December 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

eliminating fax header line
by Bruce Lefko 26 Mar '15

26 Mar '15

I'd like to completely eliminate the header line sent at the top of each fax page using spandsp. i've set fax-header to a null string, but that doesn't seem to be removing it, and it defaults back to 'Spandsp fax header'. This seems to be the default in the source code in 1.2. What do I have to do to eliminate the header line? Do I have to make changes to freeswitch source? Thanks!

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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 1.4.15 Released
by Ken Rice 03 Mar '15

03 Mar '15

New Post on freeswitch.org from krice387 check it out at http://freeswitch.org/freeswitch-1-4-15-released/ FreeSWITCH 1.4.15 Released FreeSWITCH 1.4.14 has been released! This is routine maintenance release. Source Tarball available at http://files.freeswitch.org/freeswitch-1.4.15.tar.bz2 Debian and Yum Repos have been updated as well. See the release notes below for a list of notable changes. Happy New Years From the FreeSWITCH Team! Release Notes: New features that were added: e55aee1 FS-7025 Add drop_dtmf_masking_tone channel_variable [Jira: https://jira.freeswitch.org/browse/FS-7025] a8c5a0c FS-7048 Add timezone support to mod_say_{de,es,ja,nl,th,zh} 17574a8 Add bert stats to mod_bert::lost_sync event a26e29c vs2010 support for recent unimrcp changes cee8b30 Set rtp_has_crypto for dtls calls 5fcff50 FS-7093 Create uuid_drop_dtmf [Jira: https://jira.freeswitch.org/browse/FS-7093] f024ea3 FS-7047 Arbitrary MRCP headers can now be sent to unimrcp input components in mod_rayo [Jira: https://jira.freeswitch.org/browse/FS-7047] e783999 Some changes to webrtc to make it work with iDoubs in rtcweb profile mode d189e98 Allow 10ms jb 750b1dd FS-7114 Allow streaming binary data from mod_memcache Improvements in performance: 4bcf1d8 Use cached time to save cpu Improvements in cross platform build supports: 32c27b3 Added a Debian dependency to the CentOS6 makefile f4876d5 FS-7031 [unimrcp] update sofia-sip.m4 so that it can build when relative path is used in configure.gnu –with-sofia-sip [Jira: https://jira.freeswitch.org/browse/FS-7031] 061f3cb FS-7031 #resolve #comment [unimrcp] update library again to pull in upstream fix for –with-sofia-sip=../sofia-sip [Jira: https://jira.freeswitch.org/browse/FS-7031] 382e683 Use FTDM_UINT64_FMT macro to log uint64_t values, in order to not break x86 builds. dc9e904 FS-7025 fix compiler warning introduced from e55aee14 [Jira: https://jira.freeswitch.org/browse/FS-7025] b69c93e FS-7030 More work toward fixing FS build on Windows Visual Studio 2012 [Jira: https://jira.freeswitch.org/browse/FS-7030] db66cdb Fix mrcp libraries to build correctly c327455 FS-7030 More work toward getting FS to build on Windows Visual Studio 2012 [Jira: https://jira.freeswitch.org/browse/FS-7030] b341ff7 FS-7046: fix data types and casting on some vars to silence windows build warnings in mod_verto [Jira: https://jira.freeswitch.org/browse/FS-7046] 7ce5171 FS-7046 follow up on type change in mod_verto [Jira: https://jira.freeswitch.org/browse/FS-7046] 357ffad Fix windows build error 0b414a8 vs2010 unimrcp working build 0c1e698 Update build deps for debian list 0a0b926 Build fix for gcc 4.9 fixing a variable set but not used error in mod_commands af6b23a FS-7046 Fix some additional Windows build warnings for mod_verto [Jira: https://jira.freeswitch.org/browse/FS-7046] Additional documentation: f63f868 FS-7049 – Documentation for state optional paramenter in callcenter_config queue list and count [Jira: https://jira.freeswitch.org/browse/FS-7049] In terms of stability these were the use cases that were fixed: 392c687 FS-7055 Fix for a stability race condition in FS [Jira: https://jira.freeswitch.org/browse/FS-7055] d5119a7 FS-7091 Removed unnecessary mutex lock inside input component’s cleanup function since the input component won’t be cleaned up unless all references have been released, in mod_rayo [Jira: https://jira.freeswitch.org/browse/FS-7091] These were the packaging improvements: 3c8dd3e Handle missing `lsb_release` 505cd29 Refactor distro detection and handling 430433a Improve error message d88bae1 Support optional debian parallel builds The following bugs were squashed: 5bbef7f FS-7015 Fix for inbound call on hold issue in mod_sofia [Jira: https://jira.freeswitch.org/browse/FS-7015] 99a5b50 FS-7063 Fix for media delay issue [Jira: https://jira.freeswitch.org/browse/FS-7063] 21458f8 FS-7062 On redirect, when uri are passed in without <> with multiple uris, automatically add the q= header param in decending order in mod_sofia. [Jira: https://jira.freeswitch.org/browse/FS-7062] 5376e82 FS-6688 This will fix the normal case of record route from a proxy without breaking normal changing of a contact in mod_sofia [Jira: https://jira.freeswitch.org/browse/FS-6688] 06c241a FS-6891 FS-7002 FS-7059 FS-7072 FS-7073 