Monthly Archives: April 2011

Asterisk: Remotely retrieving voicemail by pressing *

Many howtos around the internet on how to remotely access your voicemail box involve a dedicated extension reachable from the outside or an IVR menu entry. But wouldn’t it be much nicer if you could just press the * DTMF key during the announcement? Turns out, this is quite simple:

exten => s,1,Dial(SIP/1234,20)
exten => s,n,Voicemail(1234,us)
exten => a,1,VoiceMailMain(1234)
exten => a,n,Hangup()

And it even works when you’re using macros (like I am):

exten => 5551234,1,Macro(incoming-plus-voicemail,SIP/1234,20,1234)
exten => 5551337,1,Macro(incoming-plus-voicemail,SIP/1337,20,1337)

[macro-incoming-plus-voicemail] ; SIP/xxx, wait time, voicemail
exten => s,1,Dial(${ARG1},${ARG2}
exten => s,n,Voicemail(${ARG3},us)
push * during the announcement to access your mailbox
exten => a,1,VoiceMailMain(${ARG3})
exten => a,n,Hangup()

Asterisk: Compile SRTP Module without recompiling Asterisk

I recently installed Asterisk 1.8.3 (the Asterisk team now provides pre-built Debian packages at
Unfortunately, that package came without the res_srtp SRTP module. (UPDATE: Starting in 1.8.4, it does come with it.) Because I didn’t feel like re-compiling the entire package, I just took the corresponding version of res_srtp.c from the SVN, added the following lines to the beginning of it:

#ifndef AST_MODULE
#define AST_MODULE "res_srtp"

and compiled and installed it using

gcc -shared res_srtp.c -o -lsrtp
sudo cp /usr/lib/asterisk/modules/

You’ll need to have libsrtp0-dev and asterisk-dev installed, otherwise the compile will fail.
Then, you can do sudo asterisk -r and load the module using module load res_srtp (or just restart Asterisk).

I’m still working on getting SRTP working flawlessly both incoming and outgoing and with stuff like transfers. Asterisk Secure Calling Specifics are a good starting point, but I’m also planning to write another post about this in the near future.

Asterisk: Change Callee-ID using CONNECTEDLINE

It’s easy to change your Caller ID (assuming your phone provider doesn’t filter it) in Asterisk using something like Set(CALLERID(name)=blah). This is often used to choose which number to use for an outgoing call if you have multiple on a single SIP or ISDN trunk.

But did you know it’s just as easy to change the Callee ID on an outgoing call, i.e. change what your phone displays during the call? This can be very useful to display on the phone which one of several possible outgoing lines (multiple SIP providers, ISDN, …) was used or at which point in an IVR menu you are at the moment. To do this, use Set(CONNECTEDLINE(name)=blah). Before getting started, set sendrpid = pai in your sip.conf.

To make things easier, I created two macros in my dialplan:

exten => s,1,Set(CONNECTEDLINE(name,i)=${ARG1})
exten => s,n,Set(CONNECTEDLINE(number,i)=${ARG2})
exten => s,n,Set(CONNECTEDLINE(pres)=allowed)

exten => s,1,Macro(connectedline-name-number,${ARG1}, ${MACRO_EXTEN})

Now I can do things like

exten => 101,1,Answer()
exten => 101,n,Macro(connectedline-name,Hello World)
exten => 101,n,Playback(hello-world)
exten => 101,n,Hangup()

in my dialplan (IVR example).

Or how about

exten => 100,1,Macro(connectedline-name,Mailbox)
exten => 100,n,VoiceMailMain(${CALLERID(num)},s)
exten => _XXX.,n,Macro(connectedline-name,VoIP 1)
exten => _XXX.,n,Dial(SIP/${EXTEN}@voipprovider)

(outgoing line example).

The Asterisk Wiki also has an entire page on Manipulating Party ID Information.