Sonny Rajagopalan
2016-Feb-15 22:29 UTC
[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
>
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> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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Joshua Colp
2016-Feb-15 22:31 UTC
[asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work
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
Sonny Rajagopalan
2016-Feb-15 23:01 UTC
[asterisk-users] 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160215/d9471ba0/attachment.html>