asterisk/asterisk.git
3 years agoARI: Add the ability to download the media associated with a stored recording
Matt Jordan [Wed, 18 May 2016 11:19:58 +0000 (06:19 -0500)]
ARI: Add the ability to download the media associated with a stored recording

This patch adds a new feature to ARI that allows a client to download
the media associated with a stored recording. The new route is
/recordings/stored/{name}/file, and transmits the underlying binary file
using Asterisk's HTTP server's underlying file transfer facilities.

Because this REST route returns non-JSON, a few small enhancements had
to be made to the Python Swagger generation code, as well as the
mustache templates that generate the ARI bindings.

ASTERISK-26042 #close

Change-Id: I49ec5c4afdec30bb665d9c977ab423b5387e0181

3 years agores_pjsip_exten_state: Use the extension for publishing to.
Joshua Colp [Tue, 10 May 2016 16:28:04 +0000 (13:28 -0300)]
res_pjsip_exten_state: Use the extension for publishing to.

This change uses the newly added multi-user support for
outbound publish to publish to the specific user that an
extension state change is for.

This also extends the res_pjsip_outbound_publish support
to include the user specific From and To URI information in
the outbound publishing of extension state. Since the URI
is used when constructing the body it is important to ensure
that the correct local and remote URIs are used.

Finally the max string growths for the dialog-info+xml
body generator has been increased as through testing it has
proven to be too conservative.

ASTERISK-25965

Change-Id: I668fdf697b1e171d4c7e6f282b2e1590f8356ca1

3 years agores_pjsip_outbound_publish: Add multi-user support per configuration
Kevin Harwell [Tue, 3 May 2016 21:07:23 +0000 (16:07 -0500)]
res_pjsip_outbound_publish: Add multi-user support per configuration

Added a new multi_user option that when specified allows a particular
configuration to be used for multiple users. It does this by replacing
the user portion of the server uri with a dynamically created one.

Two new API calls have been added in order to make use of the new
functionality:

ast_sip_publish_user_send - Sends an outgoing publish message based on the
given user. If state for the user already exists it uses that, otherwise
it dynamically creates new outbound publishing state for the user at that
time.

ast_sip_publish_user_remove - Removes all outbound publish state objects
associated with the user. This essentially stops outbound publishing for
the user.

ASTERISK-25965 #close

Change-Id: Ib88dde024cc83c916424645d4f5bb84a0fa936cc

3 years agoMerge "CHANGES: Update formatting of items"
Joshua Colp [Wed, 18 May 2016 23:35:32 +0000 (18:35 -0500)]
Merge "CHANGES: Update formatting of items"

3 years agoMerge "ARI: Add the ability to play multiple media URIs in a single operation"
Joshua Colp [Wed, 18 May 2016 23:35:20 +0000 (18:35 -0500)]
Merge "ARI: Add the ability to play multiple media URIs in a single operation"

3 years agoMerge "chan_sip: Prevent extra Session-Expires headers from being added"
Joshua Colp [Wed, 18 May 2016 23:27:28 +0000 (18:27 -0500)]
Merge "chan_sip:  Prevent extra Session-Expires headers from being added"

3 years agoCHANGES: Update formatting of items
Matt Jordan [Sun, 15 May 2016 17:22:42 +0000 (12:22 -0500)]
CHANGES: Update formatting of items

* Provide consistent indenting of lines in bulleted paragraphs
* Respect the 80 character column width
* Group all like items together, e.g., all dialplan applications under
  "Applications", etc.
* Use a single blank line to break up functionality changes within a
  larger section
* Use two blanks lines to delineate larger sections

Change-Id: I0488554f5cb7c51da70003d69288a21c9aab9647

3 years agoARI: Add the ability to play multiple media URIs in a single operation
Matt Jordan [Mon, 18 Apr 2016 23:17:08 +0000 (18:17 -0500)]
ARI: Add the ability to play multiple media URIs in a single operation

Many ARI applications will want to play multiple media files in a row to
a resource. The most common use case is when building long-ish IVR prompts
made up of multiple, smaller sound files. Today, that requires building a
small state machine, listening for each PlaybackFinished event, and triggering
the next sound file to play. While not especially challenging, it is tedious
work. Since requiring developers to write tedious code to do normal activities
stinks, this patch adds the ability to play back a list of media files to a
resource.

Each of the 'play' operations on supported resources (channels and bridges)
now accepts a comma delineated list of media URIs to play. A single Playback
resource is created as a handle to the entire list. The operation of playing
a list is identical to playing a single media URI, save that a new event,
PlaybackContinuing, is raised instead of a PlaybackFinished for each non-final
media URI. When the entire list is finished being played, a PlaybackFinished
event is raised.

In order to help inform applications where they are in the list playback, the
Playback resource now includes a new, optional attribute, 'next_media_uri',
that contains the next URI in the list to be played.

It's important to note the following:
 - If an offset is provided to the 'play' operations, it only applies to the
   first media URI, as it would be weird to skip n seconds forward in every
   media resource.
 - Operations that control the position of the media only affect the current
   media being played. For example, once a media resource in the list
   completes, a 'reverse' operation on a subsequent media resource will not
   start a previously completed media resource at the appropiate offset.
 - This patch does not add any new operations to control the list. Hopefully,
   user feedback and/or future patches would add that if people want it.

ASTERISK-26022 #close

Change-Id: Ie1ea5356573447b8f51f2e7964915ea01792f16f

3 years agochan_sip: Prevent extra Session-Expires headers from being added
George Joseph [Tue, 17 May 2016 16:14:51 +0000 (10:14 -0600)]
chan_sip:  Prevent extra Session-Expires headers from being added

When chan_sip does a re-INVITE to refresh a session and authentication
is required, the INVITE with the Authorization header containes a
second Session-Expires header without the ";refersher=" parameter.
This is causing some proxies to return a 400.  Also, when Asterisk is
the uas and the refresher, it is including the Session-Expires and
Min-SE headers in OPTIONS messages which is not allowed per RFC4028.

This patch (based on the reporter's) Checks to see if a Session-Expires
header is already in the message before adding another one.  It also
checks that the method is INVITE or UPDATE.

ASTERISK-26030 #close

Change-Id: I58a7b07bab5a3177748d8a7034fb8ad8e11ce1d9

3 years agores_pjsip_outbound_registration: Clean up state when registration is deleted
George Joseph [Mon, 16 May 2016 20:29:38 +0000 (14:29 -0600)]
res_pjsip_outbound_registration:  Clean up state when registration is deleted

Nothing was cleaning up the registration state object when ast_sorcery_delete
was called on a registration.  So, the registration was deleted from sorcery
but the state object went right on refreshing the registration (or failing
to refresh the registration) with the peer.

* Added a 'deleted' observer on registration that removes the state object.

ASTERISK-25964 #close
Reported-by Matt Jordan

Change-Id: I2db792145cdb1f72ebbf57dd9099596dbbf12c23

3 years agoMerge "configs/samples/pjsip.conf.sample: Fix typo"
zuul [Mon, 16 May 2016 18:53:00 +0000 (13:53 -0500)]
Merge "configs/samples/pjsip.conf.sample: Fix typo"

3 years agores_pjsip: Set TCP_NODELAY on TCP transports
George Joseph [Mon, 16 May 2016 00:05:34 +0000 (18:05 -0600)]
res_pjsip:  Set TCP_NODELAY on TCP transports

Although it's perfectly legal to place multiple SIP messages in the same packet,
it can cause problems because the Linux default is to enable Path MTU Discovery
which sets the Don't Fragment bit on the packets. If adding a second message to
the packet causes the MTU to be exceeded, and the destination isn't equipped to
send a FRAGMENTATION NEEDED response to a large packet, the packet will just be
dropped.

We can't specifically tell the stack to send only 1 message per packet, but we
can turn on TCP_NODELAY when we create the transport. This will at least tell
the stack to send packets as soon as possible.

ASTERISK-26005 #close
Reported-by: Ross Beer

Change-Id: I820f23227183f2416ca5e393bec510e8fe1c8fbd

3 years agoconfigs/samples/pjsip.conf.sample: Fix typo
Matt Jordan [Sun, 15 May 2016 02:48:56 +0000 (21:48 -0500)]
configs/samples/pjsip.conf.sample: Fix typo

A ':' is not a valid token for starting a comment.

Change-Id: I123592d93a83d1bdde3e352822881eb9da85e5ad

3 years agoMerge "logger: Add PID to syslog messages."
Joshua Colp [Sun, 15 May 2016 01:37:43 +0000 (20:37 -0500)]
Merge "logger: Add PID to syslog messages."

3 years agores_ari: Correct Location headers returned by some ARI resources
Sean Bright [Sat, 14 May 2016 17:29:09 +0000 (13:29 -0400)]
res_ari: Correct Location headers returned by some ARI resources

The Location headers returned by:

 * /bridges/{bridgeId}/play
 * /bridges/{bridgeId}/record
 * /channels/{channelId}/play
 * /channels/{channelId}/record

Did not have the '/ari' prefix, and in the case of the 'play' resources, were
using 'playback' instead of 'playbacks.'

Change-Id: I957c58a3a1471bf477dae7c67faa1b74fcd9241c

3 years agoMerge "config_transport: Tell pjproject to allow all SSL/TLS protocols"
zuul [Fri, 13 May 2016 22:57:55 +0000 (17:57 -0500)]
Merge "config_transport:  Tell pjproject to allow all SSL/TLS protocols"

3 years agoMerge "pjsip_distributor: Add missing newline to NOTICE"
zuul [Fri, 13 May 2016 11:21:23 +0000 (06:21 -0500)]
Merge "pjsip_distributor:  Add missing newline to NOTICE"

3 years agoMerge "basic-cfg: asterisk.conf: don't set languages"
Joshua Colp [Fri, 13 May 2016 09:53:39 +0000 (04:53 -0500)]
Merge "basic-cfg: asterisk.conf: don't set languages"

3 years agoMerge "basic-cfg: asterisk.conf: debug level 5 spams"
Joshua Colp [Fri, 13 May 2016 09:53:27 +0000 (04:53 -0500)]
Merge "basic-cfg: asterisk.conf: debug level 5 spams"

3 years agoMerge "basic-cfg: asterisk.conf: defaults of options"
Joshua Colp [Fri, 13 May 2016 09:53:13 +0000 (04:53 -0500)]
Merge "basic-cfg: asterisk.conf: defaults of options"

3 years agoMerge "followme: delete the right recorded name file"
zuul [Fri, 13 May 2016 02:44:11 +0000 (21:44 -0500)]
Merge "followme: delete the right recorded name file"

3 years agoMerge "basic-cfg: asterisk.conf: remove [directories]"
zuul [Fri, 13 May 2016 00:52:13 +0000 (19:52 -0500)]
Merge "basic-cfg: asterisk.conf: remove [directories]"

3 years agoUse doubles instead of floats for conversions when comparing strings.
Mark Michelson [Thu, 12 May 2016 19:36:25 +0000 (14:36 -0500)]
Use doubles instead of floats for conversions when comparing strings.

In 13.9.0, there was an issue where PJSIP contacts added to an AOR would
be deleted at seemingly random times.

One reason this was happening was because of an operation to retrieve
the contacts whose expiration time was less than or equal to the current
time. When retrieving existing contacts, the contact's expiration time
and the current time were converted from a string to a float, and those
two floats were compared.

On some systems, including mine, this conversion was horribly off. For
instance, I could regularly see the string "1463079214" get converted
into 1463079168.000000. When switching from using a float to using a
double, the conversion was as expected.

Why was the conversion to float off? My best guess is that the
conversion to float was attempting to store the entire value in the 23
bit significand of the IEEE-754 floating point number. In particular, if
you take only the 23 most significant bits of 1463079214, you get the
messed up 1463079168 that we were seeing in the conversion. It likely
was possible to get a more precise value by composing the number using
an exponent, but the conversion did not work that way. With a double,
you have a 52 bit significand, allowing the entire value to fit there,
and thereby allowing an accurate conversion.

ASTERISK-26007 #close
Reported by Greg Siemon

Change-Id: I83ca7944aae8b7cd994b254c78ec02411d321070

3 years agoMerge "res_pjsip_outbound_registration: generate correct Contact URI for TLS"
zuul [Thu, 12 May 2016 19:25:43 +0000 (14:25 -0500)]
Merge "res_pjsip_outbound_registration: generate correct Contact URI for TLS"

3 years agopjsip_distributor: Add missing newline to NOTICE
George Joseph [Thu, 12 May 2016 14:13:55 +0000 (08:13 -0600)]
pjsip_distributor:  Add missing newline to NOTICE

There was a newline missing from the end of the "no matching endpoint" notice.

Change-Id: Idc11fe5bc0354072291663dbffe648c471e39181

3 years agores_pjsip_outbound_registration: generate correct Contact URI for TLS
Sebastian Damm [Tue, 10 May 2016 15:19:48 +0000 (17:19 +0200)]
res_pjsip_outbound_registration: generate correct Contact URI for TLS

There are two types of SIP URIs indicating a secure transport:
* sips:user@example.org
* sip:user@example.org;transport=tls

When using a sips URI, Asterisk checks incoming INVITEs and answers from
the other side for sips URIs, and rejects the packet if there are only
sip URIs. So Asterisk should only generate a sips Contact URI if the
other side supports it.

This patch makes Asterisk generate either a sip or sips Contact URI
depending on the format of the server URI.

If you want a sip URI, use:
server_uri=sip:example.org\;transport=tls

If you want a sips URI, use:
server_uri=sips:example.org

ASTERISK-25990 #close
Reported-by: Sebastian Damm

Change-Id: I5ae57d6531ce940b5fc64d5cd2673e60db0f9ba2

3 years agologger: Add PID to syslog messages.
Alexei Gradinari [Thu, 5 May 2016 21:41:21 +0000 (17:41 -0400)]
logger: Add PID to syslog messages.

During refactoring of this support the addition of
the PID to messages was removed. This change adds it
back in.

ASTERISK-25538 #close

Change-Id: Ie2d43b0652e59b7ac319a7dba94501540d70ba36

3 years agoconfigure: Fix errors with AST_UNDEFINED_SANITIZER/AST_LEAK_SANITIZER
Matt Jordan [Wed, 11 May 2016 19:07:17 +0000 (14:07 -0500)]
configure: Fix errors with AST_UNDEFINED_SANITIZER/AST_LEAK_SANITIZER

When running on a system that does not support or use AST_UNDEFINED_SANITIZER
or AST_LEAK_SANITIZER, the configure script would incorrectly set those
constants to a blank value, e.g., 'AST_UNDEFINED_SANITIZER='. This would
cause menuselect to error out, complaining that a blank value is not a
valid option. This patch corrects the issue by setting the value to 0 if
the options that those constants enable/disable is not found.

Change-Id: Ib39814aaf940f308d500c1e026edb3d70de47fba

3 years agoMerge "res_pjsip_outbound_publish: state potential dropped on reloads/realtime fetches"
Joshua Colp [Wed, 11 May 2016 17:57:34 +0000 (12:57 -0500)]
Merge "res_pjsip_outbound_publish: state potential dropped on reloads/realtime fetches"

3 years agoMerge "res_pjsip_outbound_publishing: After unloading the library won't load again"
Joshua Colp [Wed, 11 May 2016 17:57:24 +0000 (12:57 -0500)]
Merge "res_pjsip_outbound_publishing: After unloading the library won't load again"

3 years agoMerge "res_pjsip_outbound_publish: Won't unload if condition wait times out"
Joshua Colp [Wed, 11 May 2016 17:57:13 +0000 (12:57 -0500)]
Merge "res_pjsip_outbound_publish: Won't unload if condition wait times out"

3 years agoMerge "res_pjsip_outbound_publish: Ref leak in off nominal callback paths"
Joshua Colp [Wed, 11 May 2016 17:57:04 +0000 (12:57 -0500)]
Merge "res_pjsip_outbound_publish: Ref leak in off nominal callback paths"

3 years agoMerge "res_pjsip_outbound_publish: Potential crash due to off nominal path"
Joshua Colp [Wed, 11 May 2016 17:56:52 +0000 (12:56 -0500)]
Merge "res_pjsip_outbound_publish: Potential crash due to off nominal path"

3 years agoMerge "res_pjsip: improve realtime performance"
Joshua Colp [Wed, 11 May 2016 15:58:54 +0000 (10:58 -0500)]
Merge "res_pjsip: improve realtime performance"

3 years agoMerge "res_fax/t38_gateway: Peer V.21 session is created on wrong channel"
zuul [Wed, 11 May 2016 15:36:34 +0000 (10:36 -0500)]
Merge "res_fax/t38_gateway: Peer V.21 session is created on wrong channel"

3 years agofollowme: delete the right recorded name file
Tzafrir Cohen [Tue, 10 May 2016 13:17:29 +0000 (16:17 +0300)]
followme: delete the right recorded name file

FollowMe with the option a records the name of the caller and plays it
to the callee. However it has failed to clean up that recorded file
as it tried to delete the file name without the '.sln' extension.

