SIP to SMS bridging

I’ve recently updated my local Asterisk PBX. The main reason for the update was that processing the logs in order to set up the firewall rules to block the folk that hammer on it all day long trying to make long distance calls or run up big bills on premium rate numbers was getting too much for the original Mk i Raspberry Pi B (it now runs on a Pi 3 b+ which more up to the task).

As part of the set up I have 3G USB dongle that also supports voice calls using the chan_dongle plugin. This plugin also supports sending and receiving SMS messages (this is different from the MMS/SMS gateway I also run on a separate pi) and they are handled by the dial plan.

My first pass at the dial plan just publishes the incoming message to a MQTT topic and it is then processed by Node-RED, which emails a copy to me as well as logging it to a file.

exten => sms,1,Verbose(1,Incoming SMS from ${CALLERID(num)} ${BASE64_DECODE(${SMS_BASE64})})
exten => sms,n,AGI(/opt/asterisk/agi-mqtt/mqtt,/opt/asterisk/agi-mqtt/mqtt.cfg,foo/sms,${BASE64_DECODE(${SMS_BASE64})})
exten => sms,n,Hangup()

This works OK for incoming messages but sending them is a bit harder, as the only way to send them from outside the dialplan (from within the dialpan you can use DongleSendSMS) is to use the asterisk CLI tool which is a bit clunky.

What I was really looking for was a way to send/receive SMS messages like you do with a mobile phone. To make calls I normally use Linphone on my tablet and this includes support for SIP messaging. This lets you send messages between SIP clients and you can get Asterisk to consume these.

You can send SIP messages with the MessageSend asterisk dailplan function

The following is a basic echo bot:

exten => _.,1,Answer()
exten => _.,n,Verbose(1,Text ${MESSAGE(body)})
exten => _.,n,Verbose(1,Text from ${MESSAGE(from)})
exten => _.,n,Verbose(1,Text to ${MESSAGE(to)})
exten => _.,n,Set(FROM=${MESSAGE(from)})
exten => _.,n,Set(TO=${REPLACE(FROM,<'>,)})
exten => _.,n,MessageSend(pj${TO},${CUT(MESSAGE(to),:,2)})
exten => _.,n,Hangup()

Because I’m using the PJSIP module rather than the legacy SIP module I need to prefix the outbound address with pjsip: rather than sip:. This also matches any target extension which will be useful a little later.

To enable a specific context for SIP messages you need to add message_context to the PJSIP endpoint for the SIP user:

type = endpoint
context = internal
message_context = messaging

Now if we put the 2 bits together we get a dialplan that looks like this:

exten => sms,1,Verbose(1,Incoming SMS from ${CALLERID(num)} ${BASE64_DECODE(${SMS_BASE64})})
exten => sms,n,AGI(/opt/asterisk/agi-mqtt/mqtt,/opt/asterisk/agi-mqtt/mqtt.cfg,foo/sms,${BASE64_DECODE(${SMS_BASE64})})
exten => sms,n,Set(MESSAGE(body))=${BASE64_DECODE(${SMS_BASE64})})
exten => sms,n,MessageSend(pjsip:nexus7,${CALLERID(num)})
exten => sms,n,Hangup()

exten => _.,1,Answer()
exten => _.,n,Verbose(1,Text ${MESSAGE(body)})
exten => _.,n,Verbose(1,Text from ${MESSAGE(from)})
exten => _.,n,Verbose(1,Text to ${MESSAGE(to)})
exten => _.,n,Set(FROM=${MESSAGE(from)})
exten => _.,n,Set(TO=${REPLACE(FROM,<'>,)})
exten => _.,n,DongleSendSMS(dongle0,${EXTEN},${MESSAGE(body)},1440,no)
exten => _.,n,Hangup()

The first part handles the incoming SMS messages delivered to the dongle and passed to the sms extension in the dongle-incomming context. This logs the message to the console and via MQTT then fires it off to my tablet as a SIP message. The second is the context for the incoming SIP messages from the tablet, this will accept messages to any extension, logs the message, who it’s to/from then sends it to the number in the extension via the dongle.

