Solaris 10 nawk doesn't lthe empty pattern ike '//' for 'always'.
Just remove that. No pattern at all always matches.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226453 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) | 9 lines
Update documentation in sip.conf.sample.
Update the documentation in sip.conf.sample in order to make it more clear
that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It
is only used to stop Asterisk from generating a reINVITE, but does not stop
it from accepting them if necessary.
(closes issue #15644)
Reported by: lmadsen
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009) | 7 lines
Manager output is not always NULL-terminated, so force a NULL at the end of the filestream.
(closes issue #15495)
Reported by: pdf
Patches:
20090916__issue15495.diff.txt uploaded by tilghman (license 14)
Tested by: pdf
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226159 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change adds a configuration option to SIP peers, unsolicited_mailbox, which
configures a virtual mailbox to use for received new/old MWI information. This
virtual mailbox can then be used by any device supporting MWI.
(closes issue #13028)
Reported by: AsteriskRocks
Patches:
bug_13028_chan_sip_external_mwi_20090707.patch uploaded by cmaj (license 830)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Set OSARCH to linux-gnu even if host_os is linux-gnueabi
* When checking if we are Linux, check OSARCH rather than host_os
The newer ARM ABI ("EABI") shows the OS name 'linux-gnueabi' rather than
'linux-gnu' . This patch sets OSARCH to be 'linux-gnu' even in such a case.
OSARCH is tested for the value of 'linux-gnu' in one or two places in the
tree. This patch also fixes the check libcap to check for $OSARCH rather
than $host_os .
See also: http://wiki.debian.org/ArmEabiPort
Merged revisions 225957 via svnmerge from
http://svn.digium.com/svn/asterisk/branches/1.4
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The code committed in revision 225405 was broken; instead of removing the unreference code,
the logic used to decide when to do it should have been reversed. This patch corrects the
situation, and makes reference counting work properly again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225955 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ACL check in check_peer_ok was missing and has now been restored. The
missing check allowed for calls to be made on prohibited networks where an ACL
was defined in sip.conf and the allowguest option was set to off. See the AST
security advisory below for more information.
Merge code associated with AST-2009-007.
(closes issue #16091)
Reported by: thom4fun
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prefix PRI trace messages with the span number. This makes the trace
readable even when you have a multi-port device.
(closes issue #15054)
Reported by: tzafrir
Patches:
dahdi_pri_debug_spannum.diff uploaded by tzafrir (license 46)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch changes chan_sip to use the new astobj2 OBJ_MULTIPLE iterator support
to make pedantic mode dialog searching in find_call() not require a linear search
of all dialogs in the list of dialogs. This patch does *not* change the dialog
matching logic (more on that later), just improves the searching performance.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added handling of received HOLD/RETRIEVE messages and the optional ability
to transfer a held call on disconnect similar to an analog phone.
* Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
Will reroute/deflect an outgoing call when receive the message.
Can use the DAHDISendCallreroutingFacility to send the message for the
supported switches.
* Added ability to send/receive keypad digits in the SETUP message.
Send keypad digits in SETUP message: Dial(DAHDI/g1[/K<keypad_digits>][/extension])
Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
* Added support for BRI PTMP NT mode.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct 2009) | 10 lines
Don't force menuselect.makeopts to be rebuilt on every build.
For some reason the menuselect.makeopts file was listed as PHONY in the Makefile,
resulting in 'make' needing to rebuild it for every build. This then resulted in
the embedded module rules being rebuilt on every build, which can be slow and is
unnecessary.
This patch fixes the problem by properly allowing 'make' to know when the
menuselect.makeopts file needs to be rebuilt (defining the proper dependencies).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Update the README documentation to correctly describe which CLI command you should
use when attempting to get help from the CLI.
(closes issue #16064)
Reported by: thedavidfactor
Patches:
readme.patch uploaded by thedavidfactor (license 903)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225515 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009) | 11 lines
Clean valgrind output by suppressing false errors.
Update valgrind.txt documentation and add valgrind.supp file in order to
allow those who are creating valgrind output to have less false errors in
the logfile.
(closes issue #16007)
Reported by: atis
Patches:
valgrind.txt.diff uploaded by atis (license 242)
asterisk2.supp uploaded by atis (license 242)
Tested by: atis, amorsen
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added documentation on how to create a local git repository from
SVN. This documentation was added via doxygen.
(closes issue #15814)
Reported by: tzafrir
Patches:
git-asterisk-howto uploaded by tzafrir (license 46)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225483 65c4cc65-6c06-0410-ace0-fbb531ad65f3
What this patch fixes
1.Moves sip TCP/TLS connection setup into the TCP helper thread:
Connection setup takes awhile and before this it was being
done while holding the monitor lock.
2.Moves TCP/TLS writing to the TCP helper thread: Through the
use of a packet queue and an alert pipe, the TCP helper thread
can now be woken up to write data as well as read data.
