When this patch was originally submitted, the code allowed for the token to be
set via a channel variable. I decided that a cleaner approach would be to
integrate it into the CHANNEL() function. Unfortunately, that is not a suitable
approach. It's not possible to get the value set on the channel soon enough
using that method. So, go back to the simple channel variable method.
(closes issue #16711)
Reported by: homesick
Patches:
iax-svn.diff uploaded by homesick (license 91)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1. URI Encoding
This patch changes ast_uri_encode()'s behavior when doreserved is enabled.
Previously when doreserved was enabled only a small set of reserved
characters were encoded. This set was comprised primarily of the reserved
characters defined in RFC3261 section 25.1, but contained other characters as
well. Rather than only escaping the reserved set, doreserved now escapes
all characters not within the unreserved set as defined by RFC 3261 and
RFC 2396. Also, the 'doreserved' variable has been renamed to 'do_special_char'
in attempts to avoid confusion.
When doreserve is not enabled, the previous logic of only encoding the
characters <= 0X1F and > 0X7f remains, except for the '%' character, which
must always be encoded as it signifies a HEX escaped character during the decode
process.
2. URI Decoding: Break up URI before decode.
In chan_sip.c ast_uri_decode is called on the entire URI instead of it's
individual parts after it is parsed. This is not good as ast_uri_decode
can introduce special characters back into the URI which can mess up parsing.
This patch resolves this by not decoding a URI until parsing is completely
done. There are many instances where we check to see if pedantic checking
is enabled before we decode a URI. In these cases a new macro,
SIP_PEDANTIC_DECODE, is used on the individual parsed segments of the URI
rather than constantly putting if (pedantic) { decode() } checks everywhere
in the code. In the areas where ast_uri_decode is not dependent upon
pedantic checking this macro is not used, but decoding is still moved to
each individual part of the URI. The only behavior that should change from
this patch is the time at which decoding occurs.
Since I had to look over every place URI parsing occurs to create this
patch, I found several places where we use duplicate code for parsing.
To consolidate the code, those areas have updated to use the parse_uri()
function where possible.
3. SIP display-name decoding according to RFC3261 section 25.
To properly decode the display-name portion of a FROM header, chan_sip's
get_calleridname() function required a complete re-write. More information
about this change can be found in the comments at the beginning of this function.
4. Unit Tests.
Unit tests for ast_uri_encode, ast_uri_decode, and get_calleridname() have been
written. This involved the addition of the test_utils.c file for testing the
utils api.
(closes issue #16299)
Reported by: wdoekes
Patches:
astsvn-16299-get_calleridname.diff uploaded by wdoekes (license 717)
get_calleridname_rewrite.diff uploaded by dvossel (license 671)
Tested by: wdoekes, dvossel, Nick_Lewis
Review: https://reviewboard.asterisk.org/r/469/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Allows CDR variables added in cdr.c:set_one_cid to become visable during the call,
by executing ast_cdr_update() early in __ast_pbx run.
Reverts sig_pri changes in trunk that are specific to isdn technology only.
(closes issue #16638)
Reported by: alecdavis
Patches:
cdr_update.diff3.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r241227 | jpeeler | 2010-01-19 11:22:18 -0600 (Tue, 19 Jan 2010) | 13 lines
Fix deadlock in agent_read by removing call to agent_logoff.
One must always lock the agents list lock before the agent private. agent_read
locks the private immediately, so locking the agents list lock is not an
option (which is what agent_logoff requires). Because agent_read already
has access to the agent private all that is necessary is to do the required
hanging up that agent_logoff performed.
(closes issue #16321)
Reported by: valon24
Patches:
bug16321.patch uploaded by jpeeler (license 325)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Allows CDR variables added in cdr.c:set_one_cid to become visable during the call.
(issue #16638)
Reported by: alecdavis
Patches:
cdr_update.diff2.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241187 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If the limit was set past MAX_INT upon answering, the call was immediately
hung up due to overflow from the return of ast_tvdiff_ms (in ast_check_hangup).
The time calculation functions ast_tvdiff_sec and ast_tvdiff_ms have been
changed to return an int64_t to prevent overflow. Also the reporter suggested
adding a message indicating the reason for the call hanging up. Given that the
new limit is so much higher, the message (which would only really be useful in
the overflow scenario) has been made a debug message only.
(closes issue #16006)
Reported by: viraptor
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Allows CDR variables added in cdr.c:set_one_cid to become visable during the call.
(closes issue #16638)
Reported by: alecdavis
Patches:
cdr_update.diff.txt uploaded by alecdavis (license 585)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #15819)
Reported by: klaus3000
Patches:
asterisk-sip-show-channelstats-trunk.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000, oej
This patch is for trunk only and will be blocked in 1.6.2
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The code that handled setting 'm=text' in the sdp was not executing
in the correct order. The check to see if text was needed came after
the check to add 'm=text' to the sdp, this resulted in 'm=text' always
being set to 0 because it looked like text was never required.
(closes issue #16457)
Reported by: peterj
Patches:
textportinsdp.diff uploaded by peterj (license 951)
issue16457.diff uploaded by dvossel (license 671)
Tested by: peterj
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes a crash on Solaris.
(closes issue #16572)
Reported by: crjw
Patches:
frame_changes.patch uploaded by crjw (license 963)
Plus several others found and fixed by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239074 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010) | 10 lines
fixes crash in "scheduled_destroy" in chan_iax
A signed short was used to represent a callnumber. This is makes
it possible to attempt to access the iaxs array with a negative
index.
(closes issue #16565)
Reported by: jensvb
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Modified handle_verbose to be LOW_MEMORY aware, removed old RTP related code
in chan_sip.
(closes issue #16381)
Reported by: michael_iedema
Patches:
ast_complete_source_filename.patch uploaded by michael iedema (license 942)
modified by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009) | 11 lines
fixes issue with p->method incorrectly set to ACK
It is possible for a second ACK to come in for a retransmitted message.
If an ack does not match an unacked message in our queue, restore the previous
p->method as this ACK is completely ignored.
(closes issue #16295)
Reported by: omolenkamp
Patches:
issue16295_v2.diff uploaded by dvossel (license 671)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A registration regression caused by a code tweak in (issue #14331)
and a bug fix in (issue #15539) caused some sip registration
config entries to be constructed incorrectly. Origially
issue #14331 contained the code tweak as well as a bug fix, but since
the issue was reported as a tweak the bug fix portion was moved into
issue #15539. Both the tweak and the bug fix contained minor incorrect
logic that resulted in some SIP registrations to fail.
(issue #14331)
(issue #15539)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r234492 | oej | 2009-12-14 11:16:00 +0100 (Mån, 14 Dec 2009) | 8 lines
Stop sending 183's after call hangup.
There where still cases where the 183 keep-alive mechanism would not stop
sending 183's even though the Asterisk server had sent a final reply to
the invite.
EDVX-28
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009) | 10 lines
Do not modify the gain settings on data calls.
(The digital flag actually represents a data call.)
(closes issue #15972)
Reported by: udosw
Patches:
transcap_digital_fix.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In the process of swapping ULAW to a place in the extended codec space, we
found several unhandled cases, where a 32-bit integer was still being used to
handle a codec field. Most of these have been fixed with this commit, although
there is at least one case (codec_dahdi) which depends upon outside headers to
be altered before a conversion can be made.
(Fixes AST-278, SWP-459)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1) Allow users of UDPTL stack to associate a character-string tag with a UDPTL
session, so that log/error/debug messages generated by the UDPTL stack can
be 'connected' to the endpoint that caused them to be generated.
2) Improve comments (and process) of calculating the far end's maximum IFP size
when redundancy mode is in use for error correction.
3) When an IFP larger than the calculated 'far max IFP' size is presented for
writing, truncate it rather than putting in the buffer and allowing the buffer
to overflow; this will cause the ends to retrain to a lower bit rate that
produces IFPs of an appropriate size if possible, and if not possible, the
FAX transfer will fail completely. In these cases, it is due to the one endpoint
supplying a T38FaxMaxDatagram value that is improperly calculated and is
too low to be of use; we have configuration options available to override
this behavior.
4) Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no longer
needed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Changed areas in sig_pri to set the digital flag using a callback that will
also set the corresponding flag in chan_dahdi. Modified dahdi_request slightly
so that if a bearer is marked as digital, that information is available when
creating the new channel.
