Newbie having trouble with his first dialplan

I can dial out with not problem, just having trouble with the inbound.

Out bout is assigned to context default.
I am trying to create and inbound context.
Obviously this has to be a different context.

I would just throw the outbound under globals, but where I am going with this, that would not be a good idea so I am trying to understand how to make it work

  1. Without outbound being in the globals context.
  2. Without it being in the same context as inbound.

Can I include 2 context in my sip.conf?


And then just create those two in extensions?

I will get into arguments later is my learning progresses, I am just taking baby steps in my learning phase.

I am doing a ton of reading but finding everything “but” what I am looking for.

Here are my configs. (I know dial is in the context now, I need to get it out of there!)


exten => 100,1,Wait(1)
exten => 100,2,Answer
exten => 100,3,Playback(tt-weasles)
exten => 100,4,Voicemail(44)
exten => 100,4,Hangup
exten => 8005181896,2,DIAL(SIP/100,20)
exten => _1XXXXXXXXXX,1,DIAL(SIP/teliax/${EXTEN},30,tr)
exten => _XXXXXXXXXX,1,DIAL(SIP/teliax/${EXTEN},30,tr)


callerid=“Brad Sumrall”

Someone pointed me to a great pdf from the asterisk mailing list.

Here is my reply!

First off, I wanted to thank you for referring me to the O’Reily pdf.
It has already helped a lot and now I know exactly where I am going wrong, but still do not have an answer!

Almost every example on and O’Reily assume you are using an FXO or FXS card.
I am 100% internet based.

It hit me like a rock that I need to understand why this affects the channels differently.

O’Reily states:
exten => s,1,Answer( )
exten => s,2,Playback(hello-world)
exten => s,3,Hangup( )
If you have a channel or two configured, go ahead and try it out! Simply make a new
extensions.conf file with this short dialplan. (If it doesn’t work, check the Asterisk
console for error messages, and make sure your channels are configured to send
inbound calls to the [incoming] context.)

Go figure! I am 100% SIP based and zero IAX and I am assuming this effects how asterisk looks a Zapata.conf?

So, this would lead to the logical conclusion that if I do not configure incoming in Zapata, I configure it on the teliax authentication portion of sip.conf!

It still didn’t work.

I can see my phone number coming in on the CLI, but zero transition into the basic commands of extentions.conf

Teliax got the thing to work before. They simply stripped out everything and put in what appeared to be the exact example on their web site.

I was playing around with extensions.conf only and now it doesn’t work at all and the conceptual theory doesn’t seem to apply when dealing with 100% only SIP vs. FXx

So, can anyone point me into the right direction on documentation on “understanding the differences of SIP vs. FXx?

Got it basically working, but still need answers as to why SIP is so much different from FXx