hi guys,
I have a DID administered on my FS
but I'm not able to get any info about any calls landing on the DID on the FS
console
is this possible?
if yes, how?
attached are my config files
please let me know what I am doing wrong at the earliest
The SIP provider is sip.net4india.com
DID number: 14085121934
Please respond at the earliest
Regards,
Gayatri Kulkarni
-----
Whenever you find yourself on the side of the majority, it is time to pause and
reflect.
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
<include>
<context name="public">
<extension name="unloop">
<condition field="${unroll_loops}" expression="^true$"/>
<condition field="${sip_looped_call}" expression="^true$">
<action application="deflect" data="${destination_number}"/>
</condition>
</extension>
<extension name="public_extensions">
<condition field="destination_number" expression="^(10[01][0-9])$">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
<extension name="public_did">
<condition field="destination_number" expression="14085121934">
<action application="bridge" data="1002 XML default"/>
</condition>
</extension>
<extension name="net4india">
<condition field="destination_number" expression="14085121934">
<action application="transfer" data="$1 XML default"/>
</condition>
<extension name="14085121934">
<condition field="destination_number" expression="14085121934">
<action application="transfer" data="1200"/>
</condition>
</extension>
</extension>
<extension name="deltathree">
<condition field="destination_number" expression="14084636126">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
<extension name="1200">
<condition field="destination_number" expression="^1200$">
<action application="javascript" data="readFile.js"/>
</condition>
</extension>
</context>
</include>
<?xml version="1.0" encoding="utf-8"?>
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
<include>
<context name="default">
<extension name="unloop">
<condition field="${unroll_loops}" expression="^true$"/>
<condition field="${sip_looped_call}" expression="^true$">
<action application="deflect" data="${destination_number}"/>
</condition>
</extension>
<!-- Example of doing things based on time of day. -->
<extension name="tod_example" continue="true">
<condition field="${strftime(%H%M)}" expression="^((09|1[0-7])[0-5][0-9]|1800)$">
<action application="set" data="open=true"/>
</condition>
</extension>
<extension name="intercept">
<condition field="destination_number" expression="^886$">
<action application="answer"/>
<action application="intercept" data="${db(select/last_dial/global)}"/>
<action application="sleep" data="2000"/>
</condition>
</extension>
<extension name="intercept-ext">
<condition field="destination_number" expression="^\*\*(\d+)$">
<action application="answer"/>
<action application="intercept" data="${db(select/last_dial_ext/$1)}"/>
<action application="sleep" data="2000"/>
</condition>
</extension>
<extension name="redial">
<condition field="destination_number" expression="^870$">
<action application="transfer" data="${db(select/last_dial/${caller_id_number})}"/>
</condition>
</extension>
<extension name="global" continue="true">
<condition field="${network_addr}" expression="^$" break="never">
<action application="set" data="use_profile=${cond(${acl($${local_ip_v4} rfc1918)} == true ? nat : default)}"/>
<anti-action application="set" data="use_profile=${cond(${acl(${network_addr} rfc1918)} == true ? nat : default)}"/>
</condition>
<!-- This will setup some variables if the user isn't authenticated.
numbering_plan is required for the demo to function properly.
-->
<condition field="${numbering_plan}" expression="^$" break="never">
<action application="set_user" data="[EMAIL PROTECTED]"/>
</condition>
<condition field="${call_debug}" expression="^true$" break="never">
<action application="info"/>
</condition>
<condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never">
<action application="set" data="sip_secure_media=true"/>
<!-- Offer SRTP on outbound legs if we have it on inbound. -->
<!-- <action application="export" data="sip_secure_media=true"/> -->
</condition>
<condition>
<action application="db" data="insert/spymap/${caller_id_number}/${uuid}"/>
<action application="db" data="insert/last_dial/${caller_id_number}/${destination_number}"/>
<action application="db" data="insert/last_dial/global/${uuid}"/>
</condition>
</extension>
<!--
snom button demo, call 9000 to make button 2 mapped to transfer the current call to a conference
-->
<extension name="snom-demo-2">
<condition field="destination_number" expression="^9001$">
<action application="eval" data="${snom_bind_key(2 off DND ${sip_from_user} ${sip_from_host} ${sofia_profile_name} message notused)}"/>
<action application="transfer" data="3000"/>
</condition>
</extension>
<extension name="snom-demo-1">
<condition field="destination_number" expression="^9000$">
<!--<key> <light> <label> <user> <host> <profile> <action_name> <action>-->
<action application="eval" data="${snom_bind_key(2 on DND ${sip_from_user} ${sip_from_host} ${sofia_profile_name} message api+uuid_transfer ${uuid} 9001)}"/>
<action application="playback" data="$${hold_music}"/>
</condition>
</extension>
<extension name="eavesdrop">
<condition field="destination_number" expression="^88(.*)$|^\*0(.*)$">
<action application="answer"/>
<action application="eavesdrop" data="${db(select/spymap/$1)}"/>
</condition>
</extension>
<extension name="eavesdrop">
<condition field="destination_number" expression="^779$">
<action application="answer"/>
<action application="set" data="eavesdrop_indicate_failed=tone_stream://%(500, 0, 320)"/>
<action application="set" data="eavesdrop_indicate_new=tone_stream://%(500, 0, 620)"/>
<action application="set" data="eavesdrop_indicate_idle=tone_stream://%(250, 0, 920)"/>
<action application="eavesdrop" data="all"/>
</condition>
</extension>
<extension name="call_return">
<condition field="destination_number" expression="^\*69$|^869$">
<action application="transfer" data="${db(select/call_return/${caller_id_number})}"/>
</condition>
</extension>
<extension name="del-group">
<condition field="destination_number" expression="^80(\d{2})$">
<action application="answer"/>
<action application="group" data="delete:$1:${sofia_contact([EMAIL PROTECTED])}"/>
<action application="gentones" data="%(1000, 0, 320)"/>
</condition>
</extension>
<extension name="add-group">
<condition field="destination_number" expression="^81(\d{2})$">
<action application="answer"/>
<action application="group" data="insert:$1:${sofia_contact([EMAIL PROTECTED])}"/>
<action application="gentones" data="%(1000, 0, 640)"/>
</condition>
</extension>
<extension name="call-group-simo">
<condition field="destination_number" expression="^82(\d{2})$">
<action application="bridge" data="{ignore_early_media=true}${group(call:$1)}"/>
</condition>
</extension>
<extension name="call-group-order">
<condition field="destination_number" expression="^83(\d{2})$">
<action application="set" data="call_timeout=10"/>
<action application="bridge" data="{ignore_early_media=true}${group(call:$1:order)}"/>
</condition>
</extension>
<extension name="extension-intercom">
<!-- <condition field="${sip_to_params}" expression="intercom\=true"/> -->
<condition field="destination_number" expression="^8(10[01][0-9])$">
<action application="set" data="dialed_ext=$1"/>
<!-- This Alert-Info seems to be a case for Intercom for Polycom which sip_auto_answer=true covers already. -->
<!--<action application="export"><![CDATA[alert_info=<sip:$${domain}>;Ring;Answer]]></action>-->
<action application="export"><![CDATA[sip_h_Call-Info=<sip:$${domain}>;answer-after=0]]></action>
<action application="export" data="sip_invite_params=intercom=true"/>
<action application="export" data="sip_auto_answer=true"/>
<action application="bridge" data="user/[EMAIL PROTECTED]"/>
</condition>
</extension>
<!--
if the calling party is the called party, go to their VM
if the calling party is NOT the called party dial the extension
(1000-1019) for 30 seconds and go to voicemail if the
call fails (continue_on_fail=true), otherwise hang up after a
successful bridge (hangup_after-bridge=true)
-->
<extension name="Local_Extension">
<condition field="destination_number" expression="^(10[01][0-9])$">
<action application="set" data="dialed_ext=$1"/>
<action application="export" data="dialed_ext=$1"/>
</condition>
<condition field="destination_number" expression="^${caller_id_number}$">
<action application="set" data="voicemail_authorized=${sip_authorized}"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="voicemail" data="check default $${domain} ${dialed_ext}"/>
<!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
<anti-action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
<anti-action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<anti-action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
<anti-action application="set" data="transfer_ringback=${us-ring}"/>
<anti-action application="set" data="call_timeout=30"/>
<!-- <anti-action application="set" data="sip_exclude_contact=${network_addr}"/> -->
<anti-action application="set" data="hangup_after_bridge=true"/>
<!--<anti-action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
<anti-action application="set" data="continue_on_fail=true"/>
<anti-action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/>
<anti-action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/>
<anti-action application="bridge" data="user/[EMAIL PROTECTED]"/>
<anti-action application="answer"/>
<!--<anti-action application="send_display" data="Voicemail for ${dialed_ext}"/>-->
<anti-action application="sleep" data="1000"/>
<anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/>
</condition>
</extension>
<!-- dial via SIP uri -->
<extension name="sip_uri">
<condition field="destination_number" expression="^sip:(.*)$">
<action application="bridge" data="sofia/${use_profile}/$1"/>
</condition>
</extension>
<!--
start a dynamic conference with the settings of the "default" conference profile in conference.conf.xml
-->
<extension name="nb_conferences">
<condition field="destination_number" expression="^(30\d{2})$">
<action application="answer"/>
<!--<action application="send_display" data="8k Conference $1"/>-->
<action application="conference" data="[EMAIL PROTECTED]"/>
</condition>
</extension>
<extension name="wb_conferences">
<condition field="destination_number" expression="^(31\d{2})$">
<action application="answer"/>
<!--<action application="send_display" data="16k Conference $1"/>-->
<action application="conference" data="[EMAIL PROTECTED]"/>
</condition>
</extension>
<extension name="uwb_conferences">
<condition field="destination_number" expression="^(32\d{2})$">
<action application="answer"/>
<!--<action application="send_display" data="32k Conference $1"/>-->
<action application="conference" data="[EMAIL PROTECTED]"/>
</condition>
</extension>
<!-- dial the freeswitch conference via SIP-->
<extension name="freeswitch_public_conf_via_sip">
<condition field="destination_number" expression="^9(888|1616)$">
<action application="bridge" data="sofia/${use_profile}/[EMAIL PROTECTED]"/>
</condition>
</extension>
<!-- a sample IVR -->
<extension name="ivr_demo">
<condition field="destination_number" expression="^5000$">
<action application="answer"/>
<action application="sleep" data="2000"/>
<action application="ivr" data="demo_ivr"/>
</condition>
</extension>
<extension name="rtp_multicast_page">
<condition field="destination_number" expression="^pagegroup$|^7243">
<action application="answer"/>
<!--<action application="send_display" data="Multicast Page"/>-->
<action application="esf_page_group"/>
</condition>
</extension>
<!--
Parking extensions... transferring calls to 5900 will park them in a queue.
-->
<extension name="park">
<condition field="destination_number" expression="^5900$">
<action application="set" data="fifo_music=$${hold_music}"/>
<action application="fifo" data="[EMAIL PROTECTED] in"/>
</condition>
</extension>
<!--
Parking pickup extension. Calling 5901 will pickup the call.
-->
<extension name="unpark">
<condition field="destination_number" expression="^5901$">
<action application="answer"/>
<action application="fifo" data="[EMAIL PROTECTED] out nowait"/>
</condition>
</extension>
<!--
This extension is used with snom phones.
Set a function key to park+lot (lot being a number or name.)
Set type to Park+Orbit. You can then park and pickup using
the softkey on the phone. Should work with other phones.
-->
<extension name="park">
<condition field="source" expression="mod_sofia"/>
<condition field="destination_number" expression="park\+(\d+)">
<action application="fifo" data="[EMAIL PROTECTED] in undef $${hold_music}"/>
</condition>
</extension>
<!--
The extension is parking pickup with a to param of the fifo we are calling
Some phones send things like orbit= and you can extract that info.
