[svn-commits] oej: trunk r104137 - /trunk/channels/chan_sip.c
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Feb 26 10:51:25 CST 2008
Author: oej
Date: Tue Feb 26 10:51:25 2008
New Revision: 104137
URL: http://svn.digium.com/view/asterisk?view=rev&rev=104137
Log:
Formatting and doxygen while waiting on an airport...
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=104137&r1=104136&r2=104137
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Feb 26 10:51:25 2008
@@ -11488,7 +11488,7 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
}
-/*! \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 @@
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 */
@@ -13506,8 +13509,6 @@
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return;
}
-
-
get_msg_text(buf, sizeof(buf), req);
duration = 100; /* 100 ms */
@@ -13638,7 +13639,7 @@
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 @@
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 @@
}
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];
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