Jingle bots, offering Asterisk services to Psi+

Данная страница содержит информацию по работе с сервисом Asterisk в Psi+.

Для того, чтобы использовать Asterisk с Psi+, необходимо использовать сервис Asterisk (Jingle branch) с некоторыми патчами. (This patches on the Psi side are not necessary anymore.)

Ниже представлен список известных на данный момент jingle-ботов:

To use one of the bots listed above:

• Add the JID to your roster
• The contact should authorize you and appear online
• Make a voice call to the contact

The contact should pick up the call and offer the noted service.

• Check out the special Jingle branch of Asterisk (revision 281181 is the last revision that is still working):

svn -r 281181 co http://svn.digium.com/svn/asterisk/team/phsultan/jingle-support/

• Obtain the patch to work around some bugs and the patch to allow dealing with wideband speex

wget http://haar.student.utwente.nl/~julius/patches/asterisk-jingle-hacks-20100430.patch.txt
wget "https://issues.asterisk.org/file_download.php?file_id=25936&type=bug" -O patch-jingle_speex-1.diff

• Enter the tree and apply the patches

cd jingle-support
patch -p1 < ../asterisk-jingle-hacks-20100430.patch.txt
patch -p0 < ../patch-jingle_speex-1.diff

• Compile and install Asterisk as usual, make sure res_jabber and chan_jingle are enabled

./configure
make menuselect
make
make install
make samples

• Register a XMPP account (using Psi for example, you can remove it from Psi afterwards)

• Edit /etc/asterisk/jabber.conf and add the new account:

[echo]]
type=client
serverhost=localhost
username=echo@haar.student.utwente.nl/asterisk
secret=<password>
port=5222
usetls=no
usesasl=no
statusmessage=voice: echo test

Note that in the above case, Asterisk is running on the same server as the XMPP daemon.

• Edit /etc/asterisk/jingle.conf:

Make sure speex is allowed, the two lines below should work

disallow=all
allow=speex

Take note of the context or change it to something like public

context=public

* Edit /etc/asterisk/extensions.conf:

Edit or create a new context and set your dialing rules:

This is the example for an echo test:

exten => echo@haar.student.utwente.nl,1,Playback(demo-echotest) ; Let them know what's going on
exten => echo@haar.student.utwente.nl,n,Echo                    ; Do the echo test
exten => echo@haar.student.utwente.nl,n,Playback(demo-echodone) ; Let them know it's over
exten => echo@haar.student.utwente.nl,n,Goto(s,6)               ; Start over

You can also set up a SIP bridge:

exten => tellme@haar.student.utwente.nl,1,Dial(SIP/18005558355@proxy01.sipphone.com)

• Asterisk sometimes doesn't choose the correct IPv4 address from the offers for the actual stream, resulting in no audio being received.

• The video test with the Collabora bot is completely untested with Psi and probably won't work.