asterisk

Commit Graph

Author	SHA1	Message	Date
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
Richard Mudgett	e296e8c96f	Fix voicemail problems when using ogg/vorbis. Ogg/vorbis was fairly useless as a voicemail file format because it did not implement the seek and tell format callbacks among other problems. Since we were already using the libvorbis and libvorbisenc libraries we can use libvorbisfile as it is also part of the vorbis library package. * Made use the libvorbisfile to handle the ogg/vorbis file stream. The format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile. (closes issue ASTERISK-16926) Reported by: sque Patches: ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded by sque git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f0e24c2e24	Fix lock typo that should be unlock in cel_sqlite_custom reload. (closes issue ASTERISK-19356) Reported by: Alex Villacis Lasso Patches: asterisk-1.8.9.2-cel_sqlite3_custom-fix-reload-locking-typo.patch (license #5617) patch uploaded by Alex Villacis Lasso Review: https://reviewboard.asterisk.org/r/1740/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	a5d76e1c11	Properly invert the return of a strncmp call. This was causing identification that should have been made private to be public. (closes issue AST-814) reported by Patrick Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson (license 5430) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	8357dd7a2d	Don't enable sqlite3 CDRs by default in sample configs. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355228 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	94ade43b56	Clear the high order bit from the destination call number before sending. send_apathetic_reply takes the incoming frame's source call number as the destination call number for the outgoing frame. If the incoming frame was a full frame, then the high order bit of the source call number is set and will be interpreted as a retransmit when sent back out as the destination call number. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alexandr Anikin	7d0ee59215	call manager_event only if there is not null channel structure (Closes issue ASTERISK-19298) Reported by: robinfood Patches: issue19298.patch uploaded by may213 (License #5415) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355136 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	49ebbe0765	Fix occasional incorrectly delayed call-file execution. Since the dir timestamp is available at one second resolution, we cannot know if it was updated within the same second after we scanned it. Therefore, we will force another scan if the dir was just modified. * Changed to force another scan if the directory was just modified. (closes issue ASTERISK-19081) Reported by: Knut Bakke Review: https://reviewboard.asterisk.org/r/1688/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355056 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	c81c500276	Only allow one 'dialplan reload' to execute at a time as otherwise they would share the same common local context list. (closes issue AST-758) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	7d84a0ed69	Fix reconnecting to pgsql database after connection loss. There can only be one database connection in res_config_pgsql just like res_config_sqlite. If the connection is lost, the connection may not get reestablished to the same database if the res_pgsql.conf and extconfig.conf files are inconsistent. * Made only use the configured database from res_pgsql.conf. * Fixed potential buffer overwrite of last[] in config_pgsql(). (closes issue ASTERISK-16982) Reported by: german aracil boned Review: https://reviewboard.asterisk.org/r/1731/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	a37f262426	Fix a voicemail memory leak with heard/deleted messages. open_mailbox() was changed quite a long time ago to allocate this memory. close_mailbox() should have been changed to be responsible for freeing it. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354889 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	9ce7947aa6	Fix AMI Redirect ExtraChannel not redirecting to the same exten and context. The astman_get_header() never returns NULL so the check by the code for NULL would never fail. (closes issue ASTERISK-16974) Reported by: Nuno Borges Patches: 0018325.patch (license #6116) patch uploaded by Nuno Borges (modified) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	4dd831722a	Note that CDRs are immutable once a bridge is torn down CDRs cannot be modified after a bridge is torn down, (e.g. after Dial() returns) even though the CDR() function may be called. Since modifying the CDR code to change this behavior could very easily break all kinds of things, this patch just documents this limitation. (closes issues ASTERISK-16923) Review: https://reviewboard.asterisk.org/r/1720/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	7d5836ca78	Fix parsing of SIP headers where compact and non-compact headers are mixed Change parsing of SIP headers so that compactness of the header no longer influences which header will be chosen. Previously, a non-compact header would be chosen instead of a preceeding compact-form header. (closes issue ASTERISK-17192) Review: https://reviewboard.asterisk.org/r/1728/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354702 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	ae3a465d8f	Make the config parser remove escaping backslashes The config parser in Asterisk does not currently remove a backslash that is used to escape a semicolon which would otherwise be interpreted as the start of a comment. The change here causes that backslash to be removed, but does not create a real escape system in the config parser. The biggest complication with a real escape system would be breaking existing configs everywhere (parsing \\ as \ and breaking on escaped non-semicolon characters) even though it would be the "right" way to do things. (closes issue ASTERISK-17121) Review: https://reviewboard.asterisk.org/r/1724/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	5aa7a6986e	Remove outdated comment. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	c4334ee50a	Fix translation path choices. This change makes it so computational cost is not taken into account when deciding if a multistep path is better than a single-step path. This means that the only time a multistep path will be chosen is if no single-step path exists. This ensures a better quality translation even if it turns out to be slightly slower. (closes issue ASTERISK-16821) reported by Andrew Lindh Review: https://reviewboard.asterisk.org/r/1715 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354594 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	004babb20d	Clean-up of minor formatting issues in r354542/3/4 rmudgett pointed out some formatting issues in the check-in for ASTERISK-19290. This cleans those up. Review: https://reviewboards.asterisk.org/r/1722/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354547 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	aeadbab220	Adding reload support to res_fax.so (closes issue ASTERISK-16712) reported by Frank DiGennaro Review: https://reviewboard.asterisk.org/r/1713 