May 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Question about digit-len and inter-digit-timeout on ivr
by Nestor A Diaz 03 May '12

03 May '12

Hi Guys. Just by curiosity: Let suppose i have an ivr with just two options: * digit = '1' * digit = '/^2[0-9]$/ with digit-len = 2 and inter-digit-timeout = 5000 If i press 1 i have to wait inter-digit-timeout to perform the action, however there will be no 1[0-9] so the system can perform the action without having to wait inter-digit-timeout time. Is there any way to change digit-len to something like 'dynamic' ? so this way everytime the ivr get a digit it will compare against possible choices and if there will be just one that match then it will perform the action inmediately. Slds. -- Nestor A. Diaz nestor(a)tiendalinux.com

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preAnswer call at lua script and then bridge it to another leg
by Nikolay Rogoshchenkov 03 May '12

03 May '12

Hello, I have a dialplan where I get the incoming call to lua script <extension name="ext_call"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="ignore_early_media=false"/> <condition field="destination_number" expression="\d{7,}"/>. <condition field="${sip_from_host}" expression="^(10\.9\.0\.25)$">. <action application ="lua" data="answer.lua"/>. </condition> </extension> and then I have answering in answer.lua: =========================== session:preAnswer(); [...accessing to DB...] if (session:ready()) then repeat index = index + 1; session_call = string.format("[leg_timeout=30,ignore_early_media=true,hangup_after_bridge=true,origination_caller_id_number=".. cid .."]sofia/gateway/%s/%s",gwlist[index],dn); legB = freeswitch.Session(session_call); until ((hcause == 'USER_BUSY') or (hcause == 'SUCCESS') or (hcause == 'NO_ANSWER') or (index == #gwlist) or(session:ready() == false)) if (legB:ready()) then. freeswitch.bridge(session, legB) end =========================== But I have a one problem, when the initial legA was disconnected before legB was answered. It it still continued to call legB (during the leg_timeout). Could you please explain, how I can cancel legB right after I got disconnect on legA in this case? Thank you. -- Nikolay

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Installing FS to a non-standard directory
by Jason Moran 02 May '12

02 May '12

I do not want to install FS to the normal /usr/local/freeswitch/ Where do I make a change such that the install goes elsewhere? After running ./bootstrap.sh then ./configure it lists the install directory as /usr/local/freeswitch/ but I cannot find any inputs to change these values. Thanks, Jason

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Silence instead of spanish wav installed
by Alejandro Mejía 02 May '12

02 May '12

Hello, I installed a spanish sounds package to my Freeswitch under /usr/local/freeswitch/sounds/es/mx/ directory I'm trying it by entering an empty conference, and I only hear some seconds of silence (where conf-alone.wav should be played), and then the music on hold. I know the files are not missing, and Freeswitch is finding them because when I remove "/es" directory, I get error on CLI when executing a conference, and getting directly to MOH. Any help will be appreciated. Thanks! Alex

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Help - SIP gateway issue with fs_cli but not with freeswtich console.
by Mahendra Bhegade 02 May '12

02 May '12

Hi , I was wondering if you can help me with the issue that I am facing. I have a lua script which dials out using a gateway and works fine from freeswitch console. If I execute the same script from fs_cli I get an error identifiying an error with gateway and the complains about the numbe being invalued I am using a flowroute gateway and sip end point is sofia/gateway/flowroute/15104496650 my entry for flowroute.xml is under conf/sip_profiles/external If some one can shed some light on the matter I would appreciate it. Mahendra 510-449-6650

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Voicemail hangups 38 sec into recording
by George Sapak 02 May '12

02 May '12

Still having issue with hangup: FreeSWITCH Version 1.1.beta1 (git-5e5a2ff 2012-04-23 07-50-57 -0500) I have set the waste resources: <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="set" data="record_waste_resources=true"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> Thank You, George.

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Récuperer le digit taper ou moment de l'appel(mod_ HTTAPI)
by kader 02 May '12

02 May '12

Bonjour je serai très reconnaissant si vous pouvez me donner me débloquer dans ce problème. j'appel une extension dans dialplan, dans laquel je demande un fichier .jsp ou serveur web, ma demande arrive le plus normal ou serveur et ce dernier renvoie le fichier jsp ou freeswitch. mon problème c'est que j'arrive pas à récuperer les digits que je tape sur le softphone sur le serveur. voila mon fichier jsp: <% response.setContentType("text/xml"); %><?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="xml/freeswitch-httapi"> <work> <playback name=exten file="http://XXXXXXXXXX/tours.wav" input-timeout="5000" digit-timeout="2000"> <bind strip="#">~\d+</bind> </playback> </work> </document> -- View this message in context: http://old.nabble.com/R%C3%A9cuperer-le-digit-taper-ou-moment-de-l%27appel%… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Freeswitch speaks italian
by Chris B. Ware 02 May '12

02 May '12

Michael, is there a place where I can upload italian sound files? Just to add a link to the wiki page, once I've updated it. Thanks, Chris

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mod_voicemail and non-numeric id
by Anton Kvashenkin 02 May '12

02 May '12

Hello list. I've decided to use non-numeric id's at users configuration. <user id="001565229c95" number-alias="109" mailbox="109"> But when the call goes to voicemail app <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> switch_ivr_play_say.c:244 Handle say:[001565229c95] (ru:ru) 2012-04-27 11:39:54.452208 [INFO] mod_say_ru.c:721 ru_say!!! 001565229c95! say_opt.gender=0 say_opt.cases=0 2012-04-27 11:39:54.532157 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16@16000hz 1 channels 20ms The mod_say pronounces user id. How to force to pronounce number-alias?

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unsched
by Darcy Primrose 01 May '12

01 May '12

Hi, I am trying to get a call to drop back to voice mail on timeout after I originate it to the pstn via voip. I cannot use call_timeout since all outbound calls use early media, if I disable early media, then I get not ring back tone. So I am thinking something like sched_hangup, then use an execute on answer to disable the hangup, unsched_api or sched_del. I do not know if this is the best way to do this. I am having trouble getting the task_id from the sched_hangup so I can later delete it. Any suggestions would be most welcome. For now, I would like to accomplish this in the dial plan. <action application="sched_hangup" data="+30 alloted_timeout"/> Darcy Primrose

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