(closes issue #16151)
Reported by: alecdavis
Patch based on bug_16151.diff.txt uploaded by alecdavis (license 585)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231058 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Internally the way T.38 is negotiated has changed and the option no longer
reflects a behavior that is valid. It will now look for a CNG tone on
received calls and if present send the call to the 'fax' extension. It is
then up to the application or channel to request the switch over to T.38.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r230246 | kpfleming | 2009-11-15 11:19:06 -0600 (Sun, 15 Nov 2009) | 6 lines
Correct mistaken option name in error message.
The configuration option for allowing hosts to make non-token-based calls
is 'calltokenoptional', not 'calltokenignore'. (reported on asterisk-users)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230247 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r230038 | file | 2009-11-13 13:44:07 -0600 (Fri, 13 Nov 2009) | 9 lines
Fix a crash caused by two threads thinking they should both free the
chan_local private structure when only one should.
(closes issue #15314)
Reported by: sroberts
Patches:
Issue15314_Move_Nulling_owner.patch uploaded by davidw (license 780)
Tested by: davidw, lottc
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230039 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* chan_console accessed pvts after deallocation.
* cdr_mysql stored a pointer that was freed by realloc()
* The module loader did not check usecount on shutdown, which led to chan_iax2
reading a timer that was already unloaded.
* The event subsystem sometimes creates an event with no IEs. Due to a corner
condition, the code would read beyond the memory boundary.
* res_pktccops did not correctly check whether its monitor thread was started.
(closes issue #16062)
Reported by: alexanderheinz
Patches:
20091109__issue16062.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Dial(DAHDI/g1[/extension[/options]])
Current options:
K(<keypad_digits>)
R Reverse charging indication (Collect calls)
The earlier Dial(DAHDI/g1[/K<keypad_digits>][/extension] format was
variable and did not allow for the easy addition of more options.
The earlier 'C' prefix character for reverse charge indiation would
conflict with the a-d DTMF digits if ISDN uses them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228691 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009) | 9 lines
chan_misdn Asterisk 1.4.27-rc2 crash
Crash related to chan_misdn connection. Patch submitted by gknispel_proformatique, tested
by francesco_r. "I have many crash since i have upgraded to Asterisk 1.4.27-rc2. Attached
a full bt." This patch zeros out an ast_frame.
(closes issue #16041)
Reported by: francesco_r
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With the new code, media level proprieties should no longer be confused with session level proprieties. This change also reorganizes some of the SDP parsing code which should make it easier to manage in the future.
(closes issue #14994)
Reported by: frawd
Tested by: frawd, mnicholson, file
Review: https://reviewboard.asterisk.org/r/414/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
SIP channel names were supposed to be unique by way of a name suffix derived from the
pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but
not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with
a simple incremented unsigned int.
(closes issue #15152)
Reported by: palbrecht
Review: https://reviewboard.asterisk.org/r/420/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since ISDN works like SIP and not analog ports in regard to devices, the
device state based on the ISDN channel number could not work. This has
not been an issue until the advent of PTMP NT mode. Previously, ISDN
lines were used as trunks and did not have to keep track of specific
devices.
As an interim solution until device states are properly implemented, the
channel name is being changed to the following format to use the generic
device state support:
DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
Dialplan hints would thus be:
exten => xxx,hint,DAHDI/i2/5551212
This will work with the following restrictions:
* The number of devices/phones cannot exceed the number of B channels.
(i.e., BRI has 2)
* Each device/phone can only have one number. No shared MSN's.
* The phones/devices probably should not use subaddressing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226882 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Cleanup some flags on DAHDI PRI channel hangup. (sig_pri split)
* Make sure the outgoing flag is cleared if a new channel fails to get
created for outgoing calls.
* Remove some unused flags since sig_pri was split.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226648 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
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
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
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
Russell Bryant
David Vossel
Olle Johansson
Tilghman Lesher
Kevin P. Fleming
Alec L Davis
Jeff Peeler
Sean Bright
Richard Mudgett
Joshua Colp
Matthew Nicholson
Michiel van Baak
Mark Michelson
Jason Parker
Terry Wilson
David Brooks
Tzafrir Cohen