[asterisk-commits] tilghman: branch 1.6.0 r237966 - /branches/1.6.0/channels/chan_sip.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Jan 6 00:51:42 CST 2010


Author: tilghman
Date: Wed Jan  6 00:51:36 2010
New Revision: 237966

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=237966
Log:
Something clearly went wrong with a merge somewhere, because these are all duplicates (and therefore dead code).

Modified:
    branches/1.6.0/channels/chan_sip.c

Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=237966&r1=237965&r2=237966
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Wed Jan  6 00:51:36 2010
@@ -21884,71 +21884,6 @@
 						ast_log(LOG_WARNING, "Timer B has been set lower than recommended. (RFC 3261, 17.1.1.2)\n");
 					}
 				}
-			} else if (!strcasecmp(v->name, "subscribemwi")) {
-				ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_SUBSCRIBEMWIONLY);
-			} else if (!strcasecmp(v->name, "vmexten")) {
-				ast_copy_string(peer->vmexten, v->value, sizeof(peer->vmexten));
-			} else if (!strcasecmp(v->name, "callgroup")) {
-				peer->callgroup = ast_get_group(v->value);
-			} else if (!strcasecmp(v->name, "allowtransfer")) {
-				peer->allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED;
-			} else if (!strcasecmp(v->name, "pickupgroup")) {
-				peer->pickupgroup = ast_get_group(v->value);
-			} else if (!strcasecmp(v->name, "allow")) {
-				int error =  ast_parse_allow_disallow(&peer->prefs, &peer->capability, v->value, TRUE);
-				if (error) {
-					ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
-				}
-			} else if (!strcasecmp(v->name, "disallow")) {
-				int error =  ast_parse_allow_disallow(&peer->prefs, &peer->capability, v->value, FALSE);
-				if (error) {
-					ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
-				}
-			} else if (!strcasecmp(v->name, "registertrying")) {
-				ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_REGISTERTRYING);
-			} else if (!strcasecmp(v->name, "autoframing")) {
-				peer->autoframing = ast_true(v->value);
-			} else if (!strcasecmp(v->name, "rtptimeout")) {
-				if ((sscanf(v->value, "%30d", &peer->rtptimeout) != 1) || (peer->rtptimeout < 0)) {
-					ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d.  Using default.\n", v->value, v->lineno);
-					peer->rtptimeout = global_rtptimeout;
-				}
-			} else if (!strcasecmp(v->name, "rtpholdtimeout")) {
-				if ((sscanf(v->value, "%30d", &peer->rtpholdtimeout) != 1) || (peer->rtpholdtimeout < 0)) {
-					ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d.  Using default.\n", v->value, v->lineno);
-					peer->rtpholdtimeout = global_rtpholdtimeout;
-				}
-			} else if (!strcasecmp(v->name, "rtpkeepalive")) {
-				if ((sscanf(v->value, "%30d", &peer->rtpkeepalive) != 1) || (peer->rtpkeepalive < 0)) {
-					ast_log(LOG_WARNING, "'%s' is not a valid RTP keepalive time at line %d.  Using default.\n", v->value, v->lineno);
-					peer->rtpkeepalive = global_rtpkeepalive;
-				}
-			} else if (!strcasecmp(v->name, "timert1")) {
-				if ((sscanf(v->value, "%30d", &peer->timer_t1) != 1) || (peer->timer_t1 < 0)) {
-					ast_log(LOG_WARNING, "'%s' is not a valid T1 time at line %d.  Using default.\n", v->value, v->lineno);
-					peer->timer_t1 = global_t1;
-				}
-				/* Note that Timer B is dependent upon T1 and MUST NOT be lower
-				 * than T1 * 64, according to RFC 3261, Section 17.1.1.2 */
-				if (peer->timer_b < peer->timer_t1 * 64) {
-					peer->timer_b = peer->timer_t1 * 64;
-				}
-			} else if (!strcasecmp(v->name, "timerb")) {
-				if ((sscanf(v->value, "%30d", &peer->timer_b) != 1) || (peer->timer_b < 0)) {
-					ast_log(LOG_WARNING, "'%s' is not a valid Timer B time at line %d.  Using default.\n", v->value, v->lineno);
-					peer->timer_b = global_timer_b;
-				}
-				if (peer->timer_b < peer->timer_t1 * 64) {
-					static int warning = 0;
-					if (warning++ % 20 == 0) {
-						ast_log(LOG_WARNING, "Timer B has been set lower than recommended. (RFC 3261, 17.1.1.2)\n");
-					}
-				}
-			} else if (!strcasecmp(v->name, "rtpkeepalive")) {
-				if ((sscanf(v->value, "%30d", &peer->rtpkeepalive) != 1) || (peer->rtpkeepalive < 0)) {
-					ast_log(LOG_WARNING, "'%s' is not a valid RTP keepalive time at line %d.  Using default.\n", v->value, v->lineno);
-					peer->rtpkeepalive = global_rtpkeepalive;
-				}
 			} else if (!strcasecmp(v->name, "setvar")) {
 				peer->chanvars = add_var(v->value, peer->chanvars);
 			} else if (!strcasecmp(v->name, "qualifyfreq")) {




More information about the asterisk-commits mailing list