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. 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
