Author: brian Date: Fri Nov 9 13:03:53 2007 New Revision: 6198 Added: freeswitch/trunk/conf/sip_profiles/ freeswitch/trunk/conf/sip_profiles/default.xml freeswitch/trunk/conf/sip_profiles/nat.xml Modified: freeswitch/trunk/conf/sofia.conf.xml freeswitch/trunk/conf/vars.xml Log: house keeping.. moving things around a bit more to demo various things you can do Added: freeswitch/trunk/conf/sip_profiles/default.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/sip_profiles/default.xml Fri Nov 9 13:03:53 2007 @@ -0,0 +1,106 @@ +<profile name="default"> + <!--aliases are other names that will work as a valid profile name for this profile--> + <!-- <aliases> --> + <!-- <alias name="default"/> --> + <!-- </aliases> --> + <!-- Outbound Registrations --> + <gateways> + <!--<gateway name="asterlink.com">--> + <!--/// account username *required* ///--> + <!--<param name="username" value="cluecon"/>--> + <!--/// auth realm: *optional* same as gateway name, if blank ///--> + <!--<param name="realm" value="asterlink.com"/>--> + <!--/// domain to use in from: *optional* same as realm, if blank ///--> + <!--<param name="from-domain" value="asterlink.com"/>--> + <!--/// account password *required* ///--> + <!--<param name="password" value="2007"/>--> + <!--/// replace the INVITE from user with the channel's caller-id ///--> + <!--<param name="caller-id-in-from" value="false"/>--> + <!--/// extension for inbound calls: *optional* same as username, if blank ///--> + <!--<param name="extension" value="cluecon"/>--> + <!--/// proxy host: *optional* same as realm, if blank ///--> + <!--<param name="proxy" value="asterlink.com"/>--> + <!--/// send register to this proxy: *optional* same as proxy, if blank ///--> + <!--<param name="register-proxy" value="mysbc.com"/>--> + <!--/// expire in seconds: *optional* 3600, if blank ///--> + <!--<param name="expire-seconds" value="60"/>--> + <!--/// do not register ///--> + <!--<param name="register" value="false"/>--> + <!-- which transport to use for register --> + <!--<param name="register-transport" value="udp"/>--> + <!--How many seconds before a retry when a failure or timeout occurs --> + <!--<param name="retry_seconds" value="30"/>--> + <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway --> + <!--<param name="caller-id-in-from" value="false"/>--> + <!--extra sip params to send in the contact--> + <!--<param name="contact-params" value="tport=tcp"/>--> + <!--</gateway>--> + </gateways> + + <domains> + <!-- indicator to parse the directory for domains with parse="true" to get gateways--> + <!--<domain name="$${domain}" parse="true"/>--> + </domains> + + <settings> + <param name="debug" value="1"/> + <param name="rfc2833-pt" value="101"/> + <param name="sip-port" value="5062"/> + <param name="dialplan" value="XML,enum"/> + <param name="dtmf-duration" value="100"/> + <param name="codec-prefs" value="$${global_codec_prefs}"/> + <param name="use-rtp-timer" value="true"/> + <param name="rtp-timer-name" value="soft"/> + <param name="rtp-ip" value="$${local_ip_v4}"/> + <param name="sip-ip" value="$${local_ip_v4}"/> + <!--enable to use presense and mwi --> + <param name="manage-presence" value="true"/> + <!--max number of open dialogs in proceeding --> + <!--<param name="max-proceeding" value="1000"/>--> + <!--session timers for all call to expire after the specified seconds --> + <!--<param name="session-timeout" value="120"/>--> + <!--<param name="multiple-registrations" value="true"/>--> + <!--set to 'greedy' if you want your codec list to take precedence --> + <param name="inbound-codec-negotiation" value="generous"/> + <!-- if you want to send any special bind params of your own --> + <!--<param name="bind-params" value="transport=udp"/>--> + + <!--If you don't want to pass through timestampes from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)--> + <!--<param name="rtp-rewrite-timestampes" value="true"/>--> + + <!--If you have ODBC support and a working dsn you can use it instead of SQLite--> + <!--<param name="odbc-dsn" value="dsn:user:pass"/>--> + + <!--Uncomment to set all inbound calls to no media mode--> + <!--<param name="inbound-no-media" value="true"/>--> + + <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok--> + <!--<param name="inbound-late-negotiation" value="true"/>--> + + <!-- this lets anything register --> + <!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> + <!-- <param name="accept-blind-reg" value="true"/> --> + + <!--TTL for nonce in sip auth--> + <param name="nonce-ttl" value="60"/> + <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec + that the originator is using--> + <!--<param name="disable-transcoding" value="true"/>--> + <param name="auth-calls" value="true"/> + <!-- on authed calls, authenticate *all* the packets not just invite --> + <!-- <param name="auth-all-packets" value="true"/> --> + + <!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> --> + <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> --> + <!-- rtp inactivity timeout --> + <!--<param name="rtp-timeout-sec" value="300"/>--> + <!-- VAD choose one (out is a good choice); --> + <!-- <param name="vad" value="in"/> --> + <!-- <param name="vad" value="out"/> --> + <!-- <param name="vad" value="both"/> --> + <!--<param name="alias" value="sip:10.0.1.251:5555"/>--> + <!--all inbound reg will look in this domain for the users --> + <!--<param name="force-register-domain" value="cluecon.com"/>--> + </settings> +</profile> + Added: freeswitch/trunk/conf/sip_profiles/nat.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/sip_profiles/nat.xml Fri Nov 9 13:03:53 2007 @@ -0,0 +1,19 @@ +<profile name="nat"> + <settings> + <param name="debug" value="1"/> + <param name="rfc2833-pt" value="101"/> + <param name="sip-port" value="5061"/> + <param name="dialplan" value="XML,enum"/> + <param name="dtmf-duration" value="100"/> + <param name="codec-prefs" value="$${global_codec_prefs}"/> + <param name="use-rtp-timer" value="true"/> + <param name="rtp-timer-name" value="soft"/> + <param name="rtp-ip" value="$${local_ip_v4}"/> + <param name="sip-ip" value="$${local_ip_v4}"/> + <param name="manage-presence" value="true"/> + <param name="inbound-codec-negotiation" value="generous"/> + <param name="nonce-ttl" value="60"/> + <param name="auth-calls" value="true"/> + <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> + </settings> +</profile> Modified: freeswitch/trunk/conf/sofia.conf.xml ============================================================================== --- freeswitch/trunk/conf/sofia.conf.xml (original) +++ freeswitch/trunk/conf/sofia.conf.xml Fri Nov 9 13:03:53 2007 @@ -1,130 +1,5 @@ <configuration name="sofia.conf" description="sofia Endpoint"> <profiles> - <profile name="$${sip_profile}"> - <!--aliases are other names that will work as a valid profile name for this profile--> - <aliases> - <alias name="default"/> - </aliases> - <!-- Outbound Registrations --> - <gateways> - <!--<gateway name="asterlink.com">--> - <!--/// account username *required* ///--> - <!--<param name="username" value="cluecon"/>--> - <!--/// auth realm: *optional* same as gateway name, if blank ///--> - <!--<param name="realm" value="asterlink.com"/>--> - <!--/// domain to use in from: *optional* same as realm, if blank ///--> - <!--<param name="from-domain" value="asterlink.com"/>--> - <!--/// account password *required* ///--> - <!--<param name="password" value="2007"/>--> - <!--/// replace the INVITE from user with the channel's caller-id ///--> - <!--<param name="caller-id-in-from" value="false"/>--> - <!--/// extension for inbound calls: *optional* same as username, if blank ///--> - <!--<param name="extension" value="cluecon"/>--> - <!--/// proxy host: *optional* same as realm, if blank ///--> - <!--<param name="proxy" value="asterlink.com"/>--> - <!--/// send register to this proxy: *optional* same as proxy, if blank ///--> - <!--<param name="register-proxy" value="mysbc.com"/>--> - <!--/// expire in seconds: *optional* 3600, if blank ///--> - <!--<param name="expire-seconds" value="60"/>--> - <!--/// do not register ///--> - <!--<param name="register" value="false"/>--> - <!-- which transport to use for register --> - <!--<param name="register-transport" value="udp"/>--> - <!--How many seconds before a retry when a failure or timeout occurs --> - <!--<param name="retry_seconds" value="30"/>--> - <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway --> - <!--<param name="caller-id-in-from" value="false"/>--> - <!--extra sip params to send in the contact--> - <!--<param name="contact-params" value="tport=tcp"/>--> - <!--</gateway>--> - </gateways> - - <domains> - <!-- indicator to parse the directory for domains with parse="true" to get gateways--> - <!--<domain name="$${domain}" parse="true"/>--> - </domains> - - <settings> - <param name="debug" value="1"/> - <param name="rfc2833-pt" value="101"/> - <param name="sip-port" value="5060"/> - <param name="dialplan" value="XML,enum"/> - <param name="dtmf-duration" value="100"/> - <param name="codec-prefs" value="$${global_codec_prefs}"/> - <param name="use-rtp-timer" value="true"/> - <param name="rtp-timer-name" value="soft"/> - <param name="rtp-ip" value="$${local_ip_v4}"/> - <param name="sip-ip" value="$${local_ip_v4}"/> - <!--enable to use presense and mwi --> - <param name="manage-presence" value="true"/> - <!--max number of open dialogs in proceeding --> - <!--<param name="max-proceeding" value="1000"/>--> - <!--session timers for all call to expire after the specified seconds --> - <!--<param name="session-timeout" value="120"/>--> - <!--<param name="multiple-registrations" value="true"/>--> - <!--set to 'greedy' if you want your codec list to take precedence --> - <param name="inbound-codec-negotiation" value="generous"/> - <!-- if you want to send any special bind params of your own --> - <!--<param name="bind-params" value="transport=udp"/>--> - - <!--If you don't want to pass through timestampes from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)--> - <!--<param name="rtp-rewrite-timestampes" value="true"/>--> - - <!--If you have ODBC support and a working dsn you can use it instead of SQLite--> - <!--<param name="odbc-dsn" value="dsn:user:pass"/>--> - - <!--Uncomment to set all inbound calls to no media mode--> - <!--<param name="inbound-no-media" value="true"/>--> - - <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok--> - <!--<param name="inbound-late-negotiation" value="true"/>--> - - <!-- this lets anything register --> - <!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> - <!-- <param name="accept-blind-reg" value="true"/> --> - - <!--TTL for nonce in sip auth--> - <param name="nonce-ttl" value="60"/> - <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec - that the originator is using--> - <!--<param name="disable-transcoding" value="true"/>--> - <param name="auth-calls" value="true"/> - <!-- on authed calls, authenticate *all* the packets not just invite --> - <!-- <param name="auth-all-packets" value="true"/> --> - - <!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> --> - <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> --> - <!-- rtp inactivity timeout --> - <!--<param name="rtp-timeout-sec" value="300"/>--> - <!-- VAD choose one (out is a good choice); --> - <!-- <param name="vad" value="in"/> --> - <!-- <param name="vad" value="out"/> --> - <!-- <param name="vad" value="both"/> --> - <!--<param name="alias" value="sip:10.0.1.251:5555"/>--> - <!--all inbound reg will look in this domain for the users --> - <!--<param name="force-register-domain" value="cluecon.com"/>--> - </settings> - </profile> - - <profile name="nat"> - <settings> - <param name="debug" value="1"/> - <param name="rfc2833-pt" value="101"/> - <param name="sip-port" value="5061"/> - <param name="dialplan" value="XML,enum"/> - <param name="dtmf-duration" value="100"/> - <param name="codec-prefs" value="$${global_codec_prefs}"/> - <param name="use-rtp-timer" value="true"/> - <param name="rtp-timer-name" value="soft"/> - <param name="rtp-ip" value="$${local_ip_v4}"/> - <param name="sip-ip" value="$${local_ip_v4}"/> - <param name="manage-presence" value="true"/> - <param name="inbound-codec-negotiation" value="generous"/> - <param name="nonce-ttl" value="60"/> - <param name="auth-calls" value="true"/> - <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> - </settings> - </profile> - + <!--#include "sip_profiles/*" --> </profiles> </configuration> Modified: freeswitch/trunk/conf/vars.xml ============================================================================== --- freeswitch/trunk/conf/vars.xml (original) +++ freeswitch/trunk/conf/vars.xml Fri Nov 9 13:03:53 2007 @@ -8,6 +8,7 @@ used by: sofia.conf.xml enum.conf.xml default_context.xml directory.xml --> <!--#set "sip_profile=$${domain}"--> + <!--#set "nat_sip_profile=nat_$${domain}"--> <!-- xmpp_client_profile and xmpp_server_profile xmpp_client_profile can be any string. xmpp_server_profile is appended to "dingaling_" to form the database name