mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
19378 Commits (f70cd032b7cb3cceac5dba9e90f836670ffd1dcf)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Tilghman Lesher
|
f70cd032b7 |
Don't warn on floats and timestamps
(closes issue #17082) Reported by: coolmig git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284472 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
58b8667476 |
Merged revisions 284393 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010) | 7 lines Don't send a devstate change on poke_noanswer if the state did not change. (closes issue #17741) Reported by: schmidts Patches: chan_sip.c.patch uploaded by schmidts (license 1077) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284399 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Leif Madsen
|
928ef055fe |
Merged revisions 284316 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31 Aug 2010) | 7 lines Update say.conf.sample to match the rules in say.c (closes issue #17835) Reported by: RoadKill Patches: say.conf.sample.patch.rules uploaded by RoadKill (license 933) Tested by: RoadKill ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284317 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
7f48d17174 |
Fix 3 coding errors:
1) After we close FD, we should not be trying to write to it. 2) Call _exit(0), not exit(0), to avoid running shutdown routines in a child. 3) Use endian, not processor, detection to ensure bytes are written in the correct order. (closes issue #15706) Reported by: modelnine Patches: asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine (license 865) Tested by: gmartinez git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284280 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
David Vossel
|
9253b159fd |
Merged revisions 283960 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010) | 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests. (closes issue #17758) Reported by: ibc Patches: multiple_accept_headers_1.4.diff uploaded by dvossel (license 671) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284002 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Jason Parker
|
e8361949d4 |
Merged revisions 283880 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) | 8 lines Fix issue with decoding ^-escaped characters in realtime. (closes issue #17790) Reported by: denzs Patches: 17790-chunky.diff uploaded by qwell (license 4) Tested by: qwell, denzs ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@283881 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
015d89b40a |
Merged revisions 283690 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010) | 19 lines Fixed how Asterisk destroys a dialog on channel hangup before invite receives a response. If an ast_channel with a SIP tech pvt hangs up before the sip dialog gets a response to its outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is not rfc compliant and results in confusion at the other endpoint. sip_pretend_ack will ack and remove all the packets in the retransmit queue. This means that the INVITE will stop retransmitting, and that any response to that INVITE that comes after the pretend_ack occurs will be ignored. Instead of faking any sort of acknowledgement for an outgoing INVITE during an internal hangup, we should let the protocol stack process the INVITE transaction and terminate the dialog properly. This is achieved by setting the PENDING_BYE flag. When this flag is used, once the dialog proceeds to an escapable state the transaction will either be canceled with a SIP_CANCEL or completed followed immediately by a BYE. Attempting to do this any other way is incorrect. If the endpoint is not responding to the INVITE request, the INVITE must continue to be retransmitted until it times out which will result in the dialog being destroyed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@283691 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
8aa02d419f |
Add to and from tags to NOTIFY dialog-info xml body so pickup can occur.
When pedantic mode is used, the dialog-info xml generated during a ringing event must contain the to and from tag values. Otherwise if a pickup occurs using INVITE with replaces, Astrisk will not be able to locate the subscription. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@283594 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
4d36037db1 |
Asterisk will not advertise session timers are supported when 'session-timers=refuse' is used.
Asterisk now dynamically builds the "Supported" header depending on what is enabled/disabled in sip.conf. Session timers used to always be advertised as being supported even when they were disabled in the configuration. This caused problems with some end points. (issue #17005) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@283558 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
967a8a11d8 |
Merged revisions 283380 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010) | 11 lines This fix makes sure the ast_channel hangs up correctly when the dialog's PENDING_BYE flag is set. When the pending bye flag is used, it is possible that the dialog will terminate and leave the sip_pvt->owner channel up. This is because we never hangup the ast_channel after sending the SIP_BYE request. When we receive the response for the SIP_BYE we set need_destroy which we would expect to destroy the dialog on the next do_monitor loop, but this is not the case. The dialog will only be destroyed once the owner is hungup even with the need_destroy flag set. This patch sets the softhangup flag on the ast_channel when a SIP_BYE request is sent as a result of the pending bye flag. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@283381 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
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727cf54d50 |
CDR drivers depend upon res_odbc, not directly on the ODBC libraries
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@283318 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Richard Mudgett
|
db355da802 |
Merged revisions 283123 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r283123 | rmudgett | 2010-08-20 11:46:22 -0500 (Fri, 20 Aug 2010) | 9 lines Merged revision 278274 from https://origsvn.digium.com/svn/asterisk/trunk .......... r278274 | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1 line Reference correct struct member for unlikely event PRI_EVENT_CONFIG_ERR. .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@283124 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Richard Mudgett
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bdda8674c6 |
Merged revisions 283048 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010) | 22 lines Q931 - Sending PROGRESS after sending ALERTING is a protocol error The PRI layer in chan_dadhi will check if a PROGRESS message has already been sent, and not allow sending another (although that is technically allowed by the Q931 spec), however it does not protect against sending an ALERTING and then sending a PROGRESS message, which is a violation of the specification. Most switches don't seem to care too deeply about this, but some do, and will disconnect the call when receiving this invalid sequence. Protocol specification reference: T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview protocol control (network side) point-point (sheet 3 of 8)" (closes issue #17874) Reported by: nic_bellamy Patches: asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@283049 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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c74e4511de |
Merged revisions 282893 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) | 11 lines tos_sip option was not being set correctly When tos_sip is used, the tos of the sip socket is only set correctly if the socket binding changes on a reload. If the binding stays the same but the TOS changes, the new tos value would not take into effect. This patch fixes that. (closes issue #17712) Reported by: nickb ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282894 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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c63a2b3fa7 |
fixes sip peer memory leaks in the peer_by_ip table
(issue #17798) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282890 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Matthew Nicholson
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b3bede4744 |
Merged revisions 277944 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul 2010) | 16 lines Regression with T.38 negotiation Prior to 1.4.26.3 T.38 negotiation worked properly, in the case of the reporter. (issue #16852) Reported by: cfc (closes issue #16705) Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa, samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282859 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Terry Wilson
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0faad5f43b |
Merged revisions 282729 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18 Aug 2010) | 2 lines Add some documentation about codec negotiation to sip.conf ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282730 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
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5338d346b0 |
fixes crash with notifycid
(closes issue #17868) Reported by: francesco_r Patches: issue_17868.diff uploaded by dvossel (license 671) Tested by: francesco_r git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282668 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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943034663a |
Don't warn on callerid when completely text, instead of numeric with localdialplan prefixes.
(closes issue #16770) Reported by: jamicque Patches: 20100413__issue16770.diff.txt uploaded by tilghman (license 14) 20100811__issue16770.diff.txt uploaded by tilghman (license 14) Tested by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282607 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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93927c17ef |
fixes no default transport for temp peer creation in chan_sip
(closes issue #17829) Reported by: falves11 Patches: issue_17829.rev1.txt uploaded by russell (license 2) issue_17829.diff uploaded by dvossel (license 671) Tested by: falves11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282576 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Leif Madsen
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4c653e4cfa |
Add information about creating sounds files using
the sounds tools publically available so that others can create their own sounds prompts using the same tools we use to generate sounds releases. This allows people creating their own prompts to sound consistent with the prompts available from the open source project. SWP-595 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282469 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Terry Wilson
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79d5f46e1c |
Merged revisions 282430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010) | 16 lines Send a SRCCHANGE indication when we masquerade Masquerading a channel means that the src of the audio is potentially changing, so send a SRCCHANGE so that RTP-based media streams can get a new SSRC generated to reflect the change. Original patch by addix (along with lots of testing--thanks!). (closes issue #17007) Reported by: addix Patches: 1001-reset-SSRC-original-channel.diff uploaded by addix (license 1006) srcchange.diff uploaded by twilson (license 396) Tested by: addix, twilson Review: https://reviewboard.asterisk.org/r/862/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282467 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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a95d1e73f2 |
