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
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