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2015-04-10chan_sip: make progressinband default to noKevin Harwell
After the "progressinband" value setting of "never" was updated to never send a 183 this separated its use from the "no" value. Since "never" was the default, but most users probably expect "no" this patch updates the default for the "progressinband" setting to "no." ASTERISK-24835 #close Reported by: Andrew Nagy Review: https://reviewboard.asterisk.org/r/4606/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434654 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-27SAC: Add a few basic queuesJonathan Rose
Review: https://reviewboard.asterisk.org/r/4503/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433658 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-27SAC: Add conferencing extensions and configurationJonathan Rose
Review: https://reviewboard.asterisk.org/r/4504/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-27configs/basic-pbx - Super Awesome Company example configs Phase 1, Patch 2Rusty Newton
Example configuration files for a "basic PBX" deployment for the fictitious Super Awesome Company. Details at https://reviewboard.asterisk.org/r/4488/ and https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company Patch 4488 includes all functionality needed for SAC's outside connectivity and some externally accessed features, as well as outbound dialing. Reported by: Malcolm Davenport Tested by: Rusty Newton Review: https://reviewboard.asterisk.org/r/4488/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-24chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" ↵Richard Mudgett
messages. Incoming PJSIP call legs that have not been answered yet send unnecessary "180 Ringing" or "183 Progress" messages every time a connected line update happens. If the outgoing channel is also PJSIP then the incoming channel will always send a "180 Ringing" or "183 Progress" message when the outgoing channel sends the INVITE. Consequences of these unnecessary messages: * The caller can start hearing ringback before the far end even gets the call. * Many phones tend to grab the first connected line information and refuse to update the display if it changes. The first information is not likely to be correct if the call goes to an endpoint not under the control of the first Asterisk box. When connected line first went into Asterisk in v1.8, chan_sip received an undocumented option "rpid_immediate" that defaults to disabled. When enabled, the option immediately passes connected line update information to the caller in "180 Ringing" or "183 Progress" messages as described above. * Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or "183 Progress" messages. The default is "no" to disable sending the unnecessary messages. ASTERISK-24781 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4473/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-17res_pjsip: Allow configuration of endpoint identifier query orderKevin Harwell
This patch fixes previously reverted code that caused binary incompatibility problems with some modules. And like the original patch it makes sure that no matter what order the endpoint identifier modules were loaded, priority is given based on the ones specified in the new global 'endpoint_identifier_order' option. ASTERISK-24840 Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4489/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-14apps/app_amd: Document maximum_word_length option; fix AMDCAUSE documentationMatthew Jordan
This patch corrects the documentation for the AMD application. Specifically: * It documents the maximum_word_length option, which limits the maximum allowed length of a single utterance. * It clarifies the AMDCAUSE values MAXWORDS and MAXWORDLENGTH. MAXWORDLENGTH was documented as MAXWORDS, while MAXWORDS was undocumented. Thanks to the issue reporter, Frank DiGennaro, for pointing out the issues. ASTERISK-19470 #close Reported by: Frank DiGennaro ........ Merged revisions 432918 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432920 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-13Revert - res_pjsip: Allow configuration of endpoint identifier query orderKevin Harwell
Due to a break in binary compatibility with some other modules these changes are being reverted until the issue can be resolved. ASTERISK-24840 Reported by: Mark Michelson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-09res_pjsip: Allow configuration of endpoint identifier query orderKevin Harwell
It's possible to have a scenario that will create a conflict between endpoint identifiers. For instance an incoming call could be identified by two different endpoint identifiers and the one chosen depended upon which identifier module loaded first. This of course causes problems when, for example, the incoming call is expected to be identified by username, but instead is identified by ip. This patch adds a new 'global' option to res_pjsip called 'endpoint_identifier_order'. It is a comma separated list of endpoint identifier names that specifies the order by which identifiers are processed and checked. ASTERISK-24840 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4455/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-25configs/basic-pbx - Super Awesome Company example configs Phase 1, Patch 1Rusty Newton
Example configuration files for a "basic PBX" deployment for the fictitious Super Awesome Company. Details at https://reviewboard.asterisk.org/r/4379/ and https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company Reported by: Malcolm Davenport Tested by: Rusty Newton Review: https://reviewboard.asterisk.org/r/4379/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432301 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-15pjsip: Remove "contact" type from pjsip.conf.sampleJoshua Colp
The "contact" object is not meant to be configured from the pjsip.conf configuration file. It is meant to be created as a result of a registration and stored elsewhere. ASTERISK-24085 #close Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431860 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-10res_pjsip_config_wizard: Add ability to auto-create hints.George Joseph
Looking at the Super Awesome Company sample reminded me that creating hints is just plain gruntwork. So you can now have the pjsip conifg wizard auto-create them for you. Specifying 'hint_exten' in the wizard will create 'exten => <hint_exten>,hint/PJSIP/<wizard_id>' in whatever is specified for 'hint_context'. Specifying 'hint_application' in the wizard will create 'exten => <hint_exten>,1,<hint_application>' in whatever is specified for 'hint_context'. The default for 'hint_context' is the endpoint's context. There's no default for 'hint_application'. If not specified, no app is added. There's no default for 'hint_exten'. If not specified, neither the hint itself nor the application will be created. Some may think this is the slippery slope to users.conf but hints are a basic necessity for phones unlike voicemail, manager, etc that users.conf creates. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4383/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-30HTTP: For httpd server, need option to define server name for security purposesAshley Sanders
Added a new config property [servername] to the http.conf file; updated the http server to use the new property when sending responses, for showing http status through the CLI and when reporting status through the 'httpstatus' webpage. ASTERISK-24316 #close Reported By: Andrew Nagy Review: https://reviewboard.asterisk.org/r/4374/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-20CHANNEL(peer), chan_iax2, res_fax, SNMP agent: Fix deadlock from reaching ↵Richard Mudgett
across a bridge. Calling ast_channel_bridge_peer() cannot be done while holding any channel locks. The reported issue hit the deadlock in chan_iax2, but an audit of the ast_channel_bridge_peer() calls found three more locations where the same deadlock can occur. * Made CHANNEL(peer), res_fax, and the SNMP agent not call ast_channel_bridge_peer() with any channel locked. For CHANNEL(peer) I had to rework the logic to not hold the channel lock. * Made chan_iax2 no longer call ast_channel_bridge_peer(). It was done for legacy reasons that no longer apply. * Removed the iax.conf forcejitterbuffer option. It is now always enabled when the jitterbuffer option is enabled. If you put a jitter buffer on a channel it will be on the channel. ASTERISK-24600 #close Reported by: Jeff Collell Review: https://reviewboard.asterisk.org/r/4342/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-16Add support for the ca_list_path option for PJSIP transports.Mark Michelson
This allows for a path to be specified that has a collection of CA certificates in it. ASTERISK-24575 #close Reported by cloos Patches: pj-ca-path-trunk.diff uploaded by cloos (License #5956) Review: https://reviewboard.asterisk.org/r/4344 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-12configs/samples/features.conf.sample: Document attended transfer DTMF optionsMatthew Jordan
The sample config was missing the configuration options for DTMF attended transfer completion scenarios. The configuration options 'atxferabort', 'atxfercomplete', 'atxferthreeway', and 'atxferswap' are now documented in the appropriate configuration file. ASTERISK-24678 #close Reported by: Niklas Larsson patches: features.conf.sample.diff uploaded by Niklas Larsson (License 5068) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09res_fax: Add T.38 negotiation timeout optionKinsey Moore
This change makes the T.38 negotiation timeout configurable via 't38timeout' in res_fax.conf or FAXOPT(t38timeout). It was previously hard coded to be 5000 milliseconds. This change also handles T.38 switch failures by aborting the fax since in the case where this can happen, both sides have agreed to switch to T.38 and Asterisk is unable to do so. Review: https://reviewboard.asterisk.org/r/4320/ ........ Merged revisions 430415 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-24app_queue: Update sample conf documenationKevin Harwell
Updated the queues.conf.sample file to explicitly state which channel queue variables are propagated to. ASTERISK-24267 Reported by: Mitch Claborn ........ Merged revisions 430126 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-24res_pjsip_keepalive: Add runtime configurable keepalive module for ↵Matthew Jordan
connection-oriented transports. Note that this is backport from trunk of r425825. This change adds a module which is configurable using the keep_alive_interval setting in the global section that will send a CRLF keep alive to all active connection-oriented transports at the provided interval. This is useful because it can help keep connections open through NATs. This functionality also exists within PJSIP but can not be controlled at runtime and requires recompiling it. Review: https://reviewboard.asterisk.org/r/4084/ ASTERISK-24644 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-15res_pjsip_config_wizard: Allow streamlined config of common pjsip scenariosGeorge Joseph
res_pjsip_config_wizard ------------------ * This is a new module that adds streamlined configuration capability for chan_pjsip. It's targetted at users who have lots of basic configuration scenarios like 'phone' or 'agent' or 'trunk'. Additional information can be found in the sample configuration file at config/samples/pjsip_wizard.conf.sample. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4190/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-01main/stasis: Allow subscriptions to use a threadpool for message deliveryMatthew Jordan
Prior to this patch, all Stasis subscriptions would receive a dedicated thread for servicing published messages. In contrast, prior to r400178 (see review https://reviewboard.asterisk.org/r/2881/), the subscriptions shared a thread pool. It was discovered during some initial work on Stasis that, for a low subscription count with high message throughput, the threadpool was not as performant as simply having a dedicated thread per subscriber. For situations where a subscriber receives a substantial number of messages and is always present, the model of having a dedicated thread per subscriber makes sense. While we still have plenty of subscriptions that would follow this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into the following two categories: * Large number of subscriptions, specifically those tied to endpoints/peers. * Low number of messages. Some subscriptions exist specifically to coordinate a single message - the subscription is created, a message is published, the delivery is synchronized, and the subscription is destroyed. In both of the latter two cases, creating a dedicated thread is wasteful (and in the case of a large number of peers/endpoints, harmful). In those cases, having shared delivery threads is far more performant. This patch adds the ability of a subscriber to Stasis to choose whether or not their messages are dispatched on a dedicated thread or on a threadpool. The threadpool is configurable through stasis.conf. Review: https://reviewboard.asterisk.org/r/4193 ASTERISK-24533 #close Reported by: xrobau Tested by: xrobau ........ Merged revisions 428681 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-19res_pjsip_sdp_rtp: Add support for optimistic SRTP.Joshua Colp
Optimistic SRTP is the ability to enable SRTP but not have it be a fatal requirement. If SRTP can be used it will be, if not it won't be. This gives you a better chance of using it without having your sessions fail when it can't be. Encrypt all the things! Review: https://reviewboard.asterisk.org/r/3992/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-17Allow for transferer to retry when dialing an invalid extension.Mark Michelson
This allows for a configurable number of attempts for a transferer to dial an extension to transfer the call to. For Asterisk 13, the default values are such that upgrading between versions will not cause a behaivour change. For trunk, though, the defaults will be changed to be more user-friendly. Review: https://reviewboard.asterisk.org/r/4167 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-14Documentation: Revise explanation of cdr.conf option 'Unanswered'Jonathan Rose
ASTERISK-24279 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4109/ ........ Merged revisions 427901 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427902 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-05Make the disable_tcp_switch PJSIP system object enabled by default.Mark Michelson
Testing has shown repeatedly that PJSIP's default behavior of switching automatically to TCP for large messages can cause issues. The most common issues are that devices that we are communicating with do not handle the switch to TCP gracefully, thus causing situations such as broken calls or broken subscriptions. Now, in order to have this behavior happen, you must opt into it. The sample file has been updated to warn that enabling the TCP switch behavior may cause issues for you, so use at your own risk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-03res_pjsip: Add disable_tcp_switch option.Richard Mudgett
When a packet exceeds the MTU, pjproject will switch from UDP to TCP. In some circumstances (on some networks), this can cause some issues with messages not getting sent to the correct destination - and can also cause connections to get dropped due to quirks in pjproject deciding to terminate TCP connections with no messages. While fixing the routing/messaging issues is important, having a configuration option in Asterisk that tells pjproject to not switch over to TCP would be useful. That way, if some glitch is discovered on some other network/site, we can at least disable the behavior until a fix is put into place. AFS-197 #close Review: https://reviewboard.asterisk.org/r/4137/ ........ Merged revisions 427129 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-31pjsip: clarify tls cert and key file usageScott Griepentrog
A question arose as to whether a .pem file could be provided in place of the .crt and .key files in a PJSIP TLS configuration. I tested this and discovered that although a cert will be read from the pem file, a key will not, and thus the priv_key_file entry is still required. This update to the fine documentation clarifies the option usage. AST-1448 #close Review: https://reviewboard.asterisk.org/r/4129/ Reported by: John Bigelow ........ Merged revisions 426928 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-28ASTERISK-23512, correct inaccurate comment in manager.conf.sampleMalcolm Davenport
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426459 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-28ASTERISK-24419, fix incorrect syntax for setting language in ↵Malcolm Davenport
extensions.conf.sample git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-17Sample Configurations: make 'pjsip reload' reload all reloadable pjsip modulesJonathan Rose
AST-1432 #close Reported by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-09res_pjsip_phoneprov_provider: Provides pjsip integration with res_phoneprovGeorge Joseph
This module allows res_pjsip to integrate with res_phoneprov. It handles the pjsip 'phoneprov' object type. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3976/ ........ Merged revisions 425007 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-09res_phoneprov: Refactor phoneprov to allow pluggable config providersGeorge Joseph
This patch makes res_phoneprov more modular so other modules (like pjsip) can provide configuration information instead of res_phoneprov relying solely on users.conf and sip.conf. To accomplish this a new ast_phoneprov public API is now exposed which allows config providers to register themselves, set defaults (server profile, etc) and add user extensions. * ast_phoneprov_provider_register registers the provider and provides callbacks for loading default settings and loading users. * ast_phoneprov_provider_unregister clears the defaults and users. * ast_phoneprov_add_extension should be called once for each user/extension by the provider's load_users callback to add them. * ast_phoneprov_delete_extension deletes one extension. * ast_phoneprov_delete_extensions deletes all extensions for the provider. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3970/ ........ Merged revisions 424963 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-03PJSIP: Restore functional default for callerid_privacyKinsey Moore
The pjsip config option default fixups from r424263 altered the functional default from "allowed_not_screened" to "allowed". This change restores the functional default value when none is provided. ........ Merged revisions 424426 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-02res_pjsip: Make transport cipher option accept a comma separated list of ↵Richard Mudgett
cipher names. Improvements to the res_pjsip transport cipher option. * Made the cipher option accept a comma separated list of OpenSSL cipher names. Users of realtime will be glad if they have more than one name to list. * Added the CLI command 'pjsip list ciphers' so a user can know what OpenSSL names are available for the cipher option. * Updated the cipher option online XML documentation to specify what is expected for the value. * Updated pjsip.conf.sample to not indicate that ALL is acceptable since ALL does not imply a preference order for the ciphers and PJSIP does not simply pass the string to OpenSSL for interpretation. ASTERISK-24199 #close Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/4018/ ........ Merged revisions 424393 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424394 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-02res_pjsip: document use of rewrite_contact in sample confScott Griepentrog
Without setting rewrite_contact, an invite to an endpoint behind NAT will not reach it - unless the endpoint itself uses STUN or TURN to discover it's public URI. Thus, the use of this should be in the sample documentation. Review: https://reviewboard.asterisk.org/r/4036/ ........ Merged revisions 424337 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-01res_pjsip: Add 'dtls_fingerprint' option to configure DTLS fingerprint hash.Joshua Colp
During the latest update to DTLS-SRTP support the ability to configure the hash used for fingerprints was added. This gave us two supported ones: SHA-1 and SHA-256. The default was accordingly updated to SHA-256. Unfortunately this configuration ability was not exposed within res_pjsip. This change adds a dtls_fingerprint option that controls it. #SIPit31 ........ Merged revisions 424290 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-01PJSIP: Handle defaults properlyKinsey Moore
This updates the code behind PJSIP configuration options with custom handlers to deal with the assigned default values properly where it makes sense and adjusting the default value where it doesn't. Before applying this patch, there were several cases where the default value for an option would prevent that config section from loading properly. Reported by: Thomas Thompson Review: https://reviewboard.asterisk.org/r/4019/ ........ Merged revisions 424263 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-14chan_sip: Clarify that sipdebug=yes cannot be undone by the CLI.Walter Doekes
Document it in sip.conf. ASTERISK-24249 #close Reported by: Avinash Mohod Review: https://reviewboard.asterisk.org/r/3926/ ........ Merged revisions 423066 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423067 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423068 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-08Add note about configuring list_items on a single line.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-08Add sample configuration for resource lists.Mark Michelson
On review /r/3977, it was recommended to note in the sample configuration about the size limitation for resource lists. However, since there was no section in the sample configuration at all for resource list subscriptions, I decided to make a separate commit where I have added the necessary sample configuration as well as the size limitation warning. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422853 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-11app_queue: Add RealTime support for queue rulesMatthew Jordan
This patch gives the optional ability to keep queue rules in RealTime. It is important to note that with this patch: (a) Queue rules in RealTime are only examined on module load/reload (b) Queue rules are loaded both from the queuerules.conf file as well as the RealTime backend To inform app_queue to examine RealTime for queue rules, a new setting has been added to queuerules.conf's general section "realtime_rules". RealTime queue rules will only be used when this setting is set to "yes". The schema for the database table supports a rule_name, time, min_penalty, and max_penalty columns. min_penalty and max_penalty can be relative, if a '-' or '+' literal is provided. Otherwise, the penalties are treated as constants. For example: rule_name, time, min_penalty, max_penalty 'default', '10', '20', '30' 'test2', '20', '30', '55' 'test2', '25', '-11', '+1111' 'test2', '400', '112', '333' 'test3', '0', '4564', '46546' 'test_rule', '40', '15', '50' which would result in : Rule: default - After 10 seconds, adjust QUEUE_MAX_PENALTY to 30 and adjust QUEUE_MIN_PENALTY to 20 Rule: test2 - After 20 seconds, adjust QUEUE_MAX_PENALTY to 55 and adjust QUEUE_MIN_PENALTY to 30 - After 25 seconds, adjust QUEUE_MAX_PENALTY by 1111 and adjust QUEUE_MIN_PENALTY by -11 - After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust QUEUE_MIN_PENALTY to 112 Rule: test3 - After 0 seconds, adjust QUEUE_MAX_PENALTY to 46546 and adjust QUEUE_MIN_PENALTY to 4564 Rule: test_rule - After 40 seconds, adjust QUEUE_MAX_PENALTY to 50 and adjust QUEUE_MIN_PENALTY to 15 If you use RealTime, the queue rules will be always reloaded on a module reload, even if the underlying file did not change. With the option disabled, the rules will only be reloaded if the file was modified. Review: https://reviewboard.asterisk.org/r/3607/ ASTERISK-23823 #close Reported by: Michael K patches: app_queue.c_realtime_trunk.patch uploaded by Michael K (License 6621) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-08app_voicemail: Add the ability to specify multiple email addresses.Jason Parker
ASTERISK-24045 Reported by: Jacob Barber Review: https://reviewboard.asterisk.org/r/3833/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-06Stasis: Allow message types to be blockedKinsey Moore
This introduces stasis.conf and a mechanism to prevent certain message types from being published. Internally, this works by preventing the chosen message types from being created which ensures that those message types can never be published. This patch also adjusts message publishers such that message payloads are not created if the related message type is not available. ASTERISK-23943 #close Review: https://reviewboard.asterisk.org/r/3823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-17configs: Move sample config files into a subdirectory of configsMatthew Jordan
This moves all samples configs from configs/ to configs/samples. This allows for additional sets of sample configuration files to be added in the future. Review: https://reviewboard.asterisk.org/r/3804/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-16res_pjsip: Support setting a default accountcode on endpointsMatthew Jordan
