The sample config was missing the configuration options for DTMF attended
transfer completion scenarios. The configuration options 'atxferabort',
'atxfercomplete', 'atxferthreeway', and 'atxferswap' are now documented in the
appropriate configuration file.
ASTERISK-24678 #close
Reported by: Niklas Larsson
patches:
features.conf.sample.diff uploaded by Niklas Larsson (License 5068)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change makes the T.38 negotiation timeout configurable via
't38timeout' in res_fax.conf or FAXOPT(t38timeout). It was previously
hard coded to be 5000 milliseconds.
This change also handles T.38 switch failures by aborting the fax since
in the case where this can happen, both sides have agreed to switch to
T.38 and Asterisk is unable to do so.
Review: https://reviewboard.asterisk.org/r/4320/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Note that this is backport from trunk of r425825.
This change adds a module which is configurable using the keep_alive_interval setting in the
global section that will send a CRLF keep alive to all active connection-oriented transports at
the provided interval. This is useful because it can help keep connections open through NATs.
This functionality also exists within PJSIP but can not be controlled at runtime and requires
recompiling it.
Review: https://reviewboard.asterisk.org/r/4084/
ASTERISK-24644 #close
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
res_pjsip_config_wizard
------------------
* This is a new module that adds streamlined configuration capability for
chan_pjsip. It's targetted at users who have lots of basic configuration
scenarios like 'phone' or 'agent' or 'trunk'. Additional information
can be found in the sample configuration file at
config/samples/pjsip_wizard.conf.sample.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4190/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, all Stasis subscriptions would receive a dedicated
thread for servicing published messages. In contrast, prior to r400178
(see review https://reviewboard.asterisk.org/r/2881/), the subscriptions
shared a thread pool. It was discovered during some initial work on Stasis
that, for a low subscription count with high message throughput, the
threadpool was not as performant as simply having a dedicated thread per
subscriber.
For situations where a subscriber receives a substantial number of messages
and is always present, the model of having a dedicated thread per subscriber
makes sense. While we still have plenty of subscriptions that would follow
this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into
the following two categories:
* Large number of subscriptions, specifically those tied to endpoints/peers.
* Low number of messages. Some subscriptions exist specifically to coordinate
a single message - the subscription is created, a message is published, the
delivery is synchronized, and the subscription is destroyed.
In both of the latter two cases, creating a dedicated thread is wasteful (and
in the case of a large number of peers/endpoints, harmful). In those cases,
having shared delivery threads is far more performant.
This patch adds the ability of a subscriber to Stasis to choose whether or not
their messages are dispatched on a dedicated thread or on a threadpool. The
threadpool is configurable through stasis.conf.
Review: https://reviewboard.asterisk.org/r/4193
ASTERISK-24533 #close
Reported by: xrobau
Tested by: xrobau
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Optimistic SRTP is the ability to enable SRTP but not have it be
a fatal requirement. If SRTP can be used it will be, if not it won't be.
This gives you a better chance of using it without having your sessions
fail when it can't be.
Encrypt all the things!
Review: https://reviewboard.asterisk.org/r/3992/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This allows for a configurable number of attempts for a transferer
to dial an extension to transfer the call to. For Asterisk 13, the
default values are such that upgrading between versions will not
cause a behaivour change. For trunk, though, the defaults will be
changed to be more user-friendly.
Review: https://reviewboard.asterisk.org/r/4167
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Testing has shown repeatedly that PJSIP's default behavior of switching
automatically to TCP for large messages can cause issues. The most common
issues are that devices that we are communicating with do not handle the
switch to TCP gracefully, thus causing situations such as broken calls or
broken subscriptions. Now, in order to have this behavior happen, you must
opt into it. The sample file has been updated to warn that enabling the
TCP switch behavior may cause issues for you, so use at your own risk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a packet exceeds the MTU, pjproject will switch from UDP to TCP. In
some circumstances (on some networks), this can cause some issues with
messages not getting sent to the correct destination - and can also cause
connections to get dropped due to quirks in pjproject deciding to
terminate TCP connections with no messages.
While fixing the routing/messaging issues is important, having a
configuration option in Asterisk that tells pjproject to not switch over
to TCP would be useful. That way, if some glitch is discovered on some
other network/site, we can at least disable the behavior until a fix is
put into place.
AFS-197 #close
Review: https://reviewboard.asterisk.org/r/4137/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A question arose as to whether a .pem file
could be provided in place of the .crt and
.key files in a PJSIP TLS configuration. I
tested this and discovered that although a
cert will be read from the pem file, a key
will not, and thus the priv_key_file entry
is still required. This update to the fine
documentation clarifies the option usage.
