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4950 Commits (0135ff19edd53c46c435f9de9eab57f42eba4a4a)
Author | SHA1 | Message | Date |
---|---|---|---|
|
919ad38940 |
Improve handling of REGISTER requests with multiple contact headers.
The changes here attempt to more strictly follow RFC 3261 section 10.3. Basically the following will now cause a 400 Bad Response to be returned, if: - multiple Contact headers are present with one set to expire all bindings ("*") - wildcard parameter is specified for Contact without Expires header or Expires header is not set to zero. ABE-2442 ABE-2443 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@297603 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
8b2304108a |
If we get a NOTIFY from a non-existing subscription we should answer with 481, not bad event.
If we answer 481 the subscription that we don't want will be cancelled. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@297185 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
046f56ce1c |
Fix not stopping MOH when transfered local channel queue member is answered.
The problem here is only present when local channels are used with the MOH passthru option as well as no optimization (/nm). I will describe the slightly bizarre scenario that was used to test, where phones B and C are queue members: Phone A dials into a queue with two members using local channels and the above options. Phone B answers. Phone A blind transfers phone B into the same queue. Phone A hangs up. Phone C answers, but phone B didn't stop playing MOH. In this scenario, the unhold frame that should have gotten to phone B never arrived due to the masquerade from the blind transfer. This is usually fine since app_queue manages the starting and stopping of MOH. However, with the passthrough option enabled when app_queue attempts to stop MOH it tries to do so on the local channel rather than the real channel. The easiest solution was to just make sure to send an unhold frame during the transfer since it wouldn't make sense to have MOH playing after a transfer anyway. This only modifies SIP transfers, but the other transfers did not seem to be a problem. If DTMF based transfers were a problem it might be okay to add ast_moh_stop to finishup, but I didn't want to have to add that unless required. ABE-2624 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@297072 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
196032208f |
Get rid of the annoying startup and shutdown errors on OS X.
This mainly deals with the problem of constructors on platforms where an explicit constructor order cannot be specified (any system with gcc 4.2 or less). However, this is only a problem on those systems where we need to initialize mutexes with a constructor, because we have other code that also relies upon constructors, and we cannot specify that mutexes are initialized first (and destroyed last). There are two approaches to dealing with this issue, related to whether the code exists in the core Asterisk binary or in a separate code module. In the core case, constructors are run immediately upon load, and the file_versions list mutex needs to be already initialized, as it is referenced in the first constructor within each core source file. In this case, we use pthread_once to ensure that the mutex is initialized immediately before it is used for the first time. The only caveat is that the mutex is not ever destroyed, but because this is the core, it makes no real difference; the only time when destruction is safe would be just prior to process destruction, which takes care of that anyway. And due to using pthread_once, the mutex will never be reinitialized, which means only one structure has leaked at the end of the process. Hence, it is not a problematic leak. The second approach is to use the load_module and unload_module routines, which, for obvious reasons, exist only in loadable modules. In this second case, we don't have a problem with the constructors, but only with destructor order, because mutexes can be destroyed before their final usage is employed. However, we need the mutexes to still be destroyed, in certain scenarios: if the module is unloaded prior to the process ending, it should be clean, with no allocations by the module hanging around after that point in time. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@296867 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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728bb5ac67 |
Make sure nothing else is needed before destroying the scheduler.
(closes issue #18398) Reported by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@296670 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c98862958a |
Oneway audio to SIP phone from FXS port after FXS port gets a CallWaiting pip.
The FXS connected phone has to have CW/CID support to fail, as it will send back a DTMF 'A' or 'D' when it's ready to receive CallerID. A normal phone with no CID never fails. Also the SIP phone does not hear MOH when the CW call is answered. The DTMF end frame is suppressed when the phone acknowledges the CW signal for CID. The problem is the DTMF begin frame needs to be suppressed as well. The DTMF begin frame is causing SIP to start sending the DTMF RTP frames. Since the DTMF end frame is suppressed, SIP will not stop sending those DTMF RTP packets. * Suppress the DTMF begin and end frames when the channel driver is looking for DTMF digits. * Fixed a couple issues caused by not cleaning up the CID spill if you answer the CW call while it is sending the CID spill. * Fixed not sending CW/CID spill to the phone when the call is natively bridged. (Fixed by not using native bridge if CW/CID is possible.) * Suppress received audio when sending CW/CID spills. The other parties involved do not need to hear the CW/CID spills and may be confused if the CW call is for them. (closes issue #18129) Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett NOTE: * v1.4 does not have the main problem fixed by suppressing the DTMF start frames. The other three items fixed are relevant. * If you really must restore native bridging between analog ports, you need to disable CW/CID either by configuring chan_dahdi.conf callwaitingcallerid=no or dialing *70 before dialing the number to temporarily disable CW. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@296165 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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62d947a168 |
Discard responses with more than one Via
This is not a perfect solution as headers that are joined via commas are not detected. This is a parsing issue that to fix "correctly" would necessitate a new SIP parser. Review: https://reviewboard.asterisk.org/r/1019/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@295628 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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156f8c7d62 |
Asterisk is getting a "No D-channels available!" warning message every 4 seconds.
Asterisk is just whining too much with this message: "No D-channels available! Using Primary channel XXX as D-channel anyway!". Filtered the message so it only comes out once if there is no D channel available without an intervening D channel available period. (closes issue #17270) Reported by: jmls git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@294821 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ae7edf9a55 |
Fix problem with qualify option packets for realtime peers never stopping.
The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. This scenario has the potential to progress to the point of saturating a link just from options packets. The fix was to ensure that the poke scheduler checks to see if a peer is in the peer list before continuing to poke. The reason a peer must be in the peer list to be able to properly manage an options dialog is because otherwise the call pointer is lost when the peer is regenerated from the database, which is how existing qualify dialogs are detected. (closes issue #16382) Reported by: lftsy Patches: bug16382-3.patch uploaded by jpeeler (license 325) Tested by: zerohalo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@294688 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ea7e7cd8ed |
Modify our handling of 491 responses to drop any pending reinvite retry scheduler entries if we get a new 491.
This prevents a scheduler entry from leaking if we receive a 491 response when one is pending. If a scheduler entry leaks, the pvt it is associated my get destroyed before the scheduler entry fires, and then memory corruption and crashes can occur when the scheduled reinvite attempts to access and modify the memory of the destroyed pvt. ABE-2543 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@294163 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
1791cb12df |
Party A in an analog 3-way call would continue to hear ringback after party C answers.
All parties are analog FXS ports. 1) A calls B. 2) A flash hooks to call C. 3) A flash hooks to bring C into 3-way call before C answers. (A and B hear ringback) 4) C answers 5) A continues to hear ringback during the 3-way call. (All parties can hear each other.) * Fixed use of wrong variable in dahdi_bridge() that stopped ringback on the wrong subchannel. * Made several debug messages have more information. A similar issue happens if B and C are SIP channels. B continues to hear ringback. For some reason this only affects v1.8 and trunk. * Don't start ringback on the real and 3-way subchannels when creating the 3-way conference. Removing this code is benign on v1.6.2 and earlier. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@293805 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ab41baedbd |
Add enabled/disabled information for rtautoclear sip show settings output.
When setting to zero/"no", the numeric default was shown making it not obvious the disabled setting was respected. (closes issue #18123) Reported by: zerohalo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@293722 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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eca91ef1f3 |
Make warning message have more useful information in it.
Change "Unable to get index, and nullok is not asserted" to "Unable to get index for '<channel-name>' on channel <number> (<function>(), line <number>)". git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@293639 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
0761ad1f97 |
Remove some more code that serves no purpose.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@293416 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
bff8c6e7dd |
Remove some code that serves no purpose.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@293339 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b7adb57110 |
This patch turns chan_local pvts into astobj2 objects.
chan_local does some dangerous things involving deadlock avoidance. tech_pvt functions like hangup and queue_frame are provided with a locked channel upon entry. Those functions are completely safe as long as you don't attempt to give up that channel lock, but that is impossible to guarantee due to the required deadlock avoidance necessary to lock both the tech_pvt and both channels involved. In the past, we have tried to account for this by doing things like setting a "glare" flag that indicates what function should destroy the pvt. This was used in local_hangup and local_queue_frame to decided who should destroy the pvt if they collided in separate threads. I have removed the need to do this by converting all chan_local tech_pvts to astobj2. This means we can ref a pvt before deadlock avoidance and not have to worry about that pvt possibly getting destroyed under us. It also cleans up where we destroy the tech_pvt. The only unlink from the tech_pvt container occurs in local_hangup now, which is where it should occur. Since there still may be thread collisions on some functions like local_hangup after deadlock avoidance, I have added some checks to detect those collisions and exit appropriately. I think this patch is going to solve quite a bit of weirdness we have had with local channels in the past. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@292866 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
7922632617 |
Don't access o->next after freeing o on unload
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@291862 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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31668a2971 |
Deadlock between dahdi_exception() and dahdi_indicate().
There is a deadlock between dahdi_exception() and dahdi_indicate() for analog ports. The call-waiting and three-way-calling feature can experience deadlock if these features are trying to do something and an event from the bridged channel happens at the same time. Deadlock avoidance code added to obtain necessary channel locks before attemting an operation with call-waiting and three-way-calling. (closes issue #16847) Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett Review: https://reviewboard.asterisk.org/r/971/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@291643 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d0c73ff1b6 |
Lock pvt so pvt->owner can't disappear when queueing up a frame.
This fixes a crash due to a hangup race condition. ABE-2601 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@291392 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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85ef0289ca |
Add missing unlock to an exception condition in reload_config().
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@291109 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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8f6ca92a9d |
Change RFC2833 DTMF event duration on end to report actual elapsed time.
The scenario here is with a non P2P early media session. The reported time length of DTMF presses are coming up short when sending to the remote side. Currently the event duration is a running total that is incremented when sending continuation packets. These continuation packets are only triggered upon incoming media from the remote side, which means that the running total probably is not going to end up matching the actual length of time Asterisk received DTMF. This patch changes the end event duration to be lengthened if it is detected that the end event is going to come up short. Review: https://reviewboard.asterisk.org/r/957/ ABE-2476 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@289797 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
15 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 |
15 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 |
15 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 |
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 |