Using SIP Client to send SMS

More playing with Asterisk

I have been playing some more with Asterisk and I’ve got 2 useful bits to share.

MP3 Voicemail

The simple one first, a recent update to Android (not sure what exactly) means it won’t playback wav files attached to emails. This is a problem as when Asterisk records a voicemail it can be configured to email the recording in wav to mailbox owner. A little bit of googling turned this up It needed updating a little to get it to work on my Raspberry Pi.

First I needed to change one of the sed line to match WAV not wav to fix the file name from this:

...| sed 's/.wav/.mp3/g' > stream.part3.mp3.head

to this:

...| sed 's/.WAV/.mp3/g' > stream.part3.mp3.head

Secondly lame doesn’t like the encoding for the wav file (I think it’s because the stored values are unsigned) so we need to run it through sox to fix it first.

sox stream.part3.wav -e signed stream.part3a.wav
lame -m m -b 24 stream.part3a.wav stream.part3.mp3

This not only makes it so I can listen to the messages on my phone and tablet it also makes the mails smaller so they take up less bandwidth.

Using a 3G Stick to make calls

Having used OBi110 to hook my Asterisk VoIP rig up to a standard phone line I was looking for a way to hook a mobile phone to the system. There are 2 options with Asterisk, chan_bluetooth and chan_dongle.

Chan_bluetooth uses a bluetooth connection to a mobile phone to make and receive calls. I had a look at this but it meant keeping phone plugged into a charger and having another bluetooth adapter plugged in.

Chan_dongle work with certain Huawei 3G USB modems. These 3G sticks are basically full phones with a USB interface. I’ve already been using one of listed modems for my SMS/MMS project and I had a spare one kicking around. It needed the firmware updating to make it work, which was a bit of a challenge as it required setting up a real Windows machine as I couldn’t get it to work in a VM.

Setting up the dongle was a little tricky at first as I couldn’t get a set of udev rules to match the stick properly to ensure it always ends up with the same device names. The code does let you specify the stick using its IMEI which helps if you have multiple sticks plugged into the same computer.

Once configured it was easy to set up the extensions.conf to allow making and receiving calls. The main reason for setting this up was to have a portable VoIP rig that I can take to different places and not have to worry about a fixed phone line. There is a upcoming hackday that I have a plan for.


The title is a bit of a mouthful, but basically being able to set up recordings on my PVR by calling the my home phone number and just speaking.

This is one of the projects I wanted to play with after setting up my OBi110 and Asterisk PBX.

Setting up systems where you press digits on your phone to navigate menus in Asterisk is pretty simple, but systems that listen to what you say and then interpret that are a little trickier. To make it work you need 3 main parts:

  • A system to record the audio
  • A way to parse the audio and turn it into text
  • Something to extract the interesting bits from the text

Asterisk will record the audio if poked the right way which leaves the last two bits to sort out.

Some of the guys at work do voice recognition projects and pointed me at some of the open source toolkits1, but these normally involve lots of training to get things accurate and reasonably meaty boxes to run the systems on. Since I’m running Asterisk on a Raspberry Pi I was looking for something a little more light weight.

A bit of searching round turned up a project on Github that uses Google’s Voice to Text engine with Asterisk already. This looked spot on for what I needed. Getting hold of a Google Speech API key is a little tricky as it’s not really a public API, but support for it is build into the Chromium web browser so following these instructions helped. The API key is limited to 50 calls a day but that should be more than enough for this project.

Once installed the following flow in the Asterisk dialplan lets you dial extension 200 and it will record any speech until there are 3 seconds of silence then it forwards it on to the Google service, when it returns it puts the text into a dialplan variable called utterance, along with a value between 0 and 1 indicating how confident Google is in what it says in a variable called confidence.

exten => 200,1,Answer()
exten => 200,n,AGI(/opt/asterisk/asterisk-speech-recog/speech-recog.agi,en-GB,3,#,NOBEEP)
exten => 200,n,Verbose(1,The text you just said is: ${utterance})
exten => 200,n,Verbose(1,The probability to be right is: ${confidence})
exten => 200,n,Hangup()

An example output:

The text you just said is: record bbc1 at 9 p.m.
The probability to be right is: 0.82050169

Now I’ve got some text to work with I needed something to make sense of it and turn it into an action that can be followed. A simple way to do this would be with some regular expressions2 but I wanted to try something a little smarter that I could also use to add support for other bits of my home automation system. This means looking at some proper NLP and Text Analytics technology.

Dale has recently written about deploying some simple Text analytics tools to BlueMix which I used as a starting point along with this set of introductory tutorials for IBM Languageware Workbench.

Following the instructions I built a number of databases, the main one of television channel names to make them easy to match and to include multiple versions to help smooth out how the voice to text engine interprets things like “BBC One” which could easily end up being mapped to BBC 1 or BBC1 to name but two. Then a bunch of rules to match times. It’s a little long winded to go into here, if I get time I’ll do a separate post on writing UIMA rules. Once the rules were finished I exported them as a PEAR file and wrote a Java Servlet to feed text into the pipeline and extract the useful bits from the CAS. The source for the servlet can be found on Github here. When I get a bit more time I’ll do a more detailed post on how I actually created these rules.

Now that I had a web end point I could send text to and get it marked up with all the interesting bits I needed a way to forward text to it from within the Asterisk dialplan. I used the earlier Voice to Text example to put together this little bit of perl

#!/usr/bin/env perl

use warnings;
use strict;
use URI::Escape;
use LWP::UserAgent;
use JSON;

my %AGI;
my $ua;
my $url = "";
my $response;
my $temp;

# Store AGI input #
($AGI{arg_1}) = @ARGV;
while (<STDIN>) {
        last if (!length);
        $AGI{$1} = $2 if (/^agi_(w+):s+(.*)$/);

$temp = "text=" . uri_escape($AGI{arg_1});

$ua = LWP::UserAgent->new;
$response = $ua->post(
	Content_Type => "application/x-www-form-urlencoded",
	Content => "$temp",
if (!$response->is_success) {
	print "VERBOSE "some error"n";
} else {
	print "SET VARIABLE "action" "$response->content"n";

sub checkresponse {
        my $input = <STDIN>;
        my @values;

        chomp $input;
        if ($input =~ /^200/) {
                $input =~ /result=(-?d+)s?(.*)$/;
                if (!length($1)) {
                        warn "action.agi Command failed: $inputn";
                        @values = (-1, -1);
                } else {
                        warn "action.agi Command returned: $inputn" if ($debug);
                        @values = ("$1", "$2");
        } else {
                warn "action.agi Unexpected result: $inputn";
                @values = (-1, -1);
        return @values;

The response looks like this which I then used to feed a script that uses the MythTV Services API to query the program guide for what is showing at that time on that channel then to schedule a recording.

  "time": "9:00 am",
  "action": "record",
  "channel": "BBC ONE"

And I included the script in the dialplan like this:

exten => 200,1,Answer()
exten => 200,n,AGI(/opt/asterisk/asterisk-speech-recog/speech-recog.agi,en-GB,3,#,NOBEEP)
exten => 200,n,Verbose(1,The text you just said is: ${utterance})
exten => 200,n,Verbose(1,The probability to be right is: ${confidence})
exten => 200,n,AGI(/opt/asterisk/uima/action.agi,"${utterance}")
exten => 200,n,AGI(opt/asterisk/mythtv/record.agi,"${action}")
exten => 200,n,Hangup()

I need to add some more code to include some confirmation in cases where the confidence in the extracted text is low and also once the program look up has happened to ensure we are recording the correct show.

Now I have the basics working I plan to add some more actions to control and query other aspects of my home automation system.

1 Kaldi seams to be one of the interesting ones recently.
2 did I really say simple and RegExp in the same sentence?

Playing with Asterisk PBX

I’ve been meaning to get back and have a proper play with Asterisk again for a while. Last week Amazon sent me one of those emails about things you’ve looked at but not bought and I spotted this:

It was down from £60 to £35 so I did exactly what they wanted and bought one.

Now normally I don’t use my land line at all, it’s just there to let the internets in, it doesn’t even have a handset plugged in. But there are a few little projects kicking around the back of my mind I’ve been thinking about for a while and the OBi110 should let me play with them.

The first is to see if the (unused, never given to anybody but my ISP to set up the conection) number for the land line has ended up on any lists for scamers/spammers and people generally trying to sell me stuff. My mobile gets at least 1 call a week about payment protection and the like and even my work office number has started getting recorded calls about getting my boiler replaced.

I could have probably just used the call log on the OBi110 but I wanted to be able to potentially record these calls and a few other things so I needed something a little smarter which is were Asterisk comes in. Asterisk is a opensource VoIP PBX this basically means it acts like a telephone exchange for calls made over the internet. I’ve seen people run Asterisk on the old Linksys Slugs so I was sure it should run fine on a Raspberry Pi as long as it wasn’t dealing with too many calls and not doing much codex transcoding. As I already had a Pi running my SMS/MMS rig it seamed like a good place to put all my telephone stuff.

Installing Asterisk on the Pi was just a case of running apt-get install asterisk. It comes with a bunch of default config files (in /etc/asterisk), but there are 2 main ones that I needed to change to make some simple things work.

This file is where you can configure what clients can connect to your asterisk instance via the SIP protocol. To start with I’m going to set up 2 different clients, one for a softphone running on my laptop and one for the OBi110. It sets up few things, but the import bit for later is the context which controls which bit of the extentions.conf file we jump to when receiving a call from each client.


disallow=all ; only the sensible codecs


This file defines how Asterisk should handle calls, it has two contexts called local and external. The local context defines 2 paths, the first for extension 100, when this number is called from the softphone Asterisk calls out to a small python program called agi-mqtt which publishes a JSON object to the calls/local MQTT topic which contains all the information Asterisk has about the call. It then answers the call then plays audio file containing HelloWorld and finally hangs the call up. I’m mainly using this local context to testing things out before copying them over to the external context.

The second path through the local context uses a special case extension number “_0Z.”, this matches any number that starts with 0[1-9] (so won’t match against 100). This path forwards the dialed number on to the OBi110 to place the call via the PSTN line.

The external context only contains 1 path which matches the phone number of the PSTN line and currently matches the 100 extension (play HelloWorld). At some point later I’ll setup this path to forward calls to a local softphone or forward to a voicemail account.

exten => _0Z.,1,AGI(/opt/asterisk/agi-mqtt/mqtt,/opt/asterisk/agi-mqtt/mqtt.cfg,calls/local)
exten => _0Z.,2,Dial(SIP/${EXTEN}@obihai);
exten => _0Z.,3,Congestion()
exten => _0Z.,103,Congestion()
exten => t,1,Hangup()

exten => 100,1,AGI(/opt/asterisk/agi-mqtt/mqtt,/opt/asterisk/agi-mqtt/mqtt.cfg,calls/local)
exten => 100,2,Answer()
exten => 100,3,Playback(en_US/hello-world)
exten => 100,4,Hangup()


exten => 0123456789,1,AGI(/opt/asterisk/agi-mqtt/mqtt,/opt/asterisk/agi-mqtt/mqtt.cfg,calls/pstn-in)
exten => 0123456789,2,Answer()
exten => 0123456789,3,Playback(en_US/hello-world)
exten => 0123456789,4,Hangup()

Now Asterisk is all working properly I setup the OBi110 using the instructions found here.

After a bit of playing I have inbound and outbound calls working and some MQTT enabled logging. Next up is looking at using the SIP Client built into Android to allow calls to be made and received from my mobile phone.