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6660 Commits (eb78ce7845562ba6ee7e09ce8b22a9e65d993b9d)
Author | SHA1 | Message | Date |
---|---|---|---|
|
f8d64f8614 |
Fix up handling and indications during transfer.
Cleaned up handling of onhook indications and added indications if more than one sub on device. Also fixes issue in 12324 so that the phone can call itself without locking up. (closes issue #17692) Reported by: jmhunter Patches: chan_skinny-transfer-v4.txt uploaded by DEA (license 3) skinnytransfver.v8.diff uploaded by wedhorn (license 30) Tested by: jmhunter, salecha, wedhorn Review: NA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281257 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
394fa75a0a |
Move call answering stuff into new setsubstate_connected.
Move call answering stuff into new setsubstate_connected. Also add sub->substate var and set it to SUBSTATE_CONNECTED in setsubstate_connected. (closes issue #17772) Reported by: wedhorn Patches: cleanup.stateconnected2.diff uploaded by wedhorn (license 30) Tested by: wedhorn, salecha Review: NA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281227 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
dcb865f68a |
Start rtp on answer before the answer is queued
(closes issue #17770) Reported by: salecha Patches: skinny.answercrash.diff uploaded by wedhorn (license 30) Tested by: salecha Review: NA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281226 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
ca661f4702 |
Merged revisions 280879 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280879 | tilghman | 2010-08-04 09:04:07 -0500 (Wed, 04 Aug 2010) | 14 lines Check cur value before attempting a deref. (closes issue #17775) Reported by: svinson Patches: 20100804__issue17775.diff.txt uploaded by tilghman (license 14) Tested by: svinson (closes issue #17743) Reported by: tgruenberg Patches: 20100804__issue17775.diff.txt uploaded by tilghman (license 14) Tested by: tgruenberg ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280880 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
dfb810efc3 |
Merged revisions 280778 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280778 | simon.perreault | 2010-08-03 15:54:03 -0400 (Tue, 03 Aug 2010) | 9 lines Fixed IPv6-related SIP parsing bugs. (closes issue #17663) Reported by: oej Patches: diff uploaded by sperreault (license 252) diff2 uploaded by sperreault (license 252) get_domain.diff uploaded by sperreault (license 252) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280780 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
dc0f39a760 |
Reverted r280706 and r280707. Will commit in branch 1.8 and merge to trunk properly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280745 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
b641ad14a4 |
Fixed IPv6-related SIP parsing bugs.
(closes issue #17663) Reported by: oej Patches: diff uploaded by sperreault (license 252) diff2 uploaded by sperreault (license 252) get_domain.diff uploaded by sperreault (license 252) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280707 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
f507546498 |
if totag is not present for an ACK request, do not send an error response
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280627 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
a352ff7f60 |
Cleanup transmit_ for handle_register and keepalives
Moved inline packet sending to transmit_ subs. Removed handle_keep_alive and handle_register_message to inline in handle_message. Also moved transmit_response(d) to transmit_response_bysessions(s) and created a wrapper transmit_response(d) that calls transmit_response_bysession(d->session). (closes issue #16980) Reported by: wedhorn Patches: skinny-clean06b.diff uploaded by wedhorn (license 30) Tested by: wedhorn, DEA Review: NA git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280589 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
0443248aa7 |
PeerStatus now includes Address and Port
(closes issue #17730) Reported by: jkroon Patches: iax2-peerstate-address.patch uploaded by jkroon (license 714) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280555 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
5e2999324b |
Merged revisions 280552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280552 | dvossel | 2010-07-29 15:43:47 -0500 (Thu, 29 Jul 2010) | 17 lines Merged revisions 280551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280551 | dvossel | 2010-07-29 15:42:29 -0500 (Thu, 29 Jul 2010) | 11 lines fixes wrong SRV query for TLS connection (closes issue #17612) Reported by: marcelloceschia Patches: chan-sip_srvQuery.patch uploaded by marcelloceschia (license 1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907) chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia (license 1079) Tested by: marcelloceschia, st, pabelanger ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280553 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d5e83070b7 |
Merged revisions 280519 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280519 | seanbright | 2010-07-29 15:47:16 -0400 (Thu, 29 Jul 2010) | 7 lines Fix compilation error in chan_dahdi (strdupa -> ast_strdupa). (closes issue #17751) Reported by: b11d Patches: strdupa_oops.diff uploaded by malcolmd (license 924) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280520 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
91cfe9a93e |
respond with 481 when request requiring totag has no totag to match against
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280518 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b20da321c3 |
Merged revisions 280343 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280343 | mnicholson | 2010-07-29 10:57:57 -0500 (Thu, 29 Jul 2010) | 4 lines Use PRIx64 instead of PRId64 in format string. related to r280302 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280344 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a29c220884 |
Merged revisions 280302 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280302 | pabelanger | 2010-07-28 19:45:34 -0500 (Wed, 28 Jul 2010) | 2 lines Use PRId64 with format_t ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280342 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3329e69b12 |
Make chan_usbradio.c build on 64bit platforms.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280340 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3def1196b4 |
Merged revisions 280307 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280307 | mnicholson | 2010-07-29 08:56:35 -0500 (Thu, 29 Jul 2010) | 11 lines Merged revisions 280306 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul 2010) | 2 lines Implement support for ast_channel_queryoption on local channels. Currently only AST_OPTION_T38_STATE is supported. ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ ........ Additionally, pass AST_CONTROL_T38_PARAMETERS control frames through generic bridges. This change appears to have been unintentionally left out of rev 203699. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280308 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7a987a853e |
Merged revisions 280269 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280269 | jpeeler | 2010-07-28 15:49:26 -0500 (Wed, 28 Jul 2010) | 2 lines Give test category missing leading slash ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280270 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b75433a9e6 |
Merged revisions 280235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280235 | rmudgett | 2010-07-28 15:12:16 -0500 (Wed, 28 Jul 2010) | 9 lines Merged revisions 280229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280229 | rmudgett | 2010-07-28 14:57:49 -0500 (Wed, 28 Jul 2010) | 2 lines Add missing enum value "unknown" to the SS7 called_nai and calling_nai config options. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280247 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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8e4efe2164 |
Formatting changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280093 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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642c9534bb |
Merged revisions 280023 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280023 | pabelanger | 2010-07-27 21:37:10 -0400 (Tue, 27 Jul 2010) | 5 lines Resolve compiler warning about formatting (closes issue #17732) Reported by: pabelanger ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280024 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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538d044aca |
Merged revisions 279916 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279916 | russell | 2010-07-27 14:50:56 -0500 (Tue, 27 Jul 2010) | 12 lines Fix inband DTMF detection on outgoing ISDN calls. This is a regression from the sig_pri split from chan_dahdi. When a call is first initiated, the inband DTMF detector is not enabled if it's an outgoing ISDN call. However, it needs to be turned on once the media path starts up. This handling was put back in the open_media() callback of chan_dahdi. In sig_pri, open_media() calls were added to a few places where it was needed, including handling of PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and PRI_EVENT_PROCEEDING. Thanks to rmudgett for helping me with the patch! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279917 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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eecac589ec |
Merged revisions 279887 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279887 | mmichelson | 2010-07-27 13:54:07 -0500 (Tue, 27 Jul 2010) | 16 lines Fix parsing error in sip_sipredirect(). The code was written in a way that did a bad job of parsing the port out of a URI. Specifically, it would do badly when dealing with an IPv6 address. In this particular scenario, there was no value from parsing the port out, so I just removed that logic. And while I was messing around in the function, I changed some variable names to be more descriptive. (closes issue #17661) Reported by: oej Patches: 17661.diff uploaded by mmichelson (license 60) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279888 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d61a4088f5 |
Merged revisions 279817 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279817 | dvossel | 2010-07-27 11:09:15 -0500 (Tue, 27 Jul 2010) | 2 lines fix sip transaction match with authentication, fix confusing log message when using getaddrinfo ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279818 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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8bd241f238 |
Merged revisions 279636,279815 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279636 | russell | 2010-07-26 16:53:30 -0500 (Mon, 26 Jul 2010) | 2 lines Ignore a control subclass of -1 in ast_waitfordigit_full(). ........ r279815 | russell | 2010-07-27 11:06:58 -0500 (Tue, 27 Jul 2010) | 4 lines Support "channels" in addition to "channel" in chan_dahdi.conf. Review: https://reviewboard.asterisk.org/r/804 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279816 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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805082efd4 |
