If a Via header contained an IPv6 address, we would not properly parse
the port. We would instead get the information after the first colon in
the address.
(closes issue #17614)
Reported by: oej
Patches:
diff uploaded by sperreault (license 252)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This should fix all the CDR tests that were not passing. When they would
originate a call, all fields in the INVITE that contained the source port would
have the port set to 0. Most troubling of these was the Contact header. Tests
are passing locally now and should also pass on the bamboo build agents.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Part of the change with the IPv6 changes is to treat a host:port as
a single 'domain' entity. This test was not updated to have the correct
expectation after calling parse_uri().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274984 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.
Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.
(closes issue #17565)
Reported by: russell
Patches:
asteriskv6-test-report.pdf uploaded by russell (license 2)
Review: https://reviewboard.asterisk.org/r/743
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010) | 9 lines
Add option to not do a call forward on 482 Loop Detected
Asterisk has always set up a forwarded call when receiving a 482 Loop Detected.
This prevents handling the call failure by just continuing on in the dialplan.
Since this would be a change in behavior, the new option to disable this
behavior is forwardloopdetected which defaults to 'yes'.
Review: https://reviewboard.asterisk.org/r/764/
........
(no option for trunk, just changing the behavior)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A failure when calling the get_destination can mean multiple things. If
the extension is not found, a 404 error is appropriate, but if the URI
scheme is incorrect, a 404 is not approperiate. This patch adds the
get_destination_result enum to differentiate between these and other failure
types. The only logical difference in this patch is that we now send a "416
Unsupported URI scheme" response instead of a "404" when the scheme is not
recognized. This indicates to the initiator of the INVITE to retry the request
with a correct URI.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010) | 10 lines
Allow the "useragent" value to be restored into memory from the realtime backend.
This value is purely informational. It does not alter configuration at all.
(closes issue #16029)
Reported by: Guggemand
Patches:
realtime-useragent.patch uploaded by Guggemand (license 897)
Tested by: Guggemand
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RFC 2361 section 24.4.1 send a 400 Bad Request if the request
can not be understood due to malformed syntax. Currently we
simply ignore a packet with a missing callid, to, from, or
via header. Instead of ignoring we now send the 400 Bad request.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272981 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RFC 3261 section 8.2.2.3 states that if any unsupported options
are found in the Require header field, a "420 (Bad Extension)"
response should be sent with an Unsupported header field containing
only the unsupported options.
This is not currently being done correctly. Right now, if Asterisk
detects any unsupported sip options in a Require header the entire
list of options are returned in the Unsupported header even if some
of those options are in fact supported. This patch fixes that by
building an unsupported options character buffer when parsing the
options that can be sent with the 420 response. A unit test verifying
this functionality has been created. Some code refactoring was required.
Review: https://reviewboard.asterisk.org/r/680/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
I am doing work in this function. I noticed a large number of
coding guidline fixes that needed to be made. Rather than have
those changes distract from my functional changes I decided
to separate these into a separate patch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RFC3261 states that Timer A should start at 500ms (T1) by default.
In chan_sip this value initially started at 1000ms and I changed
it to 500ms recently. After doing that I noticed in my packet
captures that it still occasionally retransmitted starting at
1000ms instead of 500ms like I told it to. This occurs because
the scheduler runs in the do_monitor thread. If a new retransmission
is added while the do_monitor thread is sleeping then it may not
detect that retransmission for nearly 1000ms. To fix this I just
poke the do_monitor thread to wake up when a new packet is sent
reliably requiring retransmits. The thread then detects the new
scheduler entry and adjusts its sleep time to account for it.
Review: https://reviewboard.asterisk.org/r/747
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The external test suite stops Asterisk using the "core stop gracefully" command.
The logs from the tests show that there are a number of problems with Asterisk
trying to cleanly shut down. This patch addresses the following type of error
that comes from chan_iax2:
[Jun 22 16:58:11] ERROR[29884]: lock.c:129 __ast_pthread_mutex_destroy:
chan_iax2.c line 11371 (iax2_process_thread_cleanup):
Error destroying mutex &thread->lock: Device or resource busy
For an example in the context of a build, see:
http://bamboo.asterisk.org/browse/AST-TRUNK-739/log
The primary purpose of this patch is to change the thread pool shutdown
procedure to be more explicit to ensure that the thread exits from a point
where it is not holding a lock. While testing that, I encountered various
crashes due to the order of operations in unload_module() being problematic.
I reordered some things there, as well.
