Resurrect net/asterisk14-addons and upgrade to latest 1.4.13. Assign

maintainership to bamby@sippysoft.com, apparently old maintainer has no
interest in maintaining this.
This commit is contained in:
Maxim Sobolev 2011-11-09 01:28:30 +00:00
parent 983186f367
commit 42a521c1c4
Notes: svn2git 2021-03-31 03:12:20 +00:00
svn path=/head/; revision=285317
7 changed files with 293 additions and 0 deletions

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@ -30,6 +30,7 @@
SUBDIR += asterisk
SUBDIR += asterisk10
SUBDIR += asterisk14
SUBDIR += asterisk14-addons
SUBDIR += asterisk16
SUBDIR += asterisk16-addons
SUBDIR += atmsupport

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@ -0,0 +1,49 @@
# New ports collection makefile for: asterisk-addons
# Date created: 23 March 2006
# Whom: Fabian Gast <fgast@only640k.org>
#
# $FreeBSD$
#
PORTNAME= asterisk14
PORTVERSION= 1.4.13
CATEGORIES= net
MASTER_SITES= http://downloads.asterisk.org/pub/telephony/asterisk/ \
http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/
PKGNAMESUFFIX= -addons
DISTNAME= asterisk${PKGNAMESUFFIX}-${DISTVERSION}
MAINTAINER= bamby@sippysoft.com
COMMENT= GPL modules for the Asterisk Open Source PBX
BUILD_DEPENDS= asterisk:${PORTSDIR}/net/asterisk14
RUN_DEPENDS= asterisk:${PORTSDIR}/net/asterisk14
OPTIONS= SAMPLE_CONFIG "Install sample configuration files" off
ONLY_FOR_ARCHS= i386 amd64 powerpc
USE_AUTOTOOLS= libtool
USE_MYSQL= yes
USE_GMAKE= yes
GNU_CONFIGURE= yes
.include <bsd.port.pre.mk>
.if defined(WITH_SAMPLE_CONFIG)
PLIST_SUB+= SAMPLE_CONFIG=""
.else
PLIST_SUB+= SAMPLE_CONFIG="@comment "
.endif
pre-configure:
if ${GREP} -sq fixup_codecs ${LOCALBASE}/include/asterisk/channel.h; \
then \
${PATCH} ${PATCH_ARGS} < ${PATCHDIR}/codecnego.diff; \
fi
post-install:
.if defined(WITH_SAMPLE_CONFIG)
@cd ${WRKSRC} && ${GMAKE} samples
.endif
.include <bsd.port.post.mk>

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@ -0,0 +1,2 @@
SHA256 (asterisk-addons-1.4.13.tar.gz) = 4041b4a3071ce64537ad97d0961e69dc699b59436da9bb6aaf6a6394bfc8e967
SIZE (asterisk-addons-1.4.13.tar.gz) = 825032

