aboutsummaryrefslogtreecommitdiffstats
path: root/channels
AgeCommit message (Collapse)AuthorFilesLines
2011-05-26Merged revisions 320947 via svnmerge from russell1-2/+0
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320947 | russell | 2011-05-26 10:57:13 -0500 (Thu, 26 May 2011) | 2 lines Remove some variables that were set but unused. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321003 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-25Merged revisions 320883 via svnmerge from rmudgett1-11/+24
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320883 | rmudgett | 2011-05-25 17:25:18 -0500 (Wed, 25 May 2011) | 17 lines Native SIP CCSS sends bad CC cancel SUBSCRIBE message. The SUBSCRIBE message used to cancel a CC request has incorrect To/From SIP headers. They are reversed and the dialog tags are the same when they should not be. If pedantic mode was disabled, then the cancel would have succeeded despite the incorrect message. * The SIP_OUTGOING flag was not set correctly for the dialog and I had to move some CC subscribe handling code as a result. * Initialized the dialog subscribed type to CALL_COMPLETION earlier. If a CC request SUBSCRIBE message comes in and the CC instance is not found, the 404 response was duplicated. JIRA AST-568 JIRA SWP-3493 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@320884 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-23Merged revisions 320504 via svnmerge from jrose1-0/+7
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320504 | jrose | 2011-05-23 09:33:20 -0500 (Mon, 23 May 2011) | 10 lines Fixes segfault occuring in chan_sip.c at __set_address_from_contact Checks to see if domain contains anything before sending it off to ast_sockaddr_resolve which is where the segfault was occuring due to null str. (closes issue #18857) Reported by: sybasesql Review: https://reviewboard.asterisk.org/r/1225/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@320505 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-20Merged revisions 320180 via svnmerge from mnicholson1-20/+44
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320180 | mnicholson | 2011-05-20 13:48:46 -0500 (Fri, 20 May 2011) | 16 lines This commit modifies the way polling is done on TLS sockets. Because of the buffering the TLS layer does, polling is unreliable. If poll is called while there is data waiting to be read in the TLS layer but not at the network layer, the messaging processing engine will not proceed until something else writes data to the socket, which may not occur. This change modifies the logic around TLS sockets to only poll after a failed read on a non-blocking socket. This way we know that there is no data waiting to be read from the buffering layer. (closes issue #19182) Reported by: st Patches: ssl-poll-fix3.diff uploaded by mnicholson (license 96) Tested by: mnicholson ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@320181 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-20Merged revisions 319938 via svnmerge from jrose2-7/+25
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319938 | jrose | 2011-05-20 08:28:24 -0500 (Fri, 20 May 2011) | 12 lines Adds legacy_useroption_parsing to address interoperability concerns. With the new option engaged, Asterisk should interpret user fields with useroptions contained within the userfield of the uri by stripping them out of the original message whenever a semicolon is encountered in the userfield string. (closes issue #18344) Reported by: danimal Tested by: jrose Review: https://reviewboard.asterisk.org/r/1223/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319939 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-18Merged revisions 319654 via svnmerge from twilson1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319654 | twilson | 2011-05-18 16:15:58 -0700 (Wed, 18 May 2011) | 22 lines Merged revisions 319653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r319653 | twilson | 2011-05-18 16:11:57 -0700 (Wed, 18 May 2011) | 15 lines Merged revisions 319652 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011) | 8 lines Make sure everyone gets an unhold when a transfer succeeds Some phones, like the Snom phones, send a hold to the transfer target after before sending the REFER. We need to make sure that we unhold the parties that are being connected after the masquerade. If Local channels with the /nm option are used when dialing the parties, hold music would still be playing on the transfer target, even after being connected with the transferee. ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319661 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-18Merged revisions 319552 via svnmerge from twilson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319552 | twilson | 2011-05-18 13:22:36 -0700 (Wed, 18 May 2011) | 11 lines Unbreak the storing of registrations for restart The fix for issue 18882 broke retrieving non-realtime peers from the ast_db on restart/reload. This patch tries to unbreak things while leaving the intent of the original fix intact. (closes issue #19318) Reported by: remiq Patches: diff.txt uploaded by twilson (license 396) Tested by: lmadsen, remiq ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319564 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-17Merged revisions 319469 via svnmerge from rmudgett1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r319469 | rmudgett | 2011-05-17 16:57:56 -0500 (Tue, 17 May 2011) | 22 lines Merged revision 319468 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r319468 | rmudgett | 2011-05-17 16:49:31 -0500 (Tue, 17 May 2011) | 15 lines The mISDN HDLC mode is prevented on dialed channels. The use of mISDN HDLC mode is prevented if the mISDN dial technology option 'h1' is used when config option astdtmf=yes. There is a bug in channels/misdn/isdn_lib.c which prevents the use of HDLC mode. Instead of setting the channel to HDLC mode it is set to transparent(no dsp, no hdlc), although hdlc is not "no hdlc". I.e the logging message is correct, but the if condition is not. Make check the nodsp and hdlc flags. JIRA ABE-2787 JIRA SWP-3437 .......... ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319471 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-17Remove extraneous line variables.wedhorn1-13/+0
The vars were either explicitly or implicitly not used. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319470 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-17Option needed for Q931_IE_TIME_DATE to be optional in CONNECT message.rmudgett3-0/+48
The NEC SV8300 rejects the Q931_IE_TIME_DATE for Q.SIG. Add option to specify if and how much of the current time is put in Q931_IE_TIME_DATE. * Send date/time ie never. * Send date/time ie date only. * Send date/time ie date and hour. * Send date/time ie date, hour, and minute. * Send date/time ie date, hour, minute, and second. * Send date/time ie default: Libpri will send date and hhmm only when in NT PTMP mode to support ISDN phones. (closes issue #19221) Reported by: kenner JIRA SWP-3396 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319427 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-16Fix up skinny hints.wedhorn1-22/+28
Probably haven't been working for a couple of years. May still need some more love, but they are now working, both as a hint device and monitoring a hint. Changes centre around the long ago change to remove the requirement for a device name in a skinny line, and changes to the transmit_* functions. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319316 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-16Merged revisions 319204 via svnmerge from twilson1-0/+5
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: http://svn.digium.com/svn/asterisk/trunk@319212 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-16Merged revisions 319145 via svnmerge from dvossel1-4/+5
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: http://svn.digium.com/svn/asterisk/trunk@319146 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-16Merged revisions 319142 via svnmerge from mnicholson1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@319143 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-16 When a error in T.38 negotiation happens or its rejected on a channel theirroot2-8/+13
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: http://svn.digium.com/svn/asterisk/trunk@319087 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-15Add activatesub and dialandactivate sub.wedhorn1-51/+76
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: http://svn.digium.com/svn/asterisk/trunk@319024 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-13Merged revisions 318917 via svnmerge from bbryant1-7/+6
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: http://svn.digium.com/svn/asterisk/trunk@318918 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-13Move exten used for dialing from device to subchannel.wedhorn1-22/+26
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: http://svn.digium.com/svn/asterisk/trunk@318833 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-13Merged revisions 318720 via svnmerge from mnicholson1-2/+11
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: http://svn.digium.com/svn/asterisk/trunk@318785 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-13Merged revisions 318783 via svnmerge from rmudgett1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@318784 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-12Merged revisions 318671 via svnmerge from alecdavis1-13/+56
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: http://svn.digium.com/svn/asterisk/trunk@318672 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-12Consolidate setsubstate_* into setsubstate and use a switch.wedhorn1-292/+254
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: http://svn.digium.com/svn/asterisk/trunk@318635 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-12Add setsubstate_onhook.wedhorn1-182/+128
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: http://svn.digium.com/svn/asterisk/trunk@318600 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-11Merged revisions 318550 via svnmerge from twilson1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@318552 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-11Merged revisions 318549 via svnmerge from twilson1-61/+125
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: http://svn.digium.com/svn/asterisk/trunk@318551 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-10Merged revisions 318499 via svnmerge from rmudgett2-0/+15
