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2011-02-11Merged revisions 307623 via svnmerge from rmudgett1-4/+82
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r307623 | rmudgett | 2011-02-10 18:29:17 -0600 (Thu, 10 Feb 2011) | 13 lines Reentrancy problem if outgoing call gets different B channel than requested. The chan_dahdi pri_fixup_principle() routine needs to protect the Asterisk channel with the channel lock when it changes the technology private pointer to a new private structure. * Added lock protection while pri_fixup_principle() moves a call from one private structure to another. * Made some pri_fixup_principle() messages more meaningful. Partial backport from v1.8 -r300714. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@307624 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-08Merged revisions 306972 via svnmerge from twilson1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08 Feb 2011) | 2 lines Fix comparison for REFER Replaces tags with pedantic=yes ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@306973 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-07Merged revisions 306617 via svnmerge from twilson1-1/+9
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011) | 10 lines Don't allow a REFER w/replaces to replace its own dialog Asterisk currently accepts a REFER with a Refer-To with an embedded Replaces header that matches the dialog of the REFER. This would be a situation like A calls B, A calls C, A transfers B to A, which is just silly. This patch makes the transfer fail instead of making Asterisk freak out and forget to hang other channels up. Review: https://reviewboard.asterisk.org/r/1093/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@306618 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-03Merged revisions 306119 via svnmerge from twilson1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011) | 9 lines Set hangup cause in local_hangup When a call involves a local channel (like SIP -> Local -> SIP), the hangup cause was not being set. This resulted in SIP channels sometimes getting a 503 error instead of a 486 when the far side sent a busy. In Asterisk 1.8+ this also can cause issues with CCSS that involve a local channel. This patch sets the hangupcause for one side of the local channel to the other in local_hangup for outbound calls. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@306126 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-03Merged revisions 305888 via svnmerge fromrmudgett1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines Minor AST_FRAME_TEXT related issues. * Include the null terminator in the buffer length. When the frame is queued it is copied. If the null terminator is not part of the frame buffer length, the receiver could see garbage appended onto it. * Add channel lock protection with ast_sendtext(). * Fixed AMI SendText action ast_sendtext() return value check. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@305889 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-02Replace link to old doc with new wiki page.lathama1-3/+3
Link to https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@305752 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-31Merged revisions 305341 via svnmerge from rmudgett1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305341 | rmudgett | 2011-01-31 17:45:58 -0600 (Mon, 31 Jan 2011) | 7 lines Obtain the pri lock for PRI queue counters. Need to obtain the pri lock when calling pri_dump_info_str() to avoid a reentrancy problem when calculating the Q.921 Q count statistic. JIRA AST-484 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@305342 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-31Merged revisions 305252 via svnmerge from qwell1-0/+6
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) | 10 lines Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers already had code to prevent this. The attempt that app_dial was making to prevent it was not correct, so I fixed that. (closes issue #18371) Reported by: gbour Patches: 18371.patch uploaded by gbour (license 1162) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@305253 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-26Merged revisions 304241 via svnmerge from mnicholson1-0/+193
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan 2011) | 6 lines This patch modifies chan_sip to route responses to the address the request came from. It also modifies chan_sip to respect the maddr parameter in the Via header. ABE-2664 Review: https://reviewboard.asterisk.org/r/1059/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@304244 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-26Merged revisions 304148 fromrmudgett1-3/+4
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r304148 | rmudgett | 2011-01-26 13:23:46 -0600 (Wed, 26 Jan 2011) | 2 lines Update documentation for DAHDISendCallreroutingFacility() application. .......... git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@304149 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-25Merged revisions 303906 via svnmerge from twilson1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011) | 16 lines Guard against retransmitting BYEs indefinitely In the case of an attended transfer (A calls B, A atxfers to C) where A becomes unreachable before replying to Asterisk's BYE, Asterisk can sometimes retransmit the BYE indefinitely. This is because __sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0], SIP_ALREADYGONE and will then transmit a BYE. When this BYE times out, it will not ever be marked as ALREADYGONE, so when __sip_autodestruct is called again, we end up starting the cycle over. This patch adds a call to sip_alreadygone(pkt->owner) in retrans_pkt in the case of a BYE that has timed out. This should prevent Asterisk from trying to transmit new BYE messages in the future. Review: https://reviewboard.asterisk.org/r/1077/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@303960 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-25Fix "sip show user <tab>", so that it actually shows results, instead of ↵tilghman1-0/+3
