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 <j...@stratustalk.com <mailto:j...@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 <j...@stratustalk.com <mailto:j...@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:7...@voxout.voxbone.com> <mailto:sip:7...@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:7...@voxout.voxbone.com> <mailto:sip:7...@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
-- _____________________________________________________________________ -- 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
