[svn-commits] tilghman: branch 1.4 r134976 - in /branches/1.4: ./ main/ pbx/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Jul 31 16:53:20 CDT 2008
Author: tilghman
Date: Thu Jul 31 16:53:19 2008
New Revision: 134976
URL: http://svn.digium.com/view/asterisk?view=rev&rev=134976
Log:
Specify codecs in callfiles and manager, to allow video calls to be set up
from callfiles and AMI.
(closes issue #9531)
Reported by: Geisj
Patches:
20080715__bug9531__1.4.diff.txt uploaded by Corydon76 (license 14)
20080715__bug9531__1.6.0.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
Modified:
branches/1.4/main/manager.c
branches/1.4/pbx/pbx_spool.c
branches/1.4/sample.call
Modified: branches/1.4/main/manager.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/main/manager.c?view=diff&rev=134976&r1=134975&r2=134976
==============================================================================
--- branches/1.4/main/manager.c (original)
+++ branches/1.4/main/manager.c Thu Jul 31 16:53:19 2008
@@ -75,6 +75,7 @@
char tech[AST_MAX_EXTENSION];
char data[AST_MAX_EXTENSION];
int timeout;
+ int format;
char app[AST_MAX_APP];
char appdata[AST_MAX_EXTENSION];
char cid_name[AST_MAX_EXTENSION];
@@ -1783,12 +1784,12 @@
char requested_channel[AST_CHANNEL_NAME];
if (!ast_strlen_zero(in->app)) {
- res = ast_pbx_outgoing_app(in->tech, AST_FORMAT_SLINEAR, in->data, in->timeout, in->app, in->appdata, &reason, 1,
+ res = ast_pbx_outgoing_app(in->tech, in->format, in->data, in->timeout, in->app, in->appdata, &reason, 1,
S_OR(in->cid_num, NULL),
S_OR(in->cid_name, NULL),
in->vars, in->account, &chan);
} else {
- res = ast_pbx_outgoing_exten(in->tech, AST_FORMAT_SLINEAR, in->data, in->timeout, in->context, in->exten, in->priority, &reason, 1,
+ res = ast_pbx_outgoing_exten(in->tech, in->format, in->data, in->timeout, in->context, in->exten, in->priority, &reason, 1,
S_OR(in->cid_num, NULL),
S_OR(in->cid_name, NULL),
in->vars, in->account, &chan);
@@ -1851,6 +1852,7 @@
const char *appdata = astman_get_header(m, "Data");
const char *async = astman_get_header(m, "Async");
const char *id = astman_get_header(m, "ActionID");
+ const char *codecs = astman_get_header(m, "Codecs");
struct ast_variable *vars = astman_get_variables(m);
char *tech, *data;
char *l = NULL, *n = NULL;
@@ -1860,6 +1862,7 @@
int reason = 0;
char tmp[256];
char tmp2[256];
+ int format = AST_FORMAT_SLINEAR;
pthread_t th;
pthread_attr_t attr;
@@ -1895,6 +1898,10 @@
ast_shrink_phone_number(l);
if (ast_strlen_zero(l))
l = NULL;
+ }
+ if (!ast_strlen_zero(codecs)) {
+ format = 0;
+ ast_parse_allow_disallow(NULL, &format, codecs, 1);
}
if (ast_true(async)) {
struct fast_originate_helper *fast = ast_calloc(1, sizeof(*fast));
@@ -1915,6 +1922,7 @@
ast_copy_string(fast->context, context, sizeof(fast->context));
ast_copy_string(fast->exten, exten, sizeof(fast->exten));
ast_copy_string(fast->account, account, sizeof(fast->account));
+ fast->format = format;
fast->timeout = to;
fast->priority = pi;
pthread_attr_init(&attr);
@@ -1928,10 +1936,10 @@
pthread_attr_destroy(&attr);
}
} else if (!ast_strlen_zero(app)) {
- res = ast_pbx_outgoing_app(tech, AST_FORMAT_SLINEAR, data, to, app, appdata, &reason, 1, l, n, vars, account, NULL);
+ res = ast_pbx_outgoing_app(tech, format, data, to, app, appdata, &reason, 1, l, n, vars, account, NULL);
} else {
if (exten && context && pi)