Review: https://reviewboard.asterisk.org/r/736/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Testing proved that if Asterisk sent a connected line reinvite, and
the endpoint to which the reinvite were being sent sent a reinvite, Asterisk
would not properly respond with a 491 response.
The reason is that on connected line reinvites, we set the dialog's invitestate
to INV_CALLING to prevent Asterisk from sending a rapid flurry of connected line
reinvites. For other reinvites we do not do this. Because of the current invitestate,
when Asterisk received the reinvite, we interpreted this as a spiraled INVITE, and thus
did not behave properly.
The fix for this is to not enter the loop detection or spiral logic in handle_request_invite
if the channel state is currently up. This way, no mid-call reinvites will be misinterpreted,
no matter what the nature of the reinvite may have been.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This small changes prevents destroy_all_channels() from accessing a lock on an
unused dahdi_pri struct, resolving a ton of ERRORs that get spewed out when
shutting Asterisk down gracefully.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RFC 3261 section 9 states that a CANCEL has no effect on a
request to a UAS that has already given a final response. This
patch checks to make sure there is a pending invite before
allowing a CANCEL request to be processed, otherwise it responds
to the CANCEL with a "481 Call/Transaction Does Not Exist".
Review: https://reviewboard.asterisk.org/r/697/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun 2010) | 8 lines
Modify chan_sip's packet generation api to automatically calculate the Content-Length. This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated. This change was made to ensure that the Content-Length is always correct.
(closes issue #17326)
Reported by: kenner
Tested by: mnicholson, kenner
Review: https://reviewboard.asterisk.org/r/693/
........
This change also adds an ast_str_copy_string() function (similar to ast_copy_string), that copies one ast_str into another, properly handling embedded nulls.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271690 65c4cc65-6c06-0410-ace0-fbb531ad65f3
According to RFC 3261 section 17.2.2, which describes non-INVITE server
transaction, when a dialog enters the Completed state it must destroy
the dialog after Timer J (T1*64) fires. For a BYE transaction Asterisk
terminates the dialog immediately during sip_hangup() when it should be
waiting T1*64 ms. This results in some odd behavior. For instance if
Asterisk receives a BYE and transmits a 200ok in response, if the endpoint
never receives the 200ok it will retransmit the BYE to which Asterisk
responds with a "481 Call leg/transaction does not exist" because the
dialog is already gone.
To resolve this I made a function called sip_scheddestroy_final(). This
differs slightly from sip_schedestroy() in that it enables a flag that
will prevent the destruction from ever being rescheduled or canceled
afterwards. It also prevents the pvt's needdestroy flag from being set
which triggers the destruction of the dialog within the do_monitor thread().
By using this function we are guaranteed destruction will not occur
until the scheduled time. This allows Asterisk to respond to any possible
retransmits for a dialog after we process the initial BYE request for T1*64 ms.
Other changes: I removed two instances where sip_cancel_destroy is used
right before calling sip_scheddestroy. sip_scheddestroy always calls
sip_cancel_destroy before scheduling the new destruction so it is completely
unnecessary.
Review: https://reviewboard.asterisk.org/r/694/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r270866 | dvossel | 2010-06-16 12:35:29 -0500 (Wed, 16 Jun 2010) | 22 lines
fixes chan_iax2 race condition
There is code in chan_iax2.c that attempts to guarantee that only a single
active thread will handle a call number at a time. This code works once
the thread is added to an active_list of threads, but we are not currently
guaranteed that a newly activated thread will enter the active_list immediately
because it is left up to the thread to add itself after frames have been
queued to it. This means that if two frames come in for the same call number
at the same time, it is possible for them to grab two separate threads because
the first thread did not add itself to the active_list fast enough. This
causes some pretty complex problems.
This patch resolves this race condition by immediately adding an activated
thread to the active_list within the network thread and only depending on
the thread to remove itself once it is done processing the frames queued to
it. By doing this we are guaranteed that if another frame for the same call
number comes in at the same time, that this thread will immediately be found
in the active_list of threads.
Review: https://reviewboard.asterisk.org/r/720/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270867 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Clearing the callwaitcas flag in analog_call was causing the incoming D digit
to be ignored which triggers sending the caller ID.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
chan_sip's "contactdeny" feature screens the "to be registered contact".
In case of nat=yes it should not use the address information from the
Contact header (which is not used at all for routing), but the source
IP address of the request.
Thus, if nat=yes and a client sends a request from a denied IP address
(e.g. by spoofing the src-IP address) it can bypass the screening.
This commit makes contactdeny apply to the src ip when nat=yes instead.
(closes issue #17276)
Reported by: klaus3000
Patches:
patch-asterisk-trunk-contactdeny.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270658 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The "channel" option would chop the channel name at the last '-', which made
it useless for something like a channel transfer from the dialplan. The
"fullchannel" option will return the channel name as-is.
ABE-2218
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Calling dahdi_indicate() within dahdi_fixup() while the owner pointers are
in a potentially inconsistent state is a potentially bad thing in
principle.
However, calling dahdi_indicate() when the channel private lock is already
held can cause a deadlock if the PRI lock is needed because
dahdi_indicate() will also get the channel private lock. The pri_grab()
function assumes that the channel private lock is held once to avoid
deadlock.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* chan_dahdi supports dialing configuring and dialing by device file name.
DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
* A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
False by default. If set, chan_dahdi will ignore failed 'channel' entries.
Handy for the above name-based syntax as it does not depend on
initialization order.
* have my_pri_make_cc_dialstring() only manupulate dial-strings of group
(gGrR) dialing, which make it lsightly more complicated.
https://reviewboard.asterisk.org/r/535/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After 5 years in mantis and over a year on reviewboard, SRTP support is finally
being comitted. This includes generic CHANNEL dialplan functions that work for
getting the status of whether a call has secure media or signaling as defined
by the underlying channel technology and for setting whether or not a new
channel being bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples.
Original patch by mikma, updated for trunk and revised by me.
(closes issue #5413)
Reported by: mikma
Tested by: twilson, notthematrix, hemanshurpatel
Review: https://reviewboard.asterisk.org/r/191/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Extract the SS7 specific code out of chan_dahdi like what was done to
ISDN/PRI and analog signaling. The new SS7 structures were modeled on
sig_pri.
The changes to sig_pri are an enhancement and a bug fix made possible
because SS7 was extracted.
1) The sig_pri TRANSFERCAPABILITY channel variable should have been set
unconditionally in sig_pri_new_ast_channel().
2) SS7/PRI transfer capability interaction in dahdi_new() fixed because of
SS7 extraction.
3) Module ref count error in dahdi_new() if startpbx failed to start the
PBX for some reason.
Review: https://reviewboard.asterisk.org/r/661/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Changes.
1. RFC 3261 states in section 17.1.2.2 and 17.1.1.2 that retransmission
timers should initially be set to timer T1. T1 by default is 500ms.
Asterisk was starting the retransmission timers at T1*2 which shouldn't
cause any problems, but is not RFC compliant.
2. RFC 3261 states in section 17.1.2.2 that for a non-INVITE client transaction,
if the retransmit timer fires while in the proceeding state that
the request must be retransmitted. Asterisk currently ack's
requests for both INVITE and non-INVITE transactions when a
1XX response is received, this patch changes this for non-INVITE requests.
3. The 'registerattempts' option in sip.conf is supposed to set
how many registry attempts will be made before giving up. When
this option is set to 1, I would expect only one registry attempt
to be made before stopping because of a failure, but instead two are
made. In my opinion this is not expected behavior. This option does
not indicate that these are re-attempts. The logic behind this option
has been changed to only attempt registers the exact number of times
this option is set to. If this option is 0, it still continues to
re-attempt the registration forever.
Review: https://reviewboard.asterisk.org/r/687/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The problem would manifest itself if your dialplan matching could accept
more digits to match than were actually dialed. The time out waiting for
overlap digits disconnected the call instead of matching any accumulated
digits to the dialplan.
Accidental conversion of a break out of loop as a break out of switch.
(closes issue #17401)
Reported by: avalentin
Patches:
issue17401_digit_timeout.patch uploaded by rmudgett (license 664)
Tested by: avalentin, rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267928 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After some debugging, the random chan_h323 build failures appear to be due
to complications introduced by some chan_h323 specific build stuff getting
triggered during a clean. Simplify this by moving the h323 clean commands
down into channels/makefile.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add the ability to announce a call to an endpoint when there are no B
channels available. A call waiting call is a SETUP message with no B
channel selected.
Relevant specification: EN 300 056, EN 300 057, EN 300 058
For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
"no_media_path" option.
* Returns "0" if there is a B channel associated with the call.
* Returns "1" if no B channel is associated with the call. The call is
either on hold or is a call waiting call.
If you are going to allow incoming call waiting calls then you need to use
CHANNEL(no_media_path) do determine if you must drop a call to accept the
new call.
Review: https://reviewboard.asterisk.org/r/568/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Two struct sockaddr_ins are created when applying directmedia
host access rules. The addresses of these are passed to the RTP
engine to be filled in. However, the RTP engine inspects the fields
of the structs before actually taking action. This inspection caused
valgrind to be a bit unhappy.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk Generic AOC Representation
- Generic AOC encode/decode routines.
(Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame)
- AST_CONTROL_AOC frame type to represent generic encoded AOC data
- Manager events for AOC-S, AOC-D, and AOC-E messages
Asterisk App Support
- app_dial AOC-S pass-through support on call setup
- app_queue AOC-S pass-through support on call setup
AOC Unit Tests
- AOC Unit Tests for encode/decode routines
- AOC Unit Test for manager event representation.
SIP AOC Support
- Pass-through of generic AOC-D and AOC-E messages to snom phones via the
snom AOC specification.
- Creation of chan_sip page3 flags for the addition of the new
'snom_aoc_enabled' sip.conf option.
IAX AOC Support
- Natively supports AOC pass-through through the use of the new
AST_CONTROL_AOC frame type
DAHDI AOC Support
- ETSI PRI full AOC Pass-through support
- 'aoc_enable' chan_dahdi.conf option for independently enabling
pass-through of AOC-S, AOC-D, AOC-E.
- 'aoce_delayhangup' option for retrieving AOC-E on disconnect.
- DAHDI A() dial string option for requesting AOC services.
example usage:
;requests AOC-S, AOC-D, and AOC-E on call setup
exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e))
Review: https://reviewboard.asterisk.org/r/552/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added ability to send and receive ETSI Explicit Call Transfer (ECT)
messages to eliminate tromboned calls.
Note: Asterisk already supported initiating the transfer of calls to
eliminate tromboned calls to libpri so there was nothing to do for the
asterisk portion.
Review: https://reviewboard.asterisk.org/r/520/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch breaks up every part of the sip registry string during
config parsing and removes all parsing from transmit_register().
Thanks to Nick_Lewis for contributing this patch!
(closes issue #14331)
Reported by: Nick_Lewis
Patches:
chan_sip.c-domparse.patch uploaded by Nick Lewis (license 657)
chan_sip.c.patch uploaded by Nick Lewis (license 657)
chan_sip.c.domainparse3.patch uploaded by Nick Lewis (license 657)
chan_sip.c-domparse4.patch uploaded by Nick Lewis (license 657)
chan_sip.c-domparse5.patch uploaded by Nick Lewis (license 657)
nicklewispatch.diff uploaded by dvossel (license 671)
Tested by: Nick_Lewis, dvossel
Review: https://reviewboard.asterisk.org/r/628/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The logic used in transmit_register to get the outboundproxy for a peer
was flawed since this value would be overridden shortly afterwards when
create_addr was called.
In addition, this also fixes some logic used when parsing users.conf so
that the peer name is placed in the internally-generated register string
so that an outboundproxy set in the Asterisk GUI will be used for outbound
REGISTERs.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The decoding of the replace_id did not need to be broken
up in this instance. This was brought to my attention
again because it caused a segfault when the from or to
tags were not present in the "Replaces" header.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265366 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The same code was repeated in lots of different places, so I made a utility
fuction for it. This should make the merge in the v6-new branch easier.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
directmediapermit/directmediadeny support to restrict which peers can do
directmedia based on ip address. In some networks not all phones are fully
routed, i.e. not all phones can ping each other. This patch adds a way to
restrict directmedia for certain peers between certain networks.
(closes issue #16645)
Reported by: raarts
Patches:
directmediapermit.patch uploaded by raarts (license 937)
Tested by: raarts
Review: https://reviewboard.asterisk.org/r/467/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From reviewboard:
The problem here is a bit complex, so try to bear with me...
It was noticed by a Digium customer that generic PLC (as configured in
codecs.conf) did not appear to actually be having any sort of benefit when
packet loss was introduced on an RTP stream. I reproduced this issue myself
by streaming a file across an RTP stream and dropping approx. 5% of the
RTP packets. I saw no real difference between when PLC was enabled or disabled
when using wireshark to analyze the RTP streams.
After analyzing what was going on, it became clear that one of the problems
faced was that when running my tests, the translation paths were being set
up in such a way that PLC could not possibly work as expected. To illustrate,
if packets are lost on channel A's read stream, then we expect that PLC will
be applied to channel B's write stream. The problem is that generic PLC can
only be done when there is a translation path that moves from some codec to
SLINEAR. When I would run my tests, I found that every single time, read
and write translation paths would be set up on channel A instead of channel
B. There appeared to be no real way to predict which channel the translation
paths would be set up on.
This is where Kevin swooped in to let me know about the transcode_via_sln
option in asterisk.conf. It is supposed to work by placing a read translation
path on both channels from the channel's rawreadformat to SLINEAR. It also
will place a write translation path on both channels from SLINEAR to the
channel's rawwriteformat. Using this option allows one to predictably set up
translation paths on all channels. There are two problems with this, though.
First and foremost, the transcode_via_sln option did not appear to be working
properly when I was placing a SIP call between two endpoints which did not
share any common formats. Second, even if this option were to work, for PLC
to be applied, there had to be a write translation path that would go from
some format to SLINEAR. It would not work properly if the starting format
of translation was SLINEAR.
The one-line change presented in this review request in chan_sip.c fixed the
first issue for me. The problem was that in sip_request_call, the
jointcapability of the outbound channel was being set to the format passed to
sip_request_call. This is nativeformats of the inbound channel. Because of this,
when ast_channel_make_compatible was called by app_dial, both channels already
had compatibly read and write formats. Thus, no translation path was set up at
the time. My change is to set the jointcapability of the sip_pvt created during
sip_request_call to the intersection of the inbound channel's nativeformats and
the configured peer capability that we determined during the earlier call to
create_addr. Doing this got the translation paths set up as expected when using
transcode_via_sln.
The changes presented in channel.c fixed the second issue for me. First and
foremost, when Asterisk is started, we'll read codecs.conf to see the value of
the genericplc option. If this option is set, and ast_write is called for a
frame with no data, then we will attempt to fill in the missing samples for
the frame. The implementation uses a channel datastore for maintaining the
PLC state and for creating a buffer to store PLC samples in. Even when we
receive a frame with data, we'll call plc_rx so that the PLC state will have
knowledge of the previous voice frame, which it can use as a basis for when
it comes time to actually do a PLC fill-in.
So, reviewers, now I ask for your help. First off, there's the one line change
in chan_sip that I have put in. Is it right? By my logic it seems correct, but
I'm sure someone can tell me why it is not going to work. This is probably the
change I'm least concerned about, though. What concerns me much more is the
set of changes in channel.c. First off, am I even doing it right? When I run
tests, I can clearly see that when PLC is activated, I see a significant increase
in RTP traffic where I would expect it to be. However, in my humble opinion, the
audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to
me than when no PLC is used at all. I need someone to review the logic I have used
to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic
is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm
sure someone can point out somewhere where I've done something incorrectly.
As I was writing this review request up, I decided to give the code a test run under
valgrind, and I find that for some reason, calls to plc_rx are causing some invalid
reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around
a bit to see why that is the case. If it's obvious to someone reviewing, speak up!
Finally, I have one other proposal that is not reflected in my code review. Since
without transcode_via_sln set, one cannot predict or control where a translation
path will be up, it seems to me that the current practice of using PLC only when
transcoding to SLINEAR is not useful. I recommend that once it has been determined
that the method used in this code review is correct and works as expected, then
the code in translate.c that invokes PLC should be removed.
Review: https://reviewboard.asterisk.org/r/622/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
During deadlock avoidance the sip dialog pvt is locked and
unlocked. When this occurs we have no guarantee the pvt's owner
is still valid. We were trying to access the pvt's owner after
this without checking to see if it still existed first.
(closes issue #17271)
Reported by: under
Patches:
check_rtp_timeout.diff uploaded by under (license 914)
Tested by: dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Very unfortunate things happen if we add an iax_frame
to the frame queue and let go of the lock before scheduling
the frame's transmit... There is a race condition that
exists where the frame can be removed from the frame_queue
and freed before the transmit is scheduled if we do not
hold on to that lock. This results in a freed frame
being scheduled for transmit later.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Must do a deep copy of the cc_params in duplicate_pseudo(). Otherwise,
when the duplicate pseudo channel is destroyed, it frees the original
pseudo channel cc_params. The original pseudo channel is then left with a
dangling pointer for when the next duplicated pseudo channel is created.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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r262660 | rmudgett | 2010-05-12 11:46:47 -0500 (Wed, 12 May 2010) | 4 lines
Forgot some conditionals around the callrerouting facility help text.
JIRA ABE-2223
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r262657 | rmudgett | 2010-05-12 11:26:49 -0500 (Wed, 12 May 2010) | 22 lines
Add mISDN Call rerouting facility for point-to-point ISDN lines (exchange line)
In the case of ISDN point-to-multipoint (multidevice) you can use the
mISDN "facility calldeflect" application for call diversions from external
(PSTN) to external (PSTN). In that case this is the only way to get rid
of the two call legs to the PBX and let the calling number at the C party
become the number of the A party. In the case of ISDN point-to-point
(exchange line) the call deflection facility may not be used. Instead a
call rerouting facility has to be used.
