# 2005-09-03 09:47 +0100
#
# This patch adds support for non-numeric callerids being sent on over
# SIP (instead of being changed to "asterisk"), and for calling using any
# peer as a SIP proxy where the outbound domain is unknown in sip.conf.
#
# Dial(SIP/peer![extension@]host) will prepare a connection to 'peer' and
# then continue as if it was connected to 'host' but still using all of
# the settings of 'peer'.
#
# --
# Simon Arlott
# http://simon.arlott.org/
diff -U4 -r asterisk-1.2.0-beta1/channels/chan_sip.c asterisk-1.2.0-beta1+byte/channels/chan_sip.c
--- asterisk-1.2.0-beta1/channels/chan_sip.c	2005-08-26 21:04:28.000000000 +0100
+++ asterisk-1.2.0-beta1+byte/channels/chan_sip.c	2005-09-03 08:42:54.776277111 +0100
@@ -103,9 +103,9 @@
 #define MAX(a,b) ((a) > (b) ? (a) : (b))
 #endif
 
 #define CALLERID_UNKNOWN	"Unknown"
-
+#define MAX_CIDNUM_LEN      79 /* It's used in the Contact header too so it must fit. */
 
 
 #define DEFAULT_MAXMS		2000		/* Must be faster than 2 seconds by default */
 #define DEFAULT_FREQ_OK		60 * 1000	/* How often to check for the host to be up */
@@ -4356,10 +4356,12 @@
 	char to[256];
 	char tmp[80];
 	char iabuf[INET_ADDRSTRLEN];
 	char *l = default_callerid, *n=NULL;
+	char cidnum[MAX_CIDNUM_LEN+1];
 	int x;
 	char urioptions[256]="";
+	int pos=0;
 
 	if (ast_test_flag(p, SIP_USEREQPHONE)) {
 	 	char onlydigits = 1;
 		x=0;
@@ -4390,10 +4392,13 @@
 	if (p->owner) {
 		l = p->owner->cid.cid_num;
 		n = p->owner->cid.cid_name;
 	}
+/*
 	if (!l || (!ast_isphonenumber(l) && default_callerid[0]))
 			l = default_callerid;
+*/
+	if (!l) l = CALLERID_UNKNOWN;
 	/* if user want's his callerid restricted */
 	if ((p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
 		l = CALLERID_UNKNOWN;
 		n = l;
@@ -4411,12 +4416,41 @@
 		n = p->fromname;
 	else /* Save for any further attempts */
 		ast_copy_string(p->fromname, n, sizeof(p->fromname));
 
+	if (!ast_isphonenumber(l)) {
+		static char hex[] = "0123456789ABCDEF";
+		int i;
+		for (i=0; i<strlen(l); i++) {
+			if (l[i] < 32) continue;
+			if ((l[i] >= 65 && l[i] <= 90) //A-Z
+				|| (l[i] >= 97 && l[i] <= 122) //a-z
+				|| (l[i] >= 48 && l[i] <= 57) //0-9
+				|| (l[i] == 35) //#
+				|| (l[i] == 42) //*
+				|| (l[i] == 43) //+
+				|| (l[i] == 45) //-
+				|| (l[i] == 95) //_
+				) { // valid char
+				cidnum[pos]=l[i];
+				pos++;
+				if (pos == MAX_CIDNUM_LEN) break;
+			} else { // invalid char
+				if (pos+3 > MAX_CIDNUM_LEN) break;
+				cidnum[pos] = '%';
+				cidnum[pos+1] = hex[l[i] >> 4];
+				cidnum[pos+2] = hex[l[i] & 0x0f];
+				pos+=3;
+				if (pos == MAX_CIDNUM_LEN) break;
+			}
+		}
+		cidnum[pos]='\0';
+	}
+
 	if ((ourport != 5060) && ast_strlen_zero(p->fromdomain))	/* Needs to be 5060 */
-		snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s:%d>;tag=as%08x", n, l, ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain, ourport, p->tag);
+		snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s:%d>;tag=as%08x", n, (pos == 0 ? l : cidnum), ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain, ourport, p->tag);
 	else
-		snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s>;tag=as%08x", n, l, ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain, p->tag);
+		snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s>;tag=as%08x", n, (pos == 0 ? l : cidnum), ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain, p->tag);
 
 	/* If we're calling a registred SIP peer, use the fullcontact to dial to the peer */
 	if (!ast_strlen_zero(p->fullcontact)) {
 		/* If we have full contact, trust it */
@@ -4448,9 +4482,9 @@
 	add_header(req, "Via", p->via);
 	/* SLD: FIXME?: do Route: here too?  I think not cos this is the first request.
 	 * OTOH, then we won't have anything in p->route anyway */
 	add_header(req, "From", from);
-	ast_copy_string(p->exten, l, sizeof(p->exten));
+	ast_copy_string(p->exten, (pos == 0 ? l : cidnum), sizeof(p->exten));
 	build_contact(p);
 	add_header(req, "To", to);
 	add_header(req, "Contact", p->our_contact);
 	add_header(req, "Call-ID", p->callid);
@@ -10437,8 +10471,9 @@
 	struct ast_channel *tmpc = NULL;
 	char *ext, *host;
 	char tmp[256] = "";
 	char *dest = data;
+	char *fakehost = NULL;
 
 	oldformat = format;
 	format &= ((AST_FORMAT_MAX_AUDIO << 1) - 1);
 	if (!format) {
@@ -10451,33 +10486,51 @@
 		return NULL;
 	}
 
 	ast_copy_string(tmp, dest, sizeof(tmp));
-	host = strchr(tmp, '@');
-	if (host) {
-		*host = '\0';
-		host++;
-		ext = tmp;
-	} else {
-		ext = strchr(tmp, '/');
-		if (ext) {
-			*ext++ = '\0';
-			host = tmp;
+	ext = strchr(tmp, '!');
+	if (ext) {
+		*ext = '\0';
+		ext++;
+		fakehost = tmp;
+		host = strchr(ext, '@');
+		if (host) {
+			*host++ = '\0';
 		}
 		else {
-			host = tmp;
+			host = ext;
 			ext = NULL;
 		}
+	} else {
+		host = strchr(tmp, '@');
+		if (host) {
+			*host = '\0';
+			host++;
+			ext = tmp;
+		} else {
+			ext = strchr(tmp, '/');
+			if (ext) {
+				*ext++ = '\0';
+				host = tmp;
+			}
+			else {
+				host = tmp;
+				ext = NULL;
+			}
+		}
 	}
+	if (!fakehost && host) fakehost = host;
 
 	/* Assign a default capability */
 	p->capability = global_capability;
 
-	if (create_addr(p, host)) {
+	if (create_addr(p, fakehost)) {
 		*cause = AST_CAUSE_UNREGISTERED;
 		sip_destroy(p);
 		return NULL;
 	}
+	ast_copy_string(p->tohost, host, sizeof(p->tohost));
+
 	if (ast_strlen_zero(p->peername) && ext)
 		ast_copy_string(p->peername, ext, sizeof(p->peername));
 	/* Recalculate our side, and recalculate Call ID */
 	if (ast_sip_ouraddrfor(&p->sa.sin_addr,&p->ourip))