ASTERISK-26008 #close

Change-Id: I79d7b1be7d5cde57bf076d9389e2a8a4422776ec
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>

3 years agoMerge "app_confbridge: Add a regcontext option for confbridge bridge profiles."
Joshua Colp [Tue, 10 May 2016 09:48:35 +0000 (04:48 -0500)]
Merge "app_confbridge: Add a regcontext option for confbridge bridge profiles."

3 years agobasic-cfg: asterisk.conf: don't set languages
Tzafrir Cohen [Tue, 10 May 2016 08:10:55 +0000 (11:10 +0300)]
basic-cfg: asterisk.conf: don't set languages

* No need to set language in a miniml configuration. 'en' will do just
  fine.
* It would be useful to have an example of setting it to a different
  language.
* Setting the documentation language explicitly is likewise not
  required. Setting it to a different value is not common. At least
  until there is a set of translated documentation.

Change-Id: I94d91ea34e129925f25af81ef8dc0906fb568cb7
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>

3 years agobasic-cfg: asterisk.conf: debug level 5 spams
Tzafrir Cohen [Tue, 10 May 2016 08:08:33 +0000 (11:08 +0300)]
basic-cfg: asterisk.conf: debug level 5 spams

Don't suggest users to use debug level 5, which spews (usually
non-useful) debug information. Reduce the suggestion to (an
arbitrarily-selected) level 2.

Change-Id: Ib53195f78945970956ff59ef13fa89b90e0fcd60
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>

3 years agobasic-cfg: asterisk.conf: defaults of options
Tzafrir Cohen [Tue, 10 May 2016 08:06:10 +0000 (11:06 +0300)]
basic-cfg: asterisk.conf: defaults of options

Note the default of remmed-out options. To clarify that those values are
not the defaults.

Change-Id: I849c29b7a710f0abc37355fcb5bfee335ae30738
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>

3 years agobasic-cfg: asterisk.conf: remove [directories]
Tzafrir Cohen [Tue, 10 May 2016 07:56:40 +0000 (10:56 +0300)]
basic-cfg: asterisk.conf: remove [directories]

A minimal configuration does not need to explicitly spell out the
directories. The built-in defaults will do just fine. In many cases
they are wrong.

Change-Id: Id1a671e5c5e9923765a4156b57f9f7e263fdd26c
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>

3 years agoMerge "res_pjsip_authenticator_digest: Don't use source port in nonce verification"
zuul [Tue, 10 May 2016 03:56:53 +0000 (22:56 -0500)]
Merge "res_pjsip_authenticator_digest: Don't use source port in nonce verification"

3 years agoMerge "pjproject_bundled: Check for python-dev and TEST_FRAMEWORK"
Joshua Colp [Mon, 9 May 2016 23:49:42 +0000 (18:49 -0500)]
Merge "pjproject_bundled:  Check for python-dev and TEST_FRAMEWORK"

3 years agoMerge "res_pjsip_pubsub: Use common datastores container API."
Joshua Colp [Mon, 9 May 2016 23:27:35 +0000 (18:27 -0500)]
Merge "res_pjsip_pubsub: Use common datastores container API."

3 years agoMerge "datastore: Add common container based datastores API."
zuul [Mon, 9 May 2016 23:23:44 +0000 (18:23 -0500)]
Merge "datastore: Add common container based datastores API."

3 years agores_pjsip_authenticator_digest: Don't use source port in nonce verification
Kevin Harwell [Thu, 5 May 2016 16:37:37 +0000 (11:37 -0500)]
res_pjsip_authenticator_digest: Don't use source port in nonce verification

From the issue reporter:
"res_pjsip_outbound_authenticator_digest builds a nonce that is a hash of
the timestamp, the source address, the source port, a server UUID that is
calculated at startup, and the authentication realm.

Rather than caching nonces that we create, we instead attempt to re-calculate
the nonce when receiving an incoming request with authentication. We then
compare the re-calculated nonce to the incoming nonce, and if they don't match,
then authentication has failed early.

The problem is that it is possible, especially when using TCP, to receive two
requests from the same endpoint but have differing source ports for those
requests. Asterisk itself commonly will use different source ports for
outbound TCP requests."

This patch removes the source port dependency when building the nonce.

ASTERISK-25978 #close

Change-Id: I871b5f4adce102df1c4988066283095ec509dffe

3 years agoconfig_transport: Tell pjproject to allow all SSL/TLS protocols
George Joseph [Sat, 7 May 2016 19:39:25 +0000 (13:39 -0600)]
config_transport:  Tell pjproject to allow all SSL/TLS protocols

The default tls settings for pjproject only allow TLS 1, TLS 1.1 and TLS 1.2.
SSL is not allowed.   So, even if you specify "sslv3" for a transport method,
it's silently ignored and one of the TLS protocols is used.  This was a new
behavior of pjsip_tls_setting_default() in 2.4 (when tls.proto was added) that
we never caught.

Now we need to set tls.proto = 0 after we call pjsip_tls_setting_default().
This tells pjproject to set the socket protocol to match the method.

ASTERISK-26004 #close

Change-Id: Icfb55c1ebe921298dedb4b1a1d3bdc3ca41dd078

3 years agoMerge "res_pjsip: module load priority"
Joshua Colp [Mon, 9 May 2016 15:06:10 +0000 (10:06 -0500)]
Merge "res_pjsip: module load priority"

3 years agores_pjsip_pubsub: Use common datastores container API.
Joshua Colp [Thu, 5 May 2016 14:14:00 +0000 (11:14 -0300)]
res_pjsip_pubsub: Use common datastores container API.

This migrates res_pjsip_pubsub over to using the newly
introduce common datastores management API instead of using
its own implementations for both subscriptions and
publications.

