- Feb 15, 2015
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Joshua Colp authored
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 ........ Merged revisions 431860 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Feb 10, 2015
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George Joseph authored
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/ ........ Merged revisions 431643 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jan 30, 2015
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Ashley Sanders authored
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. In this version, [servername] is uncommented by default. ASTERISK-24316 #close Reported By: Andrew Nagy Review: https://reviewboard.asterisk.org/r/4374/ ........ Merged revisions 431471 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jan 20, 2015
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Richard Mudgett authored
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/ ........ Merged revisions 430817 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jan 16, 2015
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Mark Michelson authored
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 ........ Merged revisions 430709 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jan 12, 2015
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Matthew Jordan authored
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) ........ Merged revisions 430526 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Jan 09, 2015
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Kinsey Moore authored
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 ........ Merged revisions 430416 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Dec 24, 2014
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Kevin Harwell authored
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 ........ Merged revisions 430127 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430128 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Dec 22, 2014
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Matthew Jordan authored
This patch adds the ability to pass options and a command to MixMontor when recording a conference using ConfBridge. New options are - * record_options: Options to MixMontor, eg: m(), W() etc. * record_command: The command to execute when recording is over. * record_file_timestamp: Append the start time to the file name. These options can also be used with the CONFBRIDGE function, e.g., Set(CONFBRIDGE(bridge,record_command)=/path/to/command ^{MIXMONITOR_FILENAME})) Review: https://reviewboard.asterisk.org/r/4023 ASTERISK-24351 #close Reported by: Gareth Palmer patches: record_command-428838.patch uploaded by Gareth Palmer (License 5169) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Dec 15, 2014
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George Joseph authored
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/ ........ Merged revisions 429592 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429593 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Dec 01, 2014
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Matthew Jordan authored
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 ........ Merged revisions 428687 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Nov 21, 2014
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Olle Johansson authored
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Nov 19, 2014
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Joshua Colp authored
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/ ........ Merged revisions 428222 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Nov 17, 2014
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Mark Michelson authored
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 ........ Merged revisions 428145 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Nov 15, 2014
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Joshua Colp authored
Configuration of DTLS in the general section will be applied to any users or peers. If configuration exists at their level it overrides the general section values. ASTERISK-24128 #close Reported by: Michael K. patches: dtls_default_settings.patch submitted by Michael K. (license 6621) Review: https://reviewboard.asterisk.org/r/3867/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Nov 14, 2014
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Jonathan Rose authored
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 ........ Merged revisions 427902 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Nov 05, 2014
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Mark Michelson authored
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. ........ Merged revisions 427334 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Nov 04, 2014
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Joshua Colp authored
Virtual line support establishes a relationship between messages related to an outbound registration and a local endpoint. This is accomplished by attaching a parameter to the Contact of the outbound registration and looking for it on any received requests. If the parameter exists and can be matched to an outbound registration the configured endpoint is associated with the request. Review: https://reviewboard.asterisk.org/r/2964/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Nov 03, 2014
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Richard Mudgett authored
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 ........ Merged revisions 427130 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427137 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Joshua Colp authored
This change adds an option, moh_passthrough, that when enabled will pass hold and unhold requests through using a SIP re-invite. When placing on hold a re-invite with sendonly will be sent and when taking off hold a re-invite with sendrecv will be sent. This allows remote servers to handle the musiconhold instead of the local Asterisk instance being responsible. Review: https://reviewboard.asterisk.org/r/4103/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Oct 31, 2014
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Scott Griepentrog authored
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 ........ Merged revisions 426930 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Oct 28, 2014
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Malcolm Davenport authored
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Malcolm Davenport authored
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Oct 17, 2014
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Jonathan Rose authored
AST-1432 #close Reported by: John Bigelow ........ Merged revisions 425905 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Joshua Colp authored
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Oct 09, 2014
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George Joseph authored
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 ........ Merged revisions 425008 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425009 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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George Joseph authored
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 ........ Merged revisions 424964 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424965 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Oct 03, 2014
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Kinsey Moore authored
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 ........ Merged revisions 424427 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Oct 02, 2014
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Richard Mudgett authored
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 ........ Merged revisions 424394 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424395 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Scott Griepentrog authored
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 ........ Merged revisions 424338 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424339 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Oct 01, 2014
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Joshua Colp authored
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 ........ Merged revisions 424291 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Kinsey Moore authored
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 ........ Merged revisions 424266 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Sep 25, 2014
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Walter Doekes authored
The new option 'preferchannelclass' is added to musiconhold.conf. If yes (the default) the CHANNEL(musicclass) is preferred when choosing the hold music. If it is no, the class suggested by the application that calls the MoH (e.g. the Queue() app) gets preferred (new behaviour). This way you set a different hold-music from the Queue-music by setting both the CHANNEL(musicclass) and the queue-context musicclass. ASTERISK-24276 #close Reported by: Kristian Høgh Patches: app_override_channel_moh.patch uploaded by Kristian Høgh (License #6639) Review: https://reviewboard.asterisk.org/r/4010/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Sep 14, 2014
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Walter Doekes authored
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 ........ Merged revisions 423069 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Walter Doekes authored
- adds sort=randstart (next to sort=, sort=random, sort=alpha) - combines duplicate moh option parsing code into a single function - adds deprecationwarnings for application=r to sort randomly - adds deprecationwarnings for random=yes to sort randomly - removes invisible code that was supposed to stay until 1.8 The sort=randstart works like sort=alpha, except we start at a random position. Review: https://reviewboard.asterisk.org/r/3991/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423065 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Sep 08, 2014
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Mark Michelson authored
........ Merged revisions 422855 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Mark Michelson authored
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. ........ Merged revisions 422853 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422854 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Aug 28, 2014
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Paul Belanger authored
Users now have the ability to bind the rtpengine instance to a specific IP address. For example, you want chan_sip (call control) on eth0 but rtp (media) on eth1. ASTERISK-24280 #close Reported by: Paul Belanger Tested by: Paul Belanger Review: https://reviewboard.asterisk.org/r/3952/ Patches: rtpengine.diff uploaded by Paul Belanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Aug 11, 2014
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Matthew Jordan authored
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) ........ Merged revisions 420624 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Aug 08, 2014
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Jason Parker authored
ASTERISK-24045 Reported by: Jacob Barber Review: https://reviewboard.asterisk.org/r/3833/ ........ Merged revisions 420577 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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