only do magic pickup when notifycid is enabled
A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber that a device is ringing. This option should only be enabled when the new 'notifycid' option is set... but this was not the case. Instead the call-id value was included for every RINGING Notify message, which caused a regression for people who used other methods for call pickup. (closes issue #17633) Reported by: urosh Patches: chan_sip.txt uploaded by urosh (license ) blf_cid_issue.diff uploaded by dvossel (license 671) Tested by: dvossel, urosh, okrief, alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282235 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jason Parker
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fd4f594ee1 |
Merged revisions 282129 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug 2010) | 1 line Register CLI commands before parsing config, in case there is a config error. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282130 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Jeff Peeler
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ade6beaf93 |
Merged revisions 281911 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010) | 20 lines Ensure SSRC is changed when media source is changed to resolve audio delay. This change causes the SSRC to change right before the channels are bridged, which is what used to happen. It seems that fixes were made to attempt limiting SSRC changes, targeted mainly at sending DTMF. DTMF is not affecting the SSRC with this change. There are two other control frames sent in ast_channel_bridge that probably should also be changed to AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change up to the discretion of resolving issue #17007. For reference - old review implementing new control frame SRCCHANGE: https://reviewboard.asterisk.org/r/540 (closes issue #17404) Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler (license 325) Tested by: sdolloff ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281912 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Leif Madsen
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94f6ed99b5 |
Merged revisions 281819 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11 Aug 2010) | 6 lines Add Danish support to say.conf.sample (closes issue #17836) Reported by: RoadKill Patches: say.conf.sample.patch.dk uploaded by RoadKill (license 933) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281873 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Leif Madsen
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31599c811c |
Merged revisions 281762 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11 Aug 2010) | 6 lines Allow say.conf to handle large numbers ending with multiple zeros. (closes issue #17833) Reported by: RoadKill Patches: say.conf.sample.patch.largenumbers uploaded by RoadKill (license 933) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281763 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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563b986aef |
Only set status TIMEOUT, if we have no digits.
(closes issue #15188) Reported by: jcovert Patches: app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license 551) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281722 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Russell Bryant
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ee39dde50e |
Don't move the time threshold for running scheduled events on every iteration.
Instead, only calculate the time threshold each time ast_sched_runq() is called. (closes issue #17742) Reported by: schmidts Patches: sched.c.patch uploaded by schmidts (license 1077) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281574 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Russell Bryant
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08bae3a8f3 |
Merged revisions 281566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010) | 8 lines Reset visible indication after answer. (closes issue #17641) Reported by: klaus3000 Patches: ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by klaus3000 (license 65) Tested by: schmidts ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281567 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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4b089171ef |
fixes SIP peers memory leak
We zeroed out the peer's addr before it was removed from the peers_by_ip container. This made it impossible to be removed from the container as the addr is the key used by the container to find the peer. (closes issue #17774) Reported by: kkm Patches: 017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888) 017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281430 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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e4e473991e |
Merged revisions 281390 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09 Aug 2010) | 13 lines Prevent loss of Caller ID information set on local channel after masquerade. Caller ID set on the channel before a masquerade occurs when using a local channel would cause the information to be lost. The problem was that the information was set on a channel destined to be hung up. The somewhat confusing fix is to detect if any Caller ID has been set on the channel and if so preswap the Caller ID data so that basically the masquerade puts the data back. (closes issue #17138) Reported by: kobaz Review: https://reviewboard.asterisk.org/r/847/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281391 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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1f876f9663 |
Blocked revisions 281185 via svnmerge
........ r281185 | dvossel | 2010-08-06 16:34:38 -0500 (Fri, 06 Aug 2010) | 20 lines chan_sip: fixes provisional keepalive scheduled item crash There is a scheduler item in chan_sip that keeps sending the last provisional message in response to an INVITE Request for a period of time until a final response to that INVITE is sent. Because of the way this scheduler item works, it requires a reference to a sip_pvt pointer to work properly. The problem with this is that it is currently possible (but rare) for the sip_pvt to get destroyed and that scheduler item to still exist. When this occurs, the scheduler event fires and attempts to access a freed sip_pvt which causes a crash. (closes issue #17497) Reported by: anonymouz666 Patches: keepalive_diff_1.4_v2.diff uploaded by dvossel (license 671) Review: https://reviewboard.asterisk.org/r/849/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281186 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Russell Bryant
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db7878ed47 |
Cleanup default option value handling for cdr.conf [general].
