[Freeswitch-svn] [commit] r11591 - freeswitch/branches/gmaruzz/mod_skypiax/asterisk

[FreeSWITCH SVN]

Author: gmaruzz
Date: Sun Feb  1 18:59:20 2009
New Revision: 11591

Log:
skypiax: on *, incoming and outbound calls, multichannel. Concurrent calls work. Audio only on the first call for each channel. Probably because the audio threads with the skype client are not killed when call is ended, etc etc. Anyway, it is working as a proof of concept. Now is too late, let's leave something for tomorrow night ;-)

Modified:
   freeswitch/branches/gmaruzz/mod_skypiax/asterisk/chan_skypiax.c
   freeswitch/branches/gmaruzz/mod_skypiax/asterisk/skypiax.conf

Modified: freeswitch/branches/gmaruzz/mod_skypiax/asterisk/chan_skypiax.c
==============================================================================
--- freeswitch/branches/gmaruzz/mod_skypiax/asterisk/chan_skypiax.c	(original)
+++ freeswitch/branches/gmaruzz/mod_skypiax/asterisk/chan_skypiax.c	Sun Feb  1 18:59:20 2009
@@ -684,6 +684,7 @@
     DEBUGA_PBX("skypiax_call dialing idest: %s, timeout: %d, dstr: %s!\n", SKYPIAX_P_LOG,
                idest, timeout, dstr);
 
+  strcpy(p->session_uuid_str, "dialing");
   status = skypiax_skype_call(p, dstr, timeout);
   if (status) {
     WARNINGA("skypiax_call dialing failed: %d!\n", SKYPIAX_P_LOG, status);
@@ -2203,6 +2204,11 @@
                   switch_core_session_rwunlock(session);
 
 #endif
+
+              if (p->owner) {
+                ast_queue_control(p->owner, AST_CONTROL_ANSWER);
+              }
+
                   NOTICA("HERE!\n", SKYPIAX_P_LOG);
 
 	return 0;

Modified: freeswitch/branches/gmaruzz/mod_skypiax/asterisk/skypiax.conf
==============================================================================
--- freeswitch/branches/gmaruzz/mod_skypiax/asterisk/skypiax.conf	(original)
+++ freeswitch/branches/gmaruzz/mod_skypiax/asterisk/skypiax.conf	Sun Feb  1 18:59:20 2009
@@ -50,6 +50,62 @@
 ;debug_monitorlocks=yes
 
 skype=yes ; legacy setting, leave it to yes
+X11_display=:101
+tcp_cli_port=11234
+tcp_srv_port=11235
+;;;;;;;;;;;;;;;;;;;;;;;;;
+;;;;;;;;;;;;;;;;;;;;;;;;;
+; audio boost settings, valid for all platforms, to compensate for different input/output audio signal levels
+; tweak it if you get horrible (or not hearable) sound
+;
+;boost can be positive or negative (-40 to +40) in db
+;experiment to find which values are best for your computer
+playback_boost=0 ; 
+capture_boost=0 ; 
+
+[skypeclient2] 
+;;;;;;;;;;;;;;;;;;;;;;;;;
+;;;;;;;;;;;;;;;;;;;;;;;;;
+; general settings, valid on all platforms
+;
+; 
+; Default language 
+; 
+language=en 
+; 
+; Default context (in extensions.conf, can be overridden with @context syntax) 
+; 
+context=default 
+; 
+; Default extension (in extensions.conf) where incoming calls land
+; 
+extension=s 
+; 
+;;;;;;;;;;;;;;;;;;;;;;;;;
+;;;;;;;;;;;;;;;;;;;;;;;;;
+; Debugging settings, valid globally for all interfaces on all platforms
+;
+; the debug values are global for all the interfaces.
+;
+; default is no skypiax debugging output, you **have** to activate debugging here to obtain debugging from skypiax
+;
+; To see the debugging output you have to "set debug 100" from the Asterisk CLI or launch
+; Asterisk with -ddddddddddd option, and have the logger.conf file activating debug info for console and messages
+;
+; You can activate each of the following separately, but you can't disactivate. Eg: debug_at=no does not subtract debug_at from debug_all
+; debug_all activate all possible debugging info
+;
+;debug_all=yes
+debug_skype=yes
+debug_pbx=yes
+;debug_sound=yes
+;debug_locks=yes
+;debug_monitorlocks=yes
+
+skype=yes ; legacy setting, leave it to yes
+X11_display=:102
+tcp_cli_port=11236
+tcp_srv_port=11237
 ;;;;;;;;;;;;;;;;;;;;;;;;;
 ;;;;;;;;;;;;;;;;;;;;;;;;;
 ; audio boost settings, valid for all platforms, to compensate for different input/output audio signal levels