asterisk

Commit Graph

Author	SHA1	Message	Date
Terry Wilson	d34d46a16e	Merged revisions 319204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319204 \| twilson \| 2011-05-16 13:17:43 -0500 (Mon, 16 May 2011) \| 11 lines Merged revisions 319202 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r319202 \| twilson \| 2011-05-16 11:00:21 -0700 (Mon, 16 May 2011) \| 4 lines Unlink a peer from peers_by_ip when expiring a registration Review: https://reviewboard.asterisk.org/r/1218/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	980d896bde	Merged revisions 319145 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319145 \| dvossel \| 2011-05-16 10:57:26 -0500 (Mon, 16 May 2011) \| 9 lines Merged revisions 319144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r319144 \| dvossel \| 2011-05-16 10:56:16 -0500 (Mon, 16 May 2011) \| 2 lines Fixes issue with peer ref-counting during handle_request_subscribe. (closes issue #19293) Reported by: irroot ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	8e719c62b0	Merged revisions 319142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319142 \| mnicholson \| 2011-05-16 10:53:26 -0500 (Mon, 16 May 2011) \| 8 lines Make sure tcptls_session exists before dereferencing it. (closes issue #19192) Reported by: stknob Patches: 10-tcptls-unreachable-peer-segfault.patch uploaded by Chainsaw (license 723) Tested by: vois, Chainsaw ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	32d43ebe19	When a error in T.38 negotiation happens or its rejected on a channel the state of the channel reverts to unknown this should be rejected. this is important for negotiating T.38 gateway see #13405 This patch adds a option T38_REJECTED that behaves as T38_DISABLED except it reports state rejected. Trivial Change to res_fax to honnor UNAVAILABLE and REJECTED states. (closes issue #18889) Reported by: irroot Tested by: irroot, darkbasic, mnicholson Review: https://reviewboard.asterisk.org/r/1115 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319087 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	969a317d81	Add activatesub and dialandactivate sub. When called, activatesub first cleans up the active sub and then handles the sub passed. dialandactivatesub first sets sub->exten and then calls activatesub. Revise handle_offhook to utilise the callid sent to chan_skinny. Some other minor fixes especially around d->hookstate (which still needs some more work). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	547490144c	Merged revisions 318917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318917 \| bbryant \| 2011-05-13 13:56:04 -0400 (Fri, 13 May 2011) \| 11 lines This patch allows TCP peers into the ast_db where they were previously restricted. (closes issue #18882) Reported by: cmaj Patches: patch-chan_sip-1.8.3-rc2-allow-tcp-peer-store-db-and-readonly-rt-backend.diff.txt uploaded by cmaj (license 830) Tested by: cmaj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318918 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	54bb8a0ca8	Move exten used for dialing from device to subchannel. There were some issues where if a simple switch was cancelled and a new switch started before the first had timed out where the d->exten would be used for both subchannels. This was bad leading to possible invalid extensions if some digits had been entered in the abandoned simple switch and the second one was completed before the first timed out, or the second would be cancelled because d->exten would be set to nothing on the time out of the first. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	9066db4329	Merged revisions 318720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318720 \| mnicholson \| 2011-05-12 18:35:51 -0500 (Thu, 12 May 2011) \| 4 lines Handle ipv6 addresses in the sent-by Via: field. This change fixes a regression in via header parsing and ipv6 handling. (closes issue #18951) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	1ad49f46ce	Merged revisions 318783 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318783 \| rmudgett \| 2011-05-12 20:47:05 -0500 (Thu, 12 May 2011) \| 14 lines PRI early media won't ring. And another way to pass early media. Don't indicate that there is inband information present, just assume that the B channel is connected. * Restore clearing the dialing flag Rx squelch unconditionally when a PROCEEDING message comes in. (closes issue #19268) Reported by: tbsky Patches: issue19268_v1.8.patch uploaded by rmudgett (license 664) Tested by: tbsky ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318784 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alec L Davis	892b7a2efd	Merged revisions 318671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318671 \| alecdavis \| 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) \| 30 lines Fix directed group pickup feature code *8 with pickupsounds enabled Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues. 1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked. 2). dialplan applications for directed_pickups shouldn't beep. 3). feature code for directed pickup should beep on success/failure if configured. Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite. Moved app_directed:pickup_do() to features:ast_do_pickup(). Functions below, all now use the new ast_do_pickup() app_directed_pickup.c: pickup_by_channel() pickup_by_exten() pickup_by_mark() pickup_by_part() features.c: ast_pickup_call() (closes issue #18654) Reported by: Docent Patches: ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett Review: https://reviewboard.asterisk.org/r/1185/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	c37c017781	Consolidate setsubstate_* into setsubstate and use a switch. Consolidate the functions and add some debugging info. Allows to be able to set a substate without explicitly knowing what the state is. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	bdbb3a506f	Add setsubstate_onhook. Add the setsubstate_onhook to complete the initial substate handling procedures. Added dumpsub(sub, forcehangup) which is the common way of calling setsubstate_onhook. Dumpsub attempts to activate another sub after setting the current one onhook. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	475c264bd2	Merged revisions 318550 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318550 \| twilson \| 2011-05-11 13:47:33 -0500 (Wed, 11 May 2011) \| 2 lines Comment out the REF_DEBUG that slipped in during debugging ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	da4016544e	Merged revisions 318549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318549 \| twilson \| 2011-05-11 13:39:48 -0500 (Wed, 11 May 2011) \| 27 lines Merged revisions 318548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r318548 \| twilson \| 2011-05-11 12:15:39 -0500 (Wed, 11 May 2011) \| 19 lines Clean up several chan_sip reference leaks Several situations in the code could lead to peers or sip_pvt references being leaked. This would cause RTP ports to never be destroyed (leading to exhaustion of all available RTP ports) and memory leaks. The original patch for this issue from rgagnon was the result of an obscene amount of testing and hard work, for which I am very grateful. I did some cleanup and added a few additional refcount fixes that I found. (closes issue #17255) Reported by: kvveltho Patches: tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff uploaded by rgagnon (license 1202) Tested by: rgagnon, twilson, wdoekes, loloski Review: https://reviewboard.asterisk.org/r/1101/ Review: https://reviewboard.asterisk.org/r/1207/ Review: https://reviewboard.asterisk.org/r/1210/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	d1e27b1026	Merged revisions 318499 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318499 \| rmudgett \| 2011-05-10 18:41:08 -0500 (Tue, 10 May 2011) \| 15 lines Unable to pickup DAHDI/PRI call because call state is reported as DIALING. The channel state is not updated to RINGING when an ALERTING message is received. Regression caused when sig_pri.c (also sig_ss7.c) extracted from chan_dahdi.c. * Added missing channel state update to RINGING when the AST_CONTROL_RINGING frame is queued for ISDN and SS7. (closes issue #19257) Reported by: alecdavis Patches: issue19257_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318500 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	0ccfc8609a	Merged revisions 318436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318436 \| russell \| 2011-05-10 10:13:16 -0500 (Tue, 10 May 2011) \| 2 lines chan_iax2: change LOG_NOTICE to LOG_DEBUG in iax2_read(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	07b3742ad2	Merged revisions 318337 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318337 \| twilson \| 2011-05-09 15:23:15 -0500 (Mon, 09 May 2011) \| 18 lines Merged revisions 318331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r318331 \| twilson \| 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011) \| 12 lines Don't offer video to directmedia callee unless caller offered it as well Make sure that when directmedia is enabled, that video is not offered to the callee even if it supports it. p->vrtp will not exist since the caller didn't offer video. (closes issue #19195) Reported by: one47 Patches: sip_cant_add_video_rtp uploaded by one47 (license 23) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	4c35291c6b	Merged revisions 318233 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318233 \| dvossel \| 2011-05-09 12:09:55 -0500 (Mon, 09 May 2011) \| 14 lines Merged revisions 318230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r318230 \| dvossel \| 2011-05-09 11:51:45 -0500 (Mon, 09 May 2011) \| 7 lines Fixes cases where sip_set_rtp_peer can return too early during media path reset. (closes issue #19225) Reported by: one47 Patches: sip_set_rtp_peer.patch uploaded by one47 (license 23) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	d7c94e1e04	Merged revisions 318231 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r318231 \| rmudgett \| 2011-05-09 11:57:18 -0500 (Mon, 09 May 2011) \| 41 lines Don't get early media for ISDN on outgoing calls. It looks to be a long-standing misinterpretation of the progress indicator ie values: 1 - Call is not end-to-end ISDN; further call progress information may be available in-band. 8 - In-band information or an appropriate pattern is now available. Only value 8 is handled by chan_dahdi/sig_pri. The 1 value is not handled as early media probably because the meaning of the second half of it's description was overlooked. * Test to see if either PRI_PROG_CALL_NOT_E2E_ISDN(1) or PRI_PROG_INBAND_AVAILABLE(8) bits are set to open the media path. (closes issue #18868) Reported by: isrl Patches: issue18868_19246_v1.8.patch uploaded by rmudgett (license 664) Tested by: satish_lx .......... No inband progress on PRI_EVENT_RINGING even if inband flag set. My ISDN-PRI provider sends an ALERTING with "Inband information or appropriate pattern now available", but Asterisk only generates and passes the RING to the SIP extension, not the inband message. Unfortunately, the inband message is not a ringback tone but a prompt that says the number is not in service. The SIP extension then hears two rings and the call is hungup which confuses the caller. * Post an AST_CONTROL_PROGRESS as well as opening the media path if inband audio is indicated with an ALERTING message. (closes issue #19246) Reported by: cristiandimache Patches: issue19246_v1.8.patch uploaded by rmudgett (license 664) Tested by: cristiandimache ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318232 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	7002adcb3e	Add setsubstate_callwait. If a call is made to a line that already has a call and the device is offhook (ie activeish call), the call is set to CALLWAIT rather than RINGIN. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318106 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	3736b02d97	Merged revisions 318055 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318055 \| russell \| 2011-05-07 18:24:18 -0500 (Sat, 07 May 2011) \| 7 lines chan_iax2: Don't overwrite port found with an SRV lookup. (closes issue #17291) Reported by: jcovert Patches: chan_iax2.c.1.8.3-srvlookup-corrected.patch uploaded by jcovert (license 551) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318056 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	8c0b1115cd	Only allow voicemail if substate is OFFHOOK or no channel active (UNSET). (closes issue #17901) Reported by: salecha git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318019 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	a9beb8323e	Rename sub->parent to sub->line. Improve readability of code, eg, (sub->parent == d->activeline) becomes (sub->line == d->activeline). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318018 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	bc61836c1b	Move the hookstate from line to device. Long time coming, finally moving the hookstate from line to device. This may fix some issues where a device has multiple lines. Previously we had to run through all lines on a device to see if it was actually onhook or not. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317996 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	33b7cc2ef6	Merged revisions 317867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317867 \| russell \| 2011-05-06 15:01:16 -0500 (Fri, 06 May 2011) \| 10 lines chan_sip: Destroy variables on a sip_pvt before copying vars from the sip_peer. Don't duplicate variables on the sip_pvt. Just reset the variable list each time. (closes issue #19202) Reported by: wdoekes Patches: issue19202_destroy_challenged_invite_chanvars.patch uploaded by wdoekes (license 717) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	ae8dbde4a8	Merged revisions 317865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317865 \| russell \| 2011-05-06 14:46:49 -0500 (Fri, 06 May 2011) \| 11 lines chan_sip: fix a deadlock in check_rtp_timeout. Don't block doing silly deadlock avoidance. Just return and try again later. The funciton gets called often enough that it's fine. Also, this change was already made in trunk. (closes issue #18791) Reported by: irroot Patches: chan_sip.rtptimeout.patch uploaded by irroot (license 52) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	307f148adb	Merged revisions 317670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317670 \| rmudgett \| 2011-05-06 11:19:18 -0500 (Fri, 06 May 2011) \| 22 lines Fix SIP connected line updates. This patch fixes a couple SIP connected line update problems: 1) The connected line needs to be updated when the initial INVITE is sent if there is a peer callerid configured. Previously, the connected line information did not get reported until the call was connected so SIP could not report connected line information in ringing or progress messages. 2) The connected line should not be updated on initial connect if there is no connected line information. Previously, all it did was wipe out any default preset CONNECTEDLINE information set by the dialplan with empty strings. (closes issue #18367) Reported by: GeorgeKonopacki Patches: issue18367_v1.8.patch uploaded by rmudgett (license 664) Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1199/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	0938974902	Merged revisions 317478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317478 \| russell \| 2011-05-05 17:53:45 -0500 (Thu, 05 May 2011) \| 12 lines Fix some consistency issues with jitterbuffer config. Store the defaults noted in the sample config files in the jitterbuffer config data structure. This makes the CLI commands that output these settings show the right thing. Also only show the settings that are relevant in the settings CLI commands, based on which jitterbuffer is selected and whether it's enabled. (closes issue #19083) Reported by: rgagnon Patches: issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	f0f5e237bf	Merged revisions 317474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317474 \| russell \| 2011-05-05 17:36:33 -0500 (Thu, 05 May 2011) \| 2 lines Fix more "set but unused" warnings. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317475 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	e98ac1f0f4	Move hold stuff to the setsubstate arrangement. skinny_hold moved to setsubstate_hold and skinny_unhold integrated into setsubstate_connected. Removed sub->onhold and replaced with SUBSTATE_HOLD. Also fixed inbound call answering by queueing an AST_CONTROL_ANSWER on answering a SUBSTATE_RINGIN sub (was a typo). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	932e34ee62	Merged revisions 317283 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317283 \| jrose \| 2011-05-05 14:09:13 -0500 (Thu, 05 May 2011) \| 10 lines Resolves a deadlock that occurs during sip_new This is based on an uncommitted patch by jpeeler for the issue. Instead of relocking and then unlocking the channel though, we keep the lock on the channel until we are finished doing what we need to the channel. (closes issue #18441) Reported by: Alric ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317334 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	4d612d126b	Merged revisions 317281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r317281 \| russell \| 2011-05-05 13:39:44 -0500 (Thu, 05 May 2011) \| 29 lines Merged revisions 317255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r317255 \| russell \| 2011-05-05 13:29:53 -0500 (Thu, 05 May 2011) \| 22 lines Merged revisions 317211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317211 \| russell \| 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011) \| 15 lines chan_sip: fix broken realtime peer count, fix memory leak This patch addresses two bugs in chan_sip: 1) The count of realtime peers and users was off. The increment checked the value of the caching option, while the decrement did not. 2) Add a missing regfree() for a regex. (closes issue #19108) Reported by: vrban Patches: missing_regfree.patch uploaded by vrban (license 756) sip_object_counter.patch uploaded by vrban (license 756) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	89da27b780	Merged revisions 317196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317196 \| mnicholson \| 2011-05-05 13:02:52 -0500 (Thu, 05 May 2011) \| 8 lines Set SO_KEEPALIVE on SIP TCP sockets so that they eventually go away when a peer abruptly disappears. This mostly occurs after a successful registration. (closes issue #17544) Reported by: marcelloceschia Patches: (modified) tcptls.patch uploaded by st (license 907) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	2ae06c3e6d	Add setsubstate_congestion and setsubstate_progress. Move handling of both state handling from skinny_indicate to it's own sub. Also, modified behaviour to not hangup the sub and let the dialplan have a chance in doing what it wants for congestion. Added various states to substate2str and added these states where applicable for other set_substate_ procs. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317026 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	468b8229a7	Add setsubstate_busy. Move handling of setting busy state from skinny_indicate to it's own sub. Also, modified behaviour to not hangup the sub and let the dialplan have a chance in doing what it wants (eg busy(10); hangup() in the dialplan now gives a busy indication for 10 secs and then hangs up. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316994 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	b9e763ecae	Add setsubstate_ringout (equivalent to AST_STATE ringing). Renamed previous setsubstate_ringout to setsubstate_dialing for a state when attempting to dial a number, substate ringout now for when core has indicated that the channel is actually ringing on the other end. Also added substate2str for debugging purposes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316885 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	1f96380da5	Reverts rev 316218 as it breaks parsing the [general] section of sip.conf. The functionality this patch attempts to achieve should already be possible using [general](+) in the config file. issue #17957 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	3bf4b09a6e	Merged revisions 316617 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316617 \| dvossel \| 2011-05-04 08:44:41 -0500 (Wed, 04 May 2011) \| 19 lines Merged revisions 316616 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r316616 \| dvossel \| 2011-05-04 08:40:41 -0500 (Wed, 04 May 2011) \| 12 lines Fixes session-timers=refuse not being enforced for caller During handle_request_invite, the session timer mode was retrieved from a cached variable. This patch forces a peer lookup of the session timer mode in the case of an incoming invite. (closes issue #18804) Reported by: wdoekes Patches: issue18804_session_timer_refuse_caller.patch uploaded by wdoekes (license 717) issue_18804_v2.diff uploaded by dvossel (license 671) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	99d0da2a2d	Add setsubstate_ringin. Added setsubstate_ringin. skinny_call now calls sss_ringin rather than inline. Fixed previous issue so that setsubstate_connected now use SUBSTATE_RINGIN to determine is an AST_CONTROL_ANSWER should be queued. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	bc814dc84a	Make skinny_answer use setsubsate_connected. Cosolidated the code so that skinny_answer now uses the setsubstate procedures rather than doing the handling inline. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Damien Wedhorn	c6f189cd71	Cleanup skinny callinfo. Cosolidated the working out of the callinfo to be sent into transmit_callinfo. Replaced ambiguous sub->outgoing with calldirection which can be SKINNY_INCOMING or SKINNY_OUTGOING (same value as the skinny protocol). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	ed56ae3ef7	If multiple [general] contexts occur from sip.conf (usually due to external includes), merge them. The original implementation of this did the merging of all contexts with the same name in the realtime layer, but that implementation severely breaks drivers which use the same context name (e.g. iax.conf, type={peer,user}). Therefore, the implementation needs to do the merging for particular entries only, based upon what contexts would allow that in the channel driver itself. This implementation is for chan_sip only, but others could be added in the future. (closes issue #17957) Reported by: marcelloceschia Patches: chan-sip_parsing-general_branch162.patch uploaded by marcelloceschia (license 1079) Tested by: tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	95561bd37a	Merged revisions 316336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316336 \| russell \| 2011-05-03 17:13:31 -0500 (Tue, 03 May 2011) \| 8 lines Use htons() instead of ntohs() in some places. (closes issue #19200) Reported by: wdoekes Patches: issue19200-trunk.patch uploaded by wdoekes (license 717) issue19200-1.8.x.patch uploaded by wdoekes (license 717) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	bb5e875b65	Merged revisions 316330 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r316330 \| dvossel \| 2011-05-03 16:37:59 -0500 (Tue, 03 May 2011) \| 24 lines Merged revisions 316329 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r316329 \| dvossel \| 2011-05-03 16:29:55 -0500 (Tue, 03 May 2011) \| 17 lines Merged revisions 316328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r316328 \| dvossel \| 2011-05-03 16:27:59 -0500 (Tue, 03 May 2011) \| 10 lines Fixes chan_local crashs in local_fixup() Thanks OEJ for tracking down the issue and submitting the patch. (closes issue #19053) Reported by: oej Tested by: oej Review: https://reviewboard.asterisk.org/r/1158/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	37aa52fd78	Merged revisions 316265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316265 \| russell \| 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) \| 5 lines Fix a bunch of compiler warnings generated by gcc 4.6.0. Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316293 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	810b9c8879	Merged revisions 316224 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316224 \| rmudgett \| 2011-05-03 14:18:30 -0500 (Tue, 03 May 2011) \| 16 lines The dahdi_hangup() call does not clean up the channel fully. After dahdi_hangup() has supposedly hungup an ISDN channel there is still traffic on the S0-bus because the channel was not cleaned up fully. Shuffled the hangup code to include some missing cleanup. Also fixed some code formatting in the area. I think the primary missing clean up code was the call to tone_zone_play_tone() to turn off any active tones on the channel. (closes issue #19188) Reported by: jg1234 Patches: issue19188_v1.8.patch uploaded by rmudgett (license 664) Tested by: jg1234 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	db72ee299a	Merged revisions 316217 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316217 \| dvossel \| 2011-05-03 13:59:06 -0500 (Tue, 03 May 2011) \| 9 lines Never put the Require: timer header in an Invite. This has already been discussed and should have been resolved earlier. View revsion 285565's log for more information about why it is important to not put timer in the Require header. (closes issue #18704) Reported by: mfrager ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316218 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	e87639fc26	Merged revisions 315894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315894 \| mnicholson \| 2011-04-27 14:14:27 -0500 (Wed, 27 Apr 2011) \| 28 lines Merged revisions 315893 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315893 \| mnicholson \| 2011-04-27 14:03:05 -0500 (Wed, 27 Apr 2011) \| 21 lines Merged revisions 315891 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315891 \| mnicholson \| 2011-04-27 13:57:56 -0500 (Wed, 27 Apr 2011) \| 14 lines Fix our compliance with RFC 3261 section 18.2.2. This change optimizes the free_via() function and removes some redundant null checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using the port specified in the Via header for routing responses (even when maddr is not set). Also the htons() function is now used when setting the port. Additional documentation comments have been added in various places to make the logic in the code clearer. (closes issue #18951) Reported by: jmls Patches: issue18951_set_proper_port_from_via.patch uploaded by wdoekes (license 717) (modified) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	181661c617	Merged revisions 315673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315673 \| twilson \| 2011-04-26 15:56:19 -0700 (Tue, 26 Apr 2011) \| 25 lines Merged revisions 315672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315672 \| twilson \| 2011-04-26 15:52:25 -0700 (Tue, 26 Apr 2011) \| 18 lines Merged revisions 315671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315671 \| twilson \| 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011) \| 11 lines Make sure unregistering a peer unlinks it from the peer container Instead of mostly copying the code from expire_register, just use the function that "does the right thing". (closes issue #16033) Reported by: kkm Patches: 016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888) Tested by: kkm, tilghman, twilson ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	bd354a0378	Make sure to create the caps structure for autocreated peers Because crashing is bad. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315674 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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6976 Commits (d34d46a16ee2ccafb4503eefcb91c282b466a12a)