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AgeCommit message (Collapse)Author
2012-07-19Turn on debugging for pjproject so we can get a better idea of what is ↵Joshua Colp
causing the generic CCSS test crash. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-18callid logging: Issue test events when the callid is changed for a channelJonathan Rose
Review: https://reviewboard.asterisk.org/r/2054/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-18Resolve severe memory leak in CEL logging modules.Kevin P. Fleming
A customer reported a significant memory leak using Asterisk 1.8. They have tracked it down to ast_cel_fabricate_channel_from_event() in main/cel.c, which is called by both in-tree CEL logging modules (cel_custom.c and cel_sqlite3_custom.c) for each and every CEL event that they log. The cause was an incorrect assumption about how data attached to an ast_channel would be handled when the channel is destroyed; the data is now stored in a datastore attached to the channel, which is destroyed along with the channel at the proper time. (closes issue AST-916) Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2053/ ........ Merged revisions 370205 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370206 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-18Ensure that all ast_datastore_info structures are 'const'.Kevin P. Fleming
While addressing a bug, I came across a instance of 'struct ast_datastore_info' that was not declared 'const'. Since the API already expects them to be 'const', this patch changes the declarations of all existing instances that were not already declared that way. ........ Merged revisions 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370184 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370187 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-18Fix a crash in pjnath when starting an ICE connectivity check and ↵Joshua Colp
immediately destroying the ICE session. The initial ICE connectivity check is scheduled as a timer item that is to be executed immediately. It is possible for this timer item to start executing while the ICE session it is working on is destroyed. To reduce the chance of this any timer items that need to be immediately executed will be executed within the thread that has started the initial ICE connectivity check. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-18Fix a crash occurring as a result of excess stack usage.Joshua Colp
This fix involves moving the allocation of some temporary codec structures to the heap and also reduces the number of maximum payloads to something more sane for both regular and low memory builds. (closes issue ASTERISK-20140) Reported by: jonnt git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-18Added option 'interdigit_timer' to unistim.conf to make able controll ↵Igor Goncharovskiy
hardcoded dial timeout constant. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-17Add pubsub unsubscription support so subscriptions do not linger for MWI and ↵Joshua Colp
device state progatation. The pubsub code did not attempt to remove subscriptions at all. This has now changed so that if a client is being disconnected it will unsubscribe. It will also unsubscribe at connection time so if it unexpectedly disconnected duplicate subscriptions will not occur. (closes issue ASTERISK-19882) Reported by: mattvryan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-17Fix a crash as a result of propagating MWI or device state over XMPP when ↵Joshua Colp
the client is disconnected. The MWI and device state propagation code wrongly assumes that an XMPP client connection will remain established at all times. This fix corrects that by making the lifetime of the subscription the same as the lifetime of the connection itself. As the connection is established and disconnected the subscription itself is created and destroyed. (closes issue ASTERISK-18078) Reported by: elguero git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16Code cleanup and bugfix in chan_sip outboundproxy parsing.Walter Doekes
The bug was clearing the global outboundproxy when a peer-specific outboundproxy was bad. The cleanup reduces duplicate code. Review: https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark Michelson ........ Merged revisions 370131 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370132 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16Fix an issue where a service discovery request could crash Asterisk.Joshua Colp
A server sending a service discovery request to us may or may not put a from attribute in the message. If the from attribute is present use it in the to attribute for the result. If the from attribute is not present do not add a to attribute. (issue ASTERISK-16203) Reported by: wubbla git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16Fix a bug where some XMPP servers would reject authentication. We need to ↵Joshua Colp
use the user portion of the JID and not the full configured username. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16Add missing namespace for old non-SASL based authentication.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16Fix a bug exposed by the testsuite where text streams would no longer be ↵Joshua Colp
parsed correctly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16Add comments about the BUILD_NATIVE changeKinsey Moore
This is a significant change and mention of it should have gone into UPGRADE.txt and CHANGES. ........ Merged revisions 370081 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370082 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370083 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16Fix an issue where specifying the resource in the username would cause ↵Joshua Colp
authentication to fail. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16Add support for SIP over WebSocket.Joshua Colp
This allows SIP traffic to be exchanged over a WebSocket connection which is useful for rtcweb. Review: https://reviewboard.asterisk.org/r/2008 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16Deactivate timer for dialing entered number on hook switch hang up.Igor Goncharovskiy
