21766 Commits (4dd831722a37a05b02c8e176e90ffe98ab99e37c)

 

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Author	SHA1	Message	Date
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Mark Michelson	5aa7a6986e	Remove outdated comment.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	13 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	13 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	13 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	13 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	13 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	13 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	13 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	13 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	13 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	13 years ago
Jonathan Rose	3c1a9894e8	Use ast_sockaddr_stringify_fmt wrappers for various functions in chan_sip ... There are a number of cleaner looking wrappers for ast_sockaddr_stringify_fmt available which are slightly more readable than using a direct call to ast_sockaddr_stringify_fmt. This patch switches a number of those calls in chan_sip to use those wrappers and is generally harmless. (Closes issue ASTERISK-16930) Reported by: Michael L. Young Patches: chan_sip-broken-registration-1.8.diff uploaded by Michael L. Young (license 5026) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	f28f00bd6e	Resolve an overlap in the ast_audiohook_flags values. ... AST_AUDIOHOOK_TRIGGER_WRITE and AST_AUDIOHOOK_WANTS_DTMF were overlapping which may have caused unintended side effects. This patch moves AST_AUDIOHOOK_TRIGGER_WRITE, and updates AST_AUDIOHOOK_TRIGGER_MODE to reflect the original intention. This will affect existing modules that use these flags, so be sure to recompile as necessary. (closes issue ASTERISK-19246) Reported by: feyfre git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	3bf654b1dc	Added clarification for the VERBOSITY setting to etc_default_asterisk ... Clarified that using the VERBOSITY setting in etc_default_asterisk is the same as using the -v command line switch, which causes Asterisk to launch in console mode. (closes issue ASTERISK-17030) Reported by: Jonas git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353550 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	0120bde7d8	Allow res_calendar to be unloaded ... The calendaring tech modules depend on res_calendar and initially res_calendar just bumped the use count so that it couldn't be unloaded. res_calendar can potentially create many threads and I've seen issues where the Asterisk shutdown has failed where it looked like these threads could be the culprit. This patch adds unload support for res_calendar. Unloading res_calendar will also unload the dependant tech modules as well. (closes issue ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353502 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	c7c7d4dab4	Fix memory leak in error paths for action_originate(). ... * Fix memory leak of vars in error paths for action_originate(). * Moved struct fast_originate_helper tech and data members to stringfields. * Simplified ActionID header handling for fast_originate(). * Added doxygen note to ast_request() and ast_call() and the associated channel callbacks that the data/addr parameters should be treated as const char *. Review: https://reviewboard.asterisk.org/r/1690/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353454 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	d699845a55	Re-link peers by IP when dnsmgr changes the IP ... Asterisk's dnsmgr currently takes a pointer to an ast_sockaddr and updates it anytime an address resolves to something different. There are a couple of issues with this. First, the ast_sockaddr is usually the address of an ast_sockaddr inside a refcounted struct and we never bump the refcount of those structs when using dnsmgr. This makes it possible that a refresh could happen after the destructor for that object is called (despite ast_dnsmgr_release being called in that destructor). Second, the module using dnsmgr cannot be aware of an address changing without polling for it in the code. If an action needs to be taken on address update (like re-linking a SIP peer in the peers_by_ip table), then polling for this change negates many of the benefits of having dnsmgr in the first place. This patch adds a function to the dnsmgr API that calls an update callback instead of blindly updating the address itself. It also moves calls to ast_dnsmgr_release outside of the destructor functions and into cleanup functions that are called when we no longer need the objects and increments the refcount of the objects using dnsmgr since those objects are stored on the ast_dnsmgr_entry struct. A helper function for returning the proper default SIP port (non-tls vs tls) is also added and used. This patch also incorporates changes from a patch posted by Timo Teräs to ASTERISK-19106 for related dnsmgr issues. (closes issue ASTERISK-19106) Review: https://reviewboard.asterisk.org/r/1691/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353371 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	a61f99f985	prevent debug messsges displaying -ve Cseq numbers. Missed in R353320 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	8fc0050b54	RFC3261 Section 8.1.1.5. The sequence number value MUST be expressible as a 32-bit unsigned integer ... * fix: use %u instead of %d when dealing with CSeq numbers - to remove possibility of -ve numbers. * fix: change all uses of seqno and friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t. Summary of CSeq numbers. An initial CSeq number must be less than 2^31 A CSeq number can increase in value up to 2^32-1 An incrementing CSeq number must not wrap around to 0. Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1699/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353320 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	2281ba7cef	Clarify log WARNING message when port-zero SDP 'm' lines received. ... Previously, if an m-line in an SDP offer or answer had a port number of zero, that line was skipped, and resulted in an 'Unsupported SDP media type...' warning message. This was misleading, as the media type was not unsupported, but was ignored because the m-line indicated that the media stream had been rejected (in an answer) or was not going to be used (in an offer). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	d42eaedb15	Find even more network interfaces. ... The previous change made the code look for emN and pciN in addition to what it did originally, which was search for ethN. However, it needed to be looking for pciN#N, so that's what it does now. This also moves the memset() to be before every ioctl(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	1efffc8d95	Add 'L16-256' MIME subtype alias for slin16. ... Asterisk has supported the 'L16' MIME subtype for 16kHz signed linear (PCM) audio for quite some time, but some endpoints refer to it as 'L16-256'. This commit adds this as an alias for the existing format. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	45bf0ea5b0	Update ast_set_default_eid() to find more network interfaces. ... As of Fedora 15, ethN is not the name of ethernet interfaces. The names are emN or pciN. Update some code that searched for interfaces named ethN to look for the new names, as well. For more information about why this change was made, see this page: http://domsch.com/blog/?p=455 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	cda638dcfc	Make failed PauseMonitor and UnpauseMonitor with no valid channel not close AMI session. ... I also went ahead and took a little time to make sure that the manager value AMI_SUCCESS was used instead of just return 0 being thrown around everywhere since that's how we handle this stuff these days. (closes issue ASTERISK-19249) Reporter: Jamuel Starkey Patches: res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey (license 5766) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352959 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	a55030f4fa	Audit of ao2_iterator_init() usage for v1.8. ... Fixes numerous reference leaks and missing ao2_iterator_destroy() calls as a result. Review: https://reviewboard.asterisk.org/r/1697/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352955 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	4d2f8a9cfd	rfc4235 - Section 4.1: Versions MUST be representable using a non-negative 32 bit integer. ... If a BLF subscription exists for long enough, using %d may print negative version numbers. Unlikely, as 2^32 at 1 update per second is ~137 years, or half that before the versions number started going negative. Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1694/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352862 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alexandr Anikin	197662dd4d	Fix outbound DTMF for inband mode (tell asterisk core to generate DTMF ... sounds). (Closes issue ASTERISK-19233) Reported by: Matt Behrens Patches: chan_ooh323.c.patch uploaded by Matt Behrens (License #6346) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	301cc6b1c0	Copy amaflags to sip_pvt from peer during create_addr_from_peer ... For whatever reason, we don't have a single function for copying data like this from SIP peers to the SIP pvt. This patch adds the copying of amaflags to the sip_pvt, but it would probably be worth discussing this function along with the others that essentially just copy some amount of data from a peer to a private. (Closes issue ASTERISK-19029) Reported by: Matt Lehner git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	bd8d057dee	Cleanup dialog-info+xml Notify dialog ... Make similar to other Notify messages. sample output: <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="715" state="full" entity="sip:8523@192.168.x.xx"> <dialog id="8523"> <state>terminated</state> </dialog> </dialog-info> Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1693/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352704 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	b0a70ade4b	Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352643 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	d4dc9894f2	Avoid unnecessary rebuilds of main/test.c. ... main/test.c includes "asterisk/version.h", when it should include "asterisk/ast_version.h" instead (and it should use the ast_get_version() and ast_get_version_num() functions). This commit modifies it to extract the Asterisk version information using the proper APIs, and as a result means that main/test.c no longer needs to be rebuilt when a Subversion checkout is updated or modified. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	054c466a2f	Remove some extraneous debugging from registry memleak fix ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	f1dc1012ae	Clean up some SIP registry-related memory leaks ... 1) Be sure and free at unload the epa_backend we allocate at startup 2) Do the same sip_registry cleanup at unload we do at reload Review: https://reviewboard.asterisk.org/r/1689/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352514 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	5de11bfd6e	Redocuments sip types peer, user, friend in sip.conf.sample ... There was faulty information in the sample config describing user as a synonym for friend so it has been changed to better elaborate on the differences between the three entity types. (closes issue ASTERISK-15537) Reported by: yarique git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352511 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	7c7615e399	Don't do a DNS lookup on an outbound REGISTER host if there is an outbound proxy configured. ... (closes issue ASTERISK-16550) reported by: Olle Johansson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1965cea7ab	Set core sounds version to 1.4.22. ... Now that we have the right license for the Russian 1.4.22 sounds as well as the sounds for the Australian English 1.4.22 sounds, we can finally set the sounds to use 1.4.22! (closes issue ASTERISK-18978) Reported by: Cameron Twomey Patches: confbridge.tar.001 uploaded by Cameron Twomey confbridge.tar.002 uploaded by Cameron Twomey git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	cf450c7db1	Fix locking issues with channel datastores in func_odbc.c. ... * Fixed a potential memory leak when an existing datastore is manually destroyed by inline code instead of calling ast_datastore_free(). (closes issue ASTERISK-17948) Reported by: Archie Cobbs Review: https://reviewboard.asterisk.org/r/1687/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352291 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Joshua Colp	d7a25693c9	Move RTP timeout check to before bridged channel check so it is actually executed. ... (issue ASTERISK-19179) Reported by: TSAREGORODTSEV Yury (closes issue ASTERISK-14534) Reported by: kriborgen Patches: chan_sip.patch uploaded by kriborgen (license 6138) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	e41e647429	Fix grammar of comment. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352230 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	e2a7c10b97	Fix blind transfers from failing if an 'h' extension is present. ... This prevents the 'h' extension from being run on the transferee channel when it is transferred via a native transfer mechanism such as SIP REFER. (closes ASTERISK-19173) Reported by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by Mark Michelson (license 5049) Review: https://reviewboard.asterisk.org/r/1685 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	11105d0cd3	Correctly apply FAXOPT settings (V17, V27, V29) before starting spandsp layer ... While the FAXOPT function could be used to set the modem capabilities, the input to that function was not being applied correctly to the spandsp layer. This patch applies the current model capabilities before starting the spandsp layer. (closes issue: ASTERISK-16409) Reported by: Kristijan Vrban Tested by: Matt Jordan, Matthew Nicholson Patches: spandsp-modems-1.8.diff uploaded by mnicholson (license 5081) spandsp-modems-10.diff uploaded by mnicholson (license 5081) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352144 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago