automerge commit

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2-netsec@13274 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_zap.c

@@ -5856,7 +5856,7 @@ static void *ss_thread(void *data)
 					number = 0;
 			/* If set to use V23 Signalling, launch our FSK gubbins and listen for it */
 			} else if (p->cid_signalling == CID_SIG_V23) {
-				cs = callerid_new(cid_signalling);
+				cs = callerid_new(p->cid_signalling);
 				if (cs) {
 					samples = 0;
 #if 1
@@ -6027,7 +6027,7 @@ static void *ss_thread(void *data)
 			}
 		} else if (p->use_callerid && p->cid_start == CID_START_RING) {
 			/* FSK Bell202 callerID */
-			cs = callerid_new(cid_signalling);
+			cs = callerid_new(p->cid_signalling);
 			if (cs) {
 #if 1
 				bump_gains(p);

configs/sip.conf.sample

@@ -86,6 +86,7 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 ;progressinband=never		; If we should generate in-band ringing always
 				; use 'never' to never use in-band signalling, even in cases
 				; where some buggy devices might not render it
+				; Valid values: yes, no, never Default: never
 ;useragent=Asterisk PBX		; Allows you to change the user agent string
 ;promiscredir = no      	; If yes, allows 302 or REDIR to non-local SIP address
 	                       	; Note that promiscredir when redirects are made to the