asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Nicholson	1da3304813	Merged revisions 325545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325545 \| mnicholson \| 2011-06-29 11:18:39 -0500 (Wed, 29 Jun 2011) \| 2 lines make framehooks prevent native bridging (for real this time) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325547 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	6c7d437287	Merged revisions 325537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325537 \| mnicholson \| 2011-06-29 10:34:47 -0500 (Wed, 29 Jun 2011) \| 2 lines don't do native/remote bridging if a framehook is active on the channel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	db15b0010c	Merged revisions 324955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324955 \| tilghman \| 2011-06-27 11:30:50 -0500 (Mon, 27 Jun 2011) \| 5 lines Save and restore errno from within signal handlers. This is recommended by the POSIX standard, as well as by the sigaction(2) manpage for various platforms that we support (e.g. Mac OS X). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324961 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	d5ea9e5ae2	Merged revisions 324652 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r324652 \| dvossel \| 2011-06-23 13:23:21 -0500 (Thu, 23 Jun 2011) \| 20 lines Merged revisions 324634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r324634 \| dvossel \| 2011-06-23 13:18:46 -0500 (Thu, 23 Jun 2011) \| 13 lines Merged revisions 324627 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r324627 \| dvossel \| 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011) \| 7 lines Addresses AST-2011-010, remote crash in IAX2 driver Thanks to twilson for identifying the issue and providing the patches. AST-2011-010 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	385b8c6f8b	Merged revisions 324484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324484 \| twilson \| 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) \| 20 lines Stop sending IPv6 link-local scope-ids in SIP messages The idea behind the patch listed below was used, but in a more targeted manner. There are now address stringification functions for addresses that are meant to be sent to a remote party. Link-local scope-ids only make sense on the machine from which they originate and so are stripped in the new functions. There is also a host sanitization function added to chan_sip which is used for when peer and dialog tohost fields or sip_registry hostnames are used to craft a SIP message. Also added are some basic unit tests for netsock2 address parsing. (closes issue ASTERISK-17711) Reported by: ch_djalel Patches: asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251) Review: https://reviewboard.asterisk.org/r/1278/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	09a359449e	Merged revisions 324364 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324364 \| dvossel \| 2011-06-21 15:11:52 -0500 (Tue, 21 Jun 2011) \| 10 lines Fixes locking inversion issue in ast_async_goto() During this function we can not hold the "chan" lock while doing the masquerade, the explicit goto on the tmp chan, or the channel alloc. Instead we need to get the channel lock, store off information about the channel that we need, and then let the channel lock go for the remainder of the function. Review: https://reviewboard.asterisk.org/r/1275/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	3d6c1ccd91	Merged revisions 324178 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324178 \| lmadsen \| 2011-06-17 14:51:16 -0400 (Fri, 17 Jun 2011) \| 2 lines Add Username and Secret fields to manager Login action. Pointed out by Vlad Povorozniuc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324179 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	71e4b2a5d1	Merged revisions 324115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324115 \| lmadsen \| 2011-06-17 11:14:54 -0400 (Fri, 17 Jun 2011) \| 3 lines Fix grammar in documentation for Goto() and GotoIf() (closes issue ASTERISK-18023) Reported by: Tim Osman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324131 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	34e2305ae7	Merged revisions 324048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324048 \| twilson \| 2011-06-16 17:35:41 -0500 (Thu, 16 Jun 2011) \| 8 lines Lock the channel before calling the setoption callback The channel needs to be locked before calling these callback functions. Also, sip_setoption needs to lock the pvt and a check p->rtp is non-null before using it. Review: https://reviewboard.asterisk.org/r/1220/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	c33e1b0e27	Merged revisions 323754 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323754 \| twilson \| 2011-06-15 13:21:52 -0500 (Wed, 15 Jun 2011) \| 23 lines Merged revisions 323733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323733 \| twilson \| 2011-06-15 13:13:00 -0500 (Wed, 15 Jun 2011) \| 16 lines Merged revisions 323732 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323732 \| twilson \| 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011) \| 9 lines Fix DYNAMIC_FEATURES DYNAMIC_FEATURES were broken by a recent DTMF change. This patch makes sure that dynamic features are also checked when deciding whether or not to pass DTMF through or store it for interpreting. (closes issue ASTERISK-17914) Reported by: vrban ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b2d0ea5fea	Merged revisions 323669-323670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323669 \| rmudgett \| 2011-06-15 11:43:18 -0500 (Wed, 15 Jun 2011) \| 21 lines [regression] Voicemail MWI is no longer sent. When leaving a voicemail, the MWI message is never sent. The same thing happens when checking a voicemail and marking it as read. If you restart Asterisk, everything comes up at that state correctly, but changes to the messages in voicemail causes the light to not be set appropriately. Very easy to reproduce. * Made ast_event_check_subscriber() return TRUE if there are ANY subscribers to an event type when there are no restricting ie values passed. This allows an event being queued to be queued. (closes issue ASTERISK-18002) Reported by: lmadsen Tested by: lmadsen, irroot Patches: jira_asterisk_18002_v1.8.patch uploaded by rmudgett (License #5621) (closes issue ASTERISK-18019) ........ r323670 \| rmudgett \| 2011-06-15 11:43:31 -0500 (Wed, 15 Jun 2011) \| 7 lines Add a test to the event unit tests to catch ASTERISK-18002. The new tests check to see if there are ANY subscribers to the event type when ast_event_check_subscriber() is not passed any specific ie values. (issue ASTERISK-18002) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	affae67cd2	Merged revisions 323608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323608 \| seanbright \| 2011-06-15 11:31:53 -0400 (Wed, 15 Jun 2011) \| 39 lines Merged revisions 323579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323579 \| seanbright \| 2011-06-15 11:22:50 -0400 (Wed, 15 Jun 2011) \| 32 lines Merged revisions 323559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323559 \| seanbright \| 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) \| 25 lines Resolve a segfault/bus error when we try to map memory that falls on a page boundary. The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the mmap'd region. The problem with this is that reading/writing to that extra byte outside of the bounds of the underlying fd causes a bus error. The real issue is that we are working with both a FILE * and the raw fd underneath it and not synchronizing between them. The code that was removed in ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping the fd. Looking at the manager code in 1.4 reveals that the FILE * in 'struct mansession' is never used except to create a temporary file that we immediately fdopen. This means we just need to write a 0 byte to the fd and everything will just work. The other branches require a call to fflush() which, while not a guaranteed fix, should reduce the likelihood of a crash. This all makes sense in my head. (closes issue ASTERISK-16460) Reported by: Ravelomanantsoa Hoby (hoby) Patches: issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323609 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	70d9527951	Merged revisions 323456 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323456 \| rmudgett \| 2011-06-14 19:50:20 -0500 (Tue, 14 Jun 2011) \| 1 line Add missing break in ast_event_get_cached(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	9ff8844c7f	Merged revisions 323392,323394 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323392 \| rmudgett \| 2011-06-14 12:21:24 -0500 (Tue, 14 Jun 2011) \| 6 lines Add more strict hostname checking to ast_dnsmgr_lookup(). Change suggested in review. Review: https://reviewboard.asterisk.org/r/1240/ ........ r323394 \| rmudgett \| 2011-06-14 12:21:39 -0500 (Tue, 14 Jun 2011) \| 2 lines Made ast_sockaddr_split_hostport() port warning msgs more meaningful. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323397 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	abd7ef817e	Merged revisions 323370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323370 \| twilson \| 2011-06-14 09:33:55 -0700 (Tue, 14 Jun 2011) \| 10 lines Add rtpkeepalives back to 1.8 The RTP-engine conversion left out support for handling rtpkeepalives. This patch adds them back. (closes issue ASTERISK-17304) Reported by: lmadsen Review: https://reviewboard.asterisk.org/r/1226/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323374 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	dafa8a659b	Merged revisions 323213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323213 \| lmadsen \| 2011-06-13 15:51:52 -0400 (Mon, 13 Jun 2011) \| 6 lines Avoid dividing by zero with L() option to Dial() Reported by: nicolasom Patches: issue-17995.patch - nicolasom (License #5994) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323214 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	58ca560291	Merged revisions 322981 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322981 \| twilson \| 2011-06-10 08:29:00 -0700 (Fri, 10 Jun 2011) \| 11 lines Avoid a DB1 infinite loop bug Explicity check the last entry in the DB and make sure that we don't iterate past it. Since there can be no duplicates, this just makes sure that we stop after matching the last key. This patch also refactors the code to get away from some code duplication. A previous patch added many astdb tests and this patch passed them. Review: https://reviewboard.asterisk.org/r/1259/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322982 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	0a8f9d2cf0	Merged revisions 322749 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322749 \| rmudgett \| 2011-06-09 11:31:53 -0500 (Thu, 09 Jun 2011) \| 15 lines Remove potential deadlock in call pickup race. Deadlock is possible in ast_do_pickup() when holding the target channel lock and trying to get the chan channel lock. Also, holding the target lock when calling ast_channel_masquerade() is not a good idea because that routine does deadlock avoidance. * Removed the need to hold the target lock after marking the target with a datastore and getting the connected line data off of the target channel. * Moved can_pickup() to ast_can_pickup() in features.c. Now all the call pickup methods use the same basic call pickup availability check. Review: https://reviewboard.asterisk.org/r/1234/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322750 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	4b773e2ed9	Merged revisions 322425 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322425 \| rmudgett \| 2011-06-08 13:46:30 -0500 (Wed, 08 Jun 2011) \| 16 lines SRV lookup attempted for SIP peers listed as an IP address. Asterisk attempts to SRV lookup a host name even if the host name is an IP address. Regression introduced when IPv6 support was added. * Restored the check in ast_dnsmgr_lookup() to see if the given host name is an IP address. The IP address could be in either IPv4 or IPv6 formats. (closes issue ASTERISK-17815) Reported by: Byron Clark Tested by: Byron Clark, Richard Mudgett Patches: issue19248_v1.8.patch - uploaded by Richard Mudgett (License #5621) Review: https://reviewboard.asterisk.org/r/1240/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	4ab3825fe4	Merged revisions 322069 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322069 \| jrose \| 2011-06-06 14:07:56 -0500 (Mon, 06 Jun 2011) \| 8 lines Fixes level toggling for logger set levels since it was reversed (closes issue ASTERISK-17850) Reported by: Luke H Tested by: jrose, Luke H Review: https://reviewboard.asterisk.org/r/1244/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	31bcafab5b	Merged revisions 321924 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321924 \| rmudgett \| 2011-06-03 16:49:17 -0500 (Fri, 03 Jun 2011) \| 5 lines Be more explicit for CCSS generic device state event subscription. Make CCSS generic device state event subscription specify the AST_EVENT_IE_STATE ie exists to be safe. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	85aa126b34	Merged revisions 321871 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321871 \| rmudgett \| 2011-06-03 15:58:13 -0500 (Fri, 03 Jun 2011) \| 27 lines Event subscription fixes. Must commit the subscription fixes together with the integration subscription tests. The subscription fixes cause an erroneously passing test to fail. The new subscription tests detect errors without the subscription fixes. * Added missing event_names[] table entry. * Reworked ast_event_check_subscriber()/match_sub_ie_val_to_event() to correctly detect if a subscriber exists for the proposed event. * Made match_ie_val() and match_sub_ie_val_to_event() check the buffer length for RAW payload types. * Fixed error handling memory leak in ast_event_sub_activate(), ast_event_unsubscribe(), and ast_event_queue(). * Made ast_event_new() and ast_event_check_subscriber() better protect themselves from an invalid payload type. * Added container lock protection between removing old cache events and adding the new cached event in ast_event_queue_and_cache()/event_update_cache(). * Added new event subscription tests. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321872 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	397c379a7d	Merged revisions 321812-321813 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321812 \| rmudgett \| 2011-06-03 14:55:21 -0500 (Fri, 03 Jun 2011) \| 1 line Correct IAX2 and SIP event subscription description string. ........ r321813 \| rmudgett \| 2011-06-03 14:56:09 -0500 (Fri, 03 Jun 2011) \| 1 line Constify subscription description parameter string. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321814 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	6357719a82	Fix some astobj2 iterator breakage, add another unit test. Review: https://reviewboard.asterisk.org/r/1254/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321752 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	49927bcbb8	Merged revisions 321547 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321547 \| rmudgett \| 2011-06-01 18:11:55 -0500 (Wed, 01 Jun 2011) \| 1 line CDR comment tweaks. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321548 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	3f4d0e8743	Support routing text messages outside of a call. Asterisk now has protocol independent support for processing text messages outside of a call. Messages are routed through the Asterisk dialplan. SIP MESSAGE and XMPP are currently supported. There are options in sip.conf and jabber.conf that enable these features. There is a new application, MessageSend(). There are two new functions, MESSAGE() and MESSAGE_DATA(). Documentation will be available on the project wiki, wiki.asterisk.org. Thanks to Terry Wilson for the assistance with development and to David Vossel for helping with some additional testing. Review: https://reviewboard.asterisk.org/r/1042/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	9d8943868c	Merged revisions 321392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321392 \| rmudgett \| 2011-05-27 18:45:41 -0500 (Fri, 27 May 2011) \| 12 lines Crash when using hagi and no servers are available. When none of the servers returned by the SRV querey respond, asterisk crashes. The problem is that if the loop over all the SRV entries finishes then the srv_context has already been cleaned up. * Make ast_srv_cleanup() check to see if the context is already cleaned up. (closes issue #19256) Reported by: byronclark ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	a2ca0997a6	Merged revisions 321333 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321333 \| lmadsen \| 2011-05-27 17:40:23 -0400 (Fri, 27 May 2011) \| 7 lines Allow parking lot hints and musicclass to be set. (closes issue #19378) Reported by: sboily_proformatique Patches: pf_parkinghint_music_fix uploaded by sboily proformatique (license 206) Tested by: russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321334 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	7cc83a9018	Merged revisions 321211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321211 \| alecdavis \| 2011-05-27 20:31:15 +1200 (Fri, 27 May 2011) \| 16 lines Fix *8 directed pickup locks system during pickupsound play out move playout from sip_pickup_thread to bridge using BRIDGE_PLAY_SOUND method, This stop the clash of 2 threads trying to write audio to same channel. In addition fixes choppy audio beep in issue 19177. (issue #18654) (issue #19177) Reported by: Docent Patches: review1232-1.8.diff.txt alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1232/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Murawki	0648d9595b	Merged revisions 321100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321100 \| markm \| 2011-05-26 16:09:35 -0400 (Thu, 26 May 2011) \| 11 lines ast_sockaddr_resolve() in netsock2.c may deref a null pointer Added a null check in netsock2 ast_sockaddr_resolve() as well as added default initalizers in chan_sip parse_uri_legacy_check() to make sure that invalid uris will make null (and not undefined) user,pass,domain,transport variables (closes issue #19346) Reported by: kobaz Patches: netsock2.patch uploaded by kobaz (license 834) Tested by: kobaz, Marquis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321101 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	0c34e54d1a	Merged revisions 321042 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321042 \| twilson \| 2011-05-26 10:29:54 -0700 (Thu, 26 May 2011) \| 6 lines Initialize stack-allocated ast_sockaddrs before use It is important to always initialize ast_sockaddrs before use--even if they are passed to ast_sockaddr_copy as the underlying storage could be bigger than what ends up being copied--leaving part of the data unitialized. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	fc8d4e823c	Use va_copy for stringfields The ast_string_field_build_va functions were written to take to separate va_lists to work around FreeBSD 4 not having va_copy defined. In the end, we don't support anything using gcc < 3 anyway because we use va_copy all over the place anyway. This patch just simplifies things by removing the second va_list function arguments in favor of va_copy. Review: https://reviewboard.asterisk.org/r/1233/ --This line, and those below, will be ignored-- M include/asterisk/stringfields.h M main/utils.c M main/channel.c git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	0096238b52	Merged revisions 320823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320823 \| rmudgett \| 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) \| 18 lines The AMI Newstate event contains different information between v1.4 and v1.8. The addition of connected line support in v1.8 changes the behavior of the channel caller ID somewhat. The channel caller ID value no longer time shares with the connected line ID on outgoing call legs. The timing of some AMI events/responses output the connected line ID as caller ID. These party ID's are now separate. * The ConnectedLineNum and ConnectedLineName headers were added to many AMI events/responses if the CallerIDNum/CallerIDName headers were also present. (closes issue #18252) Reported by: gje Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1227/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a42bf8cc92	Merged revisions 320796 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320796 \| rmudgett \| 2011-05-25 11:23:11 -0500 (Wed, 25 May 2011) \| 17 lines Give zombies a safe channel driver to use. Recent crashes from zombie channels suggests that they need a safe home to goto. When a masquerade happens, the physical part of the zombie channel is hungup. The hangup normally sets the channel private pointer to NULL. If someone then blindly does a callback to the channel driver, a crash is likely because the private pointer is NULL. The masquerade now sets the channel technology of zombie channels to the kill channel driver. Related to the following issues: (issue #19116) (issue #19310) Review: https://reviewboard.asterisk.org/r/1224/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320820 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	024e4bd0f7	Merged revisions 320650 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320650 \| rmudgett \| 2011-05-23 12:53:44 -0500 (Mon, 23 May 2011) \| 16 lines Add ConnectedLineNum/Name headers to output of AMI action Status. * Add ConnectedLineNum and ConnectedLineName headers to the output of the AMI action Status. This makes it easier to find out who the channel is connected to without having to lookup BridgedChannel or when they are connected to an application (e.g.: VoiceMail) which has no bridged channel. * Bridged channels with no CallerID had "" instead of "<unknown>" output, that might be a bug as "<unknown>" was what older versions used. (closes issue #18158) Reported by: gareth Patches: svn-292308.diff uploaded by gareth (license 208) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320651 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	181e91a213	Merged revisions 320568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r320568 \| dvossel \| 2011-05-23 11:18:33 -0500 (Mon, 23 May 2011) \| 14 lines Merged revisions 320562 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r320562 \| dvossel \| 2011-05-23 11:15:18 -0500 (Mon, 23 May 2011) \| 9 lines Adds missing part to the ast_tcptls_server_start fails second attempt to bind patch. (closes issue #19289) Reported by: wdoekes Patches: issue19289_delay_old_address_setting_tcptls_2.patch uploaded by wdoekes (license 717) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	67637652f4	Merged revisions 320338 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r320338 \| dvossel \| 2011-05-20 16:39:36 -0500 (Fri, 20 May 2011) \| 14 lines Merged revisions 320271 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r320271 \| dvossel \| 2011-05-20 16:24:48 -0500 (Fri, 20 May 2011) \| 8 lines Fixes issue with ast_tcptls_server_start failing on second attempt to bind. (closes issue #19289) Reported by: wdoekes Patches: issue19289_delay_old_address_setting_tcptls.patch uploaded by wdoekes (license 717) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320340 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	2af231dd91	Merged revisions 320059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320059 \| rmudgett \| 2011-05-20 12:03:49 -0500 (Fri, 20 May 2011) \| 1 line Misc comment cleanup in features.c. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ae091d166a	Merged revisions 320057 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320057 \| rmudgett \| 2011-05-20 11:43:02 -0500 (Fri, 20 May 2011) \| 19 lines Crash while transferring a call during DTMF feature timeout. When a call is being attended transferred during the time between AST_FRAME_DTMF_BEGIN and AST_FRAME_DTMF_END, the transferred channel becomes a zombie (so tech data is not available), making ast_dtmf_stream() segfault when it tries to send the DTMF digit (at least with SIP channels). Patch based on feature-end-zombie.patch uploaded by Irontec (license 1256) * Check for zombies when ast_channel_bridge() returns. * Guarantee that the fo parameter value is initialized in ast_channel_bridge() before any returns. (closes issue #19116) Reported by: Irontec Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320058 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b1cfd0e059	Merged revisions 320007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320007 \| rmudgett \| 2011-05-20 11:19:01 -0500 (Fri, 20 May 2011) \| 2 lines Change some variable names to make pickup code easier to understand. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320013 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	0436c501c9	Merged revisions 319997 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319997 \| rmudgett \| 2011-05-20 10:48:25 -0500 (Fri, 20 May 2011) \| 25 lines Crash when using directed pickup applications. The directed pickup applications can cause a crash if the pickup was successful because the dialplan keeps executing. This patch does the following: * Completes the channel masquerade on a successful pickup before the application returns. The channel is now guaranteed a zombie and must not continue executing the dialplan. * Changes the return value of the directed pickup applications to return zero if the pickup failed and nonzero(-1) if the pickup succeeded. * Made some code optimizations that no longer require re-checking the pickup channel to see if it is still available to pickup. (closes issue #19310) Reported by: remiq Patches: issue19310_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, remiq, rmudgett Review: https://reviewboard.asterisk.org/r/1221/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	87004f0d9f	Merged revisions 319866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319866 \| jrose \| 2011-05-19 13:32:38 -0500 (Thu, 19 May 2011) \| 11 lines Fix Randomize option on Park() The randomize option was generally not working like it should have at all on Park(). This patch restores intended functionality. (closes issue #18862) Reported by: davidw Tested by: jrose Review: https://reviewboard.asterisk.org/r/1222/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b33fc4db48	Merged revisions 319758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319758 \| rmudgett \| 2011-05-19 11:50:48 -0500 (Thu, 19 May 2011) \| 21 lines CCSS generic agent with POTS and ISDN phones fail caller busy call-back test. If the following is true after a CCSS activation: * The generic agent is for an analog phone or ISDN phone. (Caller party) * The called party becomes available. * The caller party is not available. When the caller party becomes available, the caller is not alerted to the called party being available. The generic agent still thinks the caller is busy. * Fixed the generic agent device state event subscription to look for all device states that are considered available. * Encapsulated the device state test for CCSS generic device available in cc_generic_is_device_available(). Made the generic agent and monitor use the new function instead of the manually coded inline equivalent. JIRA AST-559 JIRA SWP-3462 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	1de75f0a4d	Merged revisions 319261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319261 \| jrose \| 2011-05-16 16:00:55 -0500 (Mon, 16 May 2011) \| 2 lines Makes busy detection in dsp.c always allow for at least one frame (20ms) of error so that 200ms tone lengths don't get ignored by single frame error lengths. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	eddc32a3b3	Merged revisions 319259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319259 \| rmudgett \| 2011-05-16 15:33:37 -0500 (Mon, 16 May 2011) \| 13 lines Deadlock between generic CCSS agent and native ISDN CCSS. Deadlock can occur when the generic CCSS agent is deleting duplicate CC offers and the native ISDN CC driver is processing an incoming CC message. The cc_core_instances container lock cannot be held when an agent or monitor callback is invoked without the possibility of a deadlock. * Make kill_duplicate_offers() remove the reference in cc_core_instances outside of the container lock. JIRA AST-566 JIRA SWP-3469 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	475ef22b20	Merged revisions 318921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318921 \| bbryant \| 2011-05-13 14:09:34 -0400 (Fri, 13 May 2011) \| 8 lines Fixes a segmentation fault in dynamic hints when a channel technology isn't loaded for a hint. (closes issue #18495) Reported by: bertrand Tested by: bertrand ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318922 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	db89abf0bd	Merged revisions 318868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318868 \| rmudgett \| 2011-05-13 11:28:26 -0500 (Fri, 13 May 2011) \| 19 lines CDR's are being written immediately on caller hangup. CDR's are being written immediately on caller hangup. The dialplan is not able to modify it in the h exten. The h exten in the initial context is not run before closing CDR's when the bridge is unlinked if a macro is active and does not have an h exten. * Make ast_bridge_call() check for an h exten in the current context and if a macro is active then the initial context. The first h exten found is then run before closing the CDR. (closes issue #18212) Reported by: leearcher Patches: issue18212_v1.8.patch uploaded by rmudgett (license 664) Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1206/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	892b7a2efd	Merged revisions 318671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318671 \| alecdavis \| 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) \| 30 lines Fix directed group pickup feature code *8 with pickupsounds enabled Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues. 1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked. 2). dialplan applications for directed_pickups shouldn't beep. 3). feature code for directed pickup should beep on success/failure if configured. Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite. Moved app_directed:pickup_do() to features:ast_do_pickup(). Functions below, all now use the new ast_do_pickup() app_directed_pickup.c: pickup_by_channel() pickup_by_exten() pickup_by_mark() pickup_by_part() features.c: ast_pickup_call() (closes issue #18654) Reported by: Docent Patches: ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett Review: https://reviewboard.asterisk.org/r/1185/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	bf57bb3c89	Merged revisions 318282 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318282 \| rmudgett \| 2011-05-09 14:07:01 -0500 (Mon, 09 May 2011) \| 24 lines Hangup extension executed twice. When a user hangs up a call, in certain circumstances, the hangup extension can end up being executed twice: 1) If a call is bridged and the 'h' extension executes the Hangup application, then the 'h' extension will be executed twice. 2) If a call is bridged within a macro (Dial or Queue), it has its own 'h' extension, the main context also has an 'h' extension, and the macro 'h' extension executes the Hangup application, then both 'h' extensions will be executed. * Revert originally commited fix for #16106 and just set AST_FLAG_BRIDGE_HANGUP_RUN unconditionally in ast_bridge_call(). The bridge code just executed an 'h' extension so the main PBX loop does not need to execute one as well. (issue #16106) Reported by: ajohnson (issue #16548) Reported by: hajekd ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318283 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	f2df0ed9f1	Increase prepend filename length. (closes issue #19238) Reported by: byronclark Patches: increase_prepend_filename_length.patch uploaded by byronclark (license 1200) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	ff4c7d46c0	Minor change to 318141 to improve parsing behavior. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318193 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	5b77bb5060	Merged revisions 318142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318142 \| mnicholson \| 2011-05-09 09:09:38 -0500 (Mon, 09 May 2011) \| 9 lines Make indicate/control frames WRITE events on framehooks. Also, if a framehook returns a non-control frame, don't forward it to the channel. (closes issue #19251) Reported by: irroot Patches: (modified) framehook_indicate.patch2 uploaded by irroot (license 52) Tested by: irroot ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	229e066dcb	Allows ParkedCall application to specify a parkinglot. When invoking the app parkedcall, the argument can now include '@parkinglot' after the extension. (closes issue #18777) Reported by: cartama Patches: 0018777.diff uploaded by cartama (license 1157) Review: https://reviewboard.asterisk.org/r/1209/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318141 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	c73ea18012	Merged revisions 317917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317917 \| russell \| 2011-05-06 16:06:33 -0500 (Fri, 06 May 2011) \| 7 lines Fix calculation of free RAM to make minmemfree option work. (closes issue #17124) Reported by: loic Patches: pbx_c.diff uploaded by loic (license 1020) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317919 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	c28e2d380c	Merged revisions 317429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317429 \| russell \| 2011-05-05 17:11:19 -0500 (Thu, 05 May 2011) \| 5 lines Only display inband DTMF warning if inband DTMF detection is enabled. (closes issue #18901) Reported by: irroot ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	19b45ad446	Merged revisions 317425 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317425 \| russell \| 2011-05-05 16:53:13 -0500 (Thu, 05 May 2011) \| 7 lines Add missing ActioID handling to Events action. (closes issue #18949) Reported by: edersohe Patches: 0018949.patch uploaded by edersohe (license 1228) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	d508a921bf	Add some new editline bindings by default, and allow for user specified configuration. I excluded the part of this patch that used the HOME environment variable since the built-in editline library goes to great lengths to disallow that. Instead only settings the EDITRC environment variable will use a user specified file. Also, the default environment variable use to determine the edit more is AST_EDITMODE instead of AST_EDITOR (although the latter is still supported). (closes issue #15929) Reported by: kkm Patches: astcli-editrc-v2.diff uploaded by kkm (license 888) 015929-astcli-editrc-trunk.240324.diff uploaded by kkm (license 888) Tested by: seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317395 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Stefan Schmidt	19eb6c7384	Adding the Move to Front Hash functionality Moving a found object to the front of its bucket to reduce the necessary traversal steps to find an object. This change improves the search time on large system with many data or in link lists. (closes issue #19233) Reported by: schmidts Review: https://reviewboard.asterisk.org/r/1201/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316962 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	fe5938c51e	Merged revisions 316917-316919 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316917 \| seanbright \| 2011-05-04 22:23:28 -0400 (Wed, 04 May 2011) \| 5 lines Make sure that tcptls_session is properly initialized. (issue #18598) Reported by: ksn ........ r316918 \| seanbright \| 2011-05-04 22:25:20 -0400 (Wed, 04 May 2011) \| 5 lines Look at the correct buffer for our digest info instead of an empty one. (issue #18598) Reported by: ksn ........ r316919 \| seanbright \| 2011-05-04 22:30:45 -0400 (Wed, 04 May 2011) \| 10 lines Use the correct HTTP method when generating our digest, otherwise we always fail. When calculating the 'A2' portion of our digest for verification, we need the HTTP method that is currently in use. Unfortunately our mapping function was incorrect, resulting in invalid hashes being generated and, in turn, failures in authentication. (closes issue #18598) Reported by: ksn ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	34734f727f	Merged revisions 316663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316663 \| seanbright \| 2011-05-04 10:35:05 -0400 (Wed, 04 May 2011) \| 8 lines Only return a single error via AMI when requesting a forbidden action. (closes issue #19216) Reported by: oej Patches: issue19216-1.8-r316204.patch uploaded by seanbright (license 71) Tested by: seanbright ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	f4417923ce	Merged revisions 316334 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316334 \| dvossel \| 2011-05-03 17:05:59 -0500 (Tue, 03 May 2011) \| 8 lines Fixes framehook segfault on indicate (closes issue #19215) Reported by: irroot Patches: framehook_indicate.patch uploaded by irroot (license 52) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316335 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	37aa52fd78	Merged revisions 316265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316265 \| russell \| 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) \| 5 lines Fix a bunch of compiler warnings generated by gcc 4.6.0. Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316293 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	a52395aaee	Merged revisions 316206 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316206 \| seanbright \| 2011-05-03 14:17:36 -0400 (Tue, 03 May 2011) \| 8 lines If we aren't interested in events, don't generate the FullyBooted event on AMI login. (closes issue #19089) Reported by: bklang Patches: issue19089-1.8-r316204.patch uploaded by seanbright (license 71) Tested by: seanbright ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	237d47b010	Clears exception flag during ast_read when func_jitterbuffer is enabled git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	98f94daf88	Merged revisions 315810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315810 \| russell \| 2011-04-27 10:55:48 -0500 (Wed, 27 Apr 2011) \| 2 lines Set the copyright year to 2011 in the startup message. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315811 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	8d2a71877a	Merged revisions 315644 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315644 \| twilson \| 2011-04-26 14:39:01 -0700 (Tue, 26 Apr 2011) \| 32 lines Merged revisions 315643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315643 \| twilson \| 2011-04-26 14:27:44 -0700 (Tue, 26 Apr 2011) \| 25 lines Merged revisions 315596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315596 \| twilson \| 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) \| 18 lines Allow transfer loops without allowing forwarding loops We try to avoid the situation where two phones may be forwarded to each other causing an infinite loop by storing each dialed interface in a channel datastore and checking the list before dialing out. This works, but currently breaks situations like A calls B, A transfers B to C, B transfers C to A, and A transfers C to B. Since human interaction is happening here and not an automated forwarding loop, it should be allowed. This patch removes the dialed_interfaces datastore when a call is bridged (a suggestion from the brilliant mmichelson). If a call is being bridged, it should be safe to assume that we aren't stuck in a loop. Since we are now handling this is the bridge code, the previous attempts at handling it in app_dial and app_queue are removed. Review: https://reviewboard.asterisk.org/r/1195/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	24b6939496	Merged revisions 315645 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r315645 \| rmudgett \| 2011-04-26 17:14:31 -0500 (Tue, 26 Apr 2011) \| 21 lines The 'e' special extension fails to trigger in at least two cases. The 'e' extension is a fall back for the 'i', 't', or 'T' extensions if any of them do not exist. Many of the places the 'e' extension was supposed to be invoked fail because the priority was set wrong. There were two places where the 'e' extension was not even checked for fall back. * Made invoke the 'e' extension similarly to the previous 'i', 't', or 'T' extension check and added the 'e' extension as a fall back to the two missing locations. * Prioritized and optimized some hangup tests associated with the 'e' extension. (closes issue #19136) Reported by: kshumard Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1196/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315649 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	079e794b1c	Merged revisions 314628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314628 \| mnicholson \| 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) \| 27 lines 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/trunk@314666 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	7f23115ad2	New HD ConfBridge conferencing application. Includes a new highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8khz-192khz. Review: https://reviewboard.asterisk.org/r/1147/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	18d591cb48	Introduction of the JITTERBUFFER dialplan function. Review: https://reviewboard.asterisk.org/r/1157/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	632cd26411	Merged revisions 314358 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314358 \| twilson \| 2011-04-19 22:25:15 -0700 (Tue, 19 Apr 2011) \| 4 lines Initialize track pointer ast_reentrancy_init checks to see if it is NULL before initializing with calloc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	02821fc5b4	Merged revisions 314251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314251 \| lmadsen \| 2011-04-19 10:42:10 -0500 (Tue, 19 Apr 2011) \| 8 lines Use SSLv23_client_method instead of old SSLv2 only. (closes issue #19095) (closes issue #19138) Reported by: tzafrir Patches: no_ssl2.diff uploaded by tzafrir (license 46) Tested by: russell, chazzam ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314252 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	4b4549106b	Merged revisions 314017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314017 \| dvossel \| 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) \| 17 lines sip codec negotiation of dynamic rtp payloads error fix This patch fixes how chan_sip handles dynamic rtp payload types it does not understand. At the moment if a dynamic payload's mime type does not match one we understand, the payload does not get removed from our payload table. As a result of this, the payload is set to whatever dynamic codec we use internally for that payload number on outgoing INVITES. This is incorrect. This patch fixes this by properly checking the rtpmap set function's return code to make sure it was found. The function can return both -1 and -2 depending on the source of the mismatch. We were just checking -1 explicitly. Review: https://reviewboard.asterisk.org/r/1169/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314018 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	05ddffccc4	Merged revisions 313860 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313860 \| jrose \| 2011-04-15 10:08:05 -0500 (Fri, 15 Apr 2011) \| 17 lines Merged revisions 313859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r313859 \| jrose \| 2011-04-15 09:58:37 -0500 (Fri, 15 Apr 2011) \| 10 lines Fix a Tab Completion bug that occurs due to multiple matches on a substring. Makes word_match function in cli.c repeat a search for a command string until a proper match is found or the string is searched to the last point. (closes issue #17494) Reported by: ffossard Review: https://reviewboard.asterisk.org/r/1180/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ae2926b5d0	Add Device State Information CCSS for Generic Devices. Add Asterisk Device State information and callbacks to the Call Completion Supplemental Services for generic agents. There are currently not many devices that have native support for CCSS. Even as the devices become available there may be other reasons why one may choose to not take advantage of the native abilities and stick with the generic implementation. The generic implementation is quite capable and could be greatly enhanced by adding device state capabilities. A phone could then subscribe to the device state with a BLF key in conjunction with Asterisk hints. The advantages of the device state information would allow a single button to: request CCSS, cancel a CCSS request, and display the current state of a CCSS request. For example, you may have a single button that when not lit, there is no active CCSS request. When you press that button, the dialplan can query the DEVICE_STATE() associated with that caller to determine whether they should be calling CallCompletionRequest() or CallCompletionCancel(). If there is currently a pending request, then the dialplan would cancel it. This also has the advantage of showing the true state of a request, which is an asynchronous call, even when CallCompletionRequest() thinks it was successful. The actual request could ultimately fail. Once lit, further feedback can be provided to the caller about the current state of their request since it will be updated by the CCSS State Machine as appropriate. The DEVICE_STATE mapping is configurable since the BLF being used on a given phone type may vary. The idea is to allow some level of customization as to the phone's behavior. As an example, you may want the BLF key to go solid once you have requested a callback. You may then want the LED to blink (typically ringing) when either the callback is in process, which is a visual indication that the incoming call is the desired callback. You may want it to blink when the callee is ready but you are busy, giving you a visual indication that the target is available as you may want to get off the line so that the callback can be successful. Device state information is sent back via the ast_devstate_prov_add() callback for any generic CCSS device as it traverses through the state machine. You simply provide a map between CC_STATE values and the corresponding AST_DEVICE state values. You could then generate hints against these states similar to what is possible today with Custom Devstates or MeetMe states. For example, you may have an extension 3000 that is currently associated with device SIP/3000. You could then create a feature code for that extension that may look something like: exten => 823000,hint,ccss:sip/3000 You would then subscribe a BLF button to 823000 which would point to the dialplan that handled CCSS requests/cancels using the available DEVICE_STATE() information about ccss:sip/3000 to make the decision about what to do. (closes issue #18788) Reported by: p_lindheimer Patches: ccss.trunk.18788.patch uploaded by p lindheimer (license 558) Modified with final reviewboard comments. Tested by: p_lindheimer, loloski Review: https://reviewboard.asterisk.org/r/1105/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	c16d39ea83	Merged revisions 313588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313588 \| rmudgett \| 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) \| 55 lines Merged revisions 313579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313579 \| rmudgett \| 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) \| 48 lines Merged revisions 313545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313545 \| rmudgett \| 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) \| 41 lines Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	530afe7d97	Merged revisions 313366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r313366 \| rmudgett \| 2011-04-11 17:27:25 -0500 (Mon, 11 Apr 2011) \| 2 lines Added "Connected Line ID" and "Connected Line ID Name" to "core show channel" output. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	68dd87ef0d	Merged revisions 313048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313048 \| jrose \| 2011-04-07 08:35:33 -0500 (Thu, 07 Apr 2011) \| 16 lines Merged revisions 313047 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r313047 \| jrose \| 2011-04-07 08:23:01 -0500 (Thu, 07 Apr 2011) \| 9 lines Makes parking lots clear and rebuild properly when features reload is invoked from CLI Before, default parkinglot in context parkedcalls with ext 700 would always be present and when reload was invoked, the previous parkinglots would not be cleared. (closes issue #18801) Reported by: mickecarlsson Review: https://reviewboard.asterisk.org/r/1161/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	a77fd545ab	Merged revisions 312766 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312766 \| mnicholson \| 2011-04-05 09:14:50 -0500 (Tue, 05 Apr 2011) \| 22 lines 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/trunk@312767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	75594e6e4a	Merged revisions 312461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r312461 \| rmudgett \| 2011-04-01 16:31:39 -0500 (Fri, 01 Apr 2011) \| 25 lines CallCompletionRequest()/CallCompletionCancel() exit non-zero if fail. The CallCompletionRequest()/CallCompletionCancel() dialplan applications exit nonzero on normal failure conditions. The nonzero exit causes the dialplan to hangup immediately. The dialplan author has no opportunity to report success/failure to the user. * Made always return zero so the dialplan can continue. * Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively. Also documented the values set. * Reduced the warning about no core instance in CallCompletionCancel() to a debug message. It is a normal event and should not be output at the WARNING level. (closes issue #18763) Reported by: p_lindheimer Patches: ccss.patch uploaded by p lindheimer (license 558) Modified Tested by: p_lindheimer, rmudgett JIRA SWP-3042 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	846cfa0ef0	New Feature for chan_dahdi. 4 length pattern matching. In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns. The s ntax remains the same and the method used to track the pattern history will only change when using the length 4 patterns. (closes issue SWP-3250) Code: jrose rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	3731fd9ccc	Merged revisions 312286,312288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312286 \| tilghman \| 2011-04-01 05:44:33 -0500 (Fri, 01 Apr 2011) \| 2 lines Reload must react correctly against a possibly changed table, so dropping the conditional reload flag. ................ r312288 \| tilghman \| 2011-04-01 05:58:45 -0500 (Fri, 01 Apr 2011) \| 21 lines Merged revisions 312287 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312287 \| tilghman \| 2011-04-01 05:51:24 -0500 (Fri, 01 Apr 2011) \| 14 lines Merged revisions 312285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312285 \| tilghman \| 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011) \| 7 lines Found some leaking file descriptors while looking at ast_FD_SETSIZE dead code. (issue #18969) Reported by: oej Patches: 20110315__issue18969__14.diff.txt uploaded by tilghman (license 14) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312289 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	581bfad2f3	Merged revisions 311141 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r311141 \| mnicholson \| 2011-03-17 10:00:33 -0500 (Thu, 17 Mar 2011) \| 11 lines Merged revisions 311140 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r311140 \| mnicholson \| 2011-03-17 09:58:52 -0500 (Thu, 17 Mar 2011) \| 4 lines Don't write items to the manager socket twice. AST-2011-003 (closes issue 0018987) Reported by: ks-steven ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	4ae1cb9456	Merged revisions 310999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310999 \| twilson \| 2011-03-16 14:47:59 -0500 (Wed, 16 Mar 2011) \| 18 lines Merged revisions 310998 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310998 \| twilson \| 2011-03-16 14:46:36 -0500 (Wed, 16 Mar 2011) \| 11 lines Fix crash on fdopen failure See security advisory AST-2011-004 (closes issue #18845) Reported by: cmaj Patches: patch-main-tcptls-1.8.3-rc2-open-session-crash-take2.diff.txt uploaded by cmaj (license 830) patch-main-tcptls-1.8.3-rc2-open-session-crash-take3.diff.txt uploaded by cmaj (license 830) Tested by: cmaj, twilson ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311001 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	d0846ea207	Merged revisions 310993 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310993 \| twilson \| 2011-03-16 14:26:57 -0500 (Wed, 16 Mar 2011) \| 11 lines Merged revisions 310992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r310992 \| twilson \| 2011-03-16 14:23:03 -0500 (Wed, 16 Mar 2011) \| 4 lines Don't keep trying to write to a closed connection See security advisory AST-2011-003. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	d958ca6953	Merged revisions 310902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310902 \| twilson \| 2011-03-16 12:19:57 -0500 (Wed, 16 Mar 2011) \| 43 lines Merged revisions 310889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310889 \| twilson \| 2011-03-16 12:03:27 -0500 (Wed, 16 Mar 2011) \| 36 lines Merged revisions 310888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310888 \| twilson \| 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011) \| 29 lines Don't delay DTMF in core bridge while listening for DTMF features This patch is mostly the work of Olle Johansson. I did some cleanup and added the silence generating code if transmit_silence is set. When a channel listens for DTMF in the core bridge, the outbound DTMF is not sent until we have received DTMF_END. For a long DTMF, this is a disaster. We send 4 seconds of DTMF to Asterisk, which sends no audio for those 4 seconds. Some products see this delay and the time skew on RTP packets that results and start ignoring the audio that is sent afterward. With this change, the DTMF_BEGIN frame is inspected and checked. If it matches a feature code, we wait for DTMF_END and activate the feature as before. If transmit_silence=yes in asterisk.conf, silence is sent if we paritally match a multi-digit feature. If it doesn't match a feature, the frame is forwarded along with the DTMF_END without delay. By doing it this way, DTMF is not delayed. (closes issue #15642) Reported by: jasonshugart Patches: issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license 396) Tested by: globalnetinc, jde (closes issue #16625) Reported by: sharvanek Review: https://reviewboard.asterisk.org/r/1092/ Review: https://reviewboard.asterisk.org/r/1125/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310941 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	858c11f075	Merged revisions 310781 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310781 \| alecdavis \| 2011-03-15 14:00:55 +1300 (Tue, 15 Mar 2011) \| 10 lines core show locks: display ThreadID in hexadecimal Allow easier cross referencing of thread ID's with GDB backtraces (closes issue #18968) Reported by: alecdavis Patches: bug18968.diff.txt uploaded by alecdavis (license 585) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	de7280fc7d	Merged revisions 310636 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r310636 \| rmudgett \| 2011-03-14 11:50:59 -0500 (Mon, 14 Mar 2011) \| 39 lines Merged revisions 310635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310635 \| rmudgett \| 2011-03-14 11:47:54 -0500 (Mon, 14 Mar 2011) \| 32 lines Merged revisions 310633 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310633 \| rmudgett \| 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011) \| 25 lines "Caller*ID failed checksum" on Wildcard TDM2400P and TDM410 The last character in the caller id message is getting a framing error. The checksum is the last character in the message. A framing error in the checksum could be because: 1) The sender did not send a full stop bit. 2) The sender cut off the FSK carrier too soon. 3) The sender opted to send zero of the specified zero to 10 trailing mark bits and round-off errors in the code resulted in the code not being where it thought it was in the demodulated bit stream. Bit 8 of 'b' is set when parity error. Bit 9 of 'b' is set when framing error. Made ignore the framing and parity error bits if the errored character is the checksum. We can tolerate a framing/parity error there. The checksum character validates the message. (closes issue #18474) Reported by: nivek Patches: callerid.c.1.patch uploaded by nivek (license 636) (with modifications) Tested by: nivek ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	16c5dda8ab	Fixes null reference bug introduced by audio hook changes that affects various OS distributions. Thanks David. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310547 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	6e36042f64	Mix Monitor: Now with r and t options. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310373 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	c7c0664bc4	Merged revisions 310287 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310287 \| alecdavis \| 2011-03-11 19:47:44 +1300 (Fri, 11 Mar 2011) \| 17 lines remote_bridge_loop: prevent segfault when after transfer of IAX2 of DAHDI call If the channel condition is one of the following after breaking out of the loop, don't try to update_peer (where x = 0/1) 1). ZOMBIE 2). cx->tech_pvt != pvtx 3). gluex != ast_rtp_instance_get_glue(cx->tech->type)) (closes issue #18781) Reported by: alecdavis Patches: bug18781.diff3.txt uploaded by alecdavis (license 585) Tested by: alecdavis, ZX81 Review: https://reviewboard.asterisk.org/r/1128/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	254092f8f6	Merged revisions 310240 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310240 \| twilson \| 2011-03-10 10:05:45 -0600 (Thu, 10 Mar 2011) \| 13 lines Add \r\n to remaining http headers passed to ast_http_send r309204 changed the behavior of ast_http_send. It now requires headers to be passed with trailing \r\n. This change updates the remaining instances in the code that did not pass the \r\n. (closes issue #18186) Reported by: nivaldomjunior Patches: res_phoneprov.c.diff uploaded by lathama (license 1028) manager.diff.txt uploaded by twilson (license 396) Tested by: lathama ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	6de1332214	Merged revisions 309808 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309808 \| tilghman \| 2011-03-06 18:54:42 -0600 (Sun, 06 Mar 2011) \| 14 lines Merged revisions 309251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309251 \| tilghman \| 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) \| 7 lines Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround. Not surprisingly, the workaround was exactly the same code as was provided by the Flex maintainers, albeit in two different places, in different macros. This should fix the FreeBSD builds, which have an older version of Flex. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	798212c828	Merged revisions 309678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309678 \| tilghman \| 2011-03-05 04:29:30 -0600 (Sat, 05 Mar 2011) \| 14 lines Merged revisions 309677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309677 \| tilghman \| 2011-03-05 04:28:24 -0600 (Sat, 05 Mar 2011) \| 7 lines Missed part of the conversion when we started passing ppid to astcanary. (closes issue #18850) Reported by: viraptor Patches: canary_ppid.patch uploaded by viraptor (license 543) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309679 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	a7bfa10472	Add HangupRequest manager event, to specify when/where a channel gets hung up. (closes issue #18226) Reported by: clegall_proformatique Patches: asterisk_1.8_293157_hanguprequests.svn.patch uploaded by clegall proformatique (license 1139) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309300 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	dc616cfe2c	Merged revisions 309204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309204 \| qwell \| 2011-03-01 16:25:44 -0600 (Tue, 01 Mar 2011) \| 7 lines Fix consistency of CRLFs on HTTP headers that get sent out. (closes issue #18186) Reported by: nivaldomjunior Patches: 18186-httpheadernewline.diff uploaded by qwell (license 4) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309209 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	e5dc4c2d8e	Merged revisions 309035 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r309035 \| tilghman \| 2011-02-28 05:10:28 -0600 (Mon, 28 Feb 2011) \| 15 lines Merged revisions 309033-309034 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r309033 \| tilghman \| 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011) \| 4 lines A later version of flex already includes the fwrite workaround code, which if used twice causes a compilation error. Detect whether Flex will compile without the workaround; if so, suppress our workaround code. ........ r309034 \| tilghman \| 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) \| 2 lines Clarify meaning, removing double negative (stupid!) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	642d6c306c	Merged revisions 308903 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308903 \| rmudgett \| 2011-02-24 15:38:41 -0600 (Thu, 24 Feb 2011) \| 9 lines Invalid read in ast_channel_set_caller_event(). Valgrind reported that ast_channel_set_caller_event() was reading data from a freed buffer when using the pre_set structure. Rearange things to pre-calculate the name and number pointer before updating the caller party structure to see if the name or number was changed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	f88ef55d74	Merged revisions 308815 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308815 \| twilson \| 2011-02-24 11:57:18 -0600 (Thu, 24 Feb 2011) \| 26 lines Merged revisions 308814 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308814 \| twilson \| 2011-02-24 11:54:49 -0600 (Thu, 24 Feb 2011) \| 19 lines Merged revisions 308813 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308813 \| twilson \| 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011) \| 12 lines Don't broadcast FullyBooted to every AMI connection The FullyBooted event should not be sent to every AMI connection every time someone connects via AMI. It should only be sent to the user who just connected. (closes issue #18168) Reported by: FeyFre Patches: bug0018168.patch uploaded by FeyFre (license 1142) Tested by: FeyFre, twilson ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	0658c5ec2e	Merged revisions 308723 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308723 \| mnicholson \| 2011-02-24 09:06:14 -0600 (Thu, 24 Feb 2011) \| 16 lines Merged revisions 308722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308722 \| mnicholson \| 2011-02-24 08:59:41 -0600 (Thu, 24 Feb 2011) \| 9 lines Merged revisions 308721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308721 \| mnicholson \| 2011-02-24 08:54:56 -0600 (Thu, 24 Feb 2011) \| 2 lines silence gcc 4.2 compiler warning ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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3232 Commits (bac5a51e2121a63549941f3a9ce7f7a4c50454ca)