AsteriskNow call redirect with different callerID
I have installed latest version of AsteriskNow.
Trying to setup a simulator system with 2 IP phones.
ext 100 and 200
When ext 200 dials example: 135 the call must be redirected to ext 100 with the callerID name and ext no from Ext 135.
By using FreePBX and adding a extension (135) with Other (Custom) Device settings I can set the "This device uses custom technology" to SIP/100
I can now from ext 200 dial 135 and beeing redirected to ext 100.
Problem is that the name and number from ext 200 is displayed in ext 100 when calling.
What I like to have displayed is 135 and the name from ext 135.
on the ext 200 when calling.
Any idea how do this? Any help is highly appreciated.
Trying to setup a simulator system with 2 IP phones.
ext 100 and 200
When ext 200 dials example: 135 the call must be redirected to ext 100 with the callerID name and ext no from Ext 135.
By using FreePBX and adding a extension (135) with Other (Custom) Device settings I can set the "This device uses custom technology" to SIP/100
I can now from ext 200 dial 135 and beeing redirected to ext 100.
Problem is that the name and number from ext 200 is displayed in ext 100 when calling.
What I like to have displayed is 135 and the name from ext 135.
on the ext 200 when calling.
Any idea how do this? Any help is highly appreciated.
DrDamnit
I would like to help you, but don't understand what you're trying to accomplish. What is the goal?
ASKER
OK, What I like to do is:
Connect 2 SIP telephones to the system. (OK done)
Telephone 1 have phone number 100, (OK done)
Telephone 2 have phone number 200, (OK done)
Now the problem:
10 extra phone numbers need to be in the system, they don't exist,
it is only simulated telephone numbers.
First phone number is 135 next phone number is 136 and so on.
In the training center I have a training room and a instructor room.
Telephone number 200 is in the training room and telephone number 100 is in the
instructor room.
So when the student dials number 135 it is forwarded to telephone number 100.
When the telehone rings in the instructor room, the instuctor need to know who the student is trying to call so he can act like the owner of telephone number 135.
If the student is dialing number 136 the instructor need to act like the owner of telephone number 136.
and so on.
Hope this make sense.
Connect 2 SIP telephones to the system. (OK done)
Telephone 1 have phone number 100, (OK done)
Telephone 2 have phone number 200, (OK done)
Now the problem:
10 extra phone numbers need to be in the system, they don't exist,
it is only simulated telephone numbers.
First phone number is 135 next phone number is 136 and so on.
In the training center I have a training room and a instructor room.
Telephone number 200 is in the training room and telephone number 100 is in the
instructor room.
So when the student dials number 135 it is forwarded to telephone number 100.
When the telehone rings in the instructor room, the instuctor need to know who the student is trying to call so he can act like the owner of telephone number 135.
If the student is dialing number 136 the instructor need to act like the owner of telephone number 136.
and so on.
Hope this make sense.
SOLUTION
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Ok. What I think you're trying to do is route a range of phone numbers to a handset.
For example, 130-139 are to be answered by extension 100, and extension 100 should see the callerID that matches from that number.
140-149 should be answered by extension 200, and that extension should show the originating number as well.
So, let's say a call comes in on 135, it would be routed to extension 100, and the callerID on that call would say "135". Right?
For example, 130-139 are to be answered by extension 100, and extension 100 should see the callerID that matches from that number.
140-149 should be answered by extension 200, and that extension should show the originating number as well.
So, let's say a call comes in on 135, it would be routed to extension 100, and the callerID on that call would say "135". Right?
ASKER
DrDamnit
That sounds exactly what I like to do!
That sounds exactly what I like to do!
ASKER
nasirbest:
Sorry I dont know how to use queues. Latest version of AsteriskNow uses 1.7 with FreePBX
I dont find any queues in the menu
Sorry I dont know how to use queues. Latest version of AsteriskNow uses 1.7 with FreePBX
I dont find any queues in the menu
To enable queue module you have to do following
1. click "Module Admin" from menu
2. click "Check for updates online"
3. local "Queue" module click on it and choose to install
4. proceed to save new configuration and reload asterisk
1. click "Module Admin" from menu
2. click "Check for updates online"
3. local "Queue" module click on it and choose to install
4. proceed to save new configuration and reload asterisk
ASKER CERTIFIED SOLUTION
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Sorry... I just realized you're using the GUI. I don't use the GUI. I do everything the coder way.
You need to add a call rule that matches numbers. You want to match 13X and route that to 200. This will give you the dialing functionality, but not the CID functionality.
You'll probably need to use an IVR in AsteriskNow. Have the IVR do the following things:
1. Match the calling party
2. Set the CallerID
3. Dial the extension to where you want to route the call.
You need to add a call rule that matches numbers. You want to match 13X and route that to 200. This will give you the dialing functionality, but not the CID functionality.
You'll probably need to use an IVR in AsteriskNow. Have the IVR do the following things:
1. Match the calling party
2. Set the CallerID
3. Dial the extension to where you want to route the call.
ASKER
DrDamnit
I dont mind to do it the "coder way", can you please kindly assist me with info where to put
the lines of code?
I dont mind to do it the "coder way", can you please kindly assist me with info where to put
the lines of code?
ASKER
My extensions.conf file looks like this
extensions.conf.txt
extensions.conf.txt
ASKER
nasirbest:
Thanks for the queues option. what happens is what the name says "CID Name Prefix"
I can now add some text before the callers original name, it will not hide the original name.
Thanks for the queues option. what happens is what the name says "CID Name Prefix"
I can now add some text before the callers original name, it will not hide the original name.
tru3533:
You can't edit your extensions.conf directly because it will be overwritten then next time you use the GUI and hit "save".
Instead, you have to add all your modifications here:
extensions_custom.conf
The way asterisk loads these files is that they "overlay" upon one another. So it will load the GUI generated extensions.conf then it will load the extensions_custom.conf.
So, you'll need to create a context that is the same name as the context that you created in the GUI in extensions_custom.conf, and then paste the code I put above there. Then, reload everything.
Let me know if that doesn't work. I have not actually tried it in AsteriskNOW, but it should work.
You can't edit your extensions.conf directly because it will be overwritten then next time you use the GUI and hit "save".
Instead, you have to add all your modifications here:
extensions_custom.conf
The way asterisk loads these files is that they "overlay" upon one another. So it will load the GUI generated extensions.conf then it will load the extensions_custom.conf.
So, you'll need to create a context that is the same name as the context that you created in the GUI in extensions_custom.conf, and then paste the code I put above there. Then, reload everything.
Let me know if that doesn't work. I have not actually tried it in AsteriskNOW, but it should work.
ASKER
Thank you both for the tips, managed to figure out the rest, so its working now. And I got a great introduction to Asterisk.