Age | Commit message (Collapse) | Author | Files | Lines |
|
Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122234 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122232 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121914 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r121861 | tilghman | 2008-06-11 13:18:16 -0500 (Wed, 11 Jun 2008) | 3 lines
Make calls to ast_assert() actually test something, so that the error message
printed is not nonsensical (reported by mvanbaak via #asterisk-bugs).
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121867 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
get a reset circuit message. Fixes crash bug
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121857 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r121751 | crichter | 2008-06-11 11:28:04 +0200 (Mi, 11 Jun 2008) | 1 line
fixed issue with previous commit, the find_free_channel test for channels which where inuse was broken.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121770 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
been.
(closes issue #12706)
Reported by: falves11
Patches:
chan_sip.c.diff uploaded by rjain (license 226)
Tested by: falves11
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121503 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r121495 | file | 2008-06-10 10:34:27 -0300 (Tue, 10 Jun 2008) | 4 lines
If we are destroying a dialog only set the MWI dialog pointer on the related peer to NULL if it is the dialog currently being destroyed.
(closes issue #12828)
Reported by: ramonpeek
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121496 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121407 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121367 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(Closes AST-38)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121334 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
(Note that this is being merged to trunk/1.6.0 because
it may affect non-callback agents with ackcall set)
........
r121229 | mmichelson | 2008-06-09 10:02:37 -0500 (Mon, 09 Jun 2008) | 16 lines
A unique situation of timeouts brought forth a failure situation for
autologoff in chan_agent. If using AgentCallbackLogin-style agents,
then if the timeout specified by the Dial() to reach the agent's phone
was shorter than the timeout specified in queues.conf, then autologoff
would only work if the caller hung up while the agent's phone was ringing.
This patch allows autologoff to work in this situation when the call in
queue transfers to the next available agent (as it would have if the timeout
in queues.conf were less than the timeout in the Dial()).
(closes issue #12754)
Reported by: Rodrigo
Patches:
12754.patch uploaded by putnopvut (license 60)
Tested by: Rodrigo
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121230 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121163 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r121078 | russell | 2008-06-07 09:10:56 -0500 (Sat, 07 Jun 2008) | 7 lines
Don't run LIST_HEAD_DESTROY on a STATIC list
(closes issue #12807)
Reported by: ys
Patches:
chan_agent_local.diff uploaded by ys (license 281)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121079 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #12562)
Reported by: michael-fig
Patches:
20080515__bug12562.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121042 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r120959 | jpeeler | 2008-06-06 13:29:14 -0500 (Fri, 06 Jun 2008) | 1 line
add another LOW_MEMORY define I forgot
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120960 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r120908 | jpeeler | 2008-06-06 13:05:15 -0500 (Fri, 06 Jun 2008) | 1 line
only define thread storage variable if necessary for LOW_MEMORY
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120909 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008) | 3 lines
This fixes a crash when LOW_MEMORY is turned on. Two allocations of the ast_rtp struct that were previously allocated on the stack have been modified to use thread local storage instead.
........
r120885 | jpeeler | 2008-06-06 11:39:20 -0500 (Fri, 06 Jun 2008) | 2 lines
Correction to commmit 120863, make sure proper destructor function is called as well define two thread storage local variables.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120906 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
to realtime less painful in the future.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120789 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
API call to access it, including maximums, minimums, standard deviatinos,
and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function
for any channel that uses RTP.
(closes issue #10590)
Reported by: gasparz
Patches:
chan_sip_c.diff uploaded by gasparz (license 219)
rtp_c.diff uploaded by gasparz (license 219)
rtp_h.diff uploaded by gasparz (license 219)
audioqos-trunk.diff uploaded by snuffy (license 35)
rtpqos-trunk-r119891.diff uploaded by sergee (license 138)
Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120635 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r120425 | tilghman | 2008-06-04 13:35:47 -0500 (Wed, 04 Jun 2008) | 6 lines
If we fail to setup the PRI request channel, don't continue, exit with an error.
(closes issue #11989)
Reported by: Corydon76
Patches:
20080213__zap_memleak.diff.txt uploaded by Corydon76 (license 14)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120426 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r120168 | russell | 2008-06-03 16:34:55 -0500 (Tue, 03 Jun 2008) | 4 lines
Fix another place where peer->callno could change at a very bad time, and also
fix a place where a peer was used after the reference was released.
(inspired by rev 120001)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120169 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r120001 | tilghman | 2008-06-03 11:10:53 -0500 (Tue, 03 Jun 2008) | 9 lines
Save the callno when we're poking, because our peer structure could change
during destruction (and thus we unlock the wrong callno, causing a
cascade failure).
(closes issue #12717)
Reported by: gewfie
Patches:
20080525__bug12717.diff.txt uploaded by Corydon76 (license 14)
Tested by: gewfie
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120012 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r119926 | file | 2008-06-03 11:46:24 -0300 (Tue, 03 Jun 2008) | 2 lines
Treat ECONNREFUSED as an error that will stop further retransmissions. (issue #AST-58, patch from Switchvox)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119927 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r119838 | russell | 2008-06-02 15:08:04 -0500 (Mon, 02 Jun 2008) | 7 lines
Revert a change made for issue #12479. This change caused a regression such that
a dial string such as (IAX2/foo) did not automatically fall back to dialing the 's'
extension anymore.
(closes issue #12770)
Reported by: dagmoller
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119839 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
jabber.conf). The actual connection is made when a call comes in
Asterisk.
Apply this fix to Jingle too.
Fix the ast_aji_get_client function that was not able to retrieve an
XMPP client from its JID.
(closes issue #12085)
Reported by: junky
Tested by: phsultan
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119741 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r119687 | russell | 2008-06-02 07:30:17 -0500 (Mon, 02 Jun 2008) | 3 lines
Even of the first PING or LAGRQ doesn't get sent because it comes up too soon,
make sure to reschedule so it gets sent later.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119688 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r119636 | crichter | 2008-06-02 11:29:21 +0200 (Mo, 02 Jun 2008) | 1 line
fixed compile issue when dev-mode is enabled
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119637 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r119585 | crichter | 2008-06-02 10:35:28 +0200 (Mo, 02 Jun 2008) | 1 line
Added counter for unhandled_bmsg Print, this prevents the logs to be flooded to fast and save CPU in this error scenario. Added 'last_used' element to bc structure, when a bchannel changes from used to free this exact time will be marked in last_used. When a new channel is requested the find_free_chan function will check if the new empty channel was used within the last second, if yes it will search for the next channel, if no it will return this channel. This simple mechanism has prooven to prevent race conditions where the NT and TE tried to allocate the exact same channel at the same time (RELEASE cause: 44).
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119586 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r119533 | russell | 2008-06-01 20:06:09 -0500 (Sun, 01 Jun 2008) | 2 lines
Change a debug message to an actual debug message
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119534 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r119238 | russell | 2008-05-30 07:55:36 -0500 (Fri, 30 May 2008) | 15 lines
Merged revisions 119237 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30 May 2008) | 7 lines
- Instead of only enforcing destination call number checking on an ACK, check
all full frames except for PING and LAGRQ, which may be sent by older versions
too quickly to contain the destination call number.
(As suggested by Tim Panton on the asterisk-dev list)
- Merge changes from team/russell/iax2-frame-race, which prevents PING and LAGRQ
from being sent before the destination call number is known.
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119239 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r119071 | tilghman | 2008-05-29 15:24:11 -0500 (Thu, 29 May 2008) | 7 lines
Call waiting tone occurs too often, because it's getting serviced by both
subchannels.
(closes issue #11354)
Reported by: cahen
Patches:
20080512__bug11354.diff.txt uploaded by Corydon76 (license 14)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119072 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r119009 | russell | 2008-05-29 13:49:12 -0500 (Thu, 29 May 2008) | 16 lines
Merged revisions 119008 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29 May 2008) | 7 lines
Merge changes from team/russell/iax2-another-fix-to-the-fix
As described in the following post to the asterisk-dev mailing list, only
enforce destination call numbers when processing an ACK.
http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html
(closes issue #12631)
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119010 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r118953 | tilghman | 2008-05-29 12:20:16 -0500 (Thu, 29 May 2008) | 3 lines
Add some debugging code that ensures that when we do deadlock avoidance, we
don't lose the information about how a lock was originally acquired.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118955 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
A lot of whitespace issues have been resolved in this commit
Also some doc updates, but that's only 6 lines
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118824 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118790 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118750 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
channel is unlocked in some cases, and because it can cause seemingly
random failures could be related to some bugs in the tracker...
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118702 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines
Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow.
(closes issue #10417)
Reported by: cstadlmann
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118647 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Jingle is completely standardized, we can set those namespaces to their
final values.
Added two attributes to the jingle_pvt struct to store the content
name attributes. Reported by Robert McQueen on Telepathy's framework
mailing list :
http://lists.freedesktop.org/archives/telepathy/2008-May/001971.html
Keeping working on our Jingle stack!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118644 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118614 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
the receiving of a packet that we've kept in memory just incase the
packet needs to be retransmitted.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118562 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r118558 | file | 2008-05-27 16:32:38 -0300 (Tue, 27 May 2008) | 4 lines
Fix an issue where codec preferences were not set on dialogs that were not authenticated via a user or peer and allow framing to work without rtpmap in the SDP.
(closes issue #12501)
Reported by: slimey
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118560 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r118251 | tilghman | 2008-05-25 11:02:04 -0500 (Sun, 25 May 2008) | 12 lines
Realtime flag affects construction in multiple ways, so consulting whether
rtcachefriends was set was done too soon (needed to be done inside build_peer,
not just as a flag to build_peer).
Also, fullcontact needed to be reconstructed, because realtime separates the
embedded ';' into multiple fields.
(closes issue #12722)
Reported by: barthpbx
Patches:
20080525__bug12722.diff.txt uploaded by Corydon76 (license 14)
Tested by: barthpbx
(Much of the discussion happened on #asterisk-dev for diagnosing this issue)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118252 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r118163 | jpeeler | 2008-05-23 16:21:35 -0500 (Fri, 23 May 2008) | 1 line
Fix a few things I missed to ensure zt_chan_conf structure is not modified in mkintf
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118164 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
them to the parser ;
- report Gtalk error messages from a buddy to the console.
This patch makes Asterisk "Google Jingle" (chan_gtalk) implementation
work with Empathy. Note that this is only true for audio streams, not
video.
Thank you to PH for his great help!
(closes issue #12647)
Reported by: PH
Patches:
trunk-12647-1.diff uploaded by phsultan (license 73)
Tested by: phsultan, PH
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118020 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117988 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117983 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117950 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117870 f38db490-d61c-443f-a65b-d21fe96a405b
|