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>
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 >> -- >> _____________________________________________________________________ >> -- 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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180508/4edcaf31/attachment.html>
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 >> -- >> _____________________________________________________________________ >> -- 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 > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180508/556f70c8/attachment-0001.html>