Describe your package here Describe your package requirements here Currently there are no FAQ items provided. siproxdsettings 0.5.13_pfs2 /usr/local/pkg/siproxd.inc /pkg_edit.php?xml=siproxd.xml&id=0 siproxd Modify siproxd users and settings.

Services

/pkg_edit.php?xml=siproxd.xml&id=0 siproxd siproxd.sh siproxd Settings /pkg_edit.php?xml=siproxd.xml&id=0 Users /pkg.php?xml=siproxdusers.xml Registered Phones /siproxd_registered_phones.php /usr/local/pkg/ 077 http://www.pfsense.com/packages/config/siproxdusers.xml /usr/local/pkg/ 077 http://www.pfsense.com/packages/config/siproxd.inc /usr/local/www/ 077 http://www.pfsense.com/packages/config/siproxd_registered_phones.php Inbound interface if_inbound Select the inbound interface. interfaces_selection Outbound interface if_outbound Select the outbound interface. interfaces_selection Listening port port Enter the port on which to listen for SIP traffic (default 5060). Do not change this unless you know what you're doing. input Default expiration timeout defaulttimeout If 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. input RTP Settings listtopic Enable RTP proxy rtpenable Enable or disable the RTP proxy. (default is enabled) select Enable 1 Disable 0 RTP port range (lower) rtplower Enter 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). input RTP port range (upper) rtpupper Enter 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). input RTP stream timeout rtptimeout After this number of seconds, an RTP stream is considered dead and proxying it will be stopped (default 300sec). input Dejittering Settings listtopic Input Dejitter rtp_input_dejitter Artificial delay to be used to de-jitter RTP data streams. This time is in microseconds. 0 - completely disable dejitter (default) input Output Dejitter rtp_output_dejitter Artificial delay to be used to de-jitter RTP data streams. This time is in microseconds. 0 - completely disable dejitter (default) input SIP over TCP Settings listtopic TCP inactivity timeout tcp_timeout Inactivity 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. input TCP Connect Timeout tcp_connect_timeout Defines 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). input TCP Keepalive tcp_keepalive For TCP SIP signalling, if > 0 empty SIP packets will be sent every 'n' seconds to keep the connection alive. Default is off. input Proxy Settings listtopic Enable proxy authentication authentication If this is checked, clients will be forced to authenticate themselves at the proxy (for registration only). checkbox Outbound proxy hostname outboundproxyhost Enter the hostname of an outbound proxy to send all traffic to. This is only useful if you have multiple masquerading firewalls to cross. input Outbound proxy port outboundproxyport Enter the port of the outbound proxy to send all traffic to. This is only useful if you have multiple masquerading firewalls to cross. input DSCP Settings listtopic Expedited RTP Forwarding expeditedforwarding This 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. checkbox Expedited SIP Forwarding expeditedsipforwarding This 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. checkbox Plugin Settings - Default Target listtopic Enable Default Target Plugin plugin_defaulttarget Redirect unknown calls to a specified target. checkbox Log redirected calls plugin_defaulttarget_log Log redirected calls. checkbox Default Target plugin_defaulttarget_target Target must be a full SIP URI with the syntax sip:user@host[:port] input Plugin Settings - Bogus VIA Networks listtopic Enable Fix Bogus Via Networks Plugin plugin_fix_bogus_via Incoming (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. checkbox Bogus Via Networks plugin_fix_bogus_via_networks Comma separated list of networks which should have their via headers rewritten. Example: 10.0.0.0/8,172.16.0.0/12,192.168.0.0/16 input Plugin Settings - STUN listtopic Enable STUN Plugin plugin_stun Uses an external STUN server to determine the public IP address of siproxd. Useful for "in-front-of-NAT-router" scenarios. checkbox STUN Hostname plugin_stun_server External STUN server hostname. input STUN Port plugin_stun_port External STUN server port. input STUN Period plugin_stun_period Period in seconds to request IP info from STUN server. input sync_package_siproxd(); sync_package_siproxd(); siproxd_generate_rules validate_form_siproxd($_POST, &$input_errors);