April 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Anyone have a good dialplan recipe for a long distance forced authorization code?
by Nathan Neulinger 05 Apr '13

05 Apr '13

I.e. to require a pin entry prior to bridging an outbound long distance call? I was thinking I could duplicate my outbound gateway bridging rules with an answer, start_dtmf, play_and_get_digits, followed by the actual bridging instruction. Is that the ideal way to do this? Intent is for open access phones that have restricted long distance unless the line-specific authorization code is entered. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Re: [Freeswitch-users] start_tone_detect, but no event is fired
by Ashish gautam 05 Apr '13

05 Apr '13

Hi Henry, You can switch the event reporting on using this command for the event you would like to listen to "event [xml/json etc.] LIST/ALL" On Fri, Apr 5, 2013 at 11:55 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: start_tone_detect, but no event is fired (henry.houfeng) > 2. Re: Call not originated through event socket (Ashish gautam) > > > ---------- Forwarded message ---------- > From: "henry.houfeng" <henry.houfeng(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Thu, 4 Apr 2013 19:16:24 -0700 (PDT) > Subject: Re: [Freeswitch-users] start_tone_detect, but no event is fired > Anyone can help me? Thanks! > > Regards, > Henry > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/start-tone-detect-but-no-even… > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > ---------- Forwarded message ---------- > From: Ashish gautam <ashish(a)nms.co.in> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Fri, 5 Apr 2013 11:54:48 +0530 > Subject: Re: [Freeswitch-users] Call not originated through event socket > Hi Steven, > > Thanks for responding. > > I am checking the console at debug level and getting > [NORMAL_CIRCUIT_CONGESTION] error. This must not happen as the number of > calls I am generating is only 4. What is happening is, the ES generates > calls to all the numbers except the last one (it always fails for the last > number and shows NORMAL_CIRCUIT_CONGESTION). If the problem would be in > originate dialstring, calls to other numbers would also be unsuccessful but > they are working fine. > > Thanks. > -Ashish > > On Fri, Apr 5, 2013 at 12:17 AM, < > freeswitch-users-request(a)lists.freeswitch.org> wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users(a)lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request(a)lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner(a)lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. Re: One of my extensions Caller ID says Anonymous! (Vik Killa) >> 2. Re: freeswitch say application currency in multipal language >> (Michael Collins) >> 3. Re: One of my extensions Caller ID says Anonymous! (Avi Marcus) >> 4. Re: Call not originated through event socket (Steven Ayre) >> >> >> ---------- Forwarded message ---------- >> From: Vik Killa <vipkilla(a)gmail.com> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Thu, 4 Apr 2013 14:13:26 -0400 >> Subject: Re: [Freeswitch-users] One of my extensions Caller ID says >> Anonymous! >> Please let us know if and how you get it resolved. I'll do the same if I >> can figure it out. Thanks. >> >> <snip> >> >> >> ---------- Forwarded message ---------- >> From: Michael Collins <msc(a)freeswitch.org> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Thu, 4 Apr 2013 11:14:52 -0700 >> Subject: Re: [Freeswitch-users] freeswitch say application currency in >> multipal language >> I don't believe that there is currently a way to do this easily right >> now. We just spoke about languages on yesterday's conference call and this >> is a prime example of the kinds of things that we will need to overcome. >> >> Additionally I don't believe that I have any currencies other than >> dollar.wav and dollars.wav for the English sounds. I'll be glad to get them >> ordered. Could the community at large send me some ideas for units of >> currency? Here are a few ideas: >> >> euro, euros >> franc, francs >> Canadian, Australian, US dollar/dollars >> pound, pounds >> >> Send me some more ideas and I will get them added to the to-be-recorded >> list. >> >> -MC >> >> On Thu, Apr 4, 2013 at 1:52 AM, bhavik patel <bhavikpatel14388(a)gmail.com>wrote: >> >>> Hi all, >>> I use free switch and i want to play sounds file like if user has credit >>> in USD then doller.wav file play and EUR then another file will be play. >>> >>> Currently it play doller.wav by default in >>> /usr/local/freeswitch/sounds/currency/en/doller.wav but i want to play EUR >>> so how can this possible. >>> >>> Is that any easy way to do this thing in multi language currency play in >>> say application. >>> >>> i use this syntax in my free-switch dial plan >>> $dialstring = "<action application=\"say\" data=\"en CURRENCY PRONOUNCED >>> $credit_balance\"/>"; >>> >>> Thanks In advance... >>> >>> -- >>> Thanks, >>> Bhavik Patel >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> ---------- Forwarded message ---------- >> From: Avi Marcus <avi(a)avimarcus.net> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Thu, 4 Apr 2013 21:16:04 +0300 >> Subject: Re: [Freeswitch-users] One of my extensions Caller ID says >> Anonymous! >> Indeed, outbound_caller_id_name is just a variable. >> >> To use it you have to tell the dialplan to set >> effective_caller_id_name=${outbound_caller_id_name}. >> effective_caller_id_name IS a variable that actually gets used. >> (The default dialplan does this.) >> >> -Avi Marcus >> BestFone >> >> >> On Thu, Apr 4, 2013 at 8:26 PM, Steven Schoch < >> schoch+freeswitch.org(a)xwin32.com> wrote: >> >>> In my directory, I set the variable "effective_caller_id_name" instead >>> of "internal_caller_id_name". (I also set the "outbound_caller_id_name" >>> variable just like your example, although that never seems to get sent to >>> the PSTN.) >>> >>> -- >>> Steve >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> ---------- Forwarded message ---------- >> From: Steven Ayre <steveayre(a)gmail.com> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Thu, 4 Apr 2013 19:46:22 +0100 >> Subject: Re: [Freeswitch-users] Call not originated through event socket >> +1 >> >> '/log 9' and watch the debug output. It'll be very clear about what's >> happening. Most likely there's something wrong about your usage of >> originate. >> >> -Steve >> >> >> On 4 April 2013 19:08, Michael Collins <msc(a)freeswitch.org> wrote: >> >>> You could also just hop on fs_cli and watch the output while you make a >>> test call with the event socket. Most likely you'll see what's going on. >>> >>> -MC >>> >>> On Thu, Apr 4, 2013 at 10:39 AM, Hector Geraldino < >>> Hector.Geraldino(a)ipsoft.com> wrote: >>> >>>> +OK doesn’t mean the call was generated, it just means that >>>> FreeSWITCH received (and processed) the command.**** >>>> >>>> ** ** >>>> >>>> You should probably be better served by listening to the events for >>>> each one of these calls. There are a few handful events that can tell you >>>> if the channel was created, the call was originated, if it was answered and >>>> when it was dropped with the drop reason.**** >>>> >>>> ** ** >>>> >>>> *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: >>>> freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Ashish >>>> gautam >>>> *Sent:* Thursday, April 04, 2013 5:26 AM >>>> *To:* freeswitch-users(a)lists.freeswitch.org >>>> *Subject:* [Freeswitch-users] Call not originated through event socket* >>>> *** >>>> >>>> ** ** >>>> >>>> Hi,**** >>>> >>>> ** ** >>>> >>>> I am generating calls through event socket using originate action to >>>> multiple numbers to PSTN network. ES is returning '+OK' for the call not >>>> generated. Still the call is not generated.**** >>>> >>>> ** ** >>>> >>>> Any help is appreciated.**** >>>> >>>> ** ** >>>> >>>> Thanks.**** >>>> >>>> ** ** >>>> >>>> -Ashish**** >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> _______________________________________________ >> 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 >> >> > > _______________________________________________ > 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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One of my extensions Caller ID says Anonymous!
by Sean Devoy 05 Apr '13

05 Apr '13

Hi, I have the current head (minus a week or so) I have many extensions/users defined almost identically in the directory: (redacted with ***'s) <user id="210"> <variables> <variable name="outbound_caller_id_name" value="BFIS Gene"/> <variable name="outbound_caller_id_number" value="4104207777"/> <variable name="internal_caller_id_name" value="Gene"/> <variable name="internal_caller_id_number" value="210"/> <variable name="user_context" value="from-internal-*********"/> <variable name="user_originated" value="true"/> <variable name="accountcode" value="****"/> <variable name="mwi-account" value="210@voicemail_********"/> <variable name="callgroup" value="1"/> </variables> <params> <param name="password" value="***"/> <param name="dial-string" value="{presence_id=210@*********}${sofia_contact(210@********)}"/> <param name="mwi-account" value="210@voicemail_********"/> </params> </user> For almost everyone, when dialing a number, the log says: 2013-04-03 *** [INFO] mod_dialplan_xml.c:557 Processing 220 <220>->141454*** in context from-internal-***** For user 210, I get: 2013-04-03 *** [INFO] mod_dialplan_xml.c:557 Processing Anonymous <210>->141454*** in context from-internal-****** The phones are cisco 504Gs. Any idea where that is coming from our how I can fix it? Thanks Sean

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error 606 user not registered
by Jason Holden 05 Apr '13

05 Apr '13

Anyone have a suggestion on how to correct this? When I do a show registrations I do see the extension registered.

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Creating a User account with no SIP login
by Alex Lake 05 Apr '13

05 Apr '13

I want to have a freeswitch user that can have voicemail but can't be logged into from a SIP handset. How can I do this?

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make current error
by Виталий Давудов 05 Apr '13

05 Apr '13

Hi, list! I've installed FreeSWITCH Version 1.2.3+git~20120920T220849Z~f718a5e8e6 (1.2.3; git at commit f718a5e8e6 on Thu, 20 Sep 2012 22:08:49 Z) Now, I try to update to the latest tree. While running "make current" in FS source directory, there is an error: ... make[1]: Leaving directory `/usr/src/freeswitch' make all make[1]: Entering directory `/usr/src/freeswitch' CDPATH="${ZSH_VERSION+.}:" && cd . && /bin/sh /usr/src/freeswitch/build/config/missing --run aclocal-1.11 cd . && /bin/sh /usr/src/freeswitch/build/config/missing --run automake-1.11 --foreign CDPATH="${ZSH_VERSION+.}:" && cd . && /bin/sh /usr/src/freeswitch/build/config/missing --run autoconf configure.in:141: error: possibly undefined macro: AC_PROG_LIBTOOL If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. make[1]: *** [configure] Error 1 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2 In addition, already installed: # rpm -qa | grep autoconf autoconf-2.62-12.3 # rpm -qa | grep libtool libtool-1.5.22-7.el5_4 What might be the reasons of this error? May be new versions of libtool/autoconf is required? -- Best regards, Vitaly.

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Re: [Freeswitch-users] Call not originated through event socket
by Ashish gautam 05 Apr '13

05 Apr '13

Hi Steven, Thanks for responding. I am checking the console at debug level and getting [NORMAL_CIRCUIT_CONGESTION] error. This must not happen as the number of calls I am generating is only 4. What is happening is, the ES generates calls to all the numbers except the last one (it always fails for the last number and shows NORMAL_CIRCUIT_CONGESTION). If the problem would be in originate dialstring, calls to other numbers would also be unsuccessful but they are working fine. Thanks. -Ashish On Fri, Apr 5, 2013 at 12:17 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: One of my extensions Caller ID says Anonymous! (Vik Killa) > 2. Re: freeswitch say application currency in multipal language > (Michael Collins) > 3. Re: One of my extensions Caller ID says Anonymous! (Avi Marcus) > 4. Re: Call not originated through event socket (Steven Ayre) > > > ---------- Forwarded message ---------- > From: Vik Killa <vipkilla(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 4 Apr 2013 14:13:26 -0400 > Subject: Re: [Freeswitch-users] One of my extensions Caller ID says > Anonymous! > Please let us know if and how you get it resolved. I'll do the same if I > can figure it out. Thanks. > > <snip> > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 4 Apr 2013 11:14:52 -0700 > Subject: Re: [Freeswitch-users] freeswitch say application currency in > multipal language > I don't believe that there is currently a way to do this easily right now. > We just spoke about languages on yesterday's conference call and this is a > prime example of the kinds of things that we will need to overcome. > > Additionally I don't believe that I have any currencies other than > dollar.wav and dollars.wav for the English sounds. I'll be glad to get them > ordered. Could the community at large send me some ideas for units of > currency? Here are a few ideas: > > euro, euros > franc, francs > Canadian, Australian, US dollar/dollars > pound, pounds > > Send me some more ideas and I will get them added to the to-be-recorded > list. > > -MC > > On Thu, Apr 4, 2013 at 1:52 AM, bhavik patel <bhavikpatel14388(a)gmail.com>wrote: > >> Hi all, >> I use free switch and i want to play sounds file like if user has credit >> in USD then doller.wav file play and EUR then another file will be play. >> >> Currently it play doller.wav by default in >> /usr/local/freeswitch/sounds/currency/en/doller.wav but i want to play EUR >> so how can this possible. >> >> Is that any easy way to do this thing in multi language currency play in >> say application. >> >> i use this syntax in my free-switch dial plan >> $dialstring = "<action application=\"say\" data=\"en CURRENCY PRONOUNCED >> $credit_balance\"/>"; >> >> Thanks In advance... >> >> -- >> Thanks, >> Bhavik Patel >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > ---------- Forwarded message ---------- > From: Avi Marcus <avi(a)avimarcus.net> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 4 Apr 2013 21:16:04 +0300 > Subject: Re: [Freeswitch-users] One of my extensions Caller ID says > Anonymous! > Indeed, outbound_caller_id_name is just a variable. > > To use it you have to tell the dialplan to set effective_caller_id_name=${ > outbound_caller_id_name}. > effective_caller_id_name IS a variable that actually gets used. > (The default dialplan does this.) > > -Avi Marcus > BestFone > > > On Thu, Apr 4, 2013 at 8:26 PM, Steven Schoch < > schoch+freeswitch.org(a)xwin32.com> wrote: > >> In my directory, I set the variable "effective_caller_id_name" instead of >> "internal_caller_id_name". (I also set the "outbound_caller_id_name" >> variable just like your example, although that never seems to get sent to >> the PSTN.) >> >> -- >> Steve >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > ---------- Forwarded message ---------- > From: Steven Ayre <steveayre(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 4 Apr 2013 19:46:22 +0100 > Subject: Re: [Freeswitch-users] Call not originated through event socket > +1 > > '/log 9' and watch the debug output. It'll be very clear about what's > happening. Most likely there's something wrong about your usage of > originate. > > -Steve > > > On 4 April 2013 19:08, Michael Collins <msc(a)freeswitch.org> wrote: > >> You could also just hop on fs_cli and watch the output while you make a >> test call with the event socket. Most likely you'll see what's going on. >> >> -MC >> >> On Thu, Apr 4, 2013 at 10:39 AM, Hector Geraldino < >> Hector.Geraldino(a)ipsoft.com> wrote: >> >>> +OK doesn’t mean the call was generated, it just means that FreeSWITCH >>> received (and processed) the command.**** >>> >>> ** ** >>> >>> You should probably be better served by listening to the events for each >>> one of these calls. There are a few handful events that can tell you if the >>> channel was created, the call was originated, if it was answered and when >>> it was dropped with the drop reason.**** >>> >>> ** ** >>> >>> *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: >>> freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Ashish >>> gautam >>> *Sent:* Thursday, April 04, 2013 5:26 AM >>> *To:* freeswitch-users(a)lists.freeswitch.org >>> *Subject:* [Freeswitch-users] Call not originated through event socket** >>> ** >>> >>> ** ** >>> >>> Hi,**** >>> >>> ** ** >>> >>> I am generating calls through event socket using originate action to >>> multiple numbers to PSTN network. ES is returning '+OK' for the call not >>> generated. Still the call is not generated.**** >>> >>> ** ** >>> >>> Any help is appreciated.**** >>> >>> ** ** >>> >>> Thanks.**** >>> >>> ** ** >>> >>> -Ashish**** >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _______________________________________________ > 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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start_tone_detect, but no event is fired
by henry.houfeng 05 Apr '13

05 Apr '13

Hi, I want to detect call progress tone with start_tone_detect, but I can't get the event fired. My dialplan is: <extension name="skynet51"> <condition field="destination_number" expression="^(\d+)$"> <action application="start_tone_detect" data="1"/> <action application="bridge" data="sofia/gateway/skynet51/$1"/> </condition> </extension> The call senario is: Caller A is a user registered on local FS. Callee B is a user on remote gateway. A-------INVITE------------>B A<------100 Trying---------B A<-----183 with SDP-------B A<------RTP of Ring Tone--B The early media of ring tone or busy tone is transferred from B to A after 183. But I can't see the DETECTED_TONE is fired. I have set loglevel to 3 in spandsp.conf.xml: <descriptors debug-level="3"> I can only find some Tone segment at a frequencey less than 10: freeswitch.log.2013-04-02-02-03-38.1:2013-04-01 21:51:12.618263 [DEBUG] mod_spandsp_dsp.c:645 Tone segment: f1 = 4, f2 = 5, duration = 48 freeswitch.log.2013-04-02-02-03-38.1:2013-04-01 21:51:12.738255 [DEBUG] mod_spandsp_dsp.c:645 Tone segment: f1 = -1, f2 = -1, duration = 128 freeswitch.log.2013-04-02-02-03-38.1:2013-04-01 21:51:12.858249 [DEBUG] mod_spandsp_dsp.c:645 Tone segment: f1 = 4, f2 = 5, duration = 112 freeswitch.log.2013-04-02-02-03-38.1:2013-04-01 21:51:12.958243 [DEBUG] mod_spandsp_dsp.c:645 Tone segment: f1 = -1, f2 = -1, duration = 112 freeswitch.log.2013-04-02-02-03-38.1:2013-04-01 21:51:13.018238 [DEBUG] mod_spandsp_dsp.c:645 Tone segment: f1 = 4, f2 = 5, duration = 48 freeswitch.log.2013-04-02-02-03-38.1:2013-04-01 21:51:13.058236 [DEBUG] mod_spandsp_dsp.c:645 Tone segment: f1 = -1, f2 = -1, duration = 48 Is my dialplan incorrect? I will very appreciate your suggestion. Thanks! Regards, Henry -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/start-tone-detect-but-no-even… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Mod Managed Conf Call
by Usama Zaidi 04 Apr '13

04 Apr '13

Hey, Any updates on when the conference call is going to be scheduled for? The thread kinda died. Regards. -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of freeswitch-users-request(a)lists.freeswitch.org Sent: Tuesday, April 02, 2013 11:34 PM To: freeswitch-users(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 82, Issue 19 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..."

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Call not originated through event socket
by Ashish gautam 04 Apr '13

04 Apr '13

Hi, I am generating calls through event socket using originate action to multiple numbers to PSTN network. ES is returning '+OK' for the call not generated. Still the call is not generated. Any help is appreciated. Thanks. -Ashish

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