asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	2c3804e06e	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	ed7570328a	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	496d1009dd	Second attempt to get optimal translation paths when codec_resample is used. This borrows code heavily from changes made in translation code in Asterisk 10. This uses the quality and sample rate change of translation in order to pick paths rather than the computational cost of translations. Computational cost is used solely in determining if a single translation step from a specific translator is better than the same translation step provided by a different translator. (closes issue ASTERISK-16821) reported by Andrew Lindh Review: https://reviewboard.asterisk.org/r/1772 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	8417ceb751	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	99a080671d	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357575 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	cc470930ba	Adding transport=udp to sample sip.conf - Also changes version of Asterisk 1.8 in UPGRADE (issue ASTERISK-19352) Reported by: jamicque Patches: asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	27ce64075f	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357455 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Tilghman Lesher	e289e9caf9	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	ec57a80169	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357407 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	0472ec7ba7	Moves UPGRADE.txt notes from r357356 to a new section specific to 1.8.12 (issue ASTERISK-19352) reported by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	60907f7df1	Adds UPGRADE.txt notes to r357266 indicating changes to transport option (issue ASTERISK-19352) Reported by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357356 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	00fc360507	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357352 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	52c50e4da7	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	f3fbe7d88f	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	100721d217	Fix callerid of Originated calls. Thanks to Matt Riddell for tracking this down. (closes issue ASTERISK-19385) Reported by: ornix git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	c026fb96b0	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	c7d587dd49	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	d3ed07d38a	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356797 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	534213a074	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	ed81d0e585	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356650 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	032962f1a2	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356604 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f49ff3ff9c	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	775b218b35	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356475 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Paul Belanger	d2cb0914e4	Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Paul Belanger	bd7d5707dd	Missed one strsep() function git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Paul Belanger	7d3cdcffd2	Add back strsep() function for previous commit git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356335 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	9a3c569772	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356291 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Paul Belanger	8dc1509465	Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) Review: https://reviewboard.asterisk.org/r/1763/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356290 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	6453352768	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/1.8@356214 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	e880b4a205	Make 'iax2 show callnumber usage' output make sense when an IP is passed in. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356107 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	cb8d4a1d50	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355997 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	a8989c5ded	This was a LOG_NOTICE, so roll it back. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	11991e8394	Change some debug messages from LOG_DEBUG to ast_debug. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	4b59946c41	Add some boilerplate documentation for IAXVAR and IAXPEER. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	3925b8fdc9	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355901 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	1b6601bc0a	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Paul Belanger	0b894ef73a	Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355839 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	0106636e42	Don't allow trunkfreq to be greater than 1000ms. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	338fd29f44	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355746 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	202d83c42c	Fix regressions with regards to route-set creation on early dialogs. This fixes two main issues: 1. Asterisk would send a CANCEL to the route created by the provisional response instead of using the same destination it did in the initial INVITE. 2. If a new route set arrives in a 200 OK than was in the 1XX response (perfectly possible if our outbound INVITE gets forked), then the route set in the 200 OK needs to overwrite the route set in the 1XX response. (closes issue ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson (license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt (license 6034) Review: https://reviewboard.asterisk.org/r/1749 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355732 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	1ba154c100	Revert change to translate.c as it has caused an infinite loop to occur in circumstances. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Paul Belanger	e31d0dfe40	Revert 355700 and 355701 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355711 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Paul Belanger	1fd85ee397	Missed a variable git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Paul Belanger	0e93bc868e	Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355700 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	706d34fade	Revert a change to audio_audiohook_write_list that had no affect. When I made this change initially, I was under the false impression that the audiohooks structure remained on the channel after all of the hooks had been detached. This is not the case, ast ast_read takes care of removing the audiohooks structure if the lists are empty. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	15f49a1497	Fix compile problem when old version of libvorbisfile v1.1.2 is used. The principle difference between libvorbisfile v1.1.2 and newer (at least v1.2.0) is the addition of the predefined callbacks OV_CALLBACKS_xxx in vorbis/vorbisfile.h used for ov_open_callbacks(). * Updated the configure script to detect if libvorbisfile.h declares OV_CALLBACKS_NOCLOSE. * Copied the declaration of OV_CALLBACKS_NOCLOSE from v1.2.0 to allow v1.1.2 to compile. (closes issue ASTERISK-19370) Reported by: Jonn Taylor git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355608 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	695e1ecbec	Fix AMI Monitor action without File header converting channel name into filename. * Fix potential Solaris crash if Monitor application has a urlbase and no fname_base option. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355574 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	dfe4ff5337	When IAX2 debugging is enabled, make sure to log 'apathetic' messages too. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	314dcc01bc	Only use maxtrunkcall and maxnontrunkcall in chan_iax2 if IAX_OLD_FIND is specified. These variables are only accessed from the IAX_OLD_FIND path, so there is no reason to keep them updated otherwise. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355458 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	62b7e35b71	Use TRUNK_CALL_START as originally intended. Back in r646, TRUNK_CALL_START was added and defined as 0x4000. That same value was also hard-coded in one part of the IAX2 code instead of using the #define. TRUNK_CALL_START has changed over the years (for dealing with LOW_MEMORY), but the hard-coded usage was never updated to match. This patch fixes that. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355448 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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