June 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Pre-Built Virtual Machine - Verto not working
by Oni Kuma 16 Jun '20

16 Jun '20

Hello, I am just starting this journey to learn Freeswitch. I've downloaded the prebuilt virtual machine from https://freeswitch.com/index.php/fs-virtual-machine/ and followed the directions accordingly. When i connect to test Verto, it does not work at all. I've tried with Firefox - I accept the cert, enter the info, it selects the audio mic and camera, and then just says "waiting for reconnect". I get nothing. I've tried with Chrome - I accept the cert, enter the info, and it gets me to the next step where it allows me to select the mic and camera ( i even see the camera light on), but I am unable to select the "save" button. I do not even see my own image from the camera. I've followed the instructions exactly, and have even followed how it is performed in the youtube video. Can someone point me in the right direction?

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calls to some registered webrtc endpoints results in NO_USER_RESPONSE
by Babak Yakhchali 16 Jun '20

16 Jun '20

Hi I'm using webrtc with chrome and sipjs as client on freeswitch 1.10. Calling to some endpoints results in NO_USER_RESPONSE. I've not access to client systems and it makes debugging the problem harder. this is the freeswitch log for one of those calls: https://pastebin.freeswitch.org/view/f9a09df4#L7 Does NO_USER_RESPONSE mean that invite is not reached to the client browser or it is reached but no reply is received? can it be related to network problems or NAT? any help would be appreciated thanks

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Re: [Freeswitch-users] playback to soundcard TML instead ringfile with portaudio
by Tom Lynn 16 Jun '20

16 Jun '20

have you considered recording it to a wave file? On Mon, Jun 1, 2020 at 7:18 AM Tamer Higazi via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Tamer Higazi <th982a(a)googlemail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Mon, 1 Jun 2020 16:17:29 +0200 > Subject: playback to soundcard TML instead ringfile with portaudio > Hi people, > > I am using on my linux system a lot portaudio to place and make phone > calls. > > Inside the portaudio.conf I am using: > <param name="ring-file" value="/opt/freeswitch10/sounds/ring_dd.wav"/> > > that if a phone call comes in it respectively rings on my headset. > > Instead the file, i'd like freeswitch to playback tones via TML: > https://freeswitch.org/confluence/display/FREESWITCH/TGML > > now what I want is to disable the ring-file and make freeswitch in the > dialplan playback to myself (the soundcard) for example this UK ring: > <action application="playback" > data="tone_stream://%(400,200,400,450);%(400,2200,400,450)"/> > > not the callee should get this ring, freeswitch should playback this > sequence before I execute "pa answer". > > > Any ideas ? > > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Mon, 01 Jun 2020 07:18:13 -0700 (PDT) > Subject: [Freeswitch-users] playback to soundcard TML instead ringfile > with portaudio > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://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 > https://freeswitch.com

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2-Stage Dialing using freeswitch
by Vishal Pai 15 Jun '20

15 Jun '20

Hello Everyone Can we use freeswitch for 2-Stage Dialing ? Does anyone tried with the freeswitch? Basically customer dials 0XX980123456 we strip 0XX dial a local number, wait for a dial tone and then send dial 980123456. Thanks Vishal P.

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Re: [Freeswitch-users] ./configure single module?
by David Villasmil 15 Jun '20

15 Jun '20

Thanks Tamer, but I was referring to the confirmation step; not the make and install. :) On Mon, 15 Jun 2020 at 11:00, Tamer Higazi via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Tamer Higazi <th982a(a)googlemail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Mon, 15 Jun 2020 11:21:20 +0200 > Subject: Re: [Freeswitch-users] ./configure single module? > Not true. > > steps to reproduce: > > 1. get the freeswitch source > 2. ./bootstrap.sh > 3. edit modules.conf (the minimal system) > 4. ./configure --prefix=/opt/freeswitch > 5. make && make install > > now you decide yourself to install at a later point of time further > modules, (my freeswitch source is in /usr/local/src/freeswitch), then > you enter the directory: > > /usr/local/src/freeswitch/src/mod/applications/mod_conference > > and just type make && make install > > Just tested it, works. > > > Have fun! > > > best, Tamer > > > On 2020-06-12 19:15, Mike Jerris wrote: > > No. Configure is one step for all. > > > >> On Jun 12, 2020, at 10:50 AM, David Villasmil < > david.villasmil.work(a)gmail.com> wrote: > >> > >> Hello > >> > >> Is it possible to issue something to configure a single module? > something like: > >> > >> ./configure mod_conference > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Mon, 15 Jun 2020 03:00:13 -0700 (PDT) > Subject: Re: [Freeswitch-users] ./configure single module? > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://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 > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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./configure single module?
by David Villasmil 15 Jun '20

15 Jun '20

Hello Is it possible to issue something to configure a single module? something like: ./configure mod_conference ? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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About att_xfer on caller leg
by 王聡 15 Jun '20

