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2011-01-04Don't authenticate SUBSCRIBE re-transmissionstwilson1-21/+26
This only skips authentication on retransmissions that are already authenticated. A similar method is already used for INVITES. This is the kind of thing we end up having to do when we don't have a transaction layer... (closes issue #18075) Reported by: mdu113 Patches: diff.txt uploaded by twilson (license 396) Tested by: twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@300216 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-25Move check for extension existence below variable inheritance, due to the ↵tilghman1-6/+6
possible use of an eswitch. (closes issue #16228) Reported by: jlaguilar git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@299624 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-20Use ast_free() instead of free()mnicholson1-4/+4
ABE-2656 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@299220 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-20Remove changes to via processing that were not supposed to go into the last ↵mnicholson1-71/+0
commit. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@299198 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-20Respond as soon as possible with a 202 Accepted to refer requests.mnicholson1-26/+101
This change also plugs a few memory leaks that can occur when parking sip calls. ABE-2656 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@299194 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-20Typos: recieved => receivedtzafrir1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@299002 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-13Outgoing PRI/BRI calls cannot do DTMF triggered transfers.rmudgett1-1/+8
Outgoing PRI/BRI calls cannot do DTMF triggered transfers if a PROCEEDING message is not received. The debug output shows that the DTMF begin event is seen, but the DTMF end event is missing. When the DTMF begin happens, the call is muted so we now have one way audio (until a DTMF end event is somehow seen). * Made set the proceeding flag when the PRI_EVENT_ANSWER event is received. * Made absorb the DTMF begin and DTMF end events if we are overlap dialing and have not seen a PROCEEDING message. * Added a debug message when absorbing a DTMF event. JIRA SWP-2690 JIRA ABE-2697 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@298193 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-09Ignore spurious REGISTER requeststwilson1-0/+7
If a REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. This info is used to generate messages for other responses in the transaction. This patch ignores REGISTER requests that match non-REGISTER transactions. (closes issue #18051) Reported by: eeman Tested by: twilson Review: https://reviewboard.asterisk.org/r/1050/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297959 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-06Improve handling of REGISTER requests with multiple contact headers.jpeeler1-6/+30
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: http://svn.digium.com/svn/asterisk/branches/1.4@297603 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-02If we get a NOTIFY from a non-existing subscription we should answer with ↵oej1-1/+1
481, not bad event. If we answer 481 the subscription that we don't want will be cancelled. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297185 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Fix not stopping MOH when transfered local channel queue member is answered.jpeeler1-0/+3
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: http://svn.digium.com/svn/asterisk/branches/1.4@297072 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Get rid of the annoying startup and shutdown errors on OS X.tilghman1-1/+13
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: http://svn.digium.com/svn/asterisk/branches/1.4@296867 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-29Make sure nothing else is needed before destroying the scheduler.pabelanger1-2/+2
(closes issue #18398) Reported by: pabelanger git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@296670 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-24Oneway audio to SIP phone from FXS port after FXS port gets a CallWaiting pip.rmudgett1-108/+202
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: http://svn.digium.com/svn/asterisk/branches/1.4@296165 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-19Discard responses with more than one Viatwilson1-3/+19
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: http://svn.digium.com/svn/asterisk/branches/1.4@295628 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-12Asterisk is getting a "No D-channels available!" warning message every 4 ↵rmudgett1-2/+11
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: http://svn.digium.com/svn/asterisk/branches/1.4@294821 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-11Fix problem with qualify option packets for realtime peers never stopping.jpeeler1-1/+24
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: http://svn.digium.com/svn/asterisk/branches/1.4@294688 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-08Modify our handling of 491 responses to drop any pending reinvite retry ↵mnicholson1-0/+8
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: http://svn.digium.com/svn/asterisk/branches/1.4@294163 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-03Party A in an analog 3-way call would continue to hear ringback after party ↵rmudgett1-12/+16
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: http://svn.digium.com/svn/asterisk/branches/1.4@293805 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-02Add enabled/disabled information for rtautoclear sip show settings output.jpeeler1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.4@293722 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-02Make warning message have more useful information in it.rmudgett1-2/+5
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: http://svn.digium.com/svn/asterisk/branches/1.4@293639 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-30Remove some more code that serves no purpose.rmudgett1-11/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@293416 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-30Remove some code that serves no purpose.rmudgett1-11/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@293339 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-25This patch turns chan_local pvts into astobj2 objects.dvossel1-139/+167
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: http://svn.digium.com/svn/asterisk/branches/1.4@292866 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-15Don't access o->next after freeing o on unloadtwilson1-3/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@291862 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Deadlock between dahdi_exception() and dahdi_indicate().rmudgett1-80/+222
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: http://svn.digium.com/svn/asterisk/branches/1.4@291643 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Lock pvt so pvt->owner can't disappear when queueing up a frame.russell1-1/+16
This fixes a crash due to a hangup race condition. ABE-2601 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@291392 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Add missing unlock to an exception condition in reload_config().rmudgett1-3/+8
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@291109 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01Change RFC2833 DTMF event duration on end to report actual elapsed time.jpeeler1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.4@289797 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01Ensure user portion of SIP URI matches dialplan when using encoded characters.jpeeler1-3/+8
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: http://svn.digium.com/svn/asterisk/branches/1.4@289699 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-01don't iterate through all dialogs to find and delete old subscribesschmitds1-26/+6
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: http://svn.digium.com/svn/asterisk/branches/1.4@289622 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-24Don't fail a masquerade if it is already being hung uptwilson1-1/+1
This avoids noise on some Local channel situations where we don't use /n. Thanks to Alec Davis for the suggestion. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@288746 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-22Don't let a Local channel get bridged to itselftwilson1-0/+9
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: http://svn.digium.com/svn/asterisk/branches/1.4@288499 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-22RFC3261 section 12.2 explicitly says out of order requests are responded ↵dvossel1-1/+1
with a 500 Server Internal Error response. ABE-2458 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@288416 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-22Handle media specific T.38 SDP informationmnicholson1-2/+10
(closes issue #16647) Reported by: kwemheuer git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@288412 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-22During check_pendings, if the dialog is terminated with a CANCEL, change the ↵dvossel1-2/+3
invitestate to INV_CANCEL like in sip_hangup. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@288343 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21In chan_iax2.c:schedule_delivery() calls ast_bridged_channel() on an ↵rmudgett1-90/+130
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: http://svn.digium.com/svn/asterisk/branches/1.4@288192 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Try both the encoded and unencoded subscription URI for a match in hints.tilghman1-9/+22
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: http://svn.digium.com/svn/asterisk/branches/1.4@288112 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-14Don't clear the username from a realtime database when a registration expires.mnicholson1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.4@286756 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Return -1 if chan_local doesn't support an optiontwilson1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@286222 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Load iax.conf before registering any functions/applications/actions.pabelanger1-17/+21
Review: https://reviewboard.asterisk.org/r/914/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@286114 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10An outgoing call may not get hung up if a pre-connect incoming ISDN call is ↵rmudgett1-0/+16
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: http://svn.digium.com/svn/asterisk/branches/1.4@286113 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Fixes sip extension state update DEADLOCKdvossel1-10/+234
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: http://svn.digium.com/svn/asterisk/branches/1.4@286070 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Inherit CHANNEL() writes to both sides of a Local channeltwilson1-0/+67
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: http://svn.digium.com/svn/asterisk/branches/1.4@286059 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-09Transmit silence when reading DTMF in ast_readstring.qwell7-197/+362
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: http://svn.digium.com/svn/asterisk/branches/1.4@285742 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-03Prevent DAHDI channels from overriding the callerid, once it's been set by ↵tilghman1-3/+3
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: http://svn.digium.com/svn/asterisk/branches/1.4@280811 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Backport fixes for sip_uri_params_cmp() from trunk.mmichelson1-39/+25
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@279053 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-22DNID does not get cleard on a new call when using immediate=yes with ISDN ↵rmudgett1-4/+4
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: http://svn.digium.com/svn/asterisk/branches/1.4@278701 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-19Regression with T.38 negotiationpabelanger1-7/+7
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: http://svn.digium.com/svn/asterisk/branches/1.4@277944 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Default to no udptl error correction so that error correction will be ↵mnicholson1-0/+5
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: http://svn.digium.com/svn/asterisk/branches/1.4@277497 f38db490-d61c-443f-a65b-d21fe96a405b
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