[asterisk-users] multi step auth?

[Kseniya Blashchuk]

Try to set fromuser=number in your sip provider peer configuration

On Tue, May 8, 2018, 11:05 PM Jeff LaCoursiere <jeff at stratustalk.com> wrote:

>
> 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>
> 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>
>> 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> <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> <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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>>
>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
--
BR, Kseniya
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