mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
14199 Commits (6f73ea7750869cf9eb2637ecd63b577ab65bb803)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
6f73ea7750 |
Use ast_strdup() instead of ast_strdupa() while processing in ast_hint_state_changed().
(related to issue #17928) Reported by: mdu113 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@287307 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
510f98c4c7 |
Add LSB headers for Debian init script, since Debian will complain if it isn't there.
Headers were taken from trunk. (closes issue #17958) Reported by: javyer git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@287197 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
18e10794d9 |
Don't limit hint processing in ast_hint_state_changed() to AST_MAX_EXTENSION length strings.
(closes issue #17928) Reported by: mdu113 Patches: 20100831__issue17928.diff.txt uploaded by tilghman (license 14) Tested by: mdu113 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@287118 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
5f9b0604b8 |
Don't stop printing cdr variables if we encounter one with a blank name or value.
(closes issue #17900) Reported by: under Patches: core-show-channel-cdr-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@287114 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
a91c96ba5d |
whitespace fix
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@286956 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
fa2b4c0196 |
Ensure mailbox is not filled to capacity before doing message forwarding.
Specifically, before prompting to record a prepended message the capacity is checked first. If the mailbox is full the extension will be reprompted. ABE-2517 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@286941 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
b34044bd3a |
Don't clear the username from a realtime database when a registration expires.
Non-realtime chan_sip does not clear the username from memory when a registration expiries so realtime probably shouldn't either. (closes issue #17551) Reported by: ricardolandim Patches: reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.8-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-trunk-fix1.diff uploaded by mnicholson (license 96) Tested by: ricardolandim, mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@286756 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
319b5134a9 |
Only drop duplicate answer frames if the channel is bridged.
Back in r3710 ast_read() was modified to drop answer frames on channels that were in the UP state. This modification prevented bridges that were up before the answer from being broken and reestablished by an ANSWER control frame. That change also prevents pickup of channels called from the ast_dial framework from working properly. The ast_dial framework expects to see an ANSWER frame after dialing and the pickup code queues one but ast_read() drops it. This new change only drops ANSWER frames when the channel is bridged, allowing the answer queued by the pickup code to properly pass through ast_read() on to the ast_dial framework. ABE-2473 (related to issue #2342) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@286679 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
b6bd325021 |
Add stuff to svn:ignore for tests/ directory.
(closes issue #17983) Reported by: oej git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@286381 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
7727748565 |
Handle error response when we can't make file compatible
Review: https://reviewboard.asterisk.org/r/911/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@286267 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
1b3cdeab25 |
Return -1 if chan_local doesn't support an option
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@286222 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
59349472bb |
Load iax.conf before registering any functions/applications/actions.
Review: https://reviewboard.asterisk.org/r/914/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@286114 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
111e145ef6 |
An outgoing call may not get hung up if a pre-connect incoming ISDN call is disconnected.
If the ISDN link a pre-connect incoming call is using fails or is reset, the outgoing leg may not hang up or be delayed in hanging up. (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER, PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the incoming call leg hangs up before connecting for any reason. It makes no sense to send a BUSY or CONGESTION control frame to the outgoing call leg under these circumstances. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@286113 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
5f8e639d61 |
Fixes sip extension state update DEADLOCK
PROBLEM: In chan_sip, and all the other channel drivers, it is common for us to hold the tech_pvt lock while we ask the Asterisk core about an extension and context. Every time we do this the locking order becomes, (1. tech_pvt lock ---> 2. global context lock). In chan_sip when a dialog subscribes to a hint, that locking order is reversed in the extensionstate callback which will occur outside of the channel_driver's monitor loop. So, on an extension state update we have (1. global context lock ----> 2. tech_pvt lock). Typically when we have to do a reversed locking order like this we'd just do some sort of deadlock avoidance to fix the problem... That will not work here. There are more locks involved here than just the context and tech_pvt. Those are the two that are colliding, but it is impossible to give up the context lock because the global hints list lock MUST be held as well and we can not give that lock up during the extensionstate callback traversal... The locking order for the context and hints are (1. global context lock ----> 2. hints list lock). Deadlock avoidance is not an option here. SOLUTION: The solution this patch implements is to queue the extension state updates into a list and send the NOTIFY messages out during the do_monitor pvt traversal. This clears out the problem of having to hold the context lock before the tech_pvt lock entirely. (closes issue #17888) Reported by: zerohalo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@286070 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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7fd5264698 |
Inherit CHANNEL() writes to both sides of a Local channel
Having Local (/n) channels as queue members and setting the language in the extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2 channel. Hold time report playbacks happen on the Local/...,1 channel and therefor do not play in the specified language. This patch modifies func_channel_write to call the setoption callback and pass the CHANNEL() write info to the callback. chan_local uses this information to look up the other side of the channel and apply the same changes to it. (closes issue #17673) Reported by: Guggemand Review: https://reviewboard.asterisk.org/r/903/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@286059 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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619807167e |
Missing newline
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@286023 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ab40f7e68d |
Fix Mac OS X build.
This also fixes a rather grievous calculation error for the offset of ast_fdset, which was masked on Linux and FreeBSD, because these platforms check the first 256 FDs regardless of the bitmask setting (due to backwards compatibility). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@285889 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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725e695859 |
GCC 4.2.x optimizations result in improper behavior of GSM codec
(closes issue #17688) Reported by: pprindeville Patches: asterisk-trunk-bugid11243.patch uploaded by pprindeville (license 347) Tested by: mkeuter, pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@285817 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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7f9358db02 |
Transmit silence when reading DTMF in ast_readstring.
Otherwise, you could get issues with DTMF timeouts causing hangups. (closes issue #17370) Reported by: makoto Patches: channel-readstring-silence-generator.patch uploaded by makoto (license 38) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@285742 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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314ac1bea6 |
Fixes an issue with MOH where it doesn't recover cleanly when it can't play a file and would just stop, instead of continuing to find the next playable file in the MOH class.
(closes issue #17807) Reported by: kshumard Review: https://reviewboard.asterisk.org/r/910/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@285638 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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49e083523d |
In retrans_pkt, do not unlock pvt until the end of the function on a transmit failure.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@285566 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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3d70384b2f |
Catch invalid extensions at the parser, instead of making the core deal with them.
(closes issue #17794) Reported by: PavelL Patches: 20100820__issue17794__1.6.2.diff.txt uploaded by tilghman (license 14) 20100820__issue17794__1.4.diff.txt uploaded by tilghman (license 14) Tested by: PavelL git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@285365 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a91eaba875 |
Use poll, if indicated to do so, in the ast_poll2 implementation.
This fixes the unit tests on FreeBSD 8.0. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@285266 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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1811246e48 |
Fixes voicemail.conf issues where mailboxes with passwords that don't precede a comma would throw unnecessary error messages.
(closes issue #15726) Reported by: 298 Patches: M15726.diff uploaded by junky (license 177) Tested by: junky Review: [full review board URL with trailing slash] git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@285194 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0653aeef5d |
Silly convenience script for BSD platforms.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@285088 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a2bf7447e2 |
Properly detect when a sound file doesn't exist
ast_fileexists returns -1 for error and 0 for a non-existant file. The existing code treated missing files as though they were existed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@284881 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a02ffb0a61 |
Fixes a bug in manager.c where the default configuration values weren't reset when the manager configuration was reloaded.
(closes issue #17917) Reported by: lmadsen Review: https://reviewboard.asterisk.org/r/883/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@284777 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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f9927f1291 |
Removed relatedpeer code from sip_autodestruct
Handling of the relatedpeer structure associated with a sip_pvt should be done during the final sip_destruction function, not in sip_autodestruct. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@284703 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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408cf5d2a2 |
Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows.
This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing a potential crash bug in all supported releases. (closes issue #17678) Reported by: russell Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select Review: https://reviewboard.asterisk.org/r/824/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@284478 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a8742f4a9a |
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/1.4@284393 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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28a0b35087 |
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/1.4@284316 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0e9fb94288 |
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/1.4@283960 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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d5c19c4c98 |
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/1.4@283880 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a0e2fe78a8 |
Use ast_free since ast_variable_new uses ast_calloc
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@283834 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c26e47f289 |
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/1.4@283690 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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3550593f30 |
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/1.4@283380 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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498cbf96bf |
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/1.4@283123 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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34806f56f5 |
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/1.4@283048 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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d04e866925 |
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/1.4@282893 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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1c0a484763 |
Add some documentation about codec negotiation to sip.conf
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@282729 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c5278fc111 |
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/1.4@282430 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e30457f021 |
Register CLI commands before parsing config, in case there is a config error.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@282129 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e90fb11e61 |
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/1.4@281911 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e002fc6cf1 |
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/1.4@281819 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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eb704b0fe3 |
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/1.4@281762 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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1c72a8ab49 |
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/1.4@281566 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5be00864ee |
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/1.4@281390 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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4f90553a45 |
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/1.4@281185 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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b7cb184823 |
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/1.4@280982 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5ec4ea248c |
Copy astcli back to 1.4 so it's available for automated testing purposes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@280944 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |