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more merge from trunk (comments and change a static function name)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44805 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Luigi Rizzo
2006-10-10 16:21:22 +00:00
parent 74171605af
commit 0681269434
1 changed files with 6 additions and 6 deletions
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12
channels/chan_sip.c
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@@ -11759,7 +11759,7 @@ static int handle_response_peerpoke(struct sip_pvt *p, int resp, struct sip_requ
}
/*! \brief Immediately stop RTP, VRTP and UDPTL as applicable */
static void stop_data_flows(struct sip_pvt *p)
static void stop_media_flows(struct sip_pvt *p)
{
/* Immediately stop RTP, VRTP and UDPTL as applicable */
if (p->rtp)
@@ -11960,7 +11960,7 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
ast_verbose(VERBOSE_PREFIX_3 "Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_inet_ntoa(p->sa.sin_addr));
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
stop_data_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
/* XXX Locking issues?? XXX */
switch(resp) {
@@ -13694,7 +13694,7 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
ast_log(LOG_DEBUG, "Got CANCEL on an answered call. Ignoring... \n");
return 0;
}
stop_data_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
if (p->owner)
ast_queue_hangup(p->owner);
@@ -13743,7 +13743,7 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
}
}
stop_data_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
if (!ast_strlen_zero(get_header(req, "Also"))) {
ast_log(LOG_NOTICE, "Client '%s' using deprecated BYE/Also transfer method. Ask vendor to support REFER instead\n",
@@ -16672,7 +16672,7 @@ static int sip_reload(int fd, int argc, char *argv[])
return 0;
}
/*! \brief reload: Part of Asterisk module interface */
/*! \brief Part of Asterisk module interface */
static int reload(void)
{
return sip_reload(0, 0, NULL);
@@ -16831,7 +16831,7 @@ static struct ast_cli_entry cli_sip[] = {
sip_reload_usage },
};
/*! \brief load_module: PBX load module - initialization */
/*! \brief PBX load module - initialization */
static int load_module(void)
{
ASTOBJ_CONTAINER_INIT(&userl); /* User object list */