Index: channels/chan_sip.c
===================================================================
--- channels/chan_sip.c	(revision 177657)
+++ channels/chan_sip.c	(working copy)
@@ -1061,6 +1061,7 @@
 	int callevents;			/*!< Whether we send manager events or not */
 	int regextenonqualify;  	/*!< Whether to add/remove regexten when qualifying peers */
 	int matchexterniplocally;	/*!< Match externip/externhost setting against localnet setting */
+  int allow_forking;	/*!< Whether to allow forking or not */	
 	int notifyringing;		/*!< Send notifications on ringing */
 	int notifyhold;			/*!< Send notifications on hold */
 	enum notifycid_setting notifycid; /*!< Send CID with ringing notifications */
@@ -1714,6 +1715,7 @@
 	int hangupcause;			/*!< Storage of hangupcause copied from our owner before we disconnect from the AST channel (only used at hangup) */
 
 	struct sip_subscription_mwi *mwi;       /*!< If this is a subscription MWI dialog, to which subscription */
+  struct sip_peer *peer;                  /*!< associated peer */	
 }; 
 
 
@@ -1839,6 +1841,7 @@
 		AST_STRING_FIELD(mohsuggest);		/*!<  Music on Hold class */
 		AST_STRING_FIELD(parkinglot);		/*!<  Parkinglot */
 		AST_STRING_FIELD(useragent);		/*!<  User agent in SIP request (saved from registration) */
+		AST_STRING_FIELD(callid);		/*!< callid, used to identify registration refreshes */
 		);
 	struct sip_socket socket;	/*!< Socket used for this peer */
 	unsigned int transports:3;      /*!< Transports (enum sip_transport) that are acceptable for this peer */
@@ -1895,7 +1898,7 @@
 	int timer_t1;			/*!<  The maximum T1 value for the peer */
 	int timer_b;			/*!<  The maximum timer B (transaction timeouts) */
 	int deprecated_username; /*!< If it's a realtime peer, are they using the deprecated "username" instead of "defaultuser" */
-	
+  AST_LIST_ENTRY(sip_peer) list;   /* multiple registration entries for same user*/	
 	/*XXX Seems like we suddenly have two flags with the same content. Why? To be continued... */
 	enum sip_peer_type type; /*!< Distinguish between "user" and "peer" types. This is used solely for CLI and manager commands */
 	char onlymatchonip;		/*!< Only match on IP for incoming calls (old type=peer) */
@@ -2331,7 +2334,8 @@
 static void ast_quiet_chan(struct ast_channel *chan);
 static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target);
 static int do_magic_pickup(struct ast_channel *channel, const char *extension, const char *context);
-
+/* add peer info to db */
+static int sip_db_add(struct sip_peer *peer, int expire, char* data, int nodepos);
 /*!
  * \brief generic function for determining if a correct transport is being 
  * used to contact a peer
@@ -2437,18 +2441,18 @@
 static void sip_dump_history(struct sip_pvt *dialog);
 
 /*--- Device object handling */
+static struct sip_peer *sip_allocpeer(const char *name);
 static struct sip_peer *temp_peer(const char *name);
 static struct sip_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime, int ispeer);
 static int update_call_counter(struct sip_pvt *fup, int event);
 static void sip_destroy_peer(struct sip_peer *peer);
 static void sip_destroy_peer_fn(void *peer);
 static void set_peer_defaults(struct sip_peer *peer);
-static struct sip_peer *temp_peer(const char *name);
 static void register_peer_exten(struct sip_peer *peer, int onoff);
 static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime, int forcenamematch, int devstate_only);
 static int sip_poke_peer_s(const void *data);
 static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *p, struct sip_request *req);
-static void reg_source_db(struct sip_peer *peer);
+static void reg_source_db(struct sip_peer *peer, int flag);
 static void destroy_association(struct sip_peer *peer);
 static void set_insecure_flags(struct ast_flags *flags, const char *value, int lineno);
 static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v);
@@ -4484,6 +4488,7 @@
 
 	if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) && !devstate_only) {
 		/* Cache peer */
+    struct sip_peer *tmppeer = peer;
 		ast_copy_flags(&peer->flags[1], &global_flags[1], SIP_PAGE2_RTAUTOCLEAR|SIP_PAGE2_RTCACHEFRIENDS);
 		if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTAUTOCLEAR)) {
 			AST_SCHED_REPLACE_UNREF(peer->expire, sched, sip_cfg.rtautoclear * 1000, expire_register, peer,
@@ -4491,10 +4496,13 @@
 					unref_peer(peer, "remove registration ref"),
 					ref_peer(peer, "add registration ref"));
 		}
-		ao2_t_link(peers, peer, "link peer into peers table");
-		if (peer->addr.sin_addr.s_addr) {
-			ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
-		}
+    while(tmppeer) {
+  		ao2_t_link(peers, tmppeer, "link peer into peers table");
+	  	if (tmppeer->addr.sin_addr.s_addr) {
+		  	ao2_t_link(peers_by_ip, tmppeer, "link peer into peers_by_ip table");
+  		}
+      tmppeer = tmppeer->list.next;
+    }
 	}
 	peer->is_realtime = 1;
 	if (peerlist)
@@ -4837,8 +4845,10 @@
 	dialog->sa.sin_family = AF_INET;
 	dialog->timer_t1 = global_t1; /* Default SIP retransmission timer T1 (RFC 3261) */
 	dialog->timer_b = global_timer_b; /* Default SIP transaction timer B (RFC 3261) */
-	peer = find_peer(peername, NULL, TRUE, FINDALLDEVICES, FALSE);
+	/*peer = find_peer(peername, NULL, TRUE, FINDALLDEVICES, FALSE); */
 
+  peer = dialog->peer;
+  
 	if (peer) {
 		int res;
 		if (newdialog)
@@ -4849,7 +4859,7 @@
 				dialog->sa.sin_port = dialog->recv.sin_port = htons(portno);
 			}
 		}
-		unref_peer(peer, "create_addr: unref peer from find_peer hashtab lookup");
+//    unref_peer(peer, "create_addr: unref peer from find_peer hashtab lookup");
 		return res;
 	}
 
@@ -4945,89 +4955,101 @@
 	struct varshead *headp;
 	struct ast_var_t *current;
 	const char *referer = NULL;   /* SIP referrer */
+  struct sip_pvt *tmpp = NULL;    /*temp var*/
 
 	if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
 		ast_log(LOG_WARNING, "sip_call called on %s, neither down nor reserved\n", ast->name);
 		return -1;
 	}
 
-	/* Check whether there is vxml_url, distinctive ring variables */
-	headp=&ast->varshead;
-	AST_LIST_TRAVERSE(headp, current, entries) {
-		/* Check whether there is a VXML_URL variable */
-		if (!p->options->vxml_url && !strcasecmp(ast_var_name(current), "VXML_URL")) {
-			p->options->vxml_url = ast_var_value(current);
-		} else if (!p->options->uri_options && !strcasecmp(ast_var_name(current), "SIP_URI_OPTIONS")) {
-			p->options->uri_options = ast_var_value(current);
-		} else if (!p->options->addsipheaders && !strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) {
-			/* Check whether there is a variable with a name starting with SIPADDHEADER */
-			p->options->addsipheaders = 1;
-		} else if (!strcasecmp(ast_var_name(current), "SIPFROMDOMAIN")) {
-			ast_string_field_set(p, fromdomain, ast_var_value(current));
-		} else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER")) {
-			/* This is a transfered call */
-			p->options->transfer = 1;
-		} else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REFERER")) {
-			/* This is the referrer */
-			referer = ast_var_value(current);
-		} else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REPLACES")) {
-			/* We're replacing a call. */
-			p->options->replaces = ast_var_value(current);
-		} else if (!strcasecmp(ast_var_name(current), "T38CALL")) {
-			p->t38.state = T38_LOCAL_DIRECT;
-			ast_debug(1, "T38State change to %d on channel %s\n", p->t38.state, ast->name);
-		}
-	}
+  /* If there is a list of sip_pvt, loop thru all of them, we will have to fork*/
+  tmpp = p;
+  if(!p) {
+	  ast_log(LOG_WARNING, "sip_pvt is NULL\n");
+    return -1;
+  }
 
-	res = 0;
-	ast_set_flag(&p->flags[0], SIP_OUTGOING);
+  do {
+  	/* Check whether there is vxml_url, distinctive ring variables */
+  	headp=&ast->varshead;
+    AST_LIST_TRAVERSE(headp, current, entries) {
+    /* Check whether there is a VXML_URL variable */
+		  if (!p->options->vxml_url && !strcasecmp(ast_var_name(current), "VXML_URL")) {
+			  p->options->vxml_url = ast_var_value(current);
+		  } else if (!p->options->uri_options && !strcasecmp(ast_var_name(current), "SIP_URI_OPTIONS")) {
+  			p->options->uri_options = ast_var_value(current);
+	  	} else if (!p->options->addsipheaders && !strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) {
+		  	/* Check whether there is a variable with a name starting with SIPADDHEADER */
+			  p->options->addsipheaders = 1;
+		  } else if (!strcasecmp(ast_var_name(current), "SIPFROMDOMAIN")) {
+  			ast_string_field_set(p, fromdomain, ast_var_value(current));
+	  	} else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER")) {
+  			/* This is a transfered call */
+	  		p->options->transfer = 1;
+  		} else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REFERER")) {
+	  		/* This is the referrer */
+		  	referer = ast_var_value(current);
+		  } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REPLACES")) {
+  			/* We're replacing a call. */
+	  		p->options->replaces = ast_var_value(current);
+  		} else if (!strcasecmp(ast_var_name(current), "T38CALL")) {
+	  		p->t38.state = T38_LOCAL_DIRECT;
+  			ast_debug(1, "T38State change to %d on channel %s\n", p->t38.state, ast->name);
+	  	}
+	  }
 
-	if (p->options->transfer) {
-		char buf[SIPBUFSIZE/2];
+  	res = 0;
+	  ast_set_flag(&p->flags[0], SIP_OUTGOING);
 
-		if (referer) {
-			if (sipdebug)
-				ast_debug(3, "Call for %s transfered by %s\n", p->username, referer);
-			snprintf(buf, sizeof(buf)-1, "-> %s (via %s)", p->cid_name, referer);
-		} else 
-			snprintf(buf, sizeof(buf)-1, "-> %s", p->cid_name);
-		ast_string_field_set(p, cid_name, buf);
-	} 
-	ast_debug(1, "Outgoing Call for %s\n", p->username);
+  	if (p->options->transfer) {
+	  	char buf[SIPBUFSIZE/2];
 
-	res = update_call_counter(p, INC_CALL_RINGING);
+		  if (referer) {
+		    if (sipdebug)
+				 ast_debug(3, "Call for %s transfered by %s\n", p->username, referer);
+			  snprintf(buf, sizeof(buf)-1, "-> %s (via %s)", p->cid_name, referer);
+		  } else 
+			  snprintf(buf, sizeof(buf)-1, "-> %s", p->cid_name);
+		  ast_string_field_set(p, cid_name, buf);
+	  } 
+	  ast_debug(1, "Outgoing Call for %s\n", p->username);
 
-	if (res == -1) {
-		return res;
-	} else {
-		ast->hangupcause = AST_CAUSE_USER_BUSY;
-	}
+	  res = update_call_counter(p, INC_CALL_RINGING);
 
-	p->callingpres = ast->cid.cid_pres;
-	p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec);
-	p->jointnoncodeccapability = p->noncodeccapability;
+	  if (res == -1) {
+  		return res;
+  	} else {
+		  ast->hangupcause = AST_CAUSE_USER_BUSY;
+	  }
 
-	/* If there are no audio formats left to offer, punt */
-	if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
-		ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username);
-		res = -1;
-	} else {
-		int xmitres;
+	  p->callingpres = ast->cid.cid_pres;
+  	p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec);
+  	p->jointnoncodeccapability = p->noncodeccapability;
 
-		p->t38.jointcapability = p->t38.capability;
-		ast_debug(2, "Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
+  	/* If there are no audio formats left to offer, punt */
+  	if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
+  		ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username);
+  		res = -1;
+  	} else {
+  		int xmitres;
 
-		xmitres = transmit_invite(p, SIP_INVITE, 1, 2);
-		if (xmitres == XMIT_ERROR)
-			return -1;
-		p->invitestate = INV_CALLING;
+	  	p->t38.jointcapability = p->t38.capability;
+	  	ast_debug(2, "Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
+
+  		xmitres = transmit_invite(p, SIP_INVITE, 1, 2);
+  		if (xmitres == XMIT_ERROR)
+  			return -1;
+	  	p->invitestate = INV_CALLING;
 	
-		/* Initialize auto-congest time */
-		AST_SCHED_REPLACE_UNREF(p->initid, sched, p->timer_b, auto_congest, p, 
-								dialog_unref(_data, "dialog ptr dec when SCHED_REPLACE del op succeeded"), 
-								dialog_unref(p, "dialog ptr dec when SCHED_REPLACE add failed"),
-								dialog_ref(p, "dialog ptr inc when SCHED_REPLACE add succeeded") );
-	}
+		  /* Initialize auto-congest time */
+	  	AST_SCHED_REPLACE_UNREF(p->initid, sched, p->timer_b, auto_congest, p, 
+		  						dialog_unref(_data, "dialog ptr dec when SCHED_REPLACE del op succeeded"), 
+			  					dialog_unref(p, "dialog ptr dec when SCHED_REPLACE add failed"),
+				  				dialog_ref(p, "dialog ptr inc when SCHED_REPLACE add succeeded") );
+	  }
+    tmpp = tmpp->next;
+    p = tmpp;
+  } while (tmpp);
 	return res;
 }
 
@@ -5087,12 +5109,15 @@
 	if (p->owner) {
 		if (lockowner)
 			ast_channel_lock(p->owner);
-		if (option_debug)
-			ast_log(LOG_DEBUG, "Detaching from %s\n", p->owner->name);
-		p->owner->tech_pvt = NULL;
-		/* Make sure that the channel knows its backend is going away */
-		p->owner->_softhangup |= AST_SOFTHANGUP_DEV;
-		if (lockowner)
+    if(p->owner->no_of_pvt_channels < 1)
+    {
+		  if (option_debug)
+			  ast_log(LOG_DEBUG, "Detaching from %s\n", p->owner->name);
+  		p->owner->tech_pvt = NULL;
+	  	/* Make sure that the channel knows its backend is going away */
+		  p->owner->_softhangup |= AST_SOFTHANGUP_DEV;
+    }
+	  if (lockowner)
 			ast_channel_unlock(p->owner);
 		/* Give the channel a chance to react before deallocation */
 		usleep(1);
@@ -5529,188 +5554,202 @@
 	int needcancel = FALSE;
 	int needdestroy = 0;
 	struct ast_channel *oldowner = ast;
+  int hangup = 0; 
 
-	if (!p) {
-		ast_debug(1, "Asked to hangup channel that was not connected\n");
-		return 0;
-	}
-	if (ast_test_flag(ast, AST_FLAG_ANSWERED_ELSEWHERE) || ast->hangupcause == AST_CAUSE_ANSWERED_ELSEWHERE) {
-		ast_debug(1, "This call was answered elsewhere");
-		if (ast->hangupcause == AST_CAUSE_ANSWERED_ELSEWHERE) {
-			ast_debug(1, "####### It's the cause code, buddy. The cause code!!!\n");
-		}
-		append_history(p, "Cancel", "Call answered elsewhere");
-		p->answered_elsewhere = TRUE;
-	}
+  /* if there are multiple calls in progress, due to forking, loop thru all and disconnect */
+  do {
+	  if (!p) {
+		  ast_debug(1, "Asked to hangup channel that was not connected\n");
+  		return 0;
+	  }
+    
+	  if (p->owner) {
+      /* decrement no of pvt channels under owner channel */
+      p->owner->no_of_pvt_channels--;
+      if(p->owner->no_of_pvt_channels < 1)
+        hangup = 1;                       /* if this is the last sip_pvt under owner, you can finally hangup */
+    }
+	  
+  	if (ast_test_flag(ast, AST_FLAG_ANSWERED_ELSEWHERE) || ast->hangupcause == AST_CAUSE_ANSWERED_ELSEWHERE) {
+	  	ast_debug(1, "This call was answered elsewhere");
+		  if (ast->hangupcause == AST_CAUSE_ANSWERED_ELSEWHERE) {
+			  ast_debug(1, "####### It's the cause code, buddy. The cause code!!!\n");
+		  }
+	  	append_history(p, "Cancel", "Call answered elsewhere");
+  		p->answered_elsewhere = TRUE;
+  	}
 
-	/* Store hangupcause locally in PVT so we still have it before disconnect */
-	if (p->owner)
-		p->hangupcause = p->owner->hangupcause;
+  	/* Store hangupcause locally in PVT so we still have it before disconnect */
+	  if (p->owner)
+		  p->hangupcause = p->owner->hangupcause;
 
-	if (ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
-		if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
-			if (sipdebug)
-				ast_debug(1, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
-			update_call_counter(p, DEC_CALL_LIMIT);
-		}
-		ast_debug(4, "SIP Transfer: Not hanging up right now... Rescheduling hangup for %s.\n", p->callid);
-		if (p->autokillid > -1 && sip_cancel_destroy(p))
-			ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
-		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
-		ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER);	/* Really hang up next time */
-		p->needdestroy = 0;
-		p->owner->tech_pvt = dialog_unref(p->owner->tech_pvt, "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);
-		return 0;
-	}
+  	if (ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
+	  	if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
+		  	if (sipdebug)
+			  	ast_debug(1, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
+			  update_call_counter(p, DEC_CALL_LIMIT);
+	  	}
+		  ast_debug(4, "SIP Transfer: Not hanging up right now... Rescheduling hangup for %s.\n", p->callid);
+		  if (p->autokillid > -1 && sip_cancel_destroy(p))
+	  		ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
+	  	sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+		  ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER);	/* Really hang up next time */
+		  p->needdestroy = 0;
+		  p->owner->tech_pvt = dialog_unref(p->owner->tech_pvt, "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);
+		  return 0;
+	  }
 
-	if (ast_test_flag(ast, AST_FLAG_ZOMBIE)) {
-		if (p->refer)
-			ast_debug(1, "SIP Transfer: Hanging up Zombie channel %s after transfer ... Call-ID: %s\n", ast->name, p->callid);
-		else
-			ast_debug(1, "Hanging up zombie call. Be scared.\n");
-	} else
-		ast_debug(1, "Hangup call %s, SIP callid %s\n", ast->name, p->callid);
+  	if (ast_test_flag(ast, AST_FLAG_ZOMBIE)) {
+	  	if (p->refer)
+		  	ast_debug(1, "SIP Transfer: Hanging up Zombie channel %s after transfer ... Call-ID: %s\n", ast->name, p->callid);
+  		else
+	  		ast_debug(1, "Hanging up zombie call. Be scared.\n");
+	  } else
+  		ast_debug(1, "Hangup call %s, SIP callid %s\n", ast->name, p->callid);
 
-	sip_pvt_lock(p);
-	if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
-		if (sipdebug)
-			ast_debug(1, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
-		update_call_counter(p, DEC_CALL_LIMIT);
-	}
+	  sip_pvt_lock(p);
+	  if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
+		  if (sipdebug)
+			  ast_debug(1, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
+		  update_call_counter(p, DEC_CALL_LIMIT);
+	  }
 
-	/* Determine how to disconnect */
-	if (p->owner != ast) {
-		ast_log(LOG_WARNING, "Huh?  We aren't the owner? Can't hangup call.\n");
-		sip_pvt_unlock(p);
-		return 0;
-	}
-	/* If the call is not UP, we need to send CANCEL instead of BYE */
-	/* In case of re-invites, the call might be UP even though we have an incomplete invite transaction */
-	if (p->invitestate < INV_COMPLETED && p->owner->_state != AST_STATE_UP) {
-		needcancel = TRUE;
-		ast_debug(4, "Hanging up channel in state %s (not UP)\n", ast_state2str(ast->_state));
-	}
+  	/* Determine how to disconnect */
+	  if (p->owner != ast) {
+  		ast_log(LOG_WARNING, "Huh?  We aren't the owner? Can't hangup call.\n");
+		  sip_pvt_unlock(p);
+		  return 0;
+	  }
+	  /* If the call is not UP, we need to send CANCEL instead of BYE */
+	  /* In case of re-invites, the call might be UP even though we have an incomplete invite transaction */
+	  if (p->invitestate < INV_COMPLETED && p->owner->_state != AST_STATE_UP) {
+  		needcancel = TRUE;
+		  ast_debug(4, "Hanging up channel in state %s (not UP)\n", ast_state2str(ast->_state));
+	  }
 
-	stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
+	  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", p->owner ? ast_cause2str(p->hangupcause) : "Unknown");
 
-	/* Disconnect */
-	if (p->vad)
-		ast_dsp_free(p->vad);
+	  /* Disconnect */
+  	if (p->vad)
+	  	ast_dsp_free(p->vad);
 
-	p->owner = NULL;
-	ast->tech_pvt = dialog_unref(ast->tech_pvt, "unref ast->tech_pvt");
+  	p->owner = NULL;
+	  ast->tech_pvt = dialog_unref(ast->tech_pvt, "unref ast->tech_pvt");
 
-	ast_module_unref(ast_module_info->self);
-	/* Do not destroy this pvt until we have timeout or
-	   get an answer to the BYE or INVITE/CANCEL 
-	   If we get no answer during retransmit period, drop the call anyway.
-	   (Sorry, mother-in-law, you can't deny a hangup by sending
-	   603 declined to BYE...)
-	*/
-	if (p->alreadygone)
-		needdestroy = 1;	/* Set destroy flag at end of this function */
-	else if (p->invitestate != INV_CALLING)
-		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+	  ast_module_unref(ast_module_info->self);
+  	/* Do not destroy this pvt until we have timeout or
+	     get an answer to the BYE or INVITE/CANCEL 
+	     If we get no answer during retransmit period, drop the call anyway.
+	     (Sorry, mother-in-law, you can't deny a hangup by sending
+	     603 declined to BYE...)
+	  */
+	  if (p->alreadygone)
+		  needdestroy = 1;	/* Set destroy flag at end of this function */
+  	else if (p->invitestate != INV_CALLING)
+	  	sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 
-	/* Start the process if it's not already started */
-	if (!p->alreadygone && p->initreq.data && !ast_strlen_zero(p->initreq.data->str)) {
-		if (needcancel) {	/* Outgoing call, not up */
-			if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
-				/* stop retransmitting an INVITE that has not received a response */
-				__sip_pretend_ack(p);
+	  /* Start the process if it's not already started */
+	  if (!p->alreadygone && p->initreq.data && !ast_strlen_zero(p->initreq.data->str)) {
+		  if (needcancel) {	/* Outgoing call, not up */
+			  if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+				  /* stop retransmitting an INVITE that has not received a response */
+  				__sip_pretend_ack(p);
 
-				/* if we can't send right now, mark it pending */
-				if (p->invitestate == INV_CALLING) {
-					/* We can't send anything in CALLING state */
-					ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
-					/* Do we need a timer here if we don't hear from them at all? */
-					sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
-					append_history(p, "DELAY", "Not sending cancel, waiting for timeout");
-				} else {
-					p->invitestate = INV_CANCELLED;
-					/* Send a new request: CANCEL */
-					transmit_request(p, SIP_CANCEL, p->lastinvite, XMIT_RELIABLE, FALSE);
-					/* Actually don't destroy us yet, wait for the 487 on our original 
-					   INVITE, but do set an autodestruct just in case we never get it. */
-					needdestroy = 0;
-					sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
-				}
-				if ( p->initid != -1 ) {
-					/* channel still up - reverse dec of inUse counter
-					   only if the channel is not auto-congested */
-					update_call_counter(p, INC_CALL_LIMIT);
-				}
-			} else {	/* Incoming call, not up */
-				const char *res;
-				if (p->hangupcause && (res = hangup_cause2sip(p->hangupcause)))
-					transmit_response_reliable(p, res, &p->initreq);
-				else 
-					transmit_response_reliable(p, "603 Declined", &p->initreq);
-				p->invitestate = INV_TERMINATED;
-			}
-		} else {	/* Call is in UP state, send BYE */
-			if (p->stimer->st_active == TRUE) {
-				stop_session_timer(p);
-			}
+	  			/* if we can't send right now, mark it pending */
+		  		if (p->invitestate == INV_CALLING) {
+			  		/* We can't send anything in CALLING state */
+				  	ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
+  					/* Do we need a timer here if we don't hear from them at all? */
+	  				sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+		  			append_history(p, "DELAY", "Not sending cancel, waiting for timeout");
+			  	} else {
+				  	p->invitestate = INV_CANCELLED;
+					  /* Send a new request: CANCEL */
+  					transmit_request(p, SIP_CANCEL, p->lastinvite, XMIT_RELIABLE, FALSE);
+	  				/* Actually don't destroy us yet, wait for the 487 on our original 
+		  			   INVITE, but do set an autodestruct just in case we never get it. */
+			  		needdestroy = 0;
+  					sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+	  			}
+		  		if ( p->initid != -1 ) {
+			  		/* channel still up - reverse dec of inUse counter
+				  	   only if the channel is not auto-congested */
+  					update_call_counter(p, INC_CALL_LIMIT);
+	  			}
+		  	} else {	/* Incoming call, not up */
+			  	const char *res;
+				  if (p->hangupcause && (res = hangup_cause2sip(p->hangupcause)))
+					  transmit_response_reliable(p, res, &p->initreq);
+  				else 
+	  				transmit_response_reliable(p, "603 Declined", &p->initreq);
+		  		p->invitestate = INV_TERMINATED;
+			  }
+  		} else {	/* Call is in UP state, send BYE */
+	  		if (p->stimer->st_active == TRUE) {
+		  		stop_session_timer(p);
+  			}
 
-			if (!p->pendinginvite) {
-				struct ast_channel *bridge = ast_bridged_channel(oldowner);
-				char *audioqos = "";
-				char *videoqos = "";
-				char *textqos = "";
+	  		if (!p->pendinginvite) {
+		  		struct ast_channel *bridge = ast_bridged_channel(oldowner);
+			  	char *audioqos = "";
+  				char *videoqos = "";
+	  			char *textqos = "";
 
-				if (p->rtp)
-					ast_rtp_set_vars(oldowner, p->rtp);
+  				if (p->rtp)
+	  				ast_rtp_set_vars(oldowner, p->rtp);
 
-				if (bridge) {
-					struct sip_pvt *q = bridge->tech_pvt;
+		  		if (bridge) {
+			  		struct sip_pvt *q = bridge->tech_pvt;
 
-					if (IS_SIP_TECH(bridge->tech) && q)
-						ast_rtp_set_vars(bridge, q->rtp);
-				}
+  					if (IS_SIP_TECH(bridge->tech) && q)
+	  					ast_rtp_set_vars(bridge, q->rtp);
+		  		}
 
-				if (p->vrtp)
-					videoqos = ast_rtp_get_quality(p->vrtp, NULL, RTPQOS_SUMMARY);
-				if (p->trtp)
-					textqos = ast_rtp_get_quality(p->trtp, NULL, RTPQOS_SUMMARY);
-				/* Send a hangup */
-				transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
+  				if (p->vrtp)
+	  				videoqos = ast_rtp_get_quality(p->vrtp, NULL, RTPQOS_SUMMARY);
+		  		if (p->trtp)
+			  		textqos = ast_rtp_get_quality(p->trtp, NULL, RTPQOS_SUMMARY);
+  				/* Send a hangup */
+	  			transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
 
-				/* Get RTCP quality before end of call */
-				if (p->do_history) {
-					if (p->rtp)
-						append_history(p, "RTCPaudio", "Quality:%s", audioqos);
-					if (p->vrtp)
-						append_history(p, "RTCPvideo", "Quality:%s", videoqos);
-					if (p->trtp)
-						append_history(p, "RTCPtext", "Quality:%s", textqos);
-				}
-				if (p->rtp && oldowner)
-					pbx_builtin_setvar_helper(oldowner, "RTPAUDIOQOS", audioqos);
-				if (p->vrtp && oldowner)
-					pbx_builtin_setvar_helper(oldowner, "RTPVIDEOQOS", videoqos);
-				if (p->trtp && oldowner)
-					pbx_builtin_setvar_helper(oldowner, "RTPTEXTQOS", textqos);
-			} else {
-				/* Note we will need a BYE when this all settles out
-				   but we can't send one while we have "INVITE" outstanding. */
-				ast_set_flag(&p->flags[0], SIP_PENDINGBYE);	
-				ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);	
-				AST_SCHED_DEL_UNREF(sched, p->waitid, dialog_unref(p, "when you delete the waitid sched, you should dec the refcount for the stored dialog ptr"));
-				if (sip_cancel_destroy(p))
-					ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
-			}
-		}
-	}
-	if (needdestroy) {
-		pvt_set_needdestroy(p, "hangup");
-	}
-	sip_pvt_unlock(p);
+		  		/* Get RTCP quality before end of call */
+			  	if (p->do_history) {
+				  	if (p->rtp)
+					  	append_history(p, "RTCPaudio", "Quality:%s", audioqos);
+  					if (p->vrtp)
+	  					append_history(p, "RTCPvideo", "Quality:%s", videoqos);
+		  			if (p->trtp)
+			  			append_history(p, "RTCPtext", "Quality:%s", textqos);
+				  }
+  				if (p->rtp && oldowner)
+	  				pbx_builtin_setvar_helper(oldowner, "RTPAUDIOQOS", audioqos);
+		  		if (p->vrtp && oldowner)
+			  		pbx_builtin_setvar_helper(oldowner, "RTPVIDEOQOS", videoqos);
+				  if (p->trtp && oldowner)
+  					pbx_builtin_setvar_helper(oldowner, "RTPTEXTQOS", textqos);
+	  		} else {
+		  		/* Note we will need a BYE when this all settles out
+			  	   but we can't send one while we have "INVITE" outstanding. */
+  				ast_set_flag(&p->flags[0], SIP_PENDINGBYE);	
+	  			ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);	
+		  		AST_SCHED_DEL_UNREF(sched, p->waitid, dialog_unref(p, "when you delete the waitid sched, you should dec the refcount for the stored dialog ptr"));
+			  	if (sip_cancel_destroy(p))
+				  	ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
+  			}
+	  	}
+	  }
+  	if (needdestroy) {
+	  	pvt_set_needdestroy(p, "hangup");
+	  }
+  	sip_pvt_unlock(p);
+    p = p->next;
+  } while (p);
+  	
 	return 0;
 }
 
@@ -11104,14 +11143,67 @@
 {
 	int realtimeregs = ast_check_realtime("sipregs");
 	char *tablename = (realtimeregs) ? "sipregs" : "sippeers";
+  struct sip_peer *tmppeer = NULL;
+  struct sip_peer *prevpeer = NULL;
+  struct sip_peer *headpeer = NULL;
+  int flag = 0, count = 0;
+  char tmpdata[SIPBUFSIZE] = {0};
+  char dbdata[SIPBUFSIZE] = {0};
+  char *tmpstr = NULL;
+  char *scan = NULL;
 
-	if (!sip_cfg.ignore_regexpire) {
-		if (peer->rt_fromcontact) {
-			ast_update_realtime(tablename, "name", peer->name, "fullcontact", "", "ipaddr", "", "port", "", "regseconds", "0", peer->deprecated_username ? "username" : "defaultuser", "", "regserver", "", "useragent", "", SENTINEL);
-			ast_update_realtime("sippeers", "name", peer->name, "lastms", "", NULL);
-		} else {
-			ast_db_del("SIP/Registry", peer->name);
-		}
+  headpeer = find_peer(peer->name, NULL, TRUE, FINDALLDEVICES, FALSE);
+  if(!headpeer) {
+    ast_log(LOG_NOTICE, "peer not found, this is an error\n");
+    return;
+  }
+  else
+    count = ast_strlen_zero(headpeer->fullcontact);
+ 
+  tmppeer = prevpeer = headpeer;
+
+  if(ast_db_get("SIP/Registry", peer->name, dbdata, sizeof(dbdata)))
+    ast_log(LOG_WARNING, "Failed to open database\n");
+
+  scan  = dbdata;
+
+  /* copy the entries in dbdata except the one for which is being destroyed */
+  while(tmppeer) {
+    if(tmppeer != peer) {
+      if(!count) {
+        tmpstr = strsep(&scan, ",");
+        strlcat(tmpdata, tmpstr, sizeof(tmpdata));
+        strlcat(tmpdata, ",", sizeof(tmpdata));
+      }
+      else
+        count = 0;
+    }
+    else {
+      if(!count)
+        tmpstr = strsep(&scan, ",");
+      else
+        count = 0;
+      prevpeer->list.next = tmppeer->list.next; /*happens only once */
+    }
+
+    prevpeer = tmppeer;
+    tmppeer = tmppeer->list.next;
+  }
+  unref_peer(headpeer, "unref peer");
+
+  flag = ast_strlen_zero(tmpdata);
+	
+  if(!flag) {
+	  ast_db_del("SIP/Registry", peer->name);             /*delete old entry */
+    ast_db_put("SIP/Registry", peer->name, tmpdata);    /*update with new one */
+  }
+  else {
+    if (!sip_cfg.ignore_regexpire) {
+		  if (peer->rt_fromcontact)
+			  ast_update_realtime(tablename, "name", peer->name, "fullcontact", "", "ipaddr", "", "port", "", "regseconds", "0", peer->deprecated_username ? "username" : "defaultuser", "", "regserver", "", "useragent", "", SENTINEL);
+  		else 
+	  		ast_db_del("SIP/Registry", peer->name);
+    }
 	}
 }
 
@@ -11119,19 +11211,25 @@
 static int expire_register(const void *data)
 {
 	struct sip_peer *peer = (struct sip_peer *)data;
-	
+	struct sip_peer *headpeer = NULL;
+  
 	if (!peer)		/* Hmmm. We have no peer. Weird. */
 		return 0;
 
+  headpeer = find_peer(peer->name, NULL, TRUE, FINDALLDEVICES, FALSE);
+ 
 	peer->expire = -1;
 	memset(&peer->addr, 0, sizeof(peer->addr));
+ 
+  destroy_association(peer);	/* remove registration data from storage */
 
-	destroy_association(peer);	/* remove registration data from storage */
-	
-	manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unregistered\r\nCause: Expired\r\n", peer->name);
-	register_peer_exten(peer, FALSE);	/* Remove regexten */
-	ast_devstate_changed(AST_DEVICE_UNKNOWN, "SIP/%s", peer->name);
+  if(headpeer == peer && !peer->list.next) {   /*unregister only if there is only one user registered as this peer */
+  	manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unregistered\r\nCause: Expired\r\n", peer->name);
+  	register_peer_exten(peer, FALSE);	/* Remove regexten */
+  	ast_devstate_changed(AST_DEVICE_UNKNOWN, "SIP/%s", peer->name);
+  }
 
+  unref_peer(headpeer, "unref tmppeer");
 	/* Do we need to release this peer from memory? 
 		Only for realtime peers and autocreated peers
 	*/
@@ -11166,65 +11264,79 @@
 }
 
 /*! \brief Get registration details from Asterisk DB */
-static void reg_source_db(struct sip_peer *peer)
+static void reg_source_db(struct sip_peer *peer, int flag)
 {
 	char data[256];
 	struct in_addr in;
 	int expire;
 	int port;
 	char *scan, *addr, *port_str, *expiry_str, *username, *contact;
+  struct sip_peer* headpeer = peer;
 
 	if (peer->rt_fromcontact) 
 		return;
 	if (ast_db_get("SIP/Registry", peer->name, data, sizeof(data)))
 		return;
 
+  /* there can be multiple comma seperated entries for the same peer, put them as a list*/
 	scan = data;
-	addr = strsep(&scan, ":");
-	port_str = strsep(&scan, ":");
-	expiry_str = strsep(&scan, ":");
-	username = strsep(&scan, ":");
-	contact = scan;	/* Contact include sip: and has to be the last part of the database entry as long as we use : as a separator */
+  while(!ast_strlen_zero(scan)) {
+  	addr = strsep(&scan, ":");
+	  port_str = strsep(&scan, ":");
+  	expiry_str = strsep(&scan, ":");
+	  username = strsep(&scan, ":");
+  	contact = strsep(&scan, ",");
 
-	if (!inet_aton(addr, &in))
-		return;
+  	if (!inet_aton(addr, &in))
+	  	return;
 
-	if (port_str)
-		port = atoi(port_str);
-	else
-		return;
+  	if (port_str)
+	  	port = atoi(port_str);
+  	else
+	  	return;
 
-	if (expiry_str)
-		expire = atoi(expiry_str);
-	else
-		return;
+  	if (expiry_str)
+	  	expire = atoi(expiry_str);
+  	else
+	  	return;
 
-	if (username)
-		ast_string_field_set(peer, username, username);
-	if (contact)
-		ast_string_field_set(peer, fullcontact, contact);
+  	if (username)
+	  	ast_string_field_set(peer, username, username);
+  	if (contact)
+	  	ast_string_field_set(peer, fullcontact, contact);
 
-	ast_debug(2, "SIP Seeding peer from astdb: '%s' at %s@%s:%d for %d\n",
-	    peer->name, peer->username, ast_inet_ntoa(in), port, expire);
+  	ast_debug(2, "SIP Seeding peer from astdb: '%s' at %s@%s:%d for %d\n",
+	      peer->name, peer->username, ast_inet_ntoa(in), port, expire);
 
-	memset(&peer->addr, 0, sizeof(peer->addr));
-	peer->addr.sin_family = AF_INET;
-	peer->addr.sin_addr = in;
-	peer->addr.sin_port = htons(port);
-	if (sipsock < 0) {
-		/* SIP isn't up yet, so schedule a poke only, pretty soon */
-		AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched, ast_random() % 5000 + 1, sip_poke_peer_s, peer,
-				unref_peer(_data, "removing poke peer ref"),
-				unref_peer(peer, "removing poke peer ref"),
-				ref_peer(peer, "adding poke peer ref"));
-	} else {
-		sip_poke_peer(peer, 0);
-	}
-	AST_SCHED_REPLACE_UNREF(peer->expire, sched, (expire + 10) * 1000, expire_register, peer,
-			unref_peer(_data, "remove registration ref"),
-			unref_peer(peer, "remove registration ref"),
-			ref_peer(peer, "add registration ref"));
-	register_peer_exten(peer, TRUE);
+  	memset(&peer->addr, 0, sizeof(peer->addr));
+  	peer->addr.sin_family = AF_INET;
+  	peer->addr.sin_addr = in;
+  	peer->addr.sin_port = htons(port);
+  
+    AST_SCHED_REPLACE_UNREF(peer->expire, sched, (expire + 10) * 1000, expire_register, peer,
+	  		unref_peer(_data, "remove registration ref"),
+		  	unref_peer(peer, "remove registration ref"),
+			  ref_peer(peer, "add registration ref"));
+  
+    if(peer->list.next || flag)
+      break;
+    else if(!ast_strlen_zero(scan) && !flag) {     /*not called from temp_peer */
+      peer->list.next = sip_allocpeer(peer->name); /*if there are multiple entries, make a list */
+      peer = peer->list.next;
+    }
+  }
+
+  if (sipsock < 0) {
+	 	/* SIP isn't up yet, so schedule a poke only, pretty soon */
+	  AST_SCHED_REPLACE_UNREF(headpeer->pokeexpire, sched, ast_random() % 5000 + 1, sip_poke_peer_s, headpeer,
+		  	unref_peer(_data, "removing poke peer ref"),
+  			unref_peer(headpeer, "removing poke peer ref"),
+	 			ref_peer(headpeer, "adding poke peer ref"));
+  } else {
+	 	sip_poke_peer(headpeer, 0);
+  }
+  if(!flag) 
+    register_peer_exten(headpeer, TRUE);
 }
 
 /*! \brief Save contact header for 200 OK on INVITE */
@@ -11336,7 +11448,11 @@
 	struct hostent *hp;
 	struct ast_hostent ahp;
 	struct sockaddr_in oldsin, testsin;
-
+  struct sip_peer *prevpeer = NULL;
+  struct sip_peer *nextpeer = NULL;
+  struct sip_peer *tmppeer = NULL;
+  int nodepos = -1;
+    
 	ast_copy_string(contact, get_header(req, "Contact"), sizeof(contact));
 
 	if (ast_strlen_zero(expires)) {	/* No expires header, try look in Contact: */
@@ -11373,29 +11489,124 @@
 		return PARSE_REGISTER_QUERY;
 	} else if (!strcasecmp(curi, "*") || !expire) {	/* Unregister this peer */
 		/* This means remove all registrations and return OK */
-		memset(&peer->addr, 0, sizeof(peer->addr));
+    int removeall = 0;
+    int freepeer = 0;
 
-		AST_SCHED_DEL_UNREF(sched, peer->expire,
-				unref_peer(peer, "remove register expire ref"));
+    if(!strcasecmp(curi, "*"))
+      removeall = 1;
+      
+    tmppeer = prevpeer = peer;
+    do {
+      if(!removeall) {
+        while(tmppeer) {
+          /* if callid matches or the contact matches, this is the required peer, update*/
+          if(!strcmp(tmppeer->callid, pvt->callid) || (!ast_strlen_zero(curi) && !strcmp(tmppeer->fullcontact, curi))) {
+            /*TODO: if peer is not the head node, not freeing head node because of some problems faced
+             * will have to free head node, and make the next in list as head*/
+            if(tmppeer != peer)
+              freepeer = 1;   
+            break;
+          }
+          prevpeer = tmppeer;
+          tmppeer = tmppeer->list.next;
+        }
+        if(!tmppeer) {
+          ast_log(LOG_NOTICE, "received unregister request for the peer not registered\n");
+          return PARSE_REGISTER_UPDATE;
+        }
+      }
+  
+  		memset(&tmppeer->addr, 0, sizeof(tmppeer->addr));
 
-		destroy_association(peer);
-		
-		register_peer_exten(peer, FALSE);	/* Remove extension from regexten= setting in sip.conf */
-		ast_string_field_set(peer, fullcontact, "");
-		ast_string_field_set(peer, useragent, "");
-		peer->sipoptions = 0;
-		peer->lastms = 0;
-		pvt->expiry = 0;
+      AST_SCHED_DEL_UNREF(sched, tmppeer->expire,
+		     	unref_peer(tmppeer, "remove register expire ref"));
 
-		ast_verb(3, "Unregistered SIP '%s'\n", peer->name);
+    	destroy_association(tmppeer);
 
-		manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unregistered\r\n", peer->name);
-		return PARSE_REGISTER_UPDATE;
+      if((!peer->list.next && freepeer != 1) || (!peer->list.next && ast_strlen_zero(peer->fullcontact))) {
+        register_peer_exten(tmppeer, FALSE);	/* Remove extension from regexten= setting in sip.conf */
+      }
+
+  		ast_string_field_set(tmppeer, fullcontact, "");
+	  	ast_string_field_set(tmppeer, useragent, "");
+  		tmppeer->sipoptions = 0;
+	  	tmppeer->lastms = 0;
+  		pvt->expiry = 0;
+
+	  	ast_verb(3, "Unregistered SIP '%s'\n", tmppeer->name);
+
+  		manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unregistered\r\n", tmppeer->name);
+     /* if * received in contact, remove all the nodes in the list */
+      if(removeall) {
+        if(tmppeer != peer) {
+          prevpeer->list.next = tmppeer->list.next;
+          ao2_t_unlink(peers, tmppeer, "ao2_unlink of peer from peer table");
+          ao2_t_unlink(peers_by_ip, tmppeer, "ao2_unlink of peer from peers_by_ip table");
+          ast_free(tmppeer);
+        }
+        tmppeer = prevpeer->list.next;
+      }
+
+      if(!tmppeer)
+        break;
+    } while(removeall);
+
+    if(freepeer) {
+      ao2_t_unlink(peers, tmppeer, "ao2_unlink of peer from peer table");
+      ao2_t_unlink(peers_by_ip, tmppeer, "ao2_unlink of peer from peers_by_ip table");
+      ast_free(tmppeer);
+    }
+    return PARSE_REGISTER_UPDATE;
 	}
-
+  
 	/* Store whatever we got as a contact from the client */
-	ast_string_field_set(peer, fullcontact, curi);
+  do {
+    tmppeer = prevpeer = peer;
 
+    if(ast_strlen_zero(peer->fullcontact))
+      nodepos--;
+
+    /* Registration refreshes */
+    while(tmppeer) {
+      nodepos++;
+      /* if allowforking=no or matches with callid or contact */
+      if(!sip_cfg.allow_forking || !strcmp(pvt->callid, tmppeer->callid) || (!ast_strlen_zero(curi) && !strcmp(tmppeer->fullcontact, curi)))
+      {
+        nextpeer = tmppeer;
+        goto contactset;
+      }
+      
+      prevpeer = tmppeer;
+      tmppeer = tmppeer->list.next;
+    }
+
+    nodepos = -1;
+
+    /*if first peer contact is null, we have to fill it first */
+    if(ast_strlen_zero(peer->fullcontact)) {
+      nextpeer = peer;
+      goto contactset;
+    }
+
+    nextpeer = temp_peer(peer->name);
+    
+    if (nextpeer) {
+      ao2_t_link(peers, nextpeer, "link peer into peer table");
+      if (nextpeer->addr.sin_addr.s_addr) {
+        ao2_t_link(peers_by_ip, nextpeer, "link peer into peers-by-ip table");
+      }
+    }
+    else
+      return PARSE_REGISTER_FAILED;
+  
+    prevpeer->list.next = nextpeer;
+  } while (0);
+
+contactset:  
+  ast_string_field_set(nextpeer, fullcontact, curi);
+  ast_string_field_set(nextpeer, callid, pvt->callid);
+
+  peer = nextpeer;
 	/* For the 200 OK, we should use the received contact */
 	ast_string_field_build(pvt, our_contact, "<%s>", curi);
 
@@ -11482,7 +11693,9 @@
 		}
 	}
 	pvt->expiry = expire;
-	snprintf(data, sizeof(data), "%s:%d:%d:%s:%s", ast_inet_ntoa(peer->addr.sin_addr), ntohs(peer->addr.sin_port), expire, peer->username, peer->fullcontact);
+
+  /* add peer info to db */
+  sip_db_add(peer, expire, data, nodepos);
 	/* Saving TCP connections is useless, we won't be able to reconnect 
 		XXX WHY???? XXX
 		\todo Fix this immediately.
@@ -15263,25 +15476,25 @@
 /*! \brief Do completion on registered peer name */
 static char *complete_sip_registered_peer(const char *word, int state, int flags2)
 {
-       char *result = NULL;
-       int wordlen = strlen(word);
-       int which = 0;
-	   struct ao2_iterator i;
-	   struct sip_peer *peer;
+  char *result = NULL;
+  int wordlen = strlen(word);
+  int which = 0;
+	struct ao2_iterator i;
+	struct sip_peer *peer;
 
-	   i = ao2_iterator_init(peers, 0);
-	   while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
-		   if (!strncasecmp(word, peer->name, wordlen) &&
-			   (!flags2 || ast_test_flag(&peer->flags[1], flags2)) &&
+	i = ao2_iterator_init(peers, 0);
+	while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
+	  if (!strncasecmp(word, peer->name, wordlen) &&
+		   (!flags2 || ast_test_flag(&peer->flags[1], flags2)) &&
 			   ++which > state && peer->expire > 0)
 			   result = ast_strdup(peer->name);
-		   if (result) {
-			   unref_peer(peer, "toss iterator peer ptr before break");
-			   break;
-		   }
-		   unref_peer(peer, "toss iterator peer ptr");
-       }
-       return result;
+	  if (result) {
+      unref_peer(peer, "toss iterator peer ptr before break");
+		  break;
+    }
+		unref_peer(peer, "toss iterator peer ptr");
+  }
+  return result;
 }
 
 /*! \brief Support routine for 'sip show history' CLI */
@@ -16505,17 +16718,32 @@
 	struct ast_channel *bridgepeer = NULL;
 	char *p_hdrval;
 	int rtn;
-	
+  int hangup = 0; 
+  struct sip_pvt *headp = NULL;
+  struct sip_pvt *tmpp = NULL;
+  
 	if (reinvite)
+  {
+		ast_log(LOG_NOTICE, "SIP response %d to RE-invite on %s call %s\n", resp, outgoing ? "outgoing" : "incoming", p->callid);
 		ast_debug(4, "SIP response %d to RE-invite on %s call %s\n", resp, outgoing ? "outgoing" : "incoming", p->callid);
+  }
 	else
+  {
+		ast_log(LOG_NOTICE, "SIP response %d to standard invite\n", resp);
 		ast_debug(4, "SIP response %d to standard invite\n", resp);
+  }
 
 	if (p->alreadygone) { /* This call is already gone */
 		ast_debug(1, "Got response on call that is already terminated: %s (ignoring)\n", p->callid);
 		return;
 	}
 
+  if(p->owner) {
+    headp = p->owner->tech_pvt;
+    if(p->owner->no_of_pvt_channels < 1)
+      hangup = 1;
+  }
+
 	/* Acknowledge sequence number - This only happens on INVITE from SIP-call */
 	/* Don't auto congest anymore since we've gotten something useful back */
 	AST_SCHED_DEL_UNREF(sched, p->initid, dialog_unref(p, "when you delete the initid sched, you should dec the refcount for the stored dialog ptr"));
@@ -16618,7 +16846,7 @@
 		if (p->owner && (p->owner->_state == AST_STATE_UP) && (bridgepeer = ast_bridged_channel(p->owner))) { /* if this is a re-invite */
 			struct sip_pvt *bridgepvt = NULL;
 
-			if (!bridgepeer->tech) {
+      if (!bridgepeer->tech) {
 				ast_log(LOG_WARNING, "Ooooh.. no tech!  That's REALLY bad\n");
 				break;
 			}
@@ -16652,6 +16880,15 @@
 
 		if (!req->ignore && p->owner) {
 			if (!reinvite) {
+        if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+          /* if tech_pvt ref in owner is not the one with which call is being established, 
+           * make it the tech_pvt in owner, others are anyhow going to die */
+	        if(p->owner->tech_pvt != p)
+          {
+            p->owner->tech_pvt = dialog_unref(p->owner->tech_pvt, "unref p->owner->tech_pvt");
+	          p->owner->tech_pvt = dialog_ref(p, "set p->owner->tech_pvt to p");
+          }
+        }
 				ast_queue_control(p->owner, AST_CONTROL_ANSWER);
 				if (sip_cfg.callevents)
 					manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
@@ -16699,12 +16936,38 @@
 			}
 		}
 
-
 		/* If I understand this right, the branch is different for a non-200 ACK only */
 		p->invitestate = INV_TERMINATED;
 		ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
 		xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
 		check_pendings(p);
+  
+    if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+      tmpp = headp;
+      /* send CANCEL or BYE to other forked calls */
+      while(tmpp) {
+        if(tmpp != p) {
+			    /* stop retransmitting an INVITE that has not received a response */
+			    __sip_pretend_ack(tmpp);
+
+    			/* if we can't send right now, mark it pending */
+		    	if (tmpp->invitestate != INV_TERMINATED && tmpp->owner != NULL) {
+            ast_set_flag(&tmpp->flags[0], SIP_PENDINGBYE);
+            tmpp->pendinginvite = 0;
+            tmpp->owner = NULL;
+            stop_media_flows(tmpp);
+		    	}
+    			if ( tmpp->initid != -1 ) {
+    				update_call_counter(tmpp, INC_CALL_LIMIT);
+			    }
+		      check_pendings(tmpp);
+        }
+        p->owner->no_of_pvt_channels--;
+        tmpp = tmpp->next;
+      }
+      p->next = NULL;
+    }
+    
 		break;
 
 	case 407: /* Proxy authentication */
@@ -16721,6 +16984,26 @@
 				p->invitestate = INV_CALLING;
 			if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, SIP_INVITE, 1)) {
 				ast_log(LOG_NOTICE, "Failed to authenticate on INVITE to '%s'\n", get_header(&p->initreq, "From"));
+        if(headp == p) {
+          struct ast_channel* owner = headp->owner;
+          owner->no_of_pvt_channels--;
+          headp = p->next;
+          /* this sip_pvt is going to die, make other in the list as the ref to owner */
+          owner->tech_pvt = dialog_unref(p, "unref p");
+          owner->tech_pvt = dialog_ref(headp, "ref headp");
+        }
+        else {
+          tmpp = p->owner->tech_pvt;
+          while(tmpp) {
+            if(tmpp->next == p) {
+              tmpp->next = p->next;
+              p->owner->no_of_pvt_channels--;
+              break;
+            }
+            tmpp = tmpp->next;
+          }
+        }
+ 
 				pvt_set_needdestroy(p, "failed to authenticate on INVITE");
 				sip_alreadygone(p);
 				if (p->owner)
@@ -16733,7 +17016,7 @@
 		/* First we ACK */
 		xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
 		ast_log(LOG_WARNING, "Received response: \"Forbidden\" from '%s'\n", get_header(&p->initreq, "From"));
-		if (!req->ignore && p->owner)
+		if (!req->ignore && p->owner && hangup)
 			ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
 		pvt_set_needdestroy(p, "received 403 response");
 		sip_alreadygone(p);
@@ -16741,7 +17024,7 @@
 
 	case 404: /* Not found */
 		xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
-		if (p->owner && !req->ignore)
+		if (p->owner && !req->ignore && hangup)
 			ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
 		sip_alreadygone(p);
 		break;
@@ -16751,7 +17034,7 @@
 		/* Could be REFER caused INVITE with replaces */
 		ast_log(LOG_WARNING, "Re-invite to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid);
 		xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
-		if (p->owner)
+		if (p->owner && hangup)
 			ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
 		sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 		break;
@@ -16760,26 +17043,27 @@
 		xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
 		ast_string_field_set(p, theirtag, NULL);
 		proc_422_rsp(p, req);
+    if(p->owner)
+      p->owner->no_of_pvt_channels++;
 		break;
 
 	case 428: /* Use identity header - rfc 4474 - not supported by Asterisk yet */
 		xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
 		append_history(p, "Identity", "SIP identity is required. Not supported by Asterisk.");
 		ast_log(LOG_WARNING, "SIP identity required by proxy. SIP dialog '%s'. Giving up.\n", p->callid);
-		if (p->owner)
+		if (p->owner && hangup)
 			ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
 		break;
 
-		
-
 	case 487: /* Cancelled transaction */
 		/* We have sent CANCEL on an outbound INVITE 
 			This transaction is already scheduled to be killed by sip_hangup().
 		*/
 		xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
-		if (p->owner && !req->ignore) {
-			ast_queue_hangup_with_cause(p->owner, AST_CAUSE_NORMAL_CLEARING);
-			append_history(p, "Hangup", "Got 487 on CANCEL request from us. Queued AST hangup request");
+
+		if (p->owner && !req->ignore && hangup) {
+			  ast_queue_hangup_with_cause(p->owner, AST_CAUSE_NORMAL_CLEARING);
+  			append_history(p, "Hangup", "Got 487 on CANCEL request from us. Queued AST hangup request");
  		} else if (!req->ignore) {
 			update_call_counter(p, DEC_CALL_LIMIT);
 			append_history(p, "Hangup", "Got 487 on CANCEL request from us on call without owner. Killing this dialog.");
@@ -16796,6 +17080,8 @@
 
 			/* Trigger a reinvite back to audio */
 			transmit_reinvite_with_sdp(p, FALSE, FALSE);
+      if(p->owner)
+        p->owner->no_of_pvt_channels++;
 		} else if (p->udptl && p->t38.state == T38_LOCAL_DIRECT) {
 			/* We tried to send T.38 out in an initial INVITE and the remote side rejected it,
 			   right now we can't fall back to audio so totally abort.
@@ -16806,13 +17092,13 @@
 
 			change_t38_state(p, T38_DISABLED);
 			/* The dialog is now terminated */
-			if (p->owner && !req->ignore)
+			if (p->owner && !req->ignore && hangup)
 				ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
 			pvt_set_needdestroy(p, "got error on T.38 initial invite");
 			sip_alreadygone(p);
 		} else {
 			/* We can't set up this call, so give up */
-			if (p->owner && !req->ignore)
+			if (p->owner && !req->ignore && hangup)
 				ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
 			pvt_set_needdestroy(p, "received 488 response");
 			/* If there's no dialog to end, then mark p as already gone */
@@ -16823,7 +17109,7 @@
 	case 491: /* Pending */
 		xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
 		if (p->owner && !req->ignore) {
-			if (p->owner->_state != AST_STATE_UP) {
+			if (p->owner->_state != AST_STATE_UP && hangup) {
 				ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
 				pvt_set_needdestroy(p, "received 491 response");
 			} else {
@@ -16840,7 +17126,7 @@
 
 	case 501: /* Not implemented */
 		xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
-		if (p->owner)
+		if (p->owner && hangup)
 			ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
 		break;
 	}
@@ -17240,6 +17526,8 @@
 	char *c_copy = ast_strdupa(c);
 	/* Skip the Cseq and its subsequent spaces */
 	const char *msg = ast_skip_blanks(ast_skip_nonblanks(c_copy));
+  int hangup = 0;
+  struct sip_pvt *tmpp = NULL;
 
 	if (!msg)
 		msg = "";
@@ -17247,9 +17535,38 @@
 	sipmethod = find_sip_method(msg);
 
 	owner = p->owner;
-	if (owner) 
+	if (owner) {
+    if(resp > 299 && resp != 401 && resp != 407 && ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+      if(owner->tech_pvt == p) {
+        /* decrementing no of pvt channels in owner as we got error response */
+        owner->no_of_pvt_channels--;
+        p->peer = NULL;
+        if(p->next)
+        {
+          tmpp = p->next;
+          /*make some others in the list as the ref to owner */
+          owner->tech_pvt = dialog_unref(p, "unref p");
+          owner->tech_pvt = dialog_ref(tmpp, "ref p");
+        }
+      }
+      else {
+        tmpp = owner->tech_pvt;
+        while(tmpp) {
+          if(tmpp->next == p) {
+            tmpp->next = p->next;
+            p->peer = NULL;
+            owner->no_of_pvt_channels--;
+            break;
+          }
+          tmpp = tmpp->next;
+        }
+      }
+    }
 		owner->hangupcause = hangup_sip2cause(resp);
-
+    if(owner->no_of_pvt_channels < 1)
+      hangup = 1;
+  }
+    
 	/* Acknowledge whatever it is destined for */
 	if ((resp >= 100) && (resp <= 199))
 		__sip_semi_ack(p, seqno, 0, sipmethod);
@@ -17472,7 +17789,7 @@
 				case 486: /* Busy here */
 				case 600: /* Busy everywhere */
 				case 603: /* Decline */
-					if (p->owner)
+					if (p->owner && hangup)
 						ast_queue_control(p->owner, AST_CONTROL_BUSY);
 					break;
 				case 482: /*!
@@ -17501,18 +17818,21 @@
 				case 502: /* Bad gateway */
 				case 503: /* Service Unavailable */
 				case 504: /* Server Timeout */
-					if (owner)
+					if (owner && hangup)
 						ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
 					break;
 				default:
 					/* Send hangup */	
-					if (owner && sipmethod != SIP_MESSAGE && sipmethod != SIP_INFO && sipmethod != SIP_BYE)
+					if (owner && hangup && sipmethod != SIP_MESSAGE && sipmethod != SIP_INFO && sipmethod != SIP_BYE)
 						ast_queue_hangup_with_cause(p->owner, AST_CAUSE_PROTOCOL_ERROR);
 					break;
 				}
 				/* ACK on invite */
-				if (sipmethod == SIP_INVITE) 
+				if (sipmethod == SIP_INVITE) { 
 					transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
+          p->pendinginvite = 0;
+					pvt_set_needdestroy(p, "transaction completed");
+        }
 				if (sipmethod != SIP_MESSAGE && sipmethod != SIP_INFO) 
 					sip_alreadygone(p);
 				if (!p->owner) {
@@ -18897,7 +19217,7 @@
 			ast_string_field_set(p, theirtag, NULL);
 			return 0;
 		}
-
+    
 		/* We have a succesful authentication, process the SDP portion if there is one */
 		if (find_sdp(req)) {
 			if (process_sdp(p, req, SDP_T38_INITIATE)) {
@@ -19776,7 +20096,6 @@
 /*! \brief Handle incoming CANCEL request */
 static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
 {
-		
 	check_via(p, req);
 	sip_alreadygone(p);
 
@@ -20899,7 +21218,11 @@
 
 	if (p->owner && !nounlock)
 		ast_channel_unlock(p->owner);
-	sip_pvt_unlock(p);
+
+  if(p->peer == NULL && ast_test_flag(&p->flags[0], SIP_OUTGOING))
+    p->owner = NULL;
+	
+  sip_pvt_unlock(p);
 	ast_mutex_unlock(&netlock);
 	ao2_t_ref(p, -1, "throw away dialog ptr from find_call at end of routine"); /* p is gone after the return */
 	return 1;
@@ -21875,7 +22198,7 @@
 */
 static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause)
 {
-	struct sip_pvt *p;
+	struct sip_pvt *p = NULL;
 	struct ast_channel *tmpc = NULL;
 	char *ext = NULL, *host;
 	char tmp[256];
@@ -21887,6 +22210,11 @@
 	char *trans = NULL;
 	enum sip_transport transport = 0;
 	int oldformat = format;
+  struct sip_peer *peer = NULL;
+  struct sip_peer *tmppeer = NULL;
+  struct sip_pvt *tmpp = NULL;
+  struct sip_pvt *headp = NULL;
+  int pvt_channels = 0;
 
 	/* mask request with some set of allowed formats.
 	 * XXX this needs to be fixed.
@@ -21903,122 +22231,161 @@
 	}
 	ast_debug(1, "Asked to create a SIP channel with formats: %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), oldformat));
 
-	if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE))) {
-		ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory or socket error)\n", dest);
-		*cause = AST_CAUSE_SWITCH_CONGESTION;
+  peer = find_peer(dest, NULL, TRUE, FINDALLDEVICES, FALSE);
+
+  if(!peer) {
+		*cause = AST_CAUSE_UNREGISTERED;
+    unref_peer(peer, "unref_peer: from sip_request_call");
 		return NULL;
-	}
+  }
+  
+  tmppeer = peer;
+  
+  if(ast_strlen_zero(tmppeer->fullcontact))
+    tmppeer = tmppeer->list.next;
 
-	p->outgoing_call = TRUE;
+  if(!tmppeer) {
+  	*cause = AST_CAUSE_UNREGISTERED;
+  	ast_debug(3, "Cant create SIP call - target device not registred\n");
+    unref_peer(peer, "unref_peer: from sip_request_call");
+    return NULL;
+  }
 
-	if (!(p->options = ast_calloc(1, sizeof(*p->options)))) {
-		dialog_unlink_all(p, TRUE, TRUE);
-		dialog_unref(p, "unref dialog p from mem fail");
-		/* sip_destroy(p); */
-		ast_log(LOG_ERROR, "Unable to build option SIP data structure - Out of memory\n");
-		*cause = AST_CAUSE_SWITCH_CONGESTION;
-		return NULL;
-	}
+  /* fork the call to all the locations this user has registerd from */
+  do {
+    tmpp = p;
+  	if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE))) {
+	  	ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory or socket error)\n", dest);
+		  *cause = AST_CAUSE_SWITCH_CONGESTION;
+      unref_peer(peer, "unref_peer: from sip_request_call");
+		  return NULL;
+	  }
+  
+  	p->outgoing_call = TRUE;
+    p->peer = tmppeer;
 
-	/* Save the destination, the SIP dial string */
-	ast_copy_string(tmp, dest, sizeof(tmp));
+  	if (!(p->options = ast_calloc(1, sizeof(*p->options)))) {
+	  	dialog_unlink_all(p, TRUE, TRUE);
+		  dialog_unref(p, "unref dialog p from mem fail");
+  		/* sip_destroy(p); */
+	  	ast_log(LOG_ERROR, "Unable to build option SIP data structure - Out of memory\n");
+  		*cause = AST_CAUSE_SWITCH_CONGESTION;
+      unref_peer(peer, "unref_peer: from sip_request_call");
+	  	return NULL;
+  	}
 
+  	/* Save the destination, the SIP dial string */
+  	ast_copy_string(tmp, dest, sizeof(tmp));
 
-	/* Find DNID and take it away */
-	dnid = strchr(tmp, '!');
-	if (dnid != NULL) {
-		*dnid++ = '\0';
-		ast_string_field_set(p, todnid, dnid);
-	}
 
-	/* Find at sign - @ */
-	host = strchr(tmp, '@');
-	if (host) {
-		*host++ = '\0';
-		ext = tmp;
-		secret = strchr(ext, ':');
-	}
-	if (secret) {
-		*secret++ = '\0';
-		md5secret = strchr(secret, ':');
-	}
-	if (md5secret) {
-		*md5secret++ = '\0';
-		authname = strchr(md5secret, ':');
-	}
-	if (authname) {
-		*authname++ = '\0';
-		trans = strchr(authname, ':');
-	}
-	if (trans) {
-		*trans++ = '\0';
-		if (!strcasecmp(trans, "tcp"))
-			transport = SIP_TRANSPORT_TCP;
-		else if (!strcasecmp(trans, "tls"))
-			transport = SIP_TRANSPORT_TLS;
-		else {
-			if (strcasecmp(trans, "udp"))
-				ast_log(LOG_WARNING, "'%s' is not a valid transport option to Dial() for SIP calls, using udp by default.\n", trans);
-			transport = SIP_TRANSPORT_UDP;
-		}
-	} else { /* use default */
-		transport = SIP_TRANSPORT_UDP;
-	}
+  	/* Find DNID and take it away */
+	  dnid = strchr(tmp, '!');
+	  if (dnid != NULL) {
+		  *dnid++ = '\0';
+		  ast_string_field_set(p, todnid, dnid);
+	  }
 
-	if (!host) {
-		ext = strchr(tmp, '/');
-		if (ext) 
-			*ext++ = '\0';
-		host = tmp;
-	}
+	  /* Find at sign - @ */
+	  host = strchr(tmp, '@');
+	  if (host) {
+  		*host++ = '\0';
+		  ext = tmp;
+  		secret = strchr(ext, ':');
+  	}
+  	if (secret) {
+	  	*secret++ = '\0';
+		  md5secret = strchr(secret, ':');
+  	}
+	  if (md5secret) {
+		  *md5secret++ = '\0';
+  		authname = strchr(md5secret, ':');
+	  }
+  	if (authname) {
+	  	*authname++ = '\0';
+		  trans = strchr(authname, ':');
+  	}
+	  if (trans) {
+		  *trans++ = '\0';
+  		if (!strcasecmp(trans, "tcp"))
+	  		transport = SIP_TRANSPORT_TCP;
+  		else if (!strcasecmp(trans, "tls"))
+	  		transport = SIP_TRANSPORT_TLS;
+  		else {
+	  		if (strcasecmp(trans, "udp"))
+		  		ast_log(LOG_WARNING, "'%s' is not a valid transport option to Dial() for SIP calls, using udp by default.\n", trans);
+			  transport = SIP_TRANSPORT_UDP;
+  		}
+	  } else { /* use default */
+  		transport = SIP_TRANSPORT_UDP;
+	  }
 
-	p->socket.fd = -1;
-	p->socket.type = transport;
+  	if (!host) {
+	  	ext = strchr(tmp, '/');
+		  if (ext) 
+			  *ext++ = '\0';
+  		host = tmp;
+  	}
 
-	/* We now have 
-		host = peer name, DNS host name or DNS domain (for SRV) 
-		ext = extension (user part of URI)
-		dnid = destination of the call (applies to the To: header)
-	*/
-	if (create_addr(p, host, NULL, 1)) {
-		*cause = AST_CAUSE_UNREGISTERED;
-		ast_debug(3, "Cant create SIP call - target device not registered\n");
-		dialog_unlink_all(p, TRUE, TRUE);
-		dialog_unref(p, "unref dialog p UNREGISTERED");
-		/* sip_destroy(p); */
-		return NULL;
-	}
-	if (ast_strlen_zero(p->peername) && ext)
-		ast_string_field_set(p, peername, ext);
-	/* Recalculate our side, and recalculate Call ID */
-	ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
-	build_via(p);
-	ao2_t_unlink(dialogs, p, "About to change the callid -- remove the old name");
-	build_callid_pvt(p);
-	ao2_t_link(dialogs, p, "Linking in under new name");
+  	p->socket.fd = -1;
+	  p->socket.type = transport;
+
+	  /* We now have 
+  		host = peer name, DNS host name or DNS domain (for SRV) 
+	  	ext = extension (user part of URI)
+  		dnid = destination of the call (applies to the To: header)
+	  */
+  	if (create_addr(p, host, NULL, 1)) {
+	  	*cause = AST_CAUSE_UNREGISTERED;
+		  ast_debug(3, "Cant create SIP call - target device not registered\n");
+  		dialog_unlink_all(p, TRUE, TRUE);
+	  	dialog_unref(p, "unref dialog p UNREGISTERED");
+      unref_peer(peer, "unref_peer: from sip_request_call");
+  		/* sip_destroy(p); */
+	  	return NULL;
+  	}
+	  if (ast_strlen_zero(p->peername) && ext)
+		  ast_string_field_set(p, peername, ext);
+  	/* Recalculate our side, and recalculate Call ID */
+	  ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
+  	build_via(p);
+	  ao2_t_unlink(dialogs, p, "About to change the callid -- remove the old name");
+  	build_callid_pvt(p);
+	  ao2_t_link(dialogs, p, "Linking in under new name");
 	
-	/* We have an extension to call, don't use the full contact here */
-	/* This to enable dialing registered peers with extension dialling,
-	   like SIP/peername/extension 	
-	   SIP/peername will still use the full contact 
-	 */
-	if (ext) {
-		ast_string_field_set(p, username, ext);
-		ast_string_field_set(p, fullcontact, NULL);
-	}
-	if (secret && !ast_strlen_zero(secret))
-		ast_string_field_set(p, peersecret, secret);
+	  /* We have an extension to call, don't use the full contact here */
+  	/* This to enable dialing registered peers with extension dialling,
+	     like SIP/peername/extension 	
+	     SIP/peername will still use the full contact 
+  	 */
+	  if (ext) {
+  		ast_string_field_set(p, username, ext);
+	  	ast_string_field_set(p, fullcontact, NULL);
+  	}
+	  if (secret && !ast_strlen_zero(secret))
+		  ast_string_field_set(p, peersecret, secret);
 
-	if (md5secret && !ast_strlen_zero(md5secret))
-		ast_string_field_set(p, peermd5secret, md5secret);
+	  if (md5secret && !ast_strlen_zero(md5secret))
+		  ast_string_field_set(p, peermd5secret, md5secret);
 
-	if (authname && !ast_strlen_zero(authname))
-		ast_string_field_set(p, authname, authname);
+  	if (authname && !ast_strlen_zero(authname))
+	  	ast_string_field_set(p, authname, authname);
 #if 0
 	printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "<none>", host);
 #endif
-	p->prefcodec = oldformat;				/* Format for this call */
-	p->jointcapability = oldformat;
+  	p->prefcodec = oldformat;				/* Format for this call */
+  	p->jointcapability = oldformat;
+  
+    if(tmpp)
+      tmpp->next = p;
+
+    if(!headp)
+      headp = p;
+    tmppeer = tmppeer->list.next;
+  } while(tmppeer);
+
+  unref_peer(peer, "unref_peer: from sip_request_call");
+
+  p = headp;
 	sip_pvt_lock(p);
 	tmpc = sip_new(p, AST_STATE_DOWN, host);	/* Place the call */
 	if (sip_cfg.callevents)
@@ -22030,8 +22397,20 @@
 		dialog_unlink_all(p, TRUE, TRUE);
 		/* sip_destroy(p); */
 	}
-	dialog_unref(p, "toss pvt ptr at end of sip_request_call");
-	ast_update_use_count();
+
+  tmpp = headp;
+
+  while(tmpp) {
+    dialog_unref(tmpp, "toss pvt ptr at end of sip_request_call");
+
+    pvt_channels++;
+    if(tmpp != headp)
+      tmpp->owner = tmpc;           /* keep a ref to owner in pvt channels */
+	  ast_update_use_count();
+    tmpp = tmpp->next;
+  }
+  
+  tmpc->no_of_pvt_channels = pvt_channels;  /* update the no of forked channels*/
 	restart_monitor();
 	return tmpc;
 }
@@ -22438,7 +22817,7 @@
 	peer->selfdestruct = TRUE;
 	peer->host_dynamic = TRUE;
 	peer->prefs = default_prefs;
-	reg_source_db(peer);
+	reg_source_db(peer, 1);
 
 	return peer;
 }
@@ -22484,7 +22863,7 @@
 	const char *srvlookup = NULL;
 	static int deprecation_warning = 1;
 	struct ast_str *fullcontact = ast_str_alloca(512);
-	
+
 	if (!realtime || ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
 		/* Note we do NOT use find_peer here, to avoid realtime recursion */
 		/* We also use a case-sensitive comparison (unlike find_peer) so
@@ -22924,7 +23303,7 @@
 	if (ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE))
 		sip_cfg.allowsubscribe = TRUE;	/* No global ban any more */
 	if (!found && peer->host_dynamic && !peer->is_realtime)
-		reg_source_db(peer);
+		reg_source_db(peer, 0);
 
 	/* If they didn't request that MWI is sent *only* on subscribe, go ahead and
 	 * subscribe to it now. */
@@ -23630,7 +24009,9 @@
 				ast_log(LOG_WARNING, "Invalid pokepeers '%s' at line %d of %s\n", v->value, v->lineno, config);
 				global_qualify_peers = DEFAULT_QUALIFY_PEERS;
 			}
-		}
+		} else if(!strcasecmp(v->name, "allowforking")) {
+			sip_cfg.allow_forking = ast_true(v->value) ? 1 : 0;
+    }
 	}
 
 	if (!sip_cfg.allow_external_domains && AST_LIST_EMPTY(&domain_list)) {
@@ -23665,10 +24046,14 @@
 				if (ast_true(hassip) || (!hassip && genhassip)) {
 					peer = build_peer(cat, gen, ast_variable_browse(ucfg, cat), 0, FALSE);
 					if (peer) {
-						ao2_t_link(peers, peer, "link peer into peer table");
-						if (peer->addr.sin_addr.s_addr) {
-							ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
-						}
+            struct sip_peer* tmppeer = peer;
+            while(tmppeer) {
+  						ao2_t_link(peers, tmppeer, "link peer into peer table");
+	  					if (tmppeer->addr.sin_addr.s_addr) {
+		  					ao2_t_link(peers_by_ip, tmppeer, "link peer into peers_by_ip table");
+			  			}
+              tmppeer = tmppeer->list.next;
+            }
 						
 						unref_peer(peer, "unref_peer: from reload_config");
 						peer_count++;
@@ -23729,10 +24114,14 @@
 			if (is_peer) {
 				peer = build_peer(cat, ast_variable_browse(cfg, cat), NULL, 0, is_peer == 2);
 				if (peer) {
-					ao2_t_link(peers, peer, "link peer into peers table");
-					if (peer->addr.sin_addr.s_addr) {
-						ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
-					}
+          struct sip_peer* tmppeer = peer;
+          while(tmppeer) {
+					  ao2_t_link(peers, tmppeer, "link peer into peers table");
+  					if (tmppeer->addr.sin_addr.s_addr) {
+	  					ao2_t_link(peers_by_ip, tmppeer, "link peer into peers_by_ip table");
+		  			}
+            tmppeer = tmppeer->list.next;
+          }
 					unref_peer(peer, "unref the result of the build_peer call. Now, the links from the tables are the only ones left.");
 					peer_count++;
 				}
@@ -24717,6 +25106,107 @@
 	return 0;
 }
 
+/* add peer info to db, there can be a user registered from multiple locations, so update all of them into db*/
+static int sip_db_add(struct sip_peer *peer, int expire, char* data, int nodepos)
+{
+  char tmpdata[SIPBUFSIZE] = {0};
+  char tmpstr[SIPBUFSIZE] = {0};
+  char *scan = NULL;
+  char *s = NULL;
+
+  if (!ast_db_get("SIP/Registry", peer->name, data, SIPBUFSIZE)) {
+    scan = data;
+    if(nodepos < 0) {
+      while(scan) {
+        s = strsep(&scan, ",");
+        if(!ast_strlen_zero(s)) {
+          strlcat(tmpdata, s, sizeof(tmpdata));
+          strlcat(tmpdata, ",", sizeof(tmpdata));
+        }
+      }
+      
+      strlcat(tmpdata, ast_inet_ntoa(peer->addr.sin_addr), sizeof(tmpdata));
+      strlcat(tmpdata, ":", sizeof(tmpdata));
+     
+      snprintf(tmpstr, sizeof(tmpstr), "%d", ntohs(peer->addr.sin_port));
+      strlcat(tmpdata, tmpstr, sizeof(tmpdata));
+      strlcat(tmpdata, ":", sizeof(tmpdata));
+          
+      snprintf(tmpstr, sizeof(tmpstr), "%d", expire);
+      strlcat(tmpdata, tmpstr, sizeof(tmpdata));
+      strlcat(tmpdata, ":", sizeof(tmpdata));
+      
+      strlcat(tmpdata, peer->username, sizeof(tmpdata));
+      strlcat(tmpdata, ":", sizeof(tmpdata));
+      
+      strlcat(tmpdata, peer->fullcontact, sizeof(tmpdata));
+      strlcat(tmpdata, ",", sizeof(tmpdata));
+    }
+    else {
+      int object = 0;
+      while(scan) {
+        s = strsep(&scan, ",");
+        if(ast_strlen_zero(s))
+          continue;
+        if(object != nodepos) {
+          strlcat(tmpdata, s, sizeof(tmpdata));
+          strlcat(tmpdata, ",", sizeof(tmpdata));
+        }
+        else {
+          strlcat(tmpdata, ast_inet_ntoa(peer->addr.sin_addr), sizeof(tmpdata));
+          strlcat(tmpdata, ":", sizeof(tmpdata));
+     
+          snprintf(tmpstr, sizeof(tmpstr), "%d", ntohs(peer->addr.sin_port));
+          strlcat(tmpdata, tmpstr, sizeof(tmpdata));
+          strlcat(tmpdata, ":", sizeof(tmpdata));
+          
+          snprintf(tmpstr, sizeof(tmpstr), "%d", expire);
+          strlcat(tmpdata, tmpstr, sizeof(tmpdata));
+          strlcat(tmpdata, ":", sizeof(tmpdata));
+      
+          strlcat(tmpdata, peer->username, sizeof(tmpdata));
+          strlcat(tmpdata, ":", sizeof(tmpdata));
+      
+          strlcat(tmpdata, peer->fullcontact, sizeof(tmpdata));
+          strlcat(tmpdata, ",", sizeof(tmpdata));
+        }
+        object++;
+      }
+    }
+    memset(data, 0, SIPBUFSIZE);
+	  snprintf(data, SIPBUFSIZE, "%s", tmpdata);
+  }
+  else
+    snprintf(data, SIPBUFSIZE, "%s:%d:%d:%s:%s,", ast_inet_ntoa(peer->addr.sin_addr), ntohs(peer->addr.sin_port), expire, peer->username, peer->fullcontact);
+  
+  return 0;
+}
+
+/* allocate a sip_peer object */
+static struct sip_peer *sip_allocpeer(const char *name)
+{
+	struct sip_peer *peer;
+
+	if (!(peer = ao2_t_alloc(sizeof(*peer), sip_destroy_peer_fn, "allocate a peer struct")))
+		return NULL;
+
+	if (ast_string_field_init(peer, 512)) {
+		ao2_t_ref(peer, -1, "failed to string_field_init, drop peer");
+		return NULL;
+	}
+
+	ast_atomic_fetchadd_int(&apeerobjs, 1);
+	set_peer_defaults(peer);
+
+	ast_copy_string(peer->name, name, sizeof(peer->name));
+
+	peer->selfdestruct = TRUE;
+	peer->host_dynamic = TRUE;
+	peer->prefs = default_prefs;
+  peer->list.next = NULL;
+  return peer;
+}
+
 AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Session Initiation Protocol (SIP)",
 		.load = load_module,
 		.unload = unload_module,