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Author	SHA1	Message	Date
Joshua Colp	732eaebec7	Defer sending the connected line reinvite if a reinvite is already in progress. ... (issue ASTERISK-19355) Reported by: tomaso (closes issue AST-825) ........ Merged revisions 358162 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@358163 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	e7c8651f06	Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors ... Asterisk was not setting pendinginvite in the upper half of handle_request_invite such that the 4xx was retransmitted repeatedly even though an ack was received for every retransmission. (closes issue ASTERISK-19303) Reported by: Jon Tsiros Patches: fix-19303.patch uploaded by Jeremiah Gowdy (license 6358) ........ Merged revisions 358115 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@358116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	ae3cb097ad	Fix unused-but-set-variable warnings ... All of these were pretty obviously unused. Some were unused because the code that used them was #if 0'd. In those cases, I just commented out the unused-but-set variables. ........ Merged revisions 358029 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@358033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	6e64180d29	Correct some set-but-unused variable warnings in the mISDN library. ... (from kpfleming's commit to trunk r356292) ........ Merged revisions 358011 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@358017 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	c2456f442e	Make chan_usbradio compile under dev mode ... x=++x and x=x=1? Really? ........ Merged revisions 357986 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357987 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	56c9cee3f6	Fix case-sensitivity for device-specific event subscriptions and CCSS ... This change fixes case-sensitivity for device-specific subscriptions such that the technology identifier is case-insensitive while the remainder of the device string is still case-sensitive. This should also preserve the original case of the device string as passed in to the event system. CCSS is the only feature affected as it is the only consumer of device-specific event subscriptions. The second part of this patch addresses similar case-sensitivity issues within CCSS itself that prevented it from functioning correctly after the fix to the events system. This adds a unit test to verify that the event system works as expected. (closes issue ASTERISK-19422) Review: https://reviewboard.asterisk.org/r/1780/ ........ Merged revisions 357940 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357941 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	f1bdfc6180	Remove ISDN hold restriction for non-bridged calls. ... The check if an ISDN call is bridged before it could be placed on hold is not necessary and is overly restrictive. The check was originally done to prevent problems with call transfers in case a user tried to transfer a call connected to an application to another call connected to an application. The ISDN transfer code has not required this restriction for quite some time because ECT could transfer any two active calls to each other. * Remove ISDN hold restriction for calls connected to applications. * Made ast_waitfordigit_full() ignore AST_CONTROL_HOLD and AST_CONTROL_UNHOLD instead of generating a warning message. (closes issue ASTERISK-19388) Reported by: Birger Harzenetter Tested by: rmudgett ........ Merged revisions 357894 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	ca5fe398a2	The default value for mohinterpret is the empty string, so when resetting to ... default values don't explicitly set the value to "default." ........ Merged revisions 357811 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357812 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	edaf107a1e	Fix channel reference leak in ChanSpy. ... * Fix next_channel() channel reference leak in ChanSpy. (closes issue ASTERISK-19461) Reported by: Irontec Patches: app_chanspy_iteartor_next_unref.patch (license #6213) patch uploaded by Irontec (issue ASTERISK-17515) ........ Merged revisions 357809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	76dc2d4705	Fix race condition that can cause important control frames (such as a hangup) to be missed. ... This takes two actions. 1. Move the reading of the alertpipe in __ast_read() to immediately before the removal of frames from the readq. This means we won't do something silly like read from the alertpipe, then ignore the fact that there's a frame to get from the readq since channel's fdno is the AST_TIMING_FD. 2. When ast_settimeout() sets the rate to 0 and the timingfunc to NULL, if the channel's fdno is the AST_TIMING_FD, then set the fdno to -1. This is because if the rate is 0 and the timingfunc is NULL, it means that the channel's timing fd is being invalidated, so any pending reads should not occur. This may actually solve more issues than the referenced one below, but it's not known at this time for sure. (closes issue ASTERISK-19223) reported by Frank-Michael Wittig Review: https://reviewboard.asterisk.org/r/1779 ........ Merged revisions 357761 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357762 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	7741e5fab3	Prevent outbound SIP NOTIFY packets from displaying a port of 0 ... In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which changed the behavior of ast_find_ourip such that port number was wiped out. This caused the port in internip (which is used for Contact and Call-ID on NOTIFYs) to be 0. This change causes ast_find_ourip to be port-preserving again. (closes issue ASTERISK-19430) ........ Merged revisions 357665 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357667 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	98b303468b	Update stringfield documentation for removed second va_list in favor of va_copy. ... In r320946, the second va_list that was passed to ast_string_field_build_va and friends, was removed. This patch updates the documentation to reflect that. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	0bd36686f2	Fix copying of CDR(accountcode) to local channels. ... In r203638, during the addition of the Channel Event Logging, in mid-2009, this got broken in trunk and ended up in asterisk 1.8 and higher. This fixes so the CDR(accountcode) from the calling channel is available to dialed channels again as well as showing up properly in the CDR's. (closes issue ASTERISK-19384) Reported by: jamicque Patches: accountcode.patch (License #6033) by jamicque Review: https://reviewboard.asterisk.org/r/1775/ Reviewed by: Richard Mudgett ........ Merged revisions 357575 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357576 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	03d77a164a	Adding transport=udp to sample sip.conf - Also changes version of Asterisk 1.8 in UPGRADE ... (issue ASTERISK-19352) Reported by: jamicque Patches: asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026) ........ Merged revisions 357490 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357497 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	770cb0d35d	Add additional character type types to supported data types for cdr_adaptive_odbc ... The reporter was uable to use varchar utf8_unicode_ci with cdr_adaptive_odbc, so this patch adds those along with some other character types to the list of types cdr_adaptive_odbc will work using the varchar conditions. The problem wasn't really UTF8 characters as much as it was a failure to respond to the exact type that was declared/in use on that database. (closes issue ASTERISK-19334) Reported By: Igor Nikolaev Patches: cdr_adaptive_odbc.patch uploaded by Igor Nikolaev (license 6236) ........ Merged revisions 357455 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357458 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	0ce4065d5e	Correctly reset the dialplan priority. ... When the stack frame is allocated, we save the address to which we should return, when the Gosub returns. However, if we just want to restore the priority, then we need to subtract 1 before setting it. Otherwise, when a Gosub goes to a nonexistent address, it will skip a priority in the dialplan. This is because when we return from an application, the PBX increments the priority for us. ........ Merged revisions 357416 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357421 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	435ebee127	Use more reasonable cause code when rejecting incoming call waiting calls. ... (closes issue ASTERISK-19397) Reported by: Birger Harzenetter Patches: nochannel-cause.patch (license #5870) patch uploaded by Birger Harzenetter ........ Merged revisions 357407 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357408 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	b8fe2da70f	revision 357386 -- oops, accidentally made it 10.3 to 10.4 instead of 10.2 to 10.3 ... (issue ASTERISK-19352) reported by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357405 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	0b6bdd2717	Moves UPGRADE.txt notes from r357356 to a new section specific to 1.8.12 ... (issue ASTERISK-19352) reported by: jamicque ........ Merged revisions 357386 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	3d627d9877	Adds UPGRADE.txt notes to r357266 indicating changes to transport option ... (issue ASTERISK-19352) Reported by: jamicque ........ Merged revisions 357356 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357357 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	6c0a0a6fd9	Remove dupliate 'i' option table entry in app_page.c. ... (closes issue ASTERISK-19310) Reported by: Makoto Dei Patches: app_page-duplicate-i-option.patch (license #5027) patch uploaded by Makoto Dei ........ Merged revisions 357352 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357353 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	6a04c78329	Add a security event for the case where fake authentication challenge is sent. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	7deeed0304	Changes transport option in sip.conf so that using multiple instances doesn't stack. ... Prior to this patch, Using "transport=" multiple times would cause them to add to one another like allow/deny. This patch changes that behavior to simply use the transport option specified last. Also, if no transport option is applied now, the default will automatically be UDP. (closes ASTERISK-19352) Reported by: jamicque Patches: asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026) issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes (license 5674) Review: https://reviewboard.asterisk.org/r/1745/diff/#index_header ........ Merged revisions 357266 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357271 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	96c3aecb0b	Make COMPILE_DOUBLE magic actually work. ... The build system has some special magic to ensure that if Asterisk is built with --enable-dev-mode *and* DONT_OPTIMIZE, that all the source is still compiled with the optimizer enabled (even though the result will be thrown away), because the compiler is able to find a great deal of coding errors and bugs as a result of running its optimizers. Unfortunately at some point this mode got broken, and the 'throwaway' compile of the code was no longer done with the optimizer enabled. This patch corrects that problem. ........ Merged revisions 357212 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	da61b9ec5f	Fix callerid of Originated calls. ... Thanks to Matt Riddell for tracking this down. (closes issue ASTERISK-19385) Reported by: ornix ........ Merged revisions 357093 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357095 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	b0b1eb823a	Copy CDR variables when set during a bridge ... This patch makes sure amaflags, accountcode, and userfield get copied to the bridge CDR when set during a bridge (like via a custom feature). (closes issue ASTERISK-16990) Review: https://reviewboard.asterisk.org/r/1721/ ........ Merged revisions 356963 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	464e75f711	Remove possible segfaults from res_odbc by adding locks around usage of odbc handle ... (closes issue ASTERISK-19011) Reported by: Walter Doekes Patches: issueA19011_combine_read_and_write_locks_WORK_IN_PROGRESS.patch uploaded by Walter Doekes (license 5674) review: https://reviewboard.asterisk.org/r/1719/ review: https://reviewboard.asterisk.org/r/1622/ ........ Merged revisions 356917 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356961 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	fe1a4e7a48	Fix crash in app_voicemail during close_mailbox ... In r354890, a memory leak in app_voicemail was fixed by properly disposing of the allocated heard/deleted pointers. However, there are situations, particularly when no messages are found in a folder, where these pointers are not allocated and not NULL. In that case, an invalid free would be attempted, which could crash app_voicemail. As there are a number of code paths where this could occur, this patch uses the number of messages detected in the folder before it attempts to free the pointers. This resolves the crash detected in the Asterisk Test Suite's check_voicemail_nominal test. ........ Merged revisions 356797 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d372c43488	Fix worker thread resource leak in SIP TCP/TLS. ... The SIP TCP/TLS worker threads were created joinable but noone could join them if they died on their own. * Fix the SIP TCP/TLS worker threads to not be created joinable. * _sip_tcp_helper_thread() only needs one parameter since the pvt parameter is only passed in as NULL and never used. (closes issue ASTERISK-19203) Reported by: Steve Davies Review: https://reviewboard.asterisk.org/r/1714/ ........ Merged revisions 356677 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356690 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	ff7665db86	Remove srtp_shutdown from res_srtp ... The patch for ASTERISK-19253 included properly shutting down the libsrtp library in the case of module unload. Unfortunately, not all distributions have the srtp_shutdown call. As such, this patch removes calling srtp_shutdown. ........ Merged revisions 356650 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356651 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	e796b45951	Allow SRTP policies to be reloaded ... Currently, when using res_srtp, once the SRTP policy has been added to the current session the policy is locked into place. Any attempt to replace an existing policy, which would be needed if the remote endpoint negotiated a new cryptographic key, is instead rejected in res_srtp. This happens in particular in transfer scenarios, where the endpoint that Asterisk is communicating with changes but uses the same RTP session. This patch modifies res_srtp to allow remote and local policies to be reloaded in the underlying SRTP library. From the perspective of users of the SRTP API, the only change is that the adding of remote and local policies are now added in a single method call, whereas they previously were added separately. This was changed to account for the differences in handling remote and local policies in libsrtp. Review: https://reviewboard.asterisk.org/r/1741/ (closes issue ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283) (with some small modifications for this check-in) ........ Merged revisions 356604 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356605 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	3b57020680	Fix blind transfer parking issues if the dialed extension is not recognized as a parking extension. ... Custom parking extensions may not be coded such that the first and only extension priority is the Park application. These custom parking extensions will not be recognized as parking extensions. When a call is blind transferred to an extension that is not recognized as a parking extension, the normal blind transfer code causes the transferred channel to start executing dialplan. Calls that get parked in this manner do not know the original channel name that parked the call so the original parker could never be called back if the parked call is not retrieved before the timeout time. The parking space is also announced to the call being parked as a side effect of not knowing the original parking channel. * Fix handling of BLINDTRANSFER channel variable for call parking. * Fixed SIP blind transfer using the wrong dialplan context variable to check for the parking extension. (closes issue ASTERISK-19322) Reported by: aragon Tested by: rmudgett, jparker Review: https://reviewboard.asterisk.org/r/1730/ JIRA AST-766 ........ Merged revisions 356521 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	4501bb048f	Fix ACK routing for non-2xx responses. ... When we send an ACK for a 2xx response to an INVITE, we are supposed to use the learned route set. However, when we receive a non-2xx final response to an INVITE, we are supposed to send the ACK to the same place we initially sent the INVITE. We had been doing this up until the changes went in that would build a route set from provisional responses. That introduced a regression where we would use the learned route set under all circumstances. With this change, we now will set the destination of our ACK based on the invitestate. If it is INV_COMPLETED then that means that we have received a non-2xx final response (INV_TERMINATED indicates a 2xx response was received). If it is INV_CANCELLED, then that means the call is being canceled, which means that we should be ACKing a 487 response. The other change introduced here is setting the invitestate to INV_CONFIRMED when we send an ACK *after* the reqprep instead of before. This way, we can tell in reqprep more easily what the invitestate is prior to sending the ACK. (closes issue ASTERISK-19389) reported by Karsten Wemheuer patches: ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049) (with some slight modifications prior to commit) ........ Merged revisions 356475 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356476 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	a5ec1fdc8a	Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) ... ........ Merged revisions 356430 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356431 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	bcadb05414	Multiple revisions 356290,356335,356337 ... ........ r356290 | pabelanger | 2012-02-22 15:20:29 -0500 (Wed, 22 Feb 2012) | 4 lines Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) Review: https://reviewboard.asterisk.org/r/1763/ ........ r356335 | pabelanger | 2012-02-22 16:29:25 -0500 (Wed, 22 Feb 2012) | 2 lines Add back strsep() function for previous commit ........ r356337 | pabelanger | 2012-02-22 16:36:37 -0500 (Wed, 22 Feb 2012) | 2 lines Missed one strsep() function ........ Merged revisions 356290,356335,356337 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	f11810c690	Track module use count for res_calendar ... If the res_calendar module was followed immediately by one of the calendar tech modules and "core stop gracefully" was run, Asterisk would crash. This patch adds use count tracking for res_calendar so that it is unloaded after the tech modules when shutting down gracefully. It is now not possible to unload all the of the calendar modules via "module unload res_calednar.so", but it is still possible to unload them all via "module unload -h res_calendar.so". Review: https://reviewboard.asterisk.org/r/1752/ ........ Merged revisions 356291 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	bc3cb5ddbc	Merged revisions 356214 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r356214 | mjordan | 2012-02-22 08:50:20 -0600 (Wed, 22 Feb 2012) | 27 lines Fix potential buffer overrun and memory leak when executing "sip show peers" The "sip show peers" command uses a fix sized array to sort the current peers in the peers ao2_container. The size of the array is based on the current number of peers in the container. However, once the size of the array is determined, the number of peers in the container can change, as the peers container is not locked. This could cause a buffer overrun when populating the array, if peers were added to the container after the array was created. Additionally, a memory leak of the allocated array would occur if a user caused the _show_peers method to return CLI_SHOWUSAGE. We now create a snapshot of the current peers using an ao2_callback with the OBJ_MULTIPLE flag. This size of the array is set to the number of peers that the iterator will iterate over; hence, if peers are added or removed from the peers container it will not affect the execution of the "sip show peers" command. Review: https://reviewboard.asterisk.org/r/1738/ (closes issue ASTERISK-19231) (closes issue ASTERISK-19361) Reported by: Thomas Arimont, Jamuel Starkey Tested by: Thomas Arimont, Jamuel Starkey Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan (license 6283) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356215 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	34179f7093	Make 'iax2 show callnumber usage' output make sense when an IP is passed in. ... ........ Merged revisions 356107 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356108 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	fe385136d7	Add missing newline to ccss state change notification ... Move along, nothing to see here... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356074 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	c6cde9e2c0	Remove spurious warning when 'qualifyfreqnotok' is set successfully. ... (closes issue ASTERISK-17176) Reported by: John Covert Tested by: Sean Bright Patches: chan_iax2.c.qualifyfreqnotok.patch uploaded by John Covert (license 5512) ........ Merged revisions 355997 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	76ddf391c6	This was a LOG_NOTICE, so roll it back. ... ........ Merged revisions 355952 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	fd99b4c811	Change some debug messages from LOG_DEBUG to ast_debug. ... ........ Merged revisions 355949 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	a61bda1a90	Add some boilerplate documentation for IAXVAR and IAXPEER. ... ........ Merged revisions 355904 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	4b13d1d36a	Set the length of the ast_sockaddr, so that we can set it's port later. ... Without this, the call to ast_sockaddr_set_port a few lines later is a noop. ........ Merged revisions 355901 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	3ac83ec54b	Blocked revisions 355839 ... ........ Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355897 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	961f71c3b0	Revert commit ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	717ea381b9	Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) ... ........ Merged revisions 355839 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	b9e0c5ecf4	push 'outgoing' flag from sig_XXX up to chan_dahdi ... 'p->outgoing' in chan_dahdi and sig_analog wern't kept in sync, particulary FXS ast_hangup didn't clear the 'outgoing' flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync. Now provides a callback for all the low level sig_XXX modules. (issue ASTERISK-19316) alecdavis (license 585) Reported by: Jeremy Pepper Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1747/ ........ Merged revisions 355850 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	7070d7acef	Don't allow trunkfreq to be greater than 1000ms. ... ........ Merged revisions 355793 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355794 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	f41e7d4a44	Pass the correct value to ast_timer_set_rate() for IAX2 trunking. ... IAX2 uses the trunkfreq variable to determine how often to send trunk packets, but this value is in milliseconds while ast_timer_set_rate() expects the rate argument to be ticks per second. So we divide 1000 by trunkfreq and pass that in instead. With a default of 20ms, this change makes IAX2 send trunk packets every 20ms instead of every 50ms. Tracked down by myself and Bob Wienholt. ........ Merged revisions 355746 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago