I am trying my custom configuration to handle freeswitch call using Elastic SIP Trunking. I am not using mod_Signalwire. I want to handle route custom with multi-tenant system.
Below is my configuration:
Profile:
<document type="freeswitch/xml">
<section name="configuration">
<configuration name="sofia.conf" description="sofia conf">
<profiles>
<profile name="MY_PROFILE">
<aliases/>
<gateways>
<gateway name="MY_GATEWAY">
<param name="username" value="MY_USER"/>
<param name="password" value="MY_PW"/>
<param name="proxy" value="fjfhjhhhk.sip.signalwire.com"/>
<param name="realm" value="ffjjhghjh.sip.signalwire.com"/>
<param name="from-domain" value="bhjhkhkkj.sip.signalwire.com"/>
<param name="register" value="true"/>
<param name="sip-trace" value="no"/>
<param name="sip-capture" value="no"/>
<param name="context" value="public"/>
<param name="extension" value="sip_uri"/>
</gateway>
</gateways>
<domains name="all" alias="true" parse="true"/>
<settings>
<params name="ext-sip-ip" value="$${external_sip_ip}"/>
<params name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<params name="sip-port" value="5060"/>
<params name="auth-calls" value="true"/>
</settings>
</profile>
</profiles>
</configuration>
</section>
</document>
Directory:
<document type="freeswitch/xml">
<section name="directory">
<domains>
<domain name="ddsadaf.sip.signalwire.com">
<params>
<param name="dial-string" value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})},${verto_contact(${dialed_user}@${dialed_domain})}"/>
</params>
<variables>
<variable name="user_context" value="public"/>
</variables>
<groups>
<group>
<users>
<user id="jssip">
<params>
<param name="$${default_password}" value="Chits123"/>
<param name="vm-password" value="MY_PASSWORD"/>
</params>
<variables>
<variable name="toll_allow" value="domestic, international, local"/>
<variable name="accountcode" value="jssip"/>
<variable name="user_context" value="dsadasdsad.sip.signalwire.com"/>
<variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
<variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
</variables>
</user>
</users>
</group>
</groups>
</domain>
</domains>
</section>
</document>
Dialplan:
<document type="freeswitch/xml">
<section name="dialplan" description="Dial Plan For FreeSwitch">
<context name="public">
<extension name="sip_uri">
<condition field="destination_number" expression="^(MY_US_PHONE_NUMBER)$"> // ex 12345678901 - eleven US number with countr cde
<action application="sleep" data="10000"/>
<action application="bridge" data="sofia/gateway/GATEWAY_NAME/$1"/>
<action application="answer"/>
</condition>
</extension>
</context>
</section>
</document>
I am getting below error:
switch_core_state_machine.c:312 No Route, Aborting
I am able to successfully create sofia profile and gateway. Both are showing Running and Registered status gradually. I am also receiving call in my freeswitch log but it stuck at above error in my dial plan.