just completing the last entry. (closes issue #16675) Reported by: pj git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@303858 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-25Merged revisions 303765 via svnmerge from rmudgett1-150/+301
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines Sending out unnecessary PROCEEDING messages breaks overlap dialing. Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@303769 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-21Merged revisions 303284 via svnmerge from qwell1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | 8 lines Reset configuration before parsing users.conf. Some values configured in chan_dahdi.conf were able to leak in to users.conf configuration. This was surprising users, and potentially setting non-sane "defaults". ASTNOW-125 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@303285 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-18Merged revisions 302311 via svnmerge from mnicholson1-5/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302311 | mnicholson | 2011-01-18 15:35:03 -0600 (Tue, 18 Jan 2011) | 4 lines URI encode the user part of the contact header. ABE-2705 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@302313 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-17Merged revisions 293493 via svnmerge from twilson1-7/+17
https://origsvn.digium.com/svn/asterisk/branches/1.8 [^] ........ r293493 | twilson | 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010) | 14 lines Only offer codecs both sides support for directmedia When using directmedia, Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (closes issue 0017403) Reported by: one47 Patches: sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ [^] ........ Backporting a bugfix that should have been included. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@302049 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-12Don't reject all SUBSCRIBE auth requeststwilson1-3/+3
When merging another SUBSCRIBE fix from 1.4, some braces were put in the wrong place. This patch fixes that. (closes issue #18597) Reported by: thsgmbh git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@301682 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-04Fix backwards and broken XML documentation.lmadsen3-5/+5
(closes issue #18547) Reported by: jcovert Patches: xmldoc.c.patch uploaded by jcovert (license 551) chan_iax2.c.doc.patch uploaded by jcovert (license 551) chan_sip.c.patch uploaded by jcovert (license 551) chan_agent.c.patch uploaded by jcovert (license 551) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@300520 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-04Merged revisions 300216 via svnmerge from twilson1-14/+20
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011) | 15 lines Don't authenticate SUBSCRIBE re-transmissions This only skips authentication on retransmissions that are already authenticated. A similar method is already used for INVITES. This is the kind of thing we end up having to do when we don't have a transaction layer... (closes issue #18075) Reported by: mdu113 Patches: diff.txt uploaded by twilson (license 396) Tested by: twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@300298 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-25Merged revisions 299624 via svnmerge from tilghman1-6/+6
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r299624 | tilghman | 2010-12-25 04:04:06 -0600 (Sat, 25 Dec 2010) | 5 lines Move check for extension existence below variable inheritance, due to the possible use of an eswitch. (closes issue #16228) Reported by: jlaguilar ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@299625 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-23do not use progress which is for PRI and SS7, add mfcr2_progress membermoy1-2/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@299533 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-23Enqueue AST_CONTROL_PROGRESS after AST_CONTROL_RINGING when MFC-R2 calls are ↵moy1-0/+13
accepted (closes issue #18438) Reported by: mariner7 Tested by: moy git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@299530 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-20Merged revisions 299194,299198,299220 via svnmerge from mnicholson1-25/+26
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec 2010) | 6 lines Respond as soon as possible with a 202 Accepted to refer requests. This change also plugs a few memory leaks that can occur when parking sip calls. ABE-2656 ........ r299198 | mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2 lines Remove changes to via processing that were not supposed to go into the last commit. ........ r299220 | mnicholson | 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines Use ast_free() instead of free() ABE-2656 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@299242 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-20Typos: recieved => receivedtzafrir1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@299003 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-13Merged revisions 298193 via svnmerge from rmudgett1-1/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298193 | rmudgett | 2010-12-13 10:56:07 -0600 (Mon, 13 Dec 2010) | 19 lines Outgoing PRI/BRI calls cannot do DTMF triggered transfers. Outgoing PRI/BRI calls cannot do DTMF triggered transfers if a PROCEEDING message is not received. The debug output shows that the DTMF begin event is seen, but the DTMF end event is missing. When the DTMF begin happens, the call is muted so we now have one way audio (until a DTMF end event is somehow seen). * Made set the proceeding flag when the PRI_EVENT_ANSWER event is received. * Made absorb the DTMF begin and DTMF end events if we are overlap dialing and have not seen a PROCEEDING message. * Added a debug message when absorbing a DTMF event. JIRA SWP-2690 JIRA ABE-2697 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@298194 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-09Merged revisions 297959 via svnmerge from twilson1-0/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010) | 14 lines Ignore spurious REGISTER requests If a REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. This info is used to generate messages for other responses in the transaction. This patch ignores REGISTER requests that match non-REGISTER transactions. (closes issue #18051) Reported by: eeman Tested by: twilson Review: https://reviewboard.asterisk.org/r/1050/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297960 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-06Merged revisions 297603 via svnmerge from jpeeler1-7/+30
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010) | 12 lines Improve handling of REGISTER requests with multiple contact headers. The changes here attempt to more strictly follow RFC 3261 section 10.3. Basically the following will now cause a 400 Bad Response to be returned, if: - multiple Contact headers are present with one set to expire all bindings ("*") - wildcard parameter is specified for Contact without Expires header or Expires header is not set to zero. ABE-2442 ABE-2443 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297605 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-03The CLI command should not contain <placeholder>s, these are for descriptions.seanbright1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297534 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-02Merged revisions 297185 via svnmerge from oej1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297185 | oej | 2010-12-02 09:37:17 +0100 (Tor, 02 Dec 2010) | 5 lines If we get a NOTIFY from a non-existing subscription we should answer with 481, not bad event. If we answer 481 the subscription that we don't want will be cancelled. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297186 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Merged revisions 297072 via svnmerge from jpeeler1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010) | 23 lines Fix not stopping MOH when transfered local channel queue member is answered. The problem here is only present when local channels are used with the MOH passthru option as well as no optimization (/nm). I will describe the slightly bizarre scenario that was used to test, where phones B and C are queue members: Phone A dials into a queue with two members using local channels and the above options. Phone B answers. Phone A blind transfers phone B into the same queue. Phone A hangs up. Phone C answers, but phone B didn't stop playing MOH. In this scenario, the unhold frame that should have gotten to phone B never arrived due to the masquerade from the blind transfer. This is usually fine since app_queue manages the starting and stopping of MOH. However, with the passthrough option enabled when app_queue attempts to stop MOH it tries to do so on the local channel rather than the real channel. The easiest solution was to just make sure to send an unhold frame during the transfer since it wouldn't make sense to have MOH playing after a transfer anyway. This only modifies SIP transfers, but the other transfers did not seem to be a problem. If DTMF based transfers were a problem it might be okay to add ast_moh_stop to finishup, but I didn't want to have to add that unless required. ABE-2624 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@297073 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Missed initializations caused startup errors on Mac OS X (and possibly ↵tilghman1-0/+10
others, too). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296950 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-29Merged revisions 296670 via svnmerge from pabelanger1-3/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296670 | pabelanger | 2010-11-29 17:49:39 -0500 (Mon, 29 Nov 2010) | 5 lines Make sure nothing else is needed before destroying the scheduler. (closes issue #18398) Reported by: pabelanger ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296671 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-24Merged revisions 296165 via svnmerge from rmudgett1-113/+197
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010) | 43 lines Oneway audio to SIP phone from FXS port after FXS port gets a CallWaiting pip. The FXS connected phone has to have CW/CID support to fail, as it will send back a DTMF 'A' or 'D' when it's ready to receive CallerID. A normal phone with no CID never fails. Also the SIP phone does not hear MOH when the CW call is answered. The DTMF end frame is suppressed when the phone acknowledges the CW signal for CID. The problem is the DTMF begin frame needs to be suppressed as well. The DTMF begin frame is causing SIP to start sending the DTMF RTP frames. Since the DTMF end frame is suppressed, SIP will not stop sending those DTMF RTP packets. * Suppress the DTMF begin and end frames when the channel driver is looking for DTMF digits. * Fixed a couple issues caused by not cleaning up the CID spill if you answer the CW call while it is sending the CID spill. * Fixed not sending CW/CID spill to the phone when the call is natively bridged. (Fixed by not using native bridge if CW/CID is possible.) * Suppress received audio when sending CW/CID spills. The other parties involved do not need to hear the CW/CID spills and may be confused if the CW call is for them. (closes issue #18129) Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett NOTE: * v1.4 does not have the main problem fixed by suppressing the DTMF start frames. The other three items fixed are relevant. * If you really must restore native bridging between analog ports, you need to disable CW/CID either by configuring chan_dahdi.conf callwaitingcallerid=no or dialing *70 before dialing the number to temporarily disable CW. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@296166 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-19Merged revisions 295628 via svnmerge from twilson1-3/+19
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295628 | twilson | 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010) | 8 lines Discard responses with more than one Via This is not a perfect solution as headers that are joined via commas are not detected. This is a parsing issue that to fix "correctly" would necessitate a new SIP parser. Review: https://reviewboard.asterisk.org/r/1019/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@295672 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-12Merged revisions 294821 via svnmerge from rmudgett1-2/+9
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010) | 11 lines Asterisk is getting a "No D-channels available!" warning message every 4 seconds. Asterisk is just whining too much with this message: "No D-channels available! Using Primary channel XXX as D-channel anyway!". Filtered the message so it only comes out once if there is no D channel available without an intervening D channel available period. (closes issue #17270) Reported by: jmls ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@294822 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-11Merged revisions 294688 via svnmerge from jpeeler1-1/+15
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010) | 18 lines Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. This scenario has the potential to progress to the point of saturating a link just from options packets. The fix was to ensure that the poke scheduler checks to see if a peer is in the peer list before continuing to poke. The reason a peer must be in the peer list to be able to properly manage an options dialog is because otherwise the call pointer is lost when the peer is regenerated from the database, which is how existing qualify dialogs are detected. (closes issue #16382) (closes issue #17779) Reported by: lftsy Patches: bug16382-3.patch uploaded by jpeeler (license 325) Tested by: zerohalo ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@294733 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-08Go off hold when we get an empty reinvite telling us to.mnicholson1-31/+40
(closes issue 0014448) Reported by: frawd (closes issue #17878) Reported by: frawd git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@294242 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-03Merged revisions 293805 via svnmerge from rmudgett1-11/+14
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010) | 20 lines Party A in an analog 3-way call would continue to hear ringback after party C answers. All parties are analog FXS ports. 1) A calls B. 2) A flash hooks to call C. 3) A flash hooks to bring C into 3-way call before C answers. (A and B hear ringback) 4) C answers 5) A continues to hear ringback during the 3-way call. (All parties can hear each other.) * Fixed use of wrong variable in dahdi_bridge() that stopped ringback on the wrong subchannel. * Made several debug messages have more information. A similar issue happens if B and C are SIP channels. B continues to hear ringback. For some reason this only affects v1.8 and trunk. * Don't start ringback on the real and 3-way subchannels when creating the 3-way conference. Removing this code is benign on v1.6.2 and earlier. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@293806 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-02Merged revisions 293722 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010) | 8 lines Add enabled/disabled information for rtautoclear sip show settings output. When setting to zero/"no", the numeric default was shown making it not obvious the disabled setting was respected. (closes issue #18123) Reported by: zerohalo ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@293723 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-02Merged revisions 293639 via svnmerge from rmudgett1-2/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02 Nov 2010) | 6 lines Make warning message have more useful information in it. Change "Unable to get index, and nullok is not asserted" to "Unable to get index for '<channel-name>' on channel <number> (<function>(), line <number>)". ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@293647 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-30Merged revisions 293416 via svnmerge from rmudgett1-11/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29 Oct 2010) | 1 line Remove some more code that serves no purpose. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@293417 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-30Merged revisions 293339 via svnmerge from rmudgett1-11/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29 Oct 2010) | 1 line Remove some code that serves no purpose. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@293340 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-25Merged revisions 292866 via svnmerge from dvossel1-150/+180
