Thank you for the speedy reply. My originate string is something like the following where xxxxx is really the sip provider's supplied IP address 1234567890 is really the phone number I am dialing
PJSIP/outbound.vitelity.net/1234567890 In the chan_sip based solution, it's... SIP/outbound.vitelity.net/1234567890 Have a great day! Dan -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Joshua Colp Sent: Wednesday, December 10, 2014 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP configuration question Kia ora, Dan Cropp wrote: > I'm working with a SIP provider to try and transition our sip > connection with them to PJSIP. I thought I had transitioned the > settings correctly, but whenever I attempt an Originate it never even > tries to send any PJSIP messages. What dial string are you providing to Originate? > I'm currently running Asterisk 13.0.0. > > Anyone have any suggestions as to what I am doing wrong? > > The SIP provider says the latest version of Asterisk they have anyone > using is Asterisk 11, so they have no PJSIP configuration experience. > > The only setting that I believe I haven't found a PJSIP settting for > is the "insecure=invite" from sip.conf That functionality exists in the form of the "identify" object. It does IP based matching of incoming traffic and to associate it with an endpoint. > > I thought that would be the equivalent of no authentication object, so > I tried that. However, that did not work either. Authentication controls authentication, it doesn't control how PJSIP associates traffic with a specific endpoint. They are separate things. I think before we get into config we need to see the dial string for your origination. Cheers, -- 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 -- 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 -- _____________________________________________________________________ -- 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
