freeswitch/libs/libcodec2/asterisk

README for codec2/asterisk
Asterisk Codec 2 support

Test Configuration
------------------

Codec 2 is used to trunk calls between two Asterisk boxes:

    A - SIP phone - Asterisk A - Codec2 - Asterisk B - SIP Phone - B

The two SIP phones are configured for mulaw.

Building
---------

Asterisk must be patched so that the core understand Codec 2 frames.

1/ First install Codec 2:

    david@cool:~$ svn co https://freetel.svn.sourceforge.net/svnroot/freetel/codec2-dev codec2-dev
    david@cool:~/codec2-dev$ cd codec2-dev
    david@cool:~/codec2-dev$ ./configure && make && sudo make install
    david@bear:~/codec2-dev$ sudo ldconfig -v
    david@cool:~/codec2-dev$ cd ~

2/ Then build Asterisk with Codec 2 support:

    david@cool:~$ tar xvzf asterisk-1.8.9.0.tar.gz
    david@cool:~/asterisk-1.8.9.0$ patch -p4 < ~/codec2-dev/asterisk/asterisk-codec2.patch
    david@cool:~/asterisk-1.8.9.0$ cp ~/codec2-dev/asterisk/codec_codec2.c .
    david@cool:~/asterisk-1.8.9.0$ cp ~/codec2-dev/asterisk/ex_codec2.h ./codecs
    david@cool:~/asterisk-1.8.9.0$ ./configure && make ASTLDFLAGS=-lcodec2
    david@cool:~/asterisk-1.8.9.0$ sudo make install
    david@cool:~/asterisk-1.8.9.0$ sudo make samples

3/ Add this to the end of sip.conf on Asterisk A:

    [6013]
    type=friend
    context=default
    host=dynamic
    user=6013
    secret=6013
    canreinvite=no
    callerid=6013
    disallow=all
    allow=ulaw

    [potato]
    type=peer
    username=potato
    fromuser=potato
    secret=password
    context=default
    disallow=all
    dtmfmode=rfc2833
    callerid=server
    canreinvite=no
    host=cool
    allow=codec2

3/ Add this to the end of sip.conf on Asterisk B:

    [6014]
    type=friend
    context=default
    host=dynamic
    user=6014
    secret=6014
    canreinvite=no
    callerid=6014
    disallow=all
    allow=ulaw

    [potato]
    type=peer
    username=potato
    fromuser=potato
    secret=password
    context=default
    disallow=all
    dtmfmode=rfc2833
    callerid=server
    canreinvite=no
    host=bear
    allow=codec2

4/ Here is the [default] section of extensions.conf on Asterisk B:

    [default]

    exten => 6013,1,Dial(SIP/potato/6013)
    ;
    ; By default we include the demo.  In a production system, you
    ; probably don't want to have the demo there.
    ;
    ;include => demo

5/ After booting see if the codec2_codec2.so module is loaded with "core show translate"

6/ To make a test call dial 6013 on the SIP phone connected to Asterisk B

7/ If codec_codec2.so won't load and you see "can't find codec2_create" try:

    david@cool:~/asterisk-1.8.9.0$ touch codecs/codec_codec2.c
    david@cool:~/asterisk-1.8.9.0$ make ASTLDFLAGS=-lcodec2
    david@cool:~/asterisk-1.8.9.0$ sudo cp codecs/codec_codec2.so /usr/lib/asterisk/modules
    david@cool:~/asterisk-1.8.9.0$ sudo asterisk -vvvcn