-->
<extension name="unpark">
<condition field="source" expression="mod_sofia"/>
<condition field="destination_number" expression="^parking$"/>
<condition field="${sip_to_params}" expression="fifo\=(\d+)">
<action application="answer"/>
<action application="fifo" data="[EMAIL PROTECTED] out nowait"/>
</condition>
</extension>
<extension name="show_info">
<condition field="destination_number" expression="^9992$">
<action application="answer"/>
<action application="info"/>
<action application="sleep" data="250"/>
<action application="hangup"/>
</condition>
</extension>
<extension name="video_record">
<condition field="destination_number" expression="^9993$">
<action application="answer"/>
<action application="record_fsv" data="/tmp/testrecord.fsv"/>
</condition>
</extension>
<extension name="video_playback">
<condition field="destination_number" expression="^9994$">
<action application="answer"/>
<action application="play_fsv" data="/tmp/testrecord.fsv"/>
</condition>
</extension>
<extension name="delay_echo">
<condition field="destination_number" expression="^9995$">
<action application="answer"/>
<!--<action application="send_display" data="Delay Echo Test (5 sec)"/>-->
<action application="delay_echo" data="5000"/>
</condition>
</extension>
<extension name="echo">
<condition field="destination_number" expression="^9996$">
<action application="answer"/>
<!--<action application="send_display" data="Echo Test"/>-->
<action application="echo"/>
</condition>
</extension>
<extension name="milliwatt">
<condition field="destination_number" expression="^9997$">
<action application="answer"/>
<!--<action application="send_display" data="Milliwatt Test"/>-->
<action application="playback" data="tone_stream://%(10000,0,1004);loops=-1"/>
</condition>
</extension>
<extension name="tone_stream">
<condition field="destination_number" expression="^9998$">
<action application="answer"/>
<action application="playback" data="tone_stream://path=${base_dir}/conf/tetris.ttml;loops=10"/>
</condition>
</extension>
<extension name="hold_music">
<condition field="destination_number" expression="^9999$"/>
<condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$">
<action application="answer"/>
<action application="execute_extension" data="is_secure XML features"/>
<action application="playback" data="$${hold_music}"/>
<anti-action application="answer"/>
<!--<anti-action application="send_display" data="Insecure Music On Hold"/>-->
<anti-action application="playback" data="$${hold_music}"/>
</condition>
</extension>
<X-PRE-PROCESS cmd="include" data="extensions/*.xml"/>
<!--
<extension name="refer">
<condition field="${sip_refer_to}">
<expression><![CDATA[<sip:[EMAIL PROTECTED]>]]></expression>
</condition>
<condition field="${sip_refer_to}">
<expression><![CDATA[<sip:(.*)@(.*)>]]></expression>
<action application="set" data="refer_user=$1"/>
<action application="set" data="refer_domain=$2"/>
<action application="info"/>
<action application="bridge" data="sofia/${use_profile}/[EMAIL PROTECTED]"/>
</condition>
</extension>
-->
<!--
This is an example of how to overide the RURI on an outgoing invite to a registered contact.
-->
<!--
<extension name="ruri">
<condition field="destination_number" expression="^ruri$">
<action application="bridge" data="sofia/${ruri_profile}/${ruri_user}${regex(${sofia_contact(${ruri_contact})}|[EMAIL PROTECTED](.*)|%1)}"/>
</condition>
</extension>
<extension name="7004">
<condition field="destination_number" expression="^7004$">
<action application="set" data="ruri_profile=default"/>
<action application="set" data="ruri_user=2000"/>
<action application="set" data="[EMAIL PROTECTED]"/>
<action application="execute_extension" data="ruri"/>
</condition>
</extension>
-->
<!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
<extension name="net4india">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<action application="set" data="effective_caller_id_number=$1"/>
<!-- If your provider does not provide ringback (180 or 183) you may simulate
ringback by uncommenting the following line. -->
<action application="ringback" />
<action application="bridge" data="sofia/gateway/net4india/14085121934"/>
</condition>
</extension>
<extension name="deltathree">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<action application="set" data="effective_caller_id_number=$1"/>
<!-- If your provider does not provide ringback (180 or 183) you may simulate
ringback by uncommenting the following line. -->
<action application="ringback" />
<action application="bridge" data="sofia/gateway/deltathree/$1"/>
</condition>
</extension>
<extension name="14085121934">
<condition field="destination_number" expression="14085121934">
<action application="bridge" data="sofia/inetrnal/1002"/>
</condition>
</extension>
<extension name="1200">
<condition field="destination_number" expression="^1200$">
<action application="javascript" data="readFile.js"/>
</condition>
</extension>
<extension name="10000">
<condition field="destination_number" expression="^10000$">
<action application="set" data="call_timeout=10"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER, NO_ROUTE_DESTINATION, INVALID_NUMBER_FORMAT, DESTINATION_OUT_OF_ORDER"/>
<!--action application="bridge" data="sofia/internal/scot.clausing"/-->
