Formatting changes, cleaning up some code

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson 2007-11-26 20:55:09 +00:00
parent d4863bb0f0
commit 96ad455115
1 changed files with 48 additions and 36 deletions

View File

@ -1492,6 +1492,15 @@ static struct sockaddr_in debugaddr;
static struct ast_config *notify_types; /*!< The list of manual NOTIFY types we know how to send */
/*! some list management macros. */
#define UNLINK(element, head, prev) do { \
if (prev) \
(prev)->next = (element)->next; \
else \
(head) = (element)->next; \
} while (0)
/*---------------------------- Forward declarations of functions in chan_sip.c */
/*! \note This is added to help splitting up chan_sip.c into several files
in coming releases */
@ -1886,14 +1895,6 @@ static int map_s_x(const struct _map_x_s *table, const char *s, int errorvalue)
return errorvalue;
}
/**--- some list management macros. **/
#define UNLINK(element, head, prev) do { \
if (prev) \
(prev)->next = (element)->next; \
else \
(head) = (element)->next; \
} while (0)
/*! \brief Interface structure with callbacks used to connect to RTP module */
static struct ast_rtp_protocol sip_rtp = {
@ -2954,6 +2955,7 @@ static void register_peer_exten(struct sip_peer *peer, int onoff)
}
}
/*! Destroy mailbox subscriptions */
static void destroy_mailbox(struct sip_mailbox *mailbox)
{
if (mailbox->mailbox)
@ -2965,6 +2967,7 @@ static void destroy_mailbox(struct sip_mailbox *mailbox)
ast_free(mailbox);
}
/*! Destroy all peer-related mailbox subscriptions */
static void clear_peer_mailboxes(struct sip_peer *peer)
{
struct sip_mailbox *mailbox;
@ -3220,7 +3223,10 @@ static int sip_addrcmp(char *name, struct sockaddr_in *sin)
/*! \brief Locate peer by name or ip address
* This is used on incoming SIP message to find matching peer on ip
or outgoing message to find matching peer on name */
or outgoing message to find matching peer on name
\note Avoid using this function in new functions if there's a way to avoid it, i
since it causes a database lookup or a traversal of the in-memory peer list.
*/
static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime)
{
struct sip_peer *p = NULL;
@ -3331,7 +3337,10 @@ static void do_setnat(struct sip_pvt *p, int natflags)
}
/*! \brief Create address structure from peer reference.
* return -1 on error, 0 on success.
* This function copies data from peer to the dialog, so we don't have to look up the peer
* again from memory or database during the life time of the dialog.
*
* \return -1 on error, 0 on success.
*/
static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
{
@ -3371,9 +3380,9 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
ast_udptl_destroy(dialog->udptl);
dialog->udptl = NULL;
}
do_setnat(dialog, ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_ROUTE );
do_setnat(dialog, ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_ROUTE);
if (dialog->rtp) {
if (dialog->rtp) { /* Audio */
ast_rtp_setdtmf(dialog->rtp, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
ast_rtp_setdtmfcompensate(dialog->rtp, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
ast_rtp_set_rtptimeout(dialog->rtp, peer->rtptimeout);
@ -3383,14 +3392,14 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
dialog->autoframing = peer->autoframing;
}
if (dialog->vrtp) {
if (dialog->vrtp) { /* Video */
ast_rtp_setdtmf(dialog->vrtp, 0);
ast_rtp_setdtmfcompensate(dialog->vrtp, 0);
ast_rtp_set_rtptimeout(dialog->vrtp, peer->rtptimeout);
ast_rtp_set_rtpholdtimeout(dialog->vrtp, peer->rtpholdtimeout);
ast_rtp_set_rtpkeepalive(dialog->vrtp, peer->rtpkeepalive);
}
if (dialog->trtp) {
if (dialog->trtp) { /* Realtime text */
ast_rtp_setdtmf(dialog->trtp, 0);
ast_rtp_setdtmfcompensate(dialog->trtp, 0);
ast_rtp_set_rtptimeout(dialog->trtp, peer->rtptimeout);
@ -3407,43 +3416,46 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
ast_string_field_set(dialog, mohinterpret, peer->mohinterpret);
ast_string_field_set(dialog, tohost, peer->tohost);
ast_string_field_set(dialog, fullcontact, peer->fullcontact);
if (!dialog->initreq.headers && !ast_strlen_zero(peer->fromdomain)) {
char *tmpcall;
char *c;
tmpcall = ast_strdupa(dialog->callid);
c = strchr(tmpcall, '@');
if (c) {
*c = '\0';
ast_string_field_build(dialog, callid, "%s@%s", tmpcall, peer->fromdomain);
}
}
ast_string_field_set(dialog, context, peer->context);
dialog->outboundproxy = obproxy_get(dialog, peer);
if (ast_strlen_zero(dialog->tohost))
ast_string_field_set(dialog, tohost, ast_inet_ntoa(dialog->sa.sin_addr));
if (!ast_strlen_zero(peer->fromdomain))
ast_string_field_set(dialog, fromdomain, peer->fromdomain);
if (!ast_strlen_zero(peer->fromuser))
ast_string_field_set(dialog, fromuser, peer->fromuser);
if (!ast_strlen_zero(peer->language))
ast_string_field_set(dialog, language, peer->language);
dialog->callgroup = peer->callgroup;
dialog->pickupgroup = peer->pickupgroup;
dialog->allowtransfer = peer->allowtransfer;
dialog->jointnoncodeccapability = dialog->noncodeccapability;
dialog->rtptimeout = peer->rtptimeout;
dialog->maxcallbitrate = peer->maxcallbitrate;
if (ast_strlen_zero(dialog->tohost))
ast_string_field_set(dialog, tohost, ast_inet_ntoa(dialog->sa.sin_addr));
if (!ast_strlen_zero(peer->fromdomain)) {
ast_string_field_set(dialog, fromdomain, peer->fromdomain);
if (!dialog->initreq.headers) {
char *c;
char *tmpcall = ast_strdupa(dialog->callid);
c = strchr(tmpcall, '@');
if (c) {
*c = '\0';
ast_string_field_build(dialog, callid, "%s@%s", tmpcall, peer->fromdomain);
}
}
}
if (!ast_strlen_zero(peer->fromuser))
ast_string_field_set(dialog, fromuser, peer->fromuser);
if (!ast_strlen_zero(peer->language))
ast_string_field_set(dialog, language, peer->language);
/* Set timer T1 to RTT for this peer (if known by qualify=) */
/* Minimum is settable or default to 100 ms */
if (peer->maxms && peer->lastms)
dialog->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
if ((ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
(ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
dialog->noncodeccapability |= AST_RTP_DTMF;
else
dialog->noncodeccapability &= ~AST_RTP_DTMF;
dialog->jointnoncodeccapability = dialog->noncodeccapability;
ast_string_field_set(dialog, context, peer->context);
dialog->rtptimeout = peer->rtptimeout;
if (peer->call_limit)
ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
dialog->maxcallbitrate = peer->maxcallbitrate;
return 0;
}