asterisk users - Feb 2016 - Asterisk 13.6.0/The simplest TCP configuration does not work

Does this help:

Session Initiation Protocol (REGISTER)
    Request-Line: REGISTER sip:1.2.3.4;transport=TCP SIP/2.0
        Method: REGISTER
        Request-URI: sip:1.2.3.4;transport=TCP
            Request-URI Host Part: 1.2.3.4
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/TCP 192.168.1.15:47053
;branch=z9hG4bK-d8754z-5e3d9f441f1de1d3-1---d8754z-;rport;transport=TCP
            Transport: TCP
            Sent-by Address: 192.168.1.15
            Sent-by port: 47053
            Branch: z9hG4bK-d8754z-5e3d9f441f1de1d3-1---d8754z-
            RPort: rport
            transport=TCP
        Max-Forwards: 70
        Contact: <sip:5678 at 192.168.1.15:47053
;rinstance=bea6f11f37c55605;transport=TCP>
            Contact URI: sip:5678 at 192.168.1.15:47053
;rinstance=bea6f11f37c55605;transport=TCP
                Contact URI User Part: 5678
                Contact URI Host Part: 192.168.1.15
                Contact URI Host Port: 47053
                Contact URI parameter: rinstance=bea6f11f37c55605
                Contact URI parameter: transport=TCP
        To: <sip:5678 at 1.2.3.4;transport=TCP>
            SIP to address: sip:5678 at 1.2.3.4;transport=TCP
                SIP to address User Part: 5678
                SIP to address Host Part: 1.2.3.4
                SIP To URI parameter: transport=TCP
        From: <sip:5678 at 1.2.3.4;transport=TCP>;tag=fc31c046
            SIP from address: sip:5678 at 1.2.3.4;transport=TCP
                SIP from address User Part: 5678
                SIP from address Host Part: 1.2.3.4
                SIP From URI parameter: transport=TCP
            SIP from tag: fc31c046
        Call-ID: ODRiMjBhNGY5MWJjMDFkNjk4MzRhYzg1ZTE3ZWM3Y2M.
        CSeq: 1 REGISTER
            Sequence Number: 1
            Method: REGISTER
        Expires: 70
        Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
INFO, SUBSCRIBE
        Supported: replaces, norefersub, extended-refer, timer,
X-cisco-serviceuri
        User-Agent: Z 3.3.25608 r25552
        Allow-Events: presence, kpml
        Content-Length: 0


On Mon, Feb 15, 2016 at 2:53 PM, Joshua Colp <jcolp at digium.com> wrote:
> Sonny Rajagopalan wrote:
>
>> Thanks for the mighty quick response, Joshua!
>>
>> I am using Zoiper on Linux softclient:
>> REGISTER sip:<ipAddr>;transport=TCP SIP/2.0
>>
>
> That's the request URI, not the Contact header. The Contact contains
the
> URI that the server should dial to reach the client. The full message would
> be useful.
>
> Cheers,
>
>
> --
> Joshua Colp
> Digium, Inc. | Senior 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:
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Sonny Rajagopalan wrote:
> Does this help:
Yes, the transport parameter is in the Contact header so it's 
interesting it didn't work. If you use pjsip show contacts what is the 
contact for the AOR?

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

Nope, there are no contacts to  show that pertain to these endpoints (only
my SIP trunks show up).

On Mon, Feb 15, 2016 at 5:31 PM, Joshua Colp <jcolp at digium.com> wrote:
> Sonny Rajagopalan wrote:
>
>> Does this help:
>>
>
> Yes, the transport parameter is in the Contact header so it's
interesting
> it didn't work. If you use pjsip show contacts what is the contact for
the
> AOR?
>
>
> --
> Joshua Colp
> Digium, Inc. | Senior 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
>
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