home
products
contribute
download
documentation
forum
Home
Forums
New posts
Search forums
What's new
New posts
All posts
Latest activity
Members
Registered members
Current visitors
Donate
Log in
Register
What's new
Search
Search
Search titles only
By:
New posts
Search forums
Search titles only
By:
Menu
Log in
Register
Navigation
Install the app
Install
More options
Contact us
Close Menu
Forums
MediaPortal 1
MediaPortal 1 Plugins
Asterisk CallerID plugin continues
Contact us
RSS
JavaScript is disabled. For a better experience, please enable JavaScript in your browser before proceeding.
You are using an out of date browser. It may not display this or other websites correctly.
You should upgrade or use an
alternative browser
.
Reply to thread
Message
<blockquote data-quote="GregorV" data-source="post: 216607" data-attributes="member: 57522"><p>Hi,</p><p></p><p>The Asterisk CallerID plugin continues !</p><p>Based on Troky's work I decided to complete his announced features and added some new ones.</p><p>This is my first attempt to code for the great MediaPortal, so don't be too hard, if things do not work as intended.</p><p></p><p></p><p>How it works:</p><p>A incoming call will display a message with CallerName, CID and optional picture (if it exists).</p><p>The plugin connects with the Asterisk via the Asterisk Call Manager API.</p><p></p><p>How to install:</p><p>Copy the DLL's 'ACID.dll' and 'Asterisk.NET.dll' into the '\plugins\process' folder.</p><p></p><p>How to configure:</p><p>Asterisk:</p><p></p><p>In manager.conf we need a user - e.g. media:</p><p></p><p>[media] // -- in AsteriskCID: Username:</p><p>secret = pwmedia // -- in AsteriskCID: Secret:</p><p>deny=0.0.0.0/0.0.0.0</p><p>permit=192.168.0.1/255.255.255.0 // -- the IP address from media pc running MediaPortal</p><p>read = system,call,log,verbose,command,agent,user</p><p>write = system,call,log,verbose,command,agent,user</p><p></p><p>Also make sure that the general settings do exist and the API is enabled:</p><p></p><p>[general]</p><p>enabled = yes</p><p>port = 5038 // -- in AsteriskCID: the field behind Asterisk server IP:</p><p>bindaddr = 0.0.0.0</p><p></p><p>Trixbox users can go to FreePBX->Tools->Asterisk API to create a user.</p><p>Note: the settings require Asterisk to be restarted, Trixbox users don't need a restart.</p><p></p><p>AsteriskCID plugin:</p><p></p><p>Asterisk server IP:</p><p>the IP address of the Asterisk server and the port configured in manager.conf</p><p>Username:</p><p>the user name configured in manager.conf (Asterisk)</p><p>Secret:</p><p>the secret configured in manager.conf (Asterisk)</p><p></p><p>Debug Level:</p><p>log level 0=info, 2=details, 3=debug</p><p></p><p>Incoming channel/trunk: e.g.: SIP/ISDN123</p><p>This can be used to filter incoming calls to show a message only, if it came in at that trunk.</p><p>Each channel/trunk must be configured in one line.</p><p>No entry will allow messages on all trunks.</p><p></p><p>Internal extensions: e.g.: SIP/30</p><p>This can be used to filter messages on internal calls and outgoing calls</p><p>Each extension must be configured in one line.</p><p>No entry will allow messages on all extensions.</p><p></p><p>Stop media when call detected:</p><p>check, if media should be paused on incomming calls</p><p>Resume media after message closed:</p><p>check, if media should resume on call established -or-</p><p>Resume media after call released:</p><p>check, if media should resume after call is finished</p><p>Notify timeout:</p><p>Timeout (seconds) after which message box is closed and media is resumed</p><p>This may override the 'resume' setting (whichever occurs first), 0 = NO timeout.</p><p></p><p>Remove leading digit</p><p>removes the first digit from the CID if it matches the configured digit (0 or 9). For an Asterisk behind setup.</p><p></p><p>Hint:</p><p>Audio may resume delayed (3-8sec.) while video is resumed immediately. Still not found the cause, it seems to be an MP issue.</p><p>Try it and send me your findings.</p><p>The configuration is stored in MediaPortal.xml. If you want to edit trunks or extensions there, make sure they are seperated by ', ' (comma and space). Do not edit the secret entry here - it is encoded and may cause an exception.</p><p></p><p>Have fun and report any found issues or wishes.</p><p></p><p>Edit:</p><p>v0.2.1.1 Message lock fixed, message 'Connected to' now configurable - see ReleaseNotes.txt.</p><p>v0.2.1.2 Connected message now also displays caller details</p><p>v0.2.1.4 for RC2</p><p></p><p>GregorV</p></blockquote><p></p>
[QUOTE="GregorV, post: 216607, member: 57522"] Hi, The Asterisk CallerID plugin continues ! Based on Troky's work I decided to complete his announced features and added some new ones. This is my first attempt to code for the great MediaPortal, so don't be too hard, if things do not work as intended. How it works: A incoming call will display a message with CallerName, CID and optional picture (if it exists). The plugin connects with the Asterisk via the Asterisk Call Manager API. How to install: Copy the DLL's 'ACID.dll' and 'Asterisk.NET.dll' into the '\plugins\process' folder. How to configure: Asterisk: In manager.conf we need a user - e.g. media: [media] // -- in AsteriskCID: Username: secret = pwmedia // -- in AsteriskCID: Secret: deny=0.0.0.0/0.0.0.0 permit=192.168.0.1/255.255.255.0 // -- the IP address from media pc running MediaPortal read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Also make sure that the general settings do exist and the API is enabled: [general] enabled = yes port = 5038 // -- in AsteriskCID: the field behind Asterisk server IP: bindaddr = 0.0.0.0 Trixbox users can go to FreePBX->Tools->Asterisk API to create a user. Note: the settings require Asterisk to be restarted, Trixbox users don't need a restart. AsteriskCID plugin: Asterisk server IP: the IP address of the Asterisk server and the port configured in manager.conf Username: the user name configured in manager.conf (Asterisk) Secret: the secret configured in manager.conf (Asterisk) Debug Level: log level 0=info, 2=details, 3=debug Incoming channel/trunk: e.g.: SIP/ISDN123 This can be used to filter incoming calls to show a message only, if it came in at that trunk. Each channel/trunk must be configured in one line. No entry will allow messages on all trunks. Internal extensions: e.g.: SIP/30 This can be used to filter messages on internal calls and outgoing calls Each extension must be configured in one line. No entry will allow messages on all extensions. Stop media when call detected: check, if media should be paused on incomming calls Resume media after message closed: check, if media should resume on call established -or- Resume media after call released: check, if media should resume after call is finished Notify timeout: Timeout (seconds) after which message box is closed and media is resumed This may override the 'resume' setting (whichever occurs first), 0 = NO timeout. Remove leading digit removes the first digit from the CID if it matches the configured digit (0 or 9). For an Asterisk behind setup. Hint: Audio may resume delayed (3-8sec.) while video is resumed immediately. Still not found the cause, it seems to be an MP issue. Try it and send me your findings. The configuration is stored in MediaPortal.xml. If you want to edit trunks or extensions there, make sure they are seperated by ', ' (comma and space). Do not edit the secret entry here - it is encoded and may cause an exception. Have fun and report any found issues or wishes. Edit: v0.2.1.1 Message lock fixed, message 'Connected to' now configurable - see ReleaseNotes.txt. v0.2.1.2 Connected message now also displays caller details v0.2.1.4 for RC2 GregorV [/QUOTE]
Insert quotes…
Verification
Post reply
Forums
MediaPortal 1
MediaPortal 1 Plugins
Asterisk CallerID plugin continues
Contact us
RSS
Top
Bottom