On Sat, Dec 2, 2017, at 06:33 PM, Carlos Chavez wrote: > I am having a really bad day trying to get incoming calls to work > on Asterisk 13 with PJSIP. We just migrated from Asterisk 1.8 where > everything was working but there seems that something got lost in > translation. No matter what I try I always get a 401 Unauthorized > message when receiving a call from the PSTN provider. I can make calls > and the registration is working. I have tried to set the identify to an > endpoint that does not have an auth defined. Anyone using Alestra SIP > trunks in Mexico?
<snip> > > My identify is: > > ============================================= > endpoint : Alestra > match : 200.94.59.150/255.255.255.255 > match_header : > srv_lookups : true > > > It does not matter if I use the original endpoint or an endpoint with no > auth. Asterisk will still reject the call. Any tips? How can I make > sure that the identify is being used? If you turn up the core debug to level 4 and send it to the console it will tell you what it is doing. I'd also suggest providing the endpoint definition, and confirming it was loaded as expected. If it's not then you can look at the Asterisk console at load time and it will tell you what it did not like. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- 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