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<TITLE>Re: [asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI,not PRI to SIP Authentication Issue</TITLE>
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<P><FONT SIZE=2>When all else fails I resort to adding this in the sip.conf peer config:<BR>
<BR>
Insecure=invite,port<BR>
<BR>
It took me a while to figure out they can be used together.<BR>
<BR>
Regards,<BR>
Scott<BR>
<BR>
----- Original Message -----<BR>
From: asterisk-users-bounces@lists.digium.com <asterisk-users-bounces@lists.digium.com><BR>
To: asterisk-users@lists.digium.com <asterisk-users@lists.digium.com><BR>
Sent: Tue Nov 07 15:23:26 2006<BR>
Subject: [asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI,not PRI to SIP Authentication Issue<BR>
<BR>
Hi All,<BR>
<BR>
I have a lab setup with two asterisk servers and a MAX TNT in the<BR>
middle like this:<BR>
<BR>
asterisk sip >< sip TNT pri >< pri asterisk<BR>
<BR>
The TNT is running 11.0.6 and the asterisk servers are running<BR>
1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to<BR>
asterisk but not the other way. The call from asterisk to pri to tnt<BR>
is good, the TNT is passing SIP invite to the SIP Asterisk server. I<BR>
have tried many variations of using sip options insecure,<BR>
autocreatepeer, permit/deny, host, user, etc.... but can't seem to get<BR>
asterisk to accept an unauthenticated call from the TNT using SIP. I<BR>
keep getting SIP/2.0 407 Proxy Authentication Required. I know others<BR>
have done this, but with older Asterisk versions, I'm wondering what<BR>
versions of Asterisk are known to work with the MAX TNT and with what<BR>
version of the TNT?<BR>
<BR>
I'm confident this is an asterisk issue, with insecure=very, I should<BR>
be able to pass calls to asterisk without trying to authenticate it<BR>
first. But this is not happening.<BR>
<BR>
Here is a debug of a call and a snip from my sip.conf:<BR>
<BR>
sip.conf<BR>
<BR>
[maxtnt]<BR>
type=friend<BR>
host=10.10.14.131<BR>
insecure=very<BR>
dtmfmode=inband<BR>
callerid="MaxTNT" <maxtnt><BR>
context=trunktntin<BR>
qualify=yes<BR>
reinvite=no<BR>
canreinvite=no<BR>
disallow=all<BR>
allow=ulaw<BR>
<BR>
debug<BR>
<BR>
lab1*CLI><BR>
<-- SIP read from 10.10.14.131:5060:<BR>
INVITE sip:2145551212@10.10.14.121:5060;user=phone SIP/2.0<BR>
t: <sip:2145551212@10.10.14.121:5060;user=phone><BR>
f: "NO CID NAME"<BR>
<sip:1239@10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a<BR>
Remote-Party-Id: "NO CID NAME"<BR>
<sip:1239@10.10.14.131:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off<BR>
i: 3a8884d9-64-1fb1f65c@10.10.14.131<BR>
CSeq: 639089 INVITE<BR>
v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d<BR>
Max-Forwards: 70<BR>
m: <sip:1239@10.10.14.131:5060;user=phone><BR>
k: replaces<BR>
c: application/sdp<BR>
Accept: application/sdp<BR>
Accept-Encoding:<BR>
Accept-Language: en<BR>
User-Agent: Lucent-Universal-Gateway<BR>
l: 232<BR>
<BR>
v=0<BR>
o=t1gw01 531756636 531756636 IN IP4 10.10.14.131<BR>
s=Session SDP<BR>
c=IN IP4 10.10.14.131<BR>
t=0 0<BR>
m=audio 40198 RTP/AVP 0 96<BR>
a=silenceSupp:on<BR>
a=ecan:b on g168<BR>
a=ptime:20<BR>
a=rtpmap:96 telephone-event/8000<BR>
a=rtpmap:0 PCMU/8000<BR>
<BR>
--- (16 headers 11 lines) ---<BR>
Using INVITE request as basis request - 3a8884d9-64-1fb1f65c@10.10.14.131<BR>
Sending to 10.10.14.131 : 5060 (non-NAT)<BR>
Reliably Transmitting (no NAT) to 10.10.14.131:5060:<BR>
SIP/2.0 407 Proxy Authentication Required<BR>
Via: SIP/2.0/UDP<BR>
10.10.14.131:5060;branch=z9hG4bK006187a112a9899d;received=10.10.14.131<BR>
From: "NO CID NAME"<BR>
<sip:1239@10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a<BR>
To: <sip:2145551212@10.10.14.121:5060;user=phone>;tag=as41f8454e<BR>
Call-ID: 3a8884d9-64-1fb1f65c@10.10.14.131<BR>
CSeq: 639089 INVITE<BR>
User-Agent: Asterisk PBX<BR>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<BR>
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ea7e98a"<BR>
Content-Length: 0<BR>
<BR>
<BR>
---<BR>
Scheduling destruction of call '3a8884d9-64-1fb1f65c@10.10.14.131' in 15000 ms<BR>
Found user '1239'<BR>
lab1*CLI><BR>
<-- SIP read from 10.10.14.131:5060:<BR>
ACK sip:2145551212@10.10.14.121:5060;user=phone SIP/2.0<BR>
t: <sip:2145551212@10.10.14.121:5060;user=phone>;tag=as41f8454e<BR>
f: "NO CID NAME"<BR>
<sip:1239@10.10.14.131:5060;user=phone>;tag=5fe9f589-1fb1f65c-830e0a0a<BR>
i: 3a8884d9-64-1fb1f65c@10.10.14.131<BR>
CSeq: 639089 ACK<BR>
v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d<BR>
Max-Forwards: 70<BR>
User-Agent: Lucent-Universal-Gateway<BR>
l: 0<BR>
<BR>
<BR>
Any guidance will be much appreciated.<BR>
<BR>
Thanks.<BR>
<BR>
JR<BR>
<BR>
--<BR>
JR Richardson<BR>
Engineering for the Masses<BR>
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