December 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

GetDigits in another thread
by yunqing 29 Dec '17

29 Dec '17

Hi there, i got a problem receiving digit within another thread, i was following the example of "Lua Script to announce members position" from https://freeswitch.org/confluence/display/FREESWITCH/mod_callcenter, everything works fine until i added GetDigits after the announcement, the voice became sorta smeared, and GetDigits was not working at all. the Lua Script was called from dialplan in unblocked mode like: <action application="set" data="result=${luarun(callcenter-announce-position.lua ${uuid} example@default 10000)}"/> and i want the caller to input dtmf after the prompt while still stay queuing, so what should i do? any ideas are appreciated. ________________________________ mangudai_(a)hotmail.com

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VTECH VSP725 MWI
by Sean Devoy 27 Dec '17

27 Dec '17

Hi All, Does anyone have any pointers about getting the Message Waiting Lamp to work on a VTECH VSP725? This is our first VTECH SIP device and it seems fairly simple except for the MWI. The only thing I have found by searching has been to add: <param name="manage-presence" value="true"/> which had no effect. Thanks, Sean

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Get digits through play_and_detect_speech
by Aqs Younas 26 Dec '17

26 Dec '17

Greeting list, I am unable to find a way to get digits through play_and_detect_speech. I have tried lua callback function but it seems to work only for stream. Searching archive and google, gives no result. Could someone please give me some pointers? Best Regards, <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

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Cannot load shared object file mod_callcenter
by Flavio Goncalves 26 Dec '17

26 Dec '17

Hi, I'm using CentOS 7 and have installed FS using yum. I'm getting the following message when loading the mod_callcenter freeswitch(a)sippulse.com> load module mod_callcenter +OK Reloading XML -ERR [module load file routine returned an error] 2017-12-26 11:10:46.097700 [CRIT] switch_loadable_module.c:1522 Error Loading module /usr/lib64/freeswitch/mod/module mod_callcenter.so **/usr/lib64/freeswitch/mod/module mod_callcenter.so: cannot open shared object file: No such file or directory** ` What shared object file is missing? Thanks Flavio E. Goncalves

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How should I configure xml_curl bindings for phrases?
by Babak Yakhchali 25 Dec '17

25 Dec '17

Hi how can I use xml_curl to fetch phrases? is this right? <binding name="languages"> <param name="gateway-url" value="http://localhost:3300/fs/phrases" bindings="phrases" /> </binding> or I should use languages instead of phrases? and what should be the format of generated xml from server? thanks

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Major change to macOS Xcode/CLT requirements
by Mario 23 Dec '17

23 Dec '17

FYI for macOS folks running FreeSWITCH: Homebrew made major changes in the last few weeks that affect the installation instructions for FreeSwitch on macOS. It also affects Homebrew update/upgrade functions. I updated 3 pages of the macOS wiki and the MacOS FreeSwitch Installer, macFI. A brief summary: Homebrew always required the full Xcode instead of the standalone Apple Command Line Tools. That is now the opposite… When Apple updated Xcode it didn’t always provide the correct version of CLT for the previous macOS. For instance, 10.13 and 10.12 use Xcode 9.2, but 9.2 may install the CLT for 10.13 on 10.12 meaning the CLT was for building 10.13 apps. Homebrew would detect this as a showstopper. This has happened a couple of times, and that’s why we sometimes had to use an older Xcode for a macOS that was not the latest to build FreeSwitch. This didn’t just affect FreeSwitch but some Homebrew formulas. So…. The Homebrew folks decided to fix this once and for all: Now, when Homebrew is installed, it looks for the correct CLT. If the latest Xcode (9.2) is already installed on the latest macOS (10.13), nothing needs to happen because the CLT that is part of Xcode is already installed and correct. If Homebrew does not detect the CLT, it will locate them online, download and install the correct version for the macOS running! The big benefit is that if you don’t need Xcode you will save about 4GB of space, and of course it takes a lot less time installing FreeSwitch from scratch. Again, the wiki and macFI are updated, I tested them on 10.13, 10.12, 10.11, and 10.10. Mario G

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XML-CURL and dynamic IVR menus
by Dmitry Mordovin 23 Dec '17

