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Formatting and doxygen while waiting on an airport... 

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104137 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson 2008-02-26 16:51:25 +00:00
parent d527ff6bb6
commit 82cef0fa88
1 changed files with 39 additions and 37 deletions
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76
channels/chan_sip.c
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@ -11488,7 +11488,7 @@ static void receive_message(struct sip_pvt *p, struct sip_request *req)
const char *content_type = get_header(req, "Content-Type");
if (strcmp(content_type, "text/plain")) { /* No text/plain attachment */
transmit_response(p, "415 Unsupported Media Type", req); /* Good enough, or? */
transmit_response(p, "415 Unsupported Media Type", req);
if (!p->owner)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return;
@ -11515,7 +11515,7 @@ static void receive_message(struct sip_pvt *p, struct sip_request *req)
transmit_response(p, "202 Accepted", req); /* We respond 202 accepted, since we relay the message */
} else { /* Message outside of a call, we do not support that */
ast_log(LOG_WARNING, "Received message to %s from %s, dropped it...\n Content-Type:%s\n Message: %s\n", get_header(req, "To"), get_header(req, "From"), content_type, buf);
transmit_response(p, "405 Method Not Allowed", req); /* Good enough, or? */
transmit_response(p, "405 Method Not Allowed", req);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
return;
@ -11880,7 +11880,7 @@ static char *sip_show_peers(struct ast_cli_entry *e, int cmd, struct ast_cli_arg
return _sip_show_peers(a->fd, NULL, NULL, NULL, a->argc, (const char **) a->argv);
}
/*! \brief _sip_show_peers: Execute sip show peers command */
/*! \brief Execute sip show peers command */
static char *_sip_show_peers(int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[])
{
regex_t regexbuf;
@ -12382,6 +12382,7 @@ static char *sip_show_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args
return _sip_show_peer(0, a->fd, NULL, NULL, a->argc, (const char **) a->argv);
}
/*! \brief list peer mailboxes to CLI */
static void peer_mailboxes_to_str(struct ast_str **mailbox_str, struct sip_peer *peer)
{
struct sip_mailbox *mailbox;
@ -12614,7 +12615,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
astman_append(s, "%s\r\n", status);
astman_append(s, "SIP-Useragent: %s\r\n", peer->useragent);
astman_append(s, "Reg-Contact : %s\r\n", peer->fullcontact);
astman_append(s, "Qualify Freq : %d ms\n", peer->qualifyfreq);
astman_append(s, "QualifyFreq : %d ms\n", peer->qualifyfreq);
if (peer->chanvars) {
for (v = peer->chanvars ; v ; v = v->next) {
astman_append(s, "ChanVariable:\n");
@ -12818,8 +12819,8 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
ast_cli(a->fd, " Bindaddress: %s\n", ast_inet_ntoa(bindaddr.sin_addr));
ast_cli(a->fd, " Videosupport: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT)));
ast_cli(a->fd, " Textsupport: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT)));
ast_cli(a->fd, " AutoCreatePeer: %s\n", cli_yesno(autocreatepeer));
ast_cli(a->fd, " MatchAuthUsername: %s\n", cli_yesno(global_match_auth_username));
ast_cli(a->fd, " AutoCreate Peer: %s\n", cli_yesno(autocreatepeer));
ast_cli(a->fd, " Match Auth Username: %s\n", cli_yesno(global_match_auth_username));
ast_cli(a->fd, " Allow unknown access: %s\n", cli_yesno(global_allowguest));
ast_cli(a->fd, " Allow subscriptions: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
ast_cli(a->fd, " Enable call counters: %s\n", cli_yesno(global_callcounter));
@ -12842,6 +12843,19 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
ast_cli(a->fd, " From: Domain: %s\n", default_fromdomain);
ast_cli(a->fd, " Record SIP history: %s\n", recordhistory ? "On" : "Off");
ast_cli(a->fd, " Call Events: %s\n", global_callevents ? "On" : "Off");
ast_cli(a->fd, " T38 fax pt UDPTL: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_UDPTL)));
#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS
ast_cli(a->fd, " T38 fax pt RTP: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_RTP)));
ast_cli(a->fd, " T38 fax pt TCP: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_TCP)));
#endif
if (!realtimepeers && !realtimeusers && !realtimeregs)
ast_cli(a->fd, " SIP realtime: Disabled\n" );
else
ast_cli(a->fd, " SIP realtime: Enabled\n" );
ast_cli(a->fd, " Qualify Freq : %d ms\n", global_qualifyfreq);
ast_cli(a->fd, "\nNetwork QoS Settings:\n");
ast_cli(a->fd, "---------------------------\n");
ast_cli(a->fd, " IP ToS SIP: %s\n", ast_tos2str(global_tos_sip));
ast_cli(a->fd, " IP ToS RTP audio: %s\n", ast_tos2str(global_tos_audio));
ast_cli(a->fd, " IP ToS RTP video: %s\n", ast_tos2str(global_tos_video));
@ -12850,24 +12864,12 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
ast_cli(a->fd, " 802.1p CoS RTP audio: %d\n", global_cos_audio);
ast_cli(a->fd, " 802.1p CoS RTP video: %d\n", global_cos_video);
ast_cli(a->fd, " 802.1p CoS RTP text: %d\n", global_cos_text);
