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+<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.
+
+ When you hear someone say "sofia profile" this is what they are talking about.
+ -->
+
+ <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
+ <!--aliases are other names that will work as a valid profile name for this profile-->
+ <aliases>
+ <!--
+ <alias name="default"/>
+ -->
+ </aliases>
+ <!-- Outbound Registrations -->
+ <gateways>
+ </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"/>
+ </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"/>-->
+
+ <!--
+ 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.
+ -->
+ <!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
+
+ <!-- Can be set to "_undef_" to remove the User-Agent header -->
+ <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
+
+ <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"/> -->
+ <param name="sip-trace" value="no"/>
+ <param name="sip-capture" value="no"/>
+
+ <!-- Use presence_map.conf.xml to convert extension regex to presence protos for routing -->
+ <!-- <param name="presence-proto-lookup" value="true"/> -->
+
+
+ <!-- Don't be picky about negotiated DTMF just always offer 2833 and accept both 2833 and INFO -->
+ <!--<param name="liberal-dtmf" value="true"/>-->
+
+
+ <!--
+ Sometimes, in extremely rare edge cases, the Sofia SIP stack may stop
+ responding. These options allow you to enable and control a watchdog
+ on the Sofia SIP stack so that if it stops responding for the
+ specified number of milliseconds, it will cause FreeSWITCH to crash
+ immediately. This is useful if you run in an HA environment and
+ need to ensure automated recovery from such a condition. Note that if
+ your server is idle a lot, the watchdog may fire due to not receiving
+ any SIP messages. Thus, if you expect your system to be idle, you
+ should leave the watchdog disabled. It can be toggled on and off
+ through the FreeSWITCH CLI either on an individual profile basis or
+ globally for all profiles. So, if you run in an HA environment with a
+ master and slave, you should use the CLI to make sure the watchdog is
+ only enabled on the master.
+ If such crash occurs, FreeSWITCH will dump core if allowed. The
+ stacktrace will include function watchdog_triggered_abort().
+ -->
+ <param name="watchdog-enabled" value="no"/>
+ <param name="watchdog-step-timeout" value="30000"/>
+ <param name="watchdog-event-timeout" value="30000"/>
+
+ <param name="log-auth-failures" value="false"/>
+ <param name="forward-unsolicited-mwi-notify" 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="$${internal_sip_port}"/>
+ <param name="dialplan" value="XML"/>
+ <param name="dtmf-duration" value="2000"/>
+ <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
+ <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
+ <param name="rtp-timer-name" value="soft"/>
+ <!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
+ <param name="rtp-ip" value="$${local_ip_v4}"/>
+ <!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
+ <param name="sip-ip" value="$${local_ip_v4}"/>
+ <param name="hold-music" value="$${hold_music}"/>
+ <param name="apply-nat-acl" value="nat.auto"/>
+
+
+ <!-- (default true) set to false if you do not wish to have called party info in 1XX responses -->
+ <!-- <param name="cid-in-1xx" value="false"/> -->
+
+ <!-- extended info parsing -->
+ <!-- <param name="extended-info-parsing" value="true"/> -->
+
+ <!--<param name="aggressive-nat-detection" value="true"/>-->
+ <!--
+ There are known issues (asserts and segfaults) when 100rel is enabled.
+ It is not recommended to enable 100rel at this time.
+ -->
+ <!--<param name="enable-100rel" value="true"/>-->
+
+ <!-- uncomment if you don't wish to try a next SRV destination on 503 response -->
+ <!-- RFC3263 Section 4.3 -->
+ <!--<param name="disable-srv503" value="true"/>-->
+
+ <!-- Enable Compact SIP headers. -->
+ <!--<param name="enable-compact-headers" value="true"/>-->
+ <!--
+ enable/disable session timers
+ -->
+ <!--<param name="enable-timer" value="false"/>-->
+ <!--<param name="minimum-session-expires" value="120"/>-->
+ <!-- <param name="apply-inbound-acl" value="domains"/>-->
+ <!--
+ 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"/>-->
+ <!--<param name="dtmf-type" value="info"/>-->
+
+
+ <!-- 'true' means every time 'first-only' means on the first register -->
+ <!--<param name="send-message-query-on-register" value="true"/>-->
+
+ <!-- 'true' means every time 'first-only' means on the first register -->
+ <!--<param name="send-presence-on-register" value="first-only"/> -->
+
+
+ <!-- Caller-ID type (choose one, can be overridden by inbound call type and/or sip_cid_type channel variable -->
+ <!-- Remote-Party-ID header -->
+ <!--<param name="caller-id-type" value="rpid"/>-->
+
+ <!-- P-*-Identity family of headers -->
+ <!--<param name="caller-id-type" value="pid"/>-->
+
+ <!-- neither one -->
+ <!--<param name="caller-id-type" value="none"/>-->
+
+
+
+ <param name="record-path" value="$${recordings_dir}"/>
+ <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
+ <!--enable to use presence -->
+ <param name="manage-presence" value="true"/>
+ <!-- send a presence probe on each register to query devices to send presence instead of sending presence with less info -->
+ <!--<param name="presence-probe-on-register" value="true"/>-->
+ <!--<param name="manage-shared-appearance" value="true"/>-->
+ <!-- used to share presence info across sofia profiles -->
+ <!-- Name of the db to use for this profile -->
+ <!--<param name="dbname" value="share_presence"/>-->
+ <param name="presence-hosts" value="$${domain},$${local_ip_v4}"/>
+ <param name="presence-privacy" value="$${presence_privacy}"/>
+ <!-- ************************************************* -->
+
+ <!-- This setting is for AAL2 bitpacking on G726 -->
+ <!-- <param name="bitpacking" value="aal2"/> -->
+ <!--max number of open dialogs in proceeding -->
+ <!--<param name="max-proceeding" value="1000"/>-->
+ <!--session timers for all call to expire after the specified seconds -->
+ <!--<param name="session-timeout" value="1800"/>-->
+ <!-- Can be 'true' or 'contact' -->
+ <!--<param name="multiple-registrations" value="contact"/>-->
+ <!--set to 'greedy' if you want your codec list to take precedence -->
+ <param name="inbound-codec-negotiation" value="generous"/>
+ <!-- if you want to send any special bind params of your own -->
+ <!--<param name="bind-params" value="transport=udp"/>-->
+ <!--<param name="unregister-on-options-fail" value="true"/>-->
+ <!-- Send an OPTIONS packet to all registered endpoints -->
+ <!--<param name="all-reg-options-ping" value="true"/>-->
+ <!-- Send an OPTIONS packet to NATed registered endpoints. Can be 'true' or 'udp-only'. -->
+ <!--<param name="nat-options-ping" value="true"/>-->
+ <!--<param name="sip-options-respond-503-on-busy" value="true"/>-->
+ <!--<param name="sip-messages-respond-200-ok" value="true"/>-->
+ <!--<param name="sip-subscribe-respond-200-ok" value="true"/>-->
+
+ <!-- TLS: disabled by default, set to "true" to enable -->
+ <param name="tls" value="$${internal_ssl_enable}"/>
+ <!-- Set to true to not bind on the normal sip-port but only on the TLS port -->
+ <param name="tls-only" value="false"/>
+ <!-- additional bind parameters for TLS -->
+ <param name="tls-bind-params" value="transport=tls"/>
+ <!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
+ <param name="tls-sip-port" value="$${internal_tls_port}"/>
+ <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
+ <!--<param name="tls-cert-dir" value=""/>-->
+ <!-- Optionally set the passphrase password used by openSSL to encrypt/decrypt TLS private key files -->
+ <param name="tls-passphrase" value=""/>
+ <!-- Verify the date on TLS certificates -->
+ <param name="tls-verify-date" value="true"/>
+ <!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate -->
+ <!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'subjects_in', 'subjects_out' and 'subjects_all' for subject validation. Multiple policies can be split with a '|' pipe -->
+ <param name="tls-verify-policy" value="none"/>
+ <!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none -->
+ <param name="tls-verify-depth" value="2"/>
+ <!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe -->
+ <param name="tls-verify-in-subjects" value=""/>
+ <!-- TLS version default: tlsv1,tlsv1.1,tlsv1.2 -->