As well the extension state data now provides a generic
datastores container instead of a subscription. This allows
the dialog-info+xml body generator to work for both
subscriptions and publications.

ASTERISK-25999 #close

Change-Id: I773f9e4f35092da0f653566736a8647e8cfebef1

3 years agodatastore: Add common container based datastores API.
Joshua Colp [Thu, 5 May 2016 14:12:59 +0000 (11:12 -0300)]
datastore: Add common container based datastores API.

This change introduces a common container based datastores
management API. This has been done in a few places across
the tree but this consolidates all of the logic into one
place in a generic fashion.

ASTERISK-25999

Change-Id: I72eb15941dcdbc2a37bb00a33ce00f8755bd336a

3 years agoMerge "file: Ensure nativeformats remains valid for lifetime of use."
Joshua Colp [Mon, 9 May 2016 13:28:16 +0000 (08:28 -0500)]
Merge "file: Ensure nativeformats remains valid for lifetime of use."

3 years agoapp_confbridge: Add a regcontext option for confbridge bridge profiles.
Jaco Kroon [Wed, 4 May 2016 07:40:55 +0000 (09:40 +0200)]
app_confbridge: Add a regcontext option for confbridge bridge profiles.

This patch allows for having app_confbridge register the name of the
conference as an extension into a specific context, similar to
regcontext for chan_sip.  This variant is not quite as involved as the
one in chan_sip and doesn't allow for multiple contexts or custom
extensions, you can only specify the context and the conference name
will always be used as the extension to register.

ASTERISK-25989 #close

Change-Id: Icacf94d9f2b5dfd31ef36f6cb702392619a7902f

3 years agopjproject_bundled: Check for python-dev and TEST_FRAMEWORK
George Joseph [Mon, 9 May 2016 01:19:50 +0000 (19:19 -0600)]
pjproject_bundled:  Check for python-dev and TEST_FRAMEWORK

The pjsua and pjsystest apps are now built only if TEST_FRAMEWORK is set.
The python bindings are now built only if TEST_FRAMEWORK is set and a
python development package is installed.

libresample was also disabled.

ASTERISK-25993 #close
Reported-by: Joshua Colp

Change-Id: If4e91c503a02f113d5b71bc8b972081fa3ff6f03

3 years agores_pjsip: module load priority
Alexei Gradinari [Fri, 6 May 2016 16:54:17 +0000 (12:54 -0400)]
res_pjsip: module load priority

The res_pjsip_authenticator_digest, res_pjsip_endpoint_identifier_*
and res_pjsip_registrar modules should load ASAP
to avoid "No matching endpoint found" for legitimate endpoint.

ASTERISK-25994

Change-Id: Iac95d95ad031e0be104189d29e923a2ad7c24a1b

3 years agostasis_endpoints: Add new Status and Headers to ContactStatus
Alexei Gradinari [Thu, 5 May 2016 20:16:16 +0000 (16:16 -0400)]
stasis_endpoints: Add new Status and Headers to ContactStatus

ASTERISK-25903 added a new headers to AMI Event ContactStatusDetail.
ASTERISK-25904 added a new Status to AMI Event ContactStatusDetail.
These additions should be also in stasis_endpoints
to include in command "manager show event ContactStatus"

Change-Id: I7610ad02a998e1f26c20caa27aa50279d0164f6a

3 years agoMerge "config_options.c: Expand #ifdef to contain whole if statement."
zuul [Fri, 6 May 2016 11:14:40 +0000 (06:14 -0500)]
Merge "config_options.c: Expand #ifdef to contain whole if statement."

3 years agores_pjsip_outbound_publish: state potential dropped on reloads/realtime fetches
Kevin Harwell [Tue, 3 May 2016 20:43:16 +0000 (15:43 -0500)]
res_pjsip_outbound_publish: state potential dropped on reloads/realtime fetches

When reloading, or fetching realtime data, if the "apply" failed for any
numerous reasons the current state object would not be maintained. This
potentially resulted in publishes being stopped for some states/clients when
they should not have been.

This patch makes it so the current state object is kept upon any type of reload/
fetch failures.

Change-Id: Iab6020c116d628ed2ae81183e987e2eaa3c90b30

3 years agores_pjsip_outbound_publishing: After unloading the library won't load again
Kevin Harwell [Tue, 3 May 2016 20:35:24 +0000 (15:35 -0500)]
res_pjsip_outbound_publishing: After unloading the library won't load again

The same thing was happening in res_pjsip_publish_asterisk. When the library
was unloaded it did not unregister the object type from sorcery. Subsequent
loads resulted in a failed load due to the sorcery type already existing.

Change-Id: Ifdc25e94e4cd40bc5a19eb4d0a00b86c2e9fedc9

3 years agores_pjsip_outbound_publish: Won't unload if condition wait times out
Kevin Harwell [Tue, 3 May 2016 20:39:32 +0000 (15:39 -0500)]
res_pjsip_outbound_publish: Won't unload if condition wait times out

When res_pjsip_outbound_publish unloads it has to wait for all current
publishing objects to get done. However if the wait condition times out
then it does not fail the unload. This sometimes results in an infinite
loop check while unloading. This patch now fails the unload operation if
the condition times out.

Change-Id: Id57b8cbed9d61222690fcba1e4f18e259df4c7ec

3 years agores_pjsip_outbound_publish: Ref leak in off nominal callback paths
Kevin Harwell [Tue, 3 May 2016 19:59:06 +0000 (14:59 -0500)]
res_pjsip_outbound_publish: Ref leak in off nominal callback paths

There were a few spots where the client object's reference was being leaked in
sip_outbound_publish_callback. This patch cleans up those leaks.

Change-Id: I485d0bc9335090f373026f77c548042e258461df

3 years agores_pjsip_outbound_publish: Potential crash due to off nominal path
Kevin Harwell [Tue, 3 May 2016 20:31:19 +0000 (15:31 -0500)]
res_pjsip_outbound_publish: Potential crash due to off nominal path

It was possible for the explicit publish destroy function to be called without
the pjsip client ever being initialized. This fix checks to make sure there is
a client to destroy before attempting.

Change-Id: I8eea1bfa3bd472149bfc255310be2a6248688f5c

3 years agofile: Ensure nativeformats remains valid for lifetime of use.
Joshua Colp [Thu, 5 May 2016 10:07:50 +0000 (07:07 -0300)]
file: Ensure nativeformats remains valid for lifetime of use.

It is possible for the nativeformats of a channel to change
throughout its lifetime. As a result a user of it needs to either
ensure the channel is locked when accessing the formats or keep
a reference to the nativeformats themselves.

This change fixes the file playback support so it keeps a
reference to the nativeformats when accessing things.

ASTERISK-25998 #close

Change-Id: Ie45b65475e1481ddf05b874ee48f63e39fff8915

3 years agores_pjsip: improve realtime performance
Alexei Gradinari [Fri, 15 Apr 2016 14:32:12 +0000 (10:32 -0400)]
res_pjsip: improve realtime performance

This patch modified pjsip_options to retrieve only
permament contacts for aor if the qualify_frequency is > 0
and persisted contacts if the qualify_frequency is > 0.

This patch also fixed a bug in res_sorcery_astdb.
res_sorcery_astdb doesn't save object data retrived from astdb.

ASTERISK-25826

Change-Id: I1831fa46c4578eae5a3e574ee3362fddf08a1f05

3 years agores_fax/t38_gateway: Peer V.21 session is created on wrong channel
Alexei Gradinari [Mon, 2 May 2016 21:52:16 +0000 (17:52 -0400)]
res_fax/t38_gateway: Peer V.21 session is created on wrong channel

The channel and peer V.21 sessions are created on the same channel now.
The peer V.21 session should be created only on peer channel
when one of channel can handle T.38.

Also this patch enable debug for T.38 gateway session
if global fax debug enabled.

ASTERISK-25982

Change-Id: I78387156ea521a77eb0faf170179ddd37a50430e

3 years agopjsip: Added "reg_server" to contacts (fixed alembic)
Alexei Gradinari [Wed, 4 May 2016 21:11:17 +0000 (17:11 -0400)]
pjsip: Added "reg_server" to contacts (fixed alembic)

ASTERISK-25931

Change-Id: Icc4321a88f5c93ff809da3f372eebbf69c6a8549

3 years agoconfig_options.c: Expand #ifdef to contain whole if statement.
Chris Trobridge [Wed, 4 May 2016 08:17:26 +0000 (09:17 +0100)]
config_options.c: Expand #ifdef to contain whole if statement.

ASTERISK-25956 #close

Change-Id: If6961ec54be276d5ab4f012ee7e7b420cb45de38

3 years agores_fax: add FAXMODE variable
Alexei Gradinari [Mon, 2 May 2016 21:08:06 +0000 (17:08 -0400)]
res_fax: add FAXMODE variable

The app_fax set FAXMODE variable, but res_fax missing this feature.
This patch add FAXMODE variable which is set to either "audio" or "T38".

ASTERISK-25980

Change-Id: Ie3dcbfb72cc681e9e267a60202f7fb8723a51b6b

3 years agoMerge "app_chanspy: fix audiohook options in non read-only mode"
Joshua Colp [Wed, 4 May 2016 09:49:29 +0000 (04:49 -0500)]
Merge "app_chanspy: fix audiohook options in non read-only mode"

3 years agoMerge "app_voicemail: always copy dynamic struct to avoid race condition"
Joshua Colp [Wed, 4 May 2016 09:49:19 +0000 (04:49 -0500)]
Merge "app_voicemail: always copy dynamic struct to avoid race condition"

3 years agoMerge "res_pjsip/AMI: add contact.updated event"
zuul [Wed, 4 May 2016 03:06:34 +0000 (22:06 -0500)]
Merge "res_pjsip/AMI: add contact.updated event"

3 years agoMerge "pjproject_bundled: Various fixes discovered during testing of OSes"
zuul [Wed, 4 May 2016 00:11:18 +0000 (19:11 -0500)]
Merge "pjproject_bundled:  Various fixes discovered during testing of OSes"

3 years agoapp_chanspy: fix audiohook options in non read-only mode
Jean Aunis [Mon, 2 May 2016 10:56:24 +0000 (12:56 +0200)]
app_chanspy: fix audiohook options in non read-only mode

When option 'o' was not set, ChanSpy created its audiohook with the flag
AST_AUDIOHOOK_MUTE_WRITE, which caused ChanSpy to listen audio from one
direction only.

ASTERISK-25866 #close

Change-Id: I5c745855eea29a3fbc4e4aed0b0c0f53580535e0

3 years agores_pjsip/AMI: add contact.updated event
Alexei Gradinari [Thu, 7 Apr 2016 21:33:49 +0000 (17:33 -0400)]
res_pjsip/AMI: add contact.updated event

With the old SIP module AMI sends PeerStatus event on every
successfully REGISTER requests, ie, on start registration,
update registration and stop registration.

With PJSIP AMI sends ContactStatus only when status is changed.
Regarding registration:
on start registration - Created
on stop registration - Removed
but on update registration nothing

This patch added contact.updated event.

ASTERISK-25904

Change-Id: I8fad8aae9305481469c38d2146e1ba3a56d3108f

3 years agoMerge "pjsip: Added "reg_server" to contacts."
zuul [Tue, 3 May 2016 19:05:45 +0000 (14:05 -0500)]
Merge "pjsip: Added "reg_server" to contacts."

