[asterisk-commits] branch russell/codetest r8996 - in
/team/russell/codetest: ./ apps/ channels/...
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Jan 31 09:56:11 MST 2006
Author: russell
Date: Tue Jan 31 10:56:07 2006
New Revision: 8996
URL: http://svn.digium.com/view/asterisk?rev=8996&view=rev
Log:
Merged revisions 8932,8938,8948-8949,8961,8976,8991 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r8932 | oej | 2006-01-30 15:36:38 -0500 (Mon, 30 Jan 2006) | 8 lines
- Doxygen and comments updates
- Moving structure declarations to top of file with the rest
- Adding some forward declarations for RTP interface functions
(All these changes to position in file are in preparation for splitting chan_sip up
into several files at some point in the future)
........
r8938 | mogorman | 2006-01-30 16:16:43 -0500 (Mon, 30 Jan 2006) | 3 lines
reverting blocks 9 and 10 from revision 7547
fixes bug 6080
........
r8948 | kpfleming | 2006-01-30 19:17:43 -0500 (Mon, 30 Jan 2006) | 2 lines
increment for recent ast_channel change
........
r8949 | russell | 2006-01-30 19:24:34 -0500 (Mon, 30 Jan 2006) | 3 lines
add a note to hopefully decrease the chance that someone forgets to increment
.cleancount after changing the ast_channel structure
........
r8961 | kpfleming | 2006-01-30 22:45:09 -0500 (Mon, 30 Jan 2006) | 2 lines
Yes Virginia, Zaptel does support native ALAW
........
r8976 | oej | 2006-01-31 09:30:09 -0500 (Tue, 31 Jan 2006) | 3 lines
- Change "prefs" to "default_prefs" and move declaration to "default" group
- Add doxygen comments
........
r8991 | oej | 2006-01-31 11:02:35 -0500 (Tue, 31 Jan 2006) | 4 lines
- Moving two session (PVT) flags to peer PAGE2 (DYNAMIC and SELFDESTRUCT) to make room for more session-related flags
This is needed for integrating patches in the bug tracker
- Adding doxygen comments
........
Modified:
team/russell/codetest/ (props changed)
team/russell/codetest/.cleancount
team/russell/codetest/apps/app_meetme.c
team/russell/codetest/channels/chan_sip.c
team/russell/codetest/channels/chan_zap.c
team/russell/codetest/include/asterisk/channel.h
Propchange: team/russell/codetest/
------------------------------------------------------------------------------
automerge = yes
Propchange: team/russell/codetest/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Jan 31 10:56:07 2006
@@ -1,1 +1,1 @@
-/trunk:1-8929
+/trunk:1-8993
Modified: team/russell/codetest/.cleancount
URL: http://svn.digium.com/view/asterisk/team/russell/codetest/.cleancount?rev=8996&r1=8995&r2=8996&view=diff
==============================================================================
--- team/russell/codetest/.cleancount (original)
+++ team/russell/codetest/.cleancount Tue Jan 31 10:56:07 2006
@@ -1,1 +1,1 @@
-8
+9
Modified: team/russell/codetest/apps/app_meetme.c
URL: http://svn.digium.com/view/asterisk/team/russell/codetest/apps/app_meetme.c?rev=8996&r1=8995&r2=8996&view=diff
==============================================================================
--- team/russell/codetest/apps/app_meetme.c (original)
+++ team/russell/codetest/apps/app_meetme.c Tue Jan 31 10:56:07 2006
@@ -426,8 +426,6 @@
unsigned char *data;
int len;
int res = -1;
- short *data2;
- int x;
if (!chan->_softhangup)
res = ast_autoservice_start(chan);
@@ -448,10 +446,7 @@
len = 0;
}
if (data) {
- data2 = alloca(len * 2);
- for (x=0;x<len;x++)
- data2[x] = AST_MULAW(data[x]);
- careful_write(conf->fd, (unsigned char *)data2, len << 1, 1);
+ careful_write(conf->fd, data, len, 1);
}
AST_LIST_UNLOCK(&confs);
Modified: team/russell/codetest/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/russell/codetest/channels/chan_sip.c?rev=8996&r1=8995&r2=8996&view=diff
==============================================================================
--- team/russell/codetest/channels/chan_sip.c (original)
+++ team/russell/codetest/channels/chan_sip.c Tue Jan 31 10:56:07 2006
@@ -34,6 +34,7 @@
* \todo Better support of forking
*
* \ingroup channel_drivers
+ *
*/
@@ -380,6 +381,7 @@
static int default_qualify; /*!< Default Qualify= setting */
static char default_vmexten[AST_MAX_EXTENSION];
static char default_musicclass[MAX_MUSICCLASS]; /*!< Global music on hold class */
+static struct ast_codec_pref default_prefs; /*!< Default codec prefs */
/* Global settings only apply to the channel */
static int global_rtautoclear = 120;
@@ -444,13 +446,12 @@
static int sip_reloading = FALSE; /*!< Flag for avoiding multiple reloads at the same time */
static enum channelreloadreason sip_reloadreason; /*!< Reason for last reload/load of configuration */
-static struct sched_context *sched;
-static struct io_context *io;
+static struct sched_context *sched; /*!< The scheduling context */
+static struct io_context *io; /*!< The IO context */
#define DEC_CALL_LIMIT 0
#define INC_CALL_LIMIT 1
-static struct ast_codec_pref prefs;
/*! \brief sip_request: The data grabbed from the UDP socket */
struct sip_request {
@@ -467,6 +468,13 @@
unsigned int flags; /*!< SIP_PKT Flags for this packet */
};
+/*! \brief structure used in transfers */
+struct sip_dual {
+ struct ast_channel *chan1;
+ struct ast_channel *chan2;
+ struct sip_request req;
+};
+
struct sip_pkt;
/*! \brief Parameters to the transmit_invite function */
@@ -481,11 +489,13 @@
enum sip_auth_type auth_type; /*!< Authentication type */
};
+/*! \brief Structure to save routing information for a SIP session */
struct sip_route {
struct sip_route *next;
char hop[0];
};
+/*! \brief Modes for SIP domain handling in the PBX */
enum domain_mode {
SIP_DOMAIN_AUTO, /*!< This domain is auto-configured */
SIP_DOMAIN_CONFIG, /*!< This domain is from configuration */
@@ -518,7 +528,9 @@
struct sip_auth *next; /*!< Next auth structure in list */
};
-/*--- Various flags for the flags field in the pvt structure */
+/*--- Various flags for the flags field in the pvt structure
+ Peer only flags should be set in PAGE2 below
+*/
#define SIP_ALREADYGONE (1 << 0) /*!< Whether or not we've already been destroyed by our peer */
#define SIP_NEEDDESTROY (1 << 1) /*!< if we need to be destroyed */
#define SIP_NOVIDEO (1 << 2) /*!< Didn't get video in invite, don't offer */
@@ -533,14 +545,13 @@
#define SIP_REALTIME (1 << 11) /*!< Flag for realtime users */
#define SIP_USECLIENTCODE (1 << 12) /*!< Trust X-ClientCode info message */
#define SIP_OUTGOING (1 << 13) /*!< Is this an outgoing call? */
-#define SIP_SELFDESTRUCT (1 << 14)
-#define SIP_DYNAMIC (1 << 15) /*!< Is this a dynamic peer? */
-/* --- Choices for DTMF support in SIP channel */
-#define SIP_DTMF (3 << 16) /*!< three settings, uses two bits */
-#define SIP_DTMF_RFC2833 (0 << 16) /*!< RTP DTMF */
-#define SIP_DTMF_INBAND (1 << 16) /*!< Inband audio, only for ULAW/ALAW */
-#define SIP_DTMF_INFO (2 << 16) /*!< SIP Info messages */
-#define SIP_DTMF_AUTO (3 << 16) /*!< AUTO switch between rfc2833 and in-band DTMF */
+#define SIP_FREEBIT (1 << 14) /*!< Free for session-related use */
+#define SIP_FREEBIT3 (1 << 15) /*!< Free for session-related use */
+#define SIP_DTMF (3 << 16) /*!< DTMF Support: four settings, uses two bits */
+#define SIP_DTMF_RFC2833 (0 << 16) /*!< DTMF Support: RTP DTMF - "rfc2833" */
+#define SIP_DTMF_INBAND (1 << 16) /*!< DTMF Support: Inband audio, only for ULAW/ALAW - "inband" */
+#define SIP_DTMF_INFO (2 << 16) /*!< DTMF Support: SIP Info messages - "info" */
+#define SIP_DTMF_AUTO (3 << 16) /*!< DTMF Support: AUTO switch between rfc2833 and in-band DTMF */
/* NAT settings */
#define SIP_NAT (3 << 18) /*!< four settings, uses two bits */
#define SIP_NAT_NEVER (0 << 18) /*!< No nat support */
@@ -578,7 +589,7 @@
SIP_PROG_INBAND | SIP_OSPAUTH | SIP_USECLIENTCODE | SIP_NAT | \
SIP_INSECURE_PORT | SIP_INSECURE_INVITE)
-/* a new page of flags for peer */
+/* a new page of flags for peers */
#define SIP_PAGE2_RTCACHEFRIENDS (1 << 0)
#define SIP_PAGE2_RTUPDATE (1 << 1)
#define SIP_PAGE2_RTAUTOCLEAR (1 << 2)
@@ -587,6 +598,8 @@
#define SIP_PAGE2_DEBUG (3 << 5)
#define SIP_PAGE2_DEBUG_CONFIG (1 << 5)
#define SIP_PAGE2_DEBUG_CONSOLE (1 << 6)
+#define SIP_PAGE2_DYNAMIC (1 << 7) /*!< Dynamic Peers register with Asterisk */
+#define SIP_PAGE2_SELFDESTRUCT (1 << 8) /*!< Automatic peers need to destruct themselves */
/* SIP packet flags */
#define SIP_PKT_DEBUG (1 << 0) /*!< Debug this packet */
@@ -956,7 +969,6 @@
static int __sip_do_register(struct sip_registry *r);
static int restart_monitor(void);
-
/*! \brief Definition of this channel for PBX channel registration */
static const struct ast_channel_tech sip_tech = {
.type = channeltype,
@@ -979,6 +991,16 @@
.send_text = sip_sendtext,
};
+/*! \brief Interface structure with callbacks used to connect to RTP module */
+static struct ast_rtp_protocol sip_rtp = {
+ type: channeltype,
+ get_rtp_info: sip_get_rtp_peer,
+ get_vrtp_info: sip_get_vrtp_peer,
+ set_rtp_peer: sip_set_rtp_peer,
+ get_codec: sip_get_codec,
+};
+
+
/*!
