[svn-commits] lmadsen: tag 1.4.26-rc6 r206223 - /tags/1.4.26-rc6/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Jul 13 13:34:54 CDT 2009
Author: lmadsen
Date: Mon Jul 13 13:34:50 2009
New Revision: 206223
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=206223
Log:
Importing files for 1.4.26-rc6 release.
Added:
tags/1.4.26-rc6/.lastclean (with props)
tags/1.4.26-rc6/.version (with props)
tags/1.4.26-rc6/ChangeLog (with props)
Added: tags/1.4.26-rc6/.lastclean
URL: http://svn.asterisk.org/svn-view/asterisk/tags/1.4.26-rc6/.lastclean?view=auto&rev=206223
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--- tags/1.4.26-rc6/ChangeLog (added)
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+2009-07-13 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.26-rc6
+
+2009-07-13 15:12 +0000 [r206126] Russell Bryant <russell at digium.com>
+
+ * main/pbx.c: Print CID match in "show dialplan". (closes issue
+ #14702) Reported by: klaus3000 Patches:
+ patch_asterisk_1.4.23_CID_matching.txt uploaded by klaus3000
+ (license 65)
+
+2009-07-10 17:39 +0000 [r205877] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Properly ACK 487 responses to canceled
+ INVITEs. From the review board request: The fix from review 298
+ has exposed a new bug in chan_sip. When we hang up an outgoing
+ call, we first will dump all the outstanding packets on the
+ sip_pvt using __sip_pretend_ack. Then, if we can, we send a
+ CANCEL. The problem with this is that since destroyed all the
+ outstanding packets on the dialog, we cannot match the incoming
+ 487 response to our INVITE. Because we cannot match the response,
+ we do not send an ACK. To correct this, instead of using
+ __sip_pretend_ack, I have changed the code to loop through the
+ list of packets and call __sip_semi_ack on each one instead. This
+ causes us to stop retransmitting the requests, but we still have
+ them around in case we get responses for them. When discussing
+ this earlier today with Josh Colp, we both agreed that in the
+ majority of cases, this would be enough of a fix. However, we
+ also agreed that we should have a safety net in place in case we
+ never receive a response to our initial INVITE. To handle this, I
+ have modified __sip_autodestruct to behave similar to the way it
+ does in Asterisk 1.4. If there are outstanding packets on the
+ sip_pvt, the needdestroy flag is not set, and the last request on
+ the dialog was either a CANCEL or BYE, then we set the
+ needdestroy flag and reschedule destruction for 10 seconds in the
+ future. If, though, the needdestroy flag is set, then we use
+ __sip_pretend_ack to kill the remaining outstanding packets so
+ that the monitor thread can destroy the sip_pvt. I ran two
+ separate tests. First, I placed a call from my Aastra phone to my
+ Polycom phone. I hung up the Aastra before the Polycom answered.
+ I verified through sip debug output that Asterisk properly ACKed
+ the 487 received from the Polycom. For my second test, I set up a
+ SIPp UAS scenario so that it would send a 200 OK in response to a
+ CANCEL but would not send a 487 for the original INVITE. I
+ verified that after about 40 seconds, Asterisk properly cleans up
+ the outgoing sip_pvt for the call. Review:
+ https://reviewboard.asterisk.org/r/308
+
+2009-07-10 16:23 +0000 [r205804] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: SIP registration auth loop caused by stale
+ nonce If an endpoint sends two registration requests in a very
+ short period of time with the same nonce, both receive 401
+ responses from Asterisk, each with a different nonce (the second
+ 401 containing the current nonce and the first one being stale).
+ If the endpoint responds to the first 401, it does not match the
+ current nonce so Asterisk sends a third 401 with a newly
+ generated nonce (which updates the current nonce)... Now if the
+ endpoint responds to the second 401, it does not match the
+ current nonce either and Asterisk sends a fourth 401 with a newly
+ generated nonce... This loop goes on and on. There appears to be
+ a simple fix for this. If the nonce from the request does not
+ match our nonce, but is a good response to a previous nonce,
+ instead of sending a 401 with a newly generated nonce, use the
+ current one instead. This breaks the loop as the nonce is not
+ updated until a response is received. Additional logic has been
+ added to make sure no nonce can be responded to twice though.
