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4929 Commits (11e86ae8f0be26bd371b4a33e4cd48fb2ac97025)
Author | SHA1 | Message | Date |
---|---|---|---|
|
11e86ae8f0 |
Ensure user portion of SIP URI matches dialplan when using encoded characters.
This commit takes a simliar approach to 288112 and checks the dialplan to determine the proper action for an incoming contact header as to whether or not it should be decoded or not. sip_new was blindly always decoding the extension, which also caused the outgoing contact header to be incorrect as well as failing to match the encoded extension in the dialplan. (closes issue #17892) Reported by: wdoekes Patches: bug17892-1.patch uploaded by jpeeler (license 325) Tested by: wdoekes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@289699 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
4d84c4d68a |
don't iterate through all dialogs to find and delete old subscribes
On every incoming subscribe there is a iteration through all dialogs to find old subscribes and delete them. This is slow and not RFC conform. This was only needed in 1.2 cause a subscribe was not deleted when a dialog was destroyed, after 1.4 a subscribe get removed when its dialog is destroyed. (closes issue #17950) Reported by: schmidts Tested by: schmidts Review: https://reviewboard.asterisk.org/r/901/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@289622 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
a4aea9fbd0 |
Don't fail a masquerade if it is already being hung up
This avoids noise on some Local channel situations where we don't use /n. Thanks to Alec Davis for the suggestion. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@288746 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
947cea177f |
Don't let a Local channel get bridged to itself
If a local channel gets bridged to itself, it becomes orphaned with no devices left to actually tell it to hang up. This patch modifies local_fixup() to detect this case and deny it. Review: https://reviewboard.asterisk.org/r/934 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@288499 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
a0579869a7 |
RFC3261 section 12.2 explicitly says out of order requests are responded with a 500 Server Internal Error response.
ABE-2458 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@288416 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
ffeacadcea |
Handle media specific T.38 SDP information
(closes issue #16647) Reported by: kwemheuer git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@288412 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
abf7e809c6 |
During check_pendings, if the dialog is terminated with a CANCEL, change the invitestate to INV_CANCEL like in sip_hangup.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@288343 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
4c511da0c1 |
In chan_iax2.c:schedule_delivery() calls ast_bridged_channel() on an unlocked channel.
Near the beginning of schedule_delivery(), ast_bridged_channel() is called on iaxs[fr->callno]->owner. However, the channel is not locked, which can result in ast_bridged_channel() crashing should owner->tech change to a technology that doesn't implement bridged_channel. I also fixed the other calls to ast_bridged_channel() in chan_iax2.c since the owner lock was not held there either. Converted the existing channel deadlock avoidance to use iax2_lock_owner(). Using the new function simplified some awkward code. In the process of fixing the locking on ast_bridged_channel(), I also found a memory leak in socket_process() for v1.6.2 and v1.8. The local struct variable ies.vars is not freed on early/abnormal function exits. (closes issue #17919) Reported by: rain Patches: issue17919_v1.4.patch uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch uploaded by rmudgett (license 664) Review: https://reviewboard.asterisk.org/r/926/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@288192 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3fec8c28ad |
Try both the encoded and unencoded subscription URI for a match in hints.
When a phone sends an encoded URI for a subscription, the URI is not matched with the actual hint that is in decoded format. For example, if we have an extension with a hint that is named: "#5601" or "*5601", the subscription will work fine if the phone subscribes with an already decoded URI, but when it's decoded like "%255601" or "%2A5601", Asterisk is unable to match it with the correct hint. (closes issue #17785) Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt uploaded by tilghman (license 14) Tested by: ramonpeek git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@288112 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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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 |
15 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 |
15 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 |
15 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 |
15 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 |
15 years ago |
|
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 |
15 years ago |
|
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 |
15 years ago |
|
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 |
15 years ago |
|
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 |
15 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 |
15 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 |
15 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 |
15 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 |
15 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 |
15 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 |
15 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 |
15 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 |
15 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 |
15 years ago |
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996a394a7d |
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/1.4@280811 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7bc66e1d77 |
Backport fixes for sip_uri_params_cmp() from trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@279053 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
8f60bcf42f |
DNID does not get cleard on a new call when using immediate=yes with ISDN signaling.
When you are using chan_dahdi ISDN signaling with immediate=yes and a call comes in without a DNID then you get the DNID of a previous call. Chan_dahdi does not touch the DNID field on a new call if it does not have a DNID. Made always copy the DNID from the new call. The patches backport the relevant changes from trunk -r210387. (closes issue #17568) Reported by: wuwu Patches: issue17568_v1.4.patch uploaded by rmudgett (license 664) issue17568_v1.6.2.patch uploaded by rmudgett (license 664) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@278701 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
b48d6c3dbe |
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/1.4@277944 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
1c8d76145b |
Default to no udptl error correction so that error correction will be disabled in the event that the remote end indicates that they do not support the error correction mode we requested.
