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Freeswitch as a SIP client
by Scholz, Dieter 29 Sep '16

29 Sep '16

Hello, this is my problem: I would like to use Freeswitch as a gateway between our Hipath PBX and an external SIP provider. For that I created a Hipath SIP account that is working (tested with X-lite). Now I would like to let Freeswitch act as a client using this SIP account. Whenever there is an incoming call Freeswitch should act on it and forward this call to an external SIP provider. Is this possible? Does that make sense? Are there alternatives? At the moment I'm stuck to configure Freeswitch as a client. I added the Hipath SIP account as a gateway. When I compare the SIP traffic of X-lite and Freeswitch using Wireshark I found out that Freeswitch dows not offer the 'Subscribe' option. Is this the reason why it fails? Can you help me? Thanks in advance. Dieter

4 19

installation of specific modules using apt-get commands
by devang nathwani 29 Sep '16

29 Sep '16

Hello, I have installed freeswitch using apt-get on my debian 8 sersver from https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie block under title 'latest release branch:' Now what if i want to install some module which are not in default vanilla setup. Specifically, I need to install mod_sangoma_codec, mod_lua, mod_json_cdr. Where can i find the apt-get command for these modules or is there any other way to install them in my freeswitch?

4 4

unable to set up a call using speex
by Dave Horton 28 Sep '16

28 Sep '16

I am trying to connect an incoming call to freeswitch using speech wideband, but freeswitch is returning a 488 Not Acceptable here. It seems like freeswitch, which is using dynamic payload 99 for speex is unable to match that up to the fact that the remote side (client) is using payload 84. If so, this would seem to be a bug. Relevant log snippets / details: 1. Freeswitch version FreeSWITCH Version 1.6.9+git~20160613T181044Z~d574870720~64bit (git d574870 2016-06-13 18:10:44Z 64bit) 2. speex seems to be enabled... > show codecs type,name,ikey … codec,Speex,CORE_SPEEX_MODULE … 3. Log snippets from call 2016-09-28 18:09:56.592985 [DEBUG] sofia.c:6943 Remote SDP: v=0 o=- 1475100596518739 1 IN IP4 10.0.1.7 s=Bria 4 release 4.4.0 stamp 79957 c=IN IP4 10.0.1.7 t=0 0 m=audio 50218 RTP/AVP 84 101 a=rtpmap:84 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 EXECUTE sofia/internal/1000(a)srf-qa-02.drachtio.org answer() 2016-09-28 18:09:56.592985 [DEBUG] switch_core_media.c:4365 Audio Codec Compare [speex:84:16000:20:0:1]/[opus:116:48000:20:0:1] 2016-09-28 18:09:56.592985 [DEBUG] switch_core_media.c:4365 Audio Codec Compare [speex:84:16000:20:0:1]/[G722:9:8000:20:64000:1] 2016-09-28 18:09:56.592985 [DEBUG] switch_core_media.c:4365 Audio Codec Compare [speex:84:16000:20:0:1]/[SPEEX:99:16000:20:42200:1] 2016-09-28 18:09:56.592985 [DEBUG] switch_core_media.c:4365 Audio Codec Compare [speex:84:16000:20:0:1]/[PCMU:0:8000:20:64000:1] 2016-09-28 18:09:56.592985 [DEBUG] switch_core_media.c:4365 Audio Codec Compare [speex:84:16000:20:0:1]/[PCMA:8:8000:20:64000:1] 2016-09-28 18:09:56.592985 [DEBUG] switch_core_media.c:4281 Set telephone-event payload to 101@8000 2016-09-28 18:09:56.592985 [DEBUG] switch_core_media.c:4693 sofia/internal/1000(a)srf-qa-02.drachtio.org Set 2833 dtmf send payload to 101 recv payload to 101 2016-09-28 18:09:56.592985 [NOTICE] switch_channel.c:3809 Hangup sofia/internal/1000(a)srf-qa-02.drachtio.org [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]' send 949 bytes to tcp/[198.0.151.126]:39915 at 18:09:56.612541: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here

2 1

How to delete entry from sip_regisration table?
by devang nathwani 28 Sep '16

28 Sep '16

We connect freeswitch using odbc connection for device registration on server. Now the problem occur when Freeswitch crashed,down or restart uncertainly that time the entry does not removed from sip_registration table and next time user are not able to make the call. Then we have to manually delete that entry from database to make it working. OS : Debian8 FS version : FreeSWITCH Version 1.6.6~64bit ( 64bit) can anybody help us to resolve this issue?

5 4

High Availability questions
by Anonim Stefan 28 Sep '16

28 Sep '16

Hi, [1] tells how one can achieve HA with Freeswitch. However, the setup is an active-passive one and for each active needs one passive. I have 2 questions: a) Currently, is there a way to achieve an active-active setup? b) Currently, is there a way to achieve a setup like this one: a group of i.e. 5 active Freeswitches and a group of i.e. 1 passive Freeswitch. When one of the 5 breaks, the passive 1 becomes active and restores the sessions for the broken one. Regards, Stefan [1] https://freeswitch.org/confluence/display/FREESWITCH/High+Availability

7 10

'no audio' while call established, can hear the ring back
by devang nathwani 28 Sep '16

28 Sep '16

Hello, Below is the call flow i am trying to achieve and getting 'no audio' while call established, i can hear the ring back tone though client registration through public ip [SERVER A], nat rule to route the request to local ip [SERVER A's CONTAINER A1] --> getting the request to local ip through Profile1; this is a freeswitch server using mod_distributor to balance the load between two freeswitch servers [SERVER A's CONTAINER A1]--> getting the request to local ip through Profile2; this is a freeswitch server1 [SERVER B's CONTAINER B1] --> sending back the response to [SERVER A's CONTAINER A1] using Profile3, nat rule is there to route the traffic to public ip [SERVER A]/to the caller sip signaling works as it should having issues with media signaling though, 'no audio' on call while call get established, let me share some params value of respective profiles regarding media on Profile1, inbound-bypass-media = false inbound-proxy-media = true on Profile2, inbound-bypass-media = true inbound-proxy-media = true on Profile3, inbound-bypass-media = true inbound-proxy-media = false Below is the file containing sip logs from [SERVER A's CONTAINER A1] http://pastebin.com/zpzT81dL Below is the file containing sip logs from [SERVER B's CONTAINER B1] http://pastebin.com/aMAQB45i caller's ip -> 182.72.122.102 public ip to register -> 51.21.91.151 [SERVER A's CONTAINER A1 eth0] -> 10.93.53.126 [SERVER A's CONTAINER A1 eth1] -> 172.16.1.2 [SERVER B's CONTAINER B1 eth1] -> 172.16.3.2 carrier/provider/gateway/termination ip -> 216.56.6.6 (Above are not the real ips, they are changed;)) What am i missig here?

1 0

Playback within condition
by Caio Assis 27 Sep '16

27 Sep '16

Hi. Here's what I'm trying to do. I want to play a sound if a certain condition is true. <extension name="test"> <condition field="destination_number" expression="^1212$"> <condition field="${cond(${destination_number} == 1212 ? 1212 : 0000)}" expression="^1212$">  <action application="log" data="INFO LOG"/> <action application="playback" data="sound1.gsm"/> </condition> <action application="playback" data="sound2.gsm"/> <action application="hangup"/> </condition> </extension> The sound 1 doesn't play, but I knew the condition was true because I can see the INFO LOG on the console. The sound 2, outside the condition, was played normally. Both PLAYBACK and READ applications don't work on nested conditions. Can anyone help me?

