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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.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Luigi Rizzo 2007-07-29 09:27:30 +00:00
parent 692e1fe0c4
commit e5f3a6ccdb
1 changed files with 27 additions and 15 deletions
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42
channels/chan_sip.c
View File

@ -831,6 +831,7 @@ struct sip_auth {
#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 @@ struct sip_auth {
#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 int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl)
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 = {
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 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
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;