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

2016-05-03 Thread George Joseph 
On Tue, May 3, 2016 at 9:39 AM, Sebastian Damm  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:1234...@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: 
> From: ;tag=vhDrzKtv9lMR53ZJFgVTnvGcACJiN6Aa
> To: 
> Call-ID: nzgHdLyliuBwecmae2Y..0oY2DqYjH0V
> CSeq: 14861 REGISTER
> Contact: 
> 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

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

2016-05-03 Thread Sebastian Damm 
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:1234...@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: 
From: ;tag=vhDrzKtv9lMR53ZJFgVTnvGcACJiN6Aa
To: 
Call-ID: nzgHdLyliuBwecmae2Y..0oY2DqYjH0V
CSeq: 14861 REGISTER
Contact: 
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