
To: <sip:10@192.168.1.2;ob>;tag=z9hG4bK2b289f5e
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: Telephone 1.0.4
Content-Type: application/sdp
Content-Length: 445
v=0
o=- 3589213432 3589213432 IN IP4 192.168.1.2
s=pjmedia
c=IN IP4 192.168.1.2
t=0 0
m=audio 4000 RTP/AVP 103 102 104 109 3 0 8 9 101
a=rtcp:4001 IN IP4 192.168.1.2
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (13 headers 19 lines) ---
Merci pour votre aide :-)