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

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sonny.rajagopalan at g...
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PostPosted: Mon Feb 15, 2016 1:37 pm    Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat Reply with quote

This question was asked by Chirag on March 4 2015 earlier, but I am following exactly the same procedure here and I cannot even get my clients to register on Asterisk.

Here's my PJSIP.conf:


[transport-tcp]
type=transport
protocol=tcp
bind=0.0.0.0:5061

...


[endpoint_internal](!)
type=endpoint
context=from-internal
disallow=all
allow=!all,ulaw
direct_media=no
rtp_symmetric=yes
message_context=text-context


[auth_userpass](!)
type=auth
auth_type=userpass


[aor_dynamic](!)
type=aor
max_contacts=1
remove_existing=yes


;;; Configuration for user 99999999

...


I used basically exactly the same procedure for UDP based registration and it works just fine.


Any help is appreciated. The specific error I see on the CLI is:


Connected to Asterisk 13.6.0 currently running on ... (pid = 30046)
*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))
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jcolp at digium.com
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PostPosted: Mon Feb 15, 2016 1:41 pm    Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat Reply with quote

Sonny Rajagopalan wrote:

<snip>

Quote:
*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
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http://lists.digium.com/mailman/listinfo/asterisk-users
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sonny.rajagopalan at g...
Guest





PostPosted: Mon Feb 15, 2016 2:23 pm    Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat Reply with quote

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@digium.com (jcolp@digium.com)> wrote:
Quote:
Sonny Rajagopalan wrote:

<snip>

Quote:
*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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jcolp at digium.com
Guest





PostPosted: Mon Feb 15, 2016 2:54 pm    Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat Reply with quote

Sonny Rajagopalan wrote:
Quote:
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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sonny.rajagopalan at g...
Guest





PostPosted: Mon Feb 15, 2016 5:30 pm    Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat Reply with quote

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@192.168.1.15:47053;rinstance=bea6f11f37c55605;transport=TCP>
            Contact URI: sip:5678@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@1.2.3.4 ([email]sip%3A5678@1.2.3.4[/email]);transport=TCP>
            SIP to address: sip:5678@1.2.3.4 ([email]sip%3A5678@1.2.3.4[/email]);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@1.2.3.4 ([email]sip%3A5678@1.2.3.4[/email]);transport=TCP>;tag=fc31c046
            SIP from address: sip:5678@1.2.3.4 ([email]sip%3A5678@1.2.3.4[/email]);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@digium.com (jcolp@digium.com)> wrote:
Quote:
Sonny Rajagopalan wrote:
Quote:
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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jcolp at digium.com
Guest





PostPosted: Mon Feb 15, 2016 5:32 pm    Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat Reply with quote

Sonny Rajagopalan wrote:
Quote:
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 at g...
Guest





PostPosted: Mon Feb 15, 2016 6:02 pm    Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat Reply with quote

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@digium.com (jcolp@digium.com)> wrote:
Quote:
Sonny Rajagopalan wrote:
Quote:
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 at g...
Guest





PostPosted: Tue Feb 16, 2016 10:09 pm    Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat Reply with quote

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@11.12.13.14 ([email]sip%3A987654321@11.12.13.14[/email])>;tag=Qb12fSdMpSBV4YJ2e4LGtM3biO.rPtcQ
            SIP from address: sip:987654321@11.12.13.14 ([email]sip%3A987654321@11.12.13.14[/email])
                SIP from address User Part: 987654321
                SIP from address Host Part: 11.12.13.14
            SIP from tag: Qb12fSdMpSBV4YJ2e4LGtM3biO.rPtcQ
        To: <sip:987654321@11.12.13.14 ([email]sip%3A987654321@11.12.13.14[/email])>
            SIP to address: sip:987654321@11.12.13.14 ([email]sip%3A987654321@11.12.13.14[/email])
                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@192.168.1.16:54402;transport=TCP;ob>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000078230be6>"
            Contact URI: sip:987654321@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@gmail.com (sonny.rajagopalan@gmail.com)> wrote:
Quote:
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@digium.com (jcolp@digium.com)> wrote:
Quote:
Sonny Rajagopalan wrote:
Quote:
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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jcolp at digium.com
Guest





PostPosted: Wed Feb 17, 2016 6:36 am    Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat Reply with quote

Sonny Rajagopalan wrote:
Quote:
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


--
_____________________________________________________________________
-- 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 at g...
Guest





PostPosted: Wed Feb 17, 2016 7:56 am    Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat Reply with quote

I receive a TCP ack back from that port (5060; owned by Asterisk) --confirmed by wireshark on the Asterisk server.

What else should I be looking for? This is on a machine on AWS that was running a UDP based Asterisk  fine (I did not make ANY other change other than changing protocol=tcp). I also tried a fresh build with protocol=tcp. Did not work.


On Wed, Feb 17, 2016 at 6:35 AM, Joshua Colp <jcolp@digium.com (jcolp@digium.com)> wrote:
Quote:
Sonny Rajagopalan wrote:
Quote:
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


--
_____________________________________________________________________
-- 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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jcolp at digium.com
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PostPosted: Wed Feb 17, 2016 7:58 am    Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat Reply with quote

Sonny Rajagopalan wrote:
Quote:
I receive a TCP ack back from that port (5060; owned by Asterisk)
--confirmed by wireshark on the Asterisk server.

That's from Wireshark, but what is Asterisk seeing? If Asterisk doesn't
show the connection or the traffic then something else is up (firewall,
etc). Try to isolate things further, start from Asterisk itself.

--
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


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sonny.rajagopalan at g...
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PostPosted: Wed Feb 17, 2016 8:03 am    Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat Reply with quote

Is there a specific place where I can set logger to log incoming TCP segments from L4?

$ netstat -tulpn | grep asterisk | grep LISTEN:

tcp        0      0 0.0.0.0:8088            0.0.0.0:*               LISTEN      10313/asterisk  
tcp        0      0 0.0.0.0:5060            0.0.0.0:*               LISTEN      10313/asterisk  
tcp        0      0 0.0.0.0:2000            0.0.0.0:*               LISTEN      10313/asterisk  




On Wed, Feb 17, 2016 at 7:57 AM, Joshua Colp <jcolp@digium.com (jcolp@digium.com)> wrote:
Quote:
Sonny Rajagopalan wrote:
Quote:
I receive a TCP ack back from that port (5060; owned by Asterisk)
--confirmed by wireshark on the Asterisk server.

That's from Wireshark, but what is Asterisk seeing? If Asterisk doesn't show the connection or the traffic then something else is up (firewall, etc). Try to isolate things further, start from Asterisk itself.

--
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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jcolp at digium.com
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PostPosted: Wed Feb 17, 2016 8:16 am    Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat Reply with quote

Sonny Rajagopalan wrote:
Quote:
Is there a specific place where I can set logger to log incoming TCP
segments from L4?

$ netstat -tulpn | grep asterisk | grep LISTEN:

tcp 0 0 0.0.0.0:8088 <http://0.0.0.0:8088>
0.0.0.0:* LISTEN 10313/asterisk
tcp 0 0 0.0.0.0:5060 <http://0.0.0.0:5060>
0.0.0.0:* LISTEN 10313/asterisk
tcp 0 0 0.0.0.0:2000 <http://0.0.0.0:2000>
0.0.0.0:* LISTEN 10313/asterisk

"pjsip set logger on" will output all incoming and outgoing SIP traffic
for all transports in PJSIP, and when a connection is established it is
logged to the console.

--
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


--
_____________________________________________________________________
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sonny.rajagopalan at g...
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PostPosted: Wed Feb 17, 2016 8:21 am    Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat Reply with quote

Sorry, I was not being very clear, Joshua, and thanks for your patience with this issue.

I had set pjsip set logger on and core set debug 99. See absolutely zilch on asterisk CLI. Or in /var/log/asterisk/messages. If the messages are not reaching Asterisk, what could be the issue? I am a little perplexed as to why Asterisk wouldn't consume those TCP segments; the port is owned by Asterisk.


On Wed, Feb 17, 2016 at 8:15 AM, Joshua Colp <jcolp@digium.com (jcolp@digium.com)> wrote:
Quote:
Sonny Rajagopalan wrote:
Quote:
Is there a specific place where I can set logger to log incoming TCP
segments from L4?

$ netstat -tulpn | grep asterisk | grep LISTEN:

tcp        0      0 0.0.0.0:8088 <http://0.0.0.0:8088>
  0.0.0.0:*               LISTEN      10313/asterisk
tcp        0      0 0.0.0.0:5060 <http://0.0.0.0:5060>
  0.0.0.0:*               LISTEN      10313/asterisk
tcp        0      0 0.0.0.0:2000 <http://0.0.0.0:2000>
  0.0.0.0:*               LISTEN      10313/asterisk

"pjsip set logger on" will output all incoming and outgoing SIP traffic for all transports in PJSIP, and when a connection is established it is logged to the console.

--
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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jcolp at digium.com
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PostPosted: Wed Feb 17, 2016 8:24 am    Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat Reply with quote

Sonny Rajagopalan wrote:
Quote:
Sorry, I was not being very clear, Joshua, and thanks for your patience
with this issue.

I had set pjsip set logger on and core set debug 99. See absolutely
zilch on asterisk CLI. Or in /var/log/asterisk/messages. If the messages
are not reaching Asterisk, what could be the issue? I am a little
perplexed as to why Asterisk wouldn't consume those TCP segments; the
port is owned by Asterisk.

Then it's likely something outside the scope of Asterisk, if the
connection (and messages) don't even seem to be reaching Asterisk at
all. You could try to just telnet to it and see if you get a connection
message on the Asterisk CLI. Do it from the machine itself and then
outside. If it works from the machine itself but not outside, then
you've narrowed it down more.

--
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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