Jingle bots, offering Asterisk services to Psi+

Jingle bots, offering Asterisk services to Psi+

Introduction

This page contains information on using Asterisk with Psi+.

Details

To be able to use Asterisk with Psi+, you need to use the Jingle branch of Asterisk with some patches. (This patches on the Psi side are not necessary anymore.)

Public Bots

Here is a list of currently known public Jingle bots:

JID	notes	Works with (tested with)
echo@haar.student.utwente.nl	A simple echo test	Psi, Psi+, Pidgin, ~~Google Talk, QIP,~~ Nokia N900, SIP
psi-dev@haar.student.utwente.nl	A conference room to talk about Psi+ (psi+ and psiplus@haar.student.utwente.nl work too)	Psi, Psi+, Pidgin, ~~Google Talk, QIP,~~ Nokia N900, SIP
tellme@haar.student.utwente.nl	A SIP bridge to the Tell Me service (hint, say Black Jack from the main menu)	Psi, Psi+, Pidgin, ~~Google Talk, QIP,~~ Nokia N900
echo@test.collabora.co.uk	A test bot by Collabora with several tests including an echo and a video test	Psi, Psi+

How to use

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.

Setting up Asterisk for use with Psi+

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)

Known issues

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.

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