[asterisk-dev] Fwd: Asterisk and Max TNT PRI to SIP Authentication Issue, a little closer

[alex at pilosoft.com]

Put a ser in between asterisk and TNT

-alex
On Wed, 8 Nov 2006, JR Richardson wrote:

> Sorry for the cross post.  I really think this needs to be brought to
> developers attention, I could be wrong though.
> 
> I've also been working with some TNT folks to see if I can get the TNT
> not to send user=phone in the sip invite.
> 
> Does anyone know of a solution to tell Asterisk sip not require
> authentication if there is a user= within the invite?
> insecure=port,invite is not doing anything for me.
> 
> Thanks.
> 
> JR
> 
> 
> ---------- Forwarded message ----------
> From: JR Richardson <jmr.richardson at gmail.com>
> Date: Nov 8, 2006 2:18 PM
> Subject: Re: Asterisk and Max TNT PRI to SIP Authentication Issue, a
> little closer
> To: asterisk-users at lists.digium.com
> 
> 
> After mocking up an unauthenticated call from a different device, a
> spa942 phone, I found something strange in the SIP debug between the
> phone and the TNT.
> 
> Asterisk is accepting unauthenticated calls as long as there is not a
> "user" in the SIP header from the calling device.
> 
> Invite from the MAX: does not get passed to the dial plan
> 
> <-- SIP read from 10.10.14.131:5060:
> INVITE sip:2145551212 at 10.10.14.121:5060;user=phone SIP/2.0
> To:   <sip:2145551212 at 10.10.14.121:5060;user=phone>
> From: "NO CID NAME"
> <sip:1239 at 10.10.14.131:5060;user=phone>;tag=1e82fc7f-1fb33c15-830e0a0a
> 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
> Call-ID: a09233dd-4a-1fb33c15 at 10.10.14.131
> CSeq: 803597 INVITE
> Via: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK007aa1ced4ace55a
> Max-Forwards: 70
> Contact: <sip:1239 at 10.10.14.131:5060;user=phone>
> Supported: replaces
> Content-Type: application/sdp
> Accept: application/sdp
> Accept-Encoding:
> Accept-Language: en
> User-Agent: Lucent-Universal-Gateway
> Content-Length: 232
> 
> Invite from the phone: gets passed to the dial plan in the [general] context=
> 
> <-- SIP read from 10.10.11.51:5060:
> INVITE sip:2145551212 at 10.10.14.121 SIP/2.0
> Via: SIP/2.0/UDP 10.10.11.51:5060;branch=z9hG4bK-9fd7c0a9
> From: "2001" <sip:2001 at 10.10.11.50>;tag=59f6242028d88691o0
> To: <sip:2145551212 at 10.10.14.121>
> Call-ID: ec4c728c-9f2aab15 at 10.10.11.51
> CSeq: 101 INVITE
> Max-Forwards: 70
> Contact: "2001" <sip:1001 at 10.10.11.51:5060>
> Expires: 240
> User-Agent: Linksys/SPA942-4.1.12(a)
> Content-Length: 391
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Content-Type: application/sdp
> 
> The invite string from the TNT:
> INVITE sip:2145551212 at 10.10.14.121:5060;user=phone SIP/2.0
> 
> The invite string from the phone:
> INVITE sip:2145551212 at 10.10.14.121 SIP/2.0
> 
> It appears that if a user= field is in the invite message, Asterisk
> looks for a user context and requires authentication.
> 
> So the insecure=port,invite option should also include an
> insecure=user option to disregard any user info in the invite.  Is
> there is another mechanism in Asterisk to disregard any user info from
> an invite?
> 
> Thanks.
> 
> JR
> --
> JR Richardson
> Engineering for the Masses
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