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Sync up the internal SIP Profile

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FusionPBX 2022-08-06 15:29:18 -06:00 committed by GitHub
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resources/templates/conf/sip_profiles/internal.xml.noload
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@ -1,9 +1,9 @@
<profile name="internal">
<!--
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.
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.
When you hear someone say "sofia profile" this is what they are talking about.
-->
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
@ -27,17 +27,24 @@
</domains>
<settings>
<!-- inject delay between dtmf digits on send to help some slow interpreters (also per channel with rtp_digit_delay var -->
<param name="rtp-digit-delay" value="40" enabled="false"/>
<!--
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" enabled="false"/>
<!--
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" enabled="false"/>
<!-- Can be set to "_undef_" to remove the User-Agent header -->
<param name="user-agent-string" value="FreeSWITCH" enabled="true"/>
<param name="debug" value="0"/>
<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
<param name="shutdown-on-fail" value="true" enabled="false"/>
@ -71,16 +78,14 @@
<param name="watchdog-step-timeout" value="30000"/>
<param name="watchdog-event-timeout" value="30000"/>
<param name="log-auth-failures" value="true"/>
<param name="log-auth-failures" value="false"/>
<param name="forward-unsolicited-mwi-notify" value="false"/>
<param name="track-calls" value="false"/>
<param name="context" value="public"/>
<param name="rfc2833-pt" value="101"/>
<!-- port to bind to for sip traffic -->
<param name="sip-port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="dtmf-type" value="rfc2833"/>
<param name="dtmf-duration" value="2000"/>
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
@ -111,20 +116,21 @@
<!-- Enable Compact SIP headers. -->
<param name="enable-compact-headers" value="true" enabled="false"/>
<!--session timers -->
<param name="session-timeout" value="0" enabled="true"/>
<!--
enable/disable session timers
-->
<param name="enable-timer" value="false" enabled="true"/>
<param name="minimum-session-expires" value="0" enabled="false"/>
<!-- apply inbound acl -->
<param name="apply-inbound-acl" value="domains"/>
<param name="apply-inbound-acl" value="providers"/>
<!--
This defines your local network, by default we detect your local network
and create this localnet.auto ACL for this.
-->
<param name="local-network-acl" value="localnet.auto"/>
<param name="apply-register-acl" value="domains" enabled="false"/>
<param name="apply-register-acl" value="providers" enabled="false"/>
<param name="dtmf-type" value="rfc2833"/>
<!-- 'true' means every time 'first-only' means on the first register -->
<param name="send-message-query-on-register" value="true" enabled="false"/>
@ -152,7 +158,7 @@
<!-- used to share presence info across sofia profiles -->
<!-- Name of the db to use for this profile -->
<param name="dbname" value="share_presence" enabled="false"/>
<param name="presence-hosts" value="$${domain},$${local_ip_v4}" enabled="false"/>
<param name="presence-hosts" value="$${domain},$${local_ip_v4}"/>
<param name="presence-privacy" value="$${presence_privacy}"/>
<!-- ************************************************* -->
@ -160,6 +166,8 @@
<param name="bitpacking" value="aal2" enabled="false"/>
<!--max number of open dialogs in proceeding -->
<param name="max-proceeding" value="1000" enabled="false"/>
<!--max number of receiving requests per second (Default: 1000, 0 - unlimited) -->
<param name="max-recv-requests-per-second" value="0" enabled="false"/>
<!--session timers for all call to expire after the specified seconds -->
<param name="session-timeout" value="1800" enabled="false"/>
<!-- Can be 'true' or 'contact' -->
@ -170,6 +178,14 @@
<param name="bind-params" value="transport=udp" enabled="false"/>
<param name="unregister-on-options-fail" value="true" enabled="false"/>
<!-- Send an OPTIONS packet to all registered endpoints -->
<!--<param name="all-reg-options-ping" value="true"/>-->
<!-- Send an OPTIONS packet to NATed registered endpoints. Can be 'true' or 'udp-only'. -->
<!--<param name="nat-options-ping" value="true"/>-->
<!--<param name="sip-options-respond-503-on-busy" value="true"/>-->
<!--<param name="sip-messages-respond-200-ok" value="true"/>-->
<!--<param name="sip-subscribe-respond-200-ok" value="true"/>-->