FS-7076 #close #comment All of these bugs are invalidated due to a botched revert [Jira: https://jira.freeswitch.org/browse/FS-6891] 922fd81 FS-7015 The code was not properly catching the 0.0.0.0 after changing it to work with ICE SDPs because it was looking in the wrong place for the 0.0.0.0 [Jira: https://jira.freeswitch.org/browse/FS-7015] 3d515cf Re-mark cur_payload as negotiated when detected as such by parser or the rtp could stop working on session re-invite 19272dc FS-7078 Fix sip_header_as_string to properly null_terminate on larger header strings [Jira: https://jira.freeswitch.org/browse/FS-7078] e268a72 FS-6994 Fix for Codec OPUS decoder error in mod_opus [Jira: https://jira.freeswitch.org/browse/FS-6994] 6dbb416 FS-7086 FS-6798 Fix for invalid codec tearing down the call request [Jira: https://jira.freeswitch.org/browse/FS-7086] 46adbec FS-7030 #comment [unimrcp] restore visual studio 2010/2012 project files added by FS project [Jira: https://jira.freeswitch.org/browse/FS-7030] bad5dc3 FS-7037 Fix for T38 fax break started by commit 5bbef7f1e50 [Jira: https://jira.freeswitch.org/browse/FS-7037] 72c3df5 FS-6891 FS-6713 #comment revert 1b612fecb6e8db11da9b15c5522b87e7b642423d [Jira: https://jira.freeswitch.org/browse/FS-6891] 2a7b022 FS-6980 #resolve don’t crash when using native recording on recordstop the redo [Jira: https://jira.freeswitch.org/browse/FS-6980] 35ba6a3 FS-6766 Fix verto caller ringback missing on conference bridge in mod_verto [Jira: https://jira.freeswitch.org/browse/FS-6766] e8cf9c7 FS-7045 Guarantee that dialed call can be joined when answered event is sent in mod_rayo [Jira: https://jira.freeswitch.org/browse/FS-7045] 4be6290 FS-7052 Moving jb queue swap operation out of the debug block. [Jira: https://jira.freeswitch.org/browse/FS-7052] 843e495 FS-7051 Preserve the annexb=no/yes status in mod_sangoma_codec [Jira: https://jira.freeswitch.org/browse/FS-7051] 158c1f2 FS-7002 Fix for recorded audio being choppy when diferent ptimes present and record session starts on bleg [Jira: https://jira.freeswitch.org/browse/FS-7002] 4ce2ce3 FS-7092 Fixed bug with Comrex OPUS [Jira: https://jira.freeswitch.org/browse/FS-7092] d786490 Fix timestamps in mod_bert broken by the cpu improvements refactoring ba016c2 FS-7095 Fix for FS sending DTLS HELLO (and STUN binding request) to wrong port [Jira: https://jira.freeswitch.org/browse/FS-7095] e0dcd17 FS-7083 #comment patch to change mod_shout to use lame_encode_buffer_interleaved on stereo channels so we don’t have to mess with the input data [Jira: https://jira.freeswitch.org/browse/FS-7083] 326289c FS-7083 This patch adds a dedicated thread for writing to the file and the channel_variable RECORD_USE_THREAD=false will disable it and sync may still be maintained at the cost of dropping more data from the audio signal. [Jira: https://jira.freeswitch.org/browse/FS-7083] 9fabbab Disable hard-mute when a session has a media bug attached 0200bc1 FS-7083 Fix divide by zero [Jira: https://jira.freeswitch.org/browse/FS-7083] 067cb0f FS-7100 Make buffer for sub contact big enough in mod_sofia [Jira: https://jira.freeswitch.org/browse/FS-7100] 7798b2f FS-6984 Set default video rates [Jira: https://jira.freeswitch.org/browse/FS-6984] 763e6aa FS-7046 Fix warning introduced from b341ff7 [Jira: https://jira.freeswitch.org/browse/FS-7046] 65e678b FS-7070 Fix mod_expr `clamp` function typo 0a66db6 FS-7111 Fix for bridge_early_media crash [Jira: https://jira.freeswitch.org/browse/FS-7111] Miscellaneous commits: 0d636af FS-7031 [unimrcp] revert configure.gnu change- doesn’t work when using non-source build dir. [Jira: https://jira.freeswitch.org/browse/FS-7031]

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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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Issue with Freeswitch Behind nat
by Frederick Pruneau 15 Jan '15

15 Jan '15

Hi guys, We have an issue with one freeswitch server behind nat. We have a setup like this: -One master Freeswitch server -One freeswitch server connected to the master (Public IP) - Server A -One freeswitch server connected to the master (behind nat) - Server B If server A call server B, nothing happens. There is no sound. After 30 sec, it times out. We have done a tcpdump. From server A to master packets are ok. From Master to server B, we have seen that there is no source and no destination ports for sip invite. If we use our cellphone and we call server B, there is no problem. I have attached the failed call pcap file and freeswitch's log file so you can take a look at them. Master = Freeswitch v1.4.13 Server A = Freeswitch v.1.4.13 Server B = Freeswitch v.1.4.14 (Updated to latest release since we have issues with this server) Thanks in advance. PS: The failed call is from 514-448-0773.

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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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