Mantis-3709 (Commit 68ff3c3, Asterisk 1.2) added support for the video format
H.263+. For this, the RTP payload ID 103 got assigned statically. Commit f1aadc8
assigned another payload ID 98 for this format in Asterisk 1.6.
Change-Id: I90e35b158487f8f1f8187da6241b54cd3b74e667
The script remains compatible with Python 2.7 but now also works with
Python 3.3 and newer; to ease the migration from chan_sip to chan_pjsip.
ASTERISK-27811
Change-Id: I59cc6b52a1a89777eebcf25b3023bdf93babf835
Due to a fixed size buffer the digest authentication could be
incorrectly calculated if a large URI was provided, causing
authentication failure. The buffer is now dynamically allocated to allow
any size URI within the normal limits of the HTTP request size.
ASTERISK-27841
Change-Id: I660609db13b8f9e5f9567f339dd804f4985d41b3
Use AST_PBX_MAX_STACK to escape if we recurse 128 times. This will
prevent crash if dialplan contains an include loop. Log an error when
this occurs, at most one message per call to Macro() so we avoid logger
spam.
ASTERISK-26570 #close
Change-Id: I6c71b76998c31434391b150de055ae9a531e31da
Previously, only an IP address would be accepted for the capture_address config
setting in hep.conf. This change allows capture_address to be a resolvable
hostname or an IP address.
ASTERISK-27796 #close
Reported-By: Sebastian Gutierrez
Change-Id: I33e1a37a8b86e20505dadeda760b861a9ef51f6f
The stream topology has no lock of its own resulting in
another lock protecting it in some way (for example the
channel lock). If multiple channels are being juggled at
the same time this can be problematic. This change makes
the topology a reference counted object instead which
guarantees it will remain valid even without the channel
lock being held.
Change-Id: I4f4d3dd856a033ed55fe218c3a4fab364afedb03
Analog phones dial overlap dialing and it is chan_dahdi's job to read the
numbers. It has three timeout constants that this commit converts to
channel-level configuration options:
* firstdigit_timeout: Default time (ms) to detect first digit
* interdigit_timeout: Default time (ms) to detect following digits
* matchdigit_timeout: Default time (ms) to wait in case of ambiguous
match. This happens when the dialed digits match a number in the current
context but are also the prefix of another number.
Change-Id: Ib728fa900a4f6ae56d1ed810aba61b6593fb7213
The "ari set debug" code for incoming requests incorrectly assumed
that all requests would contain a body. If one did not exist the
request would be incorrectly rejected. The response that was sent
was also incomplete as an incorrect function was used to construct
the response.
The code has now been changed to no longer require a request to have
a body and the response updated to use the correct function.
ASTERISK-27801
Change-Id: I4eef036ad54550a4368118cc348765ecac25e0f8
* Increase maximum number of ciphers from 100 to 256 (or whatever
PJ_SSL_SOCK_MAX_CIPHERS is #define'd to)
* Simplify logic in cipher_name_to_id()
* Make signed/unsigned comparison consistent
Re: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=897412
Reported by: Ondřej Holas
Change-Id: Iea620f03915a1b873e79743154255c3148a514e7
When the local SSRC changes we need to update the SRTP information
so that the proper key is used. This is commonly done as a result
of bridging two channels together. Previously we only updated
the SRTP information if media had already flowed, but in practice
the channel driver may have already performed SRTP negotiation and
set up the previous SSRC. We now always do it on a local SSRC
change.
ASTERISK-27795
ASTERISK-27800
Change-Id: Ia7c8e74c28841388b5244ac0b8fd6c1dc6ee4c10
How it works today:
media_cache tries to parse out the extension of the media file to be played
from the URI provided to Asterisk while caching the file.
What's expected:
Better will be to have Asterisk get extension from other ways too. One of the
common ways is to get the type of content from the CONTENT-TYPE header in the
HTTP response for fetching the media file using the URI provided.
Steps to Reproduce:
Provide a URL of the form: http://host/media/1234 to Asterisk for media
playback. It fails to play and logs show the following error line:
[Sep 15 15:48:05] WARNING [29148] [C-00000092] file.c:
File http://host/media/1234 does not exist in any format
Scenario this issue is blocking:
In the case where the media files are stored in some cloud object store,
following can block the media being played via Asterisk:
Cloud storage generally needs authenticated access to the storage. The way
to do that is by using signed URIs. With the signed URIs there's no way to
preserve the name of the file.
In most cases Cloud storage returns a key to access the object and preserving
file name is also not a thing there
ASTERISK-27286
Reporter: Gaurav Khurana
Change-Id: I1b14692a49b2c1ac67688f58757184122e92ba89
The lua_error_function assumed that lua's debug table and traceback function
are always accessible, which is not the case. This fixes the error message
'Error in the lua error handler' triggred by switch exec() function.
If this happens lua's error message is shown without traceback.
Change-Id: I34ba0a098f1ae06a3af7b4d1b098bd43f42f96c8
If during translation a codec could not handle a given frame the translation
core would return NULL, thus not passing along the "missing" frame. Due to this
there was no frame to apply generic plc to, thus rendering it useless.
This patch makes it so the translation core produces an interpolated slin frame
in the cases where an attempt was made to translate to slin, but failed. This
interpolated frame is then passed along and can be used by the generic plc
algorithms to fill in the frame.
ASTERISK-27814 #close
Change-Id: I133d084da87adef913bf2ecc9c9240e3eaf4f40a
When an externally initiated renegotiation occurred it was
possible for video streams to be incorrectly remapped,
resulting in no video flowing to some receivers.
This change ensures that only the video source sets up
mappings and also that removed streams do not have mappings
set up.
Change-Id: Iab05f2254df3606670774844bb0935f833d3a9b0
Call ooManualProgress/Ringback outside of ast_debug function
when ooh323 debugging is on
ASTERISK-27812 #close
ASTERISK-26893 #close
Reported by: Dimos, Marco Giordani
Change-Id: I5873762e4f05824e7b6e94a19dd4eb56adbbbb79
This change fixes a bug where a REMB collector may be
freed twice, and also tweaks REMB combining such that if
there is no bitrate from anyone (or there are no sources)
we report 0 instead of using an old bitrate.
ASTERISK-27804
Change-Id: Ia9dc9c150043890ee7ff85e9cdec007f1a77fcfd
