https://origsvn.digium.com/svn/asterisk/trunk
................
r204835 | rmudgett | 2009-07-02 17:01:28 -0500 (Thu, 02 Jul 2009) | 17 lines
Merged revisions 204834 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009) | 10 lines
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
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009) | 21 lines
Merged revisions 204681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines
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/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204754 65c4cc65-6c06-0410-ace0-fbb531ad65f3
dialog_unref() and dialog_ref() in 1.6.0 where only place holders
for reference counting once it was implemented. The functions
did nothing but return the pointer on ref and NULL on unref. These
calls have been removed to make way for a patch that actually does
dialog ref counting.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r204563 | tilghman | 2009-06-30 15:41:04 -0500 (Tue, 30 Jun 2009) | 13 lines
Merged revisions 204556 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009) | 6 lines
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)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r204301 | mmichelson | 2009-06-29 17:50:35 -0500 (Mon, 29 Jun 2009) | 15 lines
Merged revisions 204300 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun 2009) | 9 lines
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)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r204247 | mmichelson | 2009-06-29 16:48:54 -0500 (Mon, 29 Jun 2009) | 32 lines
Merged revisions 204243,204246 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun 2009) | 22 lines
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
........
r204246 | mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3 lines
Fix build oops.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
................
r204013 | mmichelson | 2009-06-29 10:04:39 -0500 (Mon, 29 Jun 2009) | 11 lines
Blocked revisions 204012 via svnmerge
........
r204012 | mmichelson | 2009-06-29 10:04:17 -0500 (Mon, 29 Jun 2009) | 6 lines
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/1.6.0@204015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r203909 | rmudgett | 2009-06-26 20:07:52 -0500 (Fri, 26 Jun 2009) | 23 lines
Merged revisions 203908 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009) | 16 lines
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/1.6.0@203910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r203802 | russell | 2009-06-26 16:21:48 -0500 (Fri, 26 Jun 2009) | 22 lines
Merged revisions 203785 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009) | 15 lines
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/1.6.0@203818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r203721 | dbrooks | 2009-06-26 15:13:51 -0500 (Fri, 26 Jun 2009) | 16 lines
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/1.6.0@203722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r203672 | jpeeler | 2009-06-26 14:03:25 -0500 (Fri, 26 Jun 2009) | 16 lines
Check if polarityonanswerdelay has elapsed before setting a channel as answered
after a polarity reversal.
Previously on a polarity switch event chan_dahdi would set the channel
immediately as answered. This would cause problems if a polarity reversal
occurred when the line was picked up as the dial would not have yet occurred.
Now if the polarity reversal occurs before delay has elapsed after coming off
hook or an answer, it is ignored. Also, some refactoring was done in
_handle_event.
(closes issue #13917)
Reported by: alecdavis
Patches:
chan_dahdi.bug13917.feb09.diff2.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r203376 | russell | 2009-06-25 16:04:55 -0500 (Thu, 25 Jun 2009) | 16 lines
Merged revisions 203375 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009) | 9 lines
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/1.6.0@203377 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r203304 | jpeeler | 2009-06-25 14:54:12 -0500 (Thu, 25 Jun 2009) | 6 lines
New signaling module to handle PRI/BRI operations in chan_dahdi
This merge splits the PRI/BRI signaling logic out of chan_dahdi.c into
sig_pri.c. Functionality in theory should not change (mostly). A few trivial
changes were made in sig_analog with verbose messages and commenting.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203305 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r203258 | qwell | 2009-06-25 14:22:46 -0500 (Thu, 25 Jun 2009) | 10 lines
Unmute when we get a dtmfup (we muted on dtmfdown) event.
This would occasionally cause one-way audio when using hardware DTMF detection.
(closes issue #14761)
Reported by: tzafrir
Patches:
v1-14761.patch uploaded by dimas (license 88)
Tested by: tzafrir, dimas
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r203116 | russell | 2009-06-25 11:04:10 -0500 (Thu, 25 Jun 2009) | 18 lines
Merged revisions 203115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) | 11 lines
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/1.6.0@203117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r203037 | rmudgett | 2009-06-24 16:08:55 -0500 (Wed, 24 Jun 2009) | 15 lines
Merged revisions 203036 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009) | 8 lines
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/1.6.0@203044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r202410 | dvossel | 2009-06-22 10:33:35 -0500 (Mon, 22 Jun 2009) | 5 lines
attempting to load running modules
Modules placed in the priority heap for loading were not properly removed from the linked list. This resulted in some modules attempting to load twice.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r202343 | mmichelson | 2009-06-22 09:58:24 -0500 (Mon, 22 Jun 2009) | 36 lines
Merged revisions 202341-202342 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun 2009) | 26 lines
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
........
r202342 | mmichelson | 2009-06-22 09:44:58 -0500 (Mon, 22 Jun 2009) | 3 lines
Remove an extra debug line left from previous commit.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r202337 | mmichelson | 2009-06-22 09:35:09 -0500 (Mon, 22 Jun 2009) | 31 lines
Merged revisions 202336 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun 2009) | 25 lines
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/1.6.0@202338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r201829 | tilghman | 2009-06-18 19:43:41 -0500 (Thu, 18 Jun 2009) | 13 lines
Merged revisions 201828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009) | 6 lines
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/1.6.0@201830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