Merged revisions 279785 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r279785 | mmichelson | 2010-07-27 10:15:22 -0500 (Tue, 27 Jul 2010) | 20 lines Merged revisions 279784 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r279784 | mmichelson | 2010-07-27 10:13:24 -0500 (Tue, 27 Jul 2010) | 14 lines Fix bad behavior of dynamic_exclude_static option in sip.conf. We were attempting to create a contactdeny rule based on the peer's IP address before the peer's IP address had been set. By moving the processing further down in the function, we can ensure stuff works as we expect for it to. (closes issue #17717) Reported by: mmichelson Patches: 17717.patch uploaded by mmichelson (license 60) Tested by: DennisD ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279786 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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61c782df58 |
Merged revisions 279755 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279755 | pabelanger | 2010-07-26 22:57:33 -0400 (Mon, 26 Jul 2010) | 10 lines If dringXcontext is null, fallback to default context value. (closes issue #17693) Reported by: iasgoscouk Patches: issue17693.patch uploaded by pabelanger (license 224) Tested by: iasgoscouk Review: https://reviewboard.asterisk.org/r/803/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279756 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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4a98994542 |
Merged revisions 279568 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279568 | dvossel | 2010-07-26 14:59:03 -0500 (Mon, 26 Jul 2010) | 21 lines transaction matching using top most Via header This patch modifies the way chan_sip.c does transaction to dialog matching. Asterisk now stores information in the top most Via header of the initial incoming request and compares that against other Requests that have the same call-id. This results in Asterisk being able to detect a forked call in which it has received multiple legs of the fork. I completely stripped out the previous matching code and made the comparisons a little more explicit and easier to understand. My comments in the code should offer all the details involving this patch. This patch also fixes a bug with the usage of the OBJ-MULTIPLE flag to find multiple dialogs with the same call-id. Since the callback function was returning (CMP_MATCH | CMP_STOP) only the first item found was being returned. I fixed this by making a new callback function for finding multiple dialogs that only returns (CMP_MATCH) on a match allowing for multiple items to be returned. Review: https://reviewboard.asterisk.org/r/776/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279569 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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dd9428666d |
Merged revisions 279504 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279504 | mmichelson | 2010-07-26 11:04:09 -0500 (Mon, 26 Jul 2010) | 14 lines Allow for systems without locale support to be usable. A recent change to SIP URI comparison code added a locale-specific string comparison to the mix, and certain systems do not support such functions. This fix allows for those systems to still use Asterisk 1.8 (closes issue #17697) Reported by: pprindeville Patches: asterisk-trunk-bugid17697.patch uploaded by pprindeville (license 347) Tested by: mmichelson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279533 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d7ca69ceea |
Make "dahdi show channels" show an outgoing called number.
The "dahdi show channels" extension column previously only showed the called number of an incoming call. It now shows the called number for an outgoing call as well. (closes issue #17653) Reported by: amazinzay Patches: issue17653_trunk.txt uploaded by rmudgett (license 664) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279156 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d1ad460b3d |
SIP URI comparison fixes.
This initially was created to work around the issue of using a string comparison instead of a binary comparison for IP addresses. It evolved a bit when test cases were created and it was discovered that comparison of URI parameters was not working exactly as it should. sip_uri_cmp() and its helpers have been moved to reqresp_parser.c and a new test has been added. (closes issue #17662) Reported by: oej Review: https://reviewboard.asterisk.org/r/792 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278980 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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09206a7db8 |
... just kidding. Enable SIP by default. :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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98f0f3933f |
Disable SIP support by default for Asterisk 1.8.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278944 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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301505c4c4 |
Rename sig_pri_pri to sig_pri_span. More descriptive of concept.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278942 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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57a92a6a7c |
Allow IPv6 addresses for UDPTL streams.
Review: https://reviewboard.asterisk.org/r/795 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278908 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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8b3c00a824 |
missed FXS kewl start polarityswitch when finally on hook.