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: http://svn.digium.com/svn/asterisk/trunk@318500 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-10Merged revisions 318436 via svnmerge from russell1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@318437 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-10Merged revisions 318337 via svnmerge from twilson1-2/+6
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: http://svn.digium.com/svn/asterisk/trunk@318400 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-09Merged revisions 318233 via svnmerge from dvossel1-2/+4
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: http://svn.digium.com/svn/asterisk/trunk@318234 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-09Merged revisions 318231 via svnmerge from rmudgett1-5/+13
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: http://svn.digium.com/svn/asterisk/trunk@318232 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-09Add setsubstate_callwait.wedhorn1-2/+31
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: http://svn.digium.com/svn/asterisk/trunk@318106 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-07Merged revisions 318055 via svnmerge from russell1-1/+3
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: http://svn.digium.com/svn/asterisk/trunk@318056 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-06Only allow voicemail if substate is OFFHOOK or no channel active (UNSET).wedhorn1-5/+11
(closes issue #17901) Reported by: salecha git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318019 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-06Rename sub->parent to sub->line.wedhorn1-45/+45
Improve readability of code, eg, (sub->parent == d->activeline) becomes (sub->line == d->activeline). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318018 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-06Move the hookstate from line to device.wedhorn1-36/+33
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: http://svn.digium.com/svn/asterisk/trunk@317996 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-06Merged revisions 317867 via svnmerge from russell1-1/+2
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: http://svn.digium.com/svn/asterisk/trunk@317868 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-06Merged revisions 317865 via svnmerge from russell1-4/+5
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: http://svn.digium.com/svn/asterisk/trunk@317866 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-06Merged revisions 317670 via svnmerge from rmudgett1-69/+105
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: http://svn.digium.com/svn/asterisk/trunk@317671 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-05Merged revisions 317478 via svnmerge from russell11-74/+86
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: http://svn.digium.com/svn/asterisk/trunk@317479 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-05Merged revisions 317474 via svnmerge from russell6-32/+11
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: http://svn.digium.com/svn/asterisk/trunk@317475 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-05Move hold stuff to the setsubstate arrangement.wedhorn1-65/+31
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: http://svn.digium.com/svn/asterisk/trunk@317382 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-05Merged revisions 317283 via svnmerge from jrose1-1/+2
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: http://svn.digium.com/svn/asterisk/trunk@317334 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-05Merged revisions 317281 via svnmerge from russell1-2/+9
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: http://svn.digium.com/svn/asterisk/trunk@317282 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-05Merged revisions 317196 via svnmerge from mnicholson1-0/+6
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: http://svn.digium.com/svn/asterisk/trunk@317198 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-05Add setsubstate_congestion and setsubstate_progress.wedhorn1-28/+55
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: http://svn.digium.com/svn/asterisk/trunk@317026 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-05Add setsubstate_busy.wedhorn1-16/+24
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: http://svn.digium.com/svn/asterisk/trunk@316994 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-04Add setsubstate_ringout (equivalent to AST_STATE ringing).wedhorn1-24/+61
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: http://svn.digium.com/svn/asterisk/trunk@316885 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-04Reverts rev 316218 as it breaks parsing the [general] section of sip.conf.dvossel1-490/+483
The functionality this patch attempts to achieve should already be possible using [general](+) in the config file. issue #17957 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316798 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-04Merged revisions 316617 via svnmerge from dvossel1-12/+15
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: http://svn.digium.com/svn/asterisk/trunk@316618 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-04Add setsubstate_ringin.wedhorn1-21/+25
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: http://svn.digium.com/svn/asterisk/trunk@316584 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-09 09:04:12 +0000