END; } $idx++; $var_localnet = "localnet_$idx"; $var_netmask = "netmask_$idx"; } $tabindex += 40; // make room for dynamic insertion of new fields so we can add tabindexes ?> END; } $idx++; $var_sip_custom_key = "sip_custom_key_$idx"; $var_sip_custom_val = "sip_custom_val_$idx"; } $tabindex += 60; // make room for dynamic insertion of new fields ?>

".$div_disp['div'].""; } ?>
yes = Always ignore info and assume NAT no = Use NAT mode only according to RFC3581 never = Never attempt NAT mode or RFC3581 route = Assume NAT, don't send rport")?>	/> yes /> no /> never /> route
	/> /> />
	/
	/ END; $tabindex++; echo <<< END
	" class="nat-settings" /> " class="nat-settings" />
	$codec_state) { if ($cols == 0) { echo "\n"; $cols = $cols_per_row; } $cols--; $tabindex++; $codec_trans = _($codec); $codec_checked = $codec_state ? 'checked' : ''; echo <<< END END; } ?> $codec_trans
	/> />
	/> />
	/> />
	$codec_state) { if ($cols == 0) { echo "\n"; $cols = $cols_per_row; } $cols--; $tabindex++; $codec_trans = _($codec); $codec_checked = $codec_state ? 'checked' : ''; echo <<< END END; } ?> $codec_trans
	/> /> /> nonat /> update
Asterisk: rtpholdtimeout. Terminate call if rtpholdtimeout seconds of no RTP or RTCP activity on the audio channel when we're on hold (must be > rtptimeout). Asterisk: rtpkeepalive. Send keepalives in the RTP stream to keep NAT open during periods where no RTP stream may be flowing (like on hold).")?>	(rtptimeout)   (rtpholdtimeout)   (rtpkeepalive)
	/> />
	/> />
Asterisk: registrationattempts. Number of times to try and register before giving up. A value of 0 means keep trying forever. Normally this should be set to 0 so that Asterisk will continue to register until successful in the case of network or gateway outages.")?>	(registertimeout)   (registerattempts)
Asterisk: maxepiry. Maximum allowed time of incoming registrations Asterisk: defaultexpiry. Default length of incoming and outgoing registrations.")?>	(minexpiry)   (maxexpiry)   (defaultexpiry)
	/> />
	/> />
fixed: size always equals to jbmaxsize; adaptive: with variable size (the new jb of IAX2).")?>	/> />
	/> />
Asterisk: jbresyncthreshold. Jump in the frame timestamps over which the jitterbuffer is resynchronized. Useful to improve the quality of the voice, with big jumps in/broken timestamps, usually sent from exotic devices and programs. Can be set to -1 to disable.")?>	(jbmaxsize)   (jbresyncthreshold)
	/> />
	/> />
[setting] = [value] in the boxes below. Click the Add Field box to add additional fields. Blank boxes will be deleted when submitted.")?>	=
	= END; $tabindex++; echo <<< END
	" />
" tabindex="">

"$('#".$error['id']."').addClass('validation-error');\n", 'div' => $error['message'], ); } return $error_display; } function sipsettings_check_custom_files() { global $amp_conf; $errors = array(); $custom_files[] = $amp_conf['ASTETCDIR']."/sip.conf"; $custom_files[] = $amp_conf['ASTETCDIR']."/sip_nat.conf"; $custom_files[] = $amp_conf['ASTETCDIR']."/sip_general_custom.conf"; $custom_files[] = $amp_conf['ASTETCDIR']."/sip_custom.conf"; foreach ($custom_files as $file) { if (file_exists($file)) { $sip_conf = parse_ini_file($file,true); $main = true; // 1 is sip.conf, after that don't care foreach ($sip_conf as $section => $item) { // If setting is an array, then it is a subsection // if (!is_array($item)) { $msg = sprintf(_("Settings in %s may override these. Those settings should be removed."),"$file"); $errors[] = array( 'js' => '', 'div' => $msg); break; } elseif ($main && is_array($item) && strtolower($section) == 'general' && !empty($item)) { $msg = sprintf(_("File %s should not have any settings in it. Those settings should be removed."),"$file"); $errors[] = array( 'js' => '', 'div' => $msg); break; } $main = false; } } } return $errors; } ?>