\brief Thread-safe random number generator
\return a random number
@@ -1292,7 +1314,9 @@
return 0;
}
-/*! \brief Transmit packet with retransmits */
+/*! \brief Transmit packet with retransmits
+ \return 0 on success, -1 on failure to allocate packet
+*/
static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod)
{
struct sip_pkt *pkt;
@@ -1659,7 +1683,7 @@
ast_sched_del(sched, peer->pokeexpire);
register_peer_exten(peer, 0);
ast_free_ha(peer->ha);
- if (ast_test_flag(peer, SIP_SELFDESTRUCT))
+ if (ast_test_flag((&peer->flags_page2), SIP_PAGE2_SELFDESTRUCT))
apeerobjs--;
else if (ast_test_flag(peer, SIP_REALTIME))
rpeerobjs--;
@@ -3146,7 +3170,8 @@
p->autokillid = -1;
p->subscribed = NONE;
p->stateid = -1;
- p->prefs = prefs;
+ p->prefs = default_prefs; /* Set default codecs for this call */
+
if (intended_method != SIP_OPTIONS) /* Peerpoke has it's own system */
p->timer_t1 = 500; /* Default SIP retransmission timer T1 (RFC 3261) */
#ifdef OSP_SUPPORT
@@ -5736,7 +5761,7 @@
register_peer_exten(peer, 0);
peer->expire = -1;
ast_device_state_changed("SIP/%s", peer->name);
- if (ast_test_flag(peer, SIP_SELFDESTRUCT) || ast_test_flag((&peer->flags_page2), SIP_PAGE2_RTAUTOCLEAR)) {
+ if (ast_test_flag((&peer->flags_page2), SIP_PAGE2_SELFDESTRUCT) || ast_test_flag((&peer->flags_page2), SIP_PAGE2_RTAUTOCLEAR)) {
peer = ASTOBJ_CONTAINER_UNLINK(&peerl, peer);
ASTOBJ_UNREF(peer, sip_destroy_peer);
}
@@ -6468,7 +6493,7 @@
ASTOBJ_UNREF(peer, sip_destroy_peer);
}
if (peer) {
- if (!ast_test_flag(peer, SIP_DYNAMIC)) {
+ if (!ast_test_flag((&peer->flags_page2), SIP_PAGE2_DYNAMIC)) {
ast_log(LOG_ERROR, "Peer '%s' is trying to register, but not configured as host=dynamic\n", peer->name);
} else {
ast_copy_flags(p, peer, SIP_NAT);
@@ -7628,7 +7653,7 @@
snprintf(srch, sizeof(srch), FORMAT, name,
iterator->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), iterator->addr.sin_addr) : "(Unspecified)",
- ast_test_flag(iterator, SIP_DYNAMIC) ? " D " : " ", /* Dynamic or not? */
+ ast_test_flag((&iterator->flags_page2), SIP_PAGE2_DYNAMIC) ? " D " : " ", /* Dynamic or not? */
(ast_test_flag(iterator, SIP_NAT) & SIP_NAT_ROUTE) ? " N " : " ", /* NAT=yes? */
iterator->ha ? " A " : " ", /* permit/deny */
ntohs(iterator->addr.sin_port), status);
@@ -7636,7 +7661,7 @@
if (!s) {/* Normal CLI list */
ast_cli(fd, FORMAT, name,
iterator->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), iterator->addr.sin_addr) : "(Unspecified)",
- ast_test_flag(iterator, SIP_DYNAMIC) ? " D " : " ", /* Dynamic or not? */
+ ast_test_flag((&iterator->flags_page2), SIP_PAGE2_DYNAMIC) ? " D " : " ", /* Dynamic or not? */
(ast_test_flag(iterator, SIP_NAT) & SIP_NAT_ROUTE) ? " N " : " ", /* NAT=yes? */
iterator->ha ? " A " : " ", /* permit/deny */
@@ -7658,7 +7683,7 @@
iterator->name,
iterator->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), iterator->addr.sin_addr) : "-none-",
ntohs(iterator->addr.sin_port),
- ast_test_flag(iterator, SIP_DYNAMIC) ? "yes" : "no", /* Dynamic or not? */
+ ast_test_flag((&iterator->flags_page2), SIP_PAGE2_DYNAMIC) ? "yes" : "no", /* Dynamic or not? */
(ast_test_flag(iterator, SIP_NAT) & SIP_NAT_ROUTE) ? "yes" : "no", /* NAT=yes? */
iterator->ha ? "yes" : "no", /* permit/deny */
status);
@@ -8031,7 +8056,7 @@
ast_cli(fd, " VM Extension : %s\n", peer->vmexten);
ast_cli(fd, " LastMsgsSent : %d\n", peer->lastmsgssent);
ast_cli(fd, " Call limit : %d\n", peer->call_limit);
- ast_cli(fd, " Dynamic : %s\n", (ast_test_flag(peer, SIP_DYNAMIC)?"Yes":"No"));
+ ast_cli(fd, " Dynamic : %s\n", (ast_test_flag((&peer->flags_page2), SIP_PAGE2_DYNAMIC)?"Yes":"No"));
ast_cli(fd, " Callerid : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, "<unspecified>"));
ast_cli(fd, " Expire : %d\n", peer->expire);
ast_cli(fd, " Insecure : %s\n", insecure2str(ast_test_flag(peer, SIP_INSECURE_PORT), ast_test_flag(peer, SIP_INSECURE_INVITE)));
@@ -8107,7 +8132,7 @@
ast_cli(fd, "VoiceMailbox: %s\r\n", peer->mailbox);
