Index: channels/chan_iax2.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_iax2.c,v
retrieving revision 1.237
diff -u -r1.237 chan_iax2.c
--- channels/chan_iax2.c	14 Jan 2005 21:10:16 -0000	1.237
+++ channels/chan_iax2.c	15 Jan 2005 06:31:29 -0000
@@ -2367,6 +2367,44 @@
 	return tmp;
 }
 
+/* Sub-function of iax2_call. Allows format and capability override. */
+static void iax2_call_set_codecs(struct iax_ie_data *ie_data, struct ast_channel *channel, struct chan_iax2_pvt* pvt)
+{
+	/*  Shares code with chan_sip.c::sip_call */
+	/*	CODEC_OVERRIDE can override the selection for available codecs.
+		Ensure dialplan sets var with leading underscore so it gets passed. 
+		Override does not check capabilities. This is so that a type=friend can limit what they accept via the .conf,
+		yet still allow different codecs at runtime.		*/
+
+	int format = channel->nativeformats;			/* Default to using the current native format */
+	int capability = pvt->capability;
+	int override_format = 0;
+	char *codec_override; 
+	
+	codec_override = pbx_builtin_getvar_helper(channel, "CODEC_OVERRIDE");
+	if (codec_override && !ast_strlen_zero(codec_override)) {
+		override_format = ast_getformatbyname(codec_override);
+		if (override_format) {
+			if ((override_format & pvt->capability) & (option_verbose > 2)) {
+				ast_verbose(VERBOSE_PREFIX_3 "CODEC_OVERRIDE: Forcing format for this call to %s\n", codec_override);
+			} else if (option_verbose > 2) {
+				ast_verbose(VERBOSE_PREFIX_3 "CODEC_OVERRIDE: Forcing format for this call to %s (0x%x), even though not in current capabilities (0x%x).\n",  
+						codec_override,
+						override_format,
+						pvt->capability);
+			}
+			format = override_format;
+			capability = override_format;
+		} else {
+			ast_log(LOG_NOTICE, "Ignoring CODEC_OVERRIDE variable: Unknown codec '%s'.\n", codec_override);
+		}
+	}
+
+	pvt->capability = capability;
+	iax_ie_append_int(ie_data, IAX_IE_FORMAT, format);
+	iax_ie_append_int(ie_data, IAX_IE_CAPABILITY, capability);
+}
+
 static int iax2_call(struct ast_channel *c, char *dest, int timeout)
 {
 	struct sockaddr_in sin;
@@ -2499,8 +2537,7 @@
 		} else
 			strncpy(iaxs[callno]->secret, secret, sizeof(iaxs[callno]->secret)-1);
 	}
-	iax_ie_append_int(&ied, IAX_IE_FORMAT, c->nativeformats);
-	iax_ie_append_int(&ied, IAX_IE_CAPABILITY, iaxs[callno]->capability);
+	iax2_call_set_codecs(&ied, c, iaxs[callno]);
 	iax_ie_append_short(&ied, IAX_IE_ADSICPE, c->adsicpe);
 	iax_ie_append_int(&ied, IAX_IE_DATETIME, iax2_datetime(tz));
 	/* Transmit the string in a "NEW" request */
Index: channels/chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.631
diff -u -r1.631 chan_sip.c
--- channels/chan_sip.c	13 Jan 2005 20:26:33 -0000	1.631
+++ channels/chan_sip.c	15 Jan 2005 06:31:39 -0000
@@ -1386,14 +1386,15 @@
 	struct varshead *headp;
 	struct ast_var_t *current;
 	int addsipheaders = 0;
-	
+	char *codec_override;
+	int override_format;
+
 	p = ast->pvt->pvt;
 	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 */
@@ -1432,7 +1433,31 @@
 	res = update_user_counter(p,INC_OUT_USE);
 	if ( res != -1 ) {
 		p->callingpres = ast->cid.cid_pres;
+		
+		/*   Shares code with chan_iax2.c::iax2_call_set_codecs */
+		/* Handle CODEC_OVERRIDE (override capabilities). Ensure dialplan sets var with leading underscore so it gets passed. 
+		   Override does not check capabilities. This is so that a type=friend can limit what they accept via the .conf,
+		   yet still allow different codecs at runtime. */
+		codec_override = pbx_builtin_getvar_helper(ast, "CODEC_OVERRIDE");
+		if (codec_override && !ast_strlen_zero(codec_override)) {
+			override_format = ast_getformatbyname(codec_override);
+			if (override_format) {
+				if ((override_format & p->capability) & (option_verbose > 2)) {
+					ast_verbose(VERBOSE_PREFIX_3 "CODEC_OVERRIDE: Forcing format for this call to %s\n", codec_override);
+				} else if (option_verbose > 2) {
+					ast_verbose(VERBOSE_PREFIX_3 "CODEC_OVERRIDE: Forcing format for this call to %s (0x%x), even though not in current capabilities (0x%x).\n",  
+						codec_override,
+						override_format,
+						p->capability);
+				}
+				p->prefcodec = override_format;
+				p->capability = override_format;
+			} else {
+				ast_log(LOG_NOTICE, "Ignoring CODEC_OVERRIDE variable: Unknown codec '%s'.\n", codec_override);
+			}
+		}
 		p->jointcapability = p->capability;
+
 		transmit_invite(p, "INVITE", 1, NULL, NULL, vxml_url,distinctive_ring, osptoken, addsipheaders, 1);
 		if (p->maxtime) {
 			/* Initialize auto-congest time */
@@ -3343,6 +3368,7 @@
 	add_line(req, tmp);
 	return 0;
 }
+
 
 /*--- add_sdp: Add Session Description Protocol message ---*/
 static int add_sdp(struct sip_request *resp, struct sip_pvt *p)