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pkgsrc/net/kiax/patches/patch-ab

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Changes 0.8.5: * Updated Bulgarian, French, Polish, Spanish, German, Italian translations * Fixed possible crash on systray click * Fixed accidental dialing on group collapse * Fixed DTMF from keyboard - invalid keys are filtered * Cleaner disconnect (unregister all reigstered accounts) * Contacts are now grouped in account folders * New iaxclient with tipix Echo Cancellation * Added support for Italian, Spanish and Hebrew languages * While calling just press number keys to send DTMF (no need to switch to dialpad) * Call register shows call duration * Added icons for registration status tracking * Added options for muting incoming and outgoing rings * Added option to run command on incoming call * Fixed bugs.
2006-04-14 11:56:03 +02:00
$NetBSD: patch-ab,v 1.3 2006/04/14 09:56:03 adam Exp $
Initial import of kiax 0.8.4 In the process of the development of Asterisk a new protocol has gathered the attention among the VoIP users - the Inter-Asterisk eXchange, or IAX (TM), Protocol, used as a the native communication protocol between Asterisk PBX Servers. What is particularly good in it is that it requires only one UDP port per endpoint to create a successful communication channel for VoIP calls. This makes it much friendlier for users behind NAT, which is not the case with SIP and H.323 - they required specific router configurations for their channel establishment, thus hindering faster spread of VoIP on the home and corporate desktop.
2006-01-25 01:00:03 +01:00
Changes 0.8.5: * Updated Bulgarian, French, Polish, Spanish, German, Italian translations * Fixed possible crash on systray click * Fixed accidental dialing on group collapse * Fixed DTMF from keyboard - invalid keys are filtered * Cleaner disconnect (unregister all reigstered accounts) * Contacts are now grouped in account folders * New iaxclient with tipix Echo Cancellation * Added support for Italian, Spanish and Hebrew languages * While calling just press number keys to send DTMF (no need to switch to dialpad) * Call register shows call duration * Added icons for registration status tracking * Added options for muting incoming and outgoing rings * Added option to run command on incoming call * Fixed bugs.
2006-04-14 11:56:03 +02:00
--- lib/portaudio/pa_unix_oss/pa_unix_oss.c.orig 2006-02-05 00:51:06.000000000 +0100
Initial import of kiax 0.8.4 In the process of the development of Asterisk a new protocol has gathered the attention among the VoIP users - the Inter-Asterisk eXchange, or IAX (TM), Protocol, used as a the native communication protocol between Asterisk PBX Servers. What is particularly good in it is that it requires only one UDP port per endpoint to create a successful communication channel for VoIP calls. This makes it much friendlier for users behind NAT, which is not the case with SIP and H.323 - they required specific router configurations for their channel establishment, thus hindering faster spread of VoIP on the home and corporate desktop.
2006-01-25 01:00:03 +01:00
+++ lib/portaudio/pa_unix_oss/pa_unix_oss.c
@@ -43,7 +43,7 @@
#ifdef __linux__
#include <linux/soundcard.h>
#else
Changes 0.8.5: * Updated Bulgarian, French, Polish, Spanish, German, Italian translations * Fixed possible crash on systray click * Fixed accidental dialing on group collapse * Fixed DTMF from keyboard - invalid keys are filtered * Cleaner disconnect (unregister all reigstered accounts) * Contacts are now grouped in account folders * New iaxclient with tipix Echo Cancellation * Added support for Italian, Spanish and Hebrew languages * While calling just press number keys to send DTMF (no need to switch to dialpad) * Call register shows call duration * Added icons for registration status tracking * Added options for muting incoming and outgoing rings * Added option to run command on incoming call * Fixed bugs.
2006-04-14 11:56:03 +02:00
-#if defined(__FreeBSD__) || defined(__NetBSD__)
Add DragonFly support.
2006-02-16 21:53:23 +01:00
+#if defined(__FreeBSD__) || defined(__NetBSD__) || defined(__DragonFly__)
Initial import of kiax 0.8.4 In the process of the development of Asterisk a new protocol has gathered the attention among the VoIP users - the Inter-Asterisk eXchange, or IAX (TM), Protocol, used as a the native communication protocol between Asterisk PBX Servers. What is particularly good in it is that it requires only one UDP port per endpoint to create a successful communication channel for VoIP calls. This makes it much friendlier for users behind NAT, which is not the case with SIP and H.323 - they required specific router configurations for their channel establishment, thus hindering faster spread of VoIP on the home and corporate desktop.
2006-01-25 01:00:03 +01:00
#include <sys/soundcard.h>
#else
#include <machine/soundcard.h> /* JH20010905 */
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