[Asterisk-cvs] asterisk/channels chan_zap.c,1.422,1.423
mattf at lists.digium.com
mattf at lists.digium.com
Fri Apr 1 11:07:28 CST 2005
Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv635/channels
Modified Files:
chan_zap.c
Log Message:
Merging in xylome's beaerer capabilty patch (bug 3547)
Index: chan_zap.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_zap.c,v
retrieving revision 1.422
retrieving revision 1.423
diff -u -d -r1.422 -r1.423
--- chan_zap.c 28 Mar 2005 20:48:24 -0000 1.422
+++ chan_zap.c 1 Apr 2005 17:00:50 -0000 1.423
@@ -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 @@
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 @@
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 @@
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;
More information about the svn-commits
mailing list