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2011-02-02Replace link to old doc with new wiki page.Andrew Latham
Link to https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-01Merged revisions 305692 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r305692 | qwell | 2011-02-01 16:48:16 -0600 (Tue, 01 Feb 2011) | 7 lines Reverse sense of an error test when reading from astdb. (closes issue #18545) Reported by: jcovert Patches: chan_iax2.c.patch uploaded by jcovert (license 551) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305693 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-01Merged revisions 305343 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305343 | rmudgett | 2011-01-31 18:01:09 -0600 (Mon, 31 Jan 2011) | 21 lines Merged revisions 305342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305342 | rmudgett | 2011-01-31 17:50:10 -0600 (Mon, 31 Jan 2011) | 14 lines Merged revisions 305341 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@305344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-31Merged revisions 305254 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r305254 | qwell | 2011-01-31 17:07:00 -0600 (Mon, 31 Jan 2011) | 24 lines Merged revisions 305253 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305253 | qwell | 2011-01-31 16:59:34 -0600 (Mon, 31 Jan 2011) | 17 lines Merged revisions 305252 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@305255 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-27Merged from revision 304341Richard Mudgett
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r304341 | rmudgett | 2011-01-26 16:38:39 -0600 (Wed, 26 Jan 2011) | 7 lines Add connected line chan_dahdi.conf pricpndialplan option. * Added from_channel value to prilocaldialplan option. JIRA ABE-2731 JIRA SWP-2842 .......... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-26Merged revisions 304245 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304245 | mnicholson | 2011-01-26 14:43:27 -0600 (Wed, 26 Jan 2011) | 20 lines Merged revisions 304244 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304244 | mnicholson | 2011-01-26 14:42:16 -0600 (Wed, 26 Jan 2011) | 13 lines Merged revisions 304241 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@304246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-26Merged revisions 304150 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304150 | rmudgett | 2011-01-26 13:39:35 -0600 (Wed, 26 Jan 2011) | 16 lines Merged revisions 304149 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304149 | rmudgett | 2011-01-26 13:38:38 -0600 (Wed, 26 Jan 2011) | 9 lines Merged revisions 304148 from 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: https://origsvn.digium.com/svn/asterisk/trunk@304151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-25Merged revisions 303962 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303962 | twilson | 2011-01-25 16:09:01 -0600 (Tue, 25 Jan 2011) | 30 lines Merged revisions 303960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303960 | twilson | 2011-01-25 16:02:42 -0600 (Tue, 25 Jan 2011) | 23 lines Merged revisions 303906 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@303963 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-25Merged revisions 303860 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303860 | tilghman | 2011-01-25 12:55:27 -0600 (Tue, 25 Jan 2011) | 12 lines Merged revisions 303858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r303858 | tilghman | 2011-01-25 12:41:26 -0600 (Tue, 25 Jan 2011) | 5 lines Fix "sip show user <tab>", so that it actually shows results, instead of just completing the last entry. (closes issue #16675) Reported by: pj ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-25Merged revisions 303771 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@303772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-24According to section 19.1.2 of RFC 3261:Matthew Nicholson
For each component, the set of valid BNF expansions defines exactly which characters may appear unescaped. All other characters MUST be escaped. This patch modifies ast_uri_encode() to encode strings in line with this recommendation. This patch also adds an ast_escape_quoted() function which escapes '"' and '\' characters in quoted strings in accordance with section 25.1 of RFC 3261. The ast_uri_encode() function has also been modified to take an ast_flags struct describing the set of rules it should use when escaping characters to allow for it to escape SIP URIs in addition to HTTP URIs and other types of URIs or variations of those two URI types in the future. The ast_uri_decode() function has also been modified to accept an ast_flags struct describing the set of rules to use when decoding to enable decoding '+' as ' ' in legacy http URLs. The