How to generate call to a mobile using asterisk

Hi
I am a college student and a complete newbie to asterisk.
I’m currently working on a project ‘email to voice call’.
Using python i’v extracted the email & converted it into speech and saved in a WAV file.
Now using asterisk i want to generate call to the mobile of the user through my system.

I have read the book ‘Asterisk: The Future Of Telephony’ as suggested by many. But i’m still not able to understand what all things i need to setup to generate a call to mobile.
What i understood is that i need to configure two files i.e. sip.conf where i need to give the details of VoIP provider and extensions.conf for dial-plan. Asterisk will tell the VoIP provider to generate a call.

Now can anyone please tell me what things i need to setup other than these? Also can you help me in the configuration of these two files??

Please help. Any information will be appreciated.
Thank You.

if your a complete newbie, then that’s a lofty goal, i reccommend easing yourself in by creating 2 SIP phones in SIP.conf, then writing a dialplan for each to call each other.

so create this in SIP.conf as your user (assuming you have a good “general” context already defined)

[100]
type=friend
host=dynamic
dtmfmode=rfc2833
context=bouncy
disallow=all
allow=all
secret=password
nat=yes
canreinvite=no
registertrying=yes
Subscribecontext=hints
limitonpeer=yes
notifyringing=yes
notifyhold=yes
call-limit=99

then the same again for another extension (200?)

then create [bouncy] in extensions.conf, and put in: -

exten => _X.,n,NoOp(testing the bouncy context from my phone)
exten => _X.,n,Hangup()

Hook up your SIP phones (you can use a smartphone (c-sip simple?), or x-lite for the desktop)
then watch the CLI for the phones registering, then the noop when you dial any number with 2 digits or more for the text in the NoOp defined above. If you see it, then you’re ready to start tinkering :smile:

***I’m assuming a lot here, you havent said whether you have a working asterisk server up to be able to start writing your dialplan

canreinvite is deprecated
nat=yes makes no sense here.
There are quite a few options that are not necessary for a basic configuration, or the configuration required for the application.

Generally, forums like this are not the best place for complete solutions. They should be used for filling in the details once a best effort has been made at achieving a solution. For example, I think making a call from the PSTN to voice announcement and making a call from a local phone to the PSTN fall under Asterisk/101. The provider of the SIP to PSTN gateway should be able to provide advice in configuring Asterisk to use their gateway.

thanks for the reply riva_00 & david55… i have asterisk 10.2 installed on my ubuntu os.
what i want is to generate a call to the cell phone of the user using asterisk. The voip provider will make a call when requested by asterisk…

i want to know, to complete above task what all things i need to setup. (other than configuring sip.conf, extension.conf & registering with some voip provider). do i need some additional hardware? Please reply.

I don’t know what your baseline hardware is!

Google for “call files”.