[asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue

[JR Richardson]

Hi All,

I have a lab setup with two asterisk servers and a MAX TNT in the
middle like this:

asterisk sip >< sip TNT pri >< pri asterisk

The TNT is running 11.0.6 and the asterisk servers are running
1.2.9.1.  I can get calls to pass from asterisk sip to tnt to pri to
asterisk but not the other way. The call from asterisk to pri to tnt
is good, the TNT is passing SIP invite to the SIP Asterisk server. I
have tried many variations of using sip options insecure,
autocreatepeer, permit/deny, host, user, etc.... but can't seem to get
asterisk to accept an unauthenticated call from the TNT using SIP.  I
keep getting SIP/2.0 407 Proxy Authentication Required.  I know others
have done this, but with older Asterisk versions, I'm wondering what
versions of Asterisk are known to work with the MAX TNT and with what
version of the TNT?

I'm confident this is an asterisk issue, with insecure=very, I should
be able to pass calls to asterisk without trying to authenticate it
first.  But this is not happening.

Here is a debug of a call and a snip from my sip.conf:

sip.conf

[maxtnt]
type=friend
host=10.10.14.131
insecure=very
dtmfmode=inband
callerid="MaxTNT" <maxtnt>
context=trunktntin
qualify=yes
reinvite=no
canreinvite=no
disallow=all
allow=ulaw

debug

lab1*CLI>
<-- SIP read from 10.10.14.131:5060:
INVITE sip:2145551212 at 10.10.14.121:5060;user=phone SIP/2.0
t:   <sip:2145551212 at 10.10.14.121:5060;user=phone>
f: "NO CID NAME"
<sip:1239 at 10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a
Remote-Party-Id: "NO CID NAME"
<sip:1239 at 10.10.14.131:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off
i: 3a8884d9-64-1fb1f65c at 10.10.14.131
CSeq: 639089 INVITE
v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d
Max-Forwards: 70
m: <sip:1239 at 10.10.14.131:5060;user=phone>
k: replaces
c: application/sdp
Accept: application/sdp
Accept-Encoding:
Accept-Language: en
User-Agent: Lucent-Universal-Gateway
l: 232

v=0
o=t1gw01 531756636 531756636 IN IP4 10.10.14.131
s=Session SDP
c=IN IP4 10.10.14.131
t=0 0
m=audio 40198 RTP/AVP 0 96
a=silenceSupp:on
a=ecan:b on g168
a=ptime:20
a=rtpmap:96 telephone-event/8000
a=rtpmap:0 PCMU/8000

--- (16 headers 11 lines) ---
Using INVITE request as basis request - 3a8884d9-64-1fb1f65c at 10.10.14.131
Sending to 10.10.14.131 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 10.10.14.131:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
10.10.14.131:5060;branch=z9hG4bK006187a112a9899d;received=10.10.14.131
From: "NO CID NAME"
<sip:1239 at 10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a
To: <sip:2145551212 at 10.10.14.121:5060;user=phone>;tag=as41f8454e
Call-ID: 3a8884d9-64-1fb1f65c at 10.10.14.131
CSeq: 639089 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ea7e98a"
Content-Length: 0


---
Scheduling destruction of call '3a8884d9-64-1fb1f65c at 10.10.14.131' in 15000 ms
Found user '1239'
lab1*CLI>
<-- SIP read from 10.10.14.131:5060:
ACK sip:2145551212 at 10.10.14.121:5060;user=phone SIP/2.0
t:   <sip:2145551212 at 10.10.14.121:5060;user=phone>;tag=as41f8454e
f: "NO CID NAME"
<sip:1239 at 10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a
i: 3a8884d9-64-1fb1f65c at 10.10.14.131
CSeq: 639089 ACK
v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d
Max-Forwards: 70
User-Agent: Lucent-Universal-Gateway
l: 0


Any guidance will be much appreciated.

Thanks.

JR

-- 
JR Richardson
Engineering for the Masses