3.Locking error: sip_xmit returned an XMIT_ERROR without giving
up the tcptls_session lock. This lock has been completely removed
from sip_xmit and placed in the new sip_tcptls_write() function.
4.Memory leak: When creating a tcptls_client the tls_cfg was alloced
but never freed unless the tcptls_session failed to start. Now the
session_args for a sip client are an ao2 object which frees the
tls_cfg on destruction.
5.Pointer to stack variable: During sip_prepare_socket the creation
of a client's ast_tcptls_session_args was done on the stack and
stored as a pointer in the newly created tcptls_session. Depending
on the events that followed, there was a slight possibility that
pointer could have been accessed after the stack returned. Given
the new changes, it is always accessed after the stack returns
which is why I found it.
Notable code changes
1.I broke tcptls.c's ast_tcptls_client_start() function into two
functions. One for creating and allocating the new tcptls_session,
and a separate one for starting and handling the new connection.
This allowed me to create the tcptls_session, launch the helper
thread, and then establish the connection within the helper thread.
2.Writes to a tcptls_session are now done within the helper thread.
This is done by using an alert pipe to wake up the thread if new
data needs to be sent. The thread's sip_threadinfo object contains
the alert pipe as well as the packet queue.
3.Since the threadinfo object contains the alert pipe, it must now be
accessed outside of the helper thread for every write (queuing of a
packet). For easy lookup, I moved the threadinfo objects from a
linked list to an ao2_container.
(closes issue #13136)
Reported by: pabelanger
Tested by: dvossel, whys
(closes issue #15894)
Reported by: dvossel
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/380/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225445 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When an object is being unlinked from its container *and* being returned to
the caller, we do not want to decrement the reference count after unlinking
it from the container, as the reference that the container held is what we
are returning to the caller... and if it was the only remaining reference to
the object, that could result in the object being destroyed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Telecom Specs in NZ suggests that SUB ADDRESS is always on, so doing
"desk to desk" between offices each with an asterisk box over the ISDN
should then be possible, without a whole load of DDI numbers required.
(closes issue #15604)
Reported by: alecdavis
Patches:
asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license 585)
Some minor modificatons were made.
Tested by: alecdavis, rmudgett
Review: https://reviewboard.asterisk.org/r/405/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) | 13 lines
IAX2: VNAK loop caused by signaling frames with no destination call number
It is possible for the PBX thread to queue up signaling frames before
a destination call number is received. This can result in signaling
frames being sent out with no destination call number. Since recent
versions of Asterisk require accurate destination callnumbers for all
Full Frames, this can cause a VNAK loop to occur. To resolve this
no signaling frames are sent until a destination callnumber is received,
and destination call numbers are now only required for iax_pvt matching
when the frame is an ACK.
Review: https://reviewboard.asterisk.org/r/413/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the
case where multiple results need to be returned; OBJ_NODATA mode
already was supported). In addition, it converts ast_channel_iterators
(only the targeted versions, not the ones that iterate over all
channels) to use this method.
During this work, I removed the 'ao2_flags' arguments to the
ast_channel_iterator constructor functions; there were no uses of that
argument yet, there is only one possible flag to pass, and it made the
iterators less 'opaque'. If at some point in the future someone really
needs an ast_channel_iterator that does not lock the container, we can
provide constructor(s) for that purpose.
Review: https://reviewboard.asterisk.org/r/379/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r225103 | tilghman | 2009-10-21 10:45:54 -0500 (Wed, 21 Oct 2009) | 2 lines
Suffix is not needed for a match
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225104 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is the second commit for this and documents the text stream using the configured
IP address and fixes a bug in the original patch where the UDPTL stream would also
use the different IP address.
(closes issue #14729)
Reported by: _brent_
Patches:
media_address.patch uploaded by brent (license 388)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines
IAX/SIP shrinkcallerid option
The shrinking of caller id removes '(', ' ', ')', non-trailing '.',
and '-' from the string. This means values such as 555.5555 and
test-test result in 555555 and testtest. There are instances,
such as Skype integration, where a specific value is passed via
caller id that must be preserved unmodified. This patch makes
the shrinking of caller id optional in chan_sip and chan_iax in
order to support such cases. By default this option is on to
preserve previous expected behavior.
(closes issue #15940)
Reported by: dimas
Patches:
v2-15940.patch uploaded by dimas (license 88)
15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/408/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225033 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines
Pay attention to the return value of the manipulate function.
While this looks like an optimization, it prevents a crash from occurring
when used with certain audiohook callbacks (diagnosed with SVN trunk,
backported to 1.4 to keep the source consistent across versions).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Joshua Colp
Olle Johansson
Tzafrir Cohen
Leif Madsen
Tilghman Lesher
Terry Wilson
Kevin P. Fleming
Jeff Peeler
Richard Mudgett
Sean Bright
David Vossel
Russell Bryant