less than the defined minimum digit length, and the other end only wants END
digits (SIP INFO, for example).
(closes issue #12778)
Reported by: tsearle
Patches:
12778.rev1.txt uploaded by russell (license 2)
Tested by: tsearle
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@121280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@121229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
most people, so pull in the changes that allow searching for SMDI messages
based on other components of the SMDI message. Also, update the SMDI
documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@120671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
cannot perform any translation on the extension number before searching for it
in the target context.
(closes issue #12473)
Reported by: chappell
Patches:
pbx_loopback.c.diff uploaded by chappell (license 8)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@120226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: yem
Tested by: yem
This change decreases the buffer size allocated on the stack substantially in config_text_file_load when LOW_MEMORY is turned on. This change combined with the fix from revision 117462 (making mkintf not copy the zt_chan_conf structure) was enough to prevent the crash.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@120173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
though they are allowed to connect.
(closes issue #12594)
Reported by: bkruse
Patches:
12594-managerusers-2.diff uploaded by qwell (license 4)
Tested by: bkruse
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@120061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
this for the changes I made to preserve ${EXTEN}
in switches, which affected several tests because
it adds extra priorities, and at least one needed to be updated
because of the removal of the empty extension warning
message.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@119966 65c4cc65-6c06-0410-ace0-fbb531ad65f3
which is a message from Philipp Kempgen, requesting that the WARNING
that an extension is empty be reduced to a NOTICE or less, as empty
extensions are syntactically possible, and no big deal.
With which I agree, and have removed that WARNING message entirely.
I think it is not necessary to see this message. It didn't
state that a NoOp() was inserted automatically on your behalf,
and really, as users, who cares? Why freak out dialplan writers
with unnecessary warnings? The details of the machinations a compiler goes
thru to produce working assembly code is of little interest
to most programmers-- we will follow the unix principal of
doing our work silently.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@119929 65c4cc65-6c06-0410-ace0-fbb531ad65f3
jabber.conf). The actual connection is made when a call comes in
Asterisk.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@119740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The rc/init.d scripts, and safe_asterisk work on normal sh now again.
Tested on:
OpenBSD 4.2 (me)
Debian etch (me)
Ubuntu Hardy (me and loloski)
FC9 (loloski)
(closes issue #12687)
Reported by: loloski
Patches:
20080529-12687-safe_asterisk-fixversion.diff.txt uploaded by mvanbaak (license 7)
Tested by: loloski, mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@119301 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@119238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
for a DTMF frame, or some other deferred frame type, to come in and get dropped.
(closes issue #12656)
(closes issue #12656)
Reported by: dimas
Patches:
v3-12656.patch uploaded by dimas (license 88)
-- with some modifications by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@119156 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Jeff Peeler
Christian Richter
Sean Bright
Joshua Colp
Russell Bryant
Mark Michelson
Philippe Sultan
Tilghman Lesher
Steve Murphy
Michiel van Baak