Sebastian Damm
2016-May-03 15:39 UTC
[asterisk-users] Asterisk (PJSIP) registers with sips Contact URI, but why?
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:1234567 at example.org\;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:1234567 at example.org>;tag=vhDrzKtv9lMR53ZJFgVTnvGcACJiN6Aa To: <sip:1234567 at example.org> Call-ID: nzgHdLyliuBwecmae2Y..0oY2DqYjH0V CSeq: 14861 REGISTER Contact: <sips:1234567 at 9.8.7.6:55664;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. 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
George Joseph
2016-May-03 16:07 UTC
[asterisk-users] Asterisk (PJSIP) registers with sips Contact URI, but why?
On Tue, May 3, 2016 at 9:39 AM, Sebastian Damm <damm at sipgate.de> 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:1234567 at example.org\;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:1234567 at example.org>;tag=vhDrzKtv9lMR53ZJFgVTnvGcACJiN6Aa > To: <sip:1234567 at example.org> > Call-ID: nzgHdLyliuBwecmae2Y..0oY2DqYjH0V > CSeq: 14861 REGISTER > Contact: <sips:1234567 at 9.8.7.6:55664;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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160503/391db40b/attachment.html>