(closes issue ASTERISK-19554) Reported by: Stefano Villani git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16Add French translation for chan_unistim phones on-screen menus. Igor Goncharovskiy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-13Reduce memory consumption and add the H.264 and H.263 modules I shamefully ↵Joshua Colp
neglected to add. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-13Add support for parsing SDP attributes, generating SDP attributes, and ↵Joshua Colp
passing it through. This support includes codecs such as H.263, H.264, SILK, and CELT. You are able to set up a call and have attribute information pass. This should help considerably with video calls. Review: https://reviewboard.asterisk.org/r/2005/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-13live_ast: don't set working directoryTzafrir Cohen
contrib/scripts/live_ast currently assumes that it is being run from the top-level directory of the source tree. It creates a script that will also set the working directory. This fix avoids the need to set the working directory if the caller sets LIVE_AST_BASE_DIR instead. It relies on realpath for that. If realpath is not available, it will fall back to the original behaviour. Review: https://reviewboard.asterisk.org/r/2027/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-12Handle deprecated (aliased) option names with the config options apiTerry Wilson
Add a simple way to register "deprecated" option names that alias to a different "current" name. Review: https://reviewboard.asterisk.org/r/2026/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370043 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-12Add missing ast_hangup() calls on some analog exception paths.Richard Mudgett
Make starting analog_ss_thread() or __analog_ss_thread() failure paths hangup the channel. ........ Merged revisions 370017 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370025 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-12Include Expires header for SIP PUBLISH requestsKinsey Moore
RFC3903 requres SIP PUBLISH requests to have Expires headers, so add them. Review: https://reviewboard.asterisk.org/r/2003/ Patch-by: gareth ........ Merged revisions 370014 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370015 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370016 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-12Prevent double uri_escaping in chan_sip when pedantic is enabledKinsey Moore
If pedantic mode is enabled, outbound invites will have double-escaped contacts. This avoids setting an already-escaped string into a field where it is expected to be unescaped. (closes issue ASTERISK-20023) Reported by: Walter Doekes ........ Merged revisions 369993 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369994 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-12Correct Documentation For DEC FunctionMichael L. Young
The documentation for DEC in func_math.c was incorrect. Looks like a copy and paste error. (Closes issue ASTERISK-20095) Reported by: Billy Chia Tested by: Michael L. Young Patches: func_math.patch uploaded by Billy Chia (license 6381) ........ Merged revisions 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369971 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-12Reverting last merge since it wasn't completed properly.Michael L. Young
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-12Correct Documentation For DEC FunctionMichael L. Young
The documentation for DEC in func_math.c was incorrect. Looks like a copy and paste error. (Closes issue ASTERISK-20095) Reported by: Billy Chia Tested by: Michael L. Young Patches: func_math.patch uploaded by Billy Chia (license 6381) ........ Merged revisions 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369972 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11Named ACLs: Introduces a system for creating and sharing ACLsJonathan Rose
This patch adds Named ACL functionality to Asterisk. This allows system administrators to define an ACL and refer to it by a unique name. Configurable items can then refer to that name when specifying access control lists. It also includes updates to all core supported consumers of ACLs. That includes manager, chan_sip, and chan_iax2. This feature is based on the deluxepine-trunk by Olle E. Johansson and provides a subset of the Named ACL functionality implemented in that branch. For more information on this feature, see acl.conf and/or the Asterisk wiki. Review: https://reviewboard.asterisk.org/r/1978/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11Allow the REALTIME() function to report errors back to the caller.Tilghman Lesher
Also, do more error checking on the arguments specified to the REALTIME() function and clarify the documentation. While I was editing the file, a few coding guidelines fixups, as well. Review: https://reviewboard.asterisk.org/r/2031/ ........ Merged revisions 369937 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369938 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11Don't perform an XInclude to a document node that may not always be presentMatthew Jordan
Because some of the manager events are defined in the top of the source, due to the macro calls not containing all necessary information to have the documentation colocated with the call itself, several include statements were failing when built with 'make'. While this did not cause any problems in compilation or validation, it did result in a number of warnings being dumped to stderr. This patch changes those references such that they always resolve, regardless of the documentation build options. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11Do not consider failure to read the configuration file in chan_motif to be a ↵Joshua Colp
show stopper for loading Asterisk by returning decline instead of failure. (closes issue ASTERISK-20103) Reported by: Terry Wilson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-11Fix validation errors when producing documentation using default build scriptMatthew Jordan
