May 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

outbound calls from one freeswitch to another
by Joli Martinez 01 Jun '20

01 Jun '20

Hello, I have two freeswitch boxes. They are in two different locations, but both are connected through a VPN. I am looking to send all calls from one freeswitch to another. I was able to get inbound routing working with each other. The problem is when freeswitch1 wants to make an outbound call to the PSTN out of freeswitch2. What do I have tried to add the IP 10.41.100.2 to the ACL and that did not work. What am I missing? 2020-05-27 10:50:37.774914 [WARNING] sofia_reg.c:2906 Can't find user [ FreeSWITCH(a)10.21.10.4] from 10.41.100.2 You must define a domain called '10.21.10.4' in your directory and add a user with the id="FreeSWITCH" attribute and you must configure your device to use the proper domain in it's authentication credentials. send 678 bytes to udp/[10.41.100.2]:5080 at 10:50:37.773507: ------------------------------------------------------------------------ SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.41.100.2:5080;rport=5080;branch=z9hG4bKjr90SBDZmUQKj From: "CallerID" <sip:1112223333@10.41.100.2>;tag=90SHD9m8r6X2N To: <sip:4445556666@10.21.10.4>;tag=rmB10y0mDg5yD Call-ID: 44b8eb5a-1acc-1239-f884-005056873da3 CSeq: 20741727 INVITE User-Agent: FreeSWITCH Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ 2020-05-27 10:50:37.774914 [NOTICE] sofia.c:2332 Hangup sofia/internal/ 1112223333(a)10.41.100.2 [CS_NEW] [CALL_REJECTED]

2 1

Re: [Freeswitch-users] cant locate packages to compile on WSL2
by David P 01 Jun '20

01 Jun '20

Bipin Patel <bipin(a)xbipin.com> wrote: > i was just testing WSL2 and trying to compile FS on Debian buster and for some odd reason its not able to find the below 2 packages: libcurlpp0 libpjmedia-codec2 DW: It seems like you're saying that WSL2 is Debian, but I'm pretty sure it's Ubuntu.

2 1

Anyone managed to build FS on Centos-8? libvpx fails.
by Yehavi Bourvine 30 May '20

30 May '20

Hello, I've built Freewitch 1.8.7 and 1.10.3 successfully from sources on Centos-7. however, it fails on centos-8 while building VPX. Before I dig into it, anyone noticed this also? An example for one of the errors: vpx_scale/generic/vpx_scale.c:286:26: error: ‘vp8_horizontal_line_2_1_scale’ undeclared (first use in this function); did you mean ‘horiz_line_scale’? horiz_line_scale = vp8_horizontal_line_2_1_scale; ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~ horiz_line_scale I can find the declarations inside the WIN tree, but I am not using Windows :-) Thanks, __Yehavi:

2 2

cant locate packages to compile on WSL2
by Bipin Patel 30 May '20

30 May '20

hi, i was just testing WSL2 and trying to compile FS on Debian buster and for some odd reason its not able to find the below 2 packages: libcurlpp0 libpjmedia-codec2 the same install just fine on raspbian buster so can anyone provide some hints if the package names have changed or removed or something else as im not too good with linux i use the below packages to compile on raspbian and everything works just fine: sudo apt -y install autoconf automake devscripts gawk libjpeg-dev libncurses5-dev libtool-bin python-dev 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 libsndfile-dev libopus-dev liblua5.1-0-dev g++ git-core libtool make pkg-config equivs mlocate git dpkg-dev sox flac dh-systemd yasm libogg-dev unixodbc-dev python-all-dev python-support erlang-dev doxygen uuid-dev libcurlpp0 libavformat-dev libswscale-dev libopencv-dev libhiredis-dev ladspa-sdk libmemcached-dev libsoundtouch-dev libflite1 flite1-dev libpjmedia-codec2 libasound2-dev portaudio19-dev libx11-dev librabbitmq-dev libsnmp-dev libmagickcore-dev libvorbis-dev libmp3lame-dev libmpg123-dev libshout3-dev libsndfile1-dev libflac-dev libvlc-dev default-jdk liblua5.2-dev libmono-2.0-dev mono-mcs libyaml-dev bison libbison-dev libjpeg62-turbo-dev -- Regards, Bipin Talky Communications -------------------------

1 0

Lumenvox and mod_unimrcp
by David Villasmil 29 May '20

29 May '20

Hello everyone, Has anyone integrated lumenvox successfully? I got it to “talk” to fs but I don’t seem to be able to figure out how to use it with the builtin cpa, I’ve tried like Detect_speech unimrcp builtin:cpa And variations thereof, but nothing seems to work. Many thanks David -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 3

Sched_hangup
by David Villasmil 28 May '20

28 May '20

Hello, Are sched_habgup destroyed on channel hangup before the scheduled time? Ie: I set a sched_hangup +1000 and the can ends at 10, is the schedule destroyed? Thanks -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Speech Recognition recording to URL
by mayamatakeshi 27 May '20

27 May '20

Hi, out of curiosity, I'm trying to see if I can use application record + mod_httapi (https://freeswitch.org/confluence/display/FREESWITCH/mod_httapi) to connect with Google Speech Recognition service (via a proxy app that will handle HTTP to gRCP convertion). Just as a test, I'm trying to see if using an inbound event socket I can inject a DETECTED_SPEECH event in a channel. I tried something like this and got an +OK from FS: T 192.168.88.78:60380 -> 192.168.88.74:8021 [AP] sendevent DETECTED_SPEECH Unique-ID: a03486b8-781d-430a-ab26-5815485401a8 Call-UUID: a03486b8-781d-430a-ab26-5815485401a8 call_uuid: a03486b8-781d-430a-ab26-5815485401a8 uuid: a03486b8-781d-430a-ab26-5815485401a8 Channel-UUID: a03486b8-781d-430a-ab26-5815485401a8 Speech-Type: begin-speaking T 192.168.88.74:8021 -> 192.168.88.78:60380 [AP] Content-Type: command/reply Reply-Text: +OK 64bdd4aa-9476-4db8-b56c-36a62f6a2238 but nothing got notified to the channel (and no logs about it from FS running in debug level) Would this be actually possible?

5 6

webrtc client using sip.js and conference api interaction
by Jonathan Hunter 27 May '20

27 May '20

Hi Guys, Does anyone know best practice here to be able to interact with the freeswitch conference API so I can mute users and list participants from a web browser, ideally without using dtmf? I have previously used Verto, but due to scale I am using kamailio/rtpengine as a websockets gateway which then load balances requests to freeswitch so its only dealing with SIP and RTP. All works fine, I just want to now be able to kick users/list participants and so on via the browser so wondered the best way to do this now I am now using verto, as I used to subscribe to live array for that but now I am using sip.js to fire SIP requests over websockets, how do I send conference api commands and to the correct box as well if a farm of freeswitch servers are in use. Many thanks Jon

2 3

inband DTMF detection and drop
by SamyGo 27 May '20

27 May '20

Hi all, I've followed all the previous suggestions to detect the inband DTMF coming from a provider and to drop them from going further into B leg. However the inband DTMF still goes to otherside and user can hear clicking sounds. Here's what I've done so far: *VERSION:* FreeSWITCH Version 1.10.2-release+git~20191231T140119Z~f7bdd3845a~64bit (git f7bdd38 2019-12-31 14:01:19Z 64bit) *OS:* Ubuntu 18.04.4 LTS *DIALPLAN:* <extension name="dtmf_filter"> <condition field="${acl(${network_addr} lan)}" expression="true"/> <condition field="destination_number" expression="(.*)"> <action application="set" data="rtp_manual_rtp_bugs=FLUSH_JB_ON_DTMF|ignore_dtmf_duration"/> <action application="set" data="jitterbuffer_msec=1p"/> <action application="set" data="spandsp_dtmf_rx_filter_dialtone=true"/> <action application="set" data="dtmf_verbose=true"/> <action application="set" data="drop_dtmf=true"/> <action application="spandsp_start_dtmf"/> <action application="bridge" data="{bridge_filter_dtmf=true,drop_dtmf=true}sofia/gateway/CISCO_GW/$1"/> </condition> </extension> I've also tried suggestion by other users on mailing list related to flush JB on DTMF rtp_bug but that doesnt help either. *CONSOLE OUTPUT:*2020-05-12 23:59:10.195999 [DEBUG] mod_spandsp_fax.c:295 FLOW DTMF Potentially '5' - total -15.55dB, row -32.25dB, col -40.30dB, duration 338436 - miss 2020-05-12 23:59:10.195999 [DEBUG] mod_spandsp_fax.c:295 FLOW DTMF Potentially '5' - total -17.99dB, row -30.58dB, col -40.99dB, duration 338538 - miss It does detect the number but its not a hit. Can I get some more ideas how to drop inband DTMF tones, Thanks. Best Regards, Sammy

3 2

Audio distortion on calls / streaming wav
by Sam van Herwaarden 26 May '20

26 May '20

Hi, I'm getting some distortion on calls with FreeSWITCH. This is when using the playback command to stream wav. I'm not really sure how to diagnose the issue - I have noticed it both when running FreeSWITCH on my laptop and connecting over SIP from my smartphone on my home network, and have reports that it happens when running on a server on a PRI line using a Sangoma A104 card. The distortions are a sort of crackling or noise, or audio breaking up a bit (they are not present in the source files). How can I approach debugging this? What are typical causes? Related question (as this might have to do with hardware but I'm not sure) - are there any guidelines to be found on how to choose hardware for a system streaming wav files using FreeSWITCH on many (up to 1500) channels at the same time? It seems like this might be a pretty heavy workload and I'm wondering how to do this in a way that ensures availability and quality. Kind regards, Sam

1 0