j'ai effectué un debogage. le trunk que j'établit avec le cirpack est sans authentification .
trixbox1*CLI> sip show registry
Host Username Refresh State Reg.Time
0 SIP registrations.
trixbox1*CLI> sip set debug peer *2cirpack
SIP Debugging Enabled for IP: 4.X.X.X:5060
trixbox1*CLI> sip show objects
-= User objects: 2 static, 0 realtime =-

name: cirpack2*
objflags: 0
refcount: 1

name: 100
objflags: 0
refcount: 1

-= Peer objects: 5 static, 0 realtime, 0 autocreate =-

name: 100
objflags: 0
refcount: 181

name: *2cirpack
objflags: 0
refcount: 1

-= Registry objects: 0 =-

trixbox1*CLI>
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/100-000000ef", "SIP/*2cirpack/21312407,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
Audio is at 4.Y.Y.Y port 15590
Video is at 4.Y.Y.Y port 18950
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding video codec 0x80000 (h263) to SDP
Adding video codec 0x200000 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 4.X.X.X:5060:
INVITE sip:21312407@4.X.X.X.X SIP/2.0
Via: SIP/2.0/UDP 4.Y.Y.Y:5060;branch=z9hG4bK35721b93;rport
Max-Forwards: 70
From: "21601100" <sip:21601100@4.Y.Y.Y>;tag=as07ea5cf5
To: <sip:21312407@4.X.X.X>
Contact: <sip:21601100@4.Y.Y.Y>
Call-ID: 529cc90040baf9003b44b42a1a396405@4.Y.Y.Y
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Date: Mon, 15 Oct 2012 09:58:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 401

v=0
o=root 1210728476 1210728476 IN IP4 4.Y.Y.Y
s=Asterisk PBX 1.6.0.26-FONCORE-r78
c=IN IP4 4.Y.Y.Y
b=CT:384
t=0 0
m=audio 15590 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 18950 RTP/AVP 34 99
a=rtpmap:34 H263/90000
a=rtpmap:99 H264/90000
a=sendrecv

---
-- Called *2cirpack/21312407

<--- SIP read from UDP://4.X.X.X:5060 --->
SIP/2.0 100 Trying
Allow: UPDATE,REFER,INFO
Call-ID: 529cc90040baf9003b44b42a1a396405@4.Y.Y.Y
Contact: <sip:4.X.X.X:5060>
CSeq: 102 INVITE
From: "21601100" <sip:21601100@4.Y.Y.Y>;tag=as07ea5cf5
Server: Cirpack/v4.41f (gw_sip)
To: <sip:21312407@4.X.X.X>
Via: SIP/2.0/UDP 4.Y.Y.Y:5060;received=4.Y.Y.Y;rport=5060;branch=z9 hG4bK35721b93
Content-Length: 0


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

<--- SIP read from UDP://4.X.X.X:5060 --->
SIP/2.0 484 not pnum compliant
Allow: UPDATE,REFER,INFO
Call-ID: 529cc90040baf9003b44b42a1a396405@4.Y.Y.Y
Contact: <sip:4.X.X.X:5060>
CSeq: 102 INVITE
From: "21601100" <sip:21601100@4.Y.Y.Y>;tag=as07ea5cf5
Reason: q.850;cause=21
Server: Cirpack/v4.41f (gw_sip)
To: <sip:21312407@4.X.X.X>;tag=02-08162-012bbae6-278107230
Via: SIP/2.0/UDP 4.Y.Y.Y:5060;received=4.Y.Y.Y;rport=5060;branch=z9 hG4bK35721b93
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
-- Got SIP response 484 "not pnum compliant" back from 4.X.X.X
Transmitting (no NAT) to 4.X.X.X:5060:
ACK sip:21312407@4.X.X.X SIP/2.0
Via: SIP/2.0/UDP 4.Y.Y.Y:5060;branch=z9hG4bK35721b93;rport
Max-Forwards: 70
From: "21601100" <sip:21601100@4.Y.Y.Y>;tag=as07ea5cf5
To: <sip:21312407@4.X.X.X>;tag=02-08162-012bbae6-278107230
Contact: <sip:21601100@4.Y.Y.Y>
Call-ID: 529cc90040baf9003b44b42a1a396405@4.Y.Y.Y
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Content-Length: 0


---
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:20] Goto("SIP/100-000000ef", "s-CHANUNAVAIL,1") in new stack