On Fri, Jun 9, 2017, at 11:30 AM, Daniel Tryba wrote: > With pjsip (asterisk 13.14.1) I see the problem that an anonymous from > header gets user=phone appendend to the URI if user_eq_phone=yes is > specified: > > On the incoming leg: > From: anonymous > <sip:anonymous@anonymous.invalid:5060>;tag=Q5zBj7BMnvI6Fe6O2866fox3ZHmn-smt > Get transformed to > From: "Anonymous" > <sip:anonymous@anonymous.invalid;user=phone>;tag=fa3cb748-6af9-485f-8a70-a2b9ad40b13a > on the outgoing leg. > > Setting user_eq_phone = no will result in user=phone not being added. > The upstream provide demands user=phone in URIs if the username > resembles a phonenumber, but declines the INVITE if user=phone is > present on an anonymous username.
<snip> > sip_uri->user should be "anonymous" > AST_DIGIT_ANY is: #define AST_DIGIT_ANYNUM "0123456789" > > So in the for loop the first char of sip_uri->user should result in a > NULL from strchr. Leaving i at the value 0, which is smaller than the > length of sip_uri->user. And thus the function should return before > adding the user=phone. So why is user=phone being added? What seems to be happening is that the session is being set up and the user=phone parameter added. It's only after that the values are updated to be Anonymous and the user=phone parameter is left there. Please file an issue[1] with the description above. [1] https://issues.asterisk.org/jira -- 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