diff options
| -rw-r--r-- | CHANGES | 8 | ||||
| -rw-r--r-- | channels/chan_sip.c | 16 | ||||
| -rw-r--r-- | configs/sip.conf.sample | 36 |
3 files changed, 44 insertions, 16 deletions
@@ -89,8 +89,14 @@ SIP changes * SIP now adds a header to the CANCEL if the call was answered by another phone in the same dial command, or if the new c option in dial() is used. * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically - states it is not needed. For phones, however, that do require it the registertrying option + states it is not needed. For phones, however, that do require it the "registertrying" option has been added so it can be enabled. + * The "call-limit" option is marked as deprecated. It still works in this version of + Asterisk, but will be removed in the following version. Please use the groupcount functions + in the dialplan to enforce call limits. + * A new option called "callcounter" (global/peer/user level) enables call counters needed + for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously + used to enable this functionality). IAX2 changes ------------ diff --git a/channels/chan_sip.c b/channels/chan_sip.c index ed691c650..07601f321 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -515,6 +515,7 @@ static const struct cfsip_options { #define DEFAULT_CALLERID "asterisk" #define DEFAULT_NOTIFYMIME "application/simple-message-summary" #define DEFAULT_ALLOWGUEST TRUE +#define DEFAULT_CALLCOUNTER FALSE #define DEFAULT_SRVLOOKUP TRUE /*!< Recommended setting is ON */ #define DEFAULT_COMPACTHEADERS FALSE #define DEFAULT_TOS_SIP 0 /*!< Call signalling packets should be marked as DSCP CS3, but the default is 0 to be compatible with previous versions. */ @@ -584,6 +585,9 @@ static int global_rtpkeepalive; /*!< Send RTP keepalives */ static int global_reg_timeout; static int global_regattempts_max; /*!< Registration attempts before giving up */ static int global_allowguest; /*!< allow unauthenticated users/peers to connect? */ +static int global_callcounter; /*!< Enable call counters for all devices. This is currently enabled by setting the peer + call-limit to 999. When we remove the call-limit from the code, we can make it + with just a boolean flag in the device structure */ static int global_allowsubscribe; /*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE the global setting is in globals_flags[1] */ static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */ @@ -11711,6 +11715,7 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_ ast_cli(a->fd, " MatchAuthUsername: %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)); ast_cli(a->fd, " Allow overlap dialing: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP))); ast_cli(a->fd, " Promsic. redir: %s\n", cli_yesno(ast_test_flag(&global_flags[0], SIP_PROMISCREDIR))); ast_cli(a->fd, " SIP domain support: %s\n", cli_yesno(!AST_LIST_EMPTY(&domain_list))); @@ -17469,6 +17474,8 @@ static struct sip_user *build_user(const char *name, struct ast_variable *v, int user->allowtransfer = global_allowtransfer; user->maxcallbitrate = default_maxcallbitrate; user->autoframing = global_autoframing; + if (global_callcounter) + user->call_limit=999; user->prefs = default_prefs; /* set default context */ strcpy(user->context, default_context); @@ -17516,6 +17523,8 @@ static struct sip_user *build_user(const char *name, struct ast_variable *v, int ast_copy_string(user->mohsuggest, v->value, sizeof(user->mohsuggest)); } else if (!strcasecmp(v->name, "accountcode")) { ast_copy_string(user->accountcode, v->value, sizeof(user->accountcode)); + } else if (!strcasecmp(v->name, "callcounter")) { + user->call_limit = ast_strue(v->value) ? 999 : 0; } else if (!strcasecmp(v->name, "call-limit")) { user->call_limit = atoi(v->value); if (user->call_limit < 0) @@ -17585,6 +17594,8 @@ static void set_peer_defaults(struct sip_peer *peer) peer->rtpkeepalive = global_rtpkeepalive; peer->allowtransfer = global_allowtransfer; peer->autoframing = global_autoframing; + if (global_callcounter) + peer->call_limit=999; strcpy(peer->vmexten, default_vmexten); peer->secret[0] = '\0'; peer->md5secret[0] = '\0'; @@ -17810,6 +17821,8 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str ast_copy_string(peer->regexten, v->value, sizeof(peer->regexten)); } else if (!strcasecmp(v->name, "callbackextension")) { ast_copy_string(callback, v->value, sizeof(callback)); + } else if (!strcasecmp(v->name, "callcounter")) { + peer->call_limit = ast_strue(v->value) ? 999 : 0; } else if (!strcasecmp(v->name, "call-limit")) { peer->call_limit = atoi(v->value); if (peer->call_limit < 0) @@ -18057,6 +18070,7 @@ static int reload_config(enum channelreloadreason reason) autocreatepeer = DEFAULT_AUTOCREATEPEER; global_autoframing = 0; global_allowguest = DEFAULT_ALLOWGUEST; + global_callcounter = DEFAULT_CALLCOUNTER; global_match_auth_username = FALSE; /*!