<!-- TLS: disabled by default, set to "true" to enable -->
<param name="tls" value="$${internal_ssl_enable}"/>
<!-- Set to true to not bind on the normal sip-port but only on the TLS port -->
@ -185,17 +201,28 @@
<!-- Verify the date on TLS certificates -->
<param name="tls-verify-date" value="false"/>
<!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate -->
<!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'in_subjects', 'out_subjects' and 'all_subjects' for subject validation. Multiple policies can be split with a '|' pipe -->
<param name="tls-verify-policy" value="all|subjects_all" enabled="false"/>
<!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'subjects_in', 'subjects_out' and 'subjects_all' for subject validation. Multiple policies can be split with a '|' pipe -->
<param name="tls-verify-policy" value="none"/>
<!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none -->
<param name="tls-verify-depth" value="2"/>
<!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe -->
<param name="tls-verify-in-subjects" value=""/>
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
<!-- TLS version default: tlsv1,tlsv1.1,tlsv1.2 -->
<param name="tls-version" value="$${sip_tls_version}"/>
<!-- TLS ciphers default: ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH -->
<param name="tls-ciphers" value="$${sip_tls_ciphers}"/>
<!--
Connect timeout for outgoing requests using TLS (in milliseconds).
Set the timeout and SIP engine will try again sending an outgoing request
and when possible - using an alternative address (DNS failover).
Default - 0 (disabled)
-->
<param name="tls-orq-connect-timeout" value="3000" enabled="false"/>
<!-- turn on auto-flush during bridge (skip timer sleep when the socket already has data)
(reduces delay on latent connections default true, must be disabled explicitly)-->
(reduces delay on latent connections default true, must be disabled explicitly)-->
<param name="rtp-autoflush-during-bridge" value="false" enabled="false"/>
<!--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)-->
@ -210,8 +237,11 @@
<!--Uncomment to set all inbound calls to proxy media mode-->
<param name="inbound-proxy-media" value="true" enabled="false"/>
<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
<param name="inbound-late-negotiation" value="true" enabled="false"/>
<!-- Let calls hit the dialplan before selecting codec for the a-leg -->
<param name="inbound-late-negotiation" value="true"/>
<!-- Allow ZRTP clients to negotiate end-to-end security associations (also enables late negotiation) -->
<param name="inbound-zrtp-passthru" value="true"/>
<!-- this lets anything register -->
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
@ -239,6 +269,7 @@
<!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
<param name="NDLB-received-in-nat-reg-contact" value="true" enabled="false"/>
<param name="auth-calls" value="true"/>
<!-- Force subscription requests to require authentication -->
<param name="auth-subscriptions" value="true"/>
<!-- Force the user and auth-user to match. -->
<param name="inbound-reg-force-matching-username" value="true"/>
@ -246,13 +277,13 @@
<param name="auth-all-packets" value="false"/>
<!-- external_sip_ip
Used as the public IP address for SDP.
Can be an one of:
ip address - "12.34.56.78"
a stun server lookup - "stun:stun.freeswitch.org"
a DNS name - "host:host.server.com"
auto - Use guessed ip.
auto-nat - Use ip learned from NAT-PMP or UPNP
Used as the public IP address for SDP.
Can be an one of:
ip address - "12.34.56.78"
a stun server lookup - "stun:stun.server.com"
a DNS name - "host:host.server.com"
auto - Use guessed ip.
auto-nat - Use ip learned from NAT-PMP or UPNP
-->
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
@ -263,15 +294,10 @@
<param name="media_timeout" value="300" enabled="true"/>
<param name="media_hold_timeout" value="1800" enabled="true"/>
<!-- VAD choose one (out is a good choice); -->
<!-- <param name="vad" value="in" enabled="false"/> -->
<!--<param name="vad" value="in"/>-->
<param name="vad" value="out" enabled="false"/>
<!-- <param name="vad" value="both" enabled="false"/> -->
<!--<param name="vad" value="both"/>-->
<param name="alias" value="sip:10.0.1.251:5555" enabled="false"/>
<!-- sip expire settings -->
<param name="sip-expires-max-deviation" value="600" enabled="false"/>
<param name="sip-force-expires" value="1800" enabled="false"/>
<!--
These are enabled to make the default config work better out of the box.
If you need more than ONE domain you'll need to not use these options.
@ -283,14 +309,28 @@
<!--all inbound reg will stored in the db using this domain -->
<param name="force-register-db-domain" value="$${domain}" enabled="false"/>
<!-- for sip over websocket support -->
<param name="ws-binding" value=":5066"/>
<!-- for sip over secure websocket support -->
<!-- You need wss.pem in $${certs_dir} for wss or one will be created for you -->
<param name="wss-binding" value=":7443"/>
<param name="delete-subs-on-register" value="false" enabled="false"/>
<!-- launch a new thread to process each new inbound register when using heavier backends -->
<param name="inbound-reg-in-new-thread" value="true" enabled="false"/>
<!-- enable rtcp on every channel also can be done per leg basis with rtcp_audio_interval_msec variable set to passthru to pass it across a call-->
<param name="rtcp-audio-interval-msec" value="5000" enabled="false"/>
<param name="rtcp-video-interval-msec" value="5000" enabled="false"/>
<!--force suscription expires to a lower value than requested-->
<param name="force-subscription-expires" value="60" enabled="false"/>
<!-- add a random deviation to the expires value of the 202 Accepted -->
<param name="sip-subscription-max-deviation" value="120" enabled="false"/>
<!-- disable register and transfer which may be undesirable in a public switch -->
<param name="disable-transfer" value="true" enabled="false"/>
<param name="disable-register" value="true" enabled="false"/>
@ -303,6 +343,7 @@
<!-- use at your own risk or if you know what this does.-->
<param name="NDLB-force-rport" value="safe" enabled="true"/>
<!--
Choose the realm challenge key. Default is auto_to if not set.
@ -312,28 +353,30 @@
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.
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_to"/>
<param name="disable-rtp-auto-adjust" value="true" enabled="false"/>
<!-- on inbound calls make the uuid of the session equal to the sip call id of that call -->
<!-- <param name="inbound-use-callid-as-uuid" value="false" enabled="false"/> -->
<param name="inbound-use-callid-as-uuid" value="false" enabled="false"/>
<!-- on outbound calls set the callid to match the uuid of the session -->
<!-- <param name="outbound-use-uuid-as-callid" value="false" enabled="false"/> -->
<param name="outbound-use-uuid-as-callid" value="false" enabled="false"/>
<!-- set to false disable this feature -->
<param name="rtp-autofix-timing" value="false" enabled="false"/>
<!-- set this param to false if your gateway for some reason hates X- headers that it is supposed to ignore-->
<param name="pass-callee-id" value="false" enabled="false"/>
<!-- clear clears them all or supply the name to add or the name prefixed with ~ to remove
valid values:
clear
CISCO_SKIP_MARK_BIT_2833
SONUS_SEND_INVALID_TIMESTAMP_2833
<!-- clear clears them all or supply the name to add or the name
prefixed with ~ to remove valid values:
clear
CISCO_SKIP_MARK_BIT_2833
SONUS_SEND_INVALID_TIMESTAMP_2833
-->
<param name="auto-rtp-bugs" data="clear" enabled="false"/>
@ -342,47 +385,58 @@
<param name="disable-naptr" value="false" enabled="false"/>
<!-- The following can be used to fine-tune timers within sofia's transport layer
Those settings are for advanced users and can safely be left as-is -->
Those settings are for advanced users and can safely be left as-is -->
<!-- Initial retransmission interval (in milliseconds).
Set the T1 retransmission interval used by the SIP transaction engine.
The T1 is the initial duration used by request retransmission timers A and E (UDP) as well as response retransmission timer G. -->
Set the T1 retransmission interval used by the SIP transaction engine.
The T1 is the initial duration used by request retransmission timers A and E (UDP) as well as response retransmission timer G. -->
<param name="timer-T1" value="500" enabled="false"/>
<!-- Transaction timeout (defaults to T1 * 64).
Set the T1x64 timeout value used by the SIP transaction engine.
The T1x64 is duration used for timers B, F, H, and J (UDP) by the SIP transaction engine.
The timeout value T1x64 can be adjusted separately from the initial retransmission interval T1. -->
Set the T1x64 timeout value used by the SIP transaction engine.
The T1x64 is duration used for timers B, F, H, and J (UDP) by the SIP transaction engine.
The timeout value T1x64 can be adjusted separately from the initial retransmission interval T1. -->
<param name="timer-T1X64" value="32000" enabled="false"/>
<!-- Maximum retransmission interval (in milliseconds).
Set the maximum retransmission interval used by the SIP transaction engine.
The T2 is the maximum duration used for the timers E (UDP) and G by the SIP transaction engine.
Note that the timer A is not capped by T2. Retransmission interval of INVITE requests grows exponentially
until the timer B fires. -->
Set the maximum retransmission interval used by the SIP transaction engine.
The T2 is the maximum duration used for the timers E (UDP) and G by the SIP transaction engine.
Note that the timer A is not capped by T2. Retransmission interval of INVITE requests grows exponentially
until the timer B fires. -->
<param name="timer-T2" value="4000" enabled="false"/>
<!--
Transaction lifetime (in milliseconds).
Set the lifetime for completed transactions used by the SIP transaction engine.
A completed transaction is kept around for the duration of T4 in order to catch late responses.
The T4 is the maximum duration for the messages to stay in the network and the duration of SIP timer K. -->
<param name="timer-T4" value="4000" enabled="false"/>
Transaction lifetime (in milliseconds).
Set the lifetime for completed transactions used by the SIP transaction engine.
A completed transaction is kept around for the duration of T4 in order to catch late responses.
The T4 is the maximum duration for the messages to stay in the network and the duration of SIP timer K. -->
<!-- <param name="timer-T4" value="4000" /> -->
<!-- Turn on a jitterbuffer for every call -->
<param name="auto-jitterbuffer-msec" value="60" enabled="false"/>
<!-- By default mod_sofia will ignore the codecs in the sdp for hold/unhold operations
Set this to true if you want to actually parse the sdp and re-negotiate the codec during hold/unhold.
It's probably not what you want so stick with the default unless you really need to change this.
Set this to true if you want to actually parse the sdp and re-negotiate the codec during hold/unhold.
It's probably not what you want so stick with the default unless you really need to change this.
-->
<param name="renegotiate-codec-on-hold" value="true" enabled="false"/>
<!-- Periodically send an packet to all registered endpoints that are behind NAT -->
<param name="registration-thread-frequency" value="30" enabled="false"/>
<param name="nat-options-ping" value="true" enabled="false"/>
<!-- By default mod_sofia will send "100 Trying" in response to a SIP INVITE. Set this to false if
you want to turn off this behavior and manually send the "100 Trying" via the acknowledge_call application.
-->
<!--<param name="auto-invite-100" value="false"/>-->
<!-- Secure Web Sockets -->
<param name="wss-binding" value=":7443" enabled="false"/>
<!-- sip expire settings -->
<param name="sip-expires-max-deviation" value="600" enabled="false"/>
<param name="sip-force-expires" value="1800" enabled="false"/>
<!-- save session in the database for option to restore SIP UDP calls -->
<param name="track-calls" value="false"/>
<!--number default IREG_SECONDS 30-->
<param name="registration-thread-frequency" value="30" enabled="false"/>
<!-- Periodically send an packet to all registered endpoints that are behind NAT -->
<param name="nat-options-ping" value="true" enabled="false"/>
</settings>
</profile>