August 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_callcenter record template
by Jayesh Nambiar 31 Aug '12

31 Aug '12

Hi, Could you figure out a solution to this. The variable is ${cc_agent}. --- Jayesh > In mod callcenter.conf.xml there is: > <param name="record-template" value="$${base_dir}/** > recordings/sales/${strftime(%**Y-%m-%d-%H-%M-%S)}.${** > destination_number}.${caller_**id_number}.${uuid}.wav"/> > This names the file with callers number and callcenter's assigned number. > Bit how to include in the file name the agent's number, which responded? > Which variable to use? > > >

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mod_callcenter: Call being forwarded to caller itself
by Nitin Tomer 31 Aug '12

31 Aug '12

Hi, I've configure mod_callcenter on Ubuntu 12.0.4. I've 5 extensions working on 6 soft phones - 1000 to 1005. 1000 is used to call the IVR number, rest 5 are agents. The dial plan looks like - <extension name="ivr_nitin"> <condition field="destination_number" expression="^5002$"> <action application="answer"/> <action application="sleep" data="500"/> <action application="ivr" data="nitin_ivr"/> </condition> </extension>   <extension> <condition field="destination_number" expression="^(450)$"> <action application="set" data="caller_id_name=Newgen Support" /> <action application="set" data="call_timeout=60" /> <action application="set" data="originate_timeout=60" /> <action application="callcenter" data="support@default"/> </condition> </extension>  <extension> <condition field="destination_number" expression="^(451)$"> <action application="set" data="caller_id_name=Newgen Sales" /> <action application="set" data="call_timeout=60" /> <action application="set" data="originate_timeout=60" /> <action application="callcenter" data="sales@default"/> </condition> </extension>  <extension> <condition field="destination_number" expression="^(452)$"> <action application="set" data="caller_id_name=Newgen Human Resources" /> <action application="set" data="call_timeout=60" /> <action application="set" data="originate_timeout=60" /> <action application="callcenter" data="hr@default"/> </condition> </extension> IVR.xml - <include> <menu name="nitin_ivr" greet-long="say:Welcome to Newgen Software Technologies Limited. Press 1 for support, 2 for sales or 3 for human resource" greet-short="say:Welcome to Newgen. Press 1 for support, 2 for sales or 3 for human resources" invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" exit-sound="voicemail/vm-goodbye.wav" confirm-macro="" confirm-key="" tts-engine="flite" tts-voice="slt" confirm-attempts="3" timeout="3000" inter-digit-timeout="2000" max-failures="3" max-timeouts="3" digit-len="4"> <entry action="menu-exec-app" digits="1" param="transfer 450 XML default"/>  <entry action="menu-exec-app" digits="2" param="transfer 451 XML default"/>  <entry action="menu-exec-app" digits="3" param="transfer 452 XML default"/>  </menu> </include> And callcenter.conf.xml - <configuration name="callcenter.conf" description="CallCenter"> <settings>   </settings> <queues> <queue name="sales@default"> <param name="strategy" value="agent-with-least-talk-time"/> <param name="moh-sound" value="$${hold_music}"/>  <param name="time-base-score" value="queue"/> <param name="tier-rules-apply" value="false"/> <param name="tier-rule-wait-second" value="300"/> <param name="tier-rule-wait-multiply-level" value="true"/> <param name="tier-rule-no-agent-no-wait" value="false"/> <param name="discard-abandoned-after" value="14400"/> <param name="max-wait-time" value="0"/> <param name="max-wait-time-with-no-agent" value="120"/> </queue> <queue name="support@default"> <param name="strategy" value="agent-with-least-talk-time"/> <param name="moh-sound" value="$${hold_music}"/>  <param name="time-base-score" value="queue"/> <param name="tier-rules-apply" value="false"/> <param name="tier-rule-wait-second" value="300"/> <param name="tier-rule-wait-multiply-level" value="true"/> <param name="tier-rule-no-agent-no-wait" value="false"/> <param name="discard-abandoned-after" value="60"/> <param name="abandoned-resume-allowed" value="false"/> <param name="max-wait-time" value="0"/> <param name="max-wait-time-with-no-agent" value="120"/> </queue> <queue name="hr@default"> <param name="strategy" value="agent-with-least-talk-time"/> <param name="moh-sound" value="$${hold_music}"/>  <param name="time-base-score" value="queue"/> <param name="tier-rules-apply" value="false"/> <param name="tier-rule-wait-second" value="300"/> <param name="tier-rule-wait-multiply-level" value="true"/> <param name="tier-rule-no-agent-no-wait" value="false"/> <param name="discard-abandoned-after" value="60"/> <param name="abandoned-resume-allowed" value="false"/> <param name="max-wait-time" value="0"/> <param name="max-wait-time-with-no-agent" value="120"/> </queue> </queues>    <agents> <agent name="1001" type="callback" contact="[call_timeout=10]user/1001" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" /> <agent name="1002" type="callback" contact="[call_timeout=10]user/1002" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" /> <agent name="1003" type="callback" contact="[call_timeout=10]user/1003" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" /> <agent name="1004" type="callback" contact="[call_timeout=10]user/1004" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" /> <agent name="1005" type="callback" contact="[call_timeout=10]user/1005" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" /> </agents> <tiers>   <tier agent="1001" queue="sales@default" level="1" position="1"/> <tier agent="1001" queue="support@default" level="1" position="1"/> <tier agent="1001" queue="hr@default" level="1" position="1"/> <tier agent="1002" queue="sales@default" level="1" position="1"/> <tier agent="1003" queue="support@default" level="1" position="1"/> <tier agent="1004" queue="sales@default" level="1" position="1"/> <tier agent="1004" queue="support@default" level="1" position="1"/> <tier agent="1005" queue="hr@default" level="1" position="1"/> </tiers> </configuration> When I call from extension 1000 and press 2 for sales, call is routed back to extension 1000, which is not even in agents' list. Why is it happening? Regards Nitin Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL) accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.

