Describe your package hereDescribe your package requirements hereCurrently there are no FAQ items provided.siproxdsettings0.8.0_2 pkg v1.0.2siproxd: Settings/usr/local/pkg/siproxd.inc/pkg_edit.php?xml=siproxd.xml&id=0siproxdsiproxd.shsiproxdSettings/pkg_edit.php?xml=siproxd.xml&id=0Users/pkg.php?xml=siproxdusers.xmlRegistered Phones/siproxd_registered_phones.php/usr/local/pkg/077https://packages.pfsense.org/packages/config/siproxdusers.xml/usr/local/pkg/077https://packages.pfsense.org/packages/config/siproxd.inc/usr/local/www/077https://packages.pfsense.org/packages/config/siproxd_registered_phones.phpEnable siproxdsipenableEnable or disable siproxdcheckboxInbound interfaceif_inboundSelect the inbound interface.interfaces_selectionOutbound interfaceif_outboundSelect the outbound interface.interfaces_selectionListening portportEnter the port on which to listen for SIP traffic (default 5060). Do not change this unless you know what you're doing.inputDefault expiration timeoutdefaulttimeoutIf a REGISTER request dose not contain an Expires header or expires= parameter, this number of seconds will be used and reported back to the UA in the answer.inputRTP SettingslisttopicEnable RTP proxyrtpenableEnable or disable the RTP proxy. (default is enabled)selectRTP port range (lower)rtplowerEnter the bottom edge of the port range siproxd will allocate for incoming RTP traffic. This range must be one not blocked by the firewall (default 7070).inputRTP port range (upper)rtpupperEnter the top edge of the port range siproxd will allocate for incoming RTP traffic. This range must be one not blocked by the firewall (default 7079).inputRTP stream timeoutrtptimeoutAfter this number of seconds, an RTP stream is considered dead and proxying it will be stopped (default 300sec).inputDejittering SettingslisttopicInput Dejitterrtp_input_dejitterArtificial delay to be used to de-jitter RTP data streams. This time is in microseconds. 0 - completely disable dejitter (default)inputOutput Dejitterrtp_output_dejitterArtificial delay to be used to de-jitter RTP data streams. This time is in microseconds. 0 - completely disable dejitter (default)inputSIP over TCP SettingslisttopicTCP inactivity timeouttcp_timeoutInactivity timeout (seconds). After that an idling TCP connection is disconnected. NOTE: Making this too short may cause multiple parallel registrations for the same phone. This timeout must be set larger than the used registration interval.inputTCP Connect Timeouttcp_connect_timeoutDefines How many msecs siproxd will wait for an successful connect when establishing an outgoing SIP signalling connection. This should be kept as short as possible as waiting for an TCP connection to establish is a BLOCKING operation - while waiting for a connect to succeed no SIP messages are processed (RTP is not affected).inputTCP Keepalivetcp_keepaliveFor TCP SIP signalling, if > 0 empty SIP packets will be sent every 'n' seconds to keep the connection alive. Default is off.inputProxy SettingslisttopicEnable proxy authenticationauthenticationIf this is checked, clients will be forced to authenticate themselves at the proxy (for registration only).checkboxOutbound proxy hostnameoutboundproxyhostEnter the hostname of an outbound proxy to send all traffic to. This is only useful if you have multiple masquerading firewalls to cross.inputOutbound proxy portoutboundproxyportEnter the port of the outbound proxy to send all traffic to. This is only useful if you have multiple masquerading firewalls to cross.inputDSCP SettingslisttopicExpedited RTP ForwardingexpeditedforwardingThis service is designed to allow ISPs to offer a service with attributes similar to a "leased line". This service offers the ULTIMATE IN LOW LOSS, LOW LATENCY AND LOW JITTER by ensuring that there is always sufficient room in output queues for the contracted expedited forwarding traffic.
checkboxExpedited SIP ForwardingexpeditedsipforwardingThis service is designed to allow ISPs to offer a service with attributes similar to a "leased line". This service offers the ULTIMATE IN LOW LOSS, LOW LATENCY AND LOW JITTER by ensuring that there is always sufficient room in output queues for the contracted expedited forwarding traffic.checkboxPlugin Settings - Default TargetlisttopicEnable Default Target Pluginplugin_defaulttargetRedirect unknown calls to a specified target.checkboxLog redirected callsplugin_defaulttarget_logLog redirected calls.checkboxDefault Targetplugin_defaulttarget_targetTarget must be a full SIP URI with the syntax sip:user@host[:port]inputPlugin Settings - Bogus VIA NetworkslisttopicEnable Fix Bogus Via Networks Pluginplugin_fix_bogus_viaIncoming (from public network) SIP messages are checked for broken SIP Via headers. If the IP address in the latest Via Header is part of the list below, it will be replaced by the IP where the SIP message has been received from.checkboxBogus Via Networksplugin_fix_bogus_via_networksComma separated list of networks which should have their via headers rewritten. Example: 10.0.0.0/8,172.16.0.0/12,192.168.0.0/16inputPlugin Settings - STUNlisttopicEnable STUN Pluginplugin_stunUses an external STUN server to determine the public IP address of siproxd. Useful for "in-front-of-NAT-router" scenarios.checkboxSTUN Hostnameplugin_stun_serverExternal STUN server hostname.inputSTUN Portplugin_stun_portExternal STUN server port.inputSTUN Periodplugin_stun_periodPeriod in seconds to request IP info from STUN server.inputDebug OptionslisttopicDebug Leveldebug_levelselect1TCP Debug Portdebug_portYou may connect to this port from a remote machine and receive debug output. This allows better creation of debug output on embedded systems that do not have enough memory for large disk files. Port number 0 means this feature is disabled.input
sync_package_siproxd();
sync_package_siproxd();
siproxd_generate_rules
validate_form_siproxd($_POST, $input_errors);