Asterisk CallerID plugin (1 Viewer)

ZeroX

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October 26, 2005
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YES YES!!!

I'm been waiting for this for a long time!!! THANK YOU.
 

troky

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July 30, 2007
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I will download when availalble for the debug version. I will test this with my live asterisk box.

I do have another asterisk box in my labs that I will test on as well. If need be, I will upgrade asterisk to your version and test again as well but prefer not to due to reconfiguring on upgrades......ohhh the pain.

HERE it is. Just enable extensive logging in plugin configuration and try to make some calls. All asterisk incoming event will be logged. You can send the log to troky2001_at_yahoo.com.

I think that older versions of asterisk have slightly different event descriptions and that's why detection fails.
Don't install new version of asterisk. It will be much easier that I adapt plugin to work with your version ;)

Regards,
Troky
 

charli181

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    Great stuff.....leave it with me. I will test tonight. 8pm here and family watching movies at the moment so can't work on it. I will post the logs later tonight. If it is too much coding, I can upgrade asterisk. The upgrade may fix an issue I have with it.
     

    charli181

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    here we go.....

    I have attached a zip file. In this it includes the mediaportal log file and also a log file of a telnet session directly to the astersik box for comparison. I do notice that the "status:rsrvd" is not there. Definitly looks like a version thing on asterisk side.

    the call flow for my systems is that all incoming calls are directed to a ring group, This ring group rings 2 different extensions at the same time. I did answer the call in the log and then hung up.

    let me know if this is too hard to code and I will upgrade asterisk. This could just be a clause in your plugin as minimum required astersik level.:D
     

    troky

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    July 30, 2007
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    Ok, I've found the problem... There are different events generated for SIP incoming calls (your case) and trunk/ISDN calls (my case).

    Now I've included both events in detection. If you get many unwanted pop-ups just enter 'channel' filter. You can enter just few firs letters of channel name there... something like "SIP/1" to catch all channels that start with "SIP/1" (SIP/123, SIP/1111...).

    If you still get unwanted popups report all details so I can think of new filter/configuration possibilities.

    Version 0.0.1.4 HERE

    Regards,
    Troky
     

    charli181

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    :D :D :D awesome stuff troky..... All is working great.

    I now get the popup on a sip trunk or the pstn(fxo) port with CLI.

    I know I am a pain in the @rse, but with all the options set, when the phones are ringing , the video is paused. When I answer the call, the video starts playing again but with no sound. after I hang up I still have the same symptoms of no sound. To get sound back I just pause (space bar) and then resume (space bar again).

    Is this normal or is the video supposed to pause until the call is completed (i.e. hung up).

    attached is the new log file.
     

    troky

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    July 30, 2007
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    I am experiencing similar 'no sound' issues, too... after resuming video sound is gone for few seconds and then it comes back...

    I'll now concentrate on this issue.

    BTW... you can put caller images in MP/Thumbs/YAC folder in format <callerid name>.png (or jpg I think)
    for unknown calleridname "_noImage.png" is displayed and for unknown number "_unknown.png"


    Regard,
    Troky
     

    charli181

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    once again, thanks very much for this plugin. My wife and friends are awe stuck over this feature within MP.
     

    charli181

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    Troky,

    I have just got back home after a couple of weeks away. I have 2 requests based on my wifes suggestions :

    1 - Can the call notification be configured to disappear after answering the call instead of releasing the call. If one of us has a "D&M" on the phone that goes for hours, I do not really want to see this on the screen all this time until the call is released.

    2 - I have found when an outbound call is made, I get the notification as well. I do not have a filter setup as I have a ZAP PSTN trunk and also a SIP Trunk and receive calls on both. The only other thing I can think of is setting up a sip phone on a server that is never used and placing the filter to that number. Can you think of any other way.

    On a side note, I have very many interest friends and collegues that hear about this. I work in the Comms industry and although they class this as over the top, are still very impressed.
     

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