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sonny.rajagopalan at g... Guest
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Posted: Mon Feb 15, 2016 1:37 pm Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat |
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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 Guest
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Posted: Mon Feb 15, 2016 1:41 pm Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat |
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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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sonny.rajagopalan at g... Guest
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Posted: Mon Feb 15, 2016 2:23 pm Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat |
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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
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Posted: Mon Feb 15, 2016 2:54 pm Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat |
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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
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Posted: Mon Feb 15, 2016 5:30 pm Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat |
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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
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Posted: Mon Feb 15, 2016 5:32 pm Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat |
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Sonny Rajagopalan wrote:
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
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Posted: Mon Feb 15, 2016 6:02 pm Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat |
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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:
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
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Posted: Tue Feb 16, 2016 10:09 pm Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat |
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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@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:
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
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Posted: Wed Feb 17, 2016 6:36 am Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat |
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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
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Posted: Wed Feb 17, 2016 7:56 am Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat |
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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
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
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Posted: Wed Feb 17, 2016 7:58 am Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat |
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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
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
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Posted: Wed Feb 17, 2016 8:03 am Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat |
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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
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jcolp at digium.com Guest
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Posted: Wed Feb 17, 2016 8:16 am Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat |
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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... Guest
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Posted: Wed Feb 17, 2016 8:21 am Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat |
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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:
"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
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
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Posted: Wed Feb 17, 2016 8:24 am Post subject: [asterisk-users] Asterisk 13.6.0/The simplest TCP configurat |
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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
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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