15 Jun '20

Hey all, I’m trying call transfer function in freeswitch using att_xfer app, the dialplan looks like: [features.xml] <extension name="att_xfer"> <condition field="destination_number" expression="^att_xfer$"> <action application="read" data="3 6 'tone_stream://%(10000,0,350,440)' digits 300000 #"/> <action application="set" data="origination_cancel_key=#"/> <action application="set" data="attxfer_cancel_key=#"/> <action application="set" data="attxfer_hangup_key=*"/> <action application="set" data="attxfer_conf_key=0"/> <action application="set" data="ringback=%(2000,4000,440.0,480.0)"/> <action application="att_xfer" data="user/${digits}"/> </condition> </extension> [default.xml] <extension name=“inbound_call”> ... <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features”/> … </extension> <extension name=“outbound_call”> ... <action application="bind_meta_app" data="4 a s execute_extension::att_xfer XML features”/> … </extension> The transfer should work on receiver leg in inbound call and caller leg in outbound call when I push *4, and it works correctly in my test. But an annoying problem occurred. When I tried to hangup after transfer(set as *) in outbound call, the transferred user would not hangup correctly and turned into a 3-way call. I had considered if my dialplan has some problems, but the same function works well in inbound call and transferred user could hangup after a *. Any suggestion about this problem would be appreciated. Regards. P.S.: System Information: FreeSWITCH: Version 1.8.7+git~20190702T200609Z~6047ebddfc~64bit (git 6047ebd 2019-07-02 20:06:09Z 64bit) PBX: Asterisk 15.7.4 Client: Bria 5

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Compiling c/c++
by David Villasmil 14 Jun '20

14 Jun '20

Hello all, If anyone can help with my confusion here: Let's say i'm working on an already existing module. I've created modules in the past, this is not really my problem. My problem comes from an already existing module, let's call it my_module. In this module I'm adding a new functionality where i need to decrypt a jwt token parameter encrypted in AES. The existing module main file, the file loaded by the main freeSWITCH loader is in C., let's say something like this: #include <switch.h> #include <switch_json.h> #include <switch_stun.h> #include <jwt.h> #include "token_crypto.h" <-- This is my addition ... <some stuff goes here> at some point i do this: plaintext_len = token_decrypt( *token_encoded, plaintext ); my token_crypto.h is SWITCH_BEGIN_EXTERN_C #include <stdio.h> #include <string.h> #include <openssl/ssl.h> #include <openssl/evp.h> #include <openssl/err.h> #include <openssl/bio.h> #include <openssl/buffer.h> #include <openssl/sha.h> void handleErrors(unsigned char *ciphertext); int gcm_decrypt(unsigned char *ciphertext, int ciphertext_len, unsigned char *aad, int aad_len, unsigned char *tag, unsigned char *key, unsigned char *iv, int iv_len, unsigned char *plaintext); int token_decrypt( const char token_encoded, unsigned char *plaintext ); SWITCH_END_EXTERN_C And then the actual implementation in token_crypto.cpp I added the requirement to compile token_crypto in the Makefile.am like this: mod_mymodule_la_SOURCES = \ base64url.cpp \ token_crypto.cpp \ mod_mymodule.c The code then compiles ok, but when i try to load it i'm getting: **/usr/local/freeswitch/mod/mod_mymodule.so: undefined symbol: token_decrypt** I know the linker can't find the compiled references, but i just can't figure out how to link it all up... call me stupid, but help me out 🤔 Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Re: [Freeswitch-users] Writing Custom Verto Events to Websocket
by David P 13 Jun '20

13 Jun '20

Interesting. Do you know of a way for another service to send a msg (perhaps other than via SIP) into FS that it could proxy out this way? Does FS have a chat listener? On Thu, 11 Jun 2020, 5:23 am , > ---------- Forwarded message ---------- > From: Gregor Nanger <gregor(a)infomedia.si> > > If you would like to inform verto user from fs server, I would suggest > using chat: > > https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+chat > > I am using it to send my system messages from my code logic to verto users > and it works like a charm. >

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Gist repo to package mod_av.so on CentOS 7 announcement
by Richard Chan 13 Jun '20

13 Jun '20

Hello list, Background: my most desired module for CentOS 7 is mod_av.so 1. I am announcing a gist-repo, for doing an out-of-repo package for mod_av.so for CentOS 7, based on the official freeswitch-devel. 2. mod_av.so is not packaged as CentOS 7 does not have ffmpeg. You need to wrangle your own ffmpeg-devel, e.g. from RPMFusion 3. With ffmpeg-devel installed and this gist repo, you can package mod_av.so without rebuilding the gigantic FreeSWITCH src.rpm. Essentially, all the libtool (gcc, ld) statements for mod_av.so have been extracted from a full build and put inside this Makefile. The compiler and linker flags are consistent with those from the FreeSWITCH spec file. 4. IMPORTANT: this gist repo does not distribute source. You must still obtain avcodec.c avformat.c mod_av.c mod_av.h from the tarball or src.rpm. All comments welcome. Cheers SP

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