Asterisk CallerID plugin continues (1 Viewer)

charli181

Retired Team Member
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  • August 3, 2007
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    Ok - I know that I am digging up an old thread here, but I love my asterisk (PBX In A Flash flavour) and since I want this to work with the current versions of MP, I am currently in the process of building this plugin from scratch again. Rasons for this is that the source is not available and secondly, I am doing it with VB.Net instead.

    Planned features for release one will be

    Notification for inbound calls on a single extension
    Mailbox Notification for a single mailbox.

    Future planned releases
    Ability to setup Do Not Disturb & call forwarding via GUI.
    Mailbox message retrieval/playback via MP GUI
    Possibly SIP Phone GUI - Voice Only.


    Ths will done through the Asterisk API which allows for more control than the above suggested YAC which i think is now classed as unsupported anyways.

    I will create a new thread when the first release is ready, but if anyone has any other ideas to be added in, let me know.:D

    Sean
     

    MageMinds

    Portal Pro
    January 29, 2007
    99
    5
    Canada Canada
    Count me in to test it too ... YAC works kind of good, but sometimes calls doesn't show. I did not investigate though.

    I use callerid superfecta, make sure you notify that caller id. My SIP provider doesn't supply the CID Name, I need Superfecta to lookup for a name before presenting it to the phone/MP.

    MageMinds
     

    sircolin

    Portal Member
    March 17, 2011
    14
    1
    45
    Reading UK
    United Kingdom United Kingdom
    First Off thanks for taking the time to publish your work.

    Second it's not working for me :-(

    Im using pbxiaf silver and the lastest non svn version of media portal, i have installed the plugin and added an api entry, your module authenticates ok but i still see no popup from within media portal when i place a call to my extenstion 1000

    From asterisk i have
    pbx*CLI> manager show connected
    Username IP Address
    MediaPortal 83.xxx.xx.241
    admin 127.0.0.1
    pbx*CLI> manager show eventq
    Usecount: 1
    Category: 2
    Event:
    Event: Newstate
    Privilege: call,all
    Channel: SIP/1000-00ed7640
    State: Ringing
    CallerID: 1000
    CallerIDName: <unknown>
    Uniqueid: 1300345357.59
    it seems to me that the module stays connected ok until a call comes through at which point MediaPortal (manager) seems to log off asterisk until the call is complete, and i'm left with just the admin manager logged on from localhost.

    i have played with the polling timings 30 45 60 seconds and this seems to change nothing.

    Since the media Portal can tell me when it has connected and disconnected to the api i conclude that voicemail notification is broken on this app mp (1.1.3.0) since it is unable to notify me of a voicemail waiting in ext 1000's mail box when it connects !
    Can anyone confirm this works with media portal 1.1.3.0?

    I have tried about everything i can think of.... any Ideas ?

    Col
     

    charli181

    Retired Team Member
  • Premium Supporter
  • August 3, 2007
    794
    106
    Sydney
    Australia Australia
    Country flag
    First Off thanks for taking the time to publish your work.

    Second it's not working for me :-(

    Im using pbxiaf silver and the lastest non svn version of media portal, i have installed the plugin and added an api entry, your module authenticates ok but i still see no popup from within media portal when i place a call to my extenstion 1000

    From asterisk i have
    pbx*CLI> manager show connected
    Username IP Address
    MediaPortal 83.xxx.xx.241
    admin 127.0.0.1
    pbx*CLI> manager show eventq
    Usecount: 1
    Category: 2
    Event:
    Event: Newstate
    Privilege: call,all
    Channel: SIP/1000-00ed7640
    State: Ringing
    CallerID: 1000
    CallerIDName: <unknown>
    Uniqueid: 1300345357.59
    it seems to me that the module stays connected ok until a call comes through at which point MediaPortal (manager) seems to log off asterisk until the call is complete, and i'm left with just the admin manager logged on from localhost.

    i have played with the polling timings 30 45 60 seconds and this seems to change nothing.

    Since the media Portal can tell me when it has connected and disconnected to the api i conclude that voicemail notification is broken on this app mp (1.1.3.0) since it is unable to notify me of a voicemail waiting in ext 1000's mail box when it connects !
    Can anyone confirm this works with media portal 1.1.3.0?

    I have tried about everything i can think of.... any Ideas ?

    Col
    Firstly, Thanks for trying out my plugin. I am hoping you are using the plugin I created and uploaded in the new thread Asterisk2MediaPortal Plugin
    I posted above and not the first post in this thread, if so, this report should go there so it does not get confusing for people still using this original plugin.

    I will investigate this but the event I am currently looking for from the API link is an "Event: Dial" with "Dialstring :" +your monitored extension. This maybe not the correct event to look for. My testing involved 2 asterisk boxes connected via a IAX trunk so the events maybe different from a FXO card. I will get back to you in the other thread for a resolution on this.

    Also if you can supply the mediaportal.log as this plugin does log to it. you can search for "Ast2MP" in the log for plugin events. A plugin log event is raised for an incoming call looking like

    "Plugin :- Ast2MP - Incoming Call Alert - callerID callerName"

    before the notify dialog is shown within mp GUI. this should confirm that the plugin is recognising the incoming call and it is just a matter of it not calling the dialog box within MP GUI.

    Sean
     

    sircolin

    Portal Member
    March 17, 2011
    14
    1
    45
    Reading UK
    United Kingdom United Kingdom
    Thanks For your reply charli181

    I Am using your plugin ;-) the one above

    also for your information I run pure sip and iax no pstn here. the extension (1000) i wish to use will be sip and would be nice at the later date to change the string to involve a ring group 800 (1000+1001)

    can i adjust the string myself ?

    Let me know if i can be of any assistance on this !

    Col
     

    charli181

    Retired Team Member
  • Premium Supporter
  • August 3, 2007
    794
    106
    Sydney
    Australia Australia
    Country flag
    Thanks For your reply charli181

    I Am using your plugin ;-) the one above

    also for your information I run pure sip and iax no pstn here. the extension (1000) i wish to use will be sip and would be nice at the later date to change the string to involve a ring group 800 (1000+1001)

    can i adjust the string myself ?

    Let me know if i can be of any assistance on this !

    Col
    No probs Col, I just tested with 1.1.3 and it works with mine and get a popup straight away with the other extensions callidnum and name. again this is extn to extn within 1 PIAF system this time. I will look into ring groups as well. Tomorrow (9pm here now - wife getting a bit upset I am still on PC :rolleyes:), I will post a document to manually see the events coming from the API using telnet and the command used to check the voicemail - maybe your version of PIAF is ealier/later than mine and does not support the commands I am using on API.
    Sean
     

    sircolin

    Portal Member
    March 17, 2011
    14
    1
    45
    Reading UK
    United Kingdom United Kingdom
    Thanks Sean

    maybe your version of PIAF is ealier/later than mine
    Could be im use pbxiaf silver which i compiled and upped to the pbxiaf repo

    I have

    Status Version 1.2.9 released on Date 042310
    ********************************************************************
    * PBX in a Flash Version Daemon Status *
    * Running Asterisk 1.4 *
    ********************************************************************
    * Asterisk * ONLINE * Zap/Dahdi * N/A * MySQL * ONLINE *
    * SSH * ONLINE * Apache * ONLINE * Iptables * OFFLINE *
    * Fail2ban * ONLINE * IP Connect* ONLINE * Ip6tables * OFFLINE *
    * BlueTooth * OFFLINE * Hidd * OFFLINE * NTPD * OFFLINE *
    * Sendmail * ONLINE * Samba * OFFLINE * Webmin * ONLINE *
    * Ethernet0 * N/A * Ethernet1 * N/A * Wlan0 * N/A *
    ********************************************************************
    * Running Asterisk Version : Asterisk 1.4.26.2
    * Asterisk Source Version : 1.4.26.2
    * Zap/Dahdi Source Version : N/A
    * Libpri Source Version : UNAVAILABLE
    * Addons Source Version : 1.4.9
    ********************************************************************
    pbx.berkshirecomputing.info on - eth0
    CentOS release 5.2 (Final) :64 Bit Kernel: 2.6.32-4-pve
    ********************************************************************
    For help on PBX commands than you can run type help-pbx *
    ********************************************************************
    I better not keep you ;-)

    Night

    Col

    Hi Sean,

    I have done 2 logs for completeness they are here

    Not much in them thats unexpected but hey see what you think.

    Col
     

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