asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	9c05ddbddd	pjproject_bundled: Update for pjproject 2.6 * Removed all 2.5.5 functional patches. * Updated usages of pj_release_pool to be "safe". * Updated configure options to disable webrtc. * Updated config_site.h to disable webrtc in pjmedia. * Added Richard Mudgett's recent resolver patches. Change-Id: Ib400cc4dfca68b3d07ce14d314e829bfddc252c7	9 years ago
George Joseph	456bc3c704	debug_utilities: Add ast_logescalator The escalator works by creating a set of startup commands in cli.conf that set up logger channels and issue the debug commands for the subsystems specified. If asterisk is running when it is executed, the same commands will be issued to the running instance. The original cli.conf is saved before any changes are made and can be restored by executing '$prog --reset'. The log output will be stored in... $astlogdir/message.$uniqueid $astlogdir/debug.$uniqueid $astlogdir/dtmf.$uniqueid $astlogdir/fax.$uniqueid $astlogdir/security.$uniqueid $astlogdir/pjsip_history.$uniqueid $astlogdir/sip_history.$uniqueid Some minor tweaks were made to chan_sip, and res_pjsip_history so their history output could be send to a log channel as packets are captured. A minor tweak was also made to manager so events are output to verbose when "manager set debug on" is issued. Change-Id: I799f8e5013b86dc5282961b27383d134bf09e543	9 years ago
Matt Jordan	dde7f3c1c4	res_pjsip_history: Add a module that provides PJSIP history for debugging This patch adds a new module, res_pjsip_history, that provides a slightly better way of debugging SIP message traffic on a busy Asterisk system. The existing mechanisms all rely on passively dumping a SIP message to the CLI. While this is perfectly fine for logging purposes and well controlled environments, on many installations, the amount of SIP messages Asterisk receives will quickly swamp the CLI. This makes it difficult to view/capture those messages that you want to diagnose in real time. This patch provides another way of handling this. When enabled, the module will store SIP message traffic in memory. This traffic can then be queried at leisure. In order to make the querying useful, a CLI command has been implemented, 'pjsip show history', that supports a basic expression syntax similar to SQL or other query languages. A small number of useful fields have been added in this initial patch; additional fields can easily be added in later improvements. Those fields are: - number: The entry index in the history - timestamp: The time the message was recieved - addr: The source/destination address of the message - sip.msg.request.method: The request method - sip.msg.call-id: The Call-ID header Note - this is a resurrection of the module initially proposed on Review Board here: https://reviewboard.asterisk.org/r/4053/ Change-Id: I39bd74ce998e99ad5ebc0aab3e84df3a150f8e36	10 years ago

Author

SHA1

Message

Date

George Joseph

9c05ddbddd

pjproject_bundled: Update for pjproject 2.6

* Removed all 2.5.5 functional patches.
 * Updated usages of pj_release_pool to be "safe".
 * Updated configure options to disable webrtc.
 * Updated config_site.h to disable webrtc in pjmedia.
 * Added Richard Mudgett's recent resolver patches.

Change-Id: Ib400cc4dfca68b3d07ce14d314e829bfddc252c7

George Joseph

456bc3c704

debug_utilities: Add ast_logescalator

The escalator works by creating a set of startup commands in cli.conf
that set up logger channels and issue the debug commands for the
subsystems specified.  If asterisk is running when it is executed,
the same commands will be issued to the running instance.  The original
cli.conf is saved before any changes are made and can be restored by
executing '$prog --reset'.

The log output will be stored in...
$astlogdir/message.$uniqueid
$astlogdir/debug.$uniqueid
$astlogdir/dtmf.$uniqueid
$astlogdir/fax.$uniqueid
$astlogdir/security.$uniqueid
$astlogdir/pjsip_history.$uniqueid
$astlogdir/sip_history.$uniqueid

Some minor tweaks were made to chan_sip, and res_pjsip_history
so their history output could be send to a log channel as packets
are captured.

A minor tweak was also made to manager so events are output to verbose
when "manager set debug on" is issued.

Change-Id: I799f8e5013b86dc5282961b27383d134bf09e543

Matt Jordan

dde7f3c1c4

res_pjsip_history: Add a module that provides PJSIP history for debugging

This patch adds a new module, res_pjsip_history, that provides a slightly
better way of debugging SIP message traffic on a busy Asterisk system. The
existing mechanisms all rely on passively dumping a SIP message to the CLI.
While this is perfectly fine for logging purposes and well controlled
environments, on many installations, the amount of SIP messages Asterisk
receives will quickly swamp the CLI. This makes it difficult to view/capture
those messages that you want to diagnose in real time.

This patch provides another way of handling this. When enabled, the module
will store SIP message traffic in memory. This traffic can then be queried
at leisure.

In order to make the querying useful, a CLI command has been implemented,
'pjsip show history', that supports a basic expression syntax similar to
SQL or other query languages. A small number of useful fields have been
added in this initial patch; additional fields can easily be added in
later improvements. Those fields are:
 - number: The entry index in the history
 - timestamp: The time the message was recieved
 - addr: The source/destination address of the message
 - sip.msg.request.method: The request method
 - sip.msg.call-id: The Call-ID header

Note - this is a resurrection of the module initially proposed on Review Board
here: https://reviewboard.asterisk.org/r/4053/

Change-Id: I39bd74ce998e99ad5ebc0aab3e84df3a150f8e36

3 Commits (69bebd8f1a9a210108a9afe891a5b4b406be3029)