[asterisk-users] multi step auth?

Jeff LaCoursiere jeff at stratustalk.com
Tue May 8 15:04:55 CDT 2018


Thats till doesn't change the SIP header.  Basically they want to send a 
RE INVITE and authenticate my DID number.  But my DID number does not 
have a peer or user entry in sip.conf.  Perhaps I am answering my own 
question, but is that the only way this is going to work?

Thanks,

j

On 05/08/2018 02:54 PM, Khalil Khamlichi wrote:
> try adding a + sign for the number
>
>     same => n,Set(CALLERID(all)=17864089672 <+17864089672>)
>
>
>
>
> On Tue, May 8, 2018, 8:51 PM Jeff LaCoursiere <jeff at stratustalk.com 
> <mailto:jeff at stratustalk.com>> wrote:
>
>
>     I *am* doing that, as I assumed it would be required just for the
>     911 mapping we have provided, but that doesn't change the SIP header.
>
>     Cheers,
>
>     j
>
>     On 05/08/2018 02:41 PM, Khalil Khamlichi wrote:
>>     try setting the callerid with
>>
>>     same => n,Set(CALLERID(all)=17864089672 <17864089672>)
>>
>>     ofcourse for each customer you will need to provide his own did.
>>
>>
>>     On Tue, May 8, 2018, 8:37 PM Jeff LaCoursiere
>>     <jeff at stratustalk.com <mailto:jeff at stratustalk.com>> wrote:
>>
>>         Hi,
>>
>>         We have been using Voxbone for some time for origination, and
>>         they now offer E911 services.  We are trying to set this up
>>         and having trouble meeting their authentication requirements.
>>
>>         I setup a peer as I normally would, with user/pass as they
>>         supplied ("lacoursj", "pass"), but my calls are rejected. 
>>         Their support is asking that I follow this auth mechanism:
>>
>>         1st step - You send an INVITE message.
>>         2nd step - We respond with a 407.
>>         3rd step - You send a RE INVITE message including your
>>         credentials.
>>
>>          The tricky bit seems to be that they want the original
>>         INVITE to look like:
>>
>>         From: <sip:*17864089672*@X.X.X.X:60060>;tag=as00771983.
>>         To: <sip:777 at voxout.voxbone.com>
>>         <mailto:sip:777 at voxout.voxbone.com>.
>>         Contact: <sip:*17864089672*@X.X.X.X:60060>.
>>
>>         The "1786..." above is meant to be the DID number that is
>>         placing the 911 call. Our DID numbers don't have peer or user
>>         entries in sip.conf. My peer isn't sending that, though, it
>>         is sending:
>>
>>         From: <sip:*lacoursj*@X.X.X.X:60060>;tag=as00771983.
>>         To: <sip:777 at voxout.voxbone.com>
>>         <mailto:sip:777 at voxout.voxbone.com>.
>>         Contact: <sip:*lacoursj*@X.X.X.X:60060>.
>>
>>         They claim that 'lacoursj' shouldn't be sent until step 3.
>>
>>         I have never been asked to authenticate this way... can
>>         asterisk chan_sip do it?
>>
>>         Cheers,
>>
>>         j
>>         -- 
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>>
>
>     -- 
>     _____________________________________________________________________
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>
>     Check out the new Asterisk community forum at:
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>
>     New to Asterisk? Start here:
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