Most channel drivers let you specify a default accountcode to be set on channels associated with a particular peer/endpoint/object. Prior to this patch, chan_pjsip/res_pjsip did not support such a setting. This patch adds a new setting to the res_pjsip endpoint object, 'accountcode'. When a channel is created that is associated with an endpoint with this value set, the channel will automatically have its accountcode property set to the value configured for the endpoint. Review: https://reviewboard.asterisk.org/r/3724/ ASTERISK-24000 #close Reported by: Matt Jordan ........ Merged revisions 418756 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-16cel_pgsql, cdr_pgsql, res_config_pgsql: Add PostgreSQL application_name supportMatthew Jordan
This patch adds support for the PostgreSQL application_name connection setting. When the appropriate PostgreSQL module's configuration is set with an application name, the name will be passed to PostgreSQL on connection and displayed in the database's pg_stat_activity view, as well as in CSV logs. This aids in managing which applications/servers are connected to a PostgreSQL database, as well as tracing the activity of those connections. Review: https://reviewboard.asterisk.org/r/3591 ASTERISK-23737 #close Reported by: Gergely Domodi patches: pgsql_application_name.patch uploaded by Gergely Domodi (License 6610) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-14Actually delete the removed files.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-13Remove files left behind on removal of h323, jingle and jabber.Corey Farrell
This change removes h323.conf.sample, jingle.h, jabber.h left behind by r3698. Review: https://reviewboard.asterisk.org/r/3755/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-04Remove many deprecated modulesMatthew Jordan
Billing records are fair, To get paid is quite bright, You should really use ODBC; Good-bye cdr_sqlite. Microsoft did once push H.323, Hell, we all remember NetMeeting. But try to compile chan_h323 now And you will take quite a beating. The XMPP and SIP war was fierce, And in the distant fray Was birthed res_jabber/chan_jingle; But neither to stay. For everyone did care and chase what Google professed. "Free Internet Calling" was what devotees cried, But Google did change the specs so often That the developers were happy the day chan_gtalk died. And then there was that odd application Dedicated to the Polish tongue. app_saycountpl was subsumed by Say; One could say its bell was rung. To read and parse a file from the dialplan You could (I guess) use an application. app_readfile did fill that purpose, but I think A function is perhaps better in its creation. Barging is rude, I'm not sure why we do it. Inwardly, the caller will probably sigh. But if you really must do it, Don't use app_dahdibarge, use ChanSpy. We all despise the sound of tinny robots It makes our queues so cold. To control such an abomination It's better to not use Wait/SetMusicOnHold. It's often nice to know properties of a channel It makes our calls right We have a nice function called CHANNEL And so SIPCHANINFO is sent off into the night. And now things get odd; Apparently one could delimit with a colon Properties from the SIPPEER function! Commas are in; all others are done. Finally, a word on pipes and commas. We're sorry. We can't say it enough. But those compatibility options in asterisk.conf; To maintain them forever was just too tough. This patch removes: * cdr_sqlite * chan_gtalk * chan_jingle * chan_h323 * res_jabber * app_saycountpl * app_readfile * app_dahdibarge It removes the following applications/functions: * WaitMusicOnHold * SetMusicOnHold * SIPCHANINFO It removes the colon delimiter from the SIPPEER function. Finally, it also removes all compatibility options that were configurable from asterisk.conf, as these all applied to compatibility with Asterisk 1.4 systems. Review: https://reviewboard.asterisk.org/r/3698/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-03chan_dahdi: Add inband_on_setup_ack compatibility option.Richard Mudgett
The new inband_on_setup_ack option causes Asterisk to assume inband audio may be present when a SETUP_ACKNOWLEDGE message is received. Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a dialtone is sent from the network side, progress indicator 8 "Inband info now available" MAY be sent to the CPE if no digits were received with the SETUP. It is thus implied that the ie is mandatory if digits came with the SETUP and dialtone is needed. This option should be enabled, when the network sends dialtone and you want to hear it, but the network doesn't send the progress indicator when needed. NOTE: For Q.SIG setups this option should be enabled when outgoing overlap dialing is also enabled because Q.SIG does not send the progress indicator with the SETUP ACK. The commit -r413714 (AST-1338) which causes this issue was dealing with a SIP-to-ISDN interoperability issue. This commit is a merge of the two patches indicated below. ASTERISK-23897 #close Reported by: Pavel Troller Patches: pri-4.diff (license #6302) patch uploaded by Pavel Troller jira_asterisk_23897_v11.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/3633/ ........ Merged revisions 417956 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 417957 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 417958 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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