[asterisk-users] Incoming side of SIP trunk does not work unless I add "insecure=very"

Mon Jan 5 21:41:12 CST 2009

Try it by IP address instead of hostname as reverse DNS may not be
resolving. e.g. host=123.123.123.123

On Tue, Jan 6, 2009 at 2:25 PM, Frank Bulk <frnkblk at iname.com> wrote:

> This is what I have in my configuration now:
>
> [ACME]
> host=sip.acme.com
> username=username
> secret=password
> type=friend
>
> I've done a SIP debug before, but I've done it again with the above
> configuration:
>        No user '5551236049' in SIP users list
>        Found peer 'ACME' for '5551236049' from 172.16.10.40:5060
> after which "SIP/2.0 401 Unauthorized" is issued after the un-authenticated
> INVITE and "SIP/2.0 403 Forbidden" after the authenticated INVITE.
>
> When I add "insecure=very", this is what the SIP debug shows:
>        No user '5551236049' in SIP users list
>        Found peer 'ACME' for '5551236049' from 172.16.10.40:5060
>        Found RTP audio format 0
>        Peer audio RTP is at port 172.16.10.65:36272
>        Found audio description format PCMU for ID 0
>        Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4
> (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
>        Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer -
> 0x0 (nothing), combined - 0x0 (nothing)
>        Peer audio RTP is at port 172.16.10.65:36272
>        Looking for +15552127020 in from-sip-external (domain sip.acme.com)
>        list_route: hop: <sip:5551236049 at 172.16.10.40<sip%3A5551236049 at 172.16.10.40>
> >
>
> It isn't very clear (to me) from the success how the "insecure=very" helps.
>
> Frank
>
> -----Original Message-----
> From: Andres [mailto:andres at telesip.net]
> Sent: Monday, January 05, 2009 7:43 PM
> To: frnkblk at iname.com; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [asterisk-users] Incoming side of SIP trunk does not work
> unless I add "insecure=very"
>
> Frank Bulk - iName.com wrote:
>
> >The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not
> >work unless I add "insecure=very" to my "Outgoing settings", but I don't
> >want to do that.  I do want to authenticate.  Outgoing (Asterisk PBX to
> >Class 5 switch) calls do authenticate and work.
> >
> >The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a
> username
> >and password that it's sending out.  But the INVITE is responded by the
> >Asterisk with "SIP/2.0 403 Forbidden"
> >
> >I've changed the INVITE message to mask the real telephone numbers, SIP
> >server, passwords, and IP addresses, but I did that using search and
> replace
> >so the structure is intact.
> >
> >What do I need to configure in the "Incoming Settings" panel for the CS
> >1500's INVITE to my Asterisk server to work?  I've tried all kinds of
> >combinations of user,username,authname using +15552027020,host with IP
> >and/or DNS name, but nothing appears to work.
> >
> >
> >
> Do a sip debug on the asterisk console and see if it is actually is
> matching one of your sip.conf entries during an invite from the CS1500.
> Look for a line that says something like 'Found Peer....bla bla bla'.
> If you dont see that line, then you are not even adding the correct
> sip.conf entry to match the invite from the CS1500.
>
> Andres
> http://www.telesip.net
>
> >Frank
> >
> >INVITE message from Wireshark packet capture:
> >
> >INVITE sip:+15552027020 at sip.acme.com <sip%3A%2B15552027020 at sip.acme.com>SIP/2.0
> >From:
> ><sip:5552022441 at 172.16.10.40 <sip%3A5552022441 at 172.16.10.40>
> >;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2d
> b
> >ba4
> >To: <sip:+15552027020 at sip.acme.com <sip%3A%2B15552027020 at sip.acme.com>>
> >Call-ID: f379f62-29173-3895-b14271f5-40802-45378 at 172.16.10.40
> >CSeq: 5102 INVITE
> >Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598
> >User-Agent: Nortel CS1500UA/v02.00.REL01
> >Accept: application/sdp
> >P-Asserted-Identity: <sip:5552022441 at 172.16.10.40<sip%3A5552022441 at 172.16.10.40>
> ;user=phone>
> >Privacy: none
> >Remote-Party-ID: <sip:5552022441 at 172.16.10.40<sip%3A5552022441 at 172.16.10.40>;user=phone>;
> party=calling;
> >privacy=off
> >Max-Forwards: 70
> >Supported: 100rel,replaces
> >Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, PRACK
> >Contact: <sip:5552022441 at 172.16.10.40 <sip%3A5552022441 at 172.16.10.40>>
> >Authorization: Digest
>
> >username="username",realm="asterisk",nonce="118af2b0",uri="sip:+15552027020
> @
> >sip.acme.com",response="111e63ec2a1f3ebabefe4f7dae4087a1",algorithm=MD5
> >Content-Type: application/SDP
> >Content-Length: 167
> >
> >v=0
> >o=- 2973921782 2973921782 IN IP4 172.16.10.65
> >s=SIP Call
> >c=IN IP4 172.16.10.65
> >t=0 0
> >m=audio 36224 RTP/AVP 0
> >a=rtpmap:0 PCMU/8000
> >a=ptime:20
> >a=sendrecv
> >
> >
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> >
>
>
>
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