[asterisk-commits] moy: branch moy/webrtc-11 r409360 - in /team/moy/webrtc-11: ./ apps/ channels...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Sun Mar 2 19:02:35 CST 2014


Author: moy
Date: Sun Mar  2 19:02:31 2014
New Revision: 409360

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=409360
Log:
Merged r408853:r409344 from /branches/11 to /team/moy/webrtc-11

Modified:
    team/moy/webrtc-11/   (props changed)
    team/moy/webrtc-11/Makefile.rules
    team/moy/webrtc-11/apps/app_queue.c
    team/moy/webrtc-11/channels/chan_sip.c
    team/moy/webrtc-11/configs/res_fax.conf.sample
    team/moy/webrtc-11/configs/voicemail.conf.sample
    team/moy/webrtc-11/include/asterisk/rtp_engine.h
    team/moy/webrtc-11/main/rtp_engine.c
    team/moy/webrtc-11/res/res_fax.c
    team/moy/webrtc-11/res/res_rtp_asterisk.c
    team/moy/webrtc-11/utils/astman.c

Propchange: team/moy/webrtc-11/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.

Propchange: team/moy/webrtc-11/
------------------------------------------------------------------------------
--- svn:mergeinfo (original)
+++ svn:mergeinfo Sun Mar  2 19:02:31 2014
@@ -1,1 +1,1 @@
-/branches/11:400822-408853
+/branches/11:400822-409344

Modified: team/moy/webrtc-11/Makefile.rules
URL: http://svnview.digium.com/svn/asterisk/team/moy/webrtc-11/Makefile.rules?view=diff&rev=409360&r1=409359&r2=409360
==============================================================================
--- team/moy/webrtc-11/Makefile.rules (original)
+++ team/moy/webrtc-11/Makefile.rules Sun Mar  2 19:02:31 2014
@@ -35,7 +35,7 @@
     CMD_PREFIX=
 endif
 
-OPTIMIZE?=-O6
+OPTIMIZE?=-O3
 ifneq ($(findstring darwin,$(OSARCH)),)
   ifeq ($(shell if test `/usr/bin/sw_vers -productVersion | cut -c4` -gt 5; then echo 6; else echo 0; fi),6)
     # Snow Leopard/Lion has an issue with this optimization flag on large files (like chan_sip)

Modified: team/moy/webrtc-11/apps/app_queue.c
URL: http://svnview.digium.com/svn/asterisk/team/moy/webrtc-11/apps/app_queue.c?view=diff&rev=409360&r1=409359&r2=409360
==============================================================================
--- team/moy/webrtc-11/apps/app_queue.c (original)
+++ team/moy/webrtc-11/apps/app_queue.c Sun Mar  2 19:02:31 2014
@@ -6276,22 +6276,25 @@
 
 				ast_queue_log(q->name, "NONE", mem->membername, (paused ? "PAUSE" : "UNPAUSE"), "%s", S_OR(reason, ""));
 
-				/*** DOCUMENTATION
-				<managerEventInstance>
-					<synopsis>Raised when a member is paused/unpaused in the queue with a reason.</synopsis>
-					<syntax>
-						<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Queue'])" />
-						<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Location'])" />
-						<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='MemberName'])" />
-						<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Paused'])" />
-					</syntax>
-					<see-also>
-						<ref type="application">PauseQueueMember</ref>
-						<ref type="application">UnPauseQueueMember</ref>
-					</see-also>
-				</managerEventInstance>
-				***/
 				if (!ast_strlen_zero(reason)) {
+					/*** DOCUMENTATION
+					<managerEventInstance>
+						<synopsis>Raised when a member is paused/unpaused in the queue with a reason.</synopsis>
+						<syntax>
+							<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Queue'])" />
+							<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Location'])" />
+							<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='MemberName'])" />
+							<xi:include xpointer="xpointer(/docs/managerEvent[@name='QueueMemberStatus']/managerEventInstance/syntax/parameter[@name='Paused'])" />
+							<parameter name="Reason">
+								<para>The reason given for pausing or unpausing a queue member.</para>
+							</parameter>
+						</syntax>
+						<see-also>
+							<ref type="application">PauseQueueMember</ref>
+							<ref type="application">UnPauseQueueMember</ref>
+						</see-also>
+					</managerEventInstance>
+					***/
 					manager_event(EVENT_FLAG_AGENT, "QueueMemberPaused",
 						"Queue: %s\r\n"
 						"Location: %s\r\n"

Modified: team/moy/webrtc-11/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/moy/webrtc-11/channels/chan_sip.c?view=diff&rev=409360&r1=409359&r2=409360
==============================================================================
--- team/moy/webrtc-11/channels/chan_sip.c (original)
+++ team/moy/webrtc-11/channels/chan_sip.c Sun Mar  2 19:02:31 2014
@@ -7051,10 +7051,12 @@
 		ast_debug(4, "SIP Transfer: Not hanging up right now... Rescheduling hangup for %s.\n", p->callid);
 		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 		ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER);	/* Really hang up next time */
-		ast_channel_tech_pvt_set(p->owner, dialog_unref(ast_channel_tech_pvt(p->owner), "unref p->owner->tech_pvt"));
-		sip_pvt_lock(p);
-		p->owner = NULL;  /* Owner will be gone after we return, so take it away */
-		sip_pvt_unlock(p);
+		if (p->owner) {
+			ast_channel_tech_pvt_set(p->owner, dialog_unref(ast_channel_tech_pvt(p->owner), "unref p->owner->tech_pvt"));
+			sip_pvt_lock(p);
+			p->owner = NULL;  /* Owner will be gone after we return, so take it away */
+			sip_pvt_unlock(p);
+		}
 		ast_module_unref(ast_module_info->self);
 		return 0;
 	}
@@ -7083,7 +7085,7 @@
 
 	stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
 
-	append_history(p, needcancel ? "Cancel" : "Hangup", "Cause %s", p->owner ? ast_cause2str(p->hangupcause) : "Unknown");
+	append_history(p, needcancel ? "Cancel" : "Hangup", "Cause %s", ast_cause2str(p->hangupcause));
 
 	/* Disconnect */
 	disable_dsp_detect(p);
@@ -7585,7 +7587,9 @@
 			AST_SCHED_DEL_UNREF(sched, p->t38id, dialog_unref(p, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
 			parameters.max_ifp = ast_udptl_get_far_max_ifp(p->udptl);
 			parameters.request_response = AST_T38_REQUEST_NEGOTIATE;
-			ast_queue_control_data(p->owner, AST_CONTROL_T38_PARAMETERS, &parameters, sizeof(parameters));
+			if (p->owner) {
+				ast_queue_control_data(p->owner, AST_CONTROL_T38_PARAMETERS, &parameters, sizeof(parameters));
+			}
 			/* we need to return a positive value here, so that applications that
 			 * send this request can determine conclusively whether it was accepted or not...
 			 * older versions of chan_sip would just silently accept it and return zero.
@@ -12506,9 +12510,9 @@
 {
 	struct ast_str *tmp = ast_str_alloca(256);
 	char tmp2[256];
-	char *lid_num = NULL;
-	char *lid_name = NULL;
-	int lid_pres = AST_PRES_NUMBER_NOT_AVAILABLE;
+	char *lid_num;
+	char *lid_name;
+	int lid_pres;
 	const char *fromdomain;
 	const char *privacy = NULL;
 	const char *screen = NULL;
@@ -12519,22 +12523,20 @@
 		return 0;
 	}
 
-	if (p->owner) {
-		connected_id = ast_channel_connected_effective_id(p->owner);
-
-		if (connected_id.number.valid) {
-			lid_num = connected_id.number.str;
-		}
-		if (connected_id.name.valid) {
-			lid_name = connected_id.name.str;
-		}
-		lid_pres = ast_party_id_presentation(&connected_id);
-	}
-
-	if (ast_strlen_zero(lid_num))
+	if (!p->owner) {
 		return 0;
-	if (ast_strlen_zero(lid_name))
+	}
+	connected_id = ast_channel_connected_effective_id(p->owner);
+	lid_num = S_COR(connected_id.number.valid, connected_id.number.str, NULL);
+	if (!lid_num) {
+		return 0;
+	}
+	lid_name = S_COR(connected_id.name.valid, connected_id.name.str, NULL);
+	if (!lid_name) {
 		lid_name = lid_num;
+	}
+	lid_pres = ast_party_id_presentation(&connected_id);
+
 	fromdomain = S_OR(p->fromdomain, ast_sockaddr_stringify_host_remote(&p->ourip));
 
 	lid_num = ast_uri_encode(lid_num, tmp2, sizeof(tmp2), ast_uri_sip_user);
@@ -17897,13 +17899,15 @@
 	}
 
 	/* Determine transfer context */
-	if (transferer->owner)	/* Mimic behaviour in res_features.c */
+	if (transferer->owner) {
+		/* By default, use the context in the channel sending the REFER */
 		transfer_context = pbx_builtin_getvar_helper(transferer->owner, "TRANSFER_CONTEXT");
-
-	/* By default, use the context in the channel sending the REFER */
+		if (ast_strlen_zero(transfer_context)) {
+			transfer_context = ast_channel_macrocontext(transferer->owner);
+		}
+	}
 	if (ast_strlen_zero(transfer_context)) {
-		transfer_context = S_OR(ast_channel_macrocontext(transferer->owner),
-					S_OR(transferer->context, sip_cfg.default_context));
+		transfer_context = S_OR(transferer->context, sip_cfg.default_context);
 	}
 
 	ast_string_field_set(refer, refer_to_context, transfer_context);
@@ -17957,14 +17961,18 @@
 	if (sip_debug_test_pvt(p))
 		ast_verbose("Looking for %s in %s\n", c, p->context);
 
-	if (p->owner)	/* Mimic behaviour in res_features.c */
+	/* Determine transfer context */
+	if (p->owner) {
+		/* By default, use the context in the channel sending the REFER */
 		transfer_context = pbx_builtin_getvar_helper(p->owner, "TRANSFER_CONTEXT");
-
-	/* By default, use the context in the channel sending the REFER */
+		if (ast_strlen_zero(transfer_context)) {
+			transfer_context = ast_channel_macrocontext(p->owner);
+		}
+	}
 	if (ast_strlen_zero(transfer_context)) {
-		transfer_context = S_OR(ast_channel_macrocontext(p->owner),
-					S_OR(p->context, sip_cfg.default_context));
-	}
+		transfer_context = S_OR(p->context, sip_cfg.default_context);
+	}
+
 	if (ast_exists_extension(NULL, transfer_context, c, 1, NULL)) {
 		/* This is a blind transfer */
 		ast_debug(1, "SIP Bye-also transfer to Extension %s@%s \n", c, transfer_context);
@@ -22868,9 +22876,11 @@
 					/* This 200 OK's SDP is not acceptable, so we need to ack, then hangup */
 					/* For re-invites, we try to recover */
 					ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
-					ast_channel_hangupcause_set(p->owner, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
 					p->hangupcause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
-					sip_queue_hangup_cause(p, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
+					if (p->owner) {
+						ast_channel_hangupcause_set(p->owner, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
+						sip_queue_hangup_cause(p, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
+					}
 				}
 			}
 			ast_rtp_instance_activate(p->rtp);
@@ -25453,12 +25463,6 @@
 			ast_debug(2, "No SDP in Invite, third party call control\n");
 		}
 
-		/* Queue NULL frame to prod ast_rtp_bridge if appropriate */
-		/* This seems redundant ... see !p-owner above */
-		if (p->owner)
-			ast_queue_frame(p->owner, &ast_null_frame);
-
-
 		/* Initialize the context if it hasn't been already */
 		if (ast_strlen_zero(p->context))
 			ast_string_field_set(p, context, sip_cfg.default_context);
@@ -29951,7 +29955,7 @@
 	if (sip_cfg.callevents)
 		manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
 			"Channel: %s\r\nChanneltype: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
-			p->owner? ast_channel_name(p->owner) : "", "SIP", p->callid, p->fullcontact, p->peername);
+			p->owner ? ast_channel_name(p->owner) : "", "SIP", p->callid, p->fullcontact, p->peername);
 	sip_pvt_unlock(p);
 	if (!tmpc) {
 		dialog_unlink_all(p);

Modified: team/moy/webrtc-11/configs/res_fax.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/moy/webrtc-11/configs/res_fax.conf.sample?view=diff&rev=409360&r1=409359&r2=409360
==============================================================================
--- team/moy/webrtc-11/configs/res_fax.conf.sample (original)
+++ team/moy/webrtc-11/configs/res_fax.conf.sample Sun Mar  2 19:02:31 2014
@@ -4,12 +4,12 @@
 ; Maximum Transmission Rate
 ; Possible values are { 2400 | 4800 | 7200 | 9600 | 12000 | 14400 }
 ; Set this value to the maximum desired transfer rate.  Default: 14400
-maxrate=14400
+;maxrate=14400
 
 ; Minimum Transmission Rate
 ; Possible values are { 2400 | 4800 | 7200 | 9600 | 12000 | 14400 }
 ; Set this value to the minimum desired transfer rate.  Default: 4800
-minrate=4800
+;minrate=4800
 
 ; Send Progress/Status events to manager session
 ; Manager events with 'call' class permissions will receive events indicating the
@@ -21,8 +21,8 @@
 ; modem capabilities
 ; Possible values are { v17 | v27 | v29 }
 ; Set this value to modify the default modem options.  Default: v17,v27,v29
-modems=v17,v27,v29
+;modems=v17,v27,v29
 
 ; Enable/disable T.30 ECM (error correction mode) by default.
 ; Default: Enabled
-ecm=yes
+;ecm=yes

Modified: team/moy/webrtc-11/configs/voicemail.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/moy/webrtc-11/configs/voicemail.conf.sample?view=diff&rev=409360&r1=409359&r2=409360
==============================================================================
--- team/moy/webrtc-11/configs/voicemail.conf.sample (original)
+++ team/moy/webrtc-11/configs/voicemail.conf.sample Sun Mar  2 19:02:31 2014
@@ -169,7 +169,8 @@
 ; Short 24h date format for pager use
 ;pagerdateformat=%T %D
 ;
-; You can override the default program to send e-mail if you wish, too
+; Using the mailcmd option, you can specify what command is called for
+; outbound E-mail. The default is shown below.
 ;
 ;mailcmd=/usr/sbin/sendmail -t
 ;

Modified: team/moy/webrtc-11/include/asterisk/rtp_engine.h
URL: http://svnview.digium.com/svn/asterisk/team/moy/webrtc-11/include/asterisk/rtp_engine.h?view=diff&rev=409360&r1=409359&r2=409360
==============================================================================
--- team/moy/webrtc-11/include/asterisk/rtp_engine.h (original)
+++ team/moy/webrtc-11/include/asterisk/rtp_engine.h Sun Mar  2 19:02:31 2014
@@ -539,7 +539,9 @@
 	enum ast_rtp_glue_result (*get_trtp_info)(struct ast_channel *chan, struct ast_rtp_instance **instance);
 	/*! Callback for updating the destination that the remote side should send RTP to */
 	int (*update_peer)(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, const struct ast_format_cap *cap, int nat_active);
-	/*! Callback for retrieving codecs that the channel can do.  Result returned in result_cap*/
+	/*! Callback for retrieving codecs that the channel can do.  Result returned in result_cap.
+	 * \note The channel chan will be locked during this call.
+	 */
 	void (*get_codec)(struct ast_channel *chan, struct ast_format_cap *result_cap);
 	/*! Linked list information */
 	AST_RWLIST_ENTRY(ast_rtp_glue) entry;

Modified: team/moy/webrtc-11/main/rtp_engine.c
URL: http://svnview.digium.com/svn/asterisk/team/moy/webrtc-11/main/rtp_engine.c?view=diff&rev=409360&r1=409359&r2=409360
==============================================================================
--- team/moy/webrtc-11/main/rtp_engine.c (original)
+++ team/moy/webrtc-11/main/rtp_engine.c Sun Mar  2 19:02:31 2014
@@ -1224,10 +1224,12 @@
 		if (tinstance1) {
 			ast_rtp_instance_get_remote_address(tinstance1, &tt1);
 		}
-		if (glue1->get_codec) {
+		ast_channel_lock(c1);
+		if (glue1->get_codec && ast_channel_tech_pvt(c1)) {
 			ast_format_cap_remove_all(cap1);
 			glue1->get_codec(c1, cap1);
 		}
+		ast_channel_unlock(c1);
 
 		ast_rtp_instance_get_remote_address(instance0, &t0);
 		if (vinstance0) {
@@ -1236,10 +1238,12 @@
 		if (tinstance0) {
 			ast_rtp_instance_get_remote_address(tinstance0, &tt0);
 		}
-		if (glue0->get_codec) {
+		ast_channel_lock(c0);
+		if (glue0->get_codec && ast_channel_tech_pvt(c0)) {
 			ast_format_cap_remove_all(cap0);
 			glue0->get_codec(c0, cap0);
 		}
+		ast_channel_unlock(c0);
 
 		if ((ast_sockaddr_cmp(&t1, &ac1)) ||
 		    (vinstance1 && ast_sockaddr_cmp(&vt1, &vac1)) ||
@@ -1353,6 +1357,7 @@
 				ast_rtp_instance_get_remote_address(instance1, &t1);
 				ast_sockaddr_copy(&ac1, &t1);
 				/* Update codec information */
+				ast_channel_lock(c0);
 				if (glue0->get_codec && ast_channel_tech_pvt(c0)) {
 					ast_format_cap_remove_all(cap0);
 					ast_format_cap_remove_all(oldcap0);
@@ -1360,12 +1365,15 @@
 					ast_format_cap_append(oldcap0, cap0);
 
 				}
+				ast_channel_unlock(c0);
+				ast_channel_lock(c1);
 				if (glue1->get_codec && ast_channel_tech_pvt(c1)) {
 					ast_format_cap_remove_all(cap1);
 					ast_format_cap_remove_all(oldcap1);
 					glue1->get_codec(c1, cap1);
 					ast_format_cap_append(oldcap1, cap1);
 				}
+				ast_channel_unlock(c1);
 				/* Since UPDATE_BRIDGE_PEER is only used by the bridging code, don't forward it */
 				if (fr->subclass.integer != AST_CONTROL_UPDATE_RTP_PEER) {
 					ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);

Modified: team/moy/webrtc-11/res/res_fax.c
URL: http://svnview.digium.com/svn/asterisk/team/moy/webrtc-11/res/res_fax.c?view=diff&rev=409360&r1=409359&r2=409360
==============================================================================
--- team/moy/webrtc-11/res/res_fax.c (original)
+++ team/moy/webrtc-11/res/res_fax.c Sun Mar  2 19:02:31 2014
@@ -3883,6 +3883,13 @@
 		goto end;
 	}
 
+	if (options.minrate == 2400 && (options.modems & AST_FAX_MODEM_V27) && !(options.modems & (AST_FAX_MODEM_V34))) {
+		ast_fax_modem_to_str(options.modems, modems, sizeof(modems));
+		ast_log(LOG_WARNING, "'modems' setting '%s' is no longer accepted with 'minrate' setting %d\n", modems, options.minrate);
+		ast_log(LOG_WARNING, "'minrate' has been reset to 4800, please update res_fax.conf.\n");
+		options.minrate = 4800;
+	}
+
 	if (check_modem_rate(options.modems, options.minrate)) {
 		ast_fax_modem_to_str(options.modems, modems, sizeof(modems));
 		ast_log(LOG_ERROR, "'modems' setting '%s' is incompatible with 'minrate' setting %d\n", modems, options.minrate);

Modified: team/moy/webrtc-11/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/team/moy/webrtc-11/res/res_rtp_asterisk.c?view=diff&rev=409360&r1=409359&r2=409360
==============================================================================
--- team/moy/webrtc-11/res/res_rtp_asterisk.c (original)
+++ team/moy/webrtc-11/res/res_rtp_asterisk.c Sun Mar  2 19:02:31 2014
@@ -678,7 +678,7 @@
 		return;
 	}
 
-	if (pj_ice_sess_add_cand(rtp->ice, comp_id, transport_id, type, local_pref, &foundation, addr, addr, rel_addr, addr_len, NULL) != PJ_SUCCESS) {
+	if (pj_ice_sess_add_cand(rtp->ice, comp_id, transport_id, type, local_pref, &foundation, addr, base_addr, rel_addr, addr_len, NULL) != PJ_SUCCESS) {
 		ao2_ref(candidate, -1);
 		return;
 	}
@@ -1683,15 +1683,19 @@
 	}
 
 	/* If configured to use a STUN server to get our external mapped address do so */
-	if (stunaddr.sin_addr.s_addr && ast_sockaddr_is_ipv4(addr)) {
+	if (stunaddr.sin_addr.s_addr && ast_sockaddr_is_ipv4(addr) && count) {
 		struct sockaddr_in answer;
 
-		if (!ast_stun_request(rtp->s, &stunaddr, NULL, &answer)) {
+		if (!ast_stun_request(component == AST_RTP_ICE_COMPONENT_RTCP ? rtp->rtcp->s : rtp->s, &stunaddr, NULL, &answer)) {
+			pj_sockaddr base;
 			pj_str_t mapped = pj_str(ast_strdupa(ast_inet_ntoa(answer.sin_addr)));
 
+			/* Use the first local host candidate as the base */
+			pj_sockaddr_cp(&base, &address[0]);
+
 			pj_sockaddr_init(pj_AF_INET(), &address[0], &mapped, ntohs(answer.sin_port));
 
-			ast_rtp_ice_add_cand(rtp, component, transport, PJ_ICE_CAND_TYPE_SRFLX, 65535, &address[0], &address[0],
+			ast_rtp_ice_add_cand(rtp, component, transport, PJ_ICE_CAND_TYPE_SRFLX, 65535, &address[0], &base,
 					     NULL, pj_sockaddr_get_len(&address[0]));
 		}
 	}

Modified: team/moy/webrtc-11/utils/astman.c
URL: http://svnview.digium.com/svn/asterisk/team/moy/webrtc-11/utils/astman.c?view=diff&rev=409360&r1=409359&r2=409360
==============================================================================
--- team/moy/webrtc-11/utils/astman.c (original)
+++ team/moy/webrtc-11/utils/astman.c Sun Mar  2 19:02:31 2014
@@ -737,7 +737,6 @@
 					show_message("Login Failed", get_header(m, "Message"));
 				}
 			} else {
-				memset(m, 0, sizeof(m));
 				manager_action("Login", 
 					"Username: %s\r\n"
 					"Secret: %s\r\n",




More information about the asterisk-commits mailing list