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354545 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	ead2b47907	Fix SIP INFO DTMF handling for non-numeric codes In ASTERISK-18924, SIP INFO DTMF handlingw as changed to account for both lowercase alphatbetic DTMF events, as well as uppercase alphabetic DTMF events. When this occurred, the comparison of the character buffer containing the event code was changed such that the buffer was first compared again '0' and '9' to determine if it was numeric. Unfortunately, since the first character in the buffer will typically be '1' in the case of non-numeric event codes (10-16), this caused those codes to be converted to a DTMF event of '1'. This patch fixes that, and cleans up handling of both application/dtmf-relay and application/dtmf content types. Review: https://reviewboard.asterisk.org/r/1722/ (closes issue ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	9c33d62bcb	Fix crash in ParkAndAnnounce. Well, thats embarrasing. I forgot to initialize the caller_id storage. (closes issue ASTERISK-19311) Reported by: tootai Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354495 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	fdd8444090	Remove some unnecessary locking from ast_hangup(). This patch removes some unnecessary locking of the channels container in ast_hangup(). The reason this came up is that this lock can very quickly block the entire system. If any of the channel cleanup code decides to block, it causes a problem for the whole system. For example, when audiohooks get destroyed, if that blocks for a while waiting on the mixmonitor thread to exit because it's busy blocking on some I/O, it causes a problem for many other threads in the meantime. Review: https://reviewboard.asterisk.org/r/1712/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354492 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	15d28cfdad	Fix multiple SIP realtime issues 1. Set lastms to 0 when clearing instead of "" 2. Don't set ipaddr or port to the string "(null)" when they are empty 3. Add missing required fields, set default for lastms to 0, and modify the length of the ipaddr field to 45 in the Postgresql realtime.sql file. (closes issue ASTERISK-19172) Review: https://reviewboard.asterisk.org/r/1703/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ff07b89e63	Fix column duplication bug in module reload for cdr_pgsql. Prior to this patch, attempts to reload cdr_pgsql.so would cause the column list to keep its current data and then add a second copy during the reload. This would cause attempts to log the CDR to the database to fail. This patch also cleans up some unnecessary null checks for ast_free and deals with a few potential locking problems. (closes issue ASTERISK-19216) Reported by: Jacek Konieczny Review: https://reviewboard.asterisk.org/r/1711/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	2997c5c8c6	Improved documentation of CLI "dialplan add extension" command. * Documented dialplan add extension <exten>,<priority>,<app(<app-data>)> format. * Allow acceptance of command without the app-data value. There are many applications that do no need any parameters so it is silly to require that field for all commands. * Fixed a couple ast_malloc/ast_free mismatches with ast_add_extension2() calls. (closes issue ASTERISK-19222) Reported by: Andrey Solovyev Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	294793a822	Add missing headers to AMI UnParkedCall event to uniquely identify the call. The AMI UnParkedCall event was missing the Parkinglot and Uniqueid headers that the AMI ParkedCall event contains. (closes issue ASTERISK-19240) Reported by: Michael Yara git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	65360e2ae5	Fixes deadlocks occuring in chan_agent due to r335976 Bad locking order was added to chan_agent to prevent segfaults from having no locking in a patch by irroot. This patch addresses the bad locking order by releasing locks before getting the right locking order to stop deadlocks from occuring when doing multiple interactions with agents. (closes issue ASTERISK-19285) Reported by: Alex Villacis Lasso Review: https://reviewboard.asterisk.org/r/1708/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353999 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	ea4fa3227f	Ensure entering T.38 passthrough does not cause an infinite loop After R340970 Asterisk was still polling the RTCP file descriptor after RTCP is shut down and removed. If the descriptor happened to have data ready when the removal occured then Asterisk would go into an infinite loop trying to read data that it can never actually access. This change disables the audio RTCP file descriptor for the duration of the T.38 transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan Vrban git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353915 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	c5fc58c3a0	Restore the 'w' modifier support for ISDN spans. Dial(DAHDI/g0/1234w888) This feature also causes the sending complete ie to be sent for switch types that do not automatically send the ie. (EuroISDN/ETSI) The main difference between dialing Dial(DAHDI/g0/1234w888) and Dial(DAHDI/g0/1234,,D(888)) is the sending of the sending complete ie. (closes issue ASTERISK-19176) Reported by: rmudgett Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	749b33018e	Backports some documentation for func_curl from 10 to 1.8 For some reason this function was completely undocumented in 1.8. I copied the 10 docs over to 1.8 and removed references to an enumerator that was added in the Asterisk 10 version of func_curl. That was the only change I noted. (closes issue ASTERISK-19186) Reported by: Olivier Krief git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353818 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	47b3aa6362	Fix TLS port binding behavior as well as reload behavior: * Removes references to tlsbindport from http.conf.sample and manager.conf.sample * Properly bind to port specified in tlsbindaddr, using the default port if specified. * On a reload, properly close socket if the service has been disabled. A note has been added to UPGRADE.txt to indicate how ports must be set for TLS. (closes issue ASTERISK-16959) reported by Olaf Holthausen (closes issue ASTERISK-19201) reported by Chris Mylonas (closes issue ASTERISK-19204) reported by Chris Mylonas Review: https://reviewboard.asterisk.org/r/1709 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ad24624751	Fix sip show peers port output, align columns, and fix ami port output. A previous patch I committed from ASTERISK-16930 unexpectedly changed some output for the AMI action "sippeers" which this patch changes back. Also, this aligns the output for the cli command "sip show peers" and fixes another issue that patch introduced by using ast_sockaddr_stringify calls multiple times without immediately using the pointer. I also went ahead and did a little janitorial work to clean up whitespace in _sip_show_peers. (issue ASTERISK-16930) (closes issue ASTERISK-19281) Reported by: Patrick El Youssef Patches: ASTERISK-19281.diff uploaded by Walter Doekes (license 5674) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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