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010) | 27 lines This patch turns chan_local pvts into astobj2 objects. chan_local does some dangerous things involving deadlock avoidance. tech_pvt functions like hangup and queue_frame are provided with a locked channel upon entry. Those functions are completely safe as long as you don't attempt to give up that channel lock, but that is impossible to guarantee due to the required deadlock avoidance necessary to lock both the tech_pvt and both channels involved. In the past, we have tried to account for this by doing things like setting a "glare" flag that indicates what function should destroy the pvt. This was used in local_hangup and local_queue_frame to decided who should destroy the pvt if they collided in separate threads. I have removed the need to do this by converting all chan_local tech_pvts to astobj2. This means we can ref a pvt before deadlock avoidance and not have to worry about that pvt possibly getting destroyed under us. It also cleans up where we destroy the tech_pvt. The only unlink from the tech_pvt container occurs in local_hangup now, which is where it should occur. Since there still may be thread collisions on some functions like local_hangup after deadlock avoidance, I have added some checks to detect those collisions and exit appropriately. I think this patch is going to solve quite a bit of weirdness we have had with local channels in the past. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@292867 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-22Update the LDIF file for LDAP.lmadsen1-0/+6
The LDIF file asterisk.ldif was quite a bit out of date from the asterisk.ldap-schema file, so I've now updated that to be in sync. The asterisk.ldif file being out of sync was a problem on my systems where I was doing an ldapadd to import the schema into the LDAP database, and the existing file would cause problems and ERROR messages when registering. Additional documention has been added based on feedback in the issue I'm closing. (closes issue #13861) Reported by: scramatte Patches: ldap-update.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec, rgenthner git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@292786 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Merged revisions 291643 via svnmerge from rmudgett1-76/+222
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13 Oct 2010) | 20 lines Deadlock between dahdi_exception() and dahdi_indicate(). There is a deadlock between dahdi_exception() and dahdi_indicate() for analog ports. The call-waiting and three-way-calling feature can experience deadlock if these features are trying to do something and an event from the bridged channel happens at the same time. Deadlock avoidance code added to obtain necessary channel locks before attemting an operation with call-waiting and three-way-calling. (closes issue #16847) Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett Review: https://reviewboard.asterisk.org/r/971/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@291655 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Merged revisions 291392 via svnmerge from russell1-0/+16
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010) | 6 lines Lock pvt so pvt->owner can't disappear when queueing up a frame. This fixes a crash due to a hangup race condition. ABE-2601 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@291393 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Make exit from handle_request_do() consistent.rmudgett1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@291111 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Merged revisions 291109 via svnmerge from rmudgett1-3/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 Oct 2010) | 1 line Add missing unlock to an exception condition in reload_config(). ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@291110 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01Merged revisions 289797 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines Change RFC2833 DTMF event duration on end to report actual elapsed time. The scenario here is with a non P2P early media session. The reported time length of DTMF presses are coming up short when sending to the remote side. Currently the event duration is a running total that is incremented when sending continuation packets. These continuation packets are only triggered upon incoming media from the remote side, which means that the running total probably is not going to end up matching the actual length of time Asterisk received DTMF. This patch changes the end event duration to be lengthened if it is detected that the end event is going to come up short. Review: https://reviewboard.asterisk.org/r/957/ ABE-2476 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@289798 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01Merged revisions 289699 via svnmerge from jpeeler1-3/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010) | 14 lines Ensure user portion of SIP URI matches dialplan when using encoded characters. This commit takes a simliar approach to 288112 and checks the dialplan to determine the proper action for an incoming contact header as to whether or not it should be decoded or not. sip_new was blindly always decoding the extension, which also caused the outgoing contact header to be incorrect as well as failing to match the encoded extension in the dialplan. (closes issue #17892) Reported by: wdoekes Patches: bug17892-1.patch uploaded by jpeeler (license 325) Tested by: wdoekes ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@289700 f38db490-d61c-443f-a65b-d21fe96a405b
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