Hi Joshua, Thanks for the reply. In this case we get a special SIP header in the 302, but I guess we'll need to find another solution to use it.
On Wed, 27 Apr 2022 at 21:27, Joshua C. Colp <jc...@sangoma.com> wrote: > On Wed, Apr 27, 2022 at 5:33 AM David Cunningham < > dcunning...@voisonics.com> wrote: > >> Hi Jon, >> >> Thank you for the reply. We wanted to read a particular SIP header in the >> 302 Moved response, but it seems that Asterisk creates a Local channel for >> the redirected call and the SIP_HEADER() function isn't available, so we >> can't really do what we wanted at all. >> > > Neither chan_sip or chan_pjsip provide such ability even if you had access > to the SIP or PJSIP channel. SIP_HEADER() gets headers from an incoming > INVITE, same for PJSIP_HEADER(). > > -- > Joshua C. Colp > Asterisk Technical Lead > Sangoma Technologies > Check us out at www.sangoma.com and 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 -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782
-- _____________________________________________________________________ -- 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