asterisk

Commit Graph

Author	SHA1	Message	Date
Kinsey Moore	6300aa6ae4	Make sure things compile... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	99aa02d17f	Transition MWI to Stasis-core Remove MWI's dependency on the event system by moving it to Stasis-core. This also introduces forwarding topic pools in Stasis-core which aggregate many dynamically allocated topics into a single primary topic. Review: https://reviewboard.asterisk.org/r/2368/ (closes issue ASTERISK-21097) Patch-by: Kinsey Moore git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	ad5f3a5759	tcptls: Prevent unsupported options from being set AMI, HTTP, and chan_sip all support TLS in some way, but none of them support all the options that Asterisk's TLS core is capable of interpreting. This prevents consumers of the TLS/SSL layer from setting TLS/SSL options that they do not support. This also gets tlsverifyclient closer to a working state by requesting the client certificate when tlsverifyclient is set. Currently, there is no consumer of main/tcptls.c in Asterisk that supports this feature and so it can not be properly tested. Review: https://reviewboard.asterisk.org/r/2370/ Reported-by: John Bigelow Patch-by: Kinsey Moore (closes issue AST-1093) ........ Merged revisions 383165 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383166 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383167 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	cacc356bbe	When a session timer expires during a T.38 call, re-invite with correct SDP When a session timer expires during a dialog that has re-negotiated to T.38 and Asterisk is the refresher, Asterisk will send a re-INVITE with an SDP containing audio media only. This causes some hilarity with the poor fax session under weigh. This patch corrects that by sending T.38 parameters if we are in the middle of a T.38 session. (closes issue ASTERISK-21232) Reported by: Nitesh Bansal patches: dont-send-audio-reinvite-for-sess-timer-in-t38-call.patch uploaded by nbansal (License 6418) ........ Merged revisions 383124 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383125 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	00e9ffb907	Include the Username field in SIP Registry events when Status is registered In ASTERISK-17888, the AMI Registry event during SIP registrations was supposed to include the Username field. Somehow, one of the events was missed. This patch corrects that - the Username field should be included in all AMI Registry events involving SIP registrations. (issue ASTERISK-17888) (closes issue ASTERISK-21201) Reported by: Dmitriy Serov patches: chan_sip.c.diff uploaded by Dmitriy Serov (license 6479) ........ Merged revisions 382847 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382848 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382852 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Igor Goncharovskiy	ef64b29f8b	Fix core dump on CLI usage Fix issue with 'unistim show info' CLI command when device connected not configured ........ Merged revisions 382827 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	09ecb25e08	Added an option to disallow music on hold Added an option "discard_remote_hold_retrieval" (default "no") that if set does not trigger the music on hold event. This essentially stops telling the peer to start music on hold. (issue ABE-2899) Reported by: Denis Alberto Martinez Review: https://reviewboard.asterisk.org/r/2336/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	b4a010e958	chan_sip: Update the via header when relaying SMS MESSAGE Prior to this change, certain conditions for sending the message would result in an address of '(null)' being used in the via header of the SIP message because a NULl value of pvt->ourip was used when initially generating the via header. This is fixed by adding a call to build_via when the address is set before sending the message. (closes issue ASTERISK-21148) Reported by: Zhi Cheng Patches: 700-sip_msg_send_via_fix.patch uploaded by Zhi Cheng (license 6475) ........ Merged revisions 382739 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382746 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	f6f6bc7b59	Remove unused function After r382670, get_ip_and_port_from_sdp was no longer used. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	12748bc735	Don't reset the RTP address on a glare re-INVITE Originally, way back in r201583, we added the alternate RTP address so that the RTP engine would expect to receive audio from a new source when a glare re-INVITE occurred. In r382589, we remove the alternate RTP source, as the 'secret' probation mode allows for switching to a new RTP source when a previous source stops sending RTP. At the time, it seemed appropriate to set the RTP source based on the information in the glared re-INVITE. Unfortunately, that doesn't work so well - in a glared re-INVITE that occurs with no SDP - such as in a connected line update that glances - we'll set the RTP source to an invalid address. In subsequent re-INVITE requests from this Asterisk instance, we'll then send an invalid media address, which will result in the remote side sending a 488. Whoops. There isn't any need to reset the RTP source - if we're using strictrtp, we'll simply synchronize to a new source when we stop getting packets from the old one. If we aren't using strictrtp, then again there shouldn't be a problem. Note that the Asterisk Test Suite's connectedline test caught this error. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	80b8c2349c	Add a 'secret' probation strictrtp mode to handle delayed changes in RTP source Often, Asterisk may realize that a change in the source of an RTP stream is about to occur and ask that the RTP engine reset it's lock on the current RTP source. In certain scenarios, it may take awhile for the new remote system to send RTP packets, while the old remote system may continue providing RTP during that time period. This causes Asterisk to re-lock onto the old source, thereby rejecting the new source when the old source stops sending RTP and the new source begins. This patch prevents that by having a constant secondary, 'secret' probation mode enabled when an RTP source has been chosen. RTP packets from other sources are always considered, but never chosen unless the current RTP source stops sending RTP. Review: https://reviewboard.asterisk.org/r/2364 (closes issue AST-1124) Reported by: John Bigelow Tested by: John Bigelow (closes issue AST-1125) Reported by: John Bigelow Tested by: John Bigelow ........ Merged revisions 382573 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	8d5c36c9bb	Add RFC 3327 Path header support to chan_sip This patch adds support for RFC 3327 "Path" headers. This can be enabled in sip.conf using the 'supportpath' setting, either on a global basis or on a peer basis. This setting enables Asterisk to route outgoing out-of-dialog requests via a set of proxies by using a pre-loaded route-set defined by the Path headers in the REGISTER request. This patch also adds Realtime support for dynamically updating the Path information for a peer. A huge thank-you to Klaus Darillion and Olle E Johansson for their efforts in writing this patch. Review: https://reviewboard.asterisk.org/r/2235/ Review: https://reviewboard.asterisk.org/r/991/ (closes issue ASTERISK-16884) Reported by: klaus3000 Tested by: klaus3000, oej, mjordan patches: path-1.8.0-patch.txt uploaded by klaus3000 (License 5054) oolong-path-support-trunk in team branch by oej (License 5267) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Igor Goncharovskiy	469ca1c71d	Fix several unreleased mutex locks that cause problem with processing calls	12 years ago
Michael L. Young	a3ad8b28e6	Fix / Clean Up Some Items To Handle The New auto_* NAT Options The original report had to do with a realtime peer behind NAT being pruned and the peer's private address being used instead of its external address. Upon debugging, it was discovered that this was being caused by the addition of the auto_force_rport and auto_comedia settings. This patch does the following: * Adds a missing note to the CHANGES file indicating that the default global nat setting is auto_force_rport * Constify the 'req' parameter for check_via() * Add calls to check_via() in a couple of places in order for the auto_* settings to do their job in attempting to determine if NAT is involved * Set the flags SIP_NAT_FORCE_RPORT and SIP_PAGE2_SYMMETRICRTP if the auto_* settings are in use where it was needed * Moves the copying of peer flags up in build_peer() to before they are used; this fixes the realtime prune issue * Update the contrib/realtime schemas to allow the nat column to handle the different nat setting combinations we have This patch received a review and "Ship It!" on the issue itself. (closes issue ASTERISK-20904) Reported by: JoshE Tested by: JoshE, Michael L. Young Patches: asterisk-20904-nat-auto-and-rt-peersv2.diff Michael L. Young (license 5026) ........ Merged revisions 382322 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382323 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	1a34b465bc	Prevent deadlock in chan_iax2 when attempting to set caller ID A deadlock can occur in chan_iax2 when it attempts to set the caller ID, as it already holds the iax2 private lock and improperly fails to obtain the channel lock before calling ast_set_callerid. By not safely obtaining the channel lock, a locking inversion can take place, causing a deadlock. This patch solves this by calling the required deadlock avoidance functions that obtain the channel lock before setting the caller ID. Thanks to Pavel for fixing my syntax errors and testing this patch out. (closes issue ASTERISK-21128) Reported by: Pavel Troller Tested by: Pavel Troller patches: ASTERISK-21128-1.8.diff uploaded by mjordan (license 6283) ASTERISK-21128-modified-1.8.diff uploaded by Pavel Troller (license 6302) ........ Merged revisions 382233 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382234 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382236 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	de90681293	More places to eliminate the cast to argv but were not giving warnings. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	31f08344ee	Fix compiler warning by eliminating the need for a cast. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382203 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	e0b49e7331	Relax dialog checking in get_sip_pvt_byid_locked so it works when the dialog is forked. (closes issue ASTERISK-20638) Reported by: eelcob Patches: pedantic-call-pickup-from-tag.patch uploaded by eelcob (license 6442) ........ Merged revisions 382171 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382174 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	d33d9c1781	Correct RPID parsing for unquoted display-name. Parsing Remote-Party-ID will now succeed if display-name is of the *(token LWS) kind and not just the quoted-string kind. Review: https://reviewboard.asterisk.org/r/2341/ ........ Merged revisions 382107 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382108 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382109 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	b34479ad37	More called details fixup for skinny. Basically sets the callerid and callername to the first device talked to for the purposes of putting the the calls made log on the device. Does not affect the device displaying who the device is currently talking to. Also some minor changes to use sub->exten in lieu of l->lastnumberdialed. (closes issue ASTERISK-21095) Reported by: wedhorn Tested by: snuffy, myself Patches: skinny-calllogsoutbound03.diff uploaded by wedhorn (license 5019) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	0928f45794	Add prinotify messages to skinny. Adds both fixed and variable prinotify messages and clearprinotify messages to skinny. Also adds cli function for pushing messages to devices. i Initial code by snuffy, expanded by myself to include fixed messages. (closes issue ASTERISK-21091) Reported by: snuffy Tested by: snuffy, myself Patches: skinny-prinotify02.diff uploaded by wedhorn (license 5019) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	f6abcce7c1	Set the sin_family on the bind address socket during initialization Somehow, chan_jingle has managed to operate for years without setting the sin_family on its bindaddr socket. This patch properly sets the field during initial module load to AF_INET. Note that the patch on the issue was modified slightly to change the initialization of the socket from allocation of a chan_jingle private to the module initialization, as the bindaddr object (which is static) only needs to have the address set once. (closes issue ASTERISK-19341) Reported by: andre valentin patches: 0105-chan_jingle.patch uploaded by avalentin (License 6064) ........ Merged revisions 381975 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 381976 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381977 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	d1f8e338b0	Add The Status Of A Module To The Output Of "CLI> module show" When a module's configuration is not loadable, we still load the module but it is not in a running state. When trying to troubleshoot, let's say, why chan_motif is ignoring inbound XMPP traffic, there is no way to indicate that a loaded module is not currently running. (closes issue ASTERISK-21108) Reported by: Rusty Newton Tested by: Michael L. Young Patches: asterisk-21108_add_status-v2.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2331/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	0d553eece8	Add serviceURL stuff to skinny. Patch adds all the packet and structure stuff to skinny to enable setting service URLs in skinny, such as corporate directories. This stuff is only relevant during load/unload as when activated. Also some minor changes removing duplicated counting of addons and speedials in handle_skinny_show_devices. Review: https://reviewboard.asterisk.org/r/2321/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381718 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	96025d5dfc	Fixup skinny CLI completion. Auto complete for skinny debug allows multiple options and negation, also add debug all option. Usage example: 'skinny debug all -packets' (each can be autocompleted including -packet). Change show device to use device name. Remove the duplicate ast_strdup's from place calling device complete return immediately from complete devicename and complete linename so that multiple options are displayed on the CLI if more than one option available. Review: https://reviewboard.asterisk.org/r/2333/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	2ebb9863ea	Don't send presencestate information if the state is invalid Previously, presencestate information was sent whenever the state was not NOT_SET. When r381594 actually returned INVALID presence state in all the places it was supposed to, it caused chan_sip to start adding presence state information to NOTIFY requests that it previously would not have added. chan_sip shouldn't be adding presence state information when the provider is in an invalid state; users can't set the state to invalid and an invalid state always implies that the provider is in an error condition. (issue AST-1084) ........ Merged revisions 381613 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381615 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	8a7dd2f408	Fix a crash that occurred when a BYE was received on a replaced dialog. Reference counting for the channel and its tech_pvt got messed up at some point between 1.8 and 11. The result was that if a BYE for a dialog that had been replaced (via an INVITE with Replaces) was received, Asterisk would crash due to trying to access data on a channel that was no longer there. The fix I introduced is to remove code that both unrefs the sip_pvt and sets the channel's tech_pvt to NULL when an INVITE with Replaces is handled. This way when a BYE is received, the tech_pvt will be non-NULL and so the BYE can be processed and not cause a crash. (closes issue ASTERISK-20929) reported by Kristopher Lalletti patches: ASTERISK-20929.patch uploaded by Mark Michelson (License #5049) ........ Merged revisions 381566 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381568 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	f008baddac	chan_sip: Use video and text crypto attributes to append RTP profiles to SDP Some bad copy/pasting resulted in using the audio crypto attribute for both text and video RTP. Also the audio crypto isn't set until after these, so it was really just bad all around. (closes ASTERISK-20905) Reported by: Kristopher Lalletti patches: rtp_crypto_video_text.diff uploaded by Jonathan Rose (license 6182) ........ Merged revisions 381553 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	d04ab3c645	Add CLI configuration documentation This patch allows a module to define its configuration in XML in source, such that it can be parsed by the XML documentation engine. Documentation is generated in a two-pass approach: 1. The documentation is first generated from the XML pulled from the source 2. The documentation is then enhanced by the registration of configuration options that use the configuration framework This patch include configuration documentation for the following modules: * chan_motif * res_xmpp * app_confbridge * app_skel * udptl Two new CLI commands have been added: * config show help - show configuration help by module, category, and item * xmldoc dump - dump the in-memory representation of the XML documentation to a new XML file. Review: https://reviewboard.asterisk.org/r/2278 Review: https://reviewboard.asterisk.org/r/2058 patches: on review 2058 uploaded by twilson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	edf0483f4f	Remove extraneous stuff from r381470. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381471 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	ce8101c6c6	Add back sending dialnumber to skinny. Don't know why it seemed to work during testing, but it really is needed for protocol v17 (and probably above). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381470 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	fffcdb0361	Respect callerid presentation in skinny. Fix chan_skinny so that it respects callerID presentation of inbound calls to device and a couple of other minor fixes: 145 packet (add OCTAL_FROM amd callerid), and dont send dialednumber message if protocol >= 17. (closes issue ASTERISK-21066) Reported by: snuffy Tested by: snuffy, myself Patches: skinny-respect-clid-restrictions-v2.diff uploaded by snuffy (license 5024) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381465 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Sean Bright	86a537c271	Use a shuffling algorithm to find unused IAX2 call numbers. While adding red-black tree containers to astobj2 in r376575, Richard pointed out the way chan_iax2 finds unused call numbers will prevent ao2_container integrity checks at runtime. This patch removes the ao2_container and instead uses fixed sized arrays and a modified Fisher-Yates-Durstenfeld shuffle to maintain the call number list. While the locking semantics are similar to the ao2_container implementation, this implementation should be faster and more memory efficient. Review: https://reviewboard.asterisk.org/r/2288/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	81fa307af7	Fix some more REF_DEBUG-related build errors When sip_ref_peer and sip_unref_peer were exported to be usable in channels/sip/security_events.c, modifications to those functions when building under REF_DEBUG were not taken into account. This change moves the necessary defines into sip.h to make them accessible to other parts of chan_sip that need them. ........ Merged revisions 381282 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381285 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	98ef0d3215	Fix some issues with skinny callid. Add extra string to transmit_callinfo_var, Only set string2 to tonum for outgoing calls and changes to send_callinfo and push_callinfo to not set callid name to last number. (closes issue ASTERISK-21063) Reported by: wedhorn Tested by: snuffy, myself Patches: skinny-callinfoupdate03.diff uploaded by wedhorn (license 5019) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381195 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	5b236ee647	Make ast_do_masquerade() a void function. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381086 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	345253a50e	Fixed failing test from r380696. When I added my extensive suite of session timer unit tests, apparently one of them was failing and I never noticed. If neither Min-SE nor Session-Expires is set in the header, it was responding with a Session-Expires of the global maxmimum instead of the configured max for the endpoint. (issue ASTERISK-20787) ........ Merged revisions 380973 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 380974 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380977 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	c0832b4765	Fix reload skinny with active devices. Patch ensures that d->activeline and l->activesub are moved over to the new device and line so that on callend the appropriate subs can be found to complete hangup before device resets. (closes issue ASTERISK-16610) Reported by: wedhorn Tested by: snuffy, myself Patches: skinny-reloadactive01.diff uploaded by wedhorn (license 5019) ........ Merged revisions 380942 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	44872e797c	Reset skinny vmexten and immeddial char on reload. Make skinny reset vmexten and immeddial to '\0' on reload to ensure that it is set to '\0' if the appropriate item is removed/commented in skinny.conf. Also small fix re immeddial char in skinny.conf and add immedial setting to skinny show settings. (closes issue ASTERISK-21037) Reported by: snuffy Tested by: snuffy, myself Patches: immed_dial_fix.diff uploaded by snuffy (license 5024) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Damien Wedhorn	8bb9aa2f6d	Add variable length displayprompt packet to skinny and use octals. Add new variable length displayprompt packet (0x0145) to skinny. Uses the new packet if the device is reporting protocol versions >= 17. Add the use of octal codes for sending prompts to both the new and old displayprompt messages (also cleaned up soft_key_template_default to use the defined octal codes). Review: https://reviewboard.asterisk.org/r/2294/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380792 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	ae1421e04d	chan_iax2: Fix compile error if MALLOC_DEBUG enabled. NEVER INCLUDE astmm.h DIRECTLY!! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380774 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Damien Wedhorn	523e472e1a	Adds variable length callinfo packets to skinny. Add packet 0x014A (variable length call info messages) to skinny for newer firmware. Plenty of unknown information but includes the equivalent functionality as the fixed size callinfo packet already included. Only send this packet if protocol reported is >= 17. Review: https://reviewboard.asterisk.org/r/2290/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	5899e13112	Process session timers, even if Session-Expires header is missing Previously, Asterisk only processed session timer information if both the 'Supported: timer' and 'Session-Expires' headers were present. However, the Session-Expires header is optional. If we were to receive a request with a Min-SE greater than our configured session-expires, we would respond with a 'Session-Expires' header that was too small. This patch cleans the situation up a bit, always processing timer information if the 'Supported: timer' header is present. (closes issue ASTERISK-20787) Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/2299/ ........ Merged revisions 380696 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 380698 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	d6e05d5bf3	Move IAX firmware related functionality into separate files. This patch is mostly a reorganization of existing code with a few exceptions: * Added doxygen comments to all of the extracted functions. * Split reload_firmware(int unload) into iax_firmware_reload() and iax_firmware_unload() for readability. * Create iax_firmware_traverse() to support the 'iax2 show firmware' CLI command. * Renamed iax_check_version() to iax_firmware_get_version() and change its arguments and return value so that it returns a success/failure value and sets the selected version into an out parameter to avoid confusion with failure and version 0. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380695 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	6458a6572b	chan_dahdi: Fix "dahdi show channels group" for groups greater than 31. The variable type used was not large enough to hold a group bit field. ........ Merged revisions 380572 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 380575 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380576 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	01309cf41e	Unregister SIP provider API if module load is declined A user in #asterisk ran into a problem where a configuration error prevented the chan_sip module from being loaded. Upon fixing their configuratione error, they could no longer load the chan_sip module. This was because the configuration checking happened after the SIP provider was registered with the Asterisk core, and subsequent attempts to load the SIP module failed as the provider was already registered. Since we want to detect any failure in registering chan_sip as early as possible (as that could be emblematic of a deeper mismatch between module and Asterisk core), this patch does not change the registration location, but does ensure that if a module load is declined, we unregister the module as the SIP api provider. ........ Merged revisions 380480 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380500 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	8018bdd8e1	Perform case insensitive comparisons for T.38 attributes RFC5347 section 2.5.2 states the following: ... The attribute "T38MaxBitRate" was once incorrectly registered with IANA as "T38maxBitRate" (lower-case "m"). In accordance with T.38 examples and common implementation practice, the form "T38MaxBitRate" SHOULD be generated by implementations conforming to this package. In general, it is RECOMMENDED that implementations of this package accept lowercase, uppercase, and mixed upper/lowercase encodings of all the T.38 attributes. ... Asterisk currently does not perform case insensitive matching on the T.38 attributes. This causes the T38MaxBitRate attribute to be negotiated at 2400 baud instead of 14400 (or whatever value you actually wanted). This patch makes it so that when we compare T.38 attributes, we do so in a case insensitive fashion. Note that while the issue reporter did not directly write the patch, they contributed to it (and would have provided one themselves if the license had gone through a tad faster), and hence get attribution for it. Review: https://reviewboard.asterisk.org/r/2298/ (closes issue ASTERISK-20897) Reported by: Eric Hill Tested by: Eric Hill patches: -- uploaded by Eric Hill ........ Merged revisions 380458 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 380465 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	693d609081	Move the ancillary iax2 source files into a separate sub-directory. This patch just moves the IAX2 source and header files into a separate iax2 sub-directory in the channels directory, similar to how the sip source files are structured. The only thing that was added was an #ifndef to protect provision.h from multiple inclusion. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	8cc7aea09b	chan_agent: Prevent multiple channels from logging in as the same agent. Multiple channels logging in as the same agent can result in dead channels waiting for a condition signal that will never come because another channel thread stole it. A symptom is chan_sip repeatedly generating warning messages about rescheduling autodestruction of dialogs with an agent channel owner. * Made only login_exec() (the app AgentLogin) clear the agent_pvt->chan pointer to prevent multiple channels from logging in as the same agent. agent_read(), agent_call(), and agent_set_base_channel() no longer disconnect the agent channel from the agent_pvt. This also eliminates the need to keep checking for agent_pvt->chan being NULL. * Made agent_hangup() not wake up the AgentLogin agent thread until it is done. * Made agent_request() not able to get the agent until he has logged in and any wrapup time has expired. * Made agent_request() use ast_hangup() instead of agent_hangup() to correctly dispose of a channel. * Removed agent_set_base_channel(). Nobody calls it and it is a bad thing in general. * Made only agent_devicestate() determine the current device state of an agent. Note: Agent group device states have never been supported. Review: https://reviewboard.asterisk.org/r/2260/ ........ Merged revisions 380364 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 380384 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	e06cd59e04	Corrected crypto tag in SDP ANSWER for SRTP. (again) The original fix (r380043) for getting Asterisk to respond with the correct tag overlooked some corner cases, and the fact that the same code is in 1.8. This patch moves the building of the crypto line out of sdp_crypto_process(). Instead, it merely copies the accepted tag. The call to sdp_crypto_offer() will build the crypto line in all cases now, using a tag of "1" in the case of sending offers. (closes issue ASTERISK-20849) Reported by: José Luis Millán Review: https://reviewboard.asterisk.org/r/2295/ ........ Merged revisions 380347 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 380350 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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7668 Commits (05ec2860df560477546564e07058f8f9f1e17cfc)