This patch for chan_misdn.c is an extension to realize this service
(facility rerouting application). It can accept either spelling:
"callrerouting" or "callrerouteing".
The patch is tested towards Deutsche Telekom and requires a modified
version of mISDN from Digium, Inc.
Patches:
misdn_rerouteing_corrected.patch (Slightly modified.)
JIRA ABE-2223
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262661 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Clean up chan_sip.c to use new AST_CLI functions
(closes issue #17287)
Reported by: pabelanger
Patches:
issue17287.patch uploaded by pabelanger (license 224)
Tested by: russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Revision -r1489 of the libpri 1.4 branch corrected a deviation from Q.931
Section 5.3.2. However, this resulted in an unexpected behaviour change
to the upper layer (Asterisk).
This change uses pri_hangup_fix_enable() to follow Q.931 Section 5.3.2
call hangup better if the version of libpri supports it.
(issue #17104)
Reported by: shawkris
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There is a race condition between console_hangup()
and start_stream(). It is possible for console_hangup()
to be called and then the stream thread to begin after the hangup.
To avoid this a check in start_stream() to make sure the pvt-owner
still exists while the pvt lock is held is made. If the owner
is gone that means the channel hung up and start_stream should
be aborted.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The pri_dchannel thread currently violates locking order by locking the private
and then attempting to queue a frame, which needs to lock the channel. Queueing
a frame is unneccesary though and is actually a regression since sig_pri.
All the places that currently use ast_softhangup_nolock now will just set the
softhangup value directly as before.
(closes issue #17216)
Reported by: lmsteffan
Patches:
bug17216.patch uploaded by jpeeler (license 325)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made more places use pri_queue_control() instead of pri_queue_frame()
and a local frame variable.
* Made pri_queue_frame() use sig_pri_lock_owner(). pri_queue_frame() no
longer releases the libpri access lock unless it is required.
* Made the pri_queue_frame() and pri_queue_control() parameter list
similar to sig_pri_lock_owner().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The example given within the related issue showed 120 lines, which was mostly
a result of the body being XML.
(closes issue #17179)
Reported by: khw
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Refer-To header field containing the Replaces header in the URI
was not being decoded properly. This caused invalid parsing between
the caller id field and the domain resulting in a failed transfer.
(closes issue #17284)
Reported by: dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a recognized media type was present, but the media type was not
enabled for the channel, then a warning would be emitted. For instance,
attempting to get CHANNEL(rtpsource,video) on a call with no video would
cause a warning message to appear.
With this change, the warning will only appear if the stream argument
is not recognized as being a media type that can be specified.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The CLI "dahdi show channel" command was not correctly reporting the
InAlarm status.
The inalarm flag is now consistently passed between chan_dahdi and
submodules.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010) | 11 lines
Ensure channel state is not incorrectly set in the case of a very early answer.
The needringing bit was being read in dahdi_read after answering thereby
setting the state to ringing from up. This clears needringing upon answering
so that is no longer possible.
(closes issue #17067)
Reported by: tzafrir
Patches:
needringing.diff uploaded by tzafrir (license 46)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Created
SIG_PRI_MAX_CHANNELS, SIG_PRI_NUM_DCHANS
SIG_SS7_MAX_CHANNELS, SIG_SS7_NUM_DCHANS
SIG_MFCR2_MAX_CHANNELS
Also fixed the declaration of pollers[] in mfcr2_monitor(). It was
dimensioned to the number of bytes in struct dahdi_mfcr2.pvts[] and not to
the same dimension of the struct dahdi_mfcr2.pvts[].
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010) | 26 lines
DTMF CallerID detection problems.
The code handling DTMF CallerID drops digits on long CallerID numbers and
may timeout waiting for the first ring with shorter numbers.
The DTMF emulation mode was not turned off when processing DTMF CallerID.
When the emulation code gets behind in processing the DTMF digits it can
skip a digit.
For shorter numbers, the timeout may have been too short. I increased it
from 2 seconds to 4 seconds. Four seconds is a typical time between rings
for many countries.
(closes issue #16460)
Reported by: sum
Patches:
issue16460.patch uploaded by rmudgett (license 664)
issue16460_v1.6.2.patch uploaded by rmudgett (license 664)
Tested by: sum, rmudgett
Review: https://reviewboard.asterisk.org/r/634/
JIRA SWP-562
JIRA AST-334
JIRA SWP-901
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010) | 33 lines
resolves deadlocks in chan_local
Issue_1.
