Merge storage for the stats object and name string into the main
allocation for struct ast_taskprocessor.
Change-Id: I74fe9a7f357f0e6d63152f163cf5eef6428218e1
* Ignore console=yes configuration option in remote console processes.
* Use new flag to tell consolethread to run el_end and exit when needed.
ASTERISK-28158
Change-Id: I9e23b31d4211417ddc88c6bbfd83ea4c9f3e5438
pack_string crashed on non-NULL strings returned when s->has_error was
true if the string was the result of 's' format without '#', '%' or '+'.
Change-Id: Ic125df691d81ba2cbc413e37bdae657b304d20d0
When Asterisk's taskprocessors get overloaded we need to reduce the work
load. res_pjsip currently ignores new SIP requests and relies on SIP
retransmissions in the hope that the overload condition will clear soon
enough to handle the retransmitted SIP request.
This change adds the following code after ast_taskprocessor_alert_get()
has returned TRUE:
1- identifies transport type. If non-udp then send a 503 response
2- if transport type is udp/udp6 then ignore, as before.
Change-Id: I1c230b40d43a254ea0f226b7acf9ee480a5d3836
The use of a '|' in the "global/debug" synopsis documentation caused the
generated html table on the wiki to add an extra column that included the
text after the pipe.
This patch replaces the pipe with a comma.
ASTERISK-28150
Change-Id: I3d79a6ca6d733d9cb290e779438114884b98a719
The current round-robin method does not take the current taskprocessor
load into consideration when distributing requests. Using the least-size
method the request goes to the taskprocessor that is servicing the least
number of active tasks at the current time.
Longer running tasks with the round-robin method can delay processing
tasks.
* Change the algorithm from round-robin to least-size for picking the
PJSIP taskprocessor from the default serializer pool.
Change-Id: I7b8d8cc2c2490494f579374b6af0a4868e3a37cd
As mentioned in the comment I've added in the code there is no
ability to unsubscribe all subscribers from a topic and explicitly
destroy it. This is not currently a problem as we have two types of
topics:
Long lived topics which exist for the lifetime of the system.
Ephemeral topics which feed a long lived topic.
In the case of the ephemeral topics there is no subscriber which does
not have its lifetime managed by the same entity that has created
the topic. This ensures that when the topic is being unreferenced the
subscribers are also unsubscribed and destroyed, allowing the topic
to ultimately be destroyed as well.
Change-Id: Ic5e244da7b16b1895ba1fc5ece481ebba5809c9a
When a call pickup is performed using and invite with replaces header
the ast_do_pickup method is attempted and a PICKUP stasis message is sent.
ASTERISK-28081 #close
Reported-by: Luit van Drongelen
Change-Id: Ieb1442027a3ce6ae55faca47bc095e53972f947a
Using the --quiet or -q option in conjonction with /dev/stdout as the output
file allow the output to be used as a valid configuration.
Given a script that generates a valid sip.conf I can pipe the output of that
script into `sip_to_pjsip.py -q /dev/stdin /dev/stdout`. This allow me to use
that piped command in my pjsip.conf using the `exec` command.
ASTERISK-28136
Change-Id: I7b0e2e90e2549f3f8e01dc96701f111b5874c88d
This patch adds new options 'trust_connected_line' and 'send_connected_line'
to the endpoint.
The option 'trust_connected_line' is to control if connected line updates
are accepted from this endpoint.
The option 'send_connected_line' is to control if connected line updates
can be sent to this endpoint.
The default value is 'yes' for both options.
Change-Id: I16af967815efd904597ec2f033337e4333d097cd
The module 'res_pjsip_notify' inefficiently makes a lot of DB requests
on CLI completion on the endpoint.
For example if there are 10k endpoints the module makes 10k requests
of these 10k records.
Even if a part of the endpoint entered
the module makes the same 10k requests and then filtered them by itself.
This patch gathers endpoints container by prefix
and adds all gathered endpoints to completion at once.
ASTERISK-28137 #close
Change-Id: Ic20024912cc77bf4d3e476c4cd853293c52b254b
Given a sip.conf with the following content:
setvar FOO=1
setvar BAR=42
I want my generated pjsip.conf to containt the following set_vars
set_var FOO=1
set_var BAR=42
in the matching endpoint section.
Change-Id: I6c822401fda4133c3b44bf31e655b4eb939d4d26
Add a new global flag to res_pjsip to allow the callerid to be used
as the username in the contact header. This allows chan_pjsip to have
the same behavour as chan_sip
ASTERISK-28087 #close
Change-Id: I9a720e058323f6862a91c62f8a8c1a4b5c087b95
Default logging was not setup correctly when there was no logger.conf.
This resulted in many expected log messages not actually getting out to
the console.
Change-Id: I542e61c03b2f630ff5327f9de5641d776c6fa70c
The 'I' option currently blocks initial CONNECTEDLINE or REDIRECTING updates
from the called parties to the caller.
This patch also blocks updates in the other direction before call is
answered.
ASTERISK-27980
Change-Id: I6ce9e151a2220ce9e95aa66666933cfb9e2a4a01
Adding the "label" attribute used for participant info correlation
was previously done in app_confbridge but it wasn't working
correctly because it didn't have knowledge about which video
streams belonged to which channel. Only bridge_softmix has that
data so now it's set when the bridge topology is changed.
ASTERISK-28107
Change-Id: Ieddeca5799d710cad083af3fcc3e677fa2a2a499
These macros have been documented as legacy for a long time but are
still used in new code because they exist. Remove all references to:
* ao2_container_alloc_options
* ao2_t_container_alloc_options
* ao2_t_container_alloc
These macro's are still available for use but only in modules. Only
ao2_container_alloc remains due to it's use in over 100 places.
Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a
__ast_mutex_logger used the variable `canlog` without accepting it as a
argument. Replace with internal macro `log_mutex_error` which takes
canlog as the first arguement. This will prevent confusion when working
with lock.c code, many of the function declare the canlog variable and
in some cases it previously appeared to be unused.
Change-Id: I83b372cb0654c5c18eadc512f65a57fa6c2e9853
* Update the post-answer documentation and example. The Dial example was
incorrect and misleading for the post-answer subroutine useage.
* Fix note and warning paragraphs in option descriptions. They don't show
up in the wiki.
Change-Id: I81019a1fd75d5b9151f76b52c38e2a90da682d14
Most were in comments. A couple were in warning messages.
Pointed out by Jonathan H on the Asterisk users mailing list.
Change-Id: I6286939dff5d0a27a2758140570106f1cb351855
* The dependency ensures that res_pjproject cannot be manually unloaded
before res_rtp_asterisk.
* The dependency allows startup loading errors to report that
res_rtp_asterisk depends upon res_pjproject.
Change-Id: Icf5e7581f4ddd6189929f6174c74dd951f887377
Add attribute_warn_unused_result to ast_taskprocessor_push,
ast_taskprocessor_push_local and ast_threadpool_push. This will help
ensure we perform the necessary cleanup upon failure.
Change-Id: I7e4079bd7b21cfe52fb431ea79e41314520c3f6d
Joshua Colp
George Joseph
Corey Farrell
Kevin Harwell
Chris-Savinovich
Alexei Gradinari
Jasper Hafkenscheid
Pascal Cadotte Michaud
Richard Mudgett
Torrey Searle
Jenkins2
Sean Bright