3 years agoMerge "configs/basic-pbx/asterisk.conf: contains incorrect path separator"
Joshua Colp [Tue, 3 May 2016 17:11:21 +0000 (12:11 -0500)]
Merge "configs/basic-pbx/asterisk.conf: contains incorrect path separator"

3 years agopjproject_bundled: Various fixes discovered during testing of OSes
George Joseph [Sat, 30 Apr 2016 22:52:47 +0000 (16:52 -0600)]
pjproject_bundled:  Various fixes discovered during testing of OSes

For all OSes:
* Disabled third-party codecs in pjproject and added
  '--disable-speex-codec --disable-speex-aec --disable-gsm-codec' to the
  configure options since we don't use the pjsip codec capability.

FreeBSD:
* Added FreeBSD support to install_prereq.
* Changed pjproject/configure.m4 to use $GNU_MAKE instead of hardcoding "make".
* Added __progname and environ to asterisk.exports.in.
* Reverted the use of ldconfig to create shared library symlinks to ln.
* Only enable epoll in pjproject if `uname -s` is Linux.
* Added a patch to pjproject to take the name of the 'make' command from
  an environment variable if supplied.  This is needed for the python bindings.
  (merged by Teluu into pjproject trunk 5/3/2016)
FreeBSD support isn't complete.  Still some general issues regarding
make/gmake having nothing to do with pjproject.  With some handholding it DOES
build successfully.

CentOS:
Added 'patch' and 'bzip2' to install_prereq PACKAGES_RH.
CentOS 6/7 32/64 build and run the pjsip testsuite successfully.

Ubuntu:
No changes required.
Ubuntu 15/16 32/64 build and run the pjsip testsuite successfully.

Debian:
No changes required.
Debian 6/7/8 32/64 build and run the pjsip testsuite successfully.

There will utimately be a follow-up patch to create an install_prereq for
the testsuite as I've discovered a few missing requirements.

ASTERISK-25968 #close

Change-Id: I5756a07facfc63798115a5e73a8709382fe9259c

3 years agoMerge "res_pjsip_exten_state: Create PUBLISH messages."
zuul [Tue, 3 May 2016 11:10:52 +0000 (06:10 -0500)]
Merge "res_pjsip_exten_state: Create PUBLISH messages."

3 years agoapp_voicemail: always copy dynamic struct to avoid race condition
Andrew Nagy [Thu, 17 Mar 2016 19:29:38 +0000 (12:29 -0700)]
app_voicemail: always copy dynamic struct to avoid race condition

Voicemail email addresses can be corrupt or voicemail
emails can end up being sent to the wrong email address if asterisk is
reading voicemail.conf during a reload and processing an email at the
same time. This patch always copies the struct that would otherwise only
be copied once.

ASTERISK-24463 #close
Reported by: John Campbell
Tested by: Etienne Lessard
Tested by: Andrew Nagy
Change-Id: I3a0643813116da84e2617291903d0d489b7425fb

3 years agopjsip: Added "reg_server" to contacts.
Alexei Gradinari [Fri, 15 Apr 2016 19:26:15 +0000 (15:26 -0400)]
pjsip: Added "reg_server" to contacts.

If the Asterisk system name is set in asterisk.conf, it will be stored
into the "reg_server" field in the ps_contacts table to facilitate
multi-server setups.

ASTERISK-25931

Change-Id: Ia8f6bd2267809c78753b52bcf21835b9b59f4cb8

3 years agoconfigs/basic-pbx/asterisk.conf: contains incorrect path separator
Diederik de Groot [Sun, 1 May 2016 07:21:33 +0000 (09:21 +0200)]
configs/basic-pbx/asterisk.conf: contains incorrect path separator

Note: When packagers use these files (as an example) the paths are never
really used when they are split using '='.

Note: Thirdparty applications will also have trouble parsing the file when
expecting '=>'.

Change-Id: I0ada647f588e81f023fb1333ca15a1a333fd6004

3 years agoMerge "pjproject_bundled: Disable PJSIP_UNESCAPE_IN_PLACE"
Joshua Colp [Fri, 29 Apr 2016 19:57:19 +0000 (14:57 -0500)]
Merge "pjproject_bundled:  Disable PJSIP_UNESCAPE_IN_PLACE"

3 years agores_pjsip_exten_state: Create PUBLISH messages.
Richard Mudgett [Wed, 27 Apr 2016 22:19:53 +0000 (17:19 -0500)]
res_pjsip_exten_state: Create PUBLISH messages.

Create PUBLISH messages to update a third party when an extension state
changes because of either a device or presence state change.

A configuration example:

[exten-state-publisher]
type=outbound-publish
server_uri=sip:instance1@172.16.10.2
event=presence
; Optional regex for context filtering, if specified only extension state
; for contexts matching the regex will cause a PUBLISH to be sent.
@context=^users
; Optional regex for extension filtering, if specified only extension
; state for extensions matching the regex will cause a PUBLISH to be sent.
@exten=^[0-9]*
; Required body type for the PUBLISH message.
;
; Supported values are:
; application/pidf+xml
; application/xpidf+xml
; application/cpim-pidf+xml
; application/dialog-info+xml (Planned support but not yet)
@body=application/pidf+xml

The '@' extended variables are used because the implementation can't
extend the outbound publish type as it is provided by the outbound publish
module.  That means you either have to use extended variables, or
implement some sort of custom extended variable thing in the outbound
publish module.  Another option would be to refactor that stuff to have an
option which specifies the use of an alternate implementation's
configuration and then have that passed to the implementation.  JColp
opted for the extended variables method originally.

