Merging in xylome's beaerer capabilty patch (bug 3547)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Matthew Fredrickson
2005-04-01 17:00:50 +00:00
parent 41c684c4a3
commit b81f233e68
6 changed files with 84 additions and 43 deletions

View File

@@ -38,6 +38,7 @@
#include <asterisk/causes.h>
#include <asterisk/term.h>
#include <asterisk/utils.h>
#include <asterisk/transcap.h>
#include <sys/signal.h>
#include <errno.h>
#include <stdlib.h>
@@ -1866,13 +1867,15 @@ static int zt_call(struct ast_channel *ast, char *rdest, int timeout)
ast_log(LOG_DEBUG, "I'm being setup with no bearer right now...\n");
pri_set_crv(p->pri->pri, p->call, p->channel, 0);
}
p->digital = ast_test_flag(ast,AST_FLAG_DIGITAL);
p->digital = IS_DIGITAL(ast->transfercapability);
pri_sr_set_channel(sr, p->bearer ? PVT_TO_CHANNEL(p->bearer) : PVT_TO_CHANNEL(p),
p->pri->nodetype == PRI_NETWORK ? 0 : 1, 1);
pri_sr_set_bearer(sr, p->digital ? PRI_TRANS_CAP_DIGITAL : PRI_TRANS_CAP_SPEECH,
pri_sr_set_bearer(sr, p->digital ? PRI_TRANS_CAP_DIGITAL : ast->transfercapability,
(p->digital ? -1 :
((p->law == ZT_LAW_ALAW) ? PRI_LAYER_1_ALAW : PRI_LAYER_1_ULAW)));
dp_strip = 0;
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Requested transfer capability: 0x%.2x - %s\n", ast->transfercapability, ast_transfercapability2str(ast->transfercapability));
dp_strip = 0;
pridialplan = p->pri->dialplan - 1;
if (pridialplan == -2) { /* compute dynamically */
if (strncmp(c + p->stripmsd, p->pri->internationalprefix, strlen(p->pri->internationalprefix)) == 0) {
@@ -4604,35 +4607,7 @@ static int zt_indicate(struct ast_channel *chan, int condition)
return res;
}
#ifdef ZAPATA_PRI
static void set_calltype(struct ast_channel *chan, int ctype)
{
char *s = "UNKNOWN";
switch(ctype) {
case PRI_TRANS_CAP_SPEECH:
s = "SPEECH";
break;
case PRI_TRANS_CAP_DIGITAL:
s = "DIGITAL";
break;
case PRI_TRANS_CAP_RESTRICTED_DIGITAL:
s = "RESTRICTED_DIGITAL";
break;
case PRI_TRANS_CAP_3_1K_AUDIO:
s = "31KAUDIO";
break;
case PRI_TRANS_CAP_7K_AUDIO:
s = "7KAUDIO";
break;
case PRI_TRANS_CAP_VIDEO:
s = "VIDEO";
break;
}
pbx_builtin_setvar_helper(chan, "CALLTYPE", s);
}
#endif
static struct ast_channel *zt_new(struct zt_pvt *i, int state, int startpbx, int index, int law, int ctype)
static struct ast_channel *zt_new(struct zt_pvt *i, int state, int startpbx, int index, int law, int transfercapability)
{
struct ast_channel *tmp;
int deflaw;
@@ -4764,14 +4739,14 @@ static struct ast_channel *zt_new(struct zt_pvt *i, int state, int startpbx, int
tmp->cid.cid_pres = i->callingpres;
tmp->cid.cid_ton = i->cid_ton;
#ifdef ZAPATA_PRI
set_calltype(tmp, ctype);
tmp->transfercapability = transfercapability;
pbx_builtin_setvar_helper(tmp, "TRANSFERCAPABILITY", ast_transfercapability2str(transfercapability));
if (transfercapability & PRI_TRANS_CAP_DIGITAL) {
i->digital = 1;
}
/* Assume calls are not idle calls unless we're told differently */
i->isidlecall = 0;
i->alreadyhungup = 0;
if (ctype & PRI_TRANS_CAP_DIGITAL) {
i->digital = 1;
ast_set_flag(tmp, AST_FLAG_DIGITAL);
}
#endif
/* clear the fake event in case we posted one before we had ast_chanenl */
i->fake_event = 0;