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[asterisk-users] chan_sip.c: Hanging up call


 
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ethy.brito at inexo.co...
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PostPosted: Thu May 28, 2015 11:05 am    Post subject: [asterisk-users] chan_sip.c: Hanging up call Reply with quote

Hi All

I have a few lines like this at asterisk/messages.

[May 25 15:22:42] WARNING[27725] chan_sip.c: Hanging up call
5a2600300339934f704528bb14ed05e9@MyAsterisk:5060 - no reply to our critical
packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

Since we have hundreds of clients with hundreds of simultaneous calls, how is
it possible to know to which customer/IP those calls refer to?

The above literature don't say much to help to narrow down the problem scope.

Cheers

Ethy

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sgriepentrog at digium...
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PostPosted: Thu May 28, 2015 11:16 am    Post subject: [asterisk-users] chan_sip.c: Hanging up call Reply with quote

The string "5a2600300339934f704528bb14ed05e9@MyAsterisk:5060" is the unique identifier for the call in SIP known as the Call-ID.  If you have a packet capture of the port 5060 SIP traffic, that identifier will be in each SIP message related to the call, which also includes the full from and to details.


As an alternative to running a separate packet capture, you can enable SIP message logging in Asterisk, which puts the full SIP message into the same log file.  Be aware however that this can fill your hard drive quite rapidly, as well as put additional load on the disk storage system.


On Thu, May 28, 2015 at 11:03 AM, Ethy H. Brito <ethy.brito@inexo.com.br (ethy.brito@inexo.com.br)> wrote:
Quote:

Hi All

I have a few lines like this at asterisk/messages.

[May 25 15:22:42] WARNING[27725] chan_sip.c: Hanging up call
5a2600300339934f704528bb14ed05e9@MyAsterisk:5060 - no reply to our critical
packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

Since we have hundreds of clients with hundreds of simultaneous calls, how is
it possible to know to which customer/IP those calls refer to?

The above literature don't say much to help to narrow down the problem scope.

Cheers

Ethy

--
_____________________________________________________________________
-- 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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Scott Griepentrog
Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
Check us out at: http://digium.com · http://asterisk.org
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ethy.brito at inexo.co...
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PostPosted: Thu May 28, 2015 12:49 pm    Post subject: [asterisk-users] chan_sip.c: Hanging up call Reply with quote

On Thu, 28 May 2015 11:15:45 -0500
Scott Griepentrog <sgriepentrog@digium.com> wrote:

Quote:
The string "5a2600300339934f704528bb14ed05e9@MyAsterisk:5060" is the unique
identifier for the call in SIP known as the Call-ID. If you have a packet
capture of the port 5060 SIP traffic, that identifier will be in each SIP
message related to the call, which also includes the full from and to
details.

That is the problem. Since the message occurs typically about 2~3 times a day (or even less), I will have tons of packets to sniff.

But, I will give it a try.

Quote:

As an alternative to running a separate packet capture, you can enable SIP
message logging in Asterisk, which puts the full SIP message into the same
log file.

You mean "sip set debug on" ?

Quote:
Be aware however that this can fill your hard drive quite
rapidly, as well as put additional load on the disk storage system.

I am pretty aware of that. Learn it the hard way.

Cheers

Ethy


Quote:

On Thu, May 28, 2015 at 11:03 AM, Ethy H. Brito <ethy.brito@inexo.com.br>
wrote:

Quote:

Hi All

I have a few lines like this at asterisk/messages.

[May 25 15:22:42] WARNING[27725] chan_sip.c: Hanging up call
5a2600300339934f704528bb14ed05e9@MyAsterisk:5060 - no reply to our
critical
packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
).

Since we have hundreds of clients with hundreds of simultaneous calls, how
is
it possible to know to which customer/IP those calls refer to?

The above literature don't say much to help to narrow down the problem
scope.

Cheers

Ethy

--
_____________________________________________________________________
-- 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




--
[image: Digium logo]
Scott Griepentrog
Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
Check us out at: http://digium.com · http://asterisk.org


--

Ethy H. Brito /"\
InterNexo Ltda. \ / CAMPANHA DA FITA ASCII - CONTRA MAIL HTML
+55 (12) 3797-6860 X ASCII RIBBON CAMPAIGN - AGAINST HTML MAIL
S.J.Campos - Brasil / \

PGP key: http://www.inexo.com.br/~ethy/0xC3F222A0.asc

--
_____________________________________________________________________
-- 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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sgriepentrog at digium...
Guest





PostPosted: Thu May 28, 2015 1:31 pm    Post subject: [asterisk-users] chan_sip.c: Hanging up call Reply with quote

Quote:
 You mean "sip set debug on" ?
​Yes, that's correct for chan_sip.  Sorry, I was vague -- there is now a different command for chan_pjsip​, didn't know which you were using.




On Thu, May 28, 2015 at 12:49 PM, Ethy H. Brito <ethy.brito@inexo.com.br (ethy.brito@inexo.com.br)> wrote:
Quote:
On Thu, 28 May 2015 11:15:45 -0500
Scott Griepentrog <sgriepentrog@digium.com (sgriepentrog@digium.com)> wrote:

Quote:
The string "5a2600300339934f704528bb14ed05e9@MyAsterisk:5060" is the unique
identifier for the call in SIP known as the Call-ID.  If you have a packet
capture of the port 5060 SIP traffic, that identifier will be in each SIP
message related to the call, which also includes the full from and to
details.

That is the problem. Since the message occurs typically about 2~3 times a day (or even less), I will have tons of packets to sniff.

But, I will give it a try.

Quote:

As an alternative to running a separate packet capture, you can enable SIP
message logging in Asterisk, which puts the full SIP message into the same
log file.

You mean "sip set debug on" ?

Quote:
Be aware however that this can fill your hard drive quite
rapidly, as well as put additional load on the disk storage system.

I am pretty aware of that. Learn it the hard way.

Cheers

Ethy


Quote:

On Thu, May 28, 2015 at 11:03 AM, Ethy H. Brito <ethy.brito@inexo.com.br (ethy.brito@inexo.com.br)>
wrote:

Quote:

Hi All

I have a few lines like this at asterisk/messages.

[May 25 15:22:42] WARNING[27725] chan_sip.c: Hanging up call
5a2600300339934f704528bb14ed05e9@MyAsterisk:5060 - no reply to our
critical
packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
).

Since we have hundreds of clients with hundreds of simultaneous calls, how
is
it possible to know to which customer/IP those calls refer to?

The above literature don't say much to help to narrow down the problem
scope.

Cheers

Ethy

--
_____________________________________________________________________
-- 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




--
[image: Digium logo]
Scott Griepentrog
Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
Check us out at: http://digium.com · http://asterisk.org




--

Ethy H. Brito         /"\
InterNexo Ltda.       \ /  CAMPANHA DA FITA ASCII - CONTRA MAIL HTML
+55 (12) 3797-6860     X   ASCII RIBBON CAMPAIGN - AGAINST HTML MAIL
S.J.Campos - Brasil   / \

PGP key: http://www.inexo.com.br/~ethy/0xC3F222A0.asc

--
_____________________________________________________________________
-- 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





--
Scott Griepentrog
Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
Check us out at: http://digium.com · http://asterisk.org
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