[Asterisk-code-review] chan sip: 3PCC patch for AMI "SIPnotify" (asterisk[master])

Yasuhiko Kamata asteriskteam at digium.com
Wed Dec 6 21:07:21 CST 2017


Yasuhiko Kamata has uploaded this change for review. ( https://gerrit.asterisk.org/7461


Change subject: chan_sip: 3PCC patch for AMI "SIPnotify"
......................................................................

chan_sip: 3PCC patch for AMI "SIPnotify"

A patch for sending in-dialog SIP NOTIFY message
with "SIPnotify" AMI action for latest (14.x and 15.x) asterisk.

ASTERISK-27461

Change-Id: I5797ded4752acd966db6b13971284db684cc5ab4
---
M channels/chan_sip.c
1 file changed, 62 insertions(+), 19 deletions(-)



  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/61/7461/1

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index a829e20..3c6b4f4 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -15591,8 +15591,9 @@
 {
 	const char *channame = astman_get_header(m, "Channel");
 	struct ast_variable *vars = astman_get_variables_order(m, ORDER_NATURAL);
-	struct sip_pvt *p;
+	struct sip_pvt *p = NULL;
 	struct ast_variable *header, *var;
+	char indialog = 0;
 
 	if (ast_strlen_zero(channame)) {
 		astman_send_error(s, m, "SIPNotify requires a channel name");
@@ -15603,25 +15604,61 @@
 		channame += 4;
 	}
 
-	if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY, NULL, 0))) {
-		astman_send_error(s, m, "Unable to build sip pvt data for notify (memory/socket error)");
-		return 0;
+        // check if Call-ID variable is set
+        for (var = vars; var; var = var->next) {
+          if (!strcasecmp(var->name, "Call-ID")) {
+            struct sip_pvt tmp_dialog = {
+              .callid = var->value,
+            };
+
+            p = ao2_find(dialogs, &tmp_dialog, OBJ_POINTER);
+            if (!p) {
+              astman_send_error(s, m, "Call-ID not found");
+              return 0;
+            }
+            indialog = 1;
+          }
+        }
+
+        if (!indialog) {
+		if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY, NULL, 0))) {
+			astman_send_error(s, m, "Unable to build sip pvt data for notify (memory/socket error)");
+			return 0;
+		}
+
+		if (create_addr(p, channame, NULL, 0)) {
+			/* Maybe they're not registered, etc. */
+			dialog_unlink_all(p);
+			dialog_unref(p, "unref dialog inside for loop" );
+			/* sip_destroy(p); */
+			astman_send_error(s, m, "Could not create address");
+			return 0;
+		}
+
+		/* Notify is outgoing call */
+		ast_set_flag(&p->flags[0], SIP_OUTGOING);
+		sip_notify_alloc(p);
+
+		p->notify->headers = header = ast_variable_new("Subscription-State", "terminated", "");
+        } else {
+          if (!(p->notify)) {
+	    sip_notify_alloc(p);
+          } else {
+            ast_variables_destroy(p->notify->headers);
+          }
+
+	  p->notify->headers = header = ast_variable_new("Subscription-State", "terminated", "");
 	}
 
-	if (create_addr(p, channame, NULL, 0)) {
-		/* Maybe they're not registered, etc. */
-		dialog_unlink_all(p);
-		dialog_unref(p, "unref dialog inside for loop" );
-		/* sip_destroy(p); */
-		astman_send_error(s, m, "Could not create address");
-		return 0;
-	}
-
-	/* Notify is outgoing call */
-	ast_set_flag(&p->flags[0], SIP_OUTGOING);
-	sip_notify_alloc(p);
-
-	p->notify->headers = header = ast_variable_new("Subscription-State", "terminated", "");
+        /* if (pref) {
+          ast_string_field_set(p, callid, pref->callid);
+          ast_string_field_set(p, fromuser, pref->fromuser);
+          ast_string_field_set(p, fromname, pref->fromname);
+          ast_string_field_set(p, tag, pref->tag);
+          ast_string_field_set(p, theirtag, pref->theirtag);
+          p->ocseq = pref->ocseq;
+          (pref->ocseq)++;
+        } */
 
 	for (var = vars; var; var = var->next) {
 		if (!strcasecmp(var->name, "Content")) {
@@ -15630,21 +15667,27 @@
 			ast_str_append(&p->notify->content, 0, "%s", var->value);
 		} else if (!strcasecmp(var->name, "Content-Length")) {
 			ast_log(LOG_WARNING, "it is not necessary to specify Content-Length, ignoring\n");
+		} else if (!strcasecmp(var->name, "Call-ID")) {
+                  // do nothing here
 		} else {
 			header->next = ast_variable_new(var->name, var->value, "");
 			header = header->next;
 		}
 	}
 
+        if (!indialog) {
 	/* Now that we have the peer's address, set our ip and change callid */
 	ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
 	build_via(p);
 
-	change_callid_pvt(p, NULL);
+        change_callid_pvt(p, NULL);
 
 	sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
 	transmit_invite(p, SIP_NOTIFY, 0, 2, NULL);
 	dialog_unref(p, "bump down the count of p since we're done with it.");
+        } else {
+          transmit_invite(p, SIP_NOTIFY, 0, 1, NULL);
+        }
 
 	astman_send_ack(s, m, "Notify Sent");
 	ast_variables_destroy(vars);

-- 
To view, visit https://gerrit.asterisk.org/7461
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-MessageType: newchange
Gerrit-Change-Id: I5797ded4752acd966db6b13971284db684cc5ab4
Gerrit-Change-Number: 7461
Gerrit-PatchSet: 1
Gerrit-Owner: Yasuhiko Kamata <yasuhiko.kamata at nxtg.co.jp>
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