Author: rizzo
Date: Sun Jul 29 04:27:30 2007
New Revision: 77682
URL: http://svn.digium.com/view/asterisk?view=rev&rev=77682
Log:
remove bit position from description of SIP_* flags.
use AST_FORMAT_AUDIO_MASK instead of playing with AST_FORMAT_MAX_AUDIO
to determine audio formats.
There is a dubious use of AST_FORMAT_MAX_AUDIO in sip_request_call()
which surely needs fixing, namely:
/* mask request with some set of allowed formats.
* XXX this needs to be fixed.
* The original code uses AST_FORMAT_AUDIO_MASK, but it is
* unclear what to use here. We have global_capabilities, which is
* configured from sip.conf, and sip_tech.capabilities, which is
* hardwired to all audio formats.
*/
The latter is possibly something to backport when fixed.
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL:
http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=77682&r1=77681&r2=77682
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Sun Jul 29 04:27:30 2007
@@ -831,6 +831,7 @@
#define SIP_PAGE2_RT_FROMCONTACT (1 << 4) /*!< P: ... */
#define SIP_PAGE2_RTSAVE_SYSNAME (1 << 5) /*!< G: Save system
name at registration? */
/* Space for addition of other realtime flags in the future */
+
#define SIP_PAGE2_IGNOREREGEXPIRE (1 << 10) /*!< G: Ignore
expiration of peer */
#define SIP_PAGE2_DYNAMIC (1 << 13) /*!< P: Dynamic Peers
register with Asterisk */
#define SIP_PAGE2_SELFDESTRUCT (1 << 14) /*!< P: Automatic peers
need to destruct themselves */
@@ -838,19 +839,22 @@
#define SIP_PAGE2_ALLOWSUBSCRIBE (1 << 16) /*!< GP: Allow
subscriptions from this peer? */
#define SIP_PAGE2_ALLOWOVERLAP (1 << 17) /*!< DP: Allow overlap
dialing ? */
#define SIP_PAGE2_SUBSCRIBEMWIONLY (1 << 18) /*!< GP: Only issue MWI
notification if subscribed to */
+
#define SIP_PAGE2_T38SUPPORT (7 << 20) /*!< GDP: T38 Fax
Passthrough Support */
-#define SIP_PAGE2_T38SUPPORT_UDPTL (1 << 20) /*!< GDP: 20: T38 Fax
Passthrough Support */
-#define SIP_PAGE2_T38SUPPORT_RTP (2 << 20) /*!< GDP: 21: T38 Fax
Passthrough Support (not implemented) */
-#define SIP_PAGE2_T38SUPPORT_TCP (4 << 20) /*!< GDP: 22: T38 Fax
Passthrough Support (not implemented) */
-#define SIP_PAGE2_CALL_ONHOLD (3 << 23) /*!< D: Call hold
states */
-#define SIP_PAGE2_CALL_ONHOLD_ACTIVE (1 << 23) /*!< D: 23: Active
hold */
-#define SIP_PAGE2_CALL_ONHOLD_ONEDIR (2 << 23) /*!< D: 23: One
directional hold */
-#define SIP_PAGE2_CALL_ONHOLD_INACTIVE (3 << 23) /*!< D: 23: Inactive
hold */
-#define SIP_PAGE2_RFC2833_COMPENSATE (1 << 25) /*!< DP: 25: Compensate
for buggy RFC2833 implementations */
-#define SIP_PAGE2_BUGGY_MWI (1 << 26) /*!< DP: 26: Buggy
CISCO MWI fix */
-#define SIP_PAGE2_NOTEXT (1 << 27) /*!< GPD: 27: Text not
supported */
-#define SIP_PAGE2_TEXTSUPPORT (1 << 28) /*!< GPD: 28: Global
text enable */
-#define SIP_PAGE2_OUTGOING_CALL (1 << 30) /*!< D: 30: Is this an
outgoing call? */
+#define SIP_PAGE2_T38SUPPORT_UDPTL (1 << 20) /*!< GDP: T38 Fax
Passthrough Support */
+#define SIP_PAGE2_T38SUPPORT_RTP (2 << 20) /*!< GDP: T38 Fax
Passthrough Support (not implemented) */
+#define SIP_PAGE2_T38SUPPORT_TCP (4 << 20) /*!< GDP: T38 Fax
Passthrough Support (not implemented) */
+
+#define SIP_PAGE2_CALL_ONHOLD (3 << 23) /*!< D: Call hold
states: */
+#define SIP_PAGE2_CALL_ONHOLD_ACTIVE (1 << 23) /*!< D: Active hold */
+#define SIP_PAGE2_CALL_ONHOLD_ONEDIR (2 << 23) /*!< D: One directional
hold */
+#define SIP_PAGE2_CALL_ONHOLD_INACTIVE (3 << 23) /*!< D: Inactive hold */
+
+#define SIP_PAGE2_RFC2833_COMPENSATE (1 << 25) /*!< DP: Compensate for
buggy RFC2833 implementations */
+#define SIP_PAGE2_BUGGY_MWI (1 << 26) /*!< DP: Buggy CISCO
MWI fix */
+#define SIP_PAGE2_NOTEXT (1 << 27) /*!< GDP: Text not
supported */
+#define SIP_PAGE2_TEXTSUPPORT (1 << 28) /*!< GDP: Global text
enable */
+#define SIP_PAGE2_OUTGOING_CALL (1 << 30) /*!< D: Is this an
outgoing call? */
#define SIP_PAGE2_FLAGS_TO_COPY \
(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP |
SIP_PAGE2_VIDEOSUPPORT | \
@@ -1752,7 +1756,7 @@
static const struct ast_channel_tech sip_tech = {
.type = "SIP",
.description = "Session Initiation Protocol (SIP)",
- .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
+ .capabilities = AST_FORMAT_AUDIO_MASK, /* all audio formats */
.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
.requester = sip_request_call, /* called with chan
unlocked */
.devicestate = sip_devicestate, /* called with chan
unlocked (not chan-specific) */
@@ -1781,7 +1785,7 @@
static const struct ast_channel_tech sip_tech_info = {
.type = "SIP",
.description = "Session Initiation Protocol (SIP)",
- .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
+ .capabilities = AST_FORMAT_AUDIO_MASK, /* all audio formats */
.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
.requester = sip_request_call,
.devicestate = sip_devicestate,
@@ -16570,7 +16574,15 @@
char *dest = data;
oldformat = format;
- if (!(format &= ((AST_FORMAT_MAX_AUDIO << 1) - 1))) {
+ /* mask request with some set of allowed formats.
+ * XXX this needs to be fixed.
+ * The original code uses AST_FORMAT_AUDIO_MASK, but it is
+ * unclear what to use here. We have global_capabilities, which is
+ * configured from sip.conf, and sip_tech.capabilities, which is
+ * hardwired to all audio formats.
+ */
+ format &= AST_FORMAT_AUDIO_MASK;
+ if (!format) {
ast_log(LOG_NOTICE, "Asked to get a channel of unsupported
format %s while capability is %s\n", ast_getformatname(oldformat),
ast_getformatname(global_capability));
*cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL; /* Can't find
codec to connect to host */
return NULL;
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