ast_cli(fd, "LastMsgsSent: %d\r\n", peer->lastmsgssent);
ast_cli(fd, "Call limit: %d\r\n", peer->call_limit);
- ast_cli(fd, "Dynamic: %s\r\n", (ast_test_flag(peer, SIP_DYNAMIC)?"Y":"N"));
+ ast_cli(fd, "Dynamic: %s\r\n", (ast_test_flag((&peer->flags_page2), SIP_PAGE2_DYNAMIC)?"Y":"N"));
ast_cli(fd, "Callerid: %s\r\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, ""));
ast_cli(fd, "RegExpire: %ld seconds\r\n", ast_sched_when(sched,peer->expire));
ast_cli(fd, "SIP-AuthInsecure: %s\r\n", insecure2str(ast_test_flag(peer, SIP_INSECURE_PORT), ast_test_flag(peer, SIP_INSECURE_INVITE)));
@@ -8292,7 +8317,7 @@
ast_cli(fd, "\nGlobal Signalling Settings:\n");
ast_cli(fd, "---------------------------\n");
ast_cli(fd, " Codecs: ");
- print_codec_to_cli(fd, &prefs);
+ print_codec_to_cli(fd, &default_prefs);
ast_cli(fd, "\n");
ast_cli(fd, " T1 minimum: %d\n", global_t1min);
ast_cli(fd, " Relax DTMF: %s\n", global_relaxdtmf ? "Yes" : "No");
@@ -9346,7 +9371,7 @@
} else if (!strcasecmp(colname, "expire")) {
snprintf(buf, len, "%d", peer->expire);
} else if (!strcasecmp(colname, "dynamic")) {
- ast_copy_string(buf, (ast_test_flag(peer, SIP_DYNAMIC) ? "yes" : "no"), len);
+ ast_copy_string(buf, (ast_test_flag((&peer->flags_page2), SIP_PAGE2_DYNAMIC) ? "yes" : "no"), len);
} else if (!strcasecmp(colname, "callerid_name")) {
ast_copy_string(buf, peer->cid_name, len);
} else if (!strcasecmp(colname, "callerid_num")) {
@@ -10114,13 +10139,8 @@
}
}
-struct sip_dual {
- struct ast_channel *chan1;
- struct ast_channel *chan2;
- struct sip_request req;
-};
-
-/*! \brief sip_park_thread: Park SIP call support function */
+
+/*! \brief Park SIP call support function */
static void *sip_park_thread(void *stuff)
{
struct ast_channel *chan1, *chan2;
@@ -10143,7 +10163,7 @@
return NULL;
}
-/*! \brief sip_park: Park a call */
+/*! \brief Park a call */
static int sip_park(struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req)
{
struct sip_dual *d;
@@ -10199,7 +10219,7 @@
return -1;
}
-/*! \brief ast_quiet_chan: Turn off generator data */
+/*! \brief Turn off generator data */
static void ast_quiet_chan(struct ast_channel *chan)
{
if (chan && chan->_state == AST_STATE_UP) {
@@ -10208,7 +10228,7 @@
}
}
-/*! \brief attempt_transfer: Attempt transfer of SIP call */
+/*! \brief Attempt transfer of SIP call */
static int attempt_transfer(struct sip_pvt *p1, struct sip_pvt *p2)
{
int res = 0;
@@ -10299,7 +10319,7 @@
return thetag;
}
-/*! \brief handle_request_options: Handle incoming OPTIONS request */
+/*! \brief Handle incoming OPTIONS request */
static int handle_request_options(struct sip_pvt *p, struct sip_request *req, int debug)
{
int res;
@@ -10323,7 +10343,7 @@
return res;
}
-/*! \brief handle_request_invite: Handle incoming INVITE request */
+/*! \brief Handle incoming INVITE request */
static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, struct sockaddr_in *sin, int *recount, char *e)
{
int res = 1;
@@ -10580,7 +10600,7 @@
return res;
}
-/*! \brief handle_request_refer: Handle incoming REFER request */
+/*! \brief Handle incoming REFER request */
static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, int seqno, int *nounlock)
{
struct ast_channel *c=NULL;
@@ -10647,7 +10667,7 @@
}
return res;
}
-/*! \brief handle_request_cancel: Handle incoming CANCEL request */
+/*! \brief Handle incoming CANCEL request */
static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req, int debug, int ignore)
{
@@ -10676,7 +10696,7 @@
}
}
-/*! \brief handle_request_bye: Handle incoming BYE request */
+/*! \brief Handle incoming BYE request */
static int handle_request_bye(struct sip_pvt *p, struct sip_request *req, int debug, int ignore)
{
struct ast_channel *c=NULL;
@@ -10729,7 +10749,7 @@
return 1;
}
-/*! \brief handle_request_message: Handle incoming MESSAGE request */
+/*! \brief Handle incoming MESSAGE request */
static int handle_request_message(struct sip_pvt *p, struct sip_request *req, int debug, int ignore)
{
if (!ignore) {
@@ -10936,7 +10956,7 @@
