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g729 sound format for Intel IPP codecs
by Anita Hall 04 Jan '12

04 Jan '12

Hi! We have been playing with Intel IPP G729 code and its module for FreeSWITCH. As reported here, we have successfully compile the fs-g729 module after install IPP libraries. https://code.google.com/p/fs-g729/issues/detail?id=3#c1 However, to test the codec in a loopback SIP call (FreeSWITCH calling itself over SIP), I need some G729 encoded files (do I really?) FreeSWITCH gives errors with G729 encoded file from here http://www.enicomms.com/cutglassivr/audiofiles/Alison_Keenan-British-Englis… Dialplan <action function="play-file" data="ivr/demo-thanks.g729"/> [ERR] switch_core_file.c:122 Invalid file format [g729] for [/usr/local/freeswitch/sounds/en/us/callie/ivr/demo-thanks.g729]! And also gives errors with file encoded using Intel IPP sample G729 encoder. <action function="play-file" data="ivr/ivr-welcome_to_freeswitch.g729"/> [ERR] switch_core_file.c:122 Invalid file format [g729] for [/usr/local/freeswitch/sounds/en/us/callie/ivr/ivr-welcome_to_freeswitch.g729]! Please note that the files are actually kept in the directory /usr/local/freeswitch/sounds/en/us/callie/ivr/8000 as are other demo_ivr sound files. # file /usr/local/freeswitch/sounds/en/us/callie//ivr/8000/*.g72* /usr/local/freeswitch/sounds/en/us/callie//ivr/8000/demo-thanks.g729: data /usr/local/freeswitch/sounds/en/us/callie//ivr/8000/ivr-welcome_to_freeswitch.g729: RIFF (little-endian) data, WAVE audio, mono 8000 Hz I suspect that ivr-welcome_to_freeswitch.g729 (produced by Intel IPP G729 encoder) is not really G729. Any pointers will be most helpful. Thanks! Anita.