unit tests for these functions have also been updated. ABE-2705 Review: https://reviewboard.asterisk.org/r/1081/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-24Merged revisions 303467 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303467 | qwell | 2011-01-24 11:20:03 -0600 (Mon, 24 Jan 2011) | 22 lines Merged revisions 303285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303285 | qwell | 2011-01-21 15:48:09 -0600 (Fri, 21 Jan 2011) | 15 lines Merged revisions 303284 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@303468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-21Temporarily revert r303288Jason Parker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303376 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-21Merged revisions 303286 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303286 | qwell | 2011-01-21 15:50:11 -0600 (Fri, 21 Jan 2011) | 22 lines Merged revisions 303285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303285 | qwell | 2011-01-21 15:48:09 -0600 (Fri, 21 Jan 2011) | 15 lines Merged revisions 303284 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@303288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-19Merged revisions 302414 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302414 | seanbright | 2011-01-19 10:45:17 -0500 (Wed, 19 Jan 2011) | 7 lines Initialize an uninitialized variable. (closes issue #18640) Reported by: jcovert Patches: chan_sip.c.patch uploaded by jcovert (license 551) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-19Merged revisions 302412 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r302412 | seanbright | 2011-01-19 10:31:39 -0500 (Wed, 19 Jan 2011) | 10 lines Use appropriate type for requested format in chan_local. We were passing and storing the requested format as an int instead of format_t resulting in truncation. (closes issue #18238) Reported by: whizemen Patches: 0018238_speex16.patch uploaded by whizemen (license 1143) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-18Merged revisions 302314 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302314 | mnicholson | 2011-01-18 15:43:21 -0600 (Tue, 18 Jan 2011) | 18 lines Merged revisions 302313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302313 | mnicholson | 2011-01-18 15:40:03 -0600 (Tue, 18 Jan 2011) | 11 lines Merged revisions 302311 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@302315 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-17Merged revisions 293493 via svnmerge from Terry Wilson
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 #17403) Reported by: one47 Patches: sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-14Merged revisions 301946 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301946 | rmudgett | 2011-01-14 15:09:57 -0600 (Fri, 14 Jan 2011) | 13 lines Deadlock between dahdi_request() and pri_dchannel() processing an incomming call. The sig_pri_new_ast_channel() is called with the channel private lock held when pri_dchannel() calls it and no channel private lock held when dahdi_request() calls it. The use of pri_grab() in sig_pri_new_ast_channel() could leave the channel private lock held when it returns if the lock was not held before calling it. Make sig_pri_new_ast_channel() just lock the PRI span lock instead of using pri_grab(). It is safe to do this because dahdi_request() does not have the channel private lock and the deadlock potential with the PRI span lock is only between pri_dchannel() and other threads. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-14Merged revisions 301851 via svnmerge from Brett Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301851 | bbryant | 2011-01-14 15:11:55 -0500 (Fri, 14 Jan 2011) | 6 lines Changing previous revisions 301845/301847 to use ast_sockaddr_setnull() instead of setting the field manually to avoid uninitialized data. Review: https://reviewboard.asterisk.org/r/1076/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-14Merged revisions 301845 via svnmerge from Brett Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301845 | bbryant | 2011-01-14 14:35:23 -0500 (Fri, 14 Jan 2011) | 9 lines Fix for a consistent MulticastRTP channel driver crash due to use of unitilized data. (closes issue #18290) (closes issue #18602) Reported by: voipgate, wybecom Review: https://reviewboard.asterisk.org/r/1076/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-14Merged revisions 301790 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301790 | jpeeler | 2011-01-14 11:32:52 -0600 (Fri, 14 Jan 2011) | 42 lines Resolve deadlock involving REFER. Two fixes: 1) One must always have the private unlocked before calling pbx_builtin_setvar_helper to not invalidate locking order since it locks the channel. 