+ <param name="tls-version" value="$${sip_tls_version}"/>
+
+ <!-- TLS ciphers default: ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH -->
+ <param name="tls-ciphers" value="$${sip_tls_ciphers}"/>
+
+ <!-- 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)-->
+ <!--<param name="rtp-autoflush-during-bridge" value="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)-->
+ <!--<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"/>-->
+
+ <!-- Or, if you have PGSQL support, you can use that -->
+ <!--<param name="odbc-dsn" value="pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='' options='-c client_min_messages=NOTICE' application_name='freeswitch'" />-->
+
+ <!--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"/>-->
+
+ <!-- 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 -->
+ <!-- <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-->
+ <!--<param name="disable-transcoding" value="true"/>-->
+ <!-- Handle 302 Redirect in the dialplan -->
+ <!--<param name="manual-redirect" value="true"/> -->
+ <!-- Disable Transfer -->
+ <!--<param name="disable-transfer" value="true"/> -->
+ <!-- Disable Register -->
+ <!--<param name="disable-register" 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"/>-->
+ <!-- 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"/>-->
+ <param name="auth-calls" value="$${internal_auth_calls}"/>
+ <!-- Force the user and auth-user to match. -->
+ <param name="inbound-reg-force-matching-username" value="true"/>
+ <!-- on authed calls, authenticate *all* the packets not just invite -->
+ <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.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="127.0.0.1"/>
+ <param name="ext-sip-ip" value="127.0.0.1"/>
+
+ <!-- rtp inactivity timeout -->
+ <param name="rtp-timeout-sec" value="300"/>
+ <param name="rtp-hold-timeout-sec" value="1800"/>
+ <!-- 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"/>-->
+ <!--
+ 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.
+
+ -->
+ <!--all inbound reg will look in this domain for the users -->
+ <param name="force-register-domain" value="$${domain}"/>
+ <!--force the domain in subscriptions to this value -->
+ <param name="force-subscription-domain" value="$${domain}"/>
+ <!--all inbound reg will stored in the db using this domain -->
+ <param name="force-register-db-domain" value="$${domain}"/>
+
+
+ <!-- for sip over websocket support -->
+ <param name="ws-binding" value=":5066"/>
+
+ <!-- for sip over secure websocket support -->
+ <!-- You need wss.pem in $${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"/>-->
+
+ <!-- launch a new thread to process each new inbound register when using heavier backends -->
+ <!-- <param name="inbound-reg-in-new-thread" value="true"/> -->
+
+ <!-- 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"/>-->
+ <!--<param name="rtcp-video-interval-msec" value="5000"/>-->
+
+ <!--force suscription expires to a lower value than requested-->
+ <!--<param name="force-subscription-expires" value="60"/>-->
+
+ <!-- add a random deviation to the expires value of the 202 Accepted -->
+ <!--<param name="sip-subscription-max-deviation" value="120"/>-->
+
+ <!-- disable register and transfer which may be undesirable in a public switch -->
+ <!--<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
+ -->
+ <!--<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.
+
+ 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"/>
+ <!--<param name="disable-rtp-auto-adjust" value="true"/>-->
+ <!-- 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="true"/>-->
+ <!-- on outbound calls set the callid to match the uuid of the session -->
+ <!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
+ <!-- set to false disable this feature -->
+ <!--<param name="rtp-autofix-timing" value="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"/>-->
+
+ <!-- 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"/>-->
+
+ <!-- the following can be used as workaround with bogus SRV/NAPTR records -->
+ <!--<param name="disable-srv" value="false" />-->
+ <!--<param name="disable-naptr" value="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 -->
+
+ <!-- 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. -->
+ <!-- <param name="timer-T1" value="500" /> -->
+
+ <!-- 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. -->
+ <!-- <param name="timer-T1X64" value="32000" /> -->
+
+
+ <!-- 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. -->
+ <!-- <param name="timer-T2" value="4000" /> -->
+
+ <!--
+ 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"/> -->
+
+
+ <!-- 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.
+ -->
+ <!--<param name="renegotiate-codec-on-hold" value="true"/>-->
+
+ </settings>
+</profile>