calltest 0.7.4

dtmf

The answering channel sends a random sequence of DTMF tones. The originator then does the same thing. The receivers verify that the codes are correct.

When dtmf.may_repeat is set, the receiver is allowed to read duplicate DTMF tones. This flag might be necessary with in-band signalling.

dtmf.len is the number of digits to test. Typically, one digit will be repeated. The sequence is otherwise random.

call

The answering channel will acknowledge that it’s ringing, but then simply hang up.

audio

Like DTMF, but both sides send a sound file instead. The sounds are recorded. They should match, somewhat, but that’s not yet tested.

ring

Originate-only: call this number. It should be RINGING. Then hang up.

play

Originate-only: call this number. It should answer. Play a sound, then hang up.

If no audio.src_out sound is specified, simply hang up.

Optionally, audio.dst_in records incoming audio during the call. Recording runs while the outgoing sound is playing.

fail

TODO.

Originate-only: call this number. The call may not go through. It must be rejected, some time before ringtime.

Also TODO: Add a flag to determine whether an intermediate Ringing state is allowed / required / prohibited.

timeout

TODO.

Originate-only: call this number. The call may not go through. It must be RINGing and then be rejected between ringtime and timeout seconds later.

wait

Answer-only: wait for an incoming call, set it to RINGING, wait a few seconds, hang up.

answer

Answer-only: wait for an incoming call, answer it, optionally play a sound, optionally² record audio while the sound is playing, then hang up.

record

Answer-only: wait for an incoming call, answer it, optionally play a sound, optionally record incoming audio until the originator hangs up or presses #.

The recording will be overlaid with the outgoing sound. This is an Asterisk limitation.

Outgoing

The outgoing Asterisk channel should be able to handle both kinds directly. If not, you need to write a “Local” channel that mangles the dialled number. For instance, if you need to drop the ‘+’, use this macro:

context mangle {
    _+! => Dial(SIP/broken/${EXTEN:1});
}

and then call Local/{number}@mangle. As another example, if you need to use lcoal number format to dial out:

context mangle {
    _X! => Dial(SIP/broken/${EXTEN});  // pass-thru for local extensions
    _+49123! => Dial(SIP/broken/${EXTEN:6});  // 49123: country+city
    _+49! => Dial(SIP/broken/0${EXTEN:3});  // 49: country; 0: national prefix
    // _+! => Dial(SIP/broken/00${EXTEN:1}); // 00: international prefix
    _+! => Congestion();
}

though you can probably get by with just the first and last line.

If you’re in the NANP (USA or Canada), you probably want to use this macro instead:

context mangle {
    _N! => Dial(SIP/broken/1888${EXTEN});  // 888 is your area code
    _[01]! => Dial(SIP/broken/${EXTEN});  // pass-thru for operator and long-distance
    _+1! => Dial(SIP/broken/${EXTEN:1});  // long distance
    // _+! => Dial(SIP/broken/011${EXTEN:1});  // international
    _+! => Congestion();
}

The last line causes a “Congestion” blocker so that a mistake won’t cause international charges.

Incoming

CallTest will verify that the caller’s number on an incoming call matches the number in the test’s configuration’s source link. If that number is prefixed with a ‘+’, the incoming number is converted to international format and needs to match exactly. Otherwise, the configured number is assumed to be a local extension and must only be at the end of the caller’s.

This ensures that there’s no incoming nonsense, while acknowledging that site-local numbers often are not transmitted cleanly.

If you need Asterisk to mangle the caller’s number so that it looks sane enough for CallTest, add that to the calltest macro. For instance, if your phone system adds a leading “get a trunk line” zero (in the NANP it’s usually a 9) to incoming calls, you’d do this:

macro calltest(typ) {
    SET(cid=${CALLERID(num)})
    switch(cid) {
        pattern 0.:
            set(CALLERID(num)=${cid:1});
            break;
        default:
            break;
    }
    Stasis("calltest", ${typ});
    Hangup();
    return;
}

We know that it’s annoying to have to play with Asterisk’s dialplans for this. If you want to add a pattern-match-based number manglich scheme to CallTest, great, we’ll gladly accept a pull request. However, “sane” modern phone systems should work fine with the default setting.

Server connection

Some parameters in the asterisk section require further elucidation.

• app:

• audio: (the base of) the “sound” URL which Asterisk will use to find your test’s outgoing sound files. Should be sound:/some/absolute/path.

Test setup

Asterisk needs to know how to direct incoming calls to the tester. This is typically done with a Stasis macro. This is for extensions.ael:

macro calltest(link,nr) {
    Stasis(calltest,${link},${number});
    Hangup();
    return;
}

• calltest: the asterisk.app config parameter.

• link: the name of the link, in asterisk.link.

• nr: the incoming destination phone number.

You’d call this macro from your context:

1234 => &calltest(foo,${EXTEN});

If you want to route all incoming calls on a channel to this macro, use something like this context:

context ext_bar {
     s => &calltest(bar,);
     i => &calltest(bar,${INVALID_EXTEN});
     _+! => &calltest(bar,${EXTEN});
     _X! => &calltest(bar,${EXTEN});
     h => Hangup();
}

Warning:

These calls all have no spaces after the argument-separating commas.
This is important.

Line setup

PJSIP is not very forgiving. Sample config: TODO.

In lieu of voice quality checking, which this system does not yet do, you might want to simply set your endpoints’ DTMF mode to “inband”. Don’t use a compressing codec when you do this. You might need to set the test’s dtmf_may_repeat option. However, in-band DTMF is not particularly reliable and may break randomly.

The random DTMF sequence only uses digits because letters are not universally passed on, while # and * may be interpreted and thus swallowed by intermediate systems.

Testing

This package contains an example.cfg configuration. To use it, your asterisk.ael file should contain these lines:

macro calltest(link, nr) {
    Stasis(calltest,${link},${number});
    Hangup();
    return;
}

context direct {
    _.! => &calltest(${EXTEN},);

    answer => {
        Wait(1);
        Answer();
        Wait(5);
        Hangup();
    }
    ringing => {
        Wait(1);
        Ringing();
        Wait(5);
        Hangup();
    }
    progress => {
        Wait(1);
        Progress();
        Wait(5);
        Hangup();
    }
}

Add this to your ari.conf:

[example] type = user password = oh_no_you_do_not password_format = plain

Finally, add enabled=yes to the [general] section of http.conf.

After doing all of this and restarting your Asterisk (reloading “ael”, “http” and “res_ari” should work too), the command ./ct -c example.cfg run should pass.