asterisk

Commit Graph

Author	SHA1	Message	Date
Jason Parker	66f69a53c3	Add Digium phones context to sip_notify sample config. This makes it so that they can be reconfigured remotely. (closes issue ASTERISK-19910) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369818 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f1bb4eea3d	Update membermacro and membergosub documentation in queues.conf.sample. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	e9a0da476a	Add ability to ignore layer 1 alarms for BRI PTMP lines. Several telcos bring the BRI PTMP layer 1 down when the line is idle. When layer 1 goes down, Asterisk cannot make outgoing calls. Incoming calls could fail as well because the alarm processing is handled by a different code path than the Q.931 messages. * Add the layer1_presence configuration option to ignore layer 1 alarms when the telco brings layer 1 down. This option can be configured by span while the similar DAHDI driver teignorered=1 option is system wide. This option unlike layer2_persistence does not require libpri v1.4.13 or newer. Related to JIRA AST-598 JIRA ABE-2845 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	5fcb78840f	Remove a few more files related to chan_usbradio and app_rpt. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	fe1d703804	Set snarkiness = 0 in cdr_adaptive_odbc.conf.sample git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358438 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	cc470930ba	Adding transport=udp to sample sip.conf - Also changes version of Asterisk 1.8 in UPGRADE (issue ASTERISK-19352) Reported by: jamicque Patches: asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	0106636e42	Don't allow trunkfreq to be greater than 1000ms. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	8357dd7a2d	Don't enable sqlite3 CDRs by default in sample configs. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355228 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	7d84a0ed69	Fix reconnecting to pgsql database after connection loss. There can only be one database connection in res_config_pgsql just like res_config_sqlite. If the connection is lost, the connection may not get reestablished to the same database if the res_pgsql.conf and extconfig.conf files are inconsistent. * Made only use the configured database from res_pgsql.conf. * Fixed potential buffer overwrite of last[] in config_pgsql(). (closes issue ASTERISK-16982) Reported by: german aracil boned Review: https://reviewboard.asterisk.org/r/1731/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	47b3aa6362	Fix TLS port binding behavior as well as reload behavior: * Removes references to tlsbindport from http.conf.sample and manager.conf.sample * Properly bind to port specified in tlsbindaddr, using the default port if specified. * On a reload, properly close socket if the service has been disabled. A note has been added to UPGRADE.txt to indicate how ports must be set for TLS. (closes issue ASTERISK-16959) reported by Olaf Holthausen (closes issue ASTERISK-19201) reported by Chris Mylonas (closes issue ASTERISK-19204) reported by Chris Mylonas Review: https://reviewboard.asterisk.org/r/1709 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	5de11bfd6e	Redocuments sip types peer, user, friend in sip.conf.sample There was faulty information in the sample config describing user as a synonym for friend so it has been changed to better elaborate on the differences between the three entity types. (closes issue ASTERISK-15537) Reported by: yarique git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352511 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	6aa2cd51f9	Adds pjmedia probation concepts to res_rtp_asterisk's learning mode. In order to better handle RTP sources with strictrtp enabled (which is now default in 10) using the learning mode to figure out new sources when they change is handled by checking for a number of consecutive (by sequence number) packets received to an rtp struct based on a new configurable value called 'probation'. Also, during learning mode instead of liberally accepting all packets received, we now reject packets until a clear source has been determined. Review: https://reviewboard.asterisk.org/r/1663/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@351287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	3fa4645c0b	Correct eventtype names in cel_odbc and cel_pgsql sample files git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	26966a5cd8	Use compatible names for event extra data for various CEL backends. * Change eventextra to extra in cel_psql.c and cel_odbc.c. * Change EventExtra to Extra in cel_manager.c. (issue ASTERISK-17190) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350571 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	773074f3b7	Add missing CEL logging fields to various CEL backends. * Add missing eventextra to cel_psql.c and cel_odbc.c. * Add missing PeerAccount and EventExtra to cel_manager.c. * Add missing userdeftype support for cel_custom.conf.sample and cel_sqlite3_custom.conf.sample. (closes issue ASTERISK-17190) Reported by: Bryant Zimmerman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	6dbb78d453	Correct two flaws in sip.conf.sample related to AST-2011-013. * The sample file listed two values for the 'nat' option as being the default. Only 'force_rport' is the default. * The warning about having differing 'nat' settings confusingly referred to both peers and users. ........ Merged revisions 348515 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ea8c309ecf	Fix accidental use of tabs instead of spaces from previous features.conf.sample change git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	9b52c5f1c9	Document PARKINGSLOT variable in features.conf.sample (issue ASTERISK-16239) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	c70441c168	Re-resolve the STUN address if a STUN poll fails for res_stun_monitor. The STUN socket must remain open between polls or the external address seen by the STUN server is likely to change. However, if the STUN request poll fails then the STUN server address needs to be re-resolved and the STUN socket needs to be closed and reopened. * Re-resolve the STUN server address and create a new socket if the STUN request poll fails. * Fix ast_stun_request() return value consistency. * Fix ast_stun_request() to check the received packet for expected message type and transaction ID. * Fix ast_stun_request() to read packets until timeout or an associated response packet is found. The stun_purge_socket() hack is no longer required. * Reduce ast_stun_request() error messages to debug output. * No longer pass in the destination address to ast_stun_request() if the socket is already bound or connected to the destination. (closes issue ASTERISK-18327) Reported by: Wolfram Joost Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1595/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@346700 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	2b89c88293	Update queues.conf.sample documentation. Update the documentation surrounding the use of MONITOR_EXEC to make it more clear that it can be used for both Monitor() and MixMonitor() usage. (closes issue ASTERISK-17413) Reported by: David Woolley Patches: issue18817_mixmonitor_queues_doc.diff by Michael L. Young (License #5026) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@346472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	da4f0afd6f	Default to nat=yes; warn when nat in general and peer differ It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and 1.6.2, this would mean if one setting was nat=yes or nat=route and the other was either nat=no or nat=never. In 1.8 and 10, this would mean when one was nat=force_rport and the other was nat=no. In order to address this problem, it was decided to switch the default behavior to nat=yes/force_rport as it is the most commonly used option and to strongly discourage setting nat per-peer/user when at all possible. For more discussion of the issue, please see: http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html (closes issue ASTERISK-18862) Review: https://reviewboard.asterisk.org/r/1591/ ........ Merged revisions 345776 from http://svn.asterisk.org/svn/asterisk/branches/1.4 ........ Merged revisions 345800 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	f435f45c50	Restore SIP DTMF overlap dialing method. The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support working correctly removed a long standing ability to do overlap dialing using DTMF in the early media phase of a call. See ASTERISK-18702 it has a very good description of the issue. I started with Pavel Troller's chan_sip.diff patch on issue ASTERISK-18702. * Added 'dtmf' enum value to sip.conf allowoverlap config option. The new option value causes the Incomplte application to not send anything with chan_sip so the caller can supply more digits via DTMF. * Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE since that is what it really means. * Fixed get_destination() inconsistency with the pickup extension matching. * Fixed initialization of PAGE3 of global_flags in reload_config(). (closes issue ASTERISK-18702) Reported by: Pavel Troller Review: https://reviewboard.asterisk.org/r/1517/ Review: https://reviewboard.asterisk.org/r/1582/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	85f4b075f5	Update documentation for leastrecent strategy. In queues.conf.sample the leastrecent strategy was incorrectly described. Now updated to reflect how the strategy actually checks peers. (closes issue ASTERISK-17854) Reported by: Sebastian Denz Patches: queues.conf-doc_issue.patch (License #6139) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343047 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	ab2dacb555	Several fixes to the chan_sip dynamic realtime peer/user lookup There were several problems with the dynamic realtime peer/user lookup code. The lookup logic had become rather hard to read due to lots of incremental changes to the realtime_peer function. And, during the addition of the sipregs functionality, several possibilities for memory leaks had been introduced. The insecure=port matching has always been broken for anyone using the sipregs family. And, related, the broken implementation forced those using sipregs to still have an ipaddr column on their sippeers table. Thanks Terry Wilson for comprehensive testing and finding and fixing unexpected behaviour from the multientry realtime call which caused the realtime_peer to have a completely unused code path. This changeset fixes the leaks, the lookup inconsistenties and that you won't need an ipaddr column on your sippeers table anymore (when you're using sipregs). Beware that when you're using sipregs, peers with insecure=port will now start matching! (closes issue ASTERISK-17792) (closes issue ASTERISK-18356) Reported by: marcelloceschia, Walter Doekes Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1395 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342927 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	68b523463d	Cleanup references to sipusers and sipfriends dynamic realtime families Somewhere between 1.4 and 1.8 the sipusers family has become completely unused. Before that, the sipfriends family had been obsoleted in favor of separate sipusers and sippeers families. Apparently, they have been merged back again into a single family which is now called "sippeers". Reviewed by: irroot, oej, pabelanger Review: https://reviewboard.asterisk.org/r/1523 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	f33e20e5b1	Adds documentation for QueueMemberStatus event generation git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@338718 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	b50b776427	Comment out entries in sample res_pktccops.conf. With these options enabled, they can cause Asterisk to freak out by SYN flooding a network and eating the CPU. Obviously it would be good to fix the code so that this can't happen, but we can at least change the default configuration so it doesn't happen. This was reported downstream to the Fedora issue tracker: https://bugzilla.redhat.com/show_bug.cgi?id=658431 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337774 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	c9325708c8	default 'sipstorecause' to no We've decided to disable this feature by default in future 1.8 versions. This would be an unexpected behavior change for anyone depending on that SIP_CAUSE update in their dialplan. Please refer to the asterisk-dev mailing list more information: http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html (issue AST-580) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	56549c96ab	Review: https://reviewboard.asterisk.org/r/1364/ This update adds a new AMI event, TestEvent, which is enabled when the TEST_FRAMEWORK compiler flag is defined. It also adds initial usage of this event to app_voicemail. The TestEvent AMI event is used extensively by the voicemail tests in the Asterisk Test Suite. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332817 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	9328590ddb	Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards. France Telecom brings layer 2 and layer 1 down on BRI lines when the line is idle. When layer 1 goes down Asterisk cannot make outgoing calls and the HA8 and HB8 cards also get IRQ misses. The inability to make outgoing calls is because the line is in red alarm and Asterisk will not make calls over a line it considers unavailable. The IRQ misses for the HA8 and HB8 card are because the hardware is switching clock sources from the line which just brought layer 1 down to internal timing. There is a DAHDI option for the B410P card to not tell Asterisk that layer 1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp teignored=1". There is a similar DAHDI option for the HA8 and HB8 cards: "modprobe wctdm24xxp bri_teignored=1". Unfortunately that will not clear up the IRQ misses when the telco brings layer 1 down. * Add layer 2 persistence option to customize the layer 2 behavior on BRI PTMP lines. The new option has three settings: 1) Use libpri default layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when the peer brings it down. 3) Leave layer 2 down when the peer brings it down. Layer 2 will be brought up as needed for outgoing calls. JIRA AST-598 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332264 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	422e191e03	Fix multiple parking issues. JIRA ASTERISK-17183 Multi-parkinglot directs calls to wrong parkinglot. JIRA ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430 ParkedCall() with no extension should pickup first available call and does not. JIRA AST-576 Issues with parking lots * Removed searching for parking lots by extension. Parking lots can only be found by the parking lot name since parking lot access extensions and spaces are not guaranteed to be unique. * Added parking_lot_name option to the Park and ParkedCall applications. Updated documentation for Park and ParkedCall applications. * Add parkext_exclusive configuration option to make parking entry extensions specify which parking lot they access. (closes issue ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett, David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi Quezada (closes issue ASTERISK-17430) Reported by: Philippe Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA AST-624 'next' setting for findslot does nothing * Reimplemented since findslot feature option broken by -r114655. (closes issue ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett JIRA ASTERISK-15792 Dialplan continues execution after transfer to park. This happens for DTMF attended transfer, DTMF blind transfer, and DTMF one-touch-parking if the party initiating these features also initiated the call. * Fixed the return code from the affected builtin features when parking a call. (closes issue ASTERISK-15792) Reported by: Mat Murdock Tested by: rmudgett, twilson JIRA AST-607 The courtesytone is not playing to the expected call when picking up a parked call. This is mostly a documentation problem. However, the option is not reset to the default when features.conf is reloaded. * Updated features.conf.sample documentation for courtesytone and parkedplay options. * Reset the parkedplay option to default when features.conf is reloaded. JIRA AST-615 AMI Park action followed by features reload results in orphaned channels in parking lot. * Reloading features.conf will not touch parking lots that have calls still parked in them. Reload again at a later time. Misc additional fixes: * Added unit test for parking lot dialplan usage checking. * Made update connected line when a parked call is retrieved from a parking lot. * Made retrieved parked call stop ringing or MOH depending upon how the call was waiting in the parking lot. * Made CLI "features show" indicate if the parking lot is enabled for use. * Added PARKINGDYNEXTEN channel variable to allow dynamic parking lots to specify the parking lot access extension. * Made AMI ParkedCalls action ParkedCall events have a Parkinglot header. * Made AMI ParkedCalls action ParkedCallsComplete event have a Total header. * Fixed potential deadlock from AMI Park action holding channel locks while calling masq_park_call(). * Fixed several places where ast_strdupa() were used inside of loops. (Mostly fixed by refactoring the loop body into its own function.) * Fixed copy_parkinglot() copying too much from the source parking lot. Extracted the parking lot configuration settings into struct parkinglot_cfg. * Refactored courtesytone playing code to put the channel not playing the tone in autoservice. * Fix when pbx-parkingfailed is played that the other channel is put in autoservice if it exists. * Fixed parkinglot reference leak in parked_call_exec() error paths. * Fixed parkinglot_unref() use of parkinglot after it was unreffed. * Made destroy the struct ast_parkinglot parkings lock when done. * Refactored the features.conf parking lot configuration code to eliminate redundancy. * Fixed feature reload to better protect parking lots. * Fixed parking lot container reference leak in handle_parkedcalls(). * Fixed the total count in handle_parkedcalls(). Review: https://reviewboard.asterisk.org/r/1358/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332100 