Merging in xylome's beaerer capabilty patch (bug 3547)
authorMatthew Fredrickson <creslin@digium.com>
Fri, 1 Apr 2005 17:00:50 +0000 (17:00 +0000)
committerMatthew Fredrickson <creslin@digium.com>
Fri, 1 Apr 2005 17:00:50 +0000 (17:00 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5342 65c4cc65-6c06-0410-ace0-fbb531ad65f3

apps/Makefile
apps/app_dial.c
channel.c
channels/chan_zap.c
include/asterisk/channel.h
include/asterisk/transcap.h [new file with mode: 0755]

index 0feec31..2f6a875 100755 (executable)
@@ -31,7 +31,7 @@ APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_mp3.so\
      app_talkdetect.so app_alarmreceiver.so app_userevent.so app_verbose.so \
      app_test.so app_forkcdr.so app_math.so app_realtime.so \
      app_dumpchan.so app_waitforsilence.so app_while.so app_setrdnis.so \
-     app_md5.so app_readfile.so app_chanspy.so
+     app_md5.so app_readfile.so app_chanspy.so app_settransfercapability.so
 
 ifneq (${OSARCH},Darwin)
 ifneq (${OSARCH},SunOS)
index 9ee512c..9f7dedb 100755 (executable)
@@ -77,8 +77,8 @@ static char *descrip =
 "      'r' -- indicate ringing to the calling party, pass no audio until answered.\n"
 "      'm[(class)]' -- provide hold music to the calling party until answered (optionally\n"
 "                      with the specified class.\n"
-"      'M(x[^arg]) -- Executes the macro (x with ^ delim arg list) upon connect of the call.\n"
-"                     Also, the macro can set the MACRO_RESULT variable to do the following:\n"
+"      'M(x[^arg])' -- Executes the macro (x with ^ delim arg list) upon connect of the call.\n"
+"                      Also, the macro can set the MACRO_RESULT variable to do the following:\n"
 "                     -- ABORT - Hangup both legs of the call.\n"
 "                     -- CONGESTION - Behave as if line congestion was encountered.\n"
 "                     -- BUSY - Behave as if a busy signal was encountered. (n+101)\n"
@@ -1039,8 +1039,8 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
                tmp->chan->cid.cid_tns = chan->cid.cid_tns;
                /* Presense of ADSI CPE on outgoing channel follows ours */
                tmp->chan->adsicpe = chan->adsicpe;
-               /* pass the digital flag */
-               ast_copy_flags(tmp->chan, chan, AST_FLAG_DIGITAL);
+               /* Pass the transfer capability */
+               tmp->chan->transfercapability = chan->transfercapability;
 
                /* If we have an outbound group, set this peer channel to it */
                if (outbound_group)
index e621aec..460d1b7 100755 (executable)
--- a/channel.c
+++ b/channel.c
@@ -39,6 +39,7 @@
 #include <asterisk/utils.h>
 #include <asterisk/lock.h>
 #include <asterisk/app.h>
+#include <asterisk/transcap.h>
 #ifdef ZAPTEL_OPTIMIZATIONS
 #include <sys/ioctl.h>
 #ifdef __linux__
@@ -243,6 +244,25 @@ char *ast_state2str(int state)
        }
 }
 
+char *ast_transfercapability2str(int transfercapability)
+{
+       switch(transfercapability) {
+       case AST_TRANS_CAP_SPEECH:
+               return "SPEECH";
+       case AST_TRANS_CAP_DIGITAL:
+               return "DIGITAL";
+       case AST_TRANS_CAP_RESTRICTED_DIGITAL:
+               return "RESTRICTED_DIGITAL";
+       case AST_TRANS_CAP_3_1K_AUDIO:
+               return "3K1AUDIO";
+       case AST_TRANS_CAP_DIGITAL_W_TONES:
+               return "DIGITAL_W_TONES";
+       case AST_TRANS_CAP_VIDEO:
+               return "VIDEO";
+       default:
+               return "UNKNOWN";
+       }
+}
 
 int ast_best_codec(int fmts)
 {
index c68f0ff..8eb8911 100755 (executable)
@@ -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;
index 92c5000..4d121bb 100755 (executable)
@@ -315,6 +315,9 @@ struct ast_channel {
        /*! channel flags of AST_FLAG_ type */
        unsigned int flags;
        
+       /* ISDN Transfer Capbility - AST_FLAG_DIGITAL is not enough */
+       unsigned short transfercapability;
+
        struct ast_frame *readq;
        int alertpipe[2];
        /*! Write translation path */
@@ -338,7 +341,8 @@ struct ast_channel {
 /* Channels have this property if they can accept input with jitter; i.e. most VoIP channels */
 #define AST_CHAN_TP_WANTSJITTER        (1 << 0)        
 
-#define AST_FLAG_DIGITAL       (1 << 0)        /* if the call is a digital ISDN call */
+/* This flag has been deprecated by the transfercapbilty data member in struct ast_channel */
+/* #define AST_FLAG_DIGITAL    (1 << 0) */     /* if the call is a digital ISDN call */
 #define AST_FLAG_DEFER_DTMF    (1 << 1)        /* if dtmf should be deferred */
 #define AST_FLAG_WRITE_INT     (1 << 2)        /* if write should be interrupt generator */
 #define AST_FLAG_BLOCKING      (1 << 3)        /* if we are blocking */
@@ -811,6 +815,15 @@ int ast_channel_masquerade(struct ast_channel *original, struct ast_channel *clo
  */
 char *ast_state2str(int state);
 
+/*! Gives the string form of a given transfer capability */
+/*!
+ * \param transercapability transfercapabilty to get the name of
+ * Give a name to a transfercapbility
+ * See above
+ * Returns the text form of the binary transfer capbility
+ */
+char *ast_transfercapability2str(int transfercapability);
+
 /* Options: Some low-level drivers may implement "options" allowing fine tuning of the
    low level channel.  See frame.h for options.  Note that many channel drivers may support
    none or a subset of those features, and you should not count on this if you want your
diff --git a/include/asterisk/transcap.h b/include/asterisk/transcap.h
new file mode 100755 (executable)
index 0000000..1a77dd1
--- /dev/null
@@ -0,0 +1,33 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * General Asterisk channel definitions.
+ * 
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Matthew Fredrickson <creslin@digium.com>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#ifndef _ASTERISK_TRANSCAP_H
+#define _ASTERISK_TRANSCAP_H
+
+/* These definitions are taken directly out of libpri.h and used here.
+ * DO NOT change them as it will cause unexpected behavior in channels
+ * that utilize these fields.
+ */
+
+#define AST_TRANS_CAP_SPEECH                           0x0
+#define AST_TRANS_CAP_DIGITAL                          0x08
+#define AST_TRANS_CAP_RESTRICTED_DIGITAL               0x09
+#define AST_TRANS_CAP_3_1K_AUDIO                       0x10
+#define AST_TRANS_CAP_7K_AUDIO                         0x11    /* Depriciated ITU Q.931 (05/1998)*/
+#define AST_TRANS_CAP_DIGITAL_W_TONES                  0x11
+#define AST_TRANS_CAP_VIDEO                            0x18
+
+#define IS_DIGITAL(cap)\
+       (cap) & AST_TRANS_CAP_DIGITAL ? 1 : 0
+
+#endif /* _ASTERISK_TRANSCAP_H */