5 6

mod limit for duplicate concurrent calls.
by Rodney 04 Jan '12

04 Jan '12

thanks Vitaly, could someone tell me what date weekly call rupa talked about mod limit? i really would like to setup a method to hangup or transfer to a wav before hanging up anyone trying to call in with the same number ie. three way. if this indeed could work. where would i set this mod limit at? on the inbound extension or on the gateway. thanks. ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, December 27, 2011 8:32 PM Subject: FreeSWITCH-users Digest, Vol 66, Issue 177 ----- Forwarded Message ----- 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: prevention of duplicate calls. (Vitaly Nikolaev) 2. Re: sangoma A200 with mod_freetdm on windows?! (Tamer Higazi) 3. Re: sangoma A200 with mod_freetdm on windows?! (John) 4. Re: Call log - multiple entries CDR?? Billing? (curriegrad2004) Hello, You can try to use mod_limit http://wiki.freeswitch.org/wiki/Mod_limit hash by callerid+callid and set limit 1 call PS: i never used that this way but it might work On Tue, Dec 27, 2011 at 1:11 PM, Rodney <notlikeme75(a)yahoo.com> wrote: is there a method using xml that i can prevent callers three waying themselves. I find some idiots will do this so they can "produce" feedback into a conference room. I would like the system to automatically determine that they are already on the ivr and send them to a recorded message and hangup. or maybe auto hanging up the first call in case of "accidentals" from voips not hanging up and continuing the first call. > > > > >_________________________________________________________________________ >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 > > -- -- Vitaly Nikolaev Hi Moises! I allready sent one: http://support.sangoma.com/index.php?/Tickets/Ticket/View/460 they haven't replied since 2 days :( Tamer Am 27.12.2011 20:17, schrieb Moises Silva: > On Mon, Dec 26, 2011 at 9:00 PM, Tamer Higazi <th982a(a)googlemail.com > <mailto:th982a@googlemail.com>> wrote: > > Hi people! > I got my A200 board running with 1 FXS module on Linux along with > mod_freetdm, but I am facing problems getting it to run on Windows. From > Sangoma I followed the instructions to set up the board on Windows7 > winpipe module, which works so far. > > How do I get freeswitch with mod_freetdm to run on Windows that I can > make use of the board (pbx) on a win machine? > > > You may want to send an email to Sangoma support. They are working > already in a wiki page for Windows setup, in the meantime they can help > you with instructions via email. > > *Moises Silva > **/Software Engineer, Development Manager/*** > > msilva(a)sangoma.com <mailto:msilva@sangoma.com> > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > > > > t. +1 800 388 2475 (N. America) > > t. +1 905 474 1990 x128 > > f. +1 905 474 9223 > > > > > > ** > <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_ca…> > > Products > <http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campa…> | Solutions > <http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_camp…> | Events > <http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&ut…> | Contact > <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_ca…> | Wiki > <http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign…> | Facebook > <http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=sig…> | Twitter > <http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_ca…>`| > | YouTube > <http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=…> > > VegaStream is now part of Sangoma! > <http://sangoma.com/about_us/newsroom/news_releases/news/1488795/Sangoma_Acq…> > > Ask us about both Gateway Appliances > <http://sangoma.com/products/media_gateways/vega.html> and Internal > Gateways > <http://sangoma.com/products/software_products/netborder_suite/netborder_exp…> > > > > _________________________________________________________________________ > 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 Tamer, This is linked from the Sangoma Support front page: http://support.sangoma.com/index.php?/News/NewsItem/View/1/sangoma-holiday-… John On 27/12/11 23:51, Tamer Higazi wrote: > Hi Moises! > I allready sent one: > > http://support.sangoma.com/index.php?/Tickets/Ticket/View/460 > > they haven't replied since 2 days :( > > > Tamer > > Am 27.12.2011 20:17, schrieb Moises Silva: >> On Mon, Dec 26, 2011 at 9:00 PM, Tamer Higazi<th982a(a)googlemail.com >> <mailto:th982a@googlemail.com>> wrote: >> >> Hi people! >> I got my A200 board running with 1 FXS module on Linux along with >> mod_freetdm, but I am facing problems getting it to run on Windows. From >> Sangoma I followed the instructions to set up the board on Windows7 >> winpipe module, which works so far. >> >> How do I get freeswitch with mod_freetdm to run on Windows that I can >> make use of the board (pbx) on a win machine? >> >> >> You may want to send an email to Sangoma support. They are working >> already in a wiki page for Windows setup, in the meantime they can help >> you with instructions via email. >> >> *Moises Silva >> **/Software Engineer, Development Manager/*** >> >> msilva(a)sangoma.com<mailto:msilva@sangoma.com> >> >> Sangoma Technologies >> >> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada >> >> >> >> >> t. +1 800 388 2475 (N. America) >> >> t. +1 905 474 1990 x128 >> >> f. +1 905 474 9223 >> >> >> >> >> >> ** >> <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_ca…> >> >> Products >> <http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campa…> | Solutions >> <http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_camp…> | Events >> <http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&ut…> | Contact >> <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_ca…> | Wiki >> <http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign…> | Facebook >> <http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=sig…> | Twitter >> <http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_ca…>`| >> | YouTube >> <http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=…> >> >> VegaStream is now part of Sangoma! >> <http://sangoma.com/about_us/newsroom/news_releases/news/1488795/Sangoma_Acq…> >> >> Ask us about both Gateway Appliances >> <http://sangoma.com/products/media_gateways/vega.html> and Internal >> Gateways >> <http://sangoma.com/products/software_products/netborder_suite/netborder_exp…> >> >> >> >> _________________________________________________________________________ >> 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 > > _________________________________________________________________________ > 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 > > This looks like a misconfigured box. Try setting the cdr_csv.conf.xml conf file's legs param to a only: <param name="legs" value="a"/> If it is using XML CDR's then you'll need to go into the xml_cdr.conf.xml and set this param: <param name="log-b-leg" value="true"/> to false <param name="log-b-leg" value="false"/> All of the configuration files can be found under the autoload_configs in the conf root of your FreeSWITCH configuration folder... assuming you are using the default dialplan configuration On Tue, Dec 27, 2011 at 4:10 AM, Sherif Omran <sherifomran2000(a)yahoo.com> wrote: out 1002 1002 1001 12/27/2011 11:30:48 am 0:00:47 Details >out 1002 1002 1001 12/27/2011 11:30:48 am 0:00:47 Details > >Call logs are duplicated for 1 call? How can i prevent this? > > > > > >--- On Tue, 12/27/11, curriegrad2004 <curriegrad2004(a)gmail.com> wrote: > > >>From: curriegrad2004 <curriegrad2004(a)gmail.com> >>Subject: Re: [Freeswitch-users] Call log - multiple entries CDR?? Billing? >>To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> >>Date: Tuesday, December 27, 2011, 6:33 AM >> >> >>xml_cdr does the job just fine... uuid_bridge is what you may want to be looking for >> >> >>On Mon, Dec 26, 2011 at 7:20 PM, Sherif Omran <sherifomran2000(a)yahoo.com> wrote: >> >>Hi, >>> >>>I have the CDR enabled and see multiple logs for the same call. Can any body recommend a call log that works fine and could be extended to be used for billing? >>> >>>thanks in advance >>> >>>regards, >>>Sherif Omran >>> >>> >>>_________________________________________________________________________ >>>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 >>> >>> >> >>-----Inline Attachment Follows----- >> >> >> >>_________________________________________________________________________ >>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 >> >_________________________________________________________________________ >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

5 9

transfer to dynamic con from static
by Rodney 04 Jan '12

04 Jan '12

everytime i use transfer from caller controls i get "destination out of order" on console. so i use the execute extension and get problems. is this correct? <control action="execute_application" digits="#" data="transfer 759 xml default"/> `rodney ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, December 22, 2011 6:30 PM Subject: FreeSWITCH-users Digest, Vol 66, Issue 145 ----- Forwarded Message ----- 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: Some questions regarding IPs and gateway (Michael Collins) 2. transfer conference to conference (Rodney) 3. Re: Odp: transfer to conference from conference / caller controls (Anthony Minessale) The "domain" method you used is actually a good way to go if you're always wanting the same IP for all network comms. -MC On Thu, Dec 22, 2011 at 3:19 PM, <georg(a)riseup.net> wrote: > You want to set the IP address in the SIP profile. Remember, FreeSWITCH >> can >> use multiple IP addresses and multiple ports, so it's not like Asterisk at >> all in that respect. You control which IP/Port each SIP profile uses. If >> you tell the SIP profile explicitly to use a specific IP address then >> that's what it will use and will ignore the other NICs. > >I achieved this now trough another way, see my other mail for this. >I this "a no-go"? > >I already set the ips in my internal and my external profile, however, FS >was still binding itself to the wan-ip (which is the way it normally >works, if I'm right?). My plan was to change this and exclude especially >the wan-ip. > >Georg > > >_________________________________________________________________________ >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 > actually i just tried this and even though it looked good on screen, i was still just in the static conference i tried to transfer from, so i am back to square one :( condition destination_number ^760$ 1 action answer 3 action set conference_user_list=| 10 action play_and_get_digits 4 4 3 5000 # askprivateroom2.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+ 20 action phrase spell,${target_num} 30 action sched_hangup +10 normal_clearing bleg 35 action bridge loopback/app=conference:${target_num}-127.0.0.1@dynamic_conf You should be using transfer not execute_extension or you are going into the new conference mid digit parse and blocking in the first conference waiting to return. If you use the transfer app then you will exit the current conference and transfer to the new one. On Dec 22, 2011 2:42 PM, "Rodney" <notlikeme75(a)yahoo.com> wrote: Mariusz, using your suggestion, i worked out a solution that works. although i have to drop the bleg with sched_hangup in order for it to work with my hangup back to ivr method. it still looks like crap in my cdr but I can deal with it for now since i am crazy for wanting to transfer from static conference right to dynamic :) but doing it with this loopback method seems to give me the caller controls back. thanks for your help. ~rodney > > condition destination_number ^760$ 1 > action answer 3 > action set conference_user_list=| 10 > action play_and_get_digits 4 4 3 5000 # ask4digit.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+ 20 > action phrase spell,${target_num} 30 > action sched_hangup +10 normal_clearing bleg 35 > action bridge loopback/app=conference:${target_num}-127.0.0.1@dynamic_conf > > > > 4. Odp: transfer to conference from conference / caller controls > (Mariusz Czulada) > > >> > >Perhaps this will be useful for you: > >Dialplan: > > <extension name="bind_and_join"> > ... > <action application="bind_meta_app" data="4 a s execute_extension::sub_conf"/> > ... > <action application="conference" data="mainconf@cx"/> > ... > </extension> > > <extension name="sub_conf"> > ... > <action application="bridge" data="loopback/app=conference:${conference_name}_sub@cx_sub"/> > ... > </extension> > >conference.conf.xml: > > <group name="cx_controls"> > ... > <control action="hangup" digits="9"/> > </group> > > <group name="cx_sub_controls"> > ... > <control action="hangup" digits="4"/> > </group> > > <profile name="cx"> > ... > <param name="caller-controls" value="cx_controls"/> > </profile> > <profile name="cx_sub"> > ... > <param name="caller-controls" value="cx_sub_controls"/> > </profile> > > >Dnia 20-12-2011 o godz. 18:21 Rodney napisał(a): >I am trying to transfer to a dynamic conference from within a static conference but when I do my callers loose control and can not use any dtmf caller control options and must hang up to get back. I tried using clear digit action but do not think this works on the "conf" realm. the dynamic conference works fine with all controls if i send them directly from the ivr but for what i am trying it is important to do this transfer from another conference. is there a method transfer and release all previous controls? i even tried using a difference conference profile so i transferred from static 501@default to <digits>@dynamic to see if giving new controls would work, to no avail. is there a log or something i can post that will help you in understanding how this is happening? or is it even fixable? thanks rodney >> >> >> <group name="default"> >> <control action="mute" digits="1"/> >> <control action="hangup" digits="0"/> >> <control action="execute_application" digits="4" data="playback C:/Program Files/FreeSWITCH/recordings/conf_help.wav"/> >> <control action="execute_application" digits="8" data="execute_extension ANNOUNCE_CONF_COUNT_PRIVATE XML default"/> >> <control action="execute_application" digits="#" data="execute_extension 759 xml default"/> >> >>- <group name="dynamic"> >> <control action="mute" digits="1"/> >> <control action="execute_application" digits="0" data="execute_extension 401 XML default"/> (main ivr menu extension) >> <control action="execute_application" digits="8" data="execute_extension ANNOUNCE_CONF_COUNT_PRIVATE XML default"/> >> >> >> >> >> >>condition destination_number ^759$ 1 >> action answer 2 >> action set conference_user_list=| 11 >> action play_and_get_digits 4 4 3 5000 # askprivateroom.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+ 15 >> action phrase spell,${target_num} 16 >> action conference ${target_num}-127.0.0.1@default >> > > > >_______________________________________________ >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 > > > >_________________________________________________________________________ >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

5 9

Re: [Freeswitch-users] mod limit for duplicate concurrent calls.
by Rodney 04 Jan '12

04 Jan '12

vitaly, is it possible to put this limit before it even gets to my greeting/preamble? i don't even want them hitting the IVR. plus when does that count reset? i want the counter to reset once they hangup all current calls. ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, January 4, 2012 12:17 AM Subject: FreeSWITCH-users Digest, Vol 67, Issue 29 ----- Forwarded Message ----- 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: conference.conf missing? (Michael Collins) 2. A New Config Set aimed for New People to FreeSWITCH (curriegrad2004) 3. Re: FreeSWITCH-users Digest, Vol 67, Issue 7 (Sharad Garg) 4. Re: mod limit for duplicate concurrent calls. (Vitaly Nikolaev) It means something happened to /usr/local/freeswitch/conf/autoload_configs/conference.conf.xml, assuming that's your path. If you did not have a customized conference.conf.xml file then you can just copy the the file from your FS source directory under conf/autoload_configs/ -MC On Tue, Jan 3, 2012 at 6:00 PM, Sherif Omran <sherifomran2000(a)yahoo.com> wrote: Hello every body, > >I get the following error, any clue? > >2012-01-04 01:52:14.234297 [ERR] mod_conference.c:7491 Open of conference.conf failed > >thanks > >regards, >Sherif Omran > > > >_________________________________________________________________________ >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 > > One of the frequent complaints is that the default diaplan is huge... Well, this has been discussed about in one of the earlier conference calls last year and so far nobody has taken up any work on creating this configuration set, so I decided to take up the work on creating one. It's been done and I have it uploaded on my github. Testing and feedback is greatly appreciated: https://github.com/curriegrad2004/Freeswitch_slim_conf "A configuration set created for users new to FreeSWITCH. It doesn't expose what FreeSWITCH is truly capable of, but I hope new users won't get scared with this configuration set." as quoted from my description page on github. All these are basically IP based PBX systems having PBX features. Regards ----- Original Message ----- From: <freeswitch-users-request(a)lists.freeswitch.org> To: <freeswitch-users(a)lists.freeswitch.org> Sent: Tuesday, January 03, 2012 7:47 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 7 > 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. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > CallWeaver, and YATE. (Kaushal Shriyan) > 2. Re: mod_sofia not binding to port 5060 (Brandon McGinty) > 3. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > CallWeaver, and YATE. (Zohair Raza) > 4. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > CallWeaver, and YATE. (Giovanni Maruzzelli) > 5. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > CallWeaver, and YATE. (Kaushal Shriyan) > 6. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > CallWeaver, and YATE. (Zohair Raza) > 7. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > CallWeaver, and YATE. (Vitaly Nikolaev) > -------------------------------------------------------------------------------- > _______________________________________________ > 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 > You need to check limit in dial plan before it hit conference, here is example: <extension name="limit_exceeded"> <condition field="destination_number" expression="^limit_exceeded$"> <action application="playback" data="ivr/ivr-call_rejected.wav"/> <action application="hangup"/> </condition> </extension> <extension name="conference_test_limit" continue="true"> <condition field="destination_number" expression="^999$"> <action application="limit" data="hash ${sip_auth_username} ${destination_number} 1"/> </condition> </extension> <extension name="conference_test"> <condition field="destination_number" expression="^999$"> <action application="conference" data="3002"/> </condition> </extension> if same phone call 999 more then one time, it will get to limit_exceeded ext. On Jan 3, 2012, at 7:53 PM, Rodney wrote: thanks Vitaly, > > >could someone tell me what date weekly call rupa talked about mod limit? i really would like to setup a method to hangup or transfer to a wav before hanging up anyone trying to call in with the same number ie. three way. if this indeed could work. where would i set this mod limit at? on the inbound extension or on the gateway. thanks. > > > > > >________________________________ > From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> >To: freeswitch-users(a)lists.freeswitch.org >Sent: Tuesday, December 27, 2011 8:32 PM >Subject: FreeSWITCH-users Digest, Vol 66, Issue 177 > >----- Forwarded Message ----- > >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: prevention of duplicate calls. (Vitaly Nikolaev) > 2. Re: sangoma A200 with mod_freetdm on windows?! (Tamer Higazi) > 3. Re: sangoma A200 with mod_freetdm on windows?! (John) > 4. Re: Call log - multiple entries CDR?? Billing? (curriegrad2004) > >Hello, > >You can try to use mod_limit > >http://wiki.freeswitch.org/wiki/Mod_limit > >hash by callerid+callid and set limit 1 call > >PS: i never used that this way but it might work > > > > > >On Tue, Dec 27, 2011 at 1:11 PM, Rodney <notlikeme75(a)yahoo.com> wrote: > >is there a method using xml that i can prevent callers three waying themselves. I find some idiots will do this so they can "produce" feedback into a conference room. I would like the system to automatically determine that they are already on the ivr and send them to a recorded message and hangup. or maybe auto hanging up the first call in case of "accidentals" from voips not hanging up and continuing the first call. >> >> >> >> >>_________________________________________________________________________ >>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 >> >> > > >-- >-- >Vitaly Nikolaev >Hi Moises! >I allready sent one: > >http://support.sangoma.com/index.php?/Tickets/Ticket/View/460 > >they haven't replied since 2 days :( > > >Tamer > >Am 27.12.2011 20:17, schrieb Moises Silva: >> On Mon, Dec 26, 2011 at 9:00 PM, Tamer Higazi <th982a(a)googlemail.com >> <mailto:th982a@googlemail.com>> wrote: >> >> Hi people! >> I got my A200 board running with 1 FXS module on Linux along with >> mod_freetdm, but I am facing problems getting it to run on Windows. From >> Sangoma I followed the instructions to set up the board on Windows7 >> winpipe module, which works so far. >> >> How do I get freeswitch with mod_freetdm to run on Windows that I can >> make use of the board (pbx) on a win machine? >> >> >> You may want to send an email to Sangoma support. They are working >> already in a wiki page for Windows setup, in the meantime they can help >> you with instructions via email. >> >> *Moises Silva >> **/Software Engineer, Development Manager/*** >> >> msilva(a)sangoma.com <mailto:msilva@sangoma.com> >> >> Sangoma Technologies >> >> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada >> >> >> >> >> t. +1 800 388 2475 (N. America) >> >> t. +1 905 474 1990 x128 >> >> f. +1 905 474 9223 >> >> >> >> >> >> ** >> <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_ca…> >> >> Products >> <http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campa…> | Solutions >> <http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_camp…> | Events >> <http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&ut…> | Contact >> <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_ca…> | Wiki >> <http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign…> | Facebook >> <http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=sig…> | Twitter >> <http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_ca…>`| >> | YouTube >> <http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=…> >> >> VegaStream is now part of Sangoma! >> <http://sangoma.com/about_us/newsroom/news_releases/news/1488795/Sangoma_Acq…> >> >> Ask us about both Gateway Appliances >> <http://sangoma.com/products/media_gateways/vega.html> and Internal >> Gateways >> <http://sangoma.com/products/software_products/netborder_suite/netborder_exp…> >> >> >> >> _________________________________________________________________________ >> 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 > > > >Tamer, > >This is linked from the Sangoma Support front page: > >http://support.sangoma.com/index.php?/News/NewsItem/View/1/sangoma-holiday-… > >John > >On 27/12/11 23:51, Tamer Higazi wrote: >> Hi Moises! >> I allready sent one: >> >> http://support.sangoma.com/index.php?/Tickets/Ticket/View/460 >> >> they haven't replied since 2 days :( >> >> >> Tamer >> >> Am 27.12.2011 20:17, schrieb Moises Silva: >>> On Mon, Dec 26, 2011 at 9:00 PM, Tamer Higazi<th982a(a)googlemail.com >>> <mailto:th982a@googlemail.com>> wrote: >>> >>> Hi people! >>> I got my A200 board running with 1 FXS module on Linux along with >>> mod_freetdm, but I am facing problems getting it to run on Windows. From >>> Sangoma I followed the instructions to set up the board on Windows7 >>> winpipe module, which works so far. >>> >>> How do I get freeswitch with mod_freetdm to run on Windows that I can >>> make use of the board (pbx) on a win machine? >>> >>> >>> You may want to send an email to Sangoma support. They are working >>> already in a wiki page for Windows setup, in the meantime they can help >>> you with instructions via email. >>> >>> *Moises Silva >>> **/Software Engineer, Development Manager/*** >>> >>> msilva(a)sangoma.com<mailto:msilva@sangoma.com> >>> >>> Sangoma Technologies >>> >>> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada >>> >>> >>> >>> >>> t. +1 800 388 2475 (N. America) >>> >>> t. +1 905 474 1990 x128 >>> >>> f. +1 905 474 9223 >>> >>> >>> >>> >>> >>> ** >>> <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_ca…> >>> >>> Products >>> <http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campa…> | Solutions >>> <http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_camp…> | Events >>> <http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&ut…> | Contact >>> <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_ca…> | Wiki >>> <http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign…> | Facebook >>> <http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=sig…> | Twitter >>> <http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_ca…>`| >>> | YouTube >>> <http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=…> >>> >>> VegaStream is now part of Sangoma! >>> <http://sangoma.com/about_us/newsroom/news_releases/news/1488795/Sangoma_Acq…> >>> >>> Ask us about both Gateway Appliances >>> <http://sangoma.com/products/media_gateways/vega.html> and Internal >>> Gateways >>> <http://sangoma.com/products/software_products/netborder_suite/netborder_exp…> >>> >>> >>> >>> _________________________________________________________________________ >>> 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 >> >> _________________________________________________________________________ >> 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 >> >> > > > > >This looks like a misconfigured box. Try setting the cdr_csv.conf.xml conf file's legs param to a only: > > ><param name="legs" value="a"/> > > >If it is using XML CDR's then you'll need to go into the xml_cdr.conf.xml and set this param: > <param name="log-b-leg" value="true"/> >to false > <param name="log-b-leg" value="false"/> > > >All of the configuration files can be found under the autoload_configs in the conf root of your FreeSWITCH configuration folder... assuming you are using the default dialplan configuration > > >On Tue, Dec 27, 2011 at 4:10 AM, Sherif Omran <sherifomran2000(a)yahoo.com> wrote: > >out 1002 1002 1001 12/27/2011 11:30:48 am 0:00:47 Details >>out 1002 1002 1001 12/27/2011 11:30:48 am 0:00:47 Details >> >>Call logs are duplicated for 1 call? How can i prevent this? >> >> >> >> >> >>--- On Tue, 12/27/11, curriegrad2004 <curriegrad2004(a)gmail.com> wrote: >> >> >>>From: curriegrad2004 <curriegrad2004(a)gmail.com> >>>Subject: Re: [Freeswitch-users] Call log - multiple entries CDR?? Billing? >>>To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> >>>Date: Tuesday, December 27, 2011, 6:33 AM >>> >>> >>>xml_cdr does the job just fine... uuid_bridge is what you may want to be looking for >>> >>> >>>On Mon, Dec 26, 2011 at 7:20 PM, Sherif Omran <sherifomran2000(a)yahoo.com> wrote: >>> >>>Hi, >>>> >>>>I have the CDR enabled and see multiple logs for the same call. Can any body recommend a call log that works fine and could be extended to be used for billing? >>>> >>>>thanks in advance >>>> >>>>regards, >>>>Sherif Omran >>>> >>>> >>>>_________________________________________________________________________ >>>>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 >>>> >>>> >>> >>>-----Inline Attachment Follows----- >>> >>> >>> >>>_________________________________________________________________________ >>>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 >>> >>_________________________________________________________________________ >>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 > > >_________________________________________________________________________ >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

2 1

ENUM
by Miha Zoubek 04 Jan '12

04 Jan '12

Hi, I need a pit of help for setting up ENUM. My outbount calls are going throught just fine. What I need is that, number get from enum are using for dialing in my dial plan. In enum.conf I have changed:(xxx.xxx.xxx.xxx is dns of my enum): <param name="default-root" value="xxx.xxx.xxx.xxx"/> <param name="default-isn-root" value="xxx.xxx.xxx.xxxi"/> <route service="E2U+SIP" regex="sip:(.*)" replace="sofia/gateway/sbc_trunk/$1"/ In my dial plan I have set: <action application="enum" data="386$1xxx.xxx.xxx.xxx"/> <action application="bridge" data="${enum_auto_route}"/ I get this is wrong as my calls not working:) Please help me out how to use number from enum in dialplan. Regards, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o.

1 1

Silence ESL event
by Hynek Cihlar 04 Jan '12

04 Jan '12

Hi all, is there any way to receive a silence-detected event through ESL? The use case is to hangup the call in case nothing interesting is going on on the channel. Sent from my mobile device

2 2

error on git update
by Charles Bujold 04 Jan '12

04 Jan '12

Tried updating Freeswitch (last update from GIT done Dec 1 2010) today and I'm getting the following error. Can somebody identify what needs to be done to get a proper update. Using Ubuntu 10.4 server. Never had a problem before! In file included from /usr/src/freeswitch.git/libs/spandsp/src/spandsp.h:101, from ./src/include/private/switch_core_pvt.h:35, from src/switch_apr.c:37: /usr/src/freeswitch.git/libs/spandsp/src/spandsp/t4_tx.h:145: error: expected declaration specifiers or '...' before 'tz_t' make[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1 make[2]: Leaving directory `/usr/src/freeswitch.git' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch.git' make: *** [current] Error 2 Thanks CJB

3 2

timeout for curl
by Yuriy Nasida 04 Jan '12

04 Jan '12

Hello FS users, I do some curl request from my dialplan. It looks like this:<action application="set" data="lookup=${curl(http://mycnameprovider.com/cname.php?number=${myvar})}" inline="true"/>If cname provider doesn't answer for long time, I have issue because FS doesn't process a call.whether there are any method to add some timeout for curl (3 sec for example) ? Thus, if cname provider doesn't answer for 3 sec, FS must continue the logic of dialplan. I read about mod_cidlookup but it looks like that this module have same issuehttp://wiki.freeswitch.org/wiki/Mod_cidlookup Please advise Thanks.

2 1

Voicemail - see if user left a message?
by Avi Marcus 03 Jan '12

03 Jan '12

Can the xml_cdrs tell me if the user actually left a voicemail, rather than merely that VM picked up? I don't see where. I know it's on ESL, and it hits the voicemail email code.. what's the easiest way to integrate it with the CDRs? preferably with the xml_cdrs? Maybe a patch to the xml_cdr mod? Or the easiest is to have mod_voicemail set a channel variable once a voicemail gets marked as saved or the like? Is there currently a solution for this that I missed? Thanks, -Avi

2 4

conf count when entering room
by Rodney 03 Jan '12

03 Jan '12

MC, I have not figured out how to get the conference count upon entering the room. I would imaging loading the Conference Announce Count extension at the same time as dropping into the conference. I am unsure of how to do this so it gets passed the conference number i am entering. I would imagine this would be a good start: ? <extension name="room1"> <condition field="destination_number" expression="^301$"> <action application="set" data="conference_user_list=|"/>  <action application="clear_digit_action" data="all"/> <action application="bind_digit_action" data="moderator,5,exec:transfer,555 XML default"/> <action application="bind_digit_action" data="moderator,6,exec:transfer,319 XML default"/> <action application="bind_digit_action" data="moderator,7,exec:transfer,302 XML default"/> <action application="bind_digit_action" data="moderator,#,exec:transfer,759 XML default"/> <action application="bind_digit_action" data="moderator,0,exec:transfer,300 XML default"/> <action application="digit_action_set_realm" data="moderator"/> <action application="playback" data="C:/Program Files/FreeSWITCH/sounds/en/us/callie/digits/8000/1.wav"/>  <action application="conference" data="301-127.0.0.1@default"/> <action application="transfer" data="300 xml default" /> </condition></extension> ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, January 3, 2012 4:47 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 20 ----- Forwarded Message ----- 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: conf count (Michael Collins) 2. Re: Radio stream binding ? (Michael Collins) 3. Re: Radio stream binding ? (Rob Hutton) 4. Re: int/ext dial tones not synchronous (Michael Collins) 5. timeout for curl (Yuriy Nasida) 6. Re: Silence ESL event (Michael Collins) Did you figure this one out yet? -MC On Sun, Dec 25, 2011 at 8:20 AM, Rodney <notlikeme75(a)yahoo.com> wrote: > >I can select the conf-alone sound in caller controls and even set up a bind digit for caller count once i am already in the conference using the > >Conference Announce Count Inline extension. > >but what i cant figure out is how to speak the caller count when entering the conference room. can anyone help with this? currently with conf-alone it only speaks the file when i am the only one left or there isnt anyone else in the room when i get there. i would like it to tell me the total callers of the room i am heading into. > >thanks. > >_________________________________________________________________________ >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 > > On Fri, Dec 23, 2011 at 9:05 PM, Sherif Omran <sherifomran2000(a)yahoo.com> wrote: but the station i need is this > >mmsh://live.sis.gov.eg/live?MSWMExt=.asf > >and it is not there. I would appreciate your help > Bummer. ASF is a proprietary Microsoft technology, so I would contact the people using it and let them know that they are limiting their potential audience by using a locked-down container format. If you *really* want this then you can probably hire someone to do a workaround. My guess is that you'll need to pay a fair amount because the people with the requisite skills to do this will be in high demand. -MC If you can get permission to rebroadcast, you could set up a machine to transcode it into a format that is supportable... On Tuesday 03 January 2012 15:44:34 Michael Collins wrote: > On Fri, Dec 23, 2011 at 9:05 PM, Sherif Omran <sherifomran2000(a)yahoo.com>wrote: > > > but the station i need is this > > > > mmsh://live.sis.gov.eg/live?MSWMExt=.asf > > > > and it is not there. I would appreciate your help > > > Bummer. > > ASF is a proprietary Microsoft technology, so I would contact the people > using it and let them know that they are limiting their potential audience > by using a locked-down container format. > > If you *really* want this then you can probably hire someone to do a > workaround. My guess is that you'll need to pay a fair amount because the > people with the requisite skills to do this will be in high demand. > > -MC > I don't believe you can reliable put these tones in synch. I've never heard of any telephony system that does so, although I admit my experience is limited to legacy PBXes and FreeSWITCH. -MC On Fri, Dec 23, 2011 at 8:47 PM, <georg(a)riseup.net> wrote: Hi, > >> I'm not sure that I understand what the issue is here. What two "tones" >> are >> not in sync? You said "dial tone" but I'm pretty sure that's not what you >> mean. Are you talking about the ringback tone that the caller hears? > >Exactly Michael, I was talking about the ringback tone. >I didn't know the correct expression and used a translation engine... > >Georg > > >_________________________________________________________________________ >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 > Hello FS users, I do some curl request from my dialplan. It looks like this: <action application="set" data="lookup=${curl(http://mycnameprovider.com/cname.php?number=${myvar})}" inline="true"/> If cname provider doesn't answer for long time, I have issue because FS doesn't process a call. whether there are any method to add some timeout for curl (3 sec for example) ? Thus, if cname provider doesn't answer for 3 sec, FS must continue the logic of dialplan. I read about mod_cidlookup but it looks like that this module have same issue http://wiki.freeswitch.org/wiki/Mod_cidlookup Please advise Thanks. You might be able to produce the same behavior with the "wait_for_silence" dp application. -MC On Wed, Dec 28, 2011 at 3:20 AM, Hynek Cihlar <hynek.cihlar(a)gmail.com> wrote: Hi all, > >is there any way to receive a silence-detected event through ESL? > >The use case is to hangup the call in case nothing interesting is >going on on the channel. > >Sent from my mobile device > >_________________________________________________________________________ >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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