return 1;
}
-/*! \brief handle_request_register: Handle incoming REGISTER request */
+/*! \brief Handle incoming REGISTER request */
static int handle_request_register(struct sip_pvt *p, struct sip_request *req, int debug, int ignore, struct sockaddr_in *sin, char *e)
{
int res = 0;
@@ -10957,8 +10977,8 @@
return res;
}
-/*! \brief handle_request: Handle SIP requests (methods) */
-/* this is where all incoming requests go first */
+/*! \brief Handle incoming SIP requests (methods)
+\note This is where all incoming requests go first */
static int handle_request(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int *recount, int *nounlock)
{
/* Called with p->lock held, as well as p->owner->lock if appropriate, keeping things
@@ -11160,8 +11180,8 @@
return res;
}
-/*! \brief sipsock_read: Read data from SIP socket */
-/* Successful messages is connected to SIP call and forwarded to handle_request() */
+/*! \brief Read data from SIP socket
+\note Successful messages is connected to SIP call and forwarded to handle_request() */
static int sipsock_read(int *id, int fd, short events, void *ignore)
{
struct sip_request req;
@@ -11247,7 +11267,7 @@
return 1;
}
-/*! \brief sip_send_mwi_to_peer: Send message waiting indication */
+/*! \brief Send message waiting indication to alert peer that they've got voicemail */
static int sip_send_mwi_to_peer(struct sip_peer *peer)
{
/* Called with peerl lock, but releases it */
@@ -11285,7 +11305,7 @@
return 0;
}
-/*! \brief do_monitor: The SIP monitoring thread */
+/*! \brief The SIP monitoring thread */
static void *do_monitor(void *data)
{
int res;
@@ -11416,7 +11436,7 @@
}
-/*! \brief restart_monitor: Start the channel monitor thread */
+/*! \brief Start the channel monitor thread */
static int restart_monitor(void)
{
pthread_attr_t attr;
@@ -11449,7 +11469,7 @@
return 0;
}
-/*! \brief sip_poke_noanswer: No answer to Qualify poke */
+/*! \brief React to lack of answer to Qualify poke */
static int sip_poke_noanswer(void *data)
{
struct sip_peer *peer = data;
@@ -11469,9 +11489,9 @@
return 0;
}
-/*! \brief sip_poke_peer: Check availability of peer, also keep NAT open */
-/* This is done with the interval in qualify= option in sip.conf */
-/* Default is 2 seconds */
+/*! \brief Check availability of peer, also keep NAT open
+\note This is done with the interval in qualify= configuration option
+ Default is 2 seconds */
static int sip_poke_peer(struct sip_peer *peer)
{
struct sip_pvt *p;
@@ -11531,20 +11551,21 @@
return 0;
}
-/*! \brief sip_devicestate: Part of PBX channel interface */
-
-/* Return values:---
+/*! \brief Part of PBX channel interface
+\note
+\par Return values:---
+
If we have qualify on and the device is not reachable, regardless of registration
state we return AST_DEVICE_UNAVAILABLE
For peers with call limit:
- not registered AST_DEVICE_UNAVAILABLE
- registered, no call AST_DEVICE_NOT_INUSE
- registered, calls possible AST_DEVICE_INUSE
- registered, call limit reached AST_DEVICE_BUSY
+ - not registered AST_DEVICE_UNAVAILABLE
+ - registered, no call AST_DEVICE_NOT_INUSE
+ - registered, calls possible AST_DEVICE_INUSE
+ - registered, call limit reached AST_DEVICE_BUSY
For peers without call limit:
- not registered AST_DEVICE_UNAVAILABLE
- registered AST_DEVICE_UNKNOWN
+ - not registered AST_DEVICE_UNAVAILABLE
+ - registered AST_DEVICE_UNKNOWN
*/
static int sip_devicestate(void *data)
{
@@ -11596,7 +11617,7 @@
return res;
}
-/*! \brief sip_request: PBX interface function -build SIP pvt structure */
+/*! \brief PBX interface function -build SIP pvt structure */
/* SIP calls initiated by the PBX arrive here */
static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause)
{
@@ -11672,7 +11693,7 @@
return tmpc;
}
-/*! \brief handle_common_options: Handle flag-type options common to users and peers */
+/*! \brief Handle flag-type options common to users and peers */
static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v)
{
int res = 0;
@@ -11781,7 +11802,7 @@