<action application="bridge" data="sofia/gateway/net4india/14085121934"/>
</condition>
</extension>
</context>
</include>
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<profile name="external">
<!-- This profile is only for outbound registrations to providers -->
<gateways>
<X-PRE-PROCESS cmd="include" data="external/*.xml"/>
</gateways>
<aliases>
<alias name="outbound"/>
</aliases>
<domains>
<domain name="$${domain}" parse="true"/>
<domain name="sip.net4india.com" parse="true"/>
</domains>
<settings>
<param name="debug" value="0"/>
<param name="sip-trace" value="no"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5080"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="manage-presence" value="false"/>
<param name="aggressive-nat-detection" value="true"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="false"/>
<param name="rtp-timeout-sec" value="1800"/>
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
<!--<param name="enable-3pcc" value="true"/>-->
</settings>
</profile>
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<profile name="internal" domain="$${domain}">
<!--aliases are other names that will work as a valid profile name for this profile-->
<aliases>
<alias name="$${domain}"/>
<alias name="default"/>
</aliases>
<!-- Outbound Registrations -->
<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="$${domain}" parse="true"/>-->
<!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
<!--<domain name="all" alias="true" parse="true"/>-->
</domains>
<settings>
<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
<param name="debug" value="0"/>
<param name="sip-trace" value="no"/>
<param name="context" value="public"/>
<param name="rfc2833-pt" value="101"/>
<!-- port to bind to for sip traffic -->
<param name="sip-port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${global_codec_prefs}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<!-- ip address to use for rtp -->
<param name="rtp-ip" value="$${local_ip_v4}"/>
<!-- ip address to bind to -->
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="apply-nat-acl" value="rfc1918"/>
<param name="aggressive-nat-detection" value="true"/>
<!--<param name="enable-timer" value="false"/>-->
<!--<param name="enable-100rel" value="false"/>-->
<param name="apply-inbound-acl" value="domains"/>
<param name="apply-inbound-acl" value="net4india"/>
<!--<param name="apply-register-acl" value="domains"/>-->
<!--<param name="dtmf-type" value="info"/>-->
<param name="record-template" value="$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<!--enable to use presense and mwi -->
<param name="manage-presence" value="true"/>
<!-- 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="120"/>-->
<!--<param name="multiple-registrations" value="true"/>-->
<!--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"/>-->
<!-- TLS: disabled by default, set to "true" to enable -->
<param name="tls" 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="5061"/>
<!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
<param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/>
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
<param name="tls-version" value="tlsv1"/>
<!--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"/> -->
<!-- supress CNG on this profile or per call with the 'supress_cng' variable -->
<!-- <param name="supress-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"/>-->
<!-- 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="true"/>
<!-- on authed calls, authenticate *all* the packets not just invite -->
<param name="auth-all-packets" value="false"/>
<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> -->
<!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> -->
<!-- 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"/>-->
<!--all inbound reg will look in this domain for the users -->
<!--<param name="force-register-domain" value="cluecon.com"/>-->
<!-- disable register and transfer which may be undesirable in a public switch -->
<!--<param name="disable-transfer" value="true"/>-->
<!--<param name="disable-register" value="true"/>-->
<!--<param name="enable-3pcc" value="true"/>-->
<!-- use stun when specified (default is true) -->
<!--<param name="stun-enabled" value="true"/>-->
<!-- use stun when specified (default is true) -->
<!-- set to true to have the profile determine stun is not useful and turn it off globally-->
<!--<param name="stun-auto-disable" value="true"/>-->
</settings>
</profile>
<?xml version="1.0"?>
<include>
<gateway name="net4india">
<param name="extension" value="14085121934"/>
<param name="username" value="052145454" />
<param name="password" value="1980" />
<param name="realm" value="sip.net4india.com" />
<param name="proxy" value="sip.net4india.com" />
<param name="expire-seconds" value="1800" />
<param name="register" value="true" />
<param name="retry_seconds" value="16" />
</gateway>
</include>
<configuration name="sofia.conf" description="sofia Endpoint">
<global_settings>
<param name="log-level" value="0"/>
</global_settings>
<profiles>
<X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml"/>
</profiles>
<profiles>
<profile name="outbound">
<gateways>
<gateway name="sip.net4india.com" />
</gateways>
</profile>
</profiles>
</configuration>
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