(issue #17318) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278841 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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85bfe38f2f |
Support FXS module Polarity Reversal on remote party Answer and Hangup
FXS lines normally connect to a telephone. However, when FXS lines are routed to an external PBX or Key System to act as "external" or "CO" lines, it is extremely difficult, if not impossible for the external PBX to know when the call has been disconnected without receiving a polarity reversal on the line. Now using answeronpolarityswitch and hanguponpolarityswitch keywords that previously were used only for FXO ports, now applies like functionality for an FXS port, but from the connected equipment's point of view. (closes issue #17318) Reported by: armeniki Patches: fxs_linepolarity.diff5.txt uploaded by alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/797/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278809 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ab0b255455 |
DNID not cleared when channel hang up (Affects PRI and SS7)
The "dahdi show channels" CLI command still reports the DNID of the previous call even if the call is already hang up. The "dahdi show channels" command of older releases clear the DNID once the channel is hang up. Regression from the sig_analog/sig_pri extraction from chan_dahdi. (closes issue #17623) Reported by: klaus3000 Patches: issue17623.patch uploaded by rmudgett (license 664) Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278777 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3819ba7ac7 |
update sip subscription debug message to a warning message
If the Expire header of a SUBSCRIBE is less that our expiremin, a log warning will be displayed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278619 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d6e1c724e5 |
Remove built-in AES code and use optional_api instead
Review: https://reviewboard.asterisk.org/r/793/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278538 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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318798e932 |
send "423 Interval too small" Response to Subscribe with Expires less that min allowed
[RFC3265]3.1.6.1.... The notifier MAY also check that the duration in the "Expires" header is not too small. If and only if the expiration interval is greater than zero AND smaller than one hour AND less than a notifier- configured minimum, the notifier MAY return a "423 Interval too small" error which contains a "Min-Expires" header field. The "Min- Expires" header field is described in SIP [1]. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278536 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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16b4813599 |
Fix invalid test for rxisoffhook in FXO channels
This fixes some cases of no outgoing calls on FXO before an incoming call. Remove an unnecessary testing of an "off-hook" bit from DAHDI for FXO (KS/GS) channels.In some cases the bit would not be initialized properly before the first inbound call and thus prevent an outgoing call. If those tests are actually required by anybody, they should define DAHDI_CHECK_HOOKSTATE in channels/sig_analog.c . (closes issue #14577) Reported by: jkroon Patches: asterisk_chan_dahdi_hookstate_fix_trunk_new.diff uploaded by frawd (license 610) Tested by: frawd Review: https://reviewboard.asterisk.org/r/699/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278501 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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43b486453b |
Properly set the port number for UDPTL media sessions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278461 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a8c843199c |
Change order so that it more closely matches the related SIP command.
(closes issue #17648) Reported by: GMLudo Review: https://reviewboard.asterisk.org/r/789/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278393 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d1b0bf0f2d |
include stat.h for everybody, needed for device2chan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278361 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7066a7f233 |
Reference correct struct member for unlikely event PRI_EVENT_CONFIG_ERR.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278274 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c26791d5f8 |
fixes sip CANCEL race condition
If Asterisk sends a 4xx error and the other side sends a CANCEl before receiving the 4xx and responding with the ACK, Asterisk will process the CANCEL and send a 487 Request Terminated as a new final response to the INVITE. Since we are issuing a new final response to the INVITE, the old one must be pretend_acked else it will keep retransmitting. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278234 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b4e18d5660 |
Add load priority order, such that preload becomes unnecessary in most cases
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d27bb2d811 |
Only call ast_channel_cc_params_init() if allocating a channel succeeds.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278051 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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cb5892bb67 |
Fix port setting of external address in SIP.
There are two changes here: 1. Since the externip setting can now have a port attached to it, calling it "externip" is misleading. The option is now documented and parsed as "externaddr." This also extends to the "matchexterniplocally" setting. It is now documented and parsed as "matchexternaddrlocally." The old names for the options may still be used, but they are no longer used in the sip.conf.sample file. 2. If no port is set for the externaddr, and UDP is the transport to be used, then we will set the port of the externaddr to that of the udpbindaddr. This was how things worked prior to the IPv6 merge, so this is a regression fix. (closes issue #17665) Reported by: mmichelson Patches: 17665.diff#2 uploaded by pprindeville (license 347) Tested by: pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277873 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |