January 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Update CLI on Freeswitch originated call
by Joshua Gigg 22 May '15

22 May '15

Hi, Is it possible to update the CLI at will once a Freeswitch originated call has been answered? I know it can update during a transfer, but I want to be able to control it directly myself. -- Joshua Gigg

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Receive chat
by Sameer Khan 05 May '15

05 May '15

Hi, Is there any command which can receive chat from sip channel so I can put that received message in to a variable.

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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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Port changes in multiple 183s cause no audio after 200
by Jeff Pyle 17 Mar '15

17 Mar '15

Hello, The following is on FreeSWITCH Version 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26 21:57:33Z 64bit). The design goal for this configuration is that of a simple transcoding SBC. SIP calls arrive with various supported codecs, SIP calls bridge out on PCMU. No users, no auth, etc. Overall, it seems to work but there is one call flow I'm struggling with. The B-leg of calls are bridged to a PSTN gateway. If the gateway's signaling follows 100, 183, 200, all is well. But if the gateway sends multiple 183s with different RTP ports, there is no audio when the call goes to 200. See the following example: - Gateway signals 183 with SDP indicating audio on port 16384. - Gateway signals 183 with SDP indicating audio on port 16386. - Gateway signals 200 with SDP indicating audio on port 16384 (same as original 183). In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio. I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200. I don't know if this is a feature, bug, or misconfigured option. Thoughts are welcome! - Jeff

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any handset recommendations that operate like polycom+cisco?
by Nathan Neulinger 10 Mar '15

10 Mar '15

Looking for any handset/desk phone recommendations that have the "every line key can be it's own SIP account, even on expansion modules" behavior. Yealink units for example only allow a set number of lines, and the rest don't operate as line keys, so even though you can have a 30 + button phone, it's not usable for a receptionist or secretary that may handle calls for 15 different people with their own direct lines. With the polycom and cisco (sccp) phones, you can define N different SIP accounts. Are there any other phones out there that operate this way that anyone can recommend? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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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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