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Old 2008-02-09, 08:26   #21 (permalink)
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Quote:
Originally Posted by charli181 View Post
the telnet function is a test. The asterisk box does not need the telnet service turned on. This is an old IT trick to ensure you can connect to a specific port on a Server where an Application is listenind. This trick can be used for POP3 Servers, SMTP, vitually anything. The command to get it to connect to the right port is as follows:

From a command prompt;

telnet servername portnumber
eg. telnet mypop3server 21

the response back should either be a blank screen or asking for a user name. If you get an error saying "Could not connect on port ... " then you have a problem on the network or the application that is supposed to be listening.
no i am just making it clear we have telnet disabled by our firewalls, this is going across a vpn on the other side of the country, so i just want to recommend that this is not the best way to test a connection... thats it... i know what telnet is and how it works

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As for
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My other question is... How in the read me did u get your picture on the caller id box? Is it an Asterisk thing?
The plugin will look for a file under C:\Program Files\Team MediaPortal\Thumbs\YAC\ with the name the same as the CallerID Name. It may even work with the number (but I don't think so) so for example my telephone number is 1234567 and my callerID Name is Charli181 and I call you, you need to have a picture in the abovementioned folder with the filename of Charli181.jpg or Charli181.png You may have to scale down the size of the picture to get it to fit. If it is too big, the picture will not be seen. There chould be some already in the directory so you can compare the pixel sizes.

sweet thanks!
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Old 2008-02-09, 17:13   #22 (permalink)
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hi fairchild,

a new release is uploaded (see 1st post).
The callerdetails are now also shown, when connected.

For your pause/release issue (scrambled noise): can you send logs, otherwise I cannot work on it.
For blanking the screen while ringing/on call I need to do some research. The MP.Pause function does not do this.
I have a sample picture attached for <unknown> callers (those, who are not in the phonebook). The number as filename for a picture does (currently) not work.

On last word concerning telnet:
telnet cannot be disabled on the network/firewall. If admins say they have disabled telnet, then they usually mean the standard telnet port (23).
For the test we use the Asterisk management port e.g.5038. So the test with telnet will work, otherwise the plugin also cannot work.

Have fun
Gregor
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Old 2008-02-09, 19:00   #23 (permalink)
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hi fairchild,

a new release is uploaded (see 1st post).
The callerdetails are now also shown, when connected.
Great ill try it and let you know

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Originally Posted by GregorV View Post
For your pause/release issue (scrambled noise): can you send logs, otherwise I cannot work on it.
For blanking the screen while ringing/on call I need to do some research. The MP.Pause function does not do this.
I have a sample picture attached for <unknown> callers (those, who are not in the phonebook). The number as filename for a picture does (currently) not work.
Question.. why dont you use your own dir and method for displaying phonebook names and images? Create your own plugin dir and throw an xml and image dir, where you can use a name.jpg/png or number.jpg/png?

Well whats wrong with using a keyboard hook?

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Originally Posted by GregorV View Post
On last word concerning telnet:
telnet cannot be disabled on the network/firewall. If admins say they have disabled telnet, then they usually mean the standard telnet port (23).
For the test we use the Asterisk management port e.g.5038. So the test with telnet will work, otherwise the plugin also cannot work.
alright.. i don't handle our network infrastructure here but i can connect to the system with mediaportal now but telnet still will not see the server, maybe we are doing something special here??!?!, but i tested it and i can telnet my cisco phone and our local managed dell switches no problem... haha
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Old 2008-02-15, 11:48   #24 (permalink)
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Default Resuming Problem

I have a problem with this plugin aspecielly on resume after it have paused. it goes into fast forward, the only way to get out of fastforward is to stop it al toghter and then resume from that

EDIT:
I just found another problem, I can't get the internal extension to work, how exactly should I type the exntension in i have tried both SIP/<ext. no.> aswell as SIP/<ext. name> <ext no.> but I can't get it to ignore outgoing calls
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Last edited by ZeroX; 2008-02-15 at 12:10.
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Old 2008-02-15, 20:09   #25 (permalink)
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Hi ZeroX,

strange, the log looks fine - so I need some more details.
I see, you have MP SVN Build 16208 running. So it may be libs have changed meanwhile. The current AsteriskCID is compiled with the latest MP DLLs as reference. Give me some time to do some research.

The configuration for extensions should be 'SIP/<ext. no.>' so the entries in the box should look like:
SIP/202
SIP/203
...
Each in a separate line. If you look into MediaPortal.xml you should see the following format (my configuration):
<entry name="Extensions">SIP/30, SIP/31, SIP/32, SIP/33, SIP/34, SIP/35, SIP/36, SIP/37, SIP/38, SIP/39</entry>
For exact syntax you also can do a dial from an extension and then check the log. Your sample says: 'no Extension configured SIP/202-09b5a2f8 OK' - so SIP/202 is the correct entry.
Hope this helps.

Gregor
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Old 2008-02-15, 21:35   #26 (permalink)
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Quote:
Originally Posted by GregorV View Post
Hi ZeroX,

strange, the log looks fine - so I need some more details.
I see, you have MP SVN Build 16208 running. So it may be libs have changed meanwhile. The current AsteriskCID is compiled with the latest MP DLLs as reference. Give me some time to do some research.

The configuration for extensions should be 'SIP/<ext. no.>' so the entries in the box should look like:
SIP/202
SIP/203
...
Each in a separate line. If you look into MediaPortal.xml you should see the following format (my configuration):
<entry name="Extensions">SIP/30, SIP/31, SIP/32, SIP/33, SIP/34, SIP/35, SIP/36, SIP/37, SIP/38, SIP/39</entry>
For exact syntax you also can do a dial from an extension and then check the log. Your sample says: 'no Extension configured SIP/202-09b5a2f8 OK' - so SIP/202 is the correct entry.
Hope this helps.

Gregor
I have tested installing the latest SVN and it solved one problem, it no longer pauses on outgoing call!

The fast farward problem is still there! what information do you need from me ?

How can I get it to look in the asterisk phonebook for the CID and display the name isteed?

Last edited by ZeroX; 2008-02-15 at 22:07.
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Old 2008-07-17, 05:36   #27 (permalink)
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I have created a page on the wiki for this plugin now. Any changes or updates should also be made there for ease of documentation

the page is Extensions-Plugins/AsteriskCID - MediaPortal Wiki Documentation
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Old 2008-07-23, 09:57   #28 (permalink)
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Does this work for ISDN only? What do I need to run it, a modem?
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Old 2008-07-23, 11:52   #29 (permalink)
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Originally Posted by Jeinhor View Post
Does this work for ISDN only? What do I need to run it, a modem?
You may want to check out the wiki documentation for this plugin. The link was posted directly above your post before.

Basically, thisplugin id for connection to an Asterisk PBX or TrixBox (google it to find out more)
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Old 2008-07-23, 13:42   #30 (permalink)
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It seems Asterix is for Linux only, so I need a seperate Linux server with Asterix and then I can connect to it from my Media Portal PC?

Would a virtual machine do?
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LinkBack to this Thread: http://forum.team-mediaportal.com/plugins-47/asterisk-callerid-plugin-continues-32870/
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