Commit Graph

13 Commits

Author SHA1 Message Date
Matthew Jordan e31bd332b8 Update config framework/sorcery with types/options without documentation
There are times when a configuration option should not have documentation.

1. Some options are registered with a particular object merely as a warning to
   users. These options aren't even really 'deprecated' - which has its own
   separate API call - they are actually provided by a different configuration
   file. The options are merely registered so that the user gets a warning that
   a different configuration file provides the item.

2. Some object types - most notably some used by modules that use sorcery - are
   completely internal and should never be shown to the user.

3. Sorcery itself has several 'hidden' fields that should never be shown to a
   user.

This patch updates the configuration framework and sorcery with additional API
calls that allow a module to register types as internal and options as not
requiring documentation. This bypasses the XML documentation checking.

This patch also re-enables the strict XML documentation checking in trunk, as
well as updates some documentation that was missing.

Review: https://reviewboard.asterisk.org/r/2785/

(closes issue ASTERISK-22359)
Reported by: Matt Jordan

(closes issue ASTERISK-22112)
Reported by: Rusty Newton



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23 15:21:40 +00:00
Kevin Harwell aefebebd37 res_sip_dtmf_info: Support sending of 'raw' DTMF
Added the ability to handle 'raw' DTMF within the body of an INFO message.
Also made it so values 10-16 are mapped to valid DTMF values.

(closes issue ASTERISK-22144)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2776/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 22:09:16 +00:00
Kinsey Moore 6fa4e8e3ab Add missing configOption close tags
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397483 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 21:39:10 +00:00
Rusty Newton 9520df86fd Fix missing xml doc configOption 'type' for for both 'system' and 'global' configObjects
(issue ASTERISK-22344)
(closes issue ASTERISK-22344)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 21:21:25 +00:00
Mark Michelson 86741bdf46 Clarify documentation for the "identify_by" option for SIP endpoints.
This also removes documentation for the options that no longer exist.

(closes issue ASTERISK-22306)
reported by Rusty Newton


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-20 15:39:38 +00:00
Mark Michelson 14ba1751f6 Add note to transport configuration that a restart is required to change transports.
(closes issue ASTERISK-22094)
reported by Rusty Newton



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-20 14:43:56 +00:00
Rusty Newton 168c679e6c xml doc changes for 'aor' config object and a few of its options
Added or modified text in the xml doc for the 'aor' config object to address a few issues:
* help for the 'mailboxes' option didn't make it clear how the "list" should be formatted.
* AoR object's involvement in inbound registration wasn't mentioned.
* help for the 'contact' option didn't describe how to specify multiple contacts.
* help for the 'max_contacts' option didn't tell whether it limited the amount of contacts defined through static configuration.

(issue ASTERISK-22118)
(closes issue ASTERISK-22118)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396902 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-17 18:13:11 +00:00
Rusty Newton 6b1f91b6fc 'domain_alias' config object XML help doesn't make it clear that the name used for the object is the domain alias
(issue ASTERISK-22114)
(closes issue ASTERISK-22114)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-17 17:47:34 +00:00
Rusty Newton fdfe1ea82e xml doc changes for clarity - 'auth' config object and auth's 'auth_type' config option
(issue ASTERISK-22108)
(closes issue ASTERISK-22108)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-17 17:40:34 +00:00
Rusty Newton 115c078e9d xml doc change for transport config object - remove non-applicable warning and add text regarding Asterisk restart
(closes issue ASTERISK-22105)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396899 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-17 17:27:40 +00:00
Kinsey Moore dfa5be76d4 Expose res_pjsip threadpool options
Expose initial size, automatic increment, maximum size, and idle
timeout as configurable parameters for the res_pjsip thread pool.

Review: https://reviewboard.asterisk.org/r/2704/
(closes issue ASTERISK-22143)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-06 13:08:13 +00:00
Kinsey Moore 41cd06e03f Add CLI/AMI commands to force chan_pjsip actions
For chan_pjsip, this introduces CLI/AMI remote unregistration commands,
reworks CLI syntax for sending NOTIFYs, adds AMI qualification support,
and adds documentation for PJSIPNotify.

This also fixes two refcounting bugs in the outbound registration code.

Review: https://reviewboard.asterisk.org/r/2695/
(closes issue ASTERISK-21939)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-02 12:40:03 +00:00
Mark Michelson 735b30ad71 The large GULP->PJSIP renaming effort.
The general gist is to have a clear boundary between old SIP stuff
and new SIP stuff by having the word "SIP" for old stuff and "PJSIP"
for new stuff. Here's a brief rundown of the changes:

* The word "Gulp" in dialstrings, functions, and CLI commands is now
  "PJSIP"
* chan_gulp.c is now chan_pjsip.c
* Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*"
* All files that were "res_sip*" are now "res_pjsip*"
* The "res_sip" directory is now "res_pjsip"
* Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*"
* The configuration file is now "pjsip.conf" instead of "res_sip.conf"
* The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP"
* CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as
the starting word instead of "sip"



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-30 18:14:50 +00:00