Hello, I am using multi-domain system in freeswitch, i have register sip-devices using domain name(demo1.xyz.com) 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 [1883317281@demo1.xyz.com]'. 192.168.1.35 is server IP 192.168.1.30 from where i am regsitring. https://pastebin.freeswitch.org/view/efb0d2e5
<extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> <!--<action application="export" data="nolocal:rtp_secure_media=${user_data(${dialed_extension}@${domain_name} var rtp_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> use <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> set a release cause you want FS to continue processing after bridge and it will hit voicemail. On 15 December 2016 at 08:02, devang nathwani < devang.nathwani31589@gmail.com> wrote:
Hello,
I am using multi-domain system in freeswitch, i have register sip-devices using domain name(demo1.xyz.com) 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 [1883317281@demo1.xyz.com]'. 192.168.1.35 is server IP 192.168.1.30 from where i am regsitring.
https://pastebin.freeswitch.org/view/efb0d2e5
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participants (2)
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devang nathwani
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Tihomir Culjaga