res_pjsip: Add contact_user to endpoint

contact_user, when specified on an endpoint, will override the user
portion of the Contact header on outgoing requests.

Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4

CHANGES

@@ -42,6 +42,11 @@ chan_pjsip
    dialplan function PJSIP_MEDIA_OFFER, this allows the formats on a PJSIP
    channel to be re-negotiated and updated after session set up.
 
+res_pjsip
+------------------
+ * A new endpoint configuration parameter 'contact_user' has been added which
+   when set will override the default user set on Contact headers in outgoing
+   requests.
 
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 13 to Asterisk 14 --------------------
@@ -407,7 +412,6 @@ cdr_csv
  * Added a new configuration option, "newcdrcolumns", which enables use of the
    post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.
 
-
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 13.10.0 to Asterisk 13.11.0 ----------
 ------------------------------------------------------------------------------

configs/samples/pjsip.conf.sample

@@ -762,6 +762,8 @@
 ;rtp_timeout_hold= ; Hang up channel if RTP is not received for the specified
                    ; number of seconds when the channel is on hold (default:
                    ; "0" or not enabled)
+;contact_user= ; On outgoing requests, force the user portion of the Contact
+               ; header to this value (default: "")
 
 ;==========================AUTH SECTION OPTIONS=========================
 ;[auth]

contrib/ast-db-manage/config/versions/4e2493ef32e6_add_contact_user_to_endpoint.py

@@ -0,0 +1,22 @@
+"""Add contact_user to endpoint
+
+Revision ID: 4e2493ef32e6
+Revises: 3772f8f828da
+Create Date: 2016-08-16 14:19:58.918466
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '4e2493ef32e6'
+down_revision = '3772f8f828da'
+
+from alembic import op
+import sqlalchemy as sa
+
+
+def upgrade():
+    op.add_column('ps_endpoints', sa.Column('contact_user', sa.String(80)))
+
+
+def downgrade():
+    op.drop_column('ps_endpoints', 'contact_user')

include/asterisk/res_pjsip.h

@@ -755,6 +755,8 @@ struct ast_sip_endpoint {
 	struct ast_acl_list *contact_acl;
 	/*! The number of seconds into call to disable fax detection.  (0 = disabled) */
 	unsigned int faxdetect_timeout;
+	/*! Override the user on the outgoing Contact header with this value. */
+	char *contact_user;
 };
 
 /*!

res/res_pjsip.c

@@ -916,6 +916,12 @@
 						then the <replaceable>context</replaceable> setting is used.
 					</para></description>
 				</configOption>
+				<configOption name="contact_user" default="">
+					<synopsis>Force the user on the outgoing Contact header to this value.</synopsis>
+					<description><para>
+						On outbound requests, force the user portion of the Contact header to this value.
+					</para></description>
+				</configOption>
 			</configObject>
 			<configObject name="auth">
 				<synopsis>Authentication type</synopsis>
@@ -2865,8 +2871,16 @@ pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint,
 	/* Update the dialog with the new local URI, we do it afterwards so we can use the dialog pool for construction */
 	pj_strdup_with_null(dlg->pool, &dlg->local.info_str, &local_uri);
 	dlg->local.info->uri = pjsip_parse_uri(dlg->pool, dlg->local.info_str.ptr, dlg->local.info_str.slen, 0);
+
 	dlg->local.contact = pjsip_parse_hdr(dlg->pool, &HCONTACT, local_uri.ptr, local_uri.slen, NULL);
 
+	if (!ast_strlen_zero(endpoint->contact_user)) {
+		pjsip_sip_uri *sip_uri;
+
+		sip_uri = pjsip_uri_get_uri(dlg->local.contact->uri);
+		pj_strdup2(dlg->pool, &sip_uri->user, endpoint->contact_user);
+	}
+
 	/* If a request user has been specified and we are permitted to change it, do so */
 	if (!ast_strlen_zero(request_user)) {
 		pjsip_sip_uri *sip_uri;
@@ -3168,6 +3182,18 @@ static int create_out_of_dialog_request(const pjsip_method *method, struct ast_s
 		return -1;
 	}
 
+	if (endpoint && !ast_strlen_zero(endpoint->contact_user)){
+		pjsip_contact_hdr *contact_hdr;
+		pjsip_sip_uri *contact_uri;
+		static const pj_str_t HCONTACT = { "Contact", 7 };
+
+		contact_hdr = pjsip_msg_find_hdr_by_name((*tdata)->msg, &HCONTACT, NULL);
+		if (contact_hdr) {
+			contact_uri = pjsip_uri_get_uri(contact_hdr->uri);
+			pj_strdup2(pool, &contact_uri->user, endpoint->contact_user);
+		}
+	}
+
 	/* Add the user=phone parameter if applicable */
 	ast_sip_add_usereqphone(endpoint, (*tdata)->pool, (*tdata)->msg->line.req.uri);
 

res/res_pjsip/pjsip_configuration.c

@@ -1208,6 +1208,31 @@ static int voicemail_extension_to_str(const void *obj, const intptr_t *args, cha
 	return 0;
 }
 
+static int contact_user_handler(const struct aco_option *opt,
+	struct ast_variable *var, void *obj)
+{
+	struct ast_sip_endpoint *endpoint = obj;
+
+	endpoint->contact_user = ast_strdup(var->value);
+	if (!endpoint->contact_user) {
+		return -1;
+	}
+
+	return 0;
+}
+
+static int contact_user_to_str(const void *obj, const intptr_t *args, char **buf)
+{
+	const struct ast_sip_endpoint *endpoint = obj;
+
+	*buf = ast_strdup(endpoint->contact_user);
+	if (!(*buf)) {
+		return -1;
+	}
+
+	return 0;
+}
+
 static void *sip_nat_hook_alloc(const char *name)
 {
 	return ast_sorcery_generic_alloc(sizeof(struct ast_sip_nat_hook), NULL);
@@ -1905,6 +1930,7 @@ int ast_res_pjsip_initialize_configuration(void)
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "contact_permit", "", endpoint_acl_handler, NULL, NULL, 0, 0);
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "contact_acl", "", endpoint_acl_handler, contact_acl_to_str, NULL, 0, 0);
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "subscribe_context", "", OPT_CHAR_ARRAY_T, 0, CHARFLDSET(struct ast_sip_endpoint, subscription.context));
+	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "contact_user", "", contact_user_handler, contact_user_to_str, NULL, 0, 0);
 
 	if (ast_sip_initialize_sorcery_transport()) {
 		ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n");
@@ -2036,6 +2062,7 @@ static void endpoint_destructor(void* obj)
 	ao2_cleanup(endpoint->persistent);
 	ast_variables_destroy(endpoint->channel_vars);
 	AST_VECTOR_FREE(&endpoint->ident_method_order);
+	ast_free(endpoint->contact_user);
 }
 
 static int init_subscription_configuration(struct ast_sip_endpoint_subscription_configuration *subscription)