From 82cef0fa883d9aca3ef6edb880dd9e3b8fdb973f Mon Sep 17 00:00:00 2001 From: Olle Johansson Date: Tue, 26 Feb 2008 16:51:25 +0000 Subject: Formatting and doxygen while waiting on an airport... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104137 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- channels/chan_sip.c | 76 +++++++++++++++++++++++++++-------------------------- 1 file changed, 39 insertions(+), 37 deletions(-) (limited to 'channels') diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 3d8ca20b4..610404767 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -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]; -- cgit v1.2.3