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	f01a484b48	Added the 'storesipcause' option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,<chan name>) on the channel. Having chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function. AST-580 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	b1a1cd5e57	Fix New Zealand indications profile based on http://www.telepermit.co.nz/TNA102.pdf (closes issue ASTERISK-16263) Reported by: richardf Patches: nz-indications.patch uploaded by richardf (License #6015) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@329709 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	31a1b94622	Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@329527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b111b763cd	Document parkinglot in chan_dahdi.conf.sample. * Document existing feature in chan_dahdi.conf.sample. * Remove some dead code related to the parkinglot option. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@329203 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	89391b8930	Add ATXFER_NULL_TECH note in features.conf.sample. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328014 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	cf8b27cd39	Misc minor changes in chan_sip. * Add load failure exit if primary SIP container(s) could not get created in chan_sip.c:load_module(). * Removed a redundant static prototype. * Some typos. * Some whitespace. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@325935 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	ce2e420902	Remove extra 'the'. Reported by Vlad Povorozniuc git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	186eeac9f4	Revert previous merge which had extra changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	0ce6617ee1	Remove extra 'the'. Reported by Vlad Povorozniuc git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324239 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	8d1596311c	Additional documentation for bindaddr. Note that bindaddr will only enable UDP instead of both UDP and TCP which is what I would expect for backwards compatibility with systems being upgraded which only support UDP transportation. (closes issue ASTERISK-17976) Reported by: Sean Darcy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@323234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	6bf7b59c8f	Use correct syntax for 'sip notify snom-reboot' (closes issue ASTERISK-17915) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322189 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	f1a3972b96	Also document the 'queue-minute' option. (closes issue #19386) Reported by: juanmol git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321685 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	b3a2f27111	Adds legacy_useroption_parsing to address interoperability concerns. With the new option engaged, Asterisk should interpret user fields with useroptions contained within the userfield of the uri by stripping them out of the original message whenever a semicolon is encountered in the userfield string. (closes issue #18344) Reported by: danimal Tested by: jrose Review: https://reviewboard.asterisk.org/r/1223/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	2f5ff86317	Documenting an observed behavior of features in features.conf. Since parkinglots use an integer for the parkinglot extensions, leading zeros specified in the configuration file are ignored. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@318148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	232db04967	Remove unused directory and clear up some documentation. (closes issue #19193) Reported by: bchia Patches: cel-csv.diff uploaded by lathama (license 1028) Tested by: lathama, Marquis42 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317058 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	4468fe047e	Merged revisions 314620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r314620 \| mnicholson \| 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) \| 20 lines Merged revisions 314607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r314607 \| mnicholson \| 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) \| 14 lines Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so. Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action. AST-2011-005 AST-2011-006 (closes issue #18787) Reported by: kobaz (related to issue #18996) Reported by: tzafrir ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@314628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	b5e04b75dc	Merged revisions 312764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312764 \| mnicholson \| 2011-04-05 09:13:07 -0500 (Tue, 05 Apr 2011) \| 15 lines Merged revisions 312761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312761 \| mnicholson \| 2011-04-05 09:10:34 -0500 (Tue, 05 Apr 2011) \| 8 lines Limit the number of unauthenticated manager sessions and also limit the time they have to authenticate. AST-2011-005 (closes issue #18996) Reported by: tzafrir Tested by: mnicholson ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@312766 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	8439e8344c	Incorrect default example; the field is actually internally named "clid", not "callerid". (closes issue #19040) Reported by: wcselby Tested by: tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311930 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	d0829474b2	Merged revisions 311049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r311049 \| alecdavis \| 2011-03-17 23:45:47 +1300 (Thu, 17 Mar 2011) \| 17 lines Merged revisions 311048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r311048 \| alecdavis \| 2011-03-17 23:43:35 +1300 (Thu, 17 Mar 2011) \| 12 lines Remove extra quote in indications.conf Picking low hanging fruit. (closes issue #18971) Reported by: IgorG Patches: based on indications.conf.sample.diff uploaded by IgorG (license 20) Tested by: IgorG ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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1481 Commits (395b4b4898e15d7dda36391260e5ab3e091507f9)