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

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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I can confirm that the server is receiving the SIP request, but simply
doesn't do anything with it (log from the server below). Does this have
anything to do with how PJSIP was compiled or configured?:

Session Initiation Protocol (REGISTER)
    Request-Line: REGISTER sip:11.12.13.14 SIP/2.0
        Method: REGISTER
        Request-URI: sip:11.12.13.14
            Request-URI Host Part: 11.12.13.14
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/TCP 192.168.1.16:54402
;rport;branch=z9hG4bKPjpaCDxnhZT22z-O183o5uZzMHNwTNrpkl;alias
            Transport: TCP
            Sent-by Address: 192.168.1.16
            Sent-by port: 54402
            RPort: rport
            Branch: z9hG4bKPjpaCDxnhZT22z-O183o5uZzMHNwTNrpkl
            alias
        Route: <sip:11.12.13.14;transport=tcp;lr>
            Route URI: sip:11.12.13.14;transport=tcp;lr
                Route Host Part: 11.12.13.14
                Route URI parameter: transport=tcp
                Route URI parameter: lr
        Max-Forwards: 70
        From: <sip:987654321 at 11.12.13.14
>;tag=Qb12fSdMpSBV4YJ2e4LGtM3biO.rPtcQ
            SIP from address: sip:987654321 at 11.12.13.14
                SIP from address User Part: 987654321
                SIP from address Host Part: 11.12.13.14
            SIP from tag: Qb12fSdMpSBV4YJ2e4LGtM3biO.rPtcQ
        To: <sip:987654321 at 11.12.13.14>
            SIP to address: sip:987654321 at 11.12.13.14
                SIP to address User Part: 987654321
                SIP to address Host Part: 11.12.13.14
        Call-ID: 8NDmEFaT2lmQRMUBf77UrRKRBIc3cT0h
        CSeq: 29457 REGISTER
            Sequence Number: 29457
            Method: REGISTER
        Supported: outbound, path
        Contact: <sip:987654321 at 192.168.1.16:54402
;transport=TCP;ob>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000078230be6>"
            Contact URI: sip:987654321 at 192.168.1.16:54402;transport=TCP;ob
                Contact URI User Part: 987654321
                Contact URI Host Part: 192.168.1.16
                Contact URI Host Port: 54402
                Contact URI parameter: transport=TCP
                Contact URI parameter: ob
            Contact parameter: reg-id=1
            Contact parameter:
+sip.instance="<urn:uuid:00000000-0000-0000-0000-000078230be6>"\r\n
        Expires: 900
        Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE,
NOTIFY, REFER, MESSAGE, OPTIONS
        Content-Length:  0


On Mon, Feb 15, 2016 at 6:01 PM, Sonny Rajagopalan <
sonny.rajagopalan at gmail.com> wrote:
> 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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Sonny Rajagopalan wrote:
> I can confirm that the server is receiving the SIP request, but simply
> doesn't do anything with it (log from the server below). Does this have
> anything to do with how PJSIP was compiled or configured?:
TCP support is enabled in PJSIP by default. If you do "pjsip set logger 
on" does the message show up? What is the COMPLETE console output when a 
client connects? We have tests which cover TCP and they are working, so 
it's likely something environment specific.

-- 
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