2) Unlock the channel before calling pbx_find_extension, which starts and stops autoservice during the lookup. The problem scenario as illustrated by the reporter: Thread: do_monitor ----------------------- handle_request_do handle_incoming handle_request_refer ast_parking_ext_valid pbx_find_extension ast_autoservice_stop while (chan_list_state == as_chan_list_state) { usleep(1000); } Thread: autoservice_run ----------------------- autoservice_run chan = ast_waitfor_n ast_waitfor_nandfds ast_waitfor_nandfds_classic / simple / complex (depending on your system) ast_channel_lock(c[x]); handle_request_do and schedule_process_request_queue locks the owner if it exists. The autoservice thread is waiting for the channel lock, which wasn't ever released since the do_monitor thread was waiting for autoservice operations to complete. Solved by unlocking the channel but keeping a reference to guarantee safety. (closes issue #18403) Reported by: jthurman Patches: 20110103-blind_deadlock.diff uploaded by jthurman (license 614) issue18403.patch uploaded by jpeeler (license 325) Tested by: jthurman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-12Merged revisions 301683 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r301683 | twilson | 2011-01-12 15:19:48 -0600 (Wed, 12 Jan 2011) | 15 lines Merged revisions 301682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r301682 | twilson | 2011-01-12 15:05:02 -0600 (Wed, 12 Jan 2011) | 9 lines Don't reject all SUBSCRIBE auth requests 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: https://origsvn.digium.com/svn/asterisk/trunk@301684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-08Merged revisions 301134 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r301134 | rmudgett | 2011-01-07 19:11:31 -0600 (Fri, 07 Jan 2011) | 7 lines The DTMF attended transfer feature cannot callback a chan_dahdi BRI phone. The DAHDI ISDN channel name is not dialable. Make a channel name like DAHDI/i3/400-12 dialable when the sequence number is stripped off of the name. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-05Merged revisions 300714 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300714 | rmudgett | 2011-01-05 14:54:21 -0600 (Wed, 05 Jan 2011) | 21 lines Merged revision 300711 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r300711 | rmudgett | 2011-01-05 13:43:55 -0600 (Wed, 05 Jan 2011) | 14 lines A call retrieved from hold may wind up with no audio. If the retrieved call is natively bridged then the call may not have any audio path. The following warning message is given: "Failed to add <dfd> to conference <chan>/<chan>: Invalid argument". * Open the media on a B channel when pri_fixup_principle() moves the call from a no_b_channel channel to a real channel. * Added lock protection while pri_fixup_principle() moves a call from one private structure to another. * Made some pri_fixup_principle() messages more meaningful. .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300716 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-04Merged revisions 300521 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300521 | lmadsen | 2011-01-04 15:53:27 -0600 (Tue, 04 Jan 2011) | 17 lines Merged revisions 300520 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r300520 | lmadsen | 2011-01-04 15:52:41 -0600 (Tue, 04 Jan 2011) | 9 lines Fix backwards and broken XML documentation. (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: https://origsvn.digium.com/svn/asterisk/trunk@300522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-04Update MFC-R2 code to use new DTMF-R2 functionality in OpenR2Moises Silva
(closes issue #18576) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-04Merged revisions 300301 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300301 | twilson | 2011-01-04 11:54:41 -0600 (Tue, 04 Jan 2011) | 29 lines Merged revisions 300298 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r300298 | twilson | 2011-01-04 11:37:26 -0600 (Tue, 04 Jan 2011) | 22 lines Merged revisions 300216 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@300302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-04Optional HOLD/RETRIEVE signaling for PTMP TE when the bridge goes on and off ↵Richard Mudgett
hold. Added the moh_signaling option to specify what to do when the channel's bridged peer puts the ISDN channel on and off of hold. Implemented as a FSM to control libpri ISDN signaling when the bridged peer places the channel on and off of hold with the AST_CONTROL_HOLD and AST_CONTROL_UNHOLD control frames. JIRA SWP-2687 JIRA ABE-2691 Review: https://reviewboard.asterisk.org/r/1063/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-25Merged revisions 299626 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r299626 | tilghman | 2010-12-25 04:07:15 -0600 (Sat, 25 Dec 2010) | 19 lines Merged revisions 299625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r299625 | tilghman | 2010-12-25 04:05:00 -0600 (Sat, 25 Dec 2010) | 12 lines Merged revisions 299624 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@299627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-23Enqueue AST_CONTROL_PROGRESS after AST_CONTROL_RINGING when MFC-R2 calls are ↵Moises Silva
accepted (closes issue #18438) Reported by: mariner7 Tested by: moy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-22Merged revisions 299405 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r299405 | rmudgett | 2010-12-21 20:10:39 -0600 (Tue, 21 Dec 2010) | 17 lines Chan_dahdi sends an empty COLP on the bridged channel. Chan_dahdi always inserts a connected party IE when you call from one dahdi channel to another dahdi channel, even if no such information was received on the 2nd channel. This clears the display of many phones. * Removed leftover artifact from before the valid flag was added. * Updated all of the channel's caller id information with the new connected line information instead of just the string parts. (closes issue #18508) Reported by: wimpy Patches: issue18508_trunk.patch uploaded by rmudgett (license 664) Tested by: wimpy, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-21Merged revisions 299353 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r299353 | mnicholson | 2010-12-21 09:25:03 -0600 (Tue, 21 Dec 2010) | 30 lines Merged revisions 299242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r299242 | mnicholson | 2010-12-20 15:25:35 -0600 (Mon, 20 Dec 2010) | 23 lines Merged revisions 299194,299198,299220 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@299355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-20Merged revisions 299248 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r299248 | mmichelson | 2010-12-20 15:38:30 -0600 (Mon, 20 Dec 2010) | 20 lines Fix a couple of CCSS issues. * Make sure to allocate a cc_params structure when creating autopeers. * Use sip_uri_cmp when retrieving SIP CC agents and monitors in case parameters appear in the URI. (closes issue #18504) Reported by: kkm (closes issue #18338) Reported by: GeorgeKonopacki Patches: 18338.diff uploaded by mmichelson (license 60) Tested by: GeorgeKonopacki ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-20Fix chan_misdn build after sched API changes.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299134 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-20Some scheduler API cleanup and improvements.Russell Bryant
Previously, I had added the ast_sched_thread stuff that was a generic scheduler thread implementation. However, if you used it, it required using different functions for modifying scheduler contents. This patch reworks how this is done and just allows you to optionally start a thread on the original scheduler context structure that has always been there. This makes it trivial to switch to the generic scheduler thread implementation without having to touch any of the other code that adds or removes scheduler entries. In passing, I made some naming tweaks to add ast_ prefixes where they were not there before. Review: https://reviewboard.asterisk.org/r/1007/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-20Typos: recieved => receivedTzafrir Cohen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-17Merged revisions 298773 via svnmerge from Brad Watkins
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r298773 | marquis | 2010-12-17 12:26:31 -0500 (Fri, 17 Dec 2010) | 10 lines Fix parsing of mwi => lines in sip.conf Reworking parsing of mwi => lines to resolve a segfault. Also add a set of unit tests for the function that does the parsing. (closes issue #18350) Reported by: gbour Tested by: Marquis, gbour Review: https://reviewboard.asterisk.org/r/1053/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-16Merged revisions 298539 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r298539 | tilghman | 2010-12-16 03:28:17 -0600 (Thu, 16 Dec 2010) | 8 lines Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses. (closes issue #18464) Reported by: IgorG Patches: realtime_ipv6store.diff uploaded by IgorG (license 20) (plus a few additional lines by tilghman) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298545 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-13Post AMI hold events on PRI spans when the remote party HOLD/RETRIEVEs the call.Richard Mudgett