- res = ast_pbx_outgoing_exten(tech, AST_FORMAT_SLINEAR, data, to, context, exten, pi, &reason, 1, l, n, vars, account, NULL);
+ res = ast_pbx_outgoing_exten(tech, format, data, to, context, exten, pi, &reason, 1, l, n, vars, account, NULL);
else {
astman_send_error(s, m, "Originate with 'Exten' requires 'Context' and 'Priority'");
return 0;
Modified: branches/1.4/pbx/pbx_spool.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/pbx/pbx_spool.c?view=diff&rev=134976&r1=134975&r2=134976
==============================================================================
--- branches/1.4/pbx/pbx_spool.c (original)
+++ branches/1.4/pbx/pbx_spool.c Thu Jul 31 16:53:19 2008
@@ -78,6 +78,8 @@
int waittime;
/* PID which is currently calling */
long callingpid;
+ /* Formats (codecs) for this call */
+ int format;
/* What to connect to outgoing */
char tech[256];
@@ -115,6 +117,7 @@
o->priority = 1;
o->retrytime = 300;
o->waittime = 45;
+ o->format = AST_FORMAT_SLINEAR;
ast_set_flag(&o->options, SPOOL_FLAG_ALWAYS_DELETE);
}
@@ -190,6 +193,8 @@
ast_log(LOG_WARNING, "Invalid max retries at line %d of %s\n", lineno, fn);
o->maxretries = 0;
}
+ } else if (!strcasecmp(buf, "codecs")) {
+ ast_parse_allow_disallow(NULL, &o->format, c, 1);
} else if (!strcasecmp(buf, "context")) {
ast_copy_string(o->context, c, sizeof(o->context));
} else if (!strcasecmp(buf, "extension")) {
@@ -340,11 +345,11 @@
if (!ast_strlen_zero(o->app)) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Attempting call on %s/%s for application %s(%s) (Retry %d)\n", o->tech, o->dest, o->app, o->data, o->retries);
- res = ast_pbx_outgoing_app(o->tech, AST_FORMAT_SLINEAR, o->dest, o->waittime * 1000, o->app, o->data, &reason, 2 /* wait to finish */, o->cid_num, o->cid_name, o->vars, o->account, NULL);
+ res = ast_pbx_outgoing_app(o->tech, o->format, o->dest, o->waittime * 1000, o->app, o->data, &reason, 2 /* wait to finish */, o->cid_num, o->cid_name, o->vars, o->account, NULL);
} else {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Attempting call on %s/%s for %s@%s:%d (Retry %d)\n", o->tech, o->dest, o->exten, o->context,o->priority, o->retries);
- res = ast_pbx_outgoing_exten(o->tech, AST_FORMAT_SLINEAR, o->dest, o->waittime * 1000, o->context, o->exten, o->priority, &reason, 2 /* wait to finish */, o->cid_num, o->cid_name, o->vars, o->account, NULL);
+ res = ast_pbx_outgoing_exten(o->tech, o->format, o->dest, o->waittime * 1000, o->context, o->exten, o->priority, &reason, 2 /* wait to finish */, o->cid_num, o->cid_name, o->vars, o->account, NULL);
}
if (res) {
ast_log(LOG_NOTICE, "Call failed to go through, reason (%d) %s\n", reason, ast_channel_reason2str(reason));
Modified: branches/1.4/sample.call
URL: http://svn.digium.com/view/asterisk/branches/1.4/sample.call?view=diff&rev=134976&r1=134975&r2=134976
==============================================================================
--- branches/1.4/sample.call (original)
+++ branches/1.4/sample.call Thu Jul 31 16:53:19 2008
@@ -14,6 +14,9 @@
# would for the "Dial" application. Only one channel name is permitted.
#
Channel: Zap/1
+#
+# You can specify codecs for the call
+Codecs: alaw, speex, h264
#
# You may also specify a wait time (default is 45 seconds) for how long to
# wait for the channel to be answered, a retry time (default is 5 mins)
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