Twilio: Difference between revisions

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I tried to get my Grandstream GXV3240 to register directly to Twilio and failed. It's too much of a pain to debug the SIP
I tried to get my Grandstream GXV3240 to register directly to Twilio and failed. It's too much of a pain to debug the SIP
session between the phone and Twilio through a Mikrotik firewall/router to be worth the work.
session between the phone and Twilio through a Mikrotik firewall/router to be worth the work.
== Programmable SMS ==
Put this in a TwiML bin and connect messaging to it to forward SMS.
<?xml version="1.0" encoding="UTF-8"?>
<Response>
  <Message to="+1707yournumberhere">{{From}}: {{Body}}</Message>
</Response>


== Elastic Trunk + Asterisk ==
== Elastic Trunk + Asterisk ==

Revision as of 21:17, 5 February 2017

Programmable Voice

Geo-CEG - TwimlBin directs Geo-CEG calls to a SIP URI that goes into my Asterisk PBX via Elastic Trunk. Asterisk rings the phones.

I tried to get my Grandstream GXV3240 to register directly to Twilio and failed. It's too much of a pain to debug the SIP session between the phone and Twilio through a Mikrotik firewall/router to be worth the work.

Programmable SMS

Put this in a TwiML bin and connect messaging to it to forward SMS.

<?xml version="1.0" encoding="UTF-8"?>
<Response>
 <Message to="+1707yournumberhere">Template:From: Template:Body</Message>
</Response>

Elastic Trunk + Asterisk

This is nothing to do with elephants or the back part of a sedan or rubber bands. It has nothing to do with the symbol above the "8" on your keyboard.

Refer to https://www.twilio.com/docs/api/sip-trunking/sample-configuration and https://www.twilio.com/resources/images/docs/Asterisk-Twilio.pdf

"Elastic" means you are not limited to a preset number of calls with one SIP account. You can make any number of outbound and accept any number of inbound calls over the one account; you just have to pay for the calls.

The price for a single phone number is $1 per month. The rate for calls is about a penny a minute.

Products -> Phone numbers
You need one. Set it up.

Products -> Elastic SIP Trunking -> Trunks
First time through, "Create Sip Trunk"
Termination - set it up so you can make calls. You need something like "bellman.pstn.twilio.com" where you choose "bellman".
Origination - set it up so you can receive calls and attach a phone number to it. Try using your server like sip:bellman.wildsong.biz

You need to set up authentication on the termination settings for the trunk. The first time through you get prompted but you can edit it later

Products -> Elastic SIP Trunking -> Authentication
IP Access Control Lists - Should have the IP addresses that your server uses for outbound traffic. 67.180.204.170
Credential lists - should have a username and password that you will put into your Asterisk set up.

You can have multiple IP addresses in one ACL, more than one ACL, and more than one credential set. That way you can use one Twilio account from more than one site if you want to.

Set up on Twilio account

  1. Go to http://twilio.com/
  2. Create an account if you don't have one
  3. Buy a phone number
  4. Attach the number to a trunk, the account comes with one trunk.

Access is controlled via IP address. You have to create an access control list for your Asterisk server and put its IP address into it.

You have to set up URLs for both termination and origination.

You have to set up a credential with a username and password.

Modifications to Asterisk

extensions.conf

sip.conf

Twilio for Geo-CEG

Geo-CEG is a small non-profit. The board members live all over the place.

I got a Twilio account through their program to support non-profits. Now I need to make it useful.

I want the number to be able to forward calls to whoever is "on call", and to store voice mails when no one is available.

SMS messages

You can use the same number to send and receive SMS calls. I tested it using the sample code by putting the PHP on my own server and then pointing the SMS URL at my server. It works. That's how I can receive SMS messages. I have not tried sending an SMS yet, except that the PHP code sends a reply back.

I set up a script in my dev SSL server, sms.php.

HTTPS server set up =

To set up HTTPS, I had to tighten up security on my Apache, test it here: https://ssllabs.com/ssltest I started with a "C" rating and moved up to "A"! How to: see https://hynek.me/articles/hardening-your-web-servers-ssl-ciphers/

See also https://wiki.mozilla.org/Security/Server_Side_TLS

Complete your certificate chain: https://certificatechain.io/ and http://superuser.com/questions/644343/how-do-you-fix-an-incomplete-ssl-chain

Goal - send text to server. Have the server phone back and read a message.

  1. Receive SMS - done
  2. Log information about SMS
  3. Initiate a call

Goal - build a system that will receive an SMS, perform an action, and send a confirmation.

This can be (and he stopped typing there)

Take aways from Signal Conference 2015

Move stuff into Asterisk

Lots of stuff in Twilio space could be implemented in Asterisk space. But here at the end of 2016 I am now working on going the other direction. I have Asterisk working just fine but am thinking of dumping it in favor of OpenVBX.

My thought back then was that TWIML could be used to design IVR / autoattendant systems in Asterisk. Doing this could make it easy to implement a failover, primary would be Asterisk but if * is down then the same code would fire up and run in Twilio.

What Twilio is doing

What Twilio is doing is building a big company by (1) doing only the back end, no applications. They provide only APIs and you write your own applications. (2) Using cloud. Everything they do lives in Amazon. Based on that there must be some backend to connect to carriers that does not require anything but IP. :-) That is, they don't have big server rooms full of HT503's

They have that big giant corporation mentality. But they do make their services available to little fleas like us. I can buy one phone line for $1 a month.

Call forwarding

Make a call go to Asterisk and to my cellphone at the same time.

The Asterisk way: Put a rule into the dialplan for outside calls that rings both inside (SIP) phones and also sends the call to my cellphone by calling out via Twilio. This works surprisingly well because Twilio bridges the two lines internally. The media stream does not have to go from Twilio to my PBX and back to Twilio.

The problem is that if my cellphone is working and my PBX is down then I will miss calls.

The Twilio way: Send the call to both places (PBX and phone) from the Twilio side. I think I can do this with Programmable Voice.