January 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

IVR Bridged Call Dropping after 2 rings
by John Carpenter 26 Jan '11

26 Jan '11

Hi, I am trying to setup a very simple IVR using LUA. Call arrives from a DID SIP trunk and is answered and message is played ok, after a particular digit is pressed it bridges the call to an extension which is remotely connected. It works but after 2 rings the call to the extension is dropped with a SIP message "BYE" from DID provider. If I just route the call directly to the extension (no IVR) it works fine. It seems like the DID hangs up when the call is bridged to the extension. Have tried same thing using the XML IVR Engine and get exactly the same result. The IVR script is below pathsep = '/' session:setAutoHangup(false); session:answer() prompt = "ivr" .. pathsep .. "247talk.wav" invalid = "ivr" .. pathsep .. "ivr-that_was_an_invalid_entry.wav" freeswitch.consoleLog("INFO", "Prompt file is '" .. prompt .. "'\n") continue = true while( session:ready() == true and continue == true) do digits = session:playAndGetDigits(1,1,3,7000,"#", prompt, invalid, "\\d+") if (digits == "1") then continue = false session:execute("bridge","sofia/external/2476% 91.xxx.xx.xx") end if (digits == "2") then session:execute("bridge","sofia/external/2475% 91.xxx.xx.xx") end if (digits == "3") then continue = false session:execute("bridge","sofia/external/2475% 91.xxx.xx.xx") end end session:hangup() Any help with this greatly appreciated it is driving me nuts. regards, John Carpenter

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FreeSWITCH Conference Call Wednesday Jan 26 - Vestec Speech Recognition
by Michael Collins 26 Jan '11

26 Jan '11

Hello! I wanted to let everyone know that we have special guests on the conference call this Wednesday: Kashif Kahn and Dr. Jin-Myung Won, both from Vestec. Dr. Won has been deeply involved in ASR development and will be making himself available to answer your technical questions. Please join the conference call and be part of this great discussion about advanced ASR technology! -Michael Agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2011_01_26

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Re: [Freeswitch-users] Bridge String Length Too Short (Resolved)
by Jerry Richards 25 Jan '11

25 Jan '11

FYI, I found the problem. I had an sprintf in the C-App that was only copying 256 characters. Thanks, Jerry From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jerry Richards Sent: Tuesday, January 25, 2011 11:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Bridge String Length Too Short No, I have a C-App to invoke the bridge statement: switch_core_session_execute_application(session, "bridge", results.dialstring); Thanks, Jerry From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, January 25, 2011 10:51 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Bridge String Length Too Short How are you generating this call? Is it by chance using fs_cli -x ? -MC On Tue, Jan 25, 2011 at 10:46 AM, Jerry Richards <jerry.richards(a)teotech.com<mailto:jerry.richards@teotech.com>> wrote: Hello All, I am seeing my bridge statement getting truncated. How may I extend the length of the string it will accept? Here is the actual bridge log: EXECUTE sofia/internal/2003@192.168.72.144:5060<http://2003@192.168.72.144:5060> bridge({ignore_early_media=true}{presence_id=2002(a)192.168.72.144<mailto:2002@192.168.72.144>}sofia/internal/sip:2002@192.168.72.140:5060;transport=udp,sofia/internal/sip:2001@192.168.72.93:5060;transport=udp,[lcr_carrier=Carrier / Location 2/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]op) But it should be: EXECUTE sofia/internal/2003@192.168.72.144:5060<http://2003@192.168.72.144:5060> bridge({ignore_early_media=true}{presence_id=2002(a)192.168.72.144<mailto:2002@192.168.72.144>}sofia/internal/sip:2002@192.168.72.140:5060;transport=udp,sofia/internal/sip:2001@192.168.72.93:5060;transport=udp,[lcr_carrier=Carrier / Location 2/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg_prid/a/15248152341@g1) Thanks, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Difference Between "realm" and "challenge-realm" in sip_profiles
by Jerry Richards 25 Jan '11

25 Jan '11

Hello All, What is the difference between the parameters "realm" and "challenge-realm" in the conf/sip_profiles tree? Thanks, Jerry

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How exactly execute_extension work?
by Boris Kovalenko 25 Jan '11

25 Jan '11

Hello! From one of my extension I call another extension in another context. I suppose that exactly this extension should be executed, but instead I get all previous extension executed to. So below is my configuration. We are calling start_hunting in context tagnet.ru. I suppose start_hunting extension should be executed but no, hunting begins from the start of context and all extensions before start_hunting are executed too. auto_hunt=true solves the problem. So the question is - is this normal behaviour of execute_extension or am I doing something wrong?  <extension name="ext_hunting"> <condition field="${v_tagnet_dst}" expression="^(73435230[0-9]{3}|73435494989|7343549499[0-1])$"> <action application="javascript" data="$${scripts_dir}/toll_check.js ${toll_allow} local"/> <action application="set" data="v_tagnet_hunting=true" inline="true"/> <action application="log" data="NOTICE [top.ctx] - hunting in tagnet.ru"/> <action application="log" data="NOTICE ${v_tagnet_dst} $1"/> <action application="execute_extension" data="start_hunting XML tagnet.ru"/> </condition> </extension> Context tagnet.ru:  <?xml version="1.0" encoding="utf-8"?> <include> <context name="tagnet.ru"> <extension name="unloop"> <condition field="${unroll_loops}" expression="^true$"/> <condition field="${sip_looped_call}" expression="^true$"> <action application="deflect" data="${destination_number}"/> </condition> </extension>  <extension name="ext_translate_dst" continue="true">  <condition field="${destination_number}" expression="^(1[0-9]{3})$" break="on-true"> <action application="set" data="v_tagnet_dst=$1" inline="true"/> </condition> <condition field="${destination_number}" expression="^(\d+)$" break="on-true"> <action application="set" data="v_tagnet_dst=$1" inline="true"/> <action application="execute_extension" data="ext_translate_dst XML features"/> </condition> </extension>  <extension name="ext_translate_src" continue="true"> <condition field="${caller_id_number}" expression="^2([0-9]{4})$" break="on-true"> <action application="set" data="caller_id_number=7343549$1" inline="true"/> <action application="set" data="ani=7343549$1" inline="true"/> </condition> </extension>  <extension name="start_hunting" continue="true"> <condition field="destination_number" expression="^start_hunting$"/> </extension>  <X-PRE-PROCESS cmd="include" data="tagnet.ru/*.xml"/>  <extension name="stop_hunting"> <condition field="${v_tagnet_hunting}" expression="true"/> </extension>  <extension name="top.ctx"> <condition field="${v_tagnet_dst}" expression="^(\d+)$"> <action application="log" data="DEBUG [top.ctx] - transfer to top.ctx"/>  <action application="execute_extension" data="ext_translate_src XML features"/> <action application="transfer" data="$1 XML top.ctx"/> </condition> </extension> </context> </include> -- С уважением, Борис Коваленко ЗАО "Тагнет" (3435) 494991

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Bridge String Length Too Short
by Jerry Richards 25 Jan '11

25 Jan '11

Hello All, I am seeing my bridge statement getting truncated. How may I extend the length of the string it will accept? Here is the actual bridge log: EXECUTE sofia/internal/2003@192.168.72.144:5060 bridge({ignore_early_media=true}{presence_id=2002@192.168.72.144}sofia/internal/sip:2002@192.168.72.140:5060;transport=udp,sofia/internal/sip:2001@192.168.72.93:5060;transport=udp,[lcr_carrier=Carrier / Location 2/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]op) But it should be: EXECUTE sofia/internal/2003@192.168.72.144:5060 bridge({ignore_early_media=true}{presence_id=2002@192.168.72.144}sofia/internal/sip:2002@192.168.72.140:5060;transport=udp,sofia/internal/sip:2001@192.168.72.93:5060;transport=udp,[lcr_carrier=Carrier / Location 2/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg_prid/a/15248152341@g1) Thanks, Jerry

