Commit 9148e24f authored by Etienne Allovon's avatar Etienne Allovon

Merge branch 'master' into 2019.05

parents e44279a4 c7ff2e09
......@@ -28,8 +28,6 @@ Network Flow table (IN) :
+------------------+------------------+----------+------+-----------+----------------+---------+
| asterisk | SIP | UDP | 5060 | 0.0.0.0 | yes | yes |
+------------------+------------------+----------+------+-----------+----------------+---------+
| asterisk | IAX | UDP | 4569 | 0.0.0.0 | yes | yes |
+------------------+------------------+----------+------+-----------+----------------+---------+
| asterisk | SCCP | TCP | 2000 | 0.0.0.0 | yes | yes |
+------------------+------------------+----------+------+-----------+----------------+---------+
| asterisk | AMI | TCP | 5038 | 127.0.0.1 | yes | yes |
......
......@@ -6,9 +6,9 @@ XiVO Solutions |version| Documentation (Callisto Edition)
.. important:: **What's new in this version ?**
* Inbound Call Routing Management
* Outgoing Call Routing Management
* Improvements in Desktop Assistant
* Flashtext
* Flashtext
* MDS Resilience
* Asterisk 16
* Postgres 11
......
......@@ -4,8 +4,12 @@ Upgrade Borealis to Callisto
In this section is listed the manual steps to do when migrating from Borealis to Callisto.
.. warning:: Postgres database will be updated from 9.4 to version 11 during upgrade.
Migration time will therefore be longer than usual.
.. warning:: Upgrade to Callisto:
* Postgres database will be updated from 9.4 to version 11 during upgrade.
Migration time will therefore be longer than usual.
* Asterisk will be updated from 13 to version 16 during upgrade.
* IAX trunks are no longer supported
Before Upgrade
==============
......
......@@ -14,11 +14,10 @@ Interconnections
Create an interconnection
=========================
There are three types of interconnections :
There are two types of interconnections :
* Customized
* SIP
* IAX
Customized interconnections
......@@ -72,7 +71,7 @@ To use your DAHDI links you must create a customized interconnection.
.. warning:: if you use a BRI card you MUST use per-port dahdi groups.
You should not use a group like g0 which spans over several spans.
For example, add an interconnection to the menu :menuselection:`Services --> IPBX --> Trunk management --> Customized` ::
......@@ -115,16 +114,16 @@ Outgoing call caller ID
There are several behavior for the outgoing caller ID.
Use internal caller ID
----------------------
.. Use internal caller ID
.. ----------------------
When you create an outgoing call, it's possible to set it to internal, using the check box in
the outgoing call configuration menu. When this option is activated, the internal caller's caller ID will be
forwarded to the trunk. This option is useful when the other side of the trunk can reach the user
with it's caller ID number.
.. When you create an outgoing call, it's possible to set it to internal, using the check box in
.. the outgoing call configuration menu. When this option is activated, the internal caller's caller ID will be
.. forwarded to the trunk. This option is useful when the other side of the trunk can reach the user
.. with it's caller ID number.
.. figure:: images/outgoing_call_internal.png
:scale: 85%
.. .. figure:: images/outgoing_call_internal.png
.. :scale: 85%
Use outgoing caller ID
......@@ -144,7 +143,7 @@ the *Outgoing Caller ID* option must be set to Customize.
.. figure:: images/user_custom_callerid.png
:scale: 85%
The user can also set his outgoing caller ID to Anonymous.
The user can also set his outgoing caller ID to Anonymous.
If you use a SIP provider trunk, and if your provider supports the RFC3325 for Anonymous calls, you have
to set the Send the Remote-Party-ID option of your SIP trunk to **PAI**:
......
......@@ -101,16 +101,14 @@ The outgoing calls configuration will allow XiVO to know which extensions will
be called through the trunk.
On the call emitting server(s), go on the page :menuselection:`Services
--> IPBX --> Call management --> Outgoing calls` and add an outgoing call.
--> IPBX --> Call management --> Outgoing calls` and add a route.
Tab General::
Trunks: xivo-trunk
Tab Exten::
Exten: **99. (note the period at the end)
Stripnum: 4
Extension: **99. (note the period at the end)
Target: \1
RegExp: .{4}(.*)
This will tell XiVO: if any extension begins with ``**99``, then try to dial it
on the trunk ``xivo-trunk``, after removing the 4 first characters (the ``**99``
......
......@@ -32,7 +32,7 @@ You need the following information from your provider:
* a public phone number
On your XiVO, go on page :menuselection:`Services --> IPBX --> Trunk management -->
SIP Protocol`, and create a SIP/IAX trunk::
SIP Protocol`, and create a SIP trunk::
Name : provider_username
Username: provider_username
......@@ -79,15 +79,12 @@ The outgoing calls configuration will allow XiVO to know which extensions will
be called through the trunk.