In the local_hangup() 3 locks must be held at the same time... pvt, pvt->chan,
and pvt->owner. Proper deadlock avoidance is done when the channel to hangup
is the outbound chan_local channel, but when it is not the outbound channel we
have an issue... We attempt to do deadlock avoidance only on the tech pvt, when
both the tech pvt and the pvt->owner are locked coming into that loop. By
never giving up the pvt->owner channel deadlock avoidance is not entirely possible.
This patch resolves that by doing deadlock avoidance on both the pvt->owner and the pvt
when trying to get the pvt->chan lock.
Issue_2.
ast_prod() is used in ast_activate_generator() to queue a frame on the channel
and make the channel's read function get called. This function is used in
ast_activate_generator() while the channel is locked, which mean's the channel
will have a lock both from the generator code and the frame_queue code by the
time it gets to chan_local.c's local_queue_frame code... local_queue_frame
contains some of the same crazy deadlock avoidance that local_hangup requires,
and this recursive lock prevents that deadlock avoidance from happening correctly.
This patch removes ast_prod() from the channel lock so only one lock is held during
the local_queue_frame function.
(closes issue #17185)
Reported by: schmoozecom
Patches:
issue_17185_v1.diff uploaded by dvossel (license 671)
issue_17185_v2.diff uploaded by dvossel (license 671)
Tested by: schmoozecom, GameGamer43
Review: https://reviewboard.asterisk.org/r/631/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010) | 11 lines
DAHDI "WARNING" message is confusing and vague
"WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed failed: Success"
Changed the warning to "Failed to decode CallerID on channel 'name'". The
message before it is likely more specific about why the CallerID decode
failed.
SWP-501
AST-283
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) | 14 lines
hidecalleridname parameter in chan_dahdi.conf
Issue #7321 implements a new chan_dahdi configuration option. However, a
change mentioned in the issue was never implemented. This is the change
that will allow the feature to work.
I added a note to chan_dahdi.conf.sample about the feature.
(closes issue #17143)
Reported by: djensen99
Patches:
diff.txt uploaded by djensen99 (license NA) (One line change)
Tested by: djensen99
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #16988)
Reported by: frawd
Patches:
chan_sip_sdp_verbose_fix.diff uploaded by frawd (license 610)
Tested by: russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is a clear mistake in logic. Future discussions
about how to avoid having to handle uri's like this
should take place in the future, but this fix needs
to go in for now.
(closes issue #15847)
Reported by: ebroad
Patches:
doublesip.patch uploaded by ebroad (license 878)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258305 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Revision 1072 of libpri added SERVICE message support for the 'national'
switchtype. The attached patch enables the use of 'pri service' CLI commands
on dahdi channels that are configured for the 'national' switchtype.
(closes issue #17142)
Reported by: dhubbard
Patches:
dw-ni2.patch uploaded by dhubbard (license 733)
Tested by: elguero, dhubbard
Review: https://reviewboard.asterisk.org/r/612/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010) | 13 lines
Don't recreate peer, when responding to a repeated deregistration attempt.
When a reply to a deregistration is lost in transmit, the client retries the
deregistration. Previously, this would cause a realtime/autocreate peer to be
loaded back into memory, after it had already been correctly purged. Instead,
we just want to resend the reply without loading the peer.
(closes issue #16908)
Reported by: kkm
Patches:
20100412__issue16908.diff.txt uploaded by tilghman (license 14)
Tested by: kkm
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
To guarantee the channel is valid when calling setvar on the MASTER_CHANNEL
dialplan function, a channel reference must be taken before unlocking. Thanks
to russell for pointing out the error.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Call completion code tries to grab the call completion parameters
from the requesting channel during local_request. When originating
a call to a local channel, however, this channel is NULL. This
was causing an issue for me when trying to run a test script.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256646 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From the original issue report opened by Nick Lewis:
Many sip headers in many sip methods contain the ABNF structure
name-andor-addr = name-addr / addr-spec
Examples include the to-header, from-header, contact-header, replyto-header
At the moment chan_sip.c makes various different attempts to parse this name-andor-addr structure for each header type and for each sip method with sometimes limited degrees of success.
I recommend that this name-andor-addr structure be parsed by a dedicated function and that it be used irrespective of the specific method or header that contains the name-andor-addr structure
Nick has also included unit tests for verifying these routines as well, so...heck yeah.
