- FreeSWITCH-users - FreeSWITCH Mailing Lists

originate
by David Villasmil 05 Apr '12

05 Apr '12

Hello guys, i'm trying to execute an originate originate sofia/external/669448337(a)1.2.3.4 0013058883456 +OK a27c762c-7f4a-11e1-9269-f52ca8b46747 2012-04-05 20:10:32.893284 [ERR] switch_core_io.c:929 Codec PROXY PASS-THROUGH encoder error! WHY do i see that error? codecs are fine both on FS and and the endpoint... I originally want to originate and then generate dtmf to the leg... if anyone can give me a help, i would appreciate it thanks! David

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Can ringback be variable in bridge command?
by Mario G 05 Apr '12

05 Apr '12

Testing the latest git has introduced an issue but I wonder if I can get around it by combining the snippet below into one bridge command. If I can set the ringback in the bridge it may fix it. I did not see this in the wiki. Can it be done? Mario G <action application="bridge" data="[originate_timeout=17]group/home@${domain_name}"/> <action application="set" data="ringback=${us-ring}"/> <action application="bridge" data="[originate_timeout=1]group/MGT@${domain_name}"/>

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Playing additional file when transferring call
by Klaus Hochlehnert 05 Apr '12

05 Apr '12

Hi, I have a question if this is possible and how: When transferring a call (but only when it was picked up and from an external source) I'd like to play a complete sound file till the end before going to "transfer_ringback". Does anyone have an idea how to do that? Thanks Klaus

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Re: [Freeswitch-users] FreeSWITCH-Billing
by David Villasmil 05 Apr '12

05 Apr '12

Hello Dihn, Again, read the readme. You need to set the variables accordingly. You need to have apache2 running and set the www directory. you also need to run that as root. David On Wed, Apr 4, 2012 at 6:43 PM, dinh thuc <dinhthuc1303(a)gmail.com> wrote: > Hi David !! > > Thank for quick reply ! I have install mysql success. But I have other > error when I copy file > > > ERROR 1227 (42000) at line 553: Access denied; you need the SUPER > privilege for this operation > > copying scripts: cp -r /root/davidcsi-FreeSWITCH-Billing-f079e0f/gateway/* > /usr/local/freeswitch/ > copying scripts: cp -r > /root/davidcsi-FreeSWITCH-Billing-f079e0f/gateway_scripts/* > /home/freeswitch/ > Creating default Sofia configuration > > Couldn't get sofia.conf.xml! at ./install.pl line 127. > > I can copy manual file ? > > B&R > > Dinh Thuc > > > Vào 21:34 Ngày 04 tháng 4 năm 2012, David Villasmil < > david.villasmil.work(a)gmail.com> đã viết: > > Dihn, >> >> You need to read the readme. You must have mysql installed. >> >> David >> >> >> On Wed, Apr 4, 2012 at 12:39 PM, dinh thuc <dinhthuc1303(a)gmail.com>wrote: >> >>> I have run "install.pl" but I have some error !! >>> >>> sh: mysql: command not found >>> sh: mysql: command not found >>> sh: mysql: command not found >>> copying scripts: cp -r >>> /root/davidcsi-FreeSWITCH-Billing-f079e0f/gateway/* /usr/local/freeswitch/ >>> cp: target `/usr/local/freeswitch/' is not a directory >>> copying scripts: cp -r >>> /root/davidcsi-FreeSWITCH-Billing-f079e0f/gateway_scripts/* /home/david/ >>> cp: target `/home/david/' is not a directory >>> Creating default Sofia configuration >>> /usr/local/freeswitch//conf/autoload_configs/sofia.conf.xml: No such >>> file or directory >>> Couldn't get sofia.conf.xml! at ./install.pl line 127. >>> >>> >>> B&R >>> >>> Dinh Thuc >>> >>> Vào 17:26 Ngày 04 tháng 4 năm 2012, David Villasmil < >>> david.villasmil.work(a)gmail.com> đã viết: >>> >>> Hello Dihn, >>>> >>>> There is a script called "install.pl", you need to execute that. >>>> >>>> David >>>> >>>> >>>> On Wed, Apr 4, 2012 at 7:02 AM, dinh thuc <dinhthuc1303(a)gmail.com>wrote: >>>> >>>>> Hi David !! >>>>> >>>>> We need to solution about billing in my system. I have read link >>>>> https://github.com/davidcsi/FreeSWITCH-Billing >>>>> >>>>> You have script for install all ? >>>>> >>>>> I want to install and test >>>>> >>>>> B&R >>>>> >>>>> Dinh Thuc >>>>> >>>> >>>> >>> >> >

1 0

(no subject)
by azza miled 05 Apr '12

05 Apr '12

I am trying to install FS in a debian machin. I usd the tutoriel on wiki.When I type make, I obtain this error msg cc1: all warnings being treated as errors make[9]: *** [nua_subnotref.lo] Erreur 1 make[8]: *** [all] Erreur 2 make[8]: *** Pas de règle pour fabriquer la cible « nua/libnua.la », nécessaire pour « libsofia-sip-ua.la ». Arrêt. make[7]: *** [all-recursive] Erreur 1 Making all in packages make[6]: *** [all-recursive] Erreur 1 make[5]: *** [all] Erreur 2 make[4]: *** [/usr/local/src/FreeSWITCH/libs/sofia-sip/libsofia-sip-ua/ libsofia-sip-ua.la] Erreur 2 make[3]: *** [mod_sofia-all] Erreur 1 make[2]: *** [all-recursive] Erreur 1 make[1]: *** [all-recursive] Erreur 1 make: *** [all] Erreur 2 Can any one help, please

2 1

collect dtmf events for a timed period
by adam harris 05 Apr '12

05 Apr '12

How can I set to recevice dtmf events for a timed period in lua. I want to evaluate each key press so this requires consuming events or callback I would guess. Any sugesstions or examples

2 1

3-user conference from 2-user call
by lorentz.nails 05 Apr '12

05 Apr '12

I would like to create a 3-user conference from a 2-user call (bridge) without calling the third user twice. The scenario is the following: 1. user 1000 calls user 1001 (that responds); 2. user 1001 may decide to create a 3-user conference by transferring the execution to an extension that use conference_set_auto_outcall to invite user 1002 and create the conference. <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <extension name="cf"> <condition field="destination_number" expression="^cf$"> <action application="answer"/> <action application="transfer" data="-both 3000 XML default"/> </condition> </extension> <extension name="nb_conferences"> <condition field="destination_number" expression="^(3000)$"> <action application="answer"/> <action application="set" data="conference_auto_outcall_timeout=20"/> <action application="set" data="conference_auto_outcall_flags=none"/> <action application="set" data="conference_auto_outcall_caller_id_name=${effective_caller_id_name}"/> <action application="set" data="conference_auto_outcall_caller_id_number=${effective_caller_id_number}"/> <action application="set" data="conference_auto_outcall_profile=default"/> <action application="conference_set_auto_outcall" data="user/1002@ ${domain}"/> <action application="conference" data="$1@default"/> </condition> </extension> At this point user 1002 receive two simultaneous calls: one from user 1000 and one from user 1001 (I'm using softphones). The behavior I would like to achieve is that user 1002 only gets one call from user 1001 or from the conference. Lorentz

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Is it possible to ring user without bridging?
by Mario G 04 Apr '12

04 Apr '12

Is there a way to ring an internal user/group but not have it participate in the incoming call/bridge? I know sounds strange but I have a situation where I need to call a user group for 1 second (different ring on the phones) and proceeding in the extension dialplan. I tried the wiki on this one too. It has been dialed using bridge but that is causing problems if a call is picked up during that 1 ring. Thanks for any help. Mario G

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Terminal Emulators
by Michael Collins 04 Apr '12

04 Apr '12

Hey folks, Today on the conference call we had a brief but passionate discussion about terminal emulators. I'm finding that more and more I'm using all of the "big 3" operating systems and that they each have their own quirks and peculiarities. I decided to start a wiki page on the topic: http://wiki.freeswitch.org/wiki/Terminal_emulators I have been spending more time than I really want to on a Windows laptop so I have been getting familiar with putty. I added some putty info to the wiki plus a few other tidbits. I would like to invite everyone to hop on and add whatever knowledge or tips & tricks that they've learned over the years. If you know of a good terminal program that you use then feel free to add it to the list. Thanks! -MC

1 0

Handing calls coming from Kamailio
by Ali Jawad 04 Apr '12

04 Apr '12

Hi I have the following setup sip phone --- Kamailio for Registration and Routing - Freeswitch for billing and media handling - VOIP Provider Calls are sent from Kamailio to FS and Authentication works as it should, but I cant get the calls to be forwarded to the VOIP provider when they arrive from Kamailio, if I register directly on FS the calls are delivered to the VOIP provider and I can make a call, I did try to create an inbound route but that did not work. Any hints on what I have missed please ? Thanks

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