ast_cli(a->fd, " T38 fax pt UDPTL: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_UDPTL)));
#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS
ast_cli(a->fd, " T38 fax pt RTP: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_RTP)));
ast_cli(a->fd, " T38 fax pt TCP: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_TCP)));
#endif
ast_cli(a->fd, " RFC2833 Compensation: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_RFC2833_COMPENSATE)));
ast_cli(a->fd, " Jitterbuffer enabled: %s\n", cli_yesno(ast_test_flag(&global_jbconf, AST_JB_ENABLED)));
ast_cli(a->fd, " Jitterbuffer forced: %s\n", cli_yesno(ast_test_flag(&global_jbconf, AST_JB_FORCED)));
ast_cli(a->fd, " Jitterbuffer max size: %ld\n", global_jbconf.max_size);
ast_cli(a->fd, " Jitterbuffer resync: %ld\n", global_jbconf.resync_threshold);
ast_cli(a->fd, " Jitterbuffer impl: %s\n", global_jbconf.impl);
ast_cli(a->fd, " Jitterbuffer log: %s\n", cli_yesno(ast_test_flag(&global_jbconf, AST_JB_LOG)));
if (!realtimepeers && !realtimeusers && !realtimeregs)
ast_cli(a->fd, " SIP realtime: Disabled\n" );
else
ast_cli(a->fd, " SIP realtime: Enabled\n" );
ast_cli(a->fd, " Qualify Freq : %d ms\n", global_qualifyfreq);
ast_cli(a->fd, "\nNetwork Settings:\n");
ast_cli(a->fd, "---------------------------\n");
@ -12909,6 +12911,7 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
print_codec_to_cli(a->fd, &default_prefs);
ast_cli(a->fd, "\n");
ast_cli(a->fd, " Relax DTMF: %s\n", cli_yesno(global_relaxdtmf));
ast_cli(a->fd, " RFC2833 Compensation: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_RFC2833_COMPENSATE)));
ast_cli(a->fd, " Compact SIP headers: %s\n", cli_yesno(compactheaders));
ast_cli(a->fd, " RTP Keepalive: %d %s\n", global_rtpkeepalive, global_rtpkeepalive ? "" : "(Disabled)" );
ast_cli(a->fd, " RTP Timeout: %d %s\n", global_rtptimeout, global_rtptimeout ? "" : "(Disabled)" );
@ -13424,8 +13427,7 @@ static void sip_dump_history(struct sip_pvt *dialog)
}
/*! \brief Receive SIP INFO Message
\note Doesn't read the duration of the DTMF signal */
/*! \brief Receive SIP INFO Message */
static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
{
char buf[1024];
@ -13499,6 +13501,7 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
transmit_response(p, "200 OK", req);
return;
} else if (!strcasecmp(c, "application/dtmf")) {
/*! \todo Note: Doesn't read the duration of the DTMF. Should be fixed. */
unsigned int duration = 0;
if (!p->owner) { /* not a PBX call */
@ -13507,8 +13510,6 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
return;
}
get_msg_text(buf, sizeof(buf), req);
duration = 100; /* 100 ms */
@ -13638,7 +13639,7 @@ static char *sip_do_debug_ip(int fd, char *arg)
return CLI_SUCCESS;
}
/*! \brief sip_do_debug_peer: Turn on SIP debugging for a given peer */
/*! \brief Turn on SIP debugging for a given peer */
static char *sip_do_debug_peer(int fd, char *arg)
{
struct sip_peer *peer = find_peer(arg, NULL, 1);
@ -19833,16 +19834,16 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
ast_log(LOG_WARNING, "'%s' is not a valid RTP keepalive time at line %d. Using default.\n", v->value, v->lineno);
peer->rtpkeepalive = global_rtpkeepalive;
}
} else if (!strcasecmp(v->name, "timert1")) {
if ((sscanf(v->value, "%d", &peer->timer_t1) != 1) || (peer->timer_t1 < 0)) {
ast_log(LOG_WARNING, "'%s' is not a valid T1 time at line %d. Using default.\n", v->value, v->lineno);
peer->timer_t1 = global_t1;
}
} else if (!strcasecmp(v->name, "timerb")) {
if ((sscanf(v->value, "%d", &peer->timer_b) != 1) || (peer->timer_b < 0)) {
ast_log(LOG_WARNING, "'%s' is not a valid Timer B time at line %d. Using default.\n", v->value, v->lineno);
peer->timer_b = global_timer_b;
}
} else if (!strcasecmp(v->name, "timert1")) {
if ((sscanf(v->value, "%d", &peer->timer_t1) != 1) || (peer->timer_t1 < 0)) {
ast_log(LOG_WARNING, "'%s' is not a valid T1 time at line %d. Using default.\n", v->value, v->lineno);
peer->timer_t1 = global_t1;
}
} else if (!strcasecmp(v->name, "timerb")) {
if ((sscanf(v->value, "%d", &peer->timer_b) != 1) || (peer->timer_b < 0)) {
ast_log(LOG_WARNING, "'%s' is not a valid Timer B time at line %d. Using default.\n", v->value, v->lineno);
peer->timer_b = global_timer_b;
}
} else if (!strcasecmp(v->name, "setvar")) {
peer->chanvars = add_var(v->value, peer->chanvars);
} else if (!strcasecmp(v->name, "qualify")) {
@ -20657,10 +20658,11 @@ static int reload_config(enum channelreloadreason reason)
}
ast_mutex_unlock(&netlock);
/* Add default domains - host name, IP address and IP:port */
/* Only do this if user added any sip domain with "localdomains" */
/* In order to *not* break backwards compatibility */
/* Some phones address us at IP only, some with additional port number */
/* Add default domains - host name, IP address and IP:port
* Only do this if user added any sip domain with "localdomains"
* In order to *not* break backwards compatibility
* Some phones address us at IP only, some with additional port number
*/
if (auto_sip_domains) {
char temp[MAXHOSTNAMELEN];