The awk script parses out the first instance of the DOCUMENTATION tag that it finds within a file. If a file did not previously have a DOCUMENTATION tag but received one due to it having an AMI event, then the XML fragment associated with the AMI event was erroneously placed in the resulting XML file. Without the python scripts, these XML fragments will not validate. This patch adds DOCUMENTATION tags at the top of those files that did not previously have them to prevent the awk script from pulling AMI event documentation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10Add some additional documentation for core AMI eventsMatthew Jordan
This patch adds some basic documentation for a number of modules. This includes core source files in Asterisk (those in main), as well as chan_agent, chan_dahdi, chan_local, sig_analog, and sig_pri. The DTD has also been updated to allow referencing of AMI commands. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10Fix failing SDP_offer_answer testKinsey Moore
Asterisk now generates image stream declinations with the same transport case that it used to before the stream declination improvements. (udptl vs UDPTL) (closes issue SWP-4736) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10Add additional description stanza names from the old Google Talk protocol ↵Joshua Colp
which is used with Google Voice. (closes issue ASTERISK-20114) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369898 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10Respect codec preference order when adding codecs to a media description.Joshua Colp
This change allows an endpoint in motif.conf to be configured with a preference of G.722 and fallback of ulaw. With Google this allows communication with Google Talk clients to use G.722 while when using Google Voice ulaw will be used. (closes issue ASTERISK-20114) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10Improve Goto and GotoIf related documentationKinsey Moore
Correct documentation on labeliftrue and labeliffalse parameters of GotoIf() and update several other locations that use the same syntax. (closes issue ASTERISK-20007) Patch-by: Leif Madsen Reported-by: WIMPy ........ Merged revisions 369869 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369871 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369872 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10Fix initial loading problem with res_curlMatthew Jordan
When the OpenSSL duplicate initialization issues were resolved in r351447, res_curl could fail to load if it checked SSL_library_init after SSL initialization completed. This is due to the SSL_library_init stub returning a value of 0 for success, as opposed to a value of 1. OpenSSL uses a value of 1 to indicate success - in fact, SSL_library_init is documented to always return 1. Interestingly, the CURL libraries actually checked the return value - the fact that nothing else that depends on OpenSSL was having problems loading probably means they don't check the return value. (closes issue AST-924) Reported by: Guenther Kelleter patches: (AST-924.patch license #6372 uploaded by Guenther Kelleter) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10Add required items for Google video support.Joshua Colp
This adds legacy STUN support for RTCP sockets, adds RTCP candidates to the Google transport information, and adds required codec parameters. (closes issue ASTERISK-20106) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09When receiving a STUN binding request send one out as the Google Talk client ↵Joshua Colp
uses this as a method to determine if the remote party is still reachable or not. Failure to do this results in the Google Talk client ignoring RTP packets after a specific period of time. This is also done as a result of receiving a STUN binding request so that the username information can be used from the inbound request, thus not requiring it to be stored on a per candidate basis. (closes issue ASTERISK-20107) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09Add support for exposing the received contact URI and also for setting the ↵Joshua Colp
request URI in messages. (closes issue AST-911) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09Force the clock rate of G.722 to be 16000 when using the Google transports ↵Joshua Colp
as it is 8000 elsewhere. (closes issue ASTERISK-20105) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09Document that multiple endpoints using the same connection is not supported.Joshua Colp
(closes issue ASTERISK-20104) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09Add Digium phones context to sip_notify sample config.Jason Parker
This makes it so that they can be reconfigured remotely. (closes issue ASTERISK-19910) ........ Merged revisions 369818 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369819 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09Fix an issue where media would not flow for situations where the legacy STUN ↵Joshua Colp
code is in use. The STUN packets should *not* be blocked by strict RTP. (closes issue ASTERISK-20102) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09Add additional namespaces for Google Talk which are used for the gmail client.Joshua Colp
(closes issue ASTERISK-20101) Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369816 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09Fix dependency to be on res_xmpp. Long ago in a galaxy far far away it used ↵Joshua Colp
to use res_jabber. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369811 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-09chan_sip: Fix small behavioral change accidentally introduced in r369750Jonathan Rose
When removing the warning for AST_CONTROL_FLASH from sip_indicate, I also inadvertently changed the return value, which would likely make the indication not be sent in audio. This fixes that while still removing the warning message. ........ Merged revisions 369792 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369793 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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