[asterisk-commits] seanbright: trunk r154187 - in /trunk: channels/ configs/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Nov 4 11:00:46 CST 2008
Author: seanbright
Date: Tue Nov 4 11:00:45 2008
New Revision: 154187
URL: http://svn.digium.com/view/asterisk?view=rev&rev=154187
Log:
Allow devices that accept dialog-info+xml (like snoms) to get the Caller ID of
the calling party when subscribed to the state of an extension that is ringing.
This has some limitations which are documented in sip.conf.sample.
(closes issue #13827)
Reported by: seanbright
Patches:
issue13827.patch uploaded by seanbright (license 71)
Reviewed by: russellb
Modified:
trunk/channels/chan_sip.c
trunk/configs/sip.conf.sample
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=154187&r1=154186&r2=154187
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Nov 4 11:00:45 2008
@@ -961,6 +961,7 @@
#define DEFAULT_ALLOW_EXT_DOM TRUE /*!< Allow external domains */
#define DEFAULT_REALM "asterisk" /*!< Realm for HTTP digest authentication */
#define DEFAULT_NOTIFYRINGING TRUE /*!< Notify devicestate system on ringing state */
+#define DEFAULT_NOTIFYCID FALSE /*!< Include CID with ringing notifications */
#define DEFAULT_PEDANTIC FALSE /*!< Avoid following SIP standards for dialog matching */
#define DEFAULT_AUTOCREATEPEER FALSE /*!< Don't create peers automagically */
#define DEFAULT_MATCHEXTERNIPLOCALLY FALSE /*!< Match extern IP locally default setting */
@@ -1030,6 +1031,7 @@
static int global_notifyringing; /*!< Send notifications on ringing */
static int global_notifyhold; /*!< Send notifications on hold */
+static int global_notifycid; /*!< Send CID with ringing notifications */
static int global_match_auth_username; /*!< Match auth username if available instead of From: Default off. */
static int global_relaxdtmf; /*!< Relax DTMF */
@@ -10029,23 +10031,42 @@
ast_str_append(&tmp, 0, "<?xml version=\"1.0\"?>\n");
ast_str_append(&tmp, 0, "<dialog-info xmlns=\"urn:ietf:params:xml:ns:dialog-info\" version=\"%d\" state=\"%s\" entity=\"%s\">\n", p->dialogver++, full ? "full" : "partial", mto);
if ((state & AST_EXTENSION_RINGING) && global_notifyringing) {
+ const char *local_display = p->exten, *local_target = mto;
+
+ /* There are some limitations to how this works. The primary one is that the
+ callee must be dialing the same extension that is being monitored. Simply dialing
+ the hint'd device is not sufficient. */
+ if (global_notifycid) {
+ struct ast_channel *caller = NULL;
+
+ while ((caller = ast_channel_walk_locked(caller))) {
+ if (caller->pbx &&
+ (!strcasecmp(caller->macroexten, p->exten) || !strcasecmp(caller->exten, p->exten)) &&
+ !strcasecmp(caller->context, p->context)) {
+ local_display = ast_strdupa(caller->cid.cid_name);
+ local_target = ast_strdupa(caller->cid.cid_num);
+ ast_channel_unlock(caller);
+ break;
+ }
+ ast_channel_unlock(caller);
+ }
+ }
+
/* We create a fake call-id which the phone will send back in an INVITE
Replaces header which we can grab and do some magic with. */
ast_str_append(&tmp, 0,
"<dialog id=\"%s\" call-id=\"pickup-%s\" direction=\"recipient\">\n"
"<remote>\n"
- /* Note that the identity and target elements for the local participant are currently
- (and may forever be) incorrect since we have no reliable way to get at that information
- at the moment. Luckily the phone seems to still live happily without it being correct */
- "<identity>%s</identity>\n"
+ /* See the limitations of this above. Luckily the phone seems to still be
+ happy when these values are not correct. */
+ "<identity display=\"%s\">%s</identity>\n"
"<target uri=\"%s\"/>\n"
"</remote>\n"
"<local>\n"
"<identity>%s</identity>\n"
"<target uri=\"%s\"/>\n"
"</local>\n",
- p->exten, p->callid,
- mto, mto, mto, mto);
+ p->exten, p->callid, local_display, local_target, local_target, mto, mto);
} else {
ast_str_append(&tmp, 0, "<dialog id=\"%s\">\n", p->exten);
}
@@ -14457,6 +14478,9 @@
ast_cli(a->fd, " Outbound reg. timeout: %d secs\n", global_reg_timeout);
ast_cli(a->fd, " Outbound reg. attempts: %d\n", global_regattempts_max);
ast_cli(a->fd, " Notify ringing state: %s\n", cli_yesno(global_notifyringing));
+ if (global_notifyringing) {
+ ast_cli(a->fd, " Include CID: %s\n", cli_yesno(global_notifycid));
+ }
ast_cli(a->fd, " Notify hold state: %s\n", cli_yesno(global_notifyhold));
ast_cli(a->fd, " SIP Transfer mode: %s\n", transfermode2str(global_allowtransfer));
ast_cli(a->fd, " Max Call Bitrate: %d kbps\n", default_maxcallbitrate);
@@ -22393,6 +22417,7 @@
global_regcontext[0] = '\0';
sip_cfg.regextenonqualify = DEFAULT_REGEXTENONQUALIFY;
global_notifyringing = DEFAULT_NOTIFYRINGING;
+ global_notifycid = DEFAULT_NOTIFYCID;
global_notifyhold = FALSE; /*!< Keep track of hold status for a peer */
sip_cfg.directrtpsetup = FALSE; /* Experimental feature, disabled by default */
sip_cfg.alwaysauthreject = DEFAULT_ALWAYSAUTHREJECT;
@@ -22608,6 +22633,8 @@
global_notifyringing = ast_true(v->value);
} else if (!strcasecmp(v->name, "notifyhold")) {
global_notifyhold = ast_true(v->value);
+ } else if (!strcasecmp(v->name, "notifycid")) {
+ global_notifycid = ast_true(v->value);
} else if (!strcasecmp(v->name, "alwaysauthreject")) {
sip_cfg.alwaysauthreject = ast_true(v->value);
} else if (!strcasecmp(v->name, "mohinterpret")) {
Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=154187&r1=154186&r2=154187
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Tue Nov 4 11:00:45 2008
@@ -376,6 +376,16 @@
;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.
+;notifycid = yes ; Control whether caller ID information is sent along with
+ ; dialog-info+xml notifications (supported by snom phones).
+ ; Note that this feature will only work properly when the
+ ; incoming call is using the same extension and context that
+ ; is being used as the hint for the called extension. This means
+ ; that it won't work when using subscribecontext for your sip
+ ; user or peer (if subscribecontext is different than context).
+ ; This is also limited to a single caller, meaning that if an
+ ; extension is ringing because multiple calls are incoming,
+ ; only one will be used as the source of caller ID.
;callcounter = yes ; Enable call counters on devices. This can be set per
; device too.
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