On Tue, May 3, 2016 at 9:39 AM, Sebastian Damm <[email protected]> wrote:
> Hi, > > I'm registering an Asterisk against my TLS capable service, using > res_pjsip. My config looks like this: > > [devtrunk_reg] > type=registration > outbound_auth=devtrunk_auth > server_uri=sip:example.org\;transport=tls > client_uri=sip:[email protected]\;transport=tls > outbound_proxy=sip:dev.example.org\;transport=tls\;lr > contact_user=1234567 > retry_interval=60 > expiration=600 > line=yes > endpoint=222 > > [devtrunk_auth] > type=auth > auth_type=userpass > username=1234567 > password=secret > realm=example.org > > > It registers fine, but this is what the REGISTER request looks like: > > <--- Transmitting SIP request (903 bytes) to TLS:1.2.3.4:5061 ---> > REGISTER sip:example.org;transport=tls SIP/2.0 > Via: SIP/2.0/TLS > 9.8.7.6:55664;rport;branch=z9hG4bKPjNlqlgmSOP7O4LqOTUqJtFZB8fTmc0ZKL;alias > Route: <sip:dev.example.org;transport=tls;lr> > From: <sip:[email protected]>;tag=vhDrzKtv9lMR53ZJFgVTnvGcACJiN6Aa > To: <sip:[email protected]> > Call-ID: nzgHdLyliuBwecmae2Y..0oY2DqYjH0V > CSeq: 14861 REGISTER > Contact: <sips:[email protected]:55664;transport=TLS;line=dhslasr> > Expires: 600 > Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, > UPDATE, PRACK, MESSAGE, REFER, REGISTER > Max-Forwards: 70 > User-Agent: Asterisk PBX 13.8.2 > Content-Length: 0 > > What I really don't like is the Contact line. It starts with sips > instead of sip. This makes inbound calls not work because the server > sends a sip Contact header instead of sips. And Asterisk rejects that. > res_pjsip_outbound_registration is hard-coded to send "sips" on a secure transport. I'd suggest opening a issue at issues.asterisk.org. We should probably use the scheme from the registration client_uri. > > In the header of the 480 response I see this line: > > Warning: 381 SIP "SIPS Required" > > Since I can't reconfigure the server to send sips Contact URIs, I need > Asterisk to send out a contact URI in the register, that starts with > sip: as well. Then inbound calls would work. > > Is there any way to get rid of this sips URI? > > Interestingly, when sending out calls, the Contact URI starts with sip > instead of sips, so outbound calls work. > > Best Regards, > Sebastian > > -- > _____________________________________________________________________ > -- 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 > -- George Joseph Digium, Inc. | 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
