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>.
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>.
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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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 > -- > _____________________________________________________________________ > -- 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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180508/1d824a80/attachment.html>
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 > -- > _____________________________________________________________________ > -- 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 > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180508/9120df08/attachment.html>