AST-1448 #close
Review: https://reviewboard.asterisk.org/r/4129/
Reported by: John Bigelow
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch makes res_phoneprov more modular so other modules (like pjsip)
can provide configuration information instead of res_phoneprov relying solely
on users.conf and sip.conf. To accomplish this a new ast_phoneprov public API
is now exposed which allows config providers to register themselves, set
defaults (server profile, etc) and add user extensions.
* ast_phoneprov_provider_register registers the provider and provides callbacks
for loading default settings and loading users.
* ast_phoneprov_provider_unregister clears the defaults and users.
* ast_phoneprov_add_extension should be called once for each user/extension
by the provider's load_users callback to add them.
* ast_phoneprov_delete_extension deletes one extension.
* ast_phoneprov_delete_extensions deletes all extensions for the provider.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/3970/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Improvements to the res_pjsip transport cipher option.
* Made the cipher option accept a comma separated list of OpenSSL cipher
names. Users of realtime will be glad if they have more than one name to
list.
* Added the CLI command 'pjsip list ciphers' so a user can know what
OpenSSL names are available for the cipher option.
* Updated the cipher option online XML documentation to specify what is
expected for the value.
* Updated pjsip.conf.sample to not indicate that ALL is acceptable since
ALL does not imply a preference order for the ciphers and PJSIP does not
simply pass the string to OpenSSL for interpretation.
ASTERISK-24199 #close
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/4018/
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Merged revisions 424393 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424394 65c4cc65-6c06-0410-ace0-fbb531ad65f3
During the latest update to DTLS-SRTP support the ability to configure
the hash used for fingerprints was added. This gave us two supported ones:
SHA-1 and SHA-256. The default was accordingly updated to SHA-256.
Unfortunately this configuration ability was not exposed within res_pjsip.
This change adds a dtls_fingerprint option that controls it.
#SIPit31
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This updates the code behind PJSIP configuration options with custom
handlers to deal with the assigned default values properly where it
makes sense and adjusting the default value where it doesn't. Before
applying this patch, there were several cases where the default value
for an option would prevent that config section from loading properly.
Reported by: Thomas Thompson
Review: https://reviewboard.asterisk.org/r/4019/
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Merged revisions 424263 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
On review /r/3977, it was recommended to note in the
sample configuration about the size limitation for
resource lists. However, since there was no section in
the sample configuration at all for resource list
subscriptions, I decided to make a separate commit
where I have added the necessary sample configuration
as well as the size limitation warning.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422853 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch gives the optional ability to keep queue rules in RealTime. It is
important to note that with this patch:
(a) Queue rules in RealTime are only examined on module load/reload
(b) Queue rules are loaded both from the queuerules.conf file as well as the
RealTime backend
To inform app_queue to examine RealTime for queue rules, a new setting has been
added to queuerules.conf's general section "realtime_rules". RealTime queue
rules will only be used when this setting is set to "yes".
The schema for the database table supports a rule_name, time, min_penalty, and
max_penalty columns. min_penalty and max_penalty can be relative, if a '-' or
'+' literal is provided. Otherwise, the penalties are treated as constants.
For example:
rule_name, time, min_penalty, max_penalty
'default', '10', '20', '30'
'test2', '20', '30', '55'
'test2', '25', '-11', '+1111'
'test2', '400', '112', '333'
'test3', '0', '4564', '46546'
'test_rule', '40', '15', '50'
which would result in :
Rule: default
- After 10 seconds, adjust QUEUE_MAX_PENALTY to 30 and adjust
QUEUE_MIN_PENALTY to 20
Rule: test2
- After 20 seconds, adjust QUEUE_MAX_PENALTY to 55 and adjust
QUEUE_MIN_PENALTY to 30
- After 25 seconds, adjust QUEUE_MAX_PENALTY by 1111 and adjust
QUEUE_MIN_PENALTY by -11
- After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust
QUEUE_MIN_PENALTY to 112
Rule: test3
- After 0 seconds, adjust QUEUE_MAX_PENALTY to 46546 and adjust
QUEUE_MIN_PENALTY to 4564
Rule: test_rule
- After 40 seconds, adjust QUEUE_MAX_PENALTY to 50 and adjust
QUEUE_MIN_PENALTY to 15
If you use RealTime, the queue rules will be always reloaded on a module
reload, even if the underlying file did not change. With the option disabled,
the rules will only be reloaded if the file was modified.
Review: https://reviewboard.asterisk.org/r/3607/
ASTERISK-23823 #close
Reported by: Michael K
patches:
app_queue.c_realtime_trunk.patch uploaded by Michael K (License 6621)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This introduces stasis.conf and a mechanism to prevent certain message
types from being published. Internally, this works by preventing the
chosen message types from being created which ensures that those
message types can never be published. This patch also adjusts message
publishers such that message payloads are not created if the related
message type is not available.
ASTERISK-23943 #close
Review: https://reviewboard.asterisk.org/r/3823/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
