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whitespace cleanup

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This commit is contained in:
Travis Cross
2012-05-27 05:44:14 +00:00
parent 9b569ec875
commit 73614127fc
15 changed files with 227 additions and 235 deletions
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14
conf/sbc/sbc_profiles/external.xml
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@@ -1,4 +1,4 @@
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<profile name="external">
<!-- This profile is only for outbound registrations to providers -->
<gateways>
@@ -7,7 +7,7 @@
<aliases>
<alias name="outbound"/>
<alias name="nat"/> <!-- for backwards compatibility -->
<alias name="nat"/> <!-- for backwards compatibility -->
</aliases>
<domains>
@@ -30,10 +30,10 @@
<!-- This could be set to "passive" -->
<param name="manage-presence" value="false"/>
<!-- used to share presence info across sofia profiles
manage-presence needs to be set to passive on this profile
if you want it to behave as if it were the internal profile
for presence.
<!-- used to share presence info across sofia profiles
manage-presence needs to be set to passive on this profile
if you want it to behave as if it were the internal profile
for presence.
-->
<!-- Name of the db to use for this profile -->
<!--<param name="dbname" value="share_presence"/>-->
@@ -49,7 +49,7 @@
<param name="auth-calls" value="false"/>
<param name="rtp-timeout-sec" value="1800"/>
<!--
DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
-->
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>

2
conf/sbc/sbc_profiles/external/example.xml vendored
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@@ -9,7 +9,7 @@
<!--/// 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"/>-->
<!--<param name="password" value="2007"/>-->
<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
<!--<param name="extension" value="cluecon"/>-->
<!--/// proxy host: *optional* same as realm, if blank ///-->

17
conf/sbc/sbc_profiles/internal-ipv6.xml
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@@ -53,36 +53,36 @@
<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
<param name="tls-version" value="$${sip_tls_version}"/>
<!--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-timestamps" value="true"/>-->
<!--<param name="pass-rfc2833" 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-bypass-media" value="true"/>-->
<!--Uncomment to set all inbound calls to proxy media mode-->
<!--<param name="inbound-proxy-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"/> -->
<!-- accept any authentication without actually checking (not a good feature for most people) -->
<!-- <param name="accept-blind-auth" value="true"/> -->
<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
<!-- <param name="suppress-cng" 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-->
<!--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"/>-->
<!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
<!--<param name="NDLB-broken-auth-hash" value="true"/>-->
@@ -128,4 +128,3 @@
</settings>
</profile>

61
conf/sbc/sbc_profiles/internal.xml
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@@ -1,7 +1,7 @@
<!--
This is a sofia sip profile/user agent. This will service exactly one ip and port.
In FreeSWITCH you can run multiple sip user agents on their own ip and port.
When you hear someone say "sofia profile" this is what they are talking about.
-->
@@ -15,24 +15,24 @@
<gateways>
<X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
</gateways>
<domains>
<!-- indicator to parse the directory for domains with parse="true" to get gateways-->
<!--<domain name="$${domain}" parse="true"/>-->
<!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
<!--<domain name="all" alias="true" parse="true"/>-->
<domain name="all" alias="true" parse="false"/>
<domain name="all" alias="true" parse="false"/>
</domains>
<settings>
<!--
When calls are in no media this will bring them back to media
when you press the hold button.
When calls are in no media this will bring them back to media
when you press the hold button.
-->
<!--<param name="media-option" value="resume-media-on-hold"/> -->
<!--
This will allow a call after an attended transfer go back to
bypass media after an attended transfer.
This will allow a call after an attended transfer go back to
bypass media after an attended transfer.
-->
<!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
@@ -69,7 +69,7 @@
<!--<param name="dbname" value="share_presence"/>-->
<!--<param name="presence-hosts" value="$${domain}"/>-->
<!-- ************************************************* -->
<!-- This setting is for AAL2 bitpacking on G726 -->
<!-- <param name="bitpacking" value="aal2"/> -->
<!--max number of open dialogs in proceeding -->
@@ -94,36 +94,36 @@
<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
<param name="tls-version" value="$${sip_tls_version}"/>
<!--If you don't want to pass through timestamps from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
<!--<param name="rtp-rewrite-timestamps" value="true"/>-->
<!--<param name="pass-rfc2833" 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-bypass-media" value="true"/>-->
<!--Uncomment to set all inbound calls to proxy media mode-->
<!--<param name="inbound-proxy-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"/> -->
<!-- accept any authentication without actually checking (not a good feature for most people) -->
<!-- <param name="accept-blind-auth" value="true"/> -->
<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
<!-- <param name="suppress-cng" 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-->
<!--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"/>-->
<!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
<!--<param name="NDLB-broken-auth-hash" value="true"/>-->
@@ -154,24 +154,24 @@
<!--<param name="disable-transfer" value="true"/>-->
<!--<param name="disable-register" value="true"/>-->
<!-- enable-3pcc can be set to either 'true' or 'proxy', true accepts the call right away, proxy waits until the call has been answered then sends accepts -->
<!-- enable-3pcc can be set to either 'true' or 'proxy', true accepts the call right away, proxy waits until the call has been answered then sends accepts -->
<!--<param name="enable-3pcc" value="true"/>-->
<!-- use at your own risk or if you know what this does.-->
<!--<param name="NDLB-force-rport" value="true"/>-->
<!--
Choose the realm challenge key. Default is auto_to if not set.
auto_from - uses the from field as the value for the sip realm.
auto_to - uses the to field as the value for the sip realm.
<anyvalue> - you can input any value to use for the sip realm.
Choose the realm challenge key. Default is auto_to if not set.
If you want URL dialing to work you'll want to set this to auto_from.
If you use any other value besides auto_to or auto_from you'll loose
the ability to do multiple domains.
Note: comment out to restore the behavior before 2008-09-29
auto_from - uses the from field as the value for the sip realm.
auto_to - uses the to field as the value for the sip realm.
<anyvalue> - you can input any value to use for the sip realm.
If you want URL dialing to work you'll want to set this to auto_from.
If you use any other value besides auto_to or auto_from you'll loose
the ability to do multiple domains.
Note: comment out to restore the behavior before 2008-09-29
-->
<param name="challenge-realm" value="auto_from"/>
@@ -182,4 +182,3 @@
<!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
</settings>
</profile>

2
conf/sbc/sbc_profiles/internal/example.xml
View File

@@ -9,7 +9,7 @@
<!--/// 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"/>-->
<!--<param name="password" value="2007"/>-->
<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
<!--<param name="extension" value="cluecon"/>-->
<!--/// proxy host: *optional* same as realm, if blank ///-->