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Issue with mod_verto enabled inside modules.conf.xml and originate not working correctly
by Andrew Keil 20 May '16

20 May '16

To FreeSWITCH Users, I thought I would pass this on since I spent all day yesterday trying to track this issue down. See below for my setup of FreeSWITCH version 1.6.8 (this problem also happens on 1.6.5 since that is why I upgraded to see if the problem was fixed or not). I can reproduce this issue on FreeSWITCH 1.6.5 and 1.6.8 100% of the time. Basically if mod_verto is not commented out inside the modules.conf.xml file then the "originate" does not function as it is supposed to. See at the bottom of this message my test Lua script (visytel-test-out.lua). I setup a simple SIP phone (Bria) to register with FreeSWITCH under user 1000 (sofia status profile internal reg checked out OK) then called the originate command: originate {ignore_early_media=true,call_timeout=15,return_ring_ready=true}user/1000 00000000 When mod_verto is commented out and FreeSWITCH is started everything works OK (except session:hangupCause() always returns NONE - but I can work around this) When mod_verto is not commented out and FreeSWITCH is started originate gets stuck and never starts the Lua script (unless answered) despite return_ring_ready=true. Luckily I currently do not need video on my FreeSWITCH installations so I can happily comment out mod_verto, however this may not be the case in the future. Kind Regards, Andrew Keil Visytel Pty Ltd New setup of FreeSWITCH 1.6.8 on up-to-date CentOS 6.7 yum -y install epel-release yum -y install git gcc-c++ autoconf automake libtool wget python ncurses-devel zlib-devel libjpeg-devel openssl-devel sqlite-devel libcurl-devel libxml2-devel libidn-devel readline-devel These should be already installed and up-to-date yum -y install e2fsprogs-devel pcre-devel speex-devel ldns-devel libedit-devel libyuv-devel opus-devel libvpx-devel unbound-devel libuuid-devel libsndfile-devel sox These should install Note: No packages for: libvpx2*, libdb4* (so mod_vpx & mod_fsv will not be supported) cd /usr/local/src git clone -b v1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git cd /usr/local/src/freeswitch ./bootstrap.sh vi modules.conf Comment out (ie. place '#' infront): mod_fsv mod_h26x Save and close (<Esc> :wq!) ./configure --disable-libvpx --disable-libyuv yum install yasm-devel {This is extra from the original yum install pre-requisites} make make install make cd-sounds-install make cd-moh-install cd /usr/local/freeswitch/conf vi vars.xml Change: <X-PRE-PROCESS cmd="set" data="default_password=1234"/> {!!set it to something different!!} Save and close (<Esc> :wq!) cd /usr/local/freeswitch/conf/autoload_configs vi modules.conf.xml comment out (ie. place '' at the end): mod_fsv Save and close (<Esc> :wq!) cd /usr/local/freeswitch/bin ./freeswitch -nonat -nonatmap version FreeSWITCH Version 1.6.8+git~20160505T153832Z~99de0ad502~64bit (git 99de0ad 2016-05-05 15:38:32Z 64bit) ... to exit from FreeSWITCH console Here is my test Lua script which was placed inside scripts/visytel-test-out.lua: local trace_string -- Script tested by using originate: originate {ignore_early_media=true,call_timeout=15,return_ring_ready=true}user/1000 00000000 --[[ Dialplan inside conf/dialplan/default/00_visytel_out.xml: <include> <extension name="visytel_out"> <condition field="destination_number" expression="00000000"> <action application="lua" data="visytel-test-out.lua"/> </condition> </extension> </include> ]]-- -- Trace some channel variables here trace_string = session:getVariable("uuid") freeswitch.consoleLog("INFO", trace_string) trace_string = session:getVariable("destination_number") freeswitch.consoleLog("INFO", trace_string) trace_string = session:getVariable("ani") freeswitch.consoleLog("INFO", trace_string) trace_string = session:getVariable("direction") freeswitch.consoleLog("INFO", trace_string) trace_string = session:getVariable("channel_name") freeswitch.consoleLog("INFO", trace_string) -- waitForAnswer here freeswitch.consoleLog("INFO", "Prior to session.waitForAnswer()") session:waitForAnswer(session) freeswitch.consoleLog("INFO", "After session.waitForAnswer()") -- Get Hangup Cause {this does not seem to work} freeswitch.consoleLog("INFO", session:hangupCause()) -- This seems to be always a value of NONE so not much use -- Handle whether outbound call was ANSWERED or not here if (session:getState() ~= "CS_HANGUP") then freeswitch.consoleLog("INFO", "OUTBOUND ANSWERED") session:sleep(2000) session:hangup() -- Just HANGUP for now else freeswitch.consoleLog("INFO", "OUTBOUND FAILED") session:hangup() end

