Hi! I am very very new to Freeswitch. I am running FreeSWITCH Version 1.5.15b+git~20141120T035109Z~79de78a0fb~64bit (git 79de78a 2014-11-20 03:51:09Z 64bit) on a CentOS 6.6 64-bit Virtual Machine. I am trying to setup freeswitch such that once it gets a call through a sip gateway, it sends the caller ID to another SIP gateway (A URI) to be processed. I am having a lot of difficulty and I am not sure if I am doing the right things correctly, and the logs give so much information it is difficult to see what is happening. Please help me verify if I am correctly configured. I've setup the gateways I expect inbound calls from and freeswitch registers correctly. example: <gateway name="sipgw081"> <param name="username" value="012345081"/> <param name="password" value="xxxxx"/> <param name="register" value="true"/> <param name="realm" value="sip.sipgwtelecoms.com"/> <param name="proxy" value="sip.sipgwtelecoms.com"/> <param name="outbound-proxy" value="sip.sipgwtelecoms.com"/> <param name="caller-id-in-from" value="true"/> </gateway> Sofia returns: external::sipgw081 gateway sip:012345081@sip.sipgwtelecoms.com REGED The gateway I expect to route caller_id_numbers to is defined, but as I was not given a password or asked to register I set it up like so (is this OK?): <gateway name="othersipgw"> <param name="username" value="user.name"/> <param name="password" value="none"/> <param name="register" value="false"/> <param name="realm" value="sip.othersipgw.in"/> <param name="proxy" value="sip.othersipgw.in"/> <param name="outbound-proxy" value="sip.othersipgw.in"/> <param name="expire-seconds" value="3600"/> <!-- <param name="caller-id-in-from" value="true"/> --> </gateway> Sofia returns: external::othersipgw gateway sip:user.name@sip.othersipgw.in NOREG

From what I can see in the documentation I've read, in order to route inbound calls anywhere in freeswitch, Calls that initially come in to the public context and are treated as untrusted—if they are not specifically routed to an extension in the default context, then they are simply disconnected.

So I created a file in conf/dailplan/public with the following to dail an extension 1212 in default( I want to route caller IDs with 11 digits, all beginning with number 0): <include> <extension name="sipgw-inbound"> <condition field="caller_id_number" expression="^0(\d+)$"> <!-- <action application="set" data="domain_name=$${domain}"/> --> <action application="transfer" data="1212 XML default"/> </condition> </extension> </include> This is where my confusion starts. I both created a file in conf/dialplan/default and edited the conf/dialplan/default.xml to route calls sent to 1212 through to the external gateway URI. I also want the caller_id prefixed with a +234 default.xml: <extension name="1212"> <condition field="destination_number" expression="^1212$"> <action application="set"data="absolute_codec_string=G729"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/gateway/othersipgw/+234$1"/> </condition> </extension> outbound.xml file in conf/dialplan/default: <include> <extension name="othersipgwoutbound"> <condition field="destination number" expression="1212"> <action application="set"data="absolute_codec_string=G729"/> <action application="bridge" data="[leg_timeout=5]sofia/gateway/othersipgw/$1"/> </condition> </extension> </include> Any assistance gratefully accepted.