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Is this possible with google voice and freeswitch
by Segmond Yunsai 25 Jan '11

25 Jan '11

Hello, I want to have multiple jingle profiles. when a call is made to a google voice number in the jingle profiles the freeswitch voice will intercept it with a lua script the script will somehow figure out which google voice number/jingle profile was called hold the call, so it doesn't disconnect/get forwarded to voicemail do some things in the lua script then make an outgoing call using google voice or a gateway and bridge em with the incoming call is this possible? i've been able to answer google calls to an extension i've been able to intercept with lua i haven't been able to get dialing out with google voice and i also can't bridge google voice incoming call without an outgoing call through my gateway. the outgoing phone rings, and the phone calling the google voice number rings and then goes to the voicemail. freeswitch gurus please help.

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How to make the heard DTMF tone more longer
by pipiwei03 25 Jan '11

25 Jan '11

Dear all: I try to send DTMF tone. I use two method in lua 1.queue_dtm after bridge 2. send_dtmf in a session these two methods are working and call my phone successfully But, I got a problem is... After bridge, I hear soon DTMF tones. The heard DTMF tones are so short. It is very strange to feel as broken. Even though I set and increase the tone_duration as below: queue_dtmf "12345678@5000" or send_dtmf "12345678@5000" The heard DTMF tones also very short, but the duration between inter-digit become longer... Somebody tell me how to make "heard DTMF tone" longer...or silent the dial tone and sent DTMF tones. Thanks a lot for your support. Best regards, Joy

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Re: [Freeswitch-users] IVRD
by Madovsky 25 Jan '11

25 Jan '11

Sorry I sent this email by mistake without to finish it. so in my dialplan I have : <extension name="gluglu"> <condition field="destination_number" expression="^999$"> <action application="set" data="ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl"/> <action application="socket" data="127.0.0.1:9090 full"/> <action application="answer"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> <condition> </extension> and when I call this extension in log level 7 I can see : Dialplan: sofia/internal/9999999999999@default Action set(ivr_path=/usr/local/freeswitch/scripts/perl/ivrd-hello_world.pl) Dialplan: sofia/internal/9999999999999@default Action socket(127.0.0.1:9090 full) Starting ivrd-hello_world.pl... and no ivrd-hello_world.pl code is executed in the dialplan unless the print "Starting ivrd-hello_world.pl"; the same if I replace fs_ivrd with server2.pl for example Thanks ----- Original Message ----- From: "Madovsky" <infos(a)madovsky.org> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Monday, January 24, 2011 12:48 PM Subject: Re: [Freeswitch-users] IVRD > Concerning the use of fs_ivrd: > > path is ok > permission is root 755 > perl -c gives ../scripts/perl/ivrd-hello_world.pl syntax OK > > in dialplan I have : > > > > > > ----- Original Message ----- > From: "Anthony Minessale" <anthony.minessale(a)gmail.com> > To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> > Sent: Monday, January 24, 2011 11:20 AM > Subject: Re: [Freeswitch-users] IVRD > > > check for proper path and execute permissions on the file and perl -c > to make sure it compiles. > > > On Mon, Jan 24, 2011 at 12:13 AM, Madovsky <infos(a)madovsky.org> wrote: >> I tried to use IVRD from wiki example >> >> http://wiki.freeswitch.org/wiki/Ivrd >> >> and server2.pl in ESL directory >> copy and paste in my dialplan ans settings >> so the daemon is running well, but if I attempt >> to call nothing happens unless hangup. >> on the log I can see only >> >> EXECUTE sofia/internal/9999999999999@default socket(127.0.0.1:8084 full) >> >> I tried the tests of troubleshooting without error >> I don't understand why the events are not received in the perl script >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org

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FreeSWITCH 1.0.7
by Brian West 25 Jan '11

25 Jan '11

http://latest.freeswitch.org/ Enjoy! /b

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