Go on the page :menuselection:`Services --> IPBX --> Call management -->
Outgoing calls` and add an outgoing call.
Outgoing calls` and add a route.
Tab General::
Trunks: provider_username
Tab Exten::
Exten: 418. (note the period at the end)
Extension: 418. (note the period at the end)
This will tell XiVO: if an internal user dials a number beginning with ``418``,
then try to dial it on the trunk ``provider_username``.
......
......@@ -237,29 +237,13 @@ management --> Outgoing calls` page :
1. Redirect calls to the PBX :
* Name : fsc-pabx
* Extension : XXXX
* Context : to-pabx
* Trunks : choose the *t2-pabx* interconnection
.. figure:: images/outgoing_call_general.png
:align: center
:scale: 80%
In the extensions tab :
* Exten : XXXX
.. figure:: images/outgoing_call_exten.png
:align: center
:scale: 75%
2. Create a rule "fsc-operateur":
* Name : fsc-operateur
* Extension = X.
* Context : to-extern
* Trunks : choose the "t2-operateur" interconnection
In the extensions tab::
exten = X.
......@@ -6,4 +6,53 @@ Outgoing Calls
You can configure outgoing calls settings in :menuselection:`Services --> IPBX --> Call Management --> Outgoing calls`.
An outgoing call is composed with a **definition** of a outgoing route (the extension patterns that should match) and a **destination** (a trunk owned by a media server).
.. figure:: images/outgoing_route.png
:align: center
:scale: 80%
Definition
----------
A route can define
* **Priority**: A number that will prioritize a route compare to another one, if both routes are available for a same matching extension.
* **Call Pattern**:
* **Extension**: The pattern that will match an extension. More details on `the Asterisk wiki <https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching>`_.
* (advanced) **RegExp**: Transformation to apply on the extension once route is found.
* (advanced) **Target**: Allow to perform filtering like stripping number based on transformation Regexp.
* (advanced) **Caller ID**: Override the presented number once call is performed.
* **Media server**: define the route only for specific media server.
* **Context**: define the route only for specific context.
Here some examples of how you can take advantage of **RegExp** and **Target**:
========== ======== ======================================================================================================================
Regexp Target Result
========== ======== ======================================================================================================================
(.*) \1 get whole called number
(.*) 445 Replace called number with 445 number
14(.*) \1 Delete 14 prefix and keep only following part
14(.*).. \1 Delete 14 prefix and last two digits and keep the rest (e.g. 1455660 鈫 556)
(.*) 33\18 Prefix the called number with 33 and suffix it with 8
14(.*).. 33\18 Delete 14 prefix and last two digits and then prefix the called number with 33 and suffix it with 8 (1455660 鈫 335568)
========== ======== ======================================================================================================================
Destination
-----------
Once route is defined and therefore can be selected if an extension match the route, you need to set the wanted destination.
A route destination is materialized by:
* **Trunk**: SIP or Custom trunk that will be used to reach desired network.
* **Subroutine**: Subroutine to apply once route has been selected.
Rights and schedules
--------------------
A route can define **rights** aka call permissions, it means that a call can be discarded if missing the right to use this route.
The same applies for **schedules** where you can define time slots of availability of the route.
......@@ -100,6 +100,9 @@ Phone
+---------------+--------+----------+-----------+------------------------------------------------------------------------------------------+
| line_protocol | string | Yes | sip, sccp | Line protocol |
+---------------+--------+----------+-----------+------------------------------------------------------------------------------------------+
| line_site | string | | | Unique name of one of the *Template line* (defined in menu :menuselection:`Configuration |
| | | | | --> Provisioning --> Template Line`) |
+---------------+--------+----------+-----------+------------------------------------------------------------------------------------------+
| sip_username | string | | | SIP username |
+---------------+--------+----------+-----------+------------------------------------------------------------------------------------------+
| sip_secret | string | | | SIP secret |
......
......@@ -7,7 +7,8 @@ XiVO Callisto Intermediate Versions
Consult the `2019.04 Roadmap <https://projects.xivo.solutions/versions/133>`_.
.. warning:: This version has the following known problems:
.. warning:: This intermediate version has the following known problems:
* upgrade from Aldebaran won't work,
* you need to create a callright for outgoing call to work (it may be a 'fake' callright applied to nothing),
* also, on MDS servers, you need to add the outcall service in the docker compose yml file for outgoing call to work.
......
......@@ -18,13 +18,21 @@ Callisto.00
New Features
^^^^^^^^^^^^
* High availibility
* CC Manager:
* DB Replic can replicate events from the slave XiVO to the XiVO CC reporting database - see :ref:`ha_interconnection_with_cc`
* System
* Add callbacks count and oldest callbacks
* Desktop Assistant:
* Upgrade to asterisk 16, the latest LTS version of asterisk.
