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<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>I
tried that before, but I just tried it again. Unfortunately, the same
thing:<o:p></o:p></span></p>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'>No user '5551236049' in SIP users list<o:p></o:p></span></p>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'>Found peer 'ACME' for '5551236049' from 172.16.10.40:5060<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>[ACME]<br>
host=172.16.10.40<br>
username=username<br>
secret=password<br>
type=friend<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Frank<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Allan Dib<br>
<b>Sent:</b> Monday, January 05, 2009 9:41 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] Incoming side of SIP trunk does not work
unless I add "insecure=very"<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'>Try it by IP address instead of
hostname as reverse DNS may not be resolving. e.g. host=123.123.123.123<o:p></o:p></p>
<div>
<p class=MsoNormal>On Tue, Jan 6, 2009 at 2:25 PM, Frank Bulk <<a
href="mailto:frnkblk@iname.com">frnkblk@iname.com</a>> wrote:<o:p></o:p></p>
<p class=MsoNormal>This is what I have in my configuration now:<br>
<br>
[ACME]<br>
host=<a href="http://sip.acme.com" target="_blank">sip.acme.com</a><br>
username=username<br>
secret=password<br>
type=friend<br>
<br>
I've done a SIP debug before, but I've done it again with the above<br>
configuration:<br>
No user '5551236049' in SIP users list<br>
Found peer 'ACME' for '5551236049' from <a
href="http://172.16.10.40:5060" target="_blank">172.16.10.40:5060</a><br>
after which "SIP/2.0 401 Unauthorized" is issued after the
un-authenticated<br>
INVITE and "SIP/2.0 403 Forbidden" after the authenticated INVITE.<br>
<br>
When I add "insecure=very", this is what the SIP debug shows:<br>
No user '5551236049' in SIP users list<br>
Found peer 'ACME' for '5551236049' from <a
href="http://172.16.10.40:5060" target="_blank">172.16.10.40:5060</a><br>
Found RTP audio format 0<br>
Peer audio RTP is at port <a
href="http://172.16.10.65:36272" target="_blank">172.16.10.65:36272</a><br>
Found audio description format PCMU for ID 0<br>
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4<br>
(ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)<br>
Non-codec capabilities (dtmf): us - 0x1
(telephone-event), peer -<br>
0x0 (nothing), combined - 0x0 (nothing)<br>
Peer audio RTP is at port <a
href="http://172.16.10.65:36272" target="_blank">172.16.10.65:36272</a><br>
Looking for +15552127020 in from-sip-external
(domain <a href="http://sip.acme.com" target="_blank">sip.acme.com</a>)<br>
list_route: hop: <<a
href="mailto:sip%3A5551236049@172.16.10.40">sip:5551236049@172.16.10.40</a>><br>
<br>
It isn't very clear (to me) from the success how the "insecure=very"
helps.<br>
<span style='color:#888888'><br>
Frank</span><o:p></o:p></p>
<div>
<p class=MsoNormal><br>
-----Original Message-----<br>
From: Andres [mailto:<a href="mailto:andres@telesip.net">andres@telesip.net</a>]<br>
Sent: Monday, January 05, 2009 7:43 PM<br>
To: <a href="mailto:frnkblk@iname.com">frnkblk@iname.com</a>; Asterisk Users
Mailing List - Non-Commercial<br>
Discussion<o:p></o:p></p>
</div>
<div>
<div>
<p class=MsoNormal>Subject: Re: [asterisk-users] Incoming side of SIP trunk
does not work<br>
unless I add "insecure=very"<br>
<br>
Frank Bulk - iName.com wrote:<br>
<br>
>The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not<br>
>work unless I add "insecure=very" to my "Outgoing
settings", but I don't<br>
>want to do that. I do want to authenticate. Outgoing (Asterisk
PBX to<br>
>Class 5 switch) calls do authenticate and work.<br>
><br>
>The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a<br>
username<br>
>and password that it's sending out. But the INVITE is responded by
the<br>
>Asterisk with "SIP/2.0 403 Forbidden"<br>
><br>
>I've changed the INVITE message to mask the real telephone numbers, SIP<br>
>server, passwords, and IP addresses, but I did that using search and<br>
replace<br>
>so the structure is intact.<br>
><br>
>What do I need to configure in the "Incoming Settings" panel for
the CS<br>
>1500's INVITE to my Asterisk server to work? I've tried all kinds of<br>
>combinations of user,username,authname using +15552027020,host with IP<br>
>and/or DNS name, but nothing appears to work.<br>
><br>
><br>
><br>
Do a sip debug on the asterisk console and see if it is actually is<br>
matching one of your sip.conf entries during an invite from the CS1500.<br>
Look for a line that says something like 'Found Peer....bla bla bla'.<br>
If you dont see that line, then you are not even adding the correct<br>
sip.conf entry to match the invite from the CS1500.<br>
<br>
Andres<br>
<a href="http://www.telesip.net" target="_blank">http://www.telesip.net</a><br>
<br>
>Frank<br>
><br>
>INVITE message from Wireshark packet capture:<br>
><br>
>INVITE <a href="mailto:sip%3A%2B15552027020@sip.acme.com">sip:+15552027020@sip.acme.com</a>
SIP/2.0<br>
>From:<br>
><<a href="mailto:sip%3A5552022441@172.16.10.40">sip:5552022441@172.16.10.40</a>>;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2d<br>
b<br>
>ba4<br>
>To: <<a href="mailto:sip%3A%2B15552027020@sip.acme.com">sip:+15552027020@sip.acme.com</a>><br>
>Call-ID: <a
href="mailto:f379f62-29173-3895-b14271f5-40802-45378@172.16.10.40">f379f62-29173-3895-b14271f5-40802-45378@172.16.10.40</a><br>
>CSeq: 5102 INVITE<br>
>Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598<br>
>User-Agent: Nortel CS1500UA/v02.00.REL01<br>
>Accept: application/sdp<br>
>P-Asserted-Identity: <<a href="mailto:sip%3A5552022441@172.16.10.40">sip:5552022441@172.16.10.40</a>;user=phone><br>
>Privacy: none<br>
>Remote-Party-ID: <<a href="mailto:sip%3A5552022441@172.16.10.40">sip:5552022441@172.16.10.40</a>;user=phone>;
party=calling;<br>
>privacy=off<br>
>Max-Forwards: 70<br>
>Supported: 100rel,replaces<br>
>Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK<br>
>Contact: <<a href="mailto:sip%3A5552022441@172.16.10.40">sip:5552022441@172.16.10.40</a>><br>
>Authorization: Digest<br>
>username="username",realm="asterisk",nonce="118af2b0",uri="sip:+15552027020<br>
@<br>
><a href="http://sip.acme.com" target="_blank">sip.acme.com</a>",response="111e63ec2a1f3ebabefe4f7dae4087a1",algorithm=MD5<br>
>Content-Type: application/SDP<br>
>Content-Length: 167<br>
><br>
>v=0<br>
>o=- 2973921782 2973921782 IN IP4 172.16.10.65<br>
>s=SIP Call<br>
>c=IN IP4 172.16.10.65<br>
>t=0 0<br>
>m=audio 36224 RTP/AVP 0<br>
>a=rtpmap:0 PCMU/8000<br>
>a=ptime:20<br>
>a=sendrecv<br>
><br>
><br>
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><br>
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<p class=MsoNormal style='margin-bottom:12.0pt'><br>
<br clear=all>
<br>
-- <br>
Personal Development Without The Silly Stuff: <a href="http://AllanDib.com">http://AllanDib.com</a><o:p></o:p></p>
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