On Wed, Jun 25, 2014 at 9:30 AM, Rafael Visser <[email protected]> wrote: > Hi gurus!!! > > I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn > Every minute asterisk sends an OPTION Request, i beleived that it's related > to qualify functions. > The every minute annoyng answer of the pstn is "403 Forbidden". > Some people told that asterisk is not sending the username in the OPTION, > required by the pstn.
> Is it wright? > How can i instruct FREEPBX to send the username in the option request? It may be worth asking on the FreePBX forums at http://community.freepbx.org/ as the Asterisk users who use FreePBX are generally monitoring that community. Many people here won't be able to answer your question *within the context* of FreePBX configuration. Your question is also not clear. You should ask the provider specifically which header and where in what URI they want to see the "username" in. If this wasn't FreePBX I'd tell you to just try setting the callerid and fromuser options for the corresponding SIP peer. I don't want to pretend to know FreePBX, so I still recommend you go ask on their forum to get better assistance. Good luck! -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
