asterisk

Commit Graph

Author	SHA1	Message	Date
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	14 years ago
Jonathan Rose	1b57da8673	Adds STRREPLACE function Adds a new STRREPLACe function to func_strings.c that allows users to search and replace against a variable in the dialplan. (closes issue #18023) Reported by: wdoekes Review: https://reviewboard.asterisk.org/r/1219/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320040 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 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	14 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	14 years ago
Jonathan Rose	f90bc95f0d	Merged revisions 319938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319938 \| jrose \| 2011-05-20 08:28:24 -0500 (Fri, 20 May 2011) \| 12 lines 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/trunk@319939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 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	14 years ago
Mark Murawki	45818de137	Merged revisions 319812 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319812 \| markm \| 2011-05-19 13:59:01 -0400 (Thu, 19 May 2011) \| 9 lines In cel_odbc, an uninitialized RWLIST is attempted to be locked. Added INIT and DESTROY for the RWLIST odbc_tables (closes issue #19331) Reported by: kobaz Patches: odbc_cel.patch uploaded by kobaz (license 834) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319813 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 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	14 years ago
Terry Wilson	573108e63c	Merged revisions 319654 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319654 \| twilson \| 2011-05-18 16:15:58 -0700 (Wed, 18 May 2011) \| 22 lines Merged revisions 319653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r319653 \| twilson \| 2011-05-18 16:11:57 -0700 (Wed, 18 May 2011) \| 15 lines Merged revisions 319652 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r319652 \| twilson \| 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011) \| 8 lines Make sure everyone gets an unhold when a transfer succeeds Some phones, like the Snom phones, send a hold to the transfer target after before sending the REFER. We need to make sure that we unhold the parties that are being connected after the masquerade. If Local channels with the /nm option are used when dialing the parties, hold music would still be playing on the transfer target, even after being connected with the transferee. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	99aaceacad	Merged revisions 319552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319552 \| twilson \| 2011-05-18 13:22:36 -0700 (Wed, 18 May 2011) \| 11 lines Unbreak the storing of registrations for restart The fix for issue 18882 broke retrieving non-realtime peers from the ast_db on restart/reload. This patch tries to unbreak things while leaving the intent of the original fix intact. (closes issue #19318) Reported by: remiq Patches: diff.txt uploaded by twilson (license 396) Tested by: lmadsen, remiq ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319564 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	2760e05dea	Merged revisions 319529 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319529 \| twilson \| 2011-05-18 13:05:34 -0700 (Wed, 18 May 2011) \| 24 lines Merged revisions 319528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r319528 \| twilson \| 2011-05-18 13:02:06 -0700 (Wed, 18 May 2011) \| 17 lines Merged revisions 319527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r319527 \| twilson \| 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011) \| 10 lines Fix app_dial ring groups Revert part of r315643. We need to remove the datastore here as well. The code in bridging code will catch anything that app_dial might miss. (closes issue #19311) Reported by: mspuhler Patches: issue_19311_no_answer.diff uploaded by elguero (license 37) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	3e25489e4c	Merged revisions 319469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319469 \| rmudgett \| 2011-05-17 16:57:56 -0500 (Tue, 17 May 2011) \| 22 lines Merged revision 319468 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r319468 \| rmudgett \| 2011-05-17 16:49:31 -0500 (Tue, 17 May 2011) \| 15 lines The mISDN HDLC mode is prevented on dialed channels. The use of mISDN HDLC mode is prevented if the mISDN dial technology option 'h1' is used when config option astdtmf=yes. There is a bug in channels/misdn/isdn_lib.c which prevents the use of HDLC mode. Instead of setting the channel to HDLC mode it is set to transparent(no dsp, no hdlc), although hdlc is not "no hdlc". I.e the logging message is correct, but the if condition is not. Make check the nodsp and hdlc flags. JIRA ABE-2787 JIRA SWP-3437 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319471 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	0bb90a372e	Remove extraneous line variables. The vars were either explicitly or implicitly not used. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319470 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	5257a915a8	Option needed for Q931_IE_TIME_DATE to be optional in CONNECT message. The NEC SV8300 rejects the Q931_IE_TIME_DATE for Q.SIG. Add option to specify if and how much of the current time is put in Q931_IE_TIME_DATE. * Send date/time ie never. * Send date/time ie date only. * Send date/time ie date and hour. * Send date/time ie date, hour, and minute. * Send date/time ie date, hour, minute, and second. * Send date/time ie default: Libpri will send date and hhmm only when in NT PTMP mode to support ISDN phones. (closes issue #19221) Reported by: kenner JIRA SWP-3396 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	380e0e3e2d	Merged revisions 319367 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319367 \| lmadsen \| 2011-05-17 07:53:50 -0500 (Tue, 17 May 2011) \| 10 lines Don't create [general] voicemail context when using users.conf Prior to this patch, app_voicemail would create a [general] context when parsing users.conf. (closes issue #18891) Reported by: pdugas Patches: app_voicemail-ignore-general.patch uploaded by pdugas (license 1222) app_voicemail-ignore-general-style-guidelines.patch uploaded by seanbright (license 71) Tested by: pdugas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	7f37d162e7	Merged revisions 319365 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319365 \| lmadsen \| 2011-05-17 07:39:37 -0500 (Tue, 17 May 2011) \| 6 lines Make Debian init script lsb compliant (closes issue #18896) Reported by: manwe Patches: debian_init_lsb.patch uploaded by manwe (license 1223) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319366 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	069b70c522	Fix up skinny hints. Probably haven't been working for a couple of years. May still need some more love, but they are now working, both as a hint device and monitoring a hint. Changes centre around the long ago change to remove the requirement for a device name in a skinny line, and changes to the transmit_* functions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 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	14 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	14 years ago
Terry Wilson	d34d46a16e	Merged revisions 319204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319204 \| twilson \| 2011-05-16 13:17:43 -0500 (Mon, 16 May 2011) \| 11 lines Merged revisions 319202 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r319202 \| twilson \| 2011-05-16 11:00:21 -0700 (Mon, 16 May 2011) \| 4 lines Unlink a peer from peers_by_ip when expiring a registration Review: https://reviewboard.asterisk.org/r/1218/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	980d896bde	Merged revisions 319145 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319145 \| dvossel \| 2011-05-16 10:57:26 -0500 (Mon, 16 May 2011) \| 9 lines Merged revisions 319144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r319144 \| dvossel \| 2011-05-16 10:56:16 -0500 (Mon, 16 May 2011) \| 2 lines Fixes issue with peer ref-counting during handle_request_subscribe. (closes issue #19293) Reported by: irroot ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	8e719c62b0	Merged revisions 319142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319142 \| mnicholson \| 2011-05-16 10:53:26 -0500 (Mon, 16 May 2011) \| 8 lines Make sure tcptls_session exists before dereferencing it. (closes issue #19192) Reported by: stknob Patches: 10-tcptls-unreachable-peer-segfault.patch uploaded by Chainsaw (license 723) Tested by: vois, Chainsaw ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	32d43ebe19	When a error in T.38 negotiation happens or its rejected on a channel the state of the channel reverts to unknown this should be rejected. this is important for negotiating T.38 gateway see #13405 This patch adds a option T38_REJECTED that behaves as T38_DISABLED except it reports state rejected. Trivial Change to res_fax to honnor UNAVAILABLE and REJECTED states. (closes issue #18889) Reported by: irroot Tested by: irroot, darkbasic, mnicholson Review: https://reviewboard.asterisk.org/r/1115 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319087 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	938290cf0d	Merged revisions 319085 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319085 \| pabelanger \| 2011-05-16 10:35:21 -0400 (Mon, 16 May 2011) \| 10 lines Support gmime-2.4 (closes issue #18863) Reported by: tzafrir Patches: gmime-2.4-18.diff uploaded by tzafrir (license 46) Tested by: tzafrir Review: https://reviewboard.asterisk.org/r/1213/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319086 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	9c73ece406	Merged revisions 319083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319083 \| dvossel \| 2011-05-16 09:26:33 -0500 (Mon, 16 May 2011) \| 5 lines Fixes Big Endian build issue. (closes issue #19298) Reported by: tzafrir ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	969a317d81	Add activatesub and dialandactivate sub. When called, activatesub first cleans up the active sub and then handles the sub passed. dialandactivatesub first sets sub->exten and then calls activatesub. Revise handle_offhook to utilise the callid sent to chan_skinny. Some other minor fixes especially around d->hookstate (which still needs some more work). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 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	14 years ago
Brett Bryant	085b7b212a	Merged revisions 318919 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318919 \| bbryant \| 2011-05-13 14:04:50 -0400 (Fri, 13 May 2011) \| 10 lines This patch fixes an issue with SRTP which makes HOLD/UNHOLD impossible when too much time has passed between sending audio. (closes issue #18206) Reported by: bernhardsi Patches: res_srtp_unhold.patch uploaded by bernhards (license 1138) Tested by: bernhards, notthematrix ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Brett Bryant	547490144c	Merged revisions 318917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318917 \| bbryant \| 2011-05-13 13:56:04 -0400 (Fri, 13 May 2011) \| 11 lines This patch allows TCP peers into the ast_db where they were previously restricted. (closes issue #18882) Reported by: cmaj Patches: patch-chan_sip-1.8.3-rc2-allow-tcp-peer-store-db-and-readonly-rt-backend.diff.txt uploaded by cmaj (license 830) Tested by: cmaj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318918 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 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	14 years ago
Damien Wedhorn	54bb8a0ca8	Move exten used for dialing from device to subchannel. There were some issues where if a simple switch was cancelled and a new switch started before the first had timed out where the d->exten would be used for both subchannels. This was bad leading to possible invalid extensions if some digits had been entered in the abandoned simple switch and the second one was completed before the first timed out, or the second would be cancelled because d->exten would be set to nothing on the time out of the first. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	9066db4329	Merged revisions 318720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318720 \| mnicholson \| 2011-05-12 18:35:51 -0500 (Thu, 12 May 2011) \| 4 lines Handle ipv6 addresses in the sent-by Via: field. This change fixes a regression in via header parsing and ipv6 handling. (closes issue #18951) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	1ad49f46ce	Merged revisions 318783 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318783 \| rmudgett \| 2011-05-12 20:47:05 -0500 (Thu, 12 May 2011) \| 14 lines PRI early media won't ring. And another way to pass early media. Don't indicate that there is inband information present, just assume that the B channel is connected. * Restore clearing the dialing flag Rx squelch unconditionally when a PROCEEDING message comes in. (closes issue #19268) Reported by: tbsky Patches: issue19268_v1.8.patch uploaded by rmudgett (license 664) Tested by: tbsky ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318784 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 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	14 years ago
Damien Wedhorn	c37c017781	Consolidate setsubstate_* into setsubstate and use a switch. Consolidate the functions and add some debugging info. Allows to be able to set a substate without explicitly knowing what the state is. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Damien Wedhorn	bdbb3a506f	Add setsubstate_onhook. Add the setsubstate_onhook to complete the initial substate handling procedures. Added dumpsub(sub, forcehangup) which is the common way of calling setsubstate_onhook. Dumpsub attempts to activate another sub after setting the current one onhook. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	475c264bd2	Merged revisions 318550 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318550 \| twilson \| 2011-05-11 13:47:33 -0500 (Wed, 11 May 2011) \| 2 lines Comment out the REF_DEBUG that slipped in during debugging ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	da4016544e	Merged revisions 318549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318549 \| twilson \| 2011-05-11 13:39:48 -0500 (Wed, 11 May 2011) \| 27 lines Merged revisions 318548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r318548 \| twilson \| 2011-05-11 12:15:39 -0500 (Wed, 11 May 2011) \| 19 lines Clean up several chan_sip reference leaks Several situations in the code could lead to peers or sip_pvt references being leaked. This would cause RTP ports to never be destroyed (leading to exhaustion of all available RTP ports) and memory leaks. The original patch for this issue from rgagnon was the result of an obscene amount of testing and hard work, for which I am very grateful. I did some cleanup and added a few additional refcount fixes that I found. (closes issue #17255) Reported by: kvveltho Patches: tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff uploaded by rgagnon (license 1202) Tested by: rgagnon, twilson, wdoekes, loloski Review: https://reviewboard.asterisk.org/r/1101/ Review: https://reviewboard.asterisk.org/r/1207/ Review: https://reviewboard.asterisk.org/r/1210/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d1e27b1026	Merged revisions 318499 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318499 \| rmudgett \| 2011-05-10 18:41:08 -0500 (Tue, 10 May 2011) \| 15 lines Unable to pickup DAHDI/PRI call because call state is reported as DIALING. The channel state is not updated to RINGING when an ALERTING message is received. Regression caused when sig_pri.c (also sig_ss7.c) extracted from chan_dahdi.c. * Added missing channel state update to RINGING when the AST_CONTROL_RINGING frame is queued for ISDN and SS7. (closes issue #19257) Reported by: alecdavis Patches: issue19257_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318500 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	0ccfc8609a	Merged revisions 318436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318436 \| russell \| 2011-05-10 10:13:16 -0500 (Tue, 10 May 2011) \| 2 lines chan_iax2: change LOG_NOTICE to LOG_DEBUG in iax2_read(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	07b3742ad2	Merged revisions 318337 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318337 \| twilson \| 2011-05-09 15:23:15 -0500 (Mon, 09 May 2011) \| 18 lines Merged revisions 318331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r318331 \| twilson \| 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011) \| 12 lines Don't offer video to directmedia callee unless caller offered it as well Make sure that when directmedia is enabled, that video is not offered to the callee even if it supports it. p->vrtp will not exist since the caller didn't offer video. (closes issue #19195) Reported by: one47 Patches: sip_cant_add_video_rtp uploaded by one47 (license 23) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0886204011	Merged revisions 318351 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318351 \| rmudgett \| 2011-05-09 18:15:32 -0500 (Mon, 09 May 2011) \| 6 lines Remove references to res_features and its export file. The contents of res/res_features.c was moved to into main/features.c awhile ago. There is no longer any need for the res/Makefile to reference res_features or the res_features linker exports file to exist. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318352 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 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	14 years ago
David Vossel	4c35291c6b	Merged revisions 318233 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318233 \| dvossel \| 2011-05-09 12:09:55 -0500 (Mon, 09 May 2011) \| 14 lines Merged revisions 318230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r318230 \| dvossel \| 2011-05-09 11:51:45 -0500 (Mon, 09 May 2011) \| 7 lines Fixes cases where sip_set_rtp_peer can return too early during media path reset. (closes issue #19225) Reported by: one47 Patches: sip_set_rtp_peer.patch uploaded by one47 (license 23) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d7c94e1e04	Merged revisions 318231 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318231 \| rmudgett \| 2011-05-09 11:57:18 -0500 (Mon, 09 May 2011) \| 41 lines Don't get early media for ISDN on outgoing calls. It looks to be a long-standing misinterpretation of the progress indicator ie values: 1 - Call is not end-to-end ISDN; further call progress information may be available in-band. 8 - In-band information or an appropriate pattern is now available. Only value 8 is handled by chan_dahdi/sig_pri. The 1 value is not handled as early media probably because the meaning of the second half of it's description was overlooked. * Test to see if either PRI_PROG_CALL_NOT_E2E_ISDN(1) or PRI_PROG_INBAND_AVAILABLE(8) bits are set to open the media path. (closes issue #18868) Reported by: isrl Patches: issue18868_19246_v1.8.patch uploaded by rmudgett (license 664) Tested by: satish_lx .......... No inband progress on PRI_EVENT_RINGING even if inband flag set. My ISDN-PRI provider sends an ALERTING with "Inband information or appropriate pattern now available", but Asterisk only generates and passes the RING to the SIP extension, not the inband message. Unfortunately, the inband message is not a ringback tone but a prompt that says the number is not in service. The SIP extension then hears two rings and the call is hungup which confuses the caller. * Post an AST_CONTROL_PROGRESS as well as opening the media path if inband audio is indicated with an ALERTING message. (closes issue #19246) Reported by: cristiandimache Patches: issue19246_v1.8.patch uploaded by rmudgett (license 664) Tested by: cristiandimache ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318232 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 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	14 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	14 years ago
Jonathan Rose	6eb9d7e1b5	Merged revisions 318148 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318148 \| jrose \| 2011-05-09 09:18:14 -0500 (Mon, 09 May 2011) \| 4 lines 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/trunk@318162 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 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	14 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	14 years ago

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