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>
Sonny Rajagopalan
2016-Feb-17 03:08 UTC
[asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work
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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Joshua Colp
2016-Feb-17 11:35 UTC
[asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work
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