[asterisk-users] Bypassing a Auth on Invite or Forbiden?

[Bryan Cramer]

Hello,

I have 2 asterisk servers that are not working well together.  One is  
acting like a registrar (PBX01) for all my PAP2's and other SIP/IAX  
devices.  And the other is acting like my sip gateway (PBX02) to  
various providers.  They are both on a private network and should be  
trusting each others IP 100%.  But the PBX02 challenges PBX01's  
requests all the time even though insecure=invite is set.   Here are  
the stanzas for each server:

PBX01

[pbx01topbx02]
type=friend
context=incomingDefault
host=10.10.10.2
qualify=600
disallow=all
allow=ulaw
canreinvite=no
insecure=invite
accountcode=pbx02
;TESTS;
;auth=pbx01topbx02 at asterisk
;fromuser=pbx01topbx02
;username=pbx01topbx02
;secret=j48dj7rjd9023jd
sendrpid=yes
trustrpid=yes


PBX02

[pbx01topbx02]
type=friend
host=10.10.10.3
qualify=600
context=dialOutPatternsAll
disallow=all
allow=ulaw
canreinvite=no
insecure=invite
accountcode=pbx01
;TESTS;
;auth=pbx01topbx02 at asterisk
;username=pbx01topbx02
;fromuser=pbx01topbx02
;secret=j48dj7rjd9023jd
sendrpid=yes
trustrpid=yes

Layout

SIPDEV -REGISTERD-> PBX01 -> PBX02 -> PROVIDER

With the above settings above I should be allowed to send calls as I  
like through the 2 boxes, but I can't I get the following messages:

PBX01
     -- Executing [dialIt at macro-dialOut:7] Dial("IAX2/4161231234-2",  
"SIP/pbx01topbx02/16041231234") in new stack
     -- Called pbx01topbx02/16041231234
[Feb  1 11:35:18] NOTICE[13507]: chan_sip.c:11983  
handle_response_invite: Failed to authenticate on INVITE to  
'"4161231234" <sip:4161231234 at 10.10.10.3>;tag=as4a9e0ae9'
     -- SIP/pbx01topbx02-009c31d0 is circuit-busy
   == Everyone is busy/congested at this time (1:0/1/0)

Then on PBX02 I don't get any debug/verbose messages unless I do a  
sip debug then I get:

<--- SIP read from 10.10.10.3:5060 --->
INVITE sip:16041231234 at 10.10.10.2 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK349431bd;rport
From: "4161231234" <sip:4161231234 at 10.10.10.3>;tag=as5c5aac02
To: <sip:16041231234 at 10.10.10.2>
Contact: <sip:4161231234 at 10.10.10.3>
Call-ID: 640b3e0405d977e779db174d78f57d6f at 10.10.10.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "4161231234" <sip: 
4161231234 at 10.10.10.3>;privacy=off;screen=yes
Date: Fri, 01 Feb 2008 19:40:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 13479 13479 IN IP4 10.10.10.3
s=session
c=IN IP4 10.10.10.3
t=0 0
m=audio 11026 RTP/AVP 0 101
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 12 lines) ---
Sending to 10.10.10.3 : 5060 (NAT)
Using INVITE request as basis request -  
640b3e0405d977e779db174d78f57d6f at 10.10.10.3

<--- Reliably Transmitting (no NAT) to 10.10.10.3:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP  
10.10.10.3:5060;branch=z9hG4bK349431bd;received=10.10.10.3;rport=5060
From: "4161231234" <sip:4161231234 at 10.10.10.3>;tag=as5c5aac02
To: <sip:16041231234 at 10.10.10.2>;tag=as19dfe789
Call-ID: 640b3e0405d977e779db174d78f57d6f at 10.10.10.3
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="6d7284c8"
Content-Length: 0

Any insights as to why insecure invite is not taking effect?

Bryan
_________________________________________________________________
Bryan Cramer

PO Box 616
Sechelt BC, V0N 3A0
Web: www.yellowions.com
Tel: 604-773-4580

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