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@ -0,0 +1,215 @@
--- channels/chan_ooh323.c.orig 2007-07-21 15:17:52.000000000 +0800
+++ channels/chan_ooh323.c 2008-02-13 15:37:17.000000000 +0800
@@ -42,8 +42,8 @@
/* Channel Definition */
-static struct ast_channel *ooh323_request(const char *type, int format,
- void *data, int *cause);
+static struct ast_channel *ooh323_request(const char *type, const struct ast_codec_pref *format, void *data, int *cause);
+
static int ooh323_digit_begin(struct ast_channel *ast, char digit);
static int ooh323_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
static int ooh323_call(struct ast_channel *ast, char *dest, int timeout);
@@ -57,8 +57,7 @@
static enum ast_rtp_get_result ooh323_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
static enum ast_rtp_get_result ooh323_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
-static int ooh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp,
- struct ast_rtp *vrtp, int codecs, int nat_active);
+static int ooh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, const struct ast_codec_pref *codecs, int nat_active);
static void print_codec_to_cli(int fd, struct ast_codec_pref *pref);
static void ast_ooh323c_exit();
@@ -244,6 +243,7 @@
{
struct ast_channel *ch = NULL;
int fmt;
+ int cap;
if(gH323Debug)
ast_log(LOG_DEBUG, "--- ooh323_new - %s\n", host);
@@ -258,10 +258,11 @@
ast_mutex_lock(&ch->lock);
ch->tech = &ooh323_tech;
- ch->nativeformats = i->capability;
-
+ ast_codec_pref_init(&ch->nativeformats);
+ ast_codec_pref_append_missing2(&ch->nativeformats, i->capability);
- fmt = ast_best_codec(ch->nativeformats);
+ cap = ast_codec_pref_bits(&ch->nativeformats);
+ fmt = ast_best_codec(cap);
ch->fds[0] = ast_rtp_fd(i->rtp);
ch->fds[1] = ast_rtcp_fd(i->rtp);
@@ -451,8 +452,7 @@
/*
Possible data values - peername, exten/peername, exten@ip
*/
-static struct ast_channel *ooh323_request(const char *type, int format,
- void *data, int *cause)
+static struct ast_channel *ooh323_request(const char *type, const struct ast_codec_pref *formats, void *data, int *cause)
{
struct ast_channel *chan=NULL;
@@ -461,22 +461,11 @@
char *dest = NULL;
char *ext = NULL;
char tmp[256];
- char formats[512];
- int oldformat;
char *sport = NULL;
int port = 0;
if(gH323Debug)
- ast_verbose("--- ooh323_request - data %s format %s\n", (char*)data,
- ast_getformatname_multiple(formats,512,format) );
-
- oldformat = format;
- format &= ((AST_FORMAT_MAX_AUDIO << 1) - 1);
- if (!format) {
- ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format "
- "'%d'\n", format);
- return NULL;
- }
+ ast_verbose("--- ooh323_request - data %s format %s\n", (char*)data, ast_codec_pref_dump(tmp, sizeof(tmp), formats) );
p = ooh323_alloc(0,0); /* Initial callRef is zero */
@@ -536,8 +525,8 @@
if(ext)
strncpy(p->exten, ext, sizeof(p->exten)-1);
- if(peer->capability & format)
- p->capability = peer->capability & format;
+ if(peer->capability & formats->audio_bits)
+ p->capability = peer->capability & formats->audio_bits;
else{
p->capability = peer->capability;
}
@@ -971,11 +960,11 @@
if(f->frametype == AST_FRAME_VOICE)
{
- if(!(f->subclass & ast->nativeformats))
+ if(!(f->subclass & ast->nativeformats.audio_bits))
{
ast_log(LOG_WARNING, "Asked to transmit frame type %d, while native "
"formats is %d (read/write = %d/%d)\n",
- f->subclass, ast->nativeformats, ast->readformat,
+ f->subclass, ast->nativeformats.audio_bits, ast->readformat,
ast->writeformat);
return 0;
}
@@ -2219,12 +2208,12 @@
int x, codec;
for(x = 0; x < 32 ; x++) {
- codec = ast_codec_pref_index(pref, x);
+ codec = ast_codec_pref_index_audio(pref, x);
if (!codec)
break;
ast_cli(fd, "%s", ast_getformatname(codec));
- ast_cli(fd, ":%d", pref->framing[x]);
- if (x < 31 && ast_codec_pref_index(pref, x + 1))
+ ast_cli(fd, ":%d", pref->audio_framing[x]);
+ if (x < 31 && ast_codec_pref_index_audio(pref, x + 1))
ast_cli(fd, ",");
}
if (!x)
@@ -3020,13 +3009,13 @@
(ooCallData *call, struct ast_codec_pref *prefs)
{
int i=0;
- int codec = ast_codec_pref_index(prefs, i);
+ int codec = ast_codec_pref_index_audio(prefs, i);
ooResetCapPrefs(call);
while(codec)
{
ooAppendCapToCapPrefs(call, ooh323_convertAsteriskCapToH323Cap(codec));
- codec = ast_codec_pref_index(prefs, ++i);
+ codec = ast_codec_pref_index_audio(prefs, ++i);
}
return 0;
@@ -3058,7 +3047,7 @@
static int ooh323_set_rtp_peer
(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp,
- int codecs, int nat_active)
+ const struct ast_codec_pref *codecs, int nat_active)
{
/* XXX Deal with Video */
struct ooh323_pvt *p;
@@ -3106,7 +3095,7 @@
strncpy(mediaInfo.lMediaIP, ast_inet_ntoa(us.sin_addr), sizeof(mediaInfo.lMediaIP)-1);
mediaInfo.lMediaPort = ntohs(us.sin_port);
mediaInfo.lMediaCntrlPort = mediaInfo.lMediaPort +1;
- for(x=0; 0 != (format=ast_codec_pref_index(&p->prefs, x)); x++)
+ for(x=0; 0 != (format=ast_codec_pref_index_audio(&p->prefs, x)); x++)
{
strcpy(mediaInfo.dir, "transmit");
mediaInfo.cap = ooh323_convertAsteriskCapToH323Cap(format);
@@ -3250,10 +3239,11 @@
/* We already hold the channel lock */
if (f->frametype == AST_FRAME_VOICE)
{
- if (f->subclass != p->owner->nativeformats)
+ if (!(f->subclass & p->owner->nativeformats.audio_bits))
{
ast_log(LOG_DEBUG, "Oooh, format changed to %d\n", f->subclass);
- p->owner->nativeformats = f->subclass;
+ ast_codec_pref_append(&p->owner->nativeformats, f->subclass);
+ p->capability |= f->subclass;
ast_set_read_format(p->owner, p->owner->readformat);
ast_set_write_format(p->owner, p->owner->writeformat);
}
--- channels/ooh323cDriver.c.orig 2007-01-20 06:27:05.000000000 +0800
+++ channels/ooh323cDriver.c 2008-02-11 16:47:24.000000000 +0800
@@ -66,7 +66,7 @@
if(gH323Debug)
ast_verbose("\tAdding capabilities to H323 endpoint\n");
- for(x=0; 0 != (format=ast_codec_pref_index(prefs, x)); x++)
+ for(x=0; 0 != (format=ast_codec_pref_index_audio(prefs, x)); x++)
{
if(format & AST_FORMAT_ULAW)
{
@@ -172,14 +172,14 @@
ret |= ooCallEnableDTMFH245Signal(call);
- for(x=0; 0 !=(format=ast_codec_pref_index(prefs, x)); x++)
+ for(x=0; 0 !=(format=ast_codec_pref_index_audio(prefs, x)); x++)
{
if(format & AST_FORMAT_ULAW)
{
if(gH323Debug)
ast_verbose("\tAdding g711 ulaw capability to call(%s, %s)\n",
call->callType, call->callToken);
- txframes = prefs->framing[x];
+ txframes = prefs->audio_framing[x];
ret= ooCallAddG711Capability(call, OO_G711ULAW64K, txframes,
grxframes, OORXANDTX,
&ooh323c_start_receive_channel,
@@ -192,7 +192,7 @@
if(gH323Debug)
ast_verbose("\tAdding g711 alaw capability to call(%s, %s)\n",
call->callType, call->callToken);
- txframes = prefs->framing[x];
+ txframes = prefs->audio_framing[x];
ret= ooCallAddG711Capability(call, OO_G711ALAW64K, txframes,
grxframes, OORXANDTX,
&ooh323c_start_receive_channel,
@@ -206,7 +206,7 @@
if(gH323Debug)
ast_verbose("\tAdding g729A capability to call(%s, %s)\n",
call->callType, call->callToken);
- txframes = (prefs->framing[x])/10;
+ txframes = (prefs->audio_framing[x])/10;
ret= ooCallAddG729Capability(call, OO_G729A, txframes, 24,
OORXANDTX, &ooh323c_start_receive_channel,
&ooh323c_start_transmit_channel,

