Asterisk CallerID plugin continues

[GregorV]

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

GregorV

[urev]

cool,
but is your plugin compatible with the yac-format? Since the AVM Fritzbox can be enabled to send out caller-ids in yac-format, this would be cool also for users of the old yac-plugin, which displays Namen, Number and Picture, but can not be configured to pause the played media.

bye, urev

[GregorV]

Originally Posted by urev

cool,
but is your plugin compatible with the yac-format? Since the AVM Fritzbox can be enabled to send out caller-ids in yac-format, this would be cool also for users of the old yac-plugin, which displays Namen, Number and Picture, but can not be configured to pause the played media.

bye, urev

Hi urev,

no it does'nt listen for YAC server. If there are more requests I'll think about that. I the YAC plugin there are some specific things, that cannon be tested outside US and without a modem - so it is probably not easy to make sure that all features will work.

GregorV

[charli181]

I will test this over the christmas break and let you know how the new version goes. Thanks.

[scastano]

I've got the plugin installed, the issue I'm having now is it only displays calls the FIRST time they come in.

After, that it doesn't display anything, and when I check in the logs it says something about "skip event"

any idea's there??

Other then that.... SUCH A GREAT IDEA.... it works really well.... everyone loves it!! Thanks man, this rocks!

*SMC

[GregorV]

Hi scastano,

this is what I have experienced also during the last days. Probably something appearing with latest SVNs.
I will look into it over the week end and fix it.
Are you using SVNs or stable release ?

Gregor

[scastano]

I'm using the stable release...

Trixbox Version 2.2.4
Core Version 2.3.1.3
Framework Version 2.3.1.3
Asterisk Version 1.2.23
Asterisk API Module Version 1.3.1

If you need anything else, copies of call manager sessions, logs from trix, asterisk or CID, let me know!

Thanks again!!

*SMC

[GregorV]

Hi scastano,

a new release is uploaded - check first post.
Found some scenarios that could lock messages on new calls.
Let me know if that works for you.

Gregor

[scastano]

PERFECT....I'm away for the weekend right now... but as soon as I get home late Sunday night/Monday morning I'll give it a shot and let you know how it goes!

Thanks again!

*SMC

[scastano]

Took me ALOT longer to get home yesterday then I thought.... but I just got the new version up and running and it works GREAT!!!

I tried to set it to NOT pause the media when the notification box pops up, but it seems to do that no matter what setting I choose.... but the call recognition works perfectly now.

*SMC