13588 Commits (38e98f42bc656c514b321a55bdfd5bb1c0a0fb55)

 

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Author	SHA1	Message	Date
Mark Michelson	a3848ec74c	Place unlock of mutex in an else block so that it does not get unlocked twice. ... (closes issue #15400) Reported by: aragon git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204012 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	f65dccafb6	The ISDN CPE side should not exclusively pick B channels normally. ... Before this patch, Asterisk unconditionally picked B channels exclusively on the CPE side and normally allowed alternative B channels on the network side. Now Asterisk does the opposite. Reasons for the CPE side to normally not pick B channels exclusively: * For CPE point-to-multipoint mode (i.e. phone side), the CPE side does not have enough information to exclusively pick B channels. (There may be other devices on the line.) * Q.931 gives preference to the network side picking B channels. * Some telcos require the CPE side to not pick B channels exclusively. (closes issue #14383) Reported by: mbrancaleoni git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203908 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	fc73897bbd	Make sure to recreate the dahdi pseudo channel after dahdi restart ... (closes issue #14477) Reported by: timking git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	fe7923abfc	Don't fast forward past the end of a message. ... This is nice change for users of the voicemail application. If someone gets a little carried away with fast forwarding through a message, they can easily get to the end and accidentally exit the voicemail application by hitting the fast forward key during the following prompt. This adds some safety by not allowing a fast forward past the end of a message. (closes issue #14554) Reported by: lacoursj Patches: 21761.patch uploaded by lacoursj (license 707) Tested by: lacoursj git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	64e75ecf80	Fixing voicemail's error in checking max silence vs min message length ... Max silence was represented in milliseconds, yet vmminsecs (minmessage) was represented as seconds. Also, the inequality was reversed. The warning, if triggered, was "Max silence should be less than minmessage or you may get empty messages", which should have been logged if max silence was greater than minmessage, but the check was for less than. Also, conforming if statement to coding guidelines. closes issue #15331) Reported by: markd Review: https://reviewboard.asterisk.org/r/293/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	5e3e234df6	I didn't see that Mark already fixed the underlying issue! ... Yay for removing useless code. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	090c2f1e4e	Fix a case where CDR answer time could be before the start time involving parking. ... (closes issue #13794) Reported by: davidw Patches: 13794.patch uploaded by murf (license 17) 13794.patch.160 uploaded by murf (license 17) Tested by: murf, dbrooks git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203375 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	03d3fb7a7a	Don't try to free NULL ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	140d84dfc5	Prevent false positives when freeing a NULL pointer with MALLOC_DEBUG enabled. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203230 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c05d6ceccd	Resolve a crash related to a T.38 reinvite race condition. ... This change resolves a crash observed locally during some T.38 testing. A call was set up using a call file, and when the T.38 reinvite came in, the channel state was still AST_STATE_DOWN. The reason is explained by a comment in the code that previously lived in the handling of AST_STATE_RINGING. This change modifies the logic to handle the same race condition for any channel state that is not UP. (closes ABE-1895) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	1ac27cf7ec	Improved chan_dahdi.conf pritimer error checking. ... Valid format is: pritimer=timer_name,timer_value * Fixed segfault if the ',' is missing. * Completely check the range returned by pri_timer2idx() to prevent possible access outside array bounds. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	a1fa4f0391	Use the handy UNLINK macro instead of hand-coding the same thing in-line. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202966 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	d6106936cb	MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port and transport ... (closes issue #14659) Reported by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel, klaus3000 Review: https://reviewboard.asterisk.org/r/288/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	f76b499923	Fix more memory leaks that may result if rtp is not successfully allocated. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202601 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	b0c0c17764	Fix potential memory leak in chan_sip when video rtp is not allocated properly. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202572 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	72b285ed96	Report CallerID change during a masquerade. ... Reported by: markster git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202496 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	dcfd8d7c7c	Make Polycom subscription type override check more explicit. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202414 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	26ba38b8f4	Remove an extra debug line left from previous commit. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202342 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	d31f78a172	Fix a situation in which Asterisk would not stop retransmitting 487s. ... If a CANCEL were received by Asterisk, we would send a 487 in response to the original INVITE and a 200 OK for the CANCEL. If there were a network hiccup which caused the 200 OK and the 487 to be lost, then the UA communicating with Asterisk may try to retransmit its CANCEL. Asterisk's response to this used to be to try sending another 487 to the canceled INVITE and another 200 OK to the CANCEL. The problem here is that the originally-sent 487 was sent "reliably" meaning that it will be retransmitted until it is received properly. So when we receive the second CANCEL it is likely that the first batch of 487s we sent is still going strong and reaches the UA. The result was that the second set of 487s would be retransmitted constantly until the maximum number of retries had been reached. The fix for this is that if we receive a second CANCEL for an INVITE, then we cancel the retransmission of the first set of 487s and start a second set. This causes the dialog to be terminated reasonably. (closes issue #14584) Reported by: klaus3000 Patches: 14584_v2.patch uploaded by mmichelson (license 60) Tested by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	1f7d3e9a01	Fix a possible infinite loop in SDP parsing during glare situation. ... There was a while loop in get_ip_and_port_from_sdp which was controlled by a call to get_sdp_iterate. The loop would exit either if what we were searching for was found or if the return was NULL. The problem is that get_sdp_iterate never returns NULL. This means that if what we were searching for was not present, the loop would run infinitely. This modification of the loop fixes the problem. (closes issue #15213) Reported by: schmidts (closes issue #15349) Reported by: samy (closes issue #14464) Reported by: pj (closes issue #15345) Reported by: aragon Patches: sip_inf_loop.patch uploaded by mmichelson (license 60) Tested by: aragon git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202336 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	f543251260	Since we don't have sip_pvt_lock() in 1.4, we need to use ast_mutex_* directly. ... (closes issue #15366) Reported by: loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	e735cdc36b	Added deadlock protection to try_suggested_sip_codec in chan_sip.c. ... Review: https://reviewboard.asterisk.org/r/287/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	f17d5d22d2	timestamp was being converted to host order as a short rather than a long ... (closes issue #15361) Reported by: ffloimair Patches: ts_issue.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@201993 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	5757b115b3	If the "h" extension fails, give it another chance in main/pbx.c. ... If the "h" extension fails, give it another chance in main/pbx.c, when it returns from the bridge code. Fixes an issue where the "h" extension may occasionally not fire, when a Dial is executed from a Macro. Debugged in #asterisk with user tompaw. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@201828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	fce4a98f7c	Fix memory corruption and leakage related reloads of non files mode MoH classes. ... For Music on Hold classes that are not files mode, meaning that we are executing an application that will feed us audio data, we use a thread to monitor the external application and read audio from it. This thread also makes use of the MoH class object. In the MoH class destructor, we used pthread_cancel() to ask the thread to exit. Unfortunately, the code did not wait to ensure that the thread actually went away. What needed to be done is a pthread_join() to ensure that the thread fully cleans up before we proceed. By adding this one line, we resolve two significant problems: 1) Since the thread was never joined, it never fully goes away. So, on every reload of non-files mode MoH, an unused thread was sticking around. 2) There was a race condition here where the application monitoring thread could still try to access the MoH class, even though the thread executing the MoH reload has already destroyed it. (issue #15109) Reported by: jvandal (issue #15123) Reported by: axisinternet (issue #15195) Reported by: amorsen (issue AST-208) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@201600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	03909de702	Change the datastore traversal in ast_do_masquerade to use a safe list traversal. ... It is possible for datastore fixup functions to remove the datastore from the list and free it. In particular, the queue_transfer_fixup in app_queue does this. While I don't yet know of this causing any crashes, it certainly could. Found while discussing a separate issue with Brian Degenhardt. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@201450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	86c204f34c	StopMixMonitor race condition (not giving up file immediately) ... StopMixMonitor only indicates to the MixMonitor thread to stop writing to the file. It does not guarantee that the recording's file handle is available to the dialplan immediately after execution. This results in a race condition. To resolve this, the filestream pointer is placed in a datastore on the channel. When StopMixMonitor is called, the datastore is retrieved from the channel and the filestream is closed immediately before returning to the dialplan. Documentation indicating the use of StopMixMonitor to free files has been updated as well. (closes issue #15259) Reported by: travisghansen Tested by: dvossel Review: https://reviewboard.asterisk.org/r/283/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@201423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	ebe2c1829b	Checks for NULL sip_pvt pointer in chan_sip.c->acf_channel_read() ... Zombie channels could be passed, and chan_sip.c wasn't checking for it. Could crash Asterisk. Now checking for NULL pointer. (closes issue #15330) Reported by: okrief Tested by: dbrooks git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@201380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	b8417b571b	Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty. ... When the list to be appended is empty, and the list to be appended to is *not*, AST_LIST_APPEND_LIST would actually cause the target list to become broken, and no longer have a pointer to its last entry. This patch fixes the problem. (reported by Stanislaw Pitucha on the asterisk-dev mailing list) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@201261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	94fa4d11b5	Improve support for media paths that can generate multiple frames at once. ... There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@200991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	a42ff13c97	Show the interface name on error, if it is not found. ... If the smdiport specified is not found, show the interface name instead of '(null)'. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@200875 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	c946be82a9	Add INFO to our allowed methods so that endpoints know they may send it to us. ... AST-223 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@200513 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	af2e7f5eab	Suppress a warning message and give a better return code when generating ... inband ringing after a call is answered. (closes issue #15158) Reported by: madkins Patches: 15158.patch uploaded by mmichelson (license 60) Tested by: madkins git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@200360 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	a34f50f08b	Backport fix for parallel build warnings from trunk r199781. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@200185 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Leif Madsen	c5ee7242d8	Fix path for .flavor and .version. ... (issue #14737) Reported by: davidw Patches: flavor.patch uploaded by davidw (license 780) Tested by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@200037 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	035b942a7a	__WORDSIZE is not available on all platforms, so use sizeof(void *) instead. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@199856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	aea9d7d060	Fix a typo in the stack size calculation just introduced. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@199628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	0d849d509d	Increase the size of our thread stack on 64 bit processors. ... We were setting the stack size for each thread to 240KB regardless of architecture, which meant that in some scenarios we actually had less available stack space on 64 bit processors (pointers use 8 bytes instead of 4). So now we calculate the stack size we reserve based on the platform's __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128 bit -> 1008KB (that's right, we're ready for 128 bit processors) Patch typed by me but written by several members of #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes issue #14932) Reported by: jpiszcz Patches: 06052009_issue14932.patch uploaded by seanbright (license 71) Tested by: seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@199626 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	d3bea6da02	Fixes issue with hints giving unexpected results. ... Hints with two or more devices that include ONHOLD gave unexpected results. (closes issue #15057) Reported by: p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel (license 671) pbx.c.1.4.patch uploaded by p (license 558) devicestate.c.trunk.patch uploaded by p (license 671) Tested by: p_lindheimer, dvossel Review: https://reviewboard.asterisk.org/r/254/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@199297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	ed94be12f0	Additional updates to AST-2009-001 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@199138 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	7605487610	Safely handle AMI connections/reload requests that occur during startup. ... During asterisk startup, a lock on the list of modules is obtained by the primary thread while each module is initialized. Issue 13778 pointed out a problem with this approach, however. Because the AMI is loaded before other modules, it is possible for a module reload to be issued by a connected client (via Action: Command), causing a deadlock. The resolution for 13778 was to move initialization of the manager to happen after the other modules had already been lodaded. While this fixed this particular issue, it caused a problem for users (like FreePBX) who call AMI scripts via an #exec in a configuration file (See issue 15189). The solution I have come up with is to defer any reload requests that come in until after the server is fully booted. When a call comes in to ast_module_reload (from wherever) before we are fully booted, the request is added to a queue of pending requests. Once we are done booting up, we then execute these deferred requests in turn. Note that I have tried to make this a bit more intelligent in that it will not queue up more than 1 request for the same module to be reloaded, and if a general reload request comes in ('module reload') the queue is flushed and we only issue a single deferred reload for the entire system. As for how this will impact existing installations - Before 13778, a reload issued before module initialization was completed would result in a deadlock. After 13778, you simply couldn't connect to the manager during startup (which causes problems with #exec-that-calls-AMI configuration files). I believe this is a good general purpose solution that won't negatively impact existing installations. (closes issue #15189) (closes issue #13778) Reported by: p_lindheimer Patches: 06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71) Tested by: p_lindheimer, seanbright Review: https://reviewboard.asterisk.org/r/272/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@199022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	f3b85fface	Fix a possible crash in pbx_spool. ... We were trying to reference members of a struct that had previously been freed. This patch makes sure that we free the struct after it has been removed from the spooler queue. (closes issue #15072) Reported by: garlew Patches: spool.diff uploaded by garlew (license 376) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@198957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	ddb4e3f2e7	Generic call forward api, ast_call_forward() ... The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and res_feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options. (closes issue #13630) Reported by: festr Review: https://reviewboard.asterisk.org/r/271/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@198891 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	af31809bcf	If using the old deprecated format, a reload would cause the class to disappear. ... (closes issue #14759) Reported by: lidocaineus Patches: 20090518__issue14759.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@198665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	8fc78eae7a	Properly terminate AMI JabberSend response messages. ... The response message (either Error or Success) needs an extra trailing \r\n after the fields to inform the client that the message is complete. (closes issue #14876) Reported by: srt Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright (license 71) asterisk_14876.patch uploaded by srt (license 378) trunk-14876-2.diff uploaded by phsultan (license 73) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@198370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	e86b26f1a8	Fix a crash that occurred when MWI SMDI messages expired. ... (closes issue #14561) Reported by: cmoss28 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@198311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	48253ef901	Treat an empty FORWARD_CONTEXT the same way we treat a missing one. ... (closes issue #15056) Reported by: p_lindheimer Patches: 05292009_bug15056.diff uploaded by seanbright (license 71) Tested by: p_lindheimer git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@198251 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	aa2fd9a4c2	Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition. ... This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels. (closes issue #12946) Reported by: meral Patches: null-cdr2.diff uploaded by mnicholson (license 96) Tested by: mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested by: sum git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@198068 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	be8d983874	Fix 'make config' target for Slackware. ... There was a missing semi-colon after the echo statement in the Makefile that was causing problems for some users. Fix suggested by reporter. (closes issue #15225) Reported by: pdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Leif Madsen	ad5f20b94b	Update MixMonitor documentation. ... Updated the MixMonitor documentation for the 'b' option so that it is more obvious that you must not optimize awat the Local channel when using this option. (issue #14829) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago