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--- en:jingle_bots [2010/10/19 18:42] – z_god
+++ en:jingle_bots [2019/04/28 13:51] (current) – external edit 127.0.0.1
@@ Line 1: / Line 1: @@
-#summary|Jingle bots, offering Asterisk services to Psi/Psi+ (en)
+====== Jingle bots, offering Asterisk services to Psi+ ======
-= Introduction =
+===== Introduction =====
 This page contains information on using Asterisk with Psi+.
+===== Details =====
-= 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 =====
-= 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, <del>Google Talk, QIP,</del> Nokia N900, SIP ||
-|| 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, <del>Google Talk, QIP,</del> 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, <del>Google Talk, QIP,</del> Nokia N900 ||
-|| 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 [[http://www.collabora.co.uk/|Collabora]] with several tests including an echo and a video test || Psi, Psi+ ||
+===== How to use =====
-= 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):
-= Setting up Asterisk for use with Psi+ =
-  * Check out the special Jingle branch of Asterisk:
 <code>
 svn -r 281181 co http://svn.digium.com/svn/asterisk/team/phsultan/jingle-support/
 </code>
@@ Line 84: / Line 39: @@
 <code>
 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
 </code>
@@ Line 98: / Line 47: @@
 <code>
 cd jingle-support
 patch -p1 < ../asterisk-jingle-hacks-20100430.patch.txt
 patch -p0 < ../patch-jingle_speex-1.diff
 </code>
   * Compile and install Asterisk as usual, make sure res_jabber and chan_jingle are enabled
 <code>
 ./configure
 make menuselect
 make
 make install
 make samples
 </code>
+  * Register a XMPP account (using Psi for example, you can remove it from Psi afterwards)
-  * Register a Jabber account (using Psi for example, you can remove it from Psi afterwards)
   * Edit /etc/asterisk/jabber.conf and add the new account:
 <code>
 [echo]]
 type=client
 serverhost=localhost
 username=echo@haar.student.utwente.nl/asterisk
 secret=<password>
 port=5222
 usetls=no
 usesasl=no
 statusmessage=voice: echo test
 </code>
+Note that in the above case, Asterisk is running on the same server as the XMPP daemon.
-    Note that in the above case, Asterisk is running on the same server as the Jabber daemon.
   * Edit /etc/asterisk/jingle.conf:
-    Make sure speex is allowed, the two lines below should work
+Make sure speex is allowed, the two lines below should work
 <code>
 disallow=all
 allow=speex
 </code>
+Take note of the context or change it to something like public
-    Take note of the context or change it to something like public
 <code>
 context=public
 </code>
   * Edit /etc/asterisk/extensions.conf:
-    Edit or create a new context and set your dialing rules:
+Edit or create a new context and set your dialing rules:
-    This is the example for an echo test:
+This is the example for an echo test:
 <code>
 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
 </code>
+You can also set up a SIP bridge:
-    You can also set up a SIP bridge:
 <code>
 exten => tellme@haar.student.utwente.nl,1,Dial(SIP/18005558355@proxy01.sipphone.com)
 </code>
+===== Known issues =====
-= 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.