return res;
}
-/*! \brief add_sip_domain: Add SIP domain to list of domains we are responsible for */
+/*! \brief Add SIP domain to list of domains we are responsible for */
static int add_sip_domain(const char *domain, const enum domain_mode mode, const char *context)
{
struct domain *d;
@@ -11833,7 +11854,7 @@
return result;
}
-/*! \brief clear_sip_domains: Clear our domain list (at reload) */
+/*! \brief Clear our domain list (at reload) */
static void clear_sip_domains(void)
{
struct domain *d;
@@ -11845,7 +11866,7 @@
}
-/*! \brief add_realm_authentication: Add realm authentication in list */
+/*! \brief Add realm authentication in list */
static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno)
{
char authcopy[256];
@@ -11908,7 +11929,7 @@
}
-/*! \brief clear_realm_authentication: Clear realm authentication list (at reload) */
+/*! \brief Clear realm authentication list (at reload) */
static int clear_realm_authentication(struct sip_auth *authlist)
{
struct sip_auth *a = authlist;
@@ -11923,7 +11944,7 @@
return 1;
}
-/*! \brief find_realm_authentication: Find authentication for a specific realm */
+/*! \brief Find authentication for a specific realm */
static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm)
{
struct sip_auth *a;
@@ -11936,7 +11957,7 @@
return a;
}
-/*! \brief build_user: Initiate a SIP user structure from sip.conf */
+/*! \brief Initiate a SIP user structure from configuration (configuration or realtime) */
static struct sip_user *build_user(const char *name, struct ast_variable *v, int realtime)
{
struct sip_user *user;
@@ -11958,7 +11979,7 @@
user->ha = NULL;
ast_copy_flags(user, &global_flags, SIP_FLAGS_TO_COPY);
user->capability = global_capability;
- user->prefs = prefs;
+ user->prefs = default_prefs;
/* set default context */
strcpy(user->context, default_context);
strcpy(user->language, default_language);
@@ -12026,7 +12047,7 @@
return user;
}
-/*! \brief temp_peer: Create temporary peer (used in autocreatepeer mode) */
+/*! \brief Create temporary peer (used in autocreatepeer mode) */
static struct sip_peer *temp_peer(const char *name)
{
struct sip_peer *peer;
@@ -12051,15 +12072,15 @@
peer->rtptimeout = global_rtptimeout;
peer->rtpholdtimeout = global_rtpholdtimeout;
peer->rtpkeepalive = global_rtpkeepalive;
- ast_set_flag(peer, SIP_SELFDESTRUCT);
- ast_set_flag(peer, SIP_DYNAMIC);
- peer->prefs = prefs;
+ ast_set_flag((&peer->flags_page2), SIP_PAGE2_SELFDESTRUCT);
+ ast_set_flag((&peer->flags_page2), SIP_PAGE2_DYNAMIC);
+ peer->prefs = default_prefs;
reg_source_db(peer);
return peer;
}
-/*! \brief build_peer: Build peer from config file */
+/*! \brief Build peer from configuration (file or realtime static/dynamic) */
static struct sip_peer *build_peer(const char *name, struct ast_variable *v, int realtime)
{
struct sip_peer *peer = NULL;
@@ -12130,7 +12151,7 @@
peer->pickupgroup = 0;
peer->rtpkeepalive = global_rtpkeepalive;
peer->maxms = default_qualify;
- peer->prefs = prefs;
+ peer->prefs = default_prefs;
oldha = peer->ha;
peer->ha = NULL;
peer->addr.sin_family = AF_INET;
@@ -12178,7 +12199,7 @@
ast_log(LOG_WARNING, "You can't have a dynamic outbound proxy, you big silly head at line %d.\n", v->lineno);
} else {
/* They'll register with us */
- ast_set_flag(peer, SIP_DYNAMIC);
+ ast_set_flag((&peer->flags_page2), SIP_PAGE2_DYNAMIC);
if (!found) {
/* Initialize stuff iff we're not found, otherwise
we keep going with what we had */
@@ -12195,7 +12216,7 @@
if (peer->expire > -1)
ast_sched_del(sched, peer->expire);
peer->expire = -1;
- ast_clear_flag(peer, SIP_DYNAMIC);
+ ast_clear_flag((&peer->flags_page2), SIP_PAGE2_DYNAMIC);
if (!obproxyfound || !strcasecmp(v->name, "outboundproxy")) {
if (ast_get_ip_or_srv(&peer->addr, v->value, "_sip._udp")) {
ASTOBJ_UNREF(peer, sip_destroy_peer);
@@ -12218,7 +12239,7 @@
} else if (!strcasecmp(v->name, "permit") || !strcasecmp(v->name, "deny")) {
peer->ha = ast_append_ha(v->name, v->value, peer->ha);
} else if (!strcasecmp(v->name, "port")) {
- if (!realtime && ast_test_flag(peer, SIP_DYNAMIC))
+ if (!realtime && ast_test_flag((&peer->flags_page2), SIP_PAGE2_DYNAMIC))
peer->defaddr.sin_port = htons(atoi(v->value));
else
peer->addr.sin_port = htons(atoi(v->value));
@@ -12300,7 +12321,7 @@
*/
v=v->next;
}
- if (!ast_test_flag((&global_flags_page2), SIP_PAGE2_IGNOREREGEXPIRE) && ast_test_flag(peer, SIP_DYNAMIC) && realtime) {
+ if (!ast_test_flag((&global_flags_page2), SIP_PAGE2_IGNOREREGEXPIRE) && ast_test_flag((&peer->flags_page2), SIP_PAGE2_DYNAMIC) && realtime) {
time_t nowtime;
time(&nowtime);
@@ -12312,15 +12333,15 @@
}
}
ast_copy_flags(peer, &peerflags, mask.flags);
- if (!found && ast_test_flag(peer, SIP_DYNAMIC) && !ast_test_flag(peer, SIP_REALTIME))
+ if (!found && ast_test_flag((&peer->flags_page2), SIP_PAGE2_DYNAMIC) && !ast_test_flag(peer, SIP_REALTIME))
reg_source_db(peer);
ASTOBJ_UNMARK(peer);
ast_free_ha(oldha);
return peer;
}
-/*! \brief reload_config: Re-read SIP.conf config file */
-/* This function reloads all config data, except for
+/*! \brief Re-read SIP.conf config file
+\note This function reloads all config data, except for
active peers (with registrations). They will only
change configuration data at restart, not at reload.
SIP debug and recordhistory state will not change
@@ -12356,7 +12377,7 @@
memset(&bindaddr, 0, sizeof(bindaddr));
memset(&localaddr, 0, sizeof(localaddr));
memset(&externip, 0, sizeof(externip));
- memset(&prefs, 0 , sizeof(prefs));
+ memset(&default_prefs, 0 , sizeof(default_prefs));
outboundproxyip.sin_port = htons(DEFAULT_SIP_PORT);
outboundproxyip.sin_family = AF_INET; /* Type of address: IPv4 */
ourport = DEFAULT_SIP_PORT;
@@ -12557,9 +12578,9 @@
externrefresh = 10;
}
} else if (!strcasecmp(v->name, "allow")) {
- ast_parse_allow_disallow(&prefs, &global_capability, v->value, 1);
+ ast_parse_allow_disallow(&default_prefs, &global_capability, v->value, 1);
} else if (!strcasecmp(v->name, "disallow")) {
- ast_parse_allow_disallow(&prefs, &global_capability, v->value, 0);
+ ast_parse_allow_disallow(&default_prefs, &global_capability, v->value, 0);
} else if (!strcasecmp(v->name, "allowexternaldomains")) {
allow_external_domains = ast_true(v->value);
} else if (!strcasecmp(v->name, "autodomain")) {
@@ -12744,7 +12765,7 @@
return 0;
}
-/*! \brief sip_get_rtp_peer: Returns null if we can't reinvite (part of RTP interface) */
+/*! \brief Returns null if we can't reinvite (part of RTP interface) */
static struct ast_rtp *sip_get_rtp_peer(struct ast_channel *chan)
{
struct sip_pvt *p;
@@ -12759,7 +12780,7 @@
return rtp;
}
-/*! \brief sip_get_vrtp_peer: Returns null if we can't reinvite video (part of RTP interface) */
+/*! \brief Returns null if we can't reinvite video (part of RTP interface) */
static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan)
{
struct sip_pvt *p;
@@ -12775,7 +12796,7 @@
return rtp;
}
-/*! \brief sip_set_rtp_peer: Set the RTP peer for this call */
+/*! \brief Set the RTP peer for this call */
static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active)
{
struct sip_pvt *p;
@@ -12836,7 +12857,7 @@
"Always returns 0\n";
-/*! \brief sip_dtmfmode: change the DTMFmode for a SIP call (application) */
+/*! \brief Set the DTMFmode for an outbound SIP call (application) */
static int sip_dtmfmode(struct ast_channel *chan, void *data)
{
struct sip_pvt *p;
@@ -12886,7 +12907,7 @@
return 0;
}
-/*! \brief sip_addheader: Add a SIP header */
+/*! \brief Add a SIP header to an outbound INVITE */
static int sip_addheader(struct ast_channel *chan, void *data)
{
int no = 0;
@@ -12919,11 +12940,11 @@
return 0;
}
-/*! \brief sip_sipredirect: Transfer call before connect with a 302 redirect
- * Called by the transfer() dialplan application through the sip_transfer()
- * pbx interface function if the call is in ringing state
- * \todo Fix this function so that we wait for reply to the REFER and
- * react to errors, denials or other issues the other end might have.
