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. 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 I thought that would be the equivalent of no authentication object, so I tried that. However, that did not work either. I tried changing the endpoint to have no auth and outbound_auth settings. I tried changing the endpoint to use the auth instead of the outbound_auth. The SIP provider even changed the username and passwords to blank. I followed suit and changed the pjsip.conf user and password related settings to blank. Our sip.conf (running in a different VM on Asterisk 13.0.0) settings look like this... [xxxxx] type = friend qualify = no nat = yes host = xxxxx defaultuser = yyyyy secret = zzzzz incominglimit = 4 accountcode = 9 port = 5060 context = TestApp dtmfmode = auto insecure = invite fromdomain = xxxxx fromuser = yyyyy sendrpid = yes trustrpid = yes canreinvite = no For the pjsip.conf settings (Asterisk 13.0.0), I have [transport1] type = transport bind = 0.0.0.0 protocol = udp [xxxxx] type = aor max_contacts = 1 remove_existing = yes contact = sip:yyyyy at xxxxx:5060 [auth9] type = auth username = yyyyy password = zzzzz [xxxxx] type = endpoint context = TestApp transport = transport1 outbound_auth = auth9 aors = xxxxx accountcode = 9 dtmf_mode = rfc4733 device_state_busy_at = 4 ;force_rport = yes ; also tried with this setting, but it still didn't help rtp_symmetric = yes rewrite_contact = yes from_domain = xxxxxx from_user = yyyyy send_rpid = yes trust_id_inbound = yes direct_media = no Have a great day! Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141210/a0defffa/attachment-0001.html>
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.confThat 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
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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] 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.confThat 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