Part of JIRA SWP-2687/ABE-2691. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-13Merged revisions 298195 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298195 | rmudgett | 2010-12-13 11:11:43 -0600 (Mon, 13 Dec 2010) | 33 lines Merged revisions 298194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r298194 | rmudgett | 2010-12-13 11:04:41 -0600 (Mon, 13 Dec 2010) | 26 lines Merged revisions 298193 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@298201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-09Merged revisions 297965 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297965 | twilson | 2010-12-09 16:18:19 -0600 (Thu, 09 Dec 2010) | 28 lines Merged revisions 297960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297960 | twilson | 2010-12-09 16:10:31 -0600 (Thu, 09 Dec 2010) | 21 lines Merged revisions 297959 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@297972 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-09Merged revisions 297957 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r297957 | dvossel | 2010-12-09 15:32:20 -0600 (Thu, 09 Dec 2010) | 11 lines Fixes issue with outbound google voice calls not working. Thanks to az1234 and nevermind_quack for their input in helping debug the issue. (closes issue #18412) Reported by: nevermind_quack Patches: fix uploaded by dvossel (license 671) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-06Merged revisions 297607 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297607 | jpeeler | 2010-12-06 16:06:37 -0600 (Mon, 06 Dec 2010) | 25 lines Merged revisions 297605 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297605 | jpeeler | 2010-12-06 16:03:04 -0600 (Mon, 06 Dec 2010) | 18 lines Merged revisions 297603 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@297608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-03Merged revisions 297535 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297535 | seanbright | 2010-12-03 12:41:30 -0500 (Fri, 03 Dec 2010) | 9 lines Merged revisions 297534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r297534 | seanbright | 2010-12-03 12:40:52 -0500 (Fri, 03 Dec 2010) | 3 lines The CLI command should not contain <placeholder>s, these are for descriptions. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-01Merged revisions 297075 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297075 | jpeeler | 2010-12-01 11:53:13 -0600 (Wed, 01 Dec 2010) | 37 lines Merged revisions 297073 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297073 | jpeeler | 2010-12-01 11:52:46 -0600 (Wed, 01 Dec 2010) | 30 lines Merged revisions 297072 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@297076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-01Merged revisions 296951 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296951 | tilghman | 2010-11-30 19:46:32 -0600 (Tue, 30 Nov 2010) | 9 lines Merged revisions 296950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r296950 | tilghman | 2010-11-30 19:38:19 -0600 (Tue, 30 Nov 2010) | 2 lines Missed initializations caused startup errors on Mac OS X (and possibly others, too). ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-29Merged revisions 296673 via svnmerge from Paul Belanger
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296673 | pabelanger | 2010-11-29 18:05:45 -0500 (Mon, 29 Nov 2010) | 19 lines Merged revisions 296671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r296671 | pabelanger | 2010-11-29 17:54:14 -0500 (Mon, 29 Nov 2010) | 12 lines Merged revisions 296670 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@296674 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-29Merged revisions 296628 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r296628 | russell | 2010-11-29 15:26:44 -0600 (Mon, 29 Nov 2010) | 6 lines Complete some error handling in transmit_publish() in chan_sip.c. This error handling block caught my eye. It was missing a couple of things, but it should be safe now. Thanks to mmichelson for the quick peer review on IRC. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296630 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-29Merged revisions 296582 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r296582 | rmudgett | 2010-11-29 14:46:03 -0600 (Mon, 29 Nov 2010) | 24 lines Merged revision 296575 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r296575 | rmudgett | 2010-11-29 14:27:37 -0600 (Mon, 29 Nov 2010) | 13 lines Invalid mISDN PTMP redirecting signaling as TE towards NT. The mISDN PTMP redirection signaling (NOTIFY redirecting number and notification code, SETUP redirecting number) is also sent in PTMP/TE mode. It should only apply in PTMP/NT mode. The call setup proceeds but the network (Deutsche Telekom) reacts with ugly ISDN STATUS messages. Also don't send the redirecting number ie when PTP is also sending the DivertingLegInformation2 facility. The redirecting number ie is redundant and the network (Deutsche Telekom) complains about it. Patches: abe_2651_v4.patch uploaded by rmudgett (license 664) JIRA ABE-2651 JIRA SWP-2537 .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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