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@ -0,0 +1,11 @@
--- Makefile.orig Thu Feb 14 00:58:11 2008
+++ Makefile Fri Feb 22 19:17:58 2008
@@ -27,7 +27,7 @@
# Overwite config files on "make samples"
OVERWRITE:=y
-ASTCFLAGS+=-fPIC
+ASTCFLAGS+=-fPIC -I${PREFIX}/include
#NOISY_BUILD=yes

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@ -0,0 +1,5 @@
This package contains additional modules for the Asterisk Open Source PBX
which are, not included in the normal base distribution, becouse of licence
incompatilities.
WWW: http://www.asteriskpbx.com

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@ -0,0 +1,10 @@
lib/asterisk/modules/app_addon_sql_mysql.so
lib/asterisk/modules/app_saycountpl.so
lib/asterisk/modules/cdr_addon_mysql.so
lib/asterisk/modules/res_config_mysql.so
lib/asterisk/modules/format_mp3.so
lib/asterisk/modules/chan_ooh323.so
%%SAMPLE_CONFIG%%etc/asterisk/cdr_mysql.conf
%%SAMPLE_CONFIG%%etc/asterisk/res_mysql.conf
%%SAMPLE_CONFIG%%etc/asterisk/ooh323.conf
%%SAMPLE_CONFIG%%etc/asterisk/mysql.conf