ASTERISK-25972 #close

Change-Id: Ic0dab4022f5cf59302129483ed38398764ee3cca

3 years agoMerge "res_pjsip_exten_state: Check if body generator is available."
Joshua Colp [Fri, 29 Apr 2016 19:33:01 +0000 (14:33 -0500)]
Merge "res_pjsip_exten_state: Check if body generator is available."

3 years agoMerge "res_pjsip_pubsub.c: Fix body generator registration race."
Joshua Colp [Fri, 29 Apr 2016 18:33:43 +0000 (13:33 -0500)]
Merge "res_pjsip_pubsub.c: Fix body generator registration race."

3 years agoMerge "res_pjsip: Start body generator users after suppliers."
zuul [Fri, 29 Apr 2016 18:01:06 +0000 (13:01 -0500)]
Merge "res_pjsip: Start body generator users after suppliers."

3 years agoMerge "chan_sip: Make autocreated peers send PeerStatus events"
Joshua Colp [Fri, 29 Apr 2016 16:44:11 +0000 (11:44 -0500)]
Merge "chan_sip: Make autocreated peers send PeerStatus events"

3 years agoMerge "res_pjsip_pubsub.c: Add useful information to some messages."
zuul [Fri, 29 Apr 2016 03:55:04 +0000 (22:55 -0500)]
Merge "res_pjsip_pubsub.c: Add useful information to some messages."

3 years agoMerge "res_pjsip_pubsub.h: Fix doxygen association."
zuul [Fri, 29 Apr 2016 03:43:29 +0000 (22:43 -0500)]
Merge "res_pjsip_pubsub.h: Fix doxygen association."

3 years agoMerge "res_pjsip_outbound_publish.c: Remove redundant flag check."
zuul [Fri, 29 Apr 2016 02:02:05 +0000 (21:02 -0500)]
Merge "res_pjsip_outbound_publish.c: Remove redundant flag check."

3 years agoMerge "res_pjsip: Add ability to identify by Authorization username"
zuul [Thu, 28 Apr 2016 23:02:41 +0000 (18:02 -0500)]
Merge "res_pjsip:  Add ability to identify by Authorization username"

3 years agoMerge "app_chanspy: reduce audio loss on the spying channel."
zuul [Thu, 28 Apr 2016 22:45:57 +0000 (17:45 -0500)]
Merge "app_chanspy: reduce audio loss on the spying channel."

3 years agores_pjsip_exten_state: Check if body generator is available.
Richard Mudgett [Tue, 26 Apr 2016 21:10:26 +0000 (16:10 -0500)]
res_pjsip_exten_state: Check if body generator is available.

When starting the extension state publishers, check if the requested
message body generator is available.  If not available give error message
and skip starting that publisher.

* res_pjsip_pubsub.c: Create new API if type/subtype generator
registered.

* res_pjsip_exten_state.c: Use new body generator API for validation.

ASTERISK-25922

Change-Id: I4ad69200666e3cc909d4619e3c81042d7f9db25c

3 years agores_pjsip: Start body generator users after suppliers.
Richard Mudgett [Thu, 28 Apr 2016 16:35:44 +0000 (11:35 -0500)]
res_pjsip: Start body generator users after suppliers.

Change-Id: I8f0b57841feaab56c8a4e821b5ccb4e05e5fbadb

3 years agores_pjsip_pubsub.c: Add useful information to some messages.
Richard Mudgett [Thu, 28 Apr 2016 21:06:57 +0000 (16:06 -0500)]
res_pjsip_pubsub.c: Add useful information to some messages.

Change-Id: Ia0b2e15773894c599e5c5748bbc70e99f434192a

3 years agores_pjsip_pubsub.c: Fix body generator registration race.
Richard Mudgett [Tue, 26 Apr 2016 20:58:06 +0000 (15:58 -0500)]
res_pjsip_pubsub.c: Fix body generator registration race.

Change-Id: Id8752073ef06472a2fd96080f4009fac42843e67

3 years agopjproject_bundled: Disable PJSIP_UNESCAPE_IN_PLACE
George Joseph [Thu, 28 Apr 2016 21:54:07 +0000 (15:54 -0600)]
pjproject_bundled:  Disable PJSIP_UNESCAPE_IN_PLACE

When pjsip_parse_uri is called with PJSIP_UNESCAPE_IN_PLACE enabled,
the input uri string will become corrupted if it contains escape sequences.
It's not possible to automatically strdup or strdupa the input string because
the output uri pj_str_t's will have pointers to chunks of the input string.
Getting around this would require more memory management code and wouldn't
be worth the savings of doing the unescape in place.

ASTERISK-25970 #close
Reported-by: Dmitriy Serov

Change-Id: I28dc0e599b5108f7959b9c46dc8278371b372f88

3 years agores_pjsip_pubsub.h: Fix doxygen association.
Richard Mudgett [Tue, 26 Apr 2016 20:13:50 +0000 (15:13 -0500)]
res_pjsip_pubsub.h: Fix doxygen association.

Change-Id: I110d3e3572598289fcd4215d966cf0c858f98632

3 years agores_pjsip_outbound_publish.c: Remove redundant flag check.
Richard Mudgett [Mon, 25 Apr 2016 21:00:30 +0000 (16:00 -0500)]
res_pjsip_outbound_publish.c: Remove redundant flag check.

Change-Id: I0da80a3c3e0eae0c52ff27e7412ba027d6f52353

3 years agoMerge "res_pjsip_exten_state: Add config support for exten state publishers."
zuul [Thu, 28 Apr 2016 20:35:08 +0000 (15:35 -0500)]
Merge "res_pjsip_exten_state: Add config support for exten state publishers."

3 years agoMerge "func_odbc: Check connection status before executing queries."
zuul [Thu, 28 Apr 2016 11:53:01 +0000 (06:53 -0500)]
Merge "func_odbc: Check connection status before executing queries."