[Serusers] Questions on using ser.cfg version 0.9.0 for call forward/busy/noanswer

[Charles Wang]

Dear Paul:

It seems the URI error on my failure_route[1] to route[6].

I force to rewriteuser("0939749xxx") before failure_route[1] to route[6], 
the call will forward to my mobile phone(although only one way voice).

I store my forward number at some UA id's "fwdbusy" attribute 
and value is "sip:0939749xxx at ser.xxx.net.tw" in usr_preferences table.
Using avp_write? How to implement it, please?

How can I change it to a dialed number?

Best Regard 
Charles

the snippet of ser.cfg:
------------------------------------------------------------------------------------------------------
failure_route[1] {
	log(1, "SER: Failure Route section failure_route(1)\n");

	# if caller hung up then don't sent to voicemail
	if (t_check_status("487")) {
		break;
	};
	if (isflagset(26) && t_check_status("486")) {
		# forward busy is flag 26
		
		if (avp_pushto("$ruri", "s:fwdbusy")) {
			log(1, "SER: fork to fwdbusy\n");
			avp_delete("s:fwdbusy");
			append_branch();
			resetflag(26);
			
			# test for domestic PSTN gateway
			if (uri=~"^sip:0[0-9]{9}@") {
			# if (avp_check("$fwd_busy_type", "eq/dom/i")) {
				# test for domestic PSTN gateway
				log(1, "SER: Busy Failure and Jump to route(6)\n");
				route(6);
			} else if (uri=~"^sip:002[1-9][0-9]*@") {
			# } else if (avp_check("$fwd_busy_type", "eq/int/i")) {
				# test for international PSTN gateway
				log(1, "SER: Busy Failure and Jump to route(3)\n");
				route(3);
			} else {
				# default to sip call
				log(1, "SER: Busy Failure and Jump to route(2)\n");
				route(2);
			};
			break;
		};
	};

	# here we can have either voicemail __OR__ forward no answer
	if (isflagset(27) && t_check_status("408")) {
		# forward no answer is flag 27
		
		if (avp_db_load("$ruri/username", "s:fwdnoanswer")) {
			avp_pushto("$ruri", "s:fwdnoanswer");
			log(1, "SER: fork to fwdnoanswer\n");
			avp_delete("s:fwdnoanswer");
			append_branch();
			resetflag(27);
			
			if (uri=~"^sip:0[0-9]{9}@") {
			# if (avp_check("$fwd_no_answer_type", "eq/dom/i")) {
				# test for domestic PSTN gateway
				log(1, "SER: No Answer Failure and Jump to route(3)\n");
				route(3);
		  } else if (uri=~"^sip:002[1-9][0-9]*@") {
			# } else if (avp_check("$fwd_no_answer_type", "eq/int/i")) {
				# test for international PSTN gateway
				log(1, "SER: No Answer Failure and Jump to route(6)\n");
				rewriteuser("0939749xxx");  <=== FORCE REWRITE HERE
				route(6);
			} else {
				# default to sip call
				log(1, "SER: No Answer Failure and Jump to route(2)\n");				
				route(2);
			};
			break;
		};
	} else if (isflagset(31) && avp_pushto("$ruri", "$voicemail")) {
		avp_delete("$voicemail");
		log(1, "SER: No Answer Failure and Jump to route(4)\n");
		route(4);
		break;
	};
}