4 6

unmet dependency on FS 1.6 on Ubuntu 16.04
by Victor Medina 27 Sep '16

27 Sep '16

Hello guys! I decided to give FS a try on Ubuntu LTS 16.04. It's a clean all vanilla install. .- Added repository .- Updated, and ran aptitude install -y freeswitch-all root@ubuntu:/home/administrador# aptitude install -y freeswitch-all The following NEW packages will be installed: adwaita-icon-theme{a} at-spi2-core{a} ca-certificates-java{a} colord{a} colord-data{a} dconf-gsettings-backend{a} dconf-service{a} default-jre-headless{a} emacsen-common{a} erlang{a} erlang-asn1{a} erlang-base{a} erlang-common-test{a} erlang-corba{a} erlang-crypto{a} erlang-debugger{a} erlang-dev{a} erlang-dialyzer{a} erlang-diameter{a} erlang-edoc{a} erlang-eldap{a} erlang-erl-docgen{a} erlang-et{a} erlang-eunit{a} erlang-examples{a} erlang-gs{a} erlang-ic{a} erlang-ic-java{a} erlang-inets{a} erlang-jinterface{a} erlang-megaco{a} erlang-mnesia{a} erlang-mode{a} erlang-observer{a} erlang-odbc{a} erlang-os-mon{a} erlang-parsetools{a} erlang-percept{a} erlang-public-key{a} erlang-reltool{a} erlang-runtime-tools{a} erlang-snmp{a} erlang-src{a} erlang-ssh{a} erlang-ssl{a} erlang-syntax-tools{a} erlang-test-server{a} erlang-tools{a} erlang-typer{a} erlang-webtool{a} erlang-wx{a} erlang-xmerl{a} flac{a} fontconfig{a} fontconfig-config{a} fonts-dejavu-core{a} freeswitch{a} freeswitch-all freeswitch-conf-curl{a} freeswitch-conf-insideout{a} freeswitch-conf-sbc{a} freeswitch-conf-softphone{a} freeswitch-conf-vanilla{a} freeswitch-lang{a} freeswitch-lang-de{a} freeswitch-lang-en{a} freeswitch-lang-es{a} freeswitch-lang-fr{a} freeswitch-lang-he{a} freeswitch-lang-pt{a} freeswitch-lang-ru{a} freeswitch-meta-all{ab} freeswitch-meta-codecs{a} freeswitch-meta-conf{a} freeswitch-meta-lang{a} freeswitch-meta-mod-say{a} freeswitch-mod-abstraction{a} freeswitch-mod-amr{a} freeswitch-mod-amrwb{a} freeswitch-mod-av{a} freeswitch-mod-avmd{a} freeswitch-mod-b64{a} freeswitch-mod-blacklist{a} freeswitch-mod-bv{a} freeswitch-mod-callcenter{a} freeswitch-mod-cdr-csv{a} freeswitch-mod-cdr-mongodb{a} freeswitch-mod-cdr-sqlite{a} freeswitch-mod-cidlookup{a} freeswitch-mod-codec2{a} freeswitch-mod-commands{a} freeswitch-mod-conference{a} freeswitch-mod-console{a} freeswitch-mod-curl{a} freeswitch-mod-dahdi-codec{a} freeswitch-mod-db{a} freeswitch-mod-dialplan-asterisk{a} freeswitch-mod-dialplan-directory{a} freeswitch-mod-dialplan-xml{a} freeswitch-mod-dingaling{a} freeswitch-mod-directory{a} freeswitch-mod-distributor{a} freeswitch-mod-dptools{a} freeswitch-mod-easyroute{a} freeswitch-mod-enum{a} freeswitch-mod-erlang-event{a} freeswitch-mod-esf{a} freeswitch-mod-esl{a} freeswitch-mod-event-multicast{a} freeswitch-mod-event-socket{a} freeswitch-mod-expr{a} freeswitch-mod-fifo{a} freeswitch-mod-fsk{a} freeswitch-mod-fsv{a} freeswitch-mod-g723-1{a} freeswitch-mod-g729{a} freeswitch-mod-h26x{a} freeswitch-mod-hash{a} freeswitch-mod-httapi{a} freeswitch-mod-http-cache{a} freeswitch-mod-isac{a} freeswitch-mod-java{a} freeswitch-mod-json-cdr{a} freeswitch-mod-kazoo{a} freeswitch-mod-lcr{a} freeswitch-mod-ldap{a} freeswitch-mod-local-stream{a} freeswitch-mod-logfile{a} freeswitch-mod-loopback{a} freeswitch-mod-lua{a} freeswitch-mod-memcache{a} freeswitch-mod-mp4v{a} freeswitch-mod-native-file{a} freeswitch-mod-nibblebill{a} freeswitch-mod-opus{a} freeswitch-mod-oreka{a} freeswitch-mod-perl{a} freeswitch-mod-png{a} freeswitch-mod-pocketsphinx{a} freeswitch-mod-portaudio{a} freeswitch-mod-portaudio-stream{a} freeswitch-mod-posix-timer{a} freeswitch-mod-python{a} freeswitch-mod-redis{a} freeswitch-mod-rss{a} freeswitch-mod-rtc{a} freeswitch-mod-rtmp{a} freeswitch-mod-say-de{a} freeswitch-mod-say-en{a} freeswitch-mod-say-es{a} freeswitch-mod-say-fa{a} freeswitch-mod-say-fr{a} freeswitch-mod-say-he{a} freeswitch-mod-say-hr{a} freeswitch-mod-say-hu{a} freeswitch-mod-say-it{a} freeswitch-mod-say-ja{a} freeswitch-mod-say-nl{a} freeswitch-mod-say-pl{a} freeswitch-mod-say-pt{a} freeswitch-mod-say-ru{a} freeswitch-mod-say-th{a} freeswitch-mod-say-zh{a} freeswitch-mod-shell-stream{a} freeswitch-mod-silk{a} freeswitch-mod-skinny{a} freeswitch-mod-skypopen{a} freeswitch-mod-sms{a} freeswitch-mod-snapshot{a} freeswitch-mod-sndfile{a} freeswitch-mod-snmp{a} freeswitch-mod-snom{a} freeswitch-mod-sofia{a} freeswitch-mod-sonar{a} freeswitch-mod-soundtouch{a} freeswitch-mod-spandsp{a} freeswitch-mod-spy{a} freeswitch-mod-stress{a} freeswitch-mod-syslog{a} freeswitch-mod-theora{a} freeswitch-mod-timerfd{a} freeswitch-mod-tone-stream{a} freeswitch-mod-translate{a} freeswitch-mod-tts-commandline{a} freeswitch-mod-unimrcp{a} freeswitch-mod-valet-parking{a} freeswitch-mod-verto{a} freeswitch-mod-voicemail{a} freeswitch-mod-voicemail-ivr{a} freeswitch-mod-xml-cdr{a} freeswitch-mod-xml-curl{a} freeswitch-mod-xml-rpc{a} freeswitch-mod-xml-scgi{a} freeswitch-mod-yaml{a} freeswitch-music-default{a} freeswitch-sounds-en-us-callie{a} freeswitch-sysvinit{a} freeswitch-timezones{a} glib-networking{a} glib-networking-common{a} glib-networking-services{a} gsettings-desktop-schemas{a} hicolor-icon-theme{a} humanity-icon-theme{a} i965-va-driver{a} java-common{a} javascript-common{a} libaacs0{a} libasound2{a} libasound2-data{a} libatk-bridge2.0-0{a} libatk1.0-0{a} libatk1.0-data{a} libatspi2.0-0{a} libavahi-client3{a} libavahi-common-data{a} libavahi-common3{a} libavcodec-ffmpeg56{a} libavformat-ffmpeg56{a} libavresample-ffmpeg2{a} libavutil-ffmpeg54{a} libbdplus0{a} libbluray1{a} libboost-filesystem1.58.0{a} libboost-system1.58.0{a} libbroadvoice1{a} libcairo-gobject2{a} libcairo2{a} libcodec21{a} libcolord2{a} libcolorhug2{a} libcroco3{a} libcrystalhd3{a} libcups2{a} libcurl3{a} libdatrie1{a} libdconf1{a} libdrm-amdgpu1{a} libdrm-intel1{a} libdrm-nouveau2{a} libdrm-radeon1{a} libegl1-mesa{a} libepoxy0{a} libexif12{a} libflac8{a} libfontconfig1{a} libfontenc1{a} libfreeswitch1{a} libgbm1{a} libgd3{a} libgdk-pixbuf2.0-0{a} libgdk-pixbuf2.0-common{a} libgl1-mesa-dri{a} libgl1-mesa-glx{a} libglapi-mesa{a} libglu1-mesa{a} libgme0{a} libgomp1{a} libgphoto2-6{a} libgphoto2-l10n{a} libgphoto2-port12{a} libgraphite2-3{a} libgsm1{a} libgtk-