(closes issue #16708)
Reported by: Nick_Lewis
Patches:
reqresp_parser-nameandoraddr2.patch uploaded by Nick Lewis (license 657
Review: https://reviewboard.asterisk.org/r/549
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From Reviewboard:
CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date
overview of the architecture can be found in the file doc/CCSS_architecture.pdf
in the CCSS branch. Off the top of my head, the big differences between what is
implemented and what is in the document are as follows:
1. We did not end up modifying the Hangup application at all.
2. The document states that a single call completion monitor may be used across
multiple calls to the same device. This proved to not be such a good idea
when implementing protocol-specific monitors, and so we ended up using one
monitor per-device per-call.
3. There are some configuration options which were conceived after the document
was written. These are documented in the ccss.conf.sample that is on this
review request.
For some basic understanding of terminology used throughout this code, see the
ccss.tex document that is on this review.
This implements CCBS and CCNR in several flavors.
First up is a "generic" implementation, which can work over any channel technology
provided that the channel technology can accurately report device state. Call
completion is requested using the dialplan application CallCompletionRequest and can
be canceled using CallCompletionCancel. Device state subscriptions are used in order
to monitor the state of called parties.
Next, there is a SIP-specific implementation of call completion. This method uses the
methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion
using SIP signaling. There are a few things to note here:
* The agent/monitor terminology used throughout Asterisk sometimes is the reverse of
what is defined in the referenced draft.
* Implementation of the draft required support for SIP PUBLISH. I attempted to write
this in a generic-enough fashion such that if someone were to want to write PUBLISH
support for other event packages, such as dialog-state or presence, most of the effort
would be in writing callbacks specific to the event package.
* A subportion of supporting PUBLISH reception was that we had to implement a PIDF
parser. The PIDF support added is a bit minimal. I first wrote a validation
routine to ensure that the PIDF document is formatted properly. The rest of the
PIDF reading is done in-line in the call-completion-specific PUBLISH-handling
code. In other words, while there is PIDF support here, it is not in any state
where it could easily be applied to other event packages as is.
Finally, there are a variety of ISDN-related call completion protocols supported. These
were written by Richard Mudgett, and as such I can't really say much about their
implementation. There are notes in the CHANGES file that indicate the ISDN protocols
over which call completion is supported.
Review: https://reviewboard.asterisk.org/r/523
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From Review Board:
There are two interrelated changes here.
First, there is the introduction of func_srv. This adds two new read-only
dialplan functions, SRVQUERY and SRVRESULT. They work very similarly to the
ENUMQUERY and ENUMRESULT functions, except that this allows one to query SRV
records instead. In order to facilitate this work, I added a couple of new API
calls to srv.h. ast_srv_get_record_count tells the number of records returned
by an SRV lookup. This number is calculated at the time of the SRV lookup.
ast_srv_get_nth_record allows one to get a numbered SRV record.
Second, there is the modification to chan_sip that allows one to specify a
hostname or IP address (along with a port) to send an outgoing INVITE to when
dialing a SIP peer. This goes hand-in-hand with func_srv. You can query SRV
records and then use the host and port from the results to dial via a specific
host instead of what is configured in sip.conf.
Review: https://reviewboard.asterisk.org/r/608
SWP-1200
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02 Apr 2010) | 9 lines
Resolve a deadlock that occurs due to a pointless call to ast_bridged_channel()
(closes issue #16840)
Reported by: bzing2
Patches:
patch.txt uploaded by bzing2 (license 902)
issue_16840.rev1.diff uploaded by russell (license 2)
Tested by: bzing2, russell
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Bulk lot of generally trivial changes for cleaning up the transmit stuff. Line state request has been modified for line only responses.
(closes issue #16994)
Reported by: wedhorn
Patches:
skinny-clean07.diff uploaded by wedhorn (license 30)
Tested by: wedhorn
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There was a bug where we split the URI on the @ sign and then attempted
to compare to "anonymous@anonymous.invalid" afterwards. This comparison
could never evaluate true. So now we keep a copy of the URI prior to the
split so that the comparison is valid.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds rtpsource options analogous to the rtpdest
functions that already exist. In addition, this fixes
potential crashes which could result due to trying to
read values from nonexistent RTP streams.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Mark Michelson
Paul Belanger
Tilghman Lesher
Richard Mudgett
Eliel C. Sardanons
David Vossel
Terry Wilson
Tzafrir Cohen
Russell Bryant
Tim Ringenbach
Matthew Nicholson
Jeff Peeler
Jason Parker
Leif Madsen
Dwayne M. Hubbard
Michiel van Baak