+/*! \brief Transfer call before connect with a 302 redirect
+\note Called by the transfer() dialplan application through the sip_transfer()
+ pbx interface function if the call is in ringing state
+\todo Fix this function so that we wait for reply to the REFER and
+ react to errors, denials or other issues the other end might have.
*/
static int sip_sipredirect(struct sip_pvt *p, const char *dest)
{
@@ -12980,23 +13001,14 @@
return -1;
}
-/*! \brief sip_get_codec: Return SIP UA's codec (part of the RTP interface) */
+/*! \brief Return SIP UA's codec (part of the RTP interface) */
static int sip_get_codec(struct ast_channel *chan)
{
struct sip_pvt *p = chan->tech_pvt;
return p->peercapability;
}
-/*! \brief sip_rtp: Interface structure with callbacks used to connect to rtp module */
-static struct ast_rtp_protocol sip_rtp = {
- type: channeltype,
- get_rtp_info: sip_get_rtp_peer,
- get_vrtp_info: sip_get_vrtp_peer,
- set_rtp_peer: sip_set_rtp_peer,
- get_codec: sip_get_codec,
-};
-
-/*! \brief sip_poke_all_peers: Send a poke to all known peers */
+/*! \brief Send a poke to all known peers */
static void sip_poke_all_peers(void)
{
ASTOBJ_CONTAINER_TRAVERSE(&peerl, 1, do {
@@ -13007,7 +13019,7 @@
);
}
-/*! \brief sip_send_all_registers: Send all known registrations */
+/*! \brief Send all known registrations */
static void sip_send_all_registers(void)
{
int ms;
@@ -13138,7 +13150,7 @@
AST_DEFINE_CODE_TEST(test_tags, "tags", test_tags_func);
-/*! \brief sip_do_reload: Reload module */
+/*! \brief Reload module */
static int sip_do_reload(enum channelreloadreason reason)
{
if (option_debug > 3)
@@ -13188,7 +13200,7 @@
return 0;
}
-/*! \brief sip_reload: Force reload of module from cli */
+/*! \brief Force reload of module from cli */
static int sip_reload(int fd, int argc, char *argv[])
{
Modified: team/russell/codetest/channels/chan_zap.c
URL: http://svn.digium.com/view/asterisk/team/russell/codetest/channels/chan_zap.c?rev=8996&r1=8995&r2=8996&view=diff
==============================================================================
--- team/russell/codetest/channels/chan_zap.c (original)
+++ team/russell/codetest/channels/chan_zap.c Tue Jan 31 10:56:07 2006
@@ -711,7 +711,7 @@
static const struct ast_channel_tech zap_tech = {
.type = type,
.description = tdesc,
- .capabilities = AST_FORMAT_SLINEAR | AST_FORMAT_ULAW,
+ .capabilities = AST_FORMAT_SLINEAR | AST_FORMAT_ULAW | AST_FORMAT_ALAW,
.requester = zt_request,
.send_digit = zt_digit,
.send_text = zt_sendtext,
Modified: team/russell/codetest/include/asterisk/channel.h
URL: http://svn.digium.com/view/asterisk/team/russell/codetest/include/asterisk/channel.h?rev=8996&r1=8995&r2=8996&view=diff
==============================================================================
--- team/russell/codetest/include/asterisk/channel.h (original)
+++ team/russell/codetest/include/asterisk/channel.h Tue Jan 31 10:56:07 2006
@@ -261,6 +261,9 @@
/*! Main Channel structure associated with a channel.
* This is the side of it mostly used by the pbx and call management.
+ *
+ * \note XXX It is important to remember to increment .cleancount each time
+ * this structure is changed. XXX
*/
struct ast_channel {
/*! ASCII unique channel name */
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