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cisco spa 500 configs?
by Michael Collins 31 Aug '12

31 Aug '12

Does anyone have a working config for a spa 5xx phone w/ s500 sidecar? I'd like to get that and the SPA 300 phones better documented on the wiki. Thanks! -MC -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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group call pickup
by Miha 31 Aug '12

31 Aug '12

Hi, I have implemented call pickup, which is not working as it should. When call comes in, user from same group can pick it up, but problem appears in this scenario: A calls B. C do call pickup with *5 (in dialplan). OK this works (A is talking with C). But when D is calling E, if someone from same group pick this call, it do not pick D, it picks call A. Why? What I am doing wrong? I hope you understand what I mean:D my dialplan: <action application="set" data="called_party_callgroup=${user_data(${destination_number}.enterprise(a)enterprise.fs2.softnet.si var callgroup)}"/> <action application="hash" data="insert/${destination_number}/${called_party_callgroup}/${uuid}"/> <action application="hash" data="insert/last_dial/${called_party_callgroup}/${uuid}"/> intercept: <extension name="group-intercept"> <condition field="destination_number" expression="^\*5$"> <action application="answer"/> <action application="intercept" data="${hash(select/last_dial/${callgroup})}"/> <action application="sleep" data="2000"/> </condition> </extension> Thanks! Miha

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DialPlan Generation
by Subhash 31 Aug '12

31 Aug '12

Hi, Which is the best way to create(static or dynamic) dialplan? Please suggest me. Thanks, Subhash.

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uuid_broadcast muxing.
by Vbvbrj 31 Aug '12

31 Aug '12

Hello. I use uuid_broadcast to play a file to some leg: api:executeString("uuid_broadcast "..caller_uuid.." phrase::queue-position,"..pos.." aleg") But this pauses the moh file which is played to member waiting in callcenter. After two broadcasts, there is an error about leaking stream and adjusting of moh play. Is there a command to do the same but mux this playback of position with moh playing? uuid_displace require absolute paths and also does not accept phrases. Also no leg specifying is accepted. Thank you.

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Best way to stream record session?
by Michael Giagnocavo 31 Aug '12

31 Aug '12

Hi all, I want to record and stream a (bridged) call to a non-channel endpoint. Sorta like mod_spy, but targeting a TCP or UDP connection or even a one-way RTP stream. What's the best way to approach this? What about using normal session record to a file, then "streaming" that file from an external program? Is FS guaranteed to write to the wav file regularly (like every second or so) in a manner that another program can keep reading? Or should I specify a file like "/dev/udp/1.2.3.4/666"? End goal is to have a separate machine that can provide a service like Internet radio and have it tap into live calls. Cheers, Michael

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testing
by reza a 30 Aug '12

30 Aug '12

testing dont know if i subscribed to this properly.

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Re: [Freeswitch-users] how to play a prompt after connection
by Dave Horton 30 Aug '12

30 Aug '12

> Set api_on_answer to execute > uuid_broadcast< > http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast > > > which > will let you pick which legs to play the recording to. > You can pass in the UUID with ${uuid} > > -Avi > I tried this, and it worked.....sort of. If I play the prompt the A leg it is fine -- caller hears the prompt, then is connected to called party. If I play it to the B leg, however, the prompt does not seem to start playing at all until the B party actually starts talking, and then the prompt is played extremely choppy and slow.....it takes about 10 seconds to play a 4 second prompt. Meanwhile, the caller and called party can't hear each other. Any idea what is going on and how to fix it? I am doing it in lua as follows: session:execute("export","nolocal:jitterbuffer_msec=100") ; session:setVariable("RECORD_STEREO","true") ; session:setVariable("aleg_uuid",uuid) ; session:execute("record_session", outfile) ; session:execute("set","progress_timeout=60") ; session:execute("set","call_timeout=120") ; session:setVariable("hangup_after_bridge","true") ; session:execute("bridge","sofia/normal_customer/${outdial}@${egress_gateway}") ; --session:execute("bridge","{api_on_answer='uuid_broadcast " .. uuid .. " " .. this_call_is_being_recorded .. " " .. "bleg'}sofia/normal_customer/${outdial}@${egress_gateway}") ;

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Eavesdrop with bind_digit_action
by Lon Baker 30 Aug '12

30 Aug '12

I'm trying to add the ability to screen voicemail being left, and allow the callee to pull the call back from voicemail with a DTMF. I thought I could use eavesdrop and bind_digit_action, but its not working. The bound DTMF is logged in the console, but the exec action is never triggered. Caller is correctly sent to voicemail Callee, who elected to eavesdrop, can hear the voicemail being left. Pressing 9 for the bind_digit_action, show up in the console. But, exec extension does not go off. Any help would be appreciated. -- Lon

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