* Upgrade to postgres 11, the latest release of postgres.
* Tray icon shows:
* if user is disconnected
* if user has missed calls
* Improved sound settings for WebRTC
* Windows location and sized is saved when exiting
* UC Assistant:
* Can empty the search box
* Translation: new German translation of Application (Desktop Assistant, UC Assistant, CC Agent and CC Manager)
* WebRTC
* Ability to chose which device (e.g. speaker or headset) will be used when ringing - see :ref:`UC - Ringing Device Selection <uc_webrtc_ringing_device>` or :ref:`CC Agent - Ringing Device Selection <agent_webrtc_ringing_device>`.
......@@ -32,15 +40,23 @@ New Features
* New outgoing calls configuration with *Routes*: more flexible and compatible with XDS sytem - see :ref:`outgoing_calls`.
* Add SRCNUM as available information for FaxToMail application
* WebI: available incoming calls number displayed when creating a new Incoming call (suggestions is limited to the 10 first available results).
* XDS
* Can call a user in different context
* Can call a group located on any MDS
* Can synchronize a device from Webi whatever its MDS
* Can specify the user line site when importing users with a CSV file
* Can specify the user line site when importing users with a CSV file - see :ref:`user_import`.
* Can specify local SIP trunks for a MDS
* Intra-MDS routing SIP peers are auto-generated
* Outgoing call routes can be configured per-MDS
* High availibility
* DB Replic can replicate events from the slave XiVO to the XiVO CC reporting database - see :ref:`ha_interconnection_with_cc`
* System
* Upgrade to asterisk 16, the latest LTS version of asterisk.
* Upgrade to postgres 11, the latest release of postgres.
Behavior Changes
......@@ -50,6 +66,7 @@ Behavior Changes
* Recording server API URL was changed. It is now prefixed with *recording*. For example */records/search* URL
was changed to */recording/records/search*.
* When creating a user using the REST API, the CTI profile is now set to a default value and the CTI client is enabled when a CTI client login and a password is set.
* Fingerboard
* It now runs inside the nginx container and the fingerboard container was removed
......@@ -63,8 +80,16 @@ Behavior Changes
* Database is now run inside a container
* XiVO PBX
* Asterisk: language now defaults to fr_FR.
To change it to english, one should:
* verify that the packages `asterisk-sounds-wav-en-us`, `xivo-sounds-en-us` are installed
* and set, in file :file:`/etc/asterisk/asterisk.conf` the *defaultlanguage* parameter to `en_US`
* **IAX trunks** are no longer supported.
* Outgoing calls were migrated to **Routes**: a more flexible routing system - see :ref:`our migration guide <callisto_route_upgrade_guide>`.
* Web Interface, Groups and Queues configuration: the **Busy** case in the *No answer* tab was removed.
* WebI : user's in select box are now displayed `number@mediaserver [context]` (instead of `number@context`)
* XDS:
* Intra-MDS routing SIP peers are auto-generated: you MUST then remove the peers you would have created manually.
......@@ -92,16 +117,26 @@ Table listing the current version of the components.
+----------------------+----------------+
| Component | current ver. |
+======================+================+
| config-mgt | 2019.05.00 |
+=======================================+
| **XiVO** |
+----------------------+----------------+
| elasticsearch | 1.7.2 |
| XiVO PBX | 2019.05.02 |
+----------------------+----------------+
| config_mgt | 2019.05.02 |
+----------------------+----------------+
| db | 2019.05.02 |
+----------------------+----------------+
| outcall | 2019.05.02 |
+----------------------+----------------+
| db_replic | 2019.05.00 |
+----------------------+----------------+
| fingerboard | 2019.05.00 |
| **XiVO CC** |
+----------------------+----------------+
| elasticsearch | 1.7.2 |
+----------------------+----------------+
| kibana_volume | 2019.05.00 |
+----------------------+----------------+
| nginx | 2019.05.00 |
| nginx | 2019.05.02 |
+----------------------+----------------+
| pack-reporting | 2019.05.00 |
+----------------------+----------------+
......@@ -109,21 +144,90 @@ Table listing the current version of the components.
+----------------------+----------------+
| recording-rsync | 1.0 |
+----------------------+----------------+
| recording-server | 2019.05.00 |
| recording-server | 2019.05.02 |
+----------------------+----------------+
| spagobi | 2019.05.00 |
+----------------------+----------------+
| xivo-db-replication | 2019.05.00 |
+----------------------+----------------+
| xivo-full-stats | 2019.05.00 |
+----------------------+----------------+
| XiVO PBX | 2019.05.00 |
+----------------------+----------------+
| xuc | 2019.05.00 |
| xuc | 2019.05.02 |
+----------------------+----------------+
| xucmgt | 2019.05.00 |
| xucmgt | 2019.05.02 |
+----------------------+----------------+
Callisto.02
-----------
Consult the `Callisto.02 Roadmap <https://projects.xivo.solutions/versions/143>`_.
