174 Commits (1a82f5b6451848dd4bce2e896c2273f7bb954c5b)

Author	SHA1	Message	Date
Matthew Jordan	d79c68d3fb	main/stasis: Allow subscriptions to use a threadpool for message delivery ... 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	11 years ago
Joshua Colp	a0d9eab389	res_pjsip_sdp_rtp: Add support for optimistic SRTP. ... 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	11 years ago
Mark Michelson	a1f1cdbd87	Allow for transferer to retry when dialing an invalid extension. ... 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	11 years ago
Jonathan Rose	8c61b4890b	Documentation: Revise explanation of cdr.conf option 'Unanswered' ... 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	11 years ago
Mark Michelson	27dc7e47d7	Make the disable_tcp_switch PJSIP system object enabled by default. ... 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	11 years ago
Richard Mudgett	b586e7f0b2	res_pjsip: Add disable_tcp_switch option. ... 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	11 years ago
Scott Griepentrog	b3b93a7c15	pjsip: clarify tls cert and key file usage ... 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	11 years ago
Malcolm Davenport	ed07535b1c	ASTERISK-23512, correct inaccurate comment in manager.conf.sample ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426459 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Malcolm Davenport	1ec27418da	ASTERISK-24419, fix incorrect syntax for setting language in extensions.conf.sample ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	8e610ab20e	Sample Configurations: make 'pjsip reload' reload all reloadable pjsip modules ... AST-1432 #close Reported by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	d3342fb98d	res_pjsip_phoneprov_provider: Provides pjsip integration with res_phoneprov ... 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	11 years ago
George Joseph	d1c9621852	res_phoneprov: Refactor phoneprov to allow pluggable config providers ... 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	11 years ago
Kinsey Moore	ef2c567597	PJSIP: Restore functional default for callerid_privacy ... 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	11 years ago
Richard Mudgett	94105b30a6	res_pjsip: Make transport cipher option accept a comma separated list of 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	11 years ago
Scott Griepentrog	1e620fe59e	res_pjsip: document use of rewrite_contact in sample conf ... 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	11 years ago
Joshua Colp	a1763a89a3	res_pjsip: Add 'dtls_fingerprint' option to configure DTLS fingerprint hash. ... 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	11 years ago
Kinsey Moore	e3da76a352	PJSIP: Handle defaults properly ... 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	11 years ago
Walter Doekes	5840c47223	chan_sip: Clarify that sipdebug=yes cannot be undone by the CLI. ... 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	11 years ago
Mark Michelson	048560f330	Add note about configuring list_items on a single line. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	a8ef508f5e	Add sample configuration for resource lists. ... 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	11 years ago
Matthew Jordan	95871451f6	app_queue: Add RealTime support for queue rules ... 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	11 years ago
Jason Parker	2b804661bc	app_voicemail: Add the ability to specify multiple email addresses. ... 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	11 years ago
Kinsey Moore	f1036f40dc	Stasis: Allow message types to be blocked ... 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	11 years ago
Matthew Jordan	fc0fecb476	configs: Move sample config files into a subdirectory of configs ... 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	11 years ago