The default values would differ depending on whether or not cdr.conf exists. That is no longer the case. Apply a default value to the unanswered option. Define all default values as named constants. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@281051 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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5da0ea7b8e |
Merged revisions 280982 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010) | 8 lines Change context lock back to a mutex, because functionality depends upon the lock being recursive. (closes issue #17643) Reported by: zerohalo Patches: 20100726__issue17643.diff.txt uploaded by tilghman (license 14) Tested by: zerohalo ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280983 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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81a24c617b |
Merged revisions 280811 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280811 | tilghman | 2010-08-03 15:49:10 -0500 (Tue, 03 Aug 2010) | 9 lines Prevent DAHDI channels from overriding the callerid, once it's been set by the user. (closes issue #16661) Reported by: jstapleton Patches: 20100414__issue16661.diff.txt uploaded by tilghman (license 14) 20100415__issue16661__1.6.2.diff.txt uploaded by tilghman (license 14) Tested by: jstapleton ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280812 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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5e354be7a6 |
Document -B and -W flags and regenerate manpage from sgml
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280739 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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81fe0bf8b5 |
Allow the pipe, but also allow the comma
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280671 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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a66be5fedc |
Change SIP NOTIFY requests to expect a response so authentication will work.
This changes the request to be sent with the transmit type XMIT_RELIABLE so that sip_ack doesn't return false and cause the 401 to be ignored in cases where authentication is required. (closes issue #14255) Reported by: zktech git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280669 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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e672faf469 |
Off-by-one error
(closes issue #17590) Reported by: atis Patches: 20100729__issue17590.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280556 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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608b11136f |
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/branches/1.6.2@280551 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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69f2056e38 |
Merged revisions 280448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010) | 12 lines fixes issue with translator frame not getting freed A translator frame even if it local storage so the translation path can be freed. This issue prevented g729 licenses from being freed up. (closes issue #17630) Reported by: manvirr Patches: encoder_fix.diff uploaded by dvossel (license 671) Tested by: manvirr, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280449 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Jean Galarneau
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a7e01edae4 |
Merged revisions 280341 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) | 2 lines Fix a dsp structure leak occuring when a local channel is put into a meetme conference, then masquaraded away. ABE-2422 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280345 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Matthew Nicholson
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ef546bb8fc |
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280306 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jason Parker
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2c6a11e2ef |
Work around some silly behavior on BSD.
A non-zero exit from a subshell should make the build fail. (closes issue #17621) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280231 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
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a92ec2969f |
Add missing enum value "unknown" to the SS7 called_nai and calling_nai config options.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280229 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jason Parker
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4259cb54b9 |
Add sha1sum-sh in case there is no util on the system.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280227 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jason Parker
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958afef96b |
Remove unnecessary subshells. Attempt to make checksumming work.
Also improves readability. (issue #17621) Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280193 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Sean Bright
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950cbd80a5 |
Plug a reference leak in app_queue when adding members dynamically.
(closes issue #17738) Reported by: bobwienholt Patches: issue17738.patch uploaded by bobwienholt (license 950) Tested by: bobwienholt, seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280160 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Leif Madsen
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e9a4fdcaff |
Merged revisions 280088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28 Jul 2010) | 1 line Update help text to be less confusing. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@280089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |