Try it by IP address instead of hostname as reverse DNS may not be resolving. e.g. host=123.123.123.123<br><br><div class="gmail_quote">On Tue, Jan 6, 2009 at 2:25 PM, Frank Bulk <span dir="ltr"><<a href="mailto:frnkblk@iname.com">frnkblk@iname.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">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>
<font color="#888888"><br>
Frank<br>
</font><div class="Ih2E3d"><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<br>
</div><div><div></div><div class="Wj3C7c">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>
><br>
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<br>
<br>
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