SIP to number connection

Asterisk Asterisk SIP
1

Good morning,
I would like to launch such an extension that permanently connects me with an internal SIP number, e.g. 200, with a given telephone number.
I see it this way:
I run the script - how? via API? or maybe from a PHP server in asterisk? (I don’t know yet) with the given number.
ie: http://localhost:80/connect?from=fromTrunk&to=7666666666&internalConn=200

Asterisk from the fixed extension dials the call to this number and at the same time calls the given SIP / extension.
So that after lifting the handset by the extension number you have established connection:
SIP (internal, e.g. 200) with a given external number via a given access number (TRUNK).

?
How to realize it?

2

Call files or one of the many variants of “oriiginate”.

3

? What ‘call files’.
How to write it in extensions?

I have this: https://www.voip-info.org/asterisk-config-extensionsconf/

And this: …
exten => s,3,Dial(SIP/goodperson)
exten => s,4,Dial(VARIABLENUMBER)

How to call to variable number? How to take IT as changable number?
How to join this two calls?

4

You need to understand the basics before you try anything complex. You seem to have a fundamental understanding of what Dial() does; it supervises, as well as dials the call, so doetn’t exit until one party clears.

Also, you need to be prepared to search and read the documentation once given a hint.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files
https://wiki.asterisk.org/wiki/display/AST/ManagerAction_Originate
https://wiki.asterisk.org/wiki/display/AST/Application_Originate
https://wiki.asterisk.org/wiki/display/AST/Creating+and+Manipulating+Channels+from+the+CLI

https://www.google.com/search?q=asterisk+click+to+cal

5

Thank You very much! :slight_smile: !!!

6

I have it :slight_smile: first working version, have to evaluate :slight_smile:

step 1.
add it into /etc/asterisk/extensions_custom.conf
(just edit form: http://ASTERISKIP/admin/config.php?display=configedit ):

[call-test-afya]
exten => 10,1,Set(TIMEOUT(digit)=5)
exten => 10,2,Set(TIMEOUT(response)=10)
exten => 10,n,Answer
exten => 10,n,NoOp(“----START EXTENSION—”)
exten => 10,n,Wait(1)
exten => 10,n,NoOp(“—CALLING TO ${CUSTOMER} USING: SIP ${PBX}—”)
exten => 10,n,Dial(SIP/${PBX}/${CUSTOMER})
exten => 10,n,NoOp(“----STOP EXTENSION—”)
;----calltestafya—;

step 2:
log in into console and type:

core reload

step 3:
do something (php/python script) , to generate (copy or create files with .call extension in: /var/spool/asterisk/outgoing )
they have to contain:

Channel: SIP/200 ; your internal phone you want to pick up
Callerid: AUTOMATIC CALL ; as you wish
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: call-test-afya ; my extension context
Extension: 10 ; my extension number
SetVar: CUSTOMER=123456789 ; where to call - change it by script
SetVar: PBX=freepbx.pl-XXXXXXXXX-OUT ; your SIP outgoing trunk, you find this name in: http://ASTERISKIP/admin/config.php?display=trunks

Thank You 4 help!

7

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