+ (closes issue #15102) Reported by: Jamuel Patches:
+ patch-bug_0015102 uploaded by Jamuel (license 809) nonce_sip.diff
+ uploaded by dvossel (license 671) Tested by: Jamuel Review:
+ https://reviewboard.asterisk.org/r/289/
+
+2009-07-10 15:51 +0000 [r205775] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Ensure that outbound NOTIFY requests are
+ properly routed through stateful proxies. With this change, we
+ make note of Record-Route headers present in any SUBSCRIBE
+ request that we receive so that our outbound NOTIFY requests will
+ have the proper Route headers in them. (closes issue #14725)
+ Reported by: ibc
+
+2009-07-09 23:37 +0000 [r205728] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: No audio on calls from Asterisk to various
+ ISDN devices until DTMF sent by caller. Add missing clearing of
+ the dialing flag when the ISDN call is CONNECTED. (i.e. When
+ libpri generates the event PRI_EVENT_ANSWER.) (closes issue
+ #15420) Reported by: scottbmilne Patches:
+ bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585)
+ Tested by: scottbmilne, alecdavis (closes issue #15416) Reported
+ by: avinoash (closes issue #15389) Reported by: alecdavis This
+ patch should also fix the following issue: (issue #15205)
+ Reported by: vinsik
+
+2009-07-09 16:18 +0000 [r205409-205599] David Vossel <dvossel at digium.com>
+
+ * include/asterisk/time.h: Changing ast_samp2tv to not use floating
+ point.
+
+ * main/rtp.c, channels/chan_iax2.c, include/asterisk/frame.h: Fixes
+ 8khz assumptions Many calculations assume 8khz is the codec rate.
+ This is not always the case. This patch only addresses chan_iax.c
+ and res_rtp_asterisk.c, but I am sure there are other areas that
+ make this assumption as well. Review:
+ https://reviewboard.asterisk.org/r/306/
+
+ * include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c,
+ include/asterisk/pbx.h: moving ast_devstate_to_extenstate to
+ pbx.c from devicestate.c ast_devstate_to_extenstate belongs in
+ pbx.c. This change fixes a compile time error with chan_vpb as
+ well.
+
+2009-07-08 19:26 +0000 [r205349] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Prevent phantom calls to queue members. If a
+ caller were to hang up while a periodic announcement or position
+ were being said, the return value for those functions would
+ incorrectly indicate that the caller was still in the queue. With
+ these changes, the problem does not occur. (closes issue #14631)
+ Reported by: latinsud Patches: queue_announce_ghost_call2.diff
+ uploaded by latinsud (license 745) (with small modification from
+ me)
+
+2009-07-08 18:19 +0000 [r205288] Jason Parker <jparker at digium.com>
+
+ * config.guess, config.sub: Update config.guess and config.sub from
+ the savannah.gnu.org git repo.
+
+2009-07-08 16:53 +0000 [r205215] David Vossel <dvossel at digium.com>
+
+ * include/asterisk/time.h: ast_samp2tv needs floating point for
+ 16khz audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate
+ is 16000. The .5 is currently stripped off because we don't
+ calculate using floating points. This causes madness with 16khz
+ audio. (issue ABE-1899) Review:
+ https://reviewboard.asterisk.org/r/305/
+
+2009-07-08 16:26 +0000 [r205188] Tilghman Lesher <tlesher at digium.com>
+
+ * main/say.c: Add redirection warnings for the invalid language
+ codes previously removed.
+
+2009-07-08 15:54 +0000 [r205149] Russell Bryant <russell at digium.com>
+
+ * res/res_crypto.c: Make OpenSSL usage thread-safe. OpenSSL is not
+ thread-safe by default. However, making it thread safe is very
+ easy. We just have to provide a couple of callbacks. One callback
+ returns a thread ID. The other handles locking. For more
+ information, start with the "Is OpenSSL thread-safe?" question on
+ the FAQ page of openssl.org.
+
+2009-07-02 21:59 +0000 [r204834] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_misdn.c: Removed confusing warning message "Got
+ Busy in Connected State" If an incoming mISDN call is answered
+ with the Answer application and a subsequent Dial gets a busy
+ endpoint then it is valid for that already connected channel to
+ get the busy indication. Asterisk will play the busy tones until
+ the dialplan plays something else or hangs up the call. (closes
+ issue #11974) Reported by: fvdb
+
+2009-07-02 18:15 +0000 [r204755] David Vossel <dvossel at digium.com>
+
+ * include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c,
+ include/asterisk/pbx.h: moving device state functions from pbx.h
+ to devicestate.h to sync with other branches
+
+2009-07-02 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.26-rc5
+
+2009-07-02 15:05 +0000 [r204681] David Vossel <dvossel at digium.com>
+
+ * include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c,
+ include/asterisk/pbx.h: Improved mapping of extension states from
+ combined device states. This fixes a few issues with incorrect
+ extension states and adds a cli command, core show
+ device2extenstate, to display all possible state mappings.
+ (closes issue #15413) Reported by: legart Patches:
+ exten_helper.diff uploaded by dvossel (license 671) Tested by:
+ dvossel, legart, amilcar Review:
+ https://reviewboard.asterisk.org/r/301/
+
+2009-06-30 20:23 +0000 [r204556] Tilghman Lesher <tlesher at digium.com>
+
+ * main/say.c, UPGRADE.txt: More incorrect language codes, plus
+ ensuring that regionalizations use the specified language, and
+ not English for grammar. (closes issue #15022) Reported by:
+ greenfieldtech Patches: 20090519__issue15022.diff.txt uploaded by
+ tilghman (license 14)
+
+2009-06-30 18:47 +0000 [r204474] Jason Parker <jparker at digium.com>
+
+ * main/say.c: Fix ast_say_counted_noun to correctly handle Polish.
+ Fix a comment typo in passing.
+
+2009-06-30 18:23 +0000 [r204469] Tilghman Lesher <tlesher at digium.com>
+
+ * main/say.c, UPGRADE.txt: "tw" is the language specification for
+ Twi (from Ghana) not Taiwanese. (closes issue #15346) Reported
+ by: volivier Patches: 20090617__issue15346__1.4.diff.txt uploaded
+ by tilghman (license 14) 20090617__issue15346__trunk.diff.txt
+ uploaded by tilghman (license 14)
+ 20090617__issue15346__1.6.0.diff.txt uploaded by tilghman
+ (license 14) 20090617__issue15346__1.6.1.diff.txt uploaded by
+ tilghman (license 14) 20090617__issue15346__1.6.2.diff.txt
+ uploaded by tilghman (license 14) Tested by: volivier
+
+2009-06-29 22:45 +0000 [r204243-204300] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Add error message so that it is clear why a
+ SIP peer was not processed when a DNS lookup fails on a host or
+ outboundproxy. (closes issue #13432) Reported by: p_lindheimer
+ Patches: outboundproxy.patch uploaded by p (license 558)
+
+ * channels/chan_sip.c: Fix build oops.
+
+ * channels/chan_sip.c: Fix a problem where chan_sip would ignore
+ "old" but valid responses. chan_sip has had a problem for quite a
+ long time that would manifest when Asterisk would send multiple
+ SIP responses on the same dialog before receiving a response. The
+ problem occurred because chan_sip only kept track of the highest
+ outgoing sequence number used on the dialog. If Asterisk sent two
+ requests out, and a response arrived for the first request sent,
+ then Asterisk would ignore the response. The result was that
+ Asterisk would continue retransmitting the requests and ignoring
+ the responses until the maximum number of retransmissions had
+ been reached. The fix here is to rearrange the code a bit so that
+ instead of simply comparing the sequence number of the response
+ to our latest outgoing sequence number, we walk our list of
+ outstanding packets and determine if there is a match. If there
+ is, we continue. If not, then we ignore the response. In doing
+ this, I found a few completely useless variables that I have now
+ removed. (closes issue #11231) Reported by: flefoll Review:
+ https://reviewboard.asterisk.org/r/298
+
+2009-06-29 19:36 +0000 [r204170] Tilghman Lesher <tlesher at digium.com>
+
+ * funcs/func_odbc.c, funcs/func_strings.c: Revision 189537 was
+ supposed to make 1.4 more correct. Instead, it broke func_odbc.
+ Reverting. (closes issue #15317, issue #14614)
+
+2009-06-29 17:04 +0000 [r204067] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: segfault after SPINLOCK schedule delete
+ Using the SPINLOCK schedule delete macro can result in the
+ iax_pvt lock being given up. This makes it possible for the
+ iax_pvt to dissappear when we thought we held the mutex the
+ entire time. To resolve this, the iax_pvt's ref count is
+ incremented. (closes issue #15377) Reported by: aragon Patches:
+ iax_spin_issue_1.4.diff uploaded by dvossel (license 671) Tested
+ by: aragon, dvossel
+
+2009-06-29 15:04 +0000 [r204012] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_mixmonitor.c: Place unlock of mutex in an else block so
+ that it does not get unlocked twice. (closes issue #15400)
+ Reported by: aragon
+
+2009-06-27 00:55 +0000 [r203908] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: 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
+
+2009-06-26 22:09 +0000 [r203848] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Make sure to recreate the dahdi pseudo
+ channel after dahdi restart (closes issue #14477) Reported by:
+ timking
+
+2009-06-26 21:16 +0000 [r203785] Russell Bryant <russell at digium.com>
+
+ * main/file.c: 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
+
+2009-06-26 20:03 +0000 [r203719] David Brooks <dbrooks at digium.com>
+
+ * apps/app_voicemail.c: 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/
+
+2009-06-25 21:13 +0000 [r203380] Terry Wilson <twilson at digium.com>
+
+ * main/cli.c: I didn't see that Mark already fixed the underlying
+ issue! Yay for removing useless code.
