[asterisk-users] Asterisk (PJSIP) registers with sips Contact URI, but why?

[George Joseph]

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
>
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-- 
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
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