I should mention, I am actually sending this via AMI in both the chan_sip and the pjsip case.
Pjsip originate... Action: Originate ActionID: S8 Channel: PJSIP/outbound.vitelity.net/1234567890 Exten: createcall Context: TestApp Priority: 1 Timeout: 60000 CallerID: Dan Cropp<1234> Variable: CALLERID(num-pres)=allowed_passed_screened Async: true Chan_sip based originate... Action: Originate ActionID: S8 Channel: SIP/outbound.vitelity.net/1234567890 Exten: createcall Context: TestApp Priority: 1 Timeout: 60000 CallerID: Dan Cropp<1234> Variable: CALLERID(num-pres)=allowed_passed_screened Async: true Have a great day! Dan -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Dan Cropp Sent: Wednesday, December 10, 2014 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP configuration question 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 -- _____________________________________________________________________ -- 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
