sipwitch 1.6.1-1build2 / etc / sipwitch.conf

This file is indexed.

/etc/sipwitch.conf is in sipwitch 1.6.1-1build2.

This file is owned by root:root, with mode 0o644.

The actual contents of the file can be viewed below. 

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<?xml version="1.0"?>
<sipwitch>
<!-- master config file.  The default config can be overriden with a
	 runtime one stored in /var/run/sipwitch which can be installed by
	 a management system.  If one is using a server executed under "user"
	 permissions, then this would be ~/.sipwitchrc.
-->
<provision>
<!-- Allows provisioning to be in main config file as well as scattered. 
     This allows one to produce a single config file that represents the
	 complete phone system.

<refer id="x"></refer>
<alias id="test"><contact>sip:xxx@yyy</contact></alias>
<user id="y"/>
<gateway id="z"/>
-->
</provision>
<access>
<!-- Access rules and cidr definitions.  By default 127.0.0.1/::1 are in
     a pre-generated "loopback" cidr.  Access rule entries are now
     automatically generated by scanning the network interface, so this
     is for special overrides or convenience naming.
  <local>172.16.59.0/24</local>
-->
</access>
<stack>
<!-- The effective names this server processes requests for, and an optional
     list of host or domain names this server will also respond to.  The
     default hostname is always accepted.
  <localnames>sip.gnutelephony.org, server.local, something somewhere</localnames>
-->

<!-- Stack configuration.  Here we restrict all access to the server under
     the local subnet, and we specify the local subnet is "trusted".  Trusted
	 means that challenge digests will be relaxed for devices that are
	 already registered with the server, and hence reduces the total sip
	 traffic needed.  We map for 200 calls, set 2 dispatch threads for
	 sip events, and bind to all interfaces.

  <restricted>local</restricted>
  <trusted>local</trusted>
-->
  <mapped>200</mapped>
  <threading>2</threading>
  <interface>*</interface>
  <dumping>false</dumping>

<!-- peering entry used for setting "proxy" ip address for external users
	 when we are behind a NAT.  This is used for determining ip address for
	 media proxy in particular.  Example entry shown.  Can be ip address or
     resolvable hostname.

	 <peering>www.example.com</peering>
-->

<!-- special user id's.  The "system" id is used when the server creates a
     sip message that is not on behalf of any registered "ua", but rather
	 from the server itself.  For example, when feeding a sms "message"
	 through the control interface, this is generated as a "system" message.
	 Attempts to dial the "system" id will always return SIP FORBIDDEN.
	 
	 The "anon" id is used when anonymous messages are generated.  These
	 always respond with SIP NOT FOUND if one wishes to contact anon.
-->
  <system>system</system>
  <anon>anonymous</anon>
</stack>
<timers>
  <!-- ring every 4 seconds -->
  <ring>4</ring>
  <!-- call forward no answer after x rings -->
  <cfna>4</cfna>
  <!-- call reset to clear cid in stack, 6 seconds -->
  <reset>6</reset>
</timers>
<!-- we have 2xx numbers plus space for external users -->
<registry>
<!-- Registry properties.  We specify support for numeric telephone
     extensions on this machine, for 100 extensions starting at
	 extension 200.  This is useful when sharing a common set of
	 user provisioning records over multiple servers which are routed
	 and segmented.  Hence if I want to call an extension outside of
	 the range of the server I register with, I initially authenticate
	 since this server has the common provisioning, but I then am referred
	 to the actual target server where the destination user is registered.
	 
	 Keysize is used for hash indexing range.  Realm is the realm presented
	 for www authentication, but is normally set uuid or in /etc/siprealm.
-->
  <prefix>200</prefix>
  <range>100</range>
  <keysize>77</keysize>
  <mapped>200</mapped>
  <!-- <realm>GNU Telephony</realm> -->
</registry>

<!-- templates may be used to set default values for automatically
     generated user accounts, such as default forwarding or password if
     not set.
<templates>
 <user>
  <secret>123</secret>
  <forwarding>
   <busy>voicemail</busy>
   <na>voicemail</na>
  </forwarding>
 </user>
 <admin>
  <forwarding>
   <busy>operator</busy>
   <na>operator</na>
  </forwarding>
 </admin>
</templates>
-->

<!-- Routing rules can do all sorts of transforms for dialed numbers.  The
     routing table can also be used to statically redirect ranges of
     extension numbers to alternate servers.  For example, we redirect 1xx
     numbers to a different server with something like:
     <redirect pattern="1xx" server="server.local"/>
     or a range of numbers to a single remote entity uri:
     <redirect pattern="3xx" target="sip:local@myserver.somewhere"/>

     Reject rules can be used to reject with specific error messages, and
     rewrite rules can add or subtract prefix or suffix codes.
-->
<routing>
</routing>
</sipwitch>

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