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Sonny Rajagopalan

2016-Feb-15 19:22 UTC

[asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Thanks for the mighty quick response, Joshua!

I am using Zoiper on Linux softclient:
REGISTER sip:<ipAddr>;transport=TCP SIP/2.0

Changed the port back to 5060.


On Mon, Feb 15, 2016 at 1:40 PM, Joshua Colp <jcolp at digium.com> wrote:
> Sonny Rajagopalan wrote:
>
> <snip>
>
>
> *CLI> pjsip set logger on
>> PJSIP Logging enabled
>> [Feb 15 18:28:12] WARNING[30075]: pjsip:0 <?>: tsx0x7f14b0003
.Temporary
>> failure in sending Request msg OPTIONS/cseq=52002 (tdta0x7f14b0001470),
>> will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
>> [Feb 15 18:28:12] WARNING[30075]: pjsip:0 <?>: tsx0x7f14b0003
.Temporary
>> failure in sending Request msg OPTIONS/cseq=52002 (tdta0x7f14b0001470),
>> will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
>> [Feb 15 18:28:12] WARNING[30075]: pjsip:0 <?>: tsx0x7f14b0003
.Temporary
>> failure in sending Request msg OPTIONS/cseq=52002 (tdta0x7f14b0001470),
>> will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
>> [Feb 15 18:28:12] WARNING[30075]: pjsip:0 <?>: tsx0x7f14b0003
.Failed to
>> send Request msg OPTIONS/cseq=52002 (tdta0x7f14b0001470)! err=171060
>> (Unsupported transport (PJSIP_EUNSUPTRANSPORT))
>> *CLI> core set debug 99
>> Core debug was OFF and is now 99.
>> [Feb 15 18:28:27] WARNING[30133]: pjsip:0 <?>: tsx0x7f14b4003
.Temporary
>> failure in sending Request msg OPTIONS/cseq=49786 (tdta0x7f14b40014f0),
>> will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
>> [Feb 15 18:28:27] WARNING[30133]: pjsip:0 <?>: tsx0x7f14b4003
.Temporary
>> failure in sending Request msg OPTIONS/cseq=49786 (tdta0x7f14b40014f0),
>> will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
>> [Feb 15 18:28:27] WARNING[30133]: pjsip:0 <?>: tsx0x7f14b4003
.Temporary
>> failure in sending Request msg OPTIONS/cseq=49786 (tdta0x7f14b40014f0),
>> will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
>> [Feb 15 18:28:27] WARNING[30133]: pjsip:0 <?>: tsx0x7f14b4003
.Failed to
>> send Request msg OPTIONS/cseq=49786 (tdta0x7f14b40014f0)! err=171060
>> (Unsupported transport (PJSIP_EUNSUPTRANSPORT))
>>
>
> This will happen if the URI added does not contain ;transport=tcp which
> informs things to use TCP. If the device registering doesn't do this
then
> it will try to use a UDP transport instead, if not available then it will
> fail.
>
> What is the REGISTER from the device?
>
> --
> 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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Joshua Colp

2016-Feb-15 19:53 UTC

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[asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

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

Sonny Rajagopalan

2016-Feb-15 22:29 UTC

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[asterisk-users] 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:
>               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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