< Match auth username if available instead of From: Default off. */ global_rtptimeout = 0; global_rtpholdtimeout = 0; @@ -18112,6 +18126,8 @@ static int reload_config(enum channelreloadreason reason) ast_copy_string(default_context, v->value, sizeof(default_context)); } else if (!strcasecmp(v->name, "subscribecontext")) { ast_copy_string(default_subscribecontext, v->value, sizeof(default_subscribecontext)); + } else if (!strcasecmp(v->name, "callcounter")) { + global_callcounter = ast_true(v->value) ? 1 : 0; } else if (!strcasecmp(v->name, "allowguest")) { global_allowguest = ast_true(v->value) ? 1 : 0; } else if (!strcasecmp(v->name, "realm")) { diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index ae04d37bd..ae5c9ba61 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -23,6 +23,15 @@ ; Active SIP peers will not be reconfigured ; +; ** Deprecated options ** +; The "call-limit" configuation option is deprecated. It still works in +; this version of Asterisk, but will disappear in the next version. +; You are encouraged to use the dialplan groupcount functionality +; to enforce call limits instead of using this channel-specific method. +; +; You can still set limits per device in sip.conf or in a database by using +; "setvar" to set variables that can be used in the dialplan for various limits. + [general] context=default ; Default context for incoming calls ;allowguest=no ; Allow or reject guest calls (default is yes) @@ -206,14 +215,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; (See extensions.conf.sample for examples) ; chan_sip support two major formats for notifications: dialog-info and SIMPLE ; -; You will get more detailed reports (busy etc) if you have a call limit set -; for a device. When the call limit is filled, we will indicate busy. Note that -; you need at least 2 in order to be able to do attended transfers. +; You will get more detailed reports (busy etc) if you have a call counter enabled +; for a device. ; -; If you set the busylevel in addition to the call limit, we will indicate busy -; when we have a number of calls that matches busylevel, but still allow calls -; up to the call-limit. This allows for transfers while still having blinking -; lamps and queues understanding that a device is busy. +; If you set the busylevel, we will indicate busy when we have a number of calls that +; matches the busylevel treshold. ; ; For queues, you will need this level of detail in status reporting, regardless ; if you use SIP subscriptions. Queues and manager use the same internal interface @@ -230,12 +236,14 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;notifyhold = yes ; Notify subscriptions on HOLD state (default: no) ; Turning on notifyringing and notifyhold will add a lot ; more database transactions if you are using realtime. -;limitonpeer = yes ; Apply call limits on peers only. This will improve +;limitonpeer = yes ; Apply call counting on peers only. This will improve ; status notification when you are using type=friend ; Inbound calls, that really apply to the user part ; of a friend will now be added to and compared with - ; the peer limit instead of applying two call limits, + ; the peer counter instead of applying two call counters, ; one for the peer and one for the user. +;callcounter = yes ; Enable call counters on devices. This can be set per + ; device too. ;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT ----------------------- ; @@ -561,7 +569,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; setvar setvar ; callerid callerid ; amaflags amaflags -; call-limit call-limit +; call-limit call-limit (deprecated) +; callcounter callcounter ; allowoverlap allowoverlap ; allowsubscribe allowsubscribe ; allowtransfer allowtransfer @@ -603,9 +612,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;fromdomain=provider.sip.domain ;host=box.provider.com ;usereqphone=yes ; This provider requires ";user=phone" on URI -;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer - ; Call-limits will not be enforced on real-time peers, - ; since they are not stored in-memory +;callcounter=yes ; Enable call counter ;busylevel=2 ; Signal busy at 2 or more calls ;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer ;port=80 ; The port number we want to connect to on the remote side @@ -690,11 +697,10 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time - ; from the phone to asterisk + ; from the phone to asterisk (deprecated) ; 1 for the explicit peer, 1 for the explicit user, ; remember that a friend equals 1 peer and 1 user in ; memory - ; This will affect your subscriptions as well. ; There is no combined call counter for a "friend" ; so there's currently no way in sip.conf to limit ; to one inbound or outbound call per phone. Use |