October 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Real-time billing application for the FreeSWITCH (mod_lua, mod_perl or ESL)
by Sami Montour 04 Mar '15

04 Mar '15

Hello Everyone, I am a long time user of OpenSIPS and have recently been tasked with developing a call control/real-time billing application for the FreeSWITCH. I would like to develop the application in either C/C++, Perl or lua (totally new to lua) in this order. I have used ESL to develop some basic functionality of the application using Perl. For instance, when the call comes in, I pass over control from FreeSWITCH to the call control application using outbound TCP socket. Then the call control application listens for CHANNEL_ANSWER event to start billing process and CHANNEL_HANGUP event to stop billing. The application does some database work as well to retrieve and check user status and balance and all that stuff. My main concern is with load testing because of the TCP connection between the FreeSWITCH and call control application and also TCP connection to database to retrieve subscriber info. The basic functionality seems to work fine but haven't tested it with high call volume, which I am about to do. The last couple of days I have been experimenting with mod_lua but can't find a method under "Session" to get information related to CHANNEL_ANSWER and CHANNEL_HANGUP events. The "session:geVariable" does not have variables related to session answer time or hangup time, something similar to "$e->getHeader("Caller-Channel-Answered-Time")" in ESL. I must be missing something very simple here. Again, my experience with mod_lua is only a couple of days old. I am new Freeswitch and thought I would seek the advice of all the experts and people with more experience with Freeswitch. For this type of application, am I better off with mod_lua, or mod_perl, or shall I stick with ESL using Perl? How about ESL with C/C++? Will I gain some performance gain there? Any feedback would be appreciated. Thanks. Sam

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freeswitch say application currency in multipal language
by bhavik patel 02 Mar '15

02 Mar '15

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

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mod_callcenter - cc_warning_tone not working
by bedgar＠vseinc.com 27 Feb '15

27 Feb '15

Hello Experts, I have been working on setting up a call center in FS and am having a problem with the warning tone notification. It does not play any tones before the caller is bridged to the agent. I am using the latest stable build for 1.2 and I am using the example in the wiki: https://wiki.freeswitch.org/wiki/Mod_callcenter#uuid-standby Snippet: <extension> <condition field="destination_number" expression="^(4099)$"> <action application="set" data="transfer_after_bridge=4099"/>  <action application="set" data="res=${callcenter_config(agent set uuid ${caller_id_number}@${domain_name} '${uuid}')}" /> <action application="set" data="res=${callcenter_config(agent set type ${caller_id_number}@${domain_name} 'uuid-standby')}" /> <action application="set" data="res=${callcenter_config(agent set status ${caller_id_number}@${domain_name} 'Available (On Demand)')}" /> <action application="set" data="res=${callcenter_config(agent set state ${caller_id_number}@${domain_name} 'Waiting')}" /> <action application="set" data="cc_warning_tone=tone_stream://%(200,0,500,600,700)"/> <action application="answer" /> <action application="playback" data="$${hold_music}"/> </condition> </extension> I have tried changing the tone values to see if making it repeat or play longer would produce a positive result. No dice... On a second note, is there a way to play a file to the agent before the call is bridged? My call center services multiple brands and would like to have the agent greet the caller appropriately. Thank you, Brian Edgar

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ESL and Event Socket for .NET
by Tony Bourdeaux 24 Feb '15

24 Feb '15

Could anyone help guide me to the correct Event Socket Libraries for Freeswitch with .NET (C#)? I see several projects that are incomplete and Visual Studio project in the source under /libs/esl/managed. But I cannot get this to compile. Is there any documentation on ESL for .NET that someone could share ? Thank You. -- Tony Bourdeaux

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Follow me implementation
by Brian Foster 25 Jan '15

25 Jan '15

I've been given an assignment. It's a little rough, and honestly I've been working on other projects and at the same time loosing my freeswitch-fu. So, here it goes. Company owner wants to be able to implement a follow me function. He's asking for the deskphones to begin ringing, then have cell phones ring N seconds later WHILE the deskphones continue to ring. The function has to be able to work using a couple different ways of dialing (we've got call groups implemented, http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups) When the mobile phone is answered, we need to be able to get some feedback from the callee to figure out if they're human. We'll use AVMD to kill the call if it detects a voicemail beep. I've looked at several different examples on the wiki and mailing list, and the only way I can figure out how to do it while keeping the requirements in mind is to at some point resort to using Loopback (something i didnt want to do). Requirements are: 1. Use a custom IVR/menu/something to get a confirmation from the callee that they are human (while also keeping it available for customization he's wanting a way to blacklist numbers on that same menu). So that rules out group_confirm_file, etc. 2. Use AVMD to kill the call if we detect the call was picked up by voicemail. 3. The custom IVR/menu/something isn't used on the deskphones 4. Deskphones need to continue to ring after the external number leg is started. I can't timeout the call on the deskphone then call the cell phone, or call the deskphone, time it out, then call the deskphone and cell phone. 5. It has to work on any type of calling method (so basically, if the deskphone rings then eventually the cell phone needs to ring to if it's assigned). Has anyone done something similar, and if so, how did you do it? Thank you, Brian Foster Project Manager/Owner's Representative Davri Investments, Incorporated P: +1-317-787-2686 M: +1-317-600-9753 Indianapolis, Indiana

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Playing with ACL and authenticate users
by Elhodred 15 Jan '15

15 Jan '15

Hi all, I was playing around trying to have some users authenticated by IP and some users to be challenged for username/password auth. The problem is that I don't get it all. If I configure 2 users in this way: <user id="6001" cidr="192.168.28.12/32"> <params> </params> <variables> <variable name="accountcode" value="6001"/> <variable name="user_context" value="GOLDEN"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> </variables> </user> <user id="6002"> <params> <param name="password" value="111111"/> </params> <variables> <variable name="accountcode" value="6002"/> <variable name="user_context" value="GOLDEN"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> </variables> </user> And I added in sofia.conf.xml <param name="apply-inbound-acl" value="domains"/> User 6001 can authenticate if it's using IP 192.168.28.12 and can place calls, but user 6002 is rejected by acl "domains" Then I tried this: <user id="6001"> <params> <param name="auth-acl" value="192.168.28.12/32"/> </params> <variables> <variable name="accountcode" value="6001"/> <variable name="user_context" value="GOLDEN"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> </variables> </user> <user id="6002"> <params> <param name="password" value="111111"/> </params> <variables> <variable name="accountcode" value="6002"/> <variable name="user_context" value="GOLDEN"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> </variables> </user> And I deleted in sofia.conf.xml <param name="apply-inbound-acl" value="domains"/> and added <param name="auth-calls" value="true"/> Now user 6002 can place calls but for user 6001 I see in the log : 2013-05-23 18:15:27.222090 [WARNING] switch_core_state_machine.c:514 bed463c1-742e-4e24-829f-a6188a6667b3 sofia/GOLDEN/XXXXXXXX(a)XXX.XXX.XXX.XXX Abandoned 2013-05-23 18:15:27.222090 [NOTICE] switch_core_state_machine.c:517 Hangup sofia/GOLDEN/XXXXXXXX(a)XXX.XXX.XXX.XXX [CS_NEW] [WRONG_CALL_STATE] This is my dial plan config: <context name="GOLDEN"> <extension name="unloop"> <condition field="$${unroll_loops}" expression="^true$"/> <condition field="$${sip_looped_call}" expression="^true$"> <action application="deflect" data="$${destination_number}"/> </condition> </extension> <extension name="outside_call" continue="true"> <condition> <action application="set" data="outside_call=true"/> </condition> </extension> <extension name="hangup"> <condition field="destination_number" expression="^(hangup)$"> <action application="hangup"/> </condition> </extension>  <extension name="remove_prefix_2" continue="true"> <condition field="network_addr" expression="^XXX\.XXX\.XXX\.XXX$"/> <condition field="destination_number" expression="^12345(\d+)$"> <action application="log" data="Removing leading digits"/> <action application="set" data="destination_number=$1"/> </condition> </extension> <extension name="distributor_TEST2_37360______"> <condition field="${sip_authorized}" expression="^true$" break="never"/> <condition field="destination_number" expression="^(\d+)$"> <action application="log" data="Dialing User"/> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="user/$1(a)XXX.XXX.XXX.XXX"/> </condition> </extension> And now I'm lost. Please, can anybody point me where the issue can be? Do I need to have different sofia profiles, one for IP authentication and one for username/password auth? Regards, Alfonso.

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Transfer hangup but b-leg ring
by FERNANDO VILLARROEL 29 Dec '14

29 Dec '14

Hi. I am trying to use nibblebill and lcr. If originator (example 1006) account have zero balance and trying to call to 1004, the call hangup but the phone 1004 ring anyway. My dialplan: <context name="default"> <extension name="Outbound to PSTN 11 Digits"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="continue_on_fail=true"/> <action application="lcr" data="$1 pg_prefix2"/> <action application="set" data="nibble_account=${accountcode}"/> <action application="set" data="nibble_rate=${lcr_rate_1}"/> <action application="bridge" data="${lcr_auto_route}"/> </condition> </extension> <extension name="hangup"> <condition field="destination_number" expression="^(hangup)$"> <action application="hangup"/> </condition> </extension> The log of call: http://pastebin.freeswitch.org/19228 How i can do?

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newbie question (first build): FreeSWITCH just "hangs" when I run it
by Craig Stevenson 22 Dec '14

22 Dec '14

Hoping someone can help out a newbie figure out what I'm doing wrong. This is my first attempt at building and I keep running into same issue each time. Any pointers would be greatly appreciated. My problem is that FreeSWITHC seems to just get stuck when I try to run the binary. And and if I execute with -ncwait optoin, it keeps spitting out a line saying that it is waiting for background process to be ready This is what happens when I try to run the compiled system on a home ESXi server in my internal home network. # ./freeswitch 2014-06-22 21:23:06.011792 [INFO] switch_event.c:670 Activate Eventing Engine. 2014-06-22 21:23:06.011947 [WARNING] switch_event.c:652 Create additional event dispatch thread 0 <and then just sits there.... > BUILD VERSION: - Debian-7.5.0-amd64-netinst.iso - FreeSWITCH master (git:git clone git://git.*freeswitch.org/freeswitch.git <http://freeswitch.org/freeswitch.git>)* TEST MACHINE: ESXi 5.1 VM - Debian GNU/Linus 6 (64 bit) - 1 CPU - 1 GB RAM - 16GB Memory - 1 NIC = E1000 INSTALLATION STEPS with output piped through 'tee' to capture logs: su (apt-get update 2>&1) | tee ~/apt-get-update.txt (apt-get upgrade 2>&1) | tee ~/apt-get-upgrade.txt (dpkg --get-selections 2>&1) | tee ~/dpkg-get-selections.txt (apt-get install autoconf automake devscripts gawk g++ git-core libjpeg-dev libncurses5-dev libtool make python-dev gawk pkg-config libtiff5-dev libperl-dev libgdbm-dev libdb-dev gettext libssl-dev libcurl4-openssl-dev libpcre3-dev libspeex-dev libspeexdsp-dev libsqlite3-dev libedit-dev libldns.dev libpq-dev 2>&1) | tee ~/apt-get.txt (dpkg --get-selections 2>&1) | tee ~/dpkg-get-selections-after-apt-get.txt cd /usr/src (git clone git://git.*freeswitch.org/freeswitch.git <http://freeswitch.org/freeswitch.git>* 2>&1) | tee ~/git.txt cd /usr/src/freeswitch (./bootstrap.sh 2>&1) | tee ~/bootstrap.txt # same result if I add flag -j (./configure --enable-core-pgsql-support 2>&1) | tee ~/configure.txt (make install 2>&1) | tee ~/make-install.txt (make all cd-sounds-install cd-moh-install 2>&1) | tee ~make-sounds.txt cd /usr/local/freeswitch/bin #./freeswitch

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Freeswitch HA advices
by Federico Castro 18 Dec '14

18 Dec '14

Hi all, I'm working on a Freeswitch HA solution. Now I'm deciding what method and DB I'll use to track calls. I have installed PostgreSQL on both servers and I configured them to replicate DB asynchronously. I would like to know if someone has experience with this kind of solution and what things do I have to contemplate to deploy a solid solution. Thanks in advanced. Federico.

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Re: [Freeswitch-users] Issues with rtp audio going in slow motion
by Joel White 15 Dec '14

15 Dec '14

Is this an issue with transcoding? I am wondering how I lock it down so that transcoding only happens at a minimum Also, would it be bad to completely remove G722 from the codec list? I think the issue is that our provider is sending G711 and all of the Polycom Phones are trying to use G722 On Mon, Oct 20, 2014 at 11:43 AM, Joel White <joelewhite(a)gmail.com> wrote: > I am having an issue where some calls start out fine and then go into slow > motion (distorted slow speech) > > I noticed something in the logs stating > > Activating write resampler > > > What does this mean and how do I avoid this in the future? > > > > Thank you in advance >

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