+
+2009-06-25 21:02 +0000 [r203375] Russell Bryant <russell at digium.com>
+
+ * res/res_features.c: 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
+
+2009-06-25 20:09 +0000 [r203311] Terry Wilson <twilson at digium.com>
+
+ * main/cli.c: Don't try to free NULL
+
+2009-06-25 18:52 +0000 [r203230] Mark Michelson <mmichelson at digium.com>
+
+ * main/astmm.c: Prevent false positives when freeing a NULL pointer
+ with MALLOC_DEBUG enabled.
+
+2009-06-25 16:02 +0000 [r203115] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: 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)
+
+2009-06-24 21:01 +0000 [r203036] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: 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.
+
+2009-06-24 18:28 +0000 [r202966] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Use the handy UNLINK macro instead of
+ hand-coding the same thing in-line.
+
+2009-06-24 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.26-rc4
+
+2009-06-23 16:28 +0000 [r202671] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: 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/
+
+2009-06-23 15:22 +0000 [r202572-202601] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Fix more memory leaks that may result if rtp
+ is not successfully allocated.
+
+ * channels/chan_sip.c: Fix potential memory leak in chan_sip when
+ video rtp is not allocated properly.
+
+2009-06-22 20:08 +0000 [r202414-202496] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: Report CallerID change during a masquerade.
+ Reported by: markster
+
+ * channels/chan_sip.c: Make Polycom subscription type override
+ check more explicit.
+
+2009-06-22 14:44 +0000 [r202336-202342] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Remove an extra debug line left from
+ previous commit.
+
+ * channels/chan_sip.c: 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
+
+ * channels/chan_sip.c: 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
+
+2009-06-20 17:51 +0000 [r202153] Sean Bright <sean at malleable.com>
+
+ * channels/chan_sip.c: Since we don't have sip_pvt_lock() in 1.4,
+ we need to use ast_mutex_* directly. (closes issue #15366)
+ Reported by: loloski
+
+2009-06-19 21:21 +0000 [r202022] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Added deadlock protection to
+ try_suggested_sip_codec in chan_sip.c. Review:
+ https://reviewboard.asterisk.org/r/287/
+
+2009-06-19 20:22 +0000 [r201993] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: 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)
+
+2009-06-19 00:40 +0000 [r201828] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_features.c: 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.
+
+2009-06-18 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.26-rc3
+
+2009-06-18 15:24 +0000 [r201600] Russell Bryant <russell at digium.com>
+
+ * res/res_musiconhold.c: 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)
+
+2009-06-17 19:59 +0000 [r201450] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: 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.
+
+2009-06-17 19:28 +0000 [r201423] David Vossel <dvossel at digium.com>
+
+ * apps/app_mixmonitor.c: 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/
+
+2009-06-17 18:45 +0000 [r201380] David Brooks <dbrooks at digium.com>
+
+ * channels/chan_sip.c: 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
+
+2009-06-17 12:03 +0000 [r200991-201261] Kevin P. Fleming <kpfleming at digium.com>
+
+ * include/asterisk/linkedlists.h: 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)
+
+ * apps/app_chanspy.c, apps/app_mixmonitor.c, main/channel.c,
+ build_tools/cflags-devmode.xml, main/autoservice.c, main/frame.c,
+ apps/app_meetme.c, main/slinfactory.c,
+ include/asterisk/linkedlists.h, main/file.c,
+ include/asterisk/channel.h, include/asterisk/frame.h: 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/
+
+2009-06-16 13:25 +0000 [r200875] Eliel C. Sardanons <eliels at gmail.com>
+
+ * res/res_smdi.c: 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)'.
+
+2009-06-15 15:21 +0000 [r200513] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Add INFO to our allowed methods so that
+ endpoints know they may send it to us. AST-223
+
+2009-06-12 19:06 +0000 [r200360] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: 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
+
+2009-06-11 22:20 +0000 [r200185] Sean Bright <sean.bright at gmail.com>
+
+ * Makefile: Backport fix for parallel build warnings from trunk
+ r199781.
+
+2009-06-11 12:12 +0000 [r200037] Leif Madsen <lmadsen at digium.com>
+
+ * build_tools/make_version_h: Fix path for .flavor and .version.
+ (issue #14737) Reported by: davidw Patches: flavor.patch uploaded
+ by davidw (license 780) Tested by: davidw
+
+2009-06-10 16:08 +0000 [r199856] Sean Bright <sean.bright at gmail.com>
+
+ * include/asterisk/utils.h: __WORDSIZE is not available on all
+ platforms, so use sizeof(void *) instead.
+
+2009-06-09 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.26-rc2
+
+2009-06-08 19:28 +0000 [r199626-199628] Sean Bright <sean.bright at gmail.com>
+
+ * include/asterisk/utils.h: Fix a typo in the stack size
+ calculation just introduced.
+
+ * include/asterisk/utils.h: 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
+
+2009-06-05 21:19 +0000 [r199297] David Vossel <dvossel at digium.com>
+
+ * main/pbx.c: 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/
+
+2009-06-04 19:00 +0000 [r199138] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: Additional updates to AST-2009-001
+
+2009-06-04 14:14 +0000 [r198957-199022] Sean Bright <sean.bright at gmail.com>
+
+ * main/asterisk.c, main/loader.c, include/asterisk.h: 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/
+
+ * pbx/pbx_spool.c: 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)
+
+2009-06-03 15:49 +0000 [r198891] David Vossel <dvossel at digium.com>
+
+ * main/channel.c, res/res_features.c, include/asterisk/channel.h:
+ 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/
+
+2009-06-01 20:07 +0000 [r198665] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_musiconhold.c: 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
+
+2009-05-30 19:36 +0000 [r198370] Sean Bright <sean.bright at gmail.com>
+
+ * res/res_jabber.c: 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)
+
+2009-05-30 03:42 +0000 [r198311] Russell Bryant <russell at digium.com>
+
+ * res/res_smdi.c: Fix a crash that occurred when MWI SMDI messages
+ expired. (closes issue #14561) Reported by: cmoss28
+
+2009-05-30 02:46 +0000 [r198251] Sean Bright <sean.bright at gmail.com>
+
+ * apps/app_dial.c: 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
+
+2009-05-29 18:53 +0000 [r198068] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/cdr.c, main/channel.c, res/res_features.c,
+ include/asterisk/cdr.h: 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
+
+2009-05-29 18:14 +0000 [r197998] Sean Bright <sean.bright at gmail.com>
+
+ * Makefile: 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
+
+2009-05-28 23:57 +0000 [r197895] Leif Madsen <lmadsen at digium.com>
+
+ * apps/app_mixmonitor.c: 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)
+
+2009-05-28 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.26-rc1
+
+2009-05-28 15:51 +0000 [r197620] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: 'iax show peer blah' now outputs whether or
+ not peer 'blah' is in trunk mode or not.
+
+2009-05-28 15:27 +0000 [r197588] Mark Michelson <mmichelson at digium.com>
+
+ * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Allow
+ for media to arrive from an alternate source when responding to a
+ reinvite with 491. When we receive a SIP reinvite, it is possible
+ that we may not be able to process the reinvite immediately since
+ we have also sent a reinvite out ourselves. The problem is that
+ whoever sent us the reinvite may have also sent a reinvite out to
+ another party, and that reinvite may have succeeded. As a result,
+ even though we are not going to accept the reinvite we just
+ received, it is important for us to not have problems if we
+ suddenly start receiving RTP from a new source. The fix for this
+ is to grab the media source information from the SDP of the
+ reinvite that we receive. This information is passed to the RTP
+ layer so that it will know about the alternate source for media.
+ Review: https://reviewboard.asterisk.org/r/252
+
+2009-05-28 15:21 +0000 [r197562] Eliel C. Sardanons <eliels at gmail.com>
+
+ * channels/chan_sip.c: Use the address we already know when
+ reloading a peer with nat=yes. If we already have an address for
+ a peer, and we are reloading the sip configuration, try to use
+ that address to contact the peer, instead of getting it from the
+ Contact. (closes issue #15194) Reported by: ibc Patches:
+ sip.patch uploaded by eliel (license 64) Tested by: manwe
+
+2009-05-28 14:49 +0000 [r197537] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_chanspy.c, include/asterisk/audiohook.h,
+ main/audiohook.c: Add flags to chanspy audiohook so that audio
+ stays in sync. There are two flags being added to the chanspy
+ audiohook here. One is the pre-existing
+ AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that
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