<div dir="ltr"><br><div><br></div><div><div> <extension name="Local_Extension"></div><div> <condition field="destination_number" expression="^(10[01][0-9])$"></div><div> <action application="export" data="dialed_extension=$1"/></div><div> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --></div><div> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/></div><div> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/></div><div> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/></div><div> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/></div><div> <action application="set" data="ringback=${us-ring}"/></div><div> <action application="set" data="transfer_ringback=$${hold_music}"/></div><div> <action application="set" data="call_timeout=30"/></div><div> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --></div><div> <action application="set" data="hangup_after_bridge=true"/></div><div> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --></div><div> <action application="set" data="continue_on_fail=true"/></div><div> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/></div><div> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/></div><div> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/></div><div> <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/></div><div> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/></div><div> <!--<action application="export" data="nolocal:rtp_secure_media=${user_data(${dialed_extension}@${domain_name} var rtp_secure_media)}"/>--></div><div> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/></div><div> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/></div><div> <action application="answer"/></div><div> <action application="sleep" data="1000"/></div><div> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/></div><div> </condition></div><div> </extension></div></div><div><br></div><div><br></div><div><br></div><div><br></div><div>use </div><div><action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/><br></div><div><br></div><div>set a release cause you want FS to continue processing after bridge and it will hit voicemail.</div><div><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On 15 December 2016 at 08:02, devang nathwani <span dir="ltr"><<a href="mailto:devang.nathwani31589@gmail.com" target="_blank">devang.nathwani31589@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hello,<br><br>I am using multi-domain system in freeswitch, i have register sip-devices using domain name(<a href="http://demo1.xyz.com" target="_blank">demo1.xyz.com</a>) and trying to make call to sip to sip(on-net/local) call and it's working but when another sip-device was busy or not register that time call should pass in voicemail but in getting ' Can't find user [<a href="mailto:1883317281@demo1.xyz.com" target="_blank">1883317281@demo1.xyz.com</a>]'.<br>192.168.1.35 is server IP<br>192.168.1.30 from where i am regsitring.<br><br><a href="https://pastebin.freeswitch.org/view/efb0d2e5" target="_blank">https://pastebin.freeswitch.<wbr>org/view/efb0d2e5</a><br></div>
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