FAX-128 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@277497 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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308d812403 |
priexclusive in chan_dahdi.conf ignored when reloading dahdi module
During a reload, the priexclusive and outsignalling parameters are not read in from the config file as intended. Unfortunately, they get set to defaults as a result. This patch makes sure that they do not get set to defaults during a reload. (closes issue #17441) Reported by: mtryfoss Patches: issue17441_v1.4.patch uploaded by rmudgett (license 664) issue17441_v1.6.2.patch uploaded by rmudgett (license 664) issue17441_trunk.patch uploaded by rmudgett (license 664) Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@277419 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
dca914c43b |
Fix logging message for stale nonce.
(closes issue #17582) Reported by: kenner Patches: chan_sip.c.diff uploaded by kenner (license 1040) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@275241 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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8c1f4940eb |
Close the DAHDI FD on error when processing chan_dahdi toneduration config parameter.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@274579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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79d795c383 |
Add option to not do a call forward on 482 Loop Detected
Asterisk has always set up a forwarded call when receiving a 482 Loop Detected. This prevents handling the call failure by just continuing on in the dialplan. Since this would be a change in behavior, the new option to disable this behavior is forwardloopdetected which defaults to 'yes'. Review: https://reviewboard.asterisk.org/r/764/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@274280 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
b0202ce6d1 |
Command 'stop gracefully' doesn't.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273981 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
14550f93d0 |
Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs.
(closes issue #17407) Reported by: pdf Patches: 20100527__issue17407.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/751/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273793 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
97f61244f9 |
Fix various typos, reported by Lintian
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273640 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
65458abbe8 |
Allow the "useragent" value to be restored into memory from the realtime backend.
This value is purely informational. It does not alter configuration at all. (closes issue #16029) Reported by: Guggemand Patches: realtime-useragent.patch uploaded by Guggemand (license 897) Tested by: Guggemand git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273060 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
2cc343faa2 |
Decode URI in contact header of 302 response.
ABE-2352 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@272804 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
fac72f08b8 |
ss_thread calls pri_grab without lock during overlap dial
Recent changes to chan_dahdi with relation to overlap dialing call pri_grab without first obtaining a lock. (closes issue #17414) Reported by: pdf Patches: bug17414.patch uploaded by jpeeler (license 325) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@272446 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
7eabbade0a |
Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the ref count correct.
(closes issue #16815) Reported by: rain Patches: chan_sip-unref-fix.diff uploaded by rain (license 327) (modified) Tested by: rain git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@271902 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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2d31f18ae9 |
Modify chan_sip's packet generation api to automatically calculate the Content-Length. This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated. This change was made to ensure that the Content-Length is always correct.
(closes issue #17326) Reported by: kenner Tested by: mnicholson, kenner Review: https://reviewboard.asterisk.org/r/693/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@271689 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5cf7c69afb |
Eliminate deadlock potential in dahdi_fixup().
(This is a backport of 269307, committed to trunk by rmudgett.) Calling dahdi_indicate() when the channel private lock is already held can cause a deadlock if the PRI lock is needed because dahdi_indicate() will also get the channel private lock. The pri_grab() function assumes that the channel private lock is held once to avoid deadlock. (closes issue #17261) Reported by: aragon git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@271335 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
25a613a993 |
Need to lock the agent chan before access its internal bits.
Pointed out by russellb on asterisk-dev mailing list. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@270980 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
3245bb4b86 |
fixes chan_iax2 race condition
There is code in chan_iax2.c that attempts to guarantee that only a single active thread will handle a call number at a time. This code works once the thread is added to an active_list of threads, but we are not currently guaranteed that a newly activated thread will enter the active_list immediately because it is left up to the thread to add itself after frames have been queued to it. This means that if two frames come in for the same call number at the same time, it is possible for them to grab two separate threads because the first thread did not add itself to the active_list fast enough. This causes some pretty complex problems. This patch resolves this race condition by immediately adding an activated thread to the active_list within the network thread and only depending on the thread to remove itself once it is done processing the frames queued to it. By doing this we are guaranteed that if another frame for the same call number comes in at the same time, that this thread will immediately be found in the active_list of threads. Review: https://reviewboard.asterisk.org/r/720/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@270866 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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21bdb89a2b |
Merged revisions 270658 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r270658 | twilson | 2010-06-15 15:18:04 -0500 (Tue, 15 Jun 2010) | 20 lines Make contactdeny apply to src ip when nat=yes chan_sip's "contactdeny" feature screens the "to be registered contact". In case of nat=yes it should not use the address information from the Contact header (which is not used at all for routing), but the source IP address of the request. Thus, if nat=yes and a client sends a request from a denied IP address (e.g. by spoofing the src-IP address) it can bypass the screening. This commit makes contactdeny apply to the src ip when nat=yes instead. (closes issue #17276) Reported by: klaus3000 Patches: patch-asterisk-trunk-contactdeny.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 Review: [full review board URL with trailing slash] ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@270724 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7d953183e6 |
fixes FXS port still ringing when answered, as reported by Tzafrir on dev-list.
(issue #17067) Reported by: tzafrir Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@270404 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a8f624dc93 |
Don't stop Asterisk if chan_oss fails to register 'Console' (due to another channel driver already claiming it).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@269495 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |