asterisk

Commit Graph

Author	SHA1	Message	Date
David Vossel	08460fc094	Fixes bug in chan_sip where nativeformats are not set correctly. The nativeformats field was being overwritten when it should have been appended too. This caused some format capabilities to be lost briefly and some log warnings to be output. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	4f57a3bb7c	Merged revisions 306979 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306979 \| twilson \| 2011-02-08 12:18:08 -0800 (Tue, 08 Feb 2011) \| 16 lines Merged revisions 306973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306973 \| twilson \| 2011-02-08 12:14:09 -0800 (Tue, 08 Feb 2011) \| 9 lines Merged revisions 306972 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306972 \| twilson \| 2011-02-08 12:05:13 -0800 (Tue, 08 Feb 2011) \| 2 lines Fix comparison for REFER Replaces tags with pedantic=yes ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307061 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	209a39f4b0	Use correct conditional for MCID send. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	49feb747ba	Pass a MCID request to the bridged channel. Pass a MCID request to the bridged channel so the bridged channel can send it to the network. The ability to send the MCID request on an ISDN span is enabled with the new chan_dahdi.conf mcid_send option. JIRA SWP-2845 JIRA ABE-2736 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	a974d1a4ce	Merged revisions 306619 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306619 \| twilson \| 2011-02-07 14:15:27 -0800 (Mon, 07 Feb 2011) \| 24 lines Merged revisions 306618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306618 \| twilson \| 2011-02-07 13:59:54 -0800 (Mon, 07 Feb 2011) \| 17 lines Merged revisions 306617 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306617 \| twilson \| 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011) \| 10 lines Don't allow a REFER w/replaces to replace its own dialog Asterisk currently accepts a REFER with a Refer-To with an embedded Replaces header that matches the dialog of the REFER. This would be a situation like A calls B, A calls C, A transfers B to A, which is just silly. This patch makes the transfer fail instead of making Asterisk freak out and forget to hang other channels up. Review: https://reviewboard.asterisk.org/r/1093/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	2db3c9e058	Fixes use of ast_format_cap_append where ast_format_cap_copy is necessary. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306541 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	484f9bec0a	Ignore voice frames in chan_dahdi native bridging. Hardware is handling them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a8aeb04a9f	Add ISDN display ie text handling options to chan_dahdi.conf. The display ie handling can be controlled independently in the send and receive directions with the following options: * Block display text data. * Use display text in SETUP/CONNECT messages for name. * Use display text for COLP name updates (FACILITY/NOTIFY as appropriate). * Pass arbitrary display text during a call. Sent in INFORMATION messages. Received from any message that the display text was not used as a name. If the display options are not set then the options default to legacy behavior. The arbitrary display text is exchanged between bridged channels using the AST_FRAME_TEXT frame type. To send display text from the dialplan use the SendText() application when the arbitrary display text option is enabled. JIRA SWP-2688 JIRA ABE-2693 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306396 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	3556e4c2d4	Replace ast_log(LOG_DEBUG, ...) with ast_debug() (closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	285d953fdf	Merged revisions 306215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306215 \| jpeeler \| 2011-02-03 17:49:28 -0600 (Thu, 03 Feb 2011) \| 20 lines Fix SIP deadlock involving state changes. Once again a call to pbx_builtin_getvar_helper (and pbx_builtin_setvar_helper) has caused locking problems. Both of these functions lock the channel when the channel argument is passed in! In this case, the suspected problem (the backtrace makes it impossible to tell) was the private being locked in sip_set_rtp_peer and then: transmit_reinvite_with_sdp try_suggested_sip_codec pbx_builtin_getvar_helper (Traced to verify that the fix was only required in 1.8 and later.) (closes issue #18491) Reported by: cmaj Patches: chan_sip_fix_deadlocks_bug_18491.txt uploaded by cmaj (license 830) Tested by: cmaj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	36da6b6286	Merged revisions 306127 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306127 \| twilson \| 2011-02-03 13:03:26 -0800 (Thu, 03 Feb 2011) \| 23 lines Merged revisions 306126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306126 \| twilson \| 2011-02-03 12:56:00 -0800 (Thu, 03 Feb 2011) \| 16 lines Merged revisions 306119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306119 \| twilson \| 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011) \| 9 lines Set hangup cause in local_hangup When a call involves a local channel (like SIP -> Local -> SIP), the hangup cause was not being set. This resulted in SIP channels sometimes getting a 503 error instead of a 486 when the far side sent a busy. In Asterisk 1.8+ this also can cause issues with CCSS that involve a local channel. This patch sets the hangupcause for one side of the local channel to the other in local_hangup for outbound calls. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306128 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	c26c190711	Asterisk media architecture conversion - no more format bitfields This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	f71322f239	Merged revisions 305923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305923 \| rmudgett \| 2011-02-02 18:24:40 -0600 (Wed, 02 Feb 2011) \| 24 lines Merged revisions 305889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305889 \| rmudgett \| 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) \| 17 lines Merged revisions 305888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305888 \| rmudgett \| 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) \| 8 lines Minor AST_FRAME_TEXT related issues. * Include the null terminator in the buffer length. When the frame is queued it is copied. If the null terminator is not part of the frame buffer length, the receiver could see garbage appended onto it. * Add channel lock protection with ast_sendtext(). * Fixed AMI SendText action ast_sendtext() return value check. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Latham	175dd0ebf6	Replace link to old doc with new wiki page. Link to https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	76cfbf7817	Merged revisions 305692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305692 \| qwell \| 2011-02-01 16:48:16 -0600 (Tue, 01 Feb 2011) \| 7 lines Reverse sense of an error test when reading from astdb. (closes issue #18545) Reported by: jcovert Patches: chan_iax2.c.patch uploaded by jcovert (license 551) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	44349de2df	Merged revisions 305343 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305343 \| rmudgett \| 2011-01-31 18:01:09 -0600 (Mon, 31 Jan 2011) \| 21 lines Merged revisions 305342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305342 \| rmudgett \| 2011-01-31 17:50:10 -0600 (Mon, 31 Jan 2011) \| 14 lines Merged revisions 305341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305341 \| rmudgett \| 2011-01-31 17:45:58 -0600 (Mon, 31 Jan 2011) \| 7 lines Obtain the pri lock for PRI queue counters. Need to obtain the pri lock when calling pri_dump_info_str() to avoid a reentrancy problem when calculating the Q.921 Q count statistic. JIRA AST-484 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	6908539952	Merged revisions 305254 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305254 \| qwell \| 2011-01-31 17:07:00 -0600 (Mon, 31 Jan 2011) \| 24 lines Merged revisions 305253 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305253 \| qwell \| 2011-01-31 16:59:34 -0600 (Mon, 31 Jan 2011) \| 17 lines Merged revisions 305252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305252 \| qwell \| 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) \| 10 lines Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers already had code to prevent this. The attempt that app_dial was making to prevent it was not correct, so I fixed that. (closes issue #18371) Reported by: gbour Patches: 18371.patch uploaded by gbour (license 1162) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ecdbb3d1d9	Merged from revision 304341 https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r304341 \| rmudgett \| 2011-01-26 16:38:39 -0600 (Wed, 26 Jan 2011) \| 7 lines Add connected line chan_dahdi.conf pricpndialplan option. * Added from_channel value to prilocaldialplan option. JIRA ABE-2731 JIRA SWP-2842 .......... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	48a9694ed0	Merged revisions 304245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304245 \| mnicholson \| 2011-01-26 14:43:27 -0600 (Wed, 26 Jan 2011) \| 20 lines Merged revisions 304244 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304244 \| mnicholson \| 2011-01-26 14:42:16 -0600 (Wed, 26 Jan 2011) \| 13 lines Merged revisions 304241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304241 \| mnicholson \| 2011-01-26 14:38:22 -0600 (Wed, 26 Jan 2011) \| 6 lines This patch modifies chan_sip to route responses to the address the request came from. It also modifies chan_sip to respect the maddr parameter in the Via header. ABE-2664 Review: https://reviewboard.asterisk.org/r/1059/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304246 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	15605be78b	Merged revisions 304150 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304150 \| rmudgett \| 2011-01-26 13:39:35 -0600 (Wed, 26 Jan 2011) \| 16 lines Merged revisions 304149 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304149 \| rmudgett \| 2011-01-26 13:38:38 -0600 (Wed, 26 Jan 2011) \| 9 lines Merged revisions 304148 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r304148 \| rmudgett \| 2011-01-26 13:23:46 -0600 (Wed, 26 Jan 2011) \| 2 lines Update documentation for DAHDISendCallreroutingFacility() application. .......... ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	cd9221d2f6	Merged revisions 303962 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303962 \| twilson \| 2011-01-25 16:09:01 -0600 (Tue, 25 Jan 2011) \| 30 lines Merged revisions 303960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303960 \| twilson \| 2011-01-25 16:02:42 -0600 (Tue, 25 Jan 2011) \| 23 lines Merged revisions 303906 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303906 \| twilson \| 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011) \| 16 lines Guard against retransmitting BYEs indefinitely In the case of an attended transfer (A calls B, A atxfers to C) where A becomes unreachable before replying to Asterisk's BYE, Asterisk can sometimes retransmit the BYE indefinitely. This is because __sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0], SIP_ALREADYGONE and will then transmit a BYE. When this BYE times out, it will not ever be marked as ALREADYGONE, so when __sip_autodestruct is called again, we end up starting the cycle over. This patch adds a call to sip_alreadygone(pkt->owner) in retrans_pkt in the case of a BYE that has timed out. This should prevent Asterisk from trying to transmit new BYE messages in the future. Review: https://reviewboard.asterisk.org/r/1077/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	50c432324b	Merged revisions 303860 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303860 \| tilghman \| 2011-01-25 12:55:27 -0600 (Tue, 25 Jan 2011) \| 12 lines Merged revisions 303858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r303858 \| tilghman \| 2011-01-25 12:41:26 -0600 (Tue, 25 Jan 2011) \| 5 lines Fix "sip show user <tab>", so that it actually shows results, instead of just completing the last entry. (closes issue #16675) Reported by: pj ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303861 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	7889af7cab	Merged revisions 303771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303771 \| rmudgett \| 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) \| 54 lines Merged revisions 303769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 \| rmudgett \| 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) \| 47 lines Merged revisions 303765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 \| rmudgett \| 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) \| 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303772 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	e706b5706e	According to section 19.1.2 of RFC 3261: For each component, the set of valid BNF expansions defines exactly which characters may appear unescaped. All other characters MUST be escaped. This patch modifies ast_uri_encode() to encode strings in line with this recommendation. This patch also adds an ast_escape_quoted() function which escapes '"' and '\' characters in quoted strings in accordance with section 25.1 of RFC 3261. The ast_uri_encode() function has also been modified to take an ast_flags struct describing the set of rules it should use when escaping characters to allow for it to escape SIP URIs in addition to HTTP URIs and other types of URIs or variations of those two URI types in the future. The ast_uri_decode() function has also been modified to accept an ast_flags struct describing the set of rules to use when decoding to enable decoding '+' as ' ' in legacy http URLs. The unit tests for these functions have also been updated. ABE-2705 Review: https://reviewboard.asterisk.org/r/1081/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	54f6c31a27	Merged revisions 303467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303467 \| qwell \| 2011-01-24 11:20:03 -0600 (Mon, 24 Jan 2011) \| 22 lines Merged revisions 303285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303285 \| qwell \| 2011-01-21 15:48:09 -0600 (Fri, 21 Jan 2011) \| 15 lines Merged revisions 303284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303284 \| qwell \| 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) \| 8 lines Reset configuration before parsing users.conf. Some values configured in chan_dahdi.conf were able to leak in to users.conf configuration. This was surprising users, and potentially setting non-sane "defaults". ASTNOW-125 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	95f5dc6644	Temporarily revert r303288 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303376 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	4272837ead	Merged revisions 303286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303286 \| qwell \| 2011-01-21 15:50:11 -0600 (Fri, 21 Jan 2011) \| 22 lines Merged revisions 303285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303285 \| qwell \| 2011-01-21 15:48:09 -0600 (Fri, 21 Jan 2011) \| 15 lines Merged revisions 303284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303284 \| qwell \| 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) \| 8 lines Reset configuration before parsing users.conf. Some values configured in chan_dahdi.conf were able to leak in to users.conf configuration. This was surprising users, and potentially setting non-sane "defaults". ASTNOW-125 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	06ac89965c	Merged revisions 302414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302414 \| seanbright \| 2011-01-19 10:45:17 -0500 (Wed, 19 Jan 2011) \| 7 lines Initialize an uninitialized variable. (closes issue #18640) Reported by: jcovert Patches: chan_sip.c.patch uploaded by jcovert (license 551) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302415 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	d42cb6fd1d	Merged revisions 302412 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302412 \| seanbright \| 2011-01-19 10:31:39 -0500 (Wed, 19 Jan 2011) \| 10 lines Use appropriate type for requested format in chan_local. We were passing and storing the requested format as an int instead of format_t resulting in truncation. (closes issue #18238) Reported by: whizemen Patches: 0018238_speex16.patch uploaded by whizemen (license 1143) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302413 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	785e3a1417	Merged revisions 302314 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302314 \| mnicholson \| 2011-01-18 15:43:21 -0600 (Tue, 18 Jan 2011) \| 18 lines Merged revisions 302313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302313 \| mnicholson \| 2011-01-18 15:40:03 -0600 (Tue, 18 Jan 2011) \| 11 lines Merged revisions 302311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302311 \| mnicholson \| 2011-01-18 15:35:03 -0600 (Tue, 18 Jan 2011) \| 4 lines URI encode the user part of the contact header. ABE-2705 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302315 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	ae6b55e4a3	Merged revisions 293493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293493 \| twilson \| 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010) \| 14 lines Only offer codecs both sides support for directmedia When using directmedia, Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (closes issue #17403) Reported by: one47 Patches: sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	c69406f384	Merged revisions 301946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301946 \| rmudgett \| 2011-01-14 15:09:57 -0600 (Fri, 14 Jan 2011) \| 13 lines Deadlock between dahdi_request() and pri_dchannel() processing an incomming call. The sig_pri_new_ast_channel() is called with the channel private lock held when pri_dchannel() calls it and no channel private lock held when dahdi_request() calls it. The use of pri_grab() in sig_pri_new_ast_channel() could leave the channel private lock held when it returns if the lock was not held before calling it. Make sig_pri_new_ast_channel() just lock the PRI span lock instead of using pri_grab(). It is safe to do this because dahdi_request() does not have the channel private lock and the deadlock potential with the PRI span lock is only between pri_dchannel() and other threads. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	ed0a2e8c31	Merged revisions 301851 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301851 \| bbryant \| 2011-01-14 15:11:55 -0500 (Fri, 14 Jan 2011) \| 6 lines Changing previous revisions 301845/301847 to use ast_sockaddr_setnull() instead of setting the field manually to avoid uninitialized data. Review: https://reviewboard.asterisk.org/r/1076/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301858 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	558c6a5a1a	Merged revisions 301845 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301845 \| bbryant \| 2011-01-14 14:35:23 -0500 (Fri, 14 Jan 2011) \| 9 lines Fix for a consistent MulticastRTP channel driver crash due to use of unitilized data. (closes issue #18290) (closes issue #18602) Reported by: voipgate, wybecom Review: https://reviewboard.asterisk.org/r/1076/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301847 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	a0e4c4ee5b	Merged revisions 301790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301790 \| jpeeler \| 2011-01-14 11:32:52 -0600 (Fri, 14 Jan 2011) \| 42 lines Resolve deadlock involving REFER. Two fixes: 1) One must always have the private unlocked before calling pbx_builtin_setvar_helper to not invalidate locking order since it locks the channel. 2) Unlock the channel before calling pbx_find_extension, which starts and stops autoservice during the lookup. The problem scenario as illustrated by the reporter: Thread: do_monitor ----------------------- handle_request_do handle_incoming handle_request_refer ast_parking_ext_valid pbx_find_extension ast_autoservice_stop while (chan_list_state == as_chan_list_state) { usleep(1000); } Thread: autoservice_run ----------------------- autoservice_run chan = ast_waitfor_n ast_waitfor_nandfds ast_waitfor_nandfds_classic / simple / complex (depending on your system) ast_channel_lock(c[x]); handle_request_do and schedule_process_request_queue locks the owner if it exists. The autoservice thread is waiting for the channel lock, which wasn't ever released since the do_monitor thread was waiting for autoservice operations to complete. Solved by unlocking the channel but keeping a reference to guarantee safety. (closes issue #18403) Reported by: jthurman Patches: 20110103-blind_deadlock.diff uploaded by jthurman (license 614) issue18403.patch uploaded by jpeeler (license 325) Tested by: jthurman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	c6858b9a1d	Merged revisions 301683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301683 \| twilson \| 2011-01-12 15:19:48 -0600 (Wed, 12 Jan 2011) \| 15 lines Merged revisions 301682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301682 \| twilson \| 2011-01-12 15:05:02 -0600 (Wed, 12 Jan 2011) \| 9 lines Don't reject all SUBSCRIBE auth requests When merging another SUBSCRIBE fix from 1.4, some braces were put in the wrong place. This patch fixes that. (closes issue #18597) Reported by: thsgmbh ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301684 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	f91340bb71	Merged revisions 301134 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301134 \| rmudgett \| 2011-01-07 19:11:31 -0600 (Fri, 07 Jan 2011) \| 7 lines The DTMF attended transfer feature cannot callback a chan_dahdi BRI phone. The DAHDI ISDN channel name is not dialable. Make a channel name like DAHDI/i3/400-12 dialable when the sequence number is stripped off of the name. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301135 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	398d633ce0	Merged revisions 300714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300714 \| rmudgett \| 2011-01-05 14:54:21 -0600 (Wed, 05 Jan 2011) \| 21 lines Merged revision 300711 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r300711 \| rmudgett \| 2011-01-05 13:43:55 -0600 (Wed, 05 Jan 2011) \| 14 lines A call retrieved from hold may wind up with no audio. If the retrieved call is natively bridged then the call may not have any audio path. The following warning message is given: "Failed to add <dfd> to conference <chan>/<chan>: Invalid argument". * Open the media on a B channel when pri_fixup_principle() moves the call from a no_b_channel channel to a real channel. * Added lock protection while pri_fixup_principle() moves a call from one private structure to another. * Made some pri_fixup_principle() messages more meaningful. .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	783ea39ba1	Merged revisions 300521 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300521 \| lmadsen \| 2011-01-04 15:53:27 -0600 (Tue, 04 Jan 2011) \| 17 lines Merged revisions 300520 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r300520 \| lmadsen \| 2011-01-04 15:52:41 -0600 (Tue, 04 Jan 2011) \| 9 lines Fix backwards and broken XML documentation. (closes issue #18547) Reported by: jcovert Patches: xmldoc.c.patch uploaded by jcovert (license 551) chan_iax2.c.doc.patch uploaded by jcovert (license 551) chan_sip.c.patch uploaded by jcovert (license 551) chan_agent.c.patch uploaded by jcovert (license 551) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Moises Silva	3b1553f281	Update MFC-R2 code to use new DTMF-R2 functionality in OpenR2 (closes issue #18576) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300345 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	94ef793caa	Merged revisions 300301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300301 \| twilson \| 2011-01-04 11:54:41 -0600 (Tue, 04 Jan 2011) \| 29 lines Merged revisions 300298 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r300298 \| twilson \| 2011-01-04 11:37:26 -0600 (Tue, 04 Jan 2011) \| 22 lines Merged revisions 300216 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300216 \| twilson \| 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011) \| 15 lines Don't authenticate SUBSCRIBE re-transmissions This only skips authentication on retransmissions that are already authenticated. A similar method is already used for INVITES. This is the kind of thing we end up having to do when we don't have a transaction layer... (closes issue #18075) Reported by: mdu113 Patches: diff.txt uploaded by twilson (license 396) Tested by: twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	90177fe708	Optional HOLD/RETRIEVE signaling for PTMP TE when the bridge goes on and off hold. Added the moh_signaling option to specify what to do when the channel's bridged peer puts the ISDN channel on and off of hold. Implemented as a FSM to control libpri ISDN signaling when the bridged peer places the channel on and off of hold with the AST_CONTROL_HOLD and AST_CONTROL_UNHOLD control frames. JIRA SWP-2687 JIRA ABE-2691 Review: https://reviewboard.asterisk.org/r/1063/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	ac87fc136d	Merged revisions 299626 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r299626 \| tilghman \| 2010-12-25 04:07:15 -0600 (Sat, 25 Dec 2010) \| 19 lines Merged revisions 299625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r299625 \| tilghman \| 2010-12-25 04:05:00 -0600 (Sat, 25 Dec 2010) \| 12 lines Merged revisions 299624 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r299624 \| tilghman \| 2010-12-25 04:04:06 -0600 (Sat, 25 Dec 2010) \| 5 lines Move check for extension existence below variable inheritance, due to the possible use of an eswitch. (closes issue #16228) Reported by: jlaguilar ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Moises Silva	eba903040d	Enqueue AST_CONTROL_PROGRESS after AST_CONTROL_RINGING when MFC-R2 calls are accepted (closes issue #18438) Reported by: mariner7 Tested by: moy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	17d2c0f787	Merged revisions 299405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r299405 \| rmudgett \| 2010-12-21 20:10:39 -0600 (Tue, 21 Dec 2010) \| 17 lines Chan_dahdi sends an empty COLP on the bridged channel. Chan_dahdi always inserts a connected party IE when you call from one dahdi channel to another dahdi channel, even if no such information was received on the 2nd channel. This clears the display of many phones. * Removed leftover artifact from before the valid flag was added. * Updated all of the channel's caller id information with the new connected line information instead of just the string parts. (closes issue #18508) Reported by: wimpy Patches: issue18508_trunk.patch uploaded by rmudgett (license 664) Tested by: wimpy, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	ef23c07447	Merged revisions 299353 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r299353 \| mnicholson \| 2010-12-21 09:25:03 -0600 (Tue, 21 Dec 2010) \| 30 lines Merged revisions 299242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r299242 \| mnicholson \| 2010-12-20 15:25:35 -0600 (Mon, 20 Dec 2010) \| 23 lines Merged revisions 299194,299198,299220 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r299194 \| mnicholson \| 2010-12-20 14:45:38 -0600 (Mon, 20 Dec 2010) \| 6 lines Respond as soon as possible with a 202 Accepted to refer requests. This change also plugs a few memory leaks that can occur when parking sip calls. ABE-2656 ........ r299198 \| mnicholson \| 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) \| 2 lines Remove changes to via processing that were not supposed to go into the last commit. ........ r299220 \| mnicholson \| 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) \| 4 lines Use ast_free() instead of free() ABE-2656 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	59ec959844	Merged revisions 299248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r299248 \| mmichelson \| 2010-12-20 15:38:30 -0600 (Mon, 20 Dec 2010) \| 20 lines Fix a couple of CCSS issues. * Make sure to allocate a cc_params structure when creating autopeers. * Use sip_uri_cmp when retrieving SIP CC agents and monitors in case parameters appear in the URI. (closes issue #18504) Reported by: kkm (closes issue #18338) Reported by: GeorgeKonopacki Patches: 18338.diff uploaded by mmichelson (license 60) Tested by: GeorgeKonopacki ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	9ae2d8024d	Fix chan_misdn build after sched API changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299134 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	cc0b7e7df5	Some scheduler API cleanup and improvements. Previously, I had added the ast_sched_thread stuff that was a generic scheduler thread implementation. However, if you used it, it required using different functions for modifying scheduler contents. This patch reworks how this is done and just allows you to optionally start a thread on the original scheduler context structure that has always been there. This makes it trivial to switch to the generic scheduler thread implementation without having to touch any of the other code that adds or removes scheduler entries. In passing, I made some naming tweaks to add ast_ prefixes where they were not there before. Review: https://reviewboard.asterisk.org/r/1007/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tzafrir Cohen	6307b6fe3a	Typos: recieved => received git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299005 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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6872 Commits (eaf73d6588821992ae08860b5c158f25d644c424)