Components updated: **config-mgt**, **nginx**, **recording-server**, **xivo-confgend**, **xivo-dao**, **xivo-db**, **xivo-manage-db**, **xivo-outcall**, **xivo-web-interface**, **xivocc-installer**, **xucmgt**, **xucserver**
**Desktop Assistant**
* `#2212 <https://projects.xivo.solutions/issues/2212>`_ - Empty the search box
**Recording**
* `#2488 <https://projects.xivo.solutions/issues/2488>`_ - Update login page to have same look and feel than ccagent or cccmanager and display logged username
**Web Assistant**
* `#2506 <https://projects.xivo.solutions/issues/2506>`_ - XDS - status of phone is randomly correct on UC
* `#2507 <https://projects.xivo.solutions/issues/2507>`_ - display flashtext from other users
**XUC Server**
* `#2470 <https://projects.xivo.solutions/issues/2470>`_ - ACD outbound call status is always dialing
* `#2505 <https://projects.xivo.solutions/issues/2505>`_ - Add username to RichDirectoryResult in XUC
**XiVO PBX**
* `#2458 <https://projects.xivo.solutions/issues/2458>`_ - XDS - Improve postgresql configuration handling
* `#2473 <https://projects.xivo.solutions/issues/2473>`_ - XDS - mds installation may fail when configuring uuid
* `#2498 <https://projects.xivo.solutions/issues/2498>`_ - Deleting trunk used in outcall causes outcall not to be
* `#2500 <https://projects.xivo.solutions/issues/2500>`_ - Postgres in docker is always restarting in auto recovery mode
* `#2504 <https://projects.xivo.solutions/issues/2504>`_ - Increase default number of connection in db container
* `#2512 <https://projects.xivo.solutions/issues/2512>`_ - Outcall - no group id for user causes sql group query to fail
**XiVOCC Infra**
* `#2383 <https://projects.xivo.solutions/issues/2383>`_ - XiVO CC services can't use domain names
Callisto.01
-----------
Consult the `Callisto.01 Roadmap <https://projects.xivo.solutions/versions/141>`_.
Components updated: **asterisk**, **xivo-config**, **xivo-db**, **xivo-monitoring**, **xivo-outcall**, **xivo-solutions-doc**, **xivo-upgrade**, **xivo-web-interface**, **xucmgt**
**Desktop Assistant**
* `#2481 <https://projects.xivo.solutions/issues/2481>`_ - Save desktop assistant windows location and size on exit
**WebRTC**
* `#2389 <https://projects.xivo.solutions/issues/2389>`_ - Optimize Chrome WebRTC settings
**XiVO PBX**
* **Asterisk**: Update asterisk to 16.3.0 `#2483 <https://projects.xivo.solutions/issues/2483>`_
* `#2465 <https://projects.xivo.solutions/issues/2465>`_ - Asterisk 16 - Voicemail supervision doesn't work
* `#2362 <https://projects.xivo.solutions/issues/2362>`_ - XDS - Database schema is shown as NOK after upgrade
* `#2453 <https://projects.xivo.solutions/issues/2453>`_ - Outcall - application - consider context inclusion
* `#2460 <https://projects.xivo.solutions/issues/2460>`_ - Outcall - make intra-mds call routing work for other contexts than default
* `#2463 <https://projects.xivo.solutions/issues/2463>`_ - Outcall - migration to Route - migration script creates routes with outgoing context
* `#2464 <https://projects.xivo.solutions/issues/2464>`_ - Outcall - application does not reconnect to database
* `#2477 <https://projects.xivo.solutions/issues/2477>`_ - Permissions not respected when calling a forwared user
* `#2478 <https://projects.xivo.solutions/issues/2478>`_ - Asterisk 16 - Not logging CEL if database is not ready when asterisk starts
* `#2479 <https://projects.xivo.solutions/issues/2479>`_ - Outcall - Callerid and forward - Wrong callerid when U1 calls U2 fwded to external user
* `#2484 <https://projects.xivo.solutions/issues/2484>`_ - Route - I should be able to create a route with prio > 10
* `#2489 <https://projects.xivo.solutions/issues/2489>`_ - Clean outcall from contextmember table
* `#2491 <https://projects.xivo.solutions/issues/2491>`_ - Bypass schedule with password doesn't work for outgoing calls
Callisto Intermediate Versions
==============================
......
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment