November 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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Re: [Freeswitch-users] Mod LCR SQL sub-query for better routing
by DP . 14 Jan '15

14 Jan '15

Victor, I take back my initial response on this old email. We ran into a couple of cases with some carriers and multiple matching prefixes. After actually trying your sub query (modified for Mysql), it actually does return the true lowest rate from a carrier while importantly respecting the longest match per carrier. Unlike the reorder_by_rate function that does not respect the longest match per carrier. That function simply returns the lowest rate, period. So thanks! From: hi-tecc(a)hotmail.com To: freeswitch-users(a)lists.freeswitch.org Subject: RE: [Freeswitch-users] Mod LCR SQL sub-query for better routing Date: Mon, 16 Dec 2013 18:36:06 -0500 I agree. This definitely sounds like he simply needed the "reorder_by_rate" param. It will reorder the initial sql results strictly by rate: reorder_by_rate - Forces the LCR module to re-order the query strictly on rate basis. By default this is turned off, but enabling this will always prefer rate over anything else. Beware this may have an adverse effect! I initially had this turned on then quickly realized it would sometimes try to route ALL calls by the lowest rate found. Ex: flowroute lists all calls for the US with a default NPA of "1" at .0098. Now a user trying to call Jamaica with "1876" at a rate of 0.19 (or whatever) will get both flowroute rates returned. The reorder by rate will assume 0.0098 is a valid rate since it will now be the "cheapest" in the list and send the call along its way to flowroute, whom will now bill you at 0.19. Now if you have another carrier in your list with 1876 at 0.15 you can see why this would be a problem. In this case you will always want the longest matched NPANXX rate. Date: Fri, 13 Dec 2013 15:58:54 -0500 From: intralanman(a)freeswitch.org To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Mod LCR SQL sub-query for better routing On 12/12/2013 02:42 PM, Victor Chukalovskiy wrote: Hello, For those interested, I added a piece to mod lcr wiki. It makes sorting / routing logic better than default logic: http://wiki.freeswitch.org/wiki/Mod_lcr#Custom_SQL_with_sub-query_-_for_rea… Why it helps: Rates rates can often be given both on per-NPA or per-NPANXX level depending on the carrier and on the NPA. Moreover, some carriers may have NPANXX rate lower than the corresponding NPA rate, while others will have it inverse. Neither simple ORDER BY rate, prefix; nor ORDER BY prefix, rate; give the truly cheapest route. The LCR logic should be two-step process to accommodate this. Cheers, -Victor Unless I misunderstand what you're saying, this is what the reorder_by_rate param does. You'll always want to pick the longest digit match per carrier. Then you probably want to grab the cheapest overall rate of the matches you got back. -Ray _________________________________________________________________________ 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

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Fwd: OverTime RAM usage keep inceasing
by indra sena 08 Jan '15

08 Jan '15

Hi Team, I have observed that overtime RAM utilization/usage is keep increasing maybe due to memory leaks ? when I put load initially it starts with 5GB/12GM RAM, I have kept for 15 hours load test and oberserved now it is using almost 11GB/12GB starts swap also increasing now. Do you have config changes to fix or to take as work around ? How can we reduce RAM utilization overtime ? Could you please provide some solution for this ? Thanks & Regards, Indra.

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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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