3-0{a} libgtk-3-bin{a} libgtk-3-common{a} libgtk2.0-0{a} libgtk2.0-bin{a} libgtk2.0-common{a} libgudev-1.0-0{a} libgusb2{a} libharfbuzz0b{a} libice6{a} libieee1284-3{a} libjack-jackd2-0{a} libjbig0{a} libjpeg-turbo8{a} libjpeg8{a} libjs-jquery{a} libjs-jquery-metadata{a} libjs-jquery-tablesorter{a} libjson-glib-1.0-0{a} libjson-glib-1.0-common{a} liblcms2-2{a} libldns1{a} libllvm3.8{a} libltdl7{a} liblua5.2-0{a} libmemcached11{a} libmirclient9{a} libmircommon5{a} libmirprotobuf3{a} libmodplug1{a} libmp3lame0{a} libnotify4{a} libnspr4{a} libnss3{a} libnss3-nssdb{a} libodbc1{a} libogg0{a} libopencore-amrnb0{a} libopencore-amrwb0{a} libopenjpeg5{a} libopus0{a} liborc-0.4-0{a} libpango-1.0-0{a} libpangocairo-1.0-0{a} libpangoft2-1.0-0{a} libpciaccess0{a} libpixman-1-0{a} libportaudio2{a} libpq5{a} libprotobuf-lite9v5{a} libproxy1v5{a} libpython2.7{a} librest-0.7-0{a} librsvg2-2{a} librsvg2-common{a} libsamplerate0{a} libsane{a} libsane-common{a} libschroedinger-1.0-0{a} libsctp1{a} libsensors4{a} libshine3{a} libsilk1{a} libsm6{a} libsnappy1v5{a} libsndfile1{a} libsnmp-base{a} libsnmp30{a} libsoundtouch1{a} libsoup-gnome2.4-1{a} libsoup2.4-1{a} libsox-fmt-alsa{a} libsox-fmt-base{a} libsox2{a} libsoxr0{a} libspeex1{a} libspeexdsp1{a} libssh-gcrypt-4{a} libswresample-ffmpeg1{a} libswscale-ffmpeg3{a} libtcl8.6{a} libthai-data{a} libthai0{a} libtheora0{a} libtiff5{a} libtk8.6{a} libtwolame0{a} libtxc-dxtn-s2tc0{a} libva1{a} libvdpau1{a} libvorbis0a{a} libvorbisenc2{a} libvorbisfile3{a} libvpx3{a} libwavpack1{a} libwayland-client0{a} libwayland-cursor0{a} libwayland-egl1-mesa{a} libwayland-server0{a} libwebp5{a} libwxbase3.0-0v5{a} libwxgtk3.0-0v5{a} libx11-xcb1{a} libx264-148{a} libx265-79{a} libxaw7{a} libxcb-dri2-0{a} libxcb-dri3-0{a} libxcb-glx0{a} libxcb-present0{a} libxcb-render0{a} libxcb-shape0{a} libxcb-shm0{a} libxcb-sync1{a} libxcb-xfixes0{a} libxcomposite1{a} libxcursor1{a} libxdamage1{a} libxfixes3{a} libxft2{a} libxi6{a} libxinerama1{a} libxkbcommon0{a} libxmu6{a} libxpm4{a} libxrandr2{a} libxrender1{a} libxshmfence1{a} libxss1{a} libxt6{a} libxtst6{a} libxv1{a} libxvidcore4{a} libxxf86dga1{a} libxxf86vm1{a} libyaml-0-2{a} libzvbi-common{a} libzvbi0{a} mesa-vdpau-drivers{a} notification-daemon{a} openjdk-8-jre-headless{a} sox{a} tcl{a} tcl8.6{a} tk{a} tk8.6{a} ubuntu-mono{a} va-driver-all{a} vdpau-driver-all{a} vdpau-va-driver{a} x11-common{a} x11-utils{a} xbitmaps{a} xterm{a} yasm{a} 0 packages upgraded, 418 newly installed, 0 to remove and 2 not upgraded. Need to get 231 MB of archives. After unpacking 735 MB will be used. The following packages have unmet dependencies: freeswitch-meta-all : Depends: freeswitch-mod-flite (= 1.7.0~64~d6bf670-1~xenial+1) which is a virtual package and is not provided by any available package. The following actions will resolve these dependencies: Keep the following packages at their current version: 1) freeswitch-all [Not Installed] 2) freeswitch-meta-all [Not Installed] -- Víctor E. Medina M. Platform Architect / Chief Infrastructure +58424 291 4561 BB #79A8AFA2 @VMCibersys

1 0

Fwd: Freeswitch uses wrong ICE candidate to establish connection
by Александр Дружилов 27 Sep '16

27 Sep '16

---------- Forwarded message ---------- From: Александр Дружилов <alexdruzhilov(a)gmail.com> Date: 2016-09-26 14:31 GMT+03:00 Subject: Freeswitch uses wrong ICE candidate to establish connection To: freeswitch-users(a)lists.freeswitch.org *The short description of problem* We are using Freeswitch for conference calls through webRTC. Chrome collects ICE candidates (from two net interfaces: ethernet and wifi), sends them to Freeswitch and Freeswitch accepts one of this candidates. After that Freeswitch tries to make DTLS handshake, but Chrome ignores it and sends STUN pings over and over again. But if I disable one of this network interfaces, then everything works fine. From my point of view the problem is that Freeswitch uses wrong ICE candidate (it takes this candidate from STUN pings, but there is no exactly such candidate in SDP). I also admit that may be the root of the problem is inside Chrome (because Firefox works properly in this case), but I interested in possible explanations an may be solutions from the Freeswitch side. *Versions* Freeswitch 1.6.9, 1.6.10 Chrome 52, 53, 54 Server OS: CentOS 7.2.1511 Client OS: Ubuntu 16.04 *IP addresses* 1) 172.15.1.1 - local VM (VirtualBox) 2) 192.168.22.57 - local ethernet interface 3) 192.168.41.185 - local wi-fi interface 4) 52.202.171.116 - Freeswitch IP *Succsess case* Chrome sends offer with this candidates (Ethernet and WI-FI addresses) a=candidate:988003974 1 udp 2122260223 192.168.22.57 56687 typ host generation 0 network-id 1 a=candidate:3404510875 1 udp 2122194687 192.168.41.185 57467 typ host generation 0 network-id 4 network-cost 10 a=candidate:988003974 2 udp 2122260222 192.168.22.57 60099 typ host generation 0 network-id 1 a=candidate:3404510875 2 udp 2122194686 192.168.41.185 54887 typ host generation 0 network-id 4 network-cost 10 a=candidate:3148593202 1 udp 1686052607 80.254.60.17 56687 typ srflx raddr 192.168.22.57 rport 56687 generation 0 network-id 1 a=candidate:3148593202 2 udp 1686052606 80.254.60.17 60099 typ srflx raddr 192.168.22.57 rport 60099 generation 0 network-id 1 Freeswitch choose srflx candidate 2016-09-26 10:54:28.337897 [DEBUG] switch_core_media.c:6434 AUDIO RTP [sofia/external576460752303423473@wrike] 172.16.106.100 port 17052 -> 80.254.60.17 port 56687 codec: 111 ms: 20 Receives STUN pings from this interface and everything goes as expected - media stream established. *Failure case (add one interface (VirtualBox) or disable one of existing interfaces)* Chrome sends offer with one additional interface (VirtualBox). a=candidate:1861086701 1 udp 2122260223 172.15.1.1 43367 typ host generation 0 network-id 3 a=candidate:988003974 1 udp 2122194687 192.168.22.57 54230 typ host generation 0 network-id 1 a=candidate:3404510875 1 udp 2122129151 192.168.41.185 42752 typ host generation 0 network-id 4 network-cost 10 a=candidate:1861086701 2 udp 2122260222 172.15.1.1 55301 typ host generation 0 network-id 3 a=candidate:988003974 2 udp 2122194686 192.168.22.57 44112 typ host generation 0 network-id 1 a=candidate:3404510875 2 udp 2122129150 192.168.41.185 46978 typ host generation 0 network-id 4 network-cost 10 a=candidate:3148593202 1 udp 1685987071 80.254.60.17 54230 typ srflx raddr 192.168.22.57 rport 54230 generation 0 network-id 1 a=candidate:3148593202 2 udp 1685987070 80.254.60.17 44112 typ srflx raddr 192.168.22.57 rport 44112 generation 0 network-id 1 Freeswitch choose VirtualBox interface as main candidate 2016-09-26 11:03:40.513295 [DEBUG] switch_core_media.c:6434 AUDIO RTP [sofia/external/576460752303423471@wrike] 172.16.106.100 port 17000 -> 172.15.1.1 port 42374 codec: 111 ms: 20 After a few STUN pings Freeswitch changes his main candidate to the public address of my Wi-Fi interface (but it was not included in SDP by chrome and I don't know exactly why). 2016-09-26 11:03:40.912987 [NOTICE] switch_rtp.c:1258 Auto Changing video stun/rtp/dtls port from 172.15.1.1:43367 to 80.254.60.17:42752 Freeswitch tries to make DTLS handshake with this candidate, but fails because Chrome ingores this "Client Hello" request. All necessary data attached for success and failure cases (freeswitch logs, wireshark dump made on client, SDP).

4 7

Verto response headers
by Oleg Stolyar 27 Sep '16

27 Sep '16

Hello, In SIP we can add custom headers to the response which the calling client can examine. Is there a way to do something like this in Verto?

1 0