23 Dec '17

Hello! I use xml-curl module - bindings="directory|dialplan" I want use IVR in dialplan, so, return this XML: <document type="freeswitch/xml"> <section name="configuration"> <configuration name="ivr.conf" description="IVR menus"> <menus> <menu name="ivr-test" invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" exit-sound="voicemail/vm-goodbye.wav" timeout ="3000" inter-digit-timeout="2000" max-failures="2" digit-len="3" phrase-lang="en"> <entry action="menu-exec-app" digits="0" param="transfer 11 XML voicemail" /> </menu> </menus> </configuration> </section> <section name="dialplan" description="IVR call"> <context name="public"> <extension name="ivr_1"> <condition field="destination_number" expression="^([1-9]\\d+)$"> <action application="ivr" data="ivr-test"/> <action application="bridge" data="sofia/internal/1000%10.10.10.10"/> </condition> </extension> </context> </section> </document> As I see in log, FS ignore IVR configuration part and ERROR happen [ERR] mod_dptools.c:2055 Unable to find menu Does it possible attach IVR menu with dialplan response? Or I need use another way to load dynamic IVR?

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[OT] baresip
by Alfred E. Heggestad 23 Dec '17

23 Dec '17

Hi, My apologies for this off-topic post, but I hope that this can be of interest to the nice Freeswitch community. We have released a new version of Baresip which is a open source SIP client that is compatible with Freeswitch. you can find the source code here: https://github.com/alfredh/baresip/releases/tag/v0.5.5 We are doing regular interop-testing with Freeswitch, mainly SIP and RTP related features. From the Baresip project we send the best greetings to the nice people on this list :) /alfred

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Play MP4 file into conference
by Richard Screene 23 Dec '17

23 Dec '17

I am having problems attempting to play a MP4 video into a conference using mod_av. I have built the 1.6.19 branch from source after removing the -pedantic compile flag to allow mod_av to compile. My simple dial plan is: ``` <extension name="test"> <condition field="destination_number" expression="test"> <action application="conference" data="conf1"/> </condition> </extension> ``` From fs_cli I then attempt to play the video using the command: "conference conf1 play /tmp/sample.mp4" No error messages are visible in the console and I can hear the audio, but I do not see the video stream. Strangely, when I do "uuid_broadcast <uuid> /tmp/sample.mp4" I get the audio and video (but obviously only to a single caller) If I try on the code from the master branch then I do not video from either conference..play or uuid_broadcast. Has anyone got any ideas why I cannot play video into a conference? Also, is there any equivalent of uuid_fileman for files played into a conference? I would like pause and seek functionality to affect the video of all participants in the conference. Many thanks, Richard

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mod_verto with standart firewall ports
by Pascal Hari 23 Dec '17

23 Dec '17

Hello I'm new to FreeSWITCH and made a setup where everything is behind the same NAT. I have a linux debian 8 server where FreeSWITCH is running on and I'm able to connect different clients (SIP and WebRTC). I'm using the demonstration configuration which come with the FreeSWITCH installation. I mostly followed the instruction "Quick Start FreeSWITCH Demo With Verto Communicator<https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian…>" but used apache2 instead of nginx. Now I want that same setup but on a server in the web with public IP. I'm running a linux debian 8 in azure and installed everything accordingly to the "behind the Nat" setup. I use real (trusted) certificates and can reach the WebRTC client over https. If I'm in the public network of our company I can log in, dial a number and connect to FreeSWITCH (according to the dialplan). If I'm in the corp (internal) network of our company I cannot log in. I only get the message "Waiting for server reconnection". Even in FreeSWITCH (fs_cli) I don't see a request. For me this sound like a firewall issue and many post say to open ports (16384-32768, 5066, ...). I don't have this option because I can't change our companies firewall. And I'm anyway looking for a solution to provide a WebRTC client which can be used in a normal company network with relatively strict firewall rules. Workarounds like VPN-tunnel are also not a wanted solution because I want a really easy setup on the client (WebRTC) side. Am I missing a step? Is my guess with the firewall right or could it be something else? Is there a way to configure FreeSWITCH and verto communicator to use other ports? Thanks already for suggestions. Best regards, Pascal Hari SW Developer CSA Engineering AG ______________________________________________________________________________________________ Confidentiality Note: This message is intended only for the use of the named recipient(s) and may contain confidential and/or privileged information. If you are not the/an intended recipient, please contact the sender and delete this message. Any unauthorized use of the information contained in this message is prohibited.

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