freeswitch/conf/vanilla/directory/default/brian.xml
Travis Cross 5a209a9680 Remove misleading tport example from configs
As an example of using mod_sofia's gateway parameter `contact-params`
we'd used the value `tport=tcp`.  Looking around, it's clear this has
misled people into believing you can specify `tport=tcp` to make the
gateway use TCP or `tport=tls` to make the gateway use TLS.  This does
not work.

The actual contact parameter is named `transport` rather than `tport`,
and you shouldn't use `transport` in `contact-params` because we
automatically add a `transport` to the Contact: based on the value of
`register-transport` (even if the gateway is set to not register).

It's clear why this would be confusing, so we'll just remove this as
an example.
2014-08-27 23:15:45 +00:00

93 lines
4.9 KiB
XML

<include>
<!--
ipauth if you have an cidr= in the user attributes ie cidr="1.2.3.4/32"
see <node type="allow" domain="$${domain}"/> in default acl.conf.xml
-->
<user id="brian" cidr="192.0.2.0/24">
<!-- Outbound Registrations Related to this user -->
<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"/>-->
<!--/// username to use in from: *optional* same as username, if blank ///-->
<!--<param name="from-user" value="cluecon"/>-->
<!--/// 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=""/>-->
<!--send an options ping every x seconds, failure will unregister and/or mark it down-->
<!--<param name="ping" value="25"/>-->
<!--</gateway>-->
</gateways>
<params>
<!-- omit password for authless registration -->
<param name="password" value="$${default_password}"/>
<param name="vm-password" value="9999"/><!--if vm-password is omitted password param is used-->
<!--<param name="email-addr" value="me@mydomain.com"/>-->
<!--<param name="vm-delete-file" value="true"/>-->
<!--<param name="vm-attach-file" value="true"/>-->
<!--<param name="vm-mailto" value="me@mydomain.com"/>-->
<!--<param name="vm-email-all-messages" value="true"/>-->
<!-- optionally use this instead if you want to store the hash of user:domain:pass-->
<!--<param name="a1-hash" value="c6440e5de50b403206989679159de89a"/>-->
<!-- What this user is allowed to acces -->
<!--<param name="http-allowed-api" value="jsapi,voicemail,status"/> -->
</params>
<variables>
<!--all variables here will be set on all inbound calls that originate from this user -->
<variable name="user_context" value="default"/>
<variable name="effective_caller_id_name" value="Brian West"/>
<variable name="effective_caller_id_number" value="1000"/>
<!-- Don't write a CDR if this is false valid values are: true, false, a_leg and b_leg -->
<variable name="process_cdr" value="true"/>
<!-- rtp_secure_media will offer mandatory SRTP on invite AES_CM_128_HMAC_SHA1_32, AES_CM_128_HMAC_SHA1_80 or true-->
<variable name="rtp_secure_media" value="true"/>
<!-- limit the max number of outgoing calls for this user -->
<!--<variable name="max_calls" value="2"/>-->
<!-- send presence information if FS is configured to do so -->
<!--<variable name="presence_id" value="1000@$${domain}"/>-->
<!-- set these to take advantage of a dialplan localized to this user -->
<!--<variable name="numbering_plan" value="US"/>-->
<!--<variable name="default_area_code" value="434"/>-->
<!--<variable name="default_gateway" value="asterlink.com"/>-->
<!--
NDLB-connectile-dysfunction - Rewrite contact ip and port
NDLB-tls-connectile-dysfunction - Rewrite contact port only.
-->
<!--<variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>-->
<!--<variable name="sip-force-expires" value="10"/>-->
<!--<variable name="sip-register-gateway" value="cluecon.com"/>-->
<!-- Set the file format for a specific user -->
<!--<variable name="vm_message_ext" value="mp3"/> -->
</variables>
<vcard>
<!-- insert optional compliant vcard xml here-->
</vcard>
</user>
</include>