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event socket error
by Nduwayezu, Joselyne 19 May '16

19 May '16

I'm having Socket Connection error when i do fs_cli. when i do netstat -nlp | grep 8021, i see that the event socket is not connecting to the localhost as defined in event_socket.conf.xml, but to another ip 0.0.31.85 i dont know where is coming from. I tried since yesterday to fix the problem. NDUWAYEZU Joselyne -- Haute École de Bruxelles

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Javascript and freeswitch
by Nduwayezu, Joselyne 19 May '16

19 May '16

Hello, How do you write script in javascript when you want to invoke it in the dialplan? I tried this: #!/usr/bin/js session = new Session ("sofia/external/xxxx(a)yyyy.fr"); session.waitForanswer(10000); if(session.ready()) { session.sleep(1000); session.streamFile("path to the file"); But when i execute it, i get the error: ReferenceError: Session is not defined How can i fix it? NDUWAYEZU Joselyne -- Haute École de Bruxelles

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Nuance cloud services HTTP integration
by Dave Horton 19 May '16

19 May '16

Has anyone tried to integrate ASR using Nuance’s NCS HTTP-based interface? It looks like it would require recording a channel and streaming it in real time via HTTP to their cloud service, just wondering if anyone has thought about how to accomplish that.

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Bypass media with callcenter
by Szalánczi Gábor 19 May '16

19 May '16

Hello! We are developing a FreeSwitch based solution for a sign-language interpreter service for hearing impaireds. The system is already somewhat operational, but we are facing serious audio/video quality issues (to the point of being completely unusable), presumably caused by our virtualized PBX environment in passthrough media mode. Indeed, by switching the calls to bypass media mode from the dialplan, most of these problems vanished. The problem is the callcenter module, which is a crucial component for the operation of our service, but seemingly does not honor any setting meant to switch the calls to bypass mode. We have tried setting, exporting (even with cc_export_vars) the bypass_media=true and bypass_media_after_bridge=true variables from the dialplan before calling the callcenter application, setting them from the dialstring of the callcenter agents, and configuring the profile with inbound-bypass-media=true. None of these worked, the media streams of the calls serviced from the queue still got relayed by FreeSwitch. The only way we were able to remove the PBX from the media path was using the uuid_media_3p off API call, issued from an ESL PHP script after the call is established between a client and an agent (from a custom event handler for the bridge-agent-start callcenter action). The main problem with this approach that the callcenter app initiates the call to the agent without ICE protocol (since it wants the PBX to relay the media streams), so our custom liblinphone-based desktop client will also respond to the b-leg REINVITE (triggered by uuid_media_3p) without ICE, resulting no media at all in bypass mode. Our custom mobile clients are ICE enabled, so the a-leg properly answers the REINVITE with ICE, and its SDP with candidate attributes gets forwarded to the b-leg, but since it had not started an ICE session at the initial call setup, it will ignore it, and send back a plain SDP just with its internal IP and ports. Is there any way to force the callcenter app to send its initial INVITE with ICE? We have noticed that the a-leg's INVITE with candidates in its SDP is responded by FreeSwitch with a single candidate line for itself (this way it also keeps the ICE session alive at the client, so it can use it later responding the REINVITE), so hopefully it is possible. We are open to any other suggestions which leads to bypass media mode with callcenter. Best Regards! Gábor Szalánczi cid:image001.gif@01D1B1E9.D64F69C0 E‑Group cím: telefon: fax: 1062 Budapest, Váci út 1-3. Westend Irodaház "C" torony, 5. emelet (+36-1) 371 25 55 (+36-1) 371 25 56 web: telephely: www.egroup.hu <http://www.egroup.hu/> 3525 Miskolc, Madarász Viktor u. 13/A Szalánczi Gábor szoftverfejlesztő / Software Developer

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Disable annexb for g729
by Mike Rice 19 May '16

19 May '16

We have a carrier the mandates that annex b be disabled on the invites to them. I have added the following to the default dialplan but nothing seems to change the invite on the B leg to the carrier. <extension name="disable-annexB" continue="true"> <condition field="${switch_r_sdp}" expression="/(.*)(m=audio \d+ RTP\/AVP)(.*)( 18 )(.*)/s"> <action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/> </condition> </extension> <action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/> and <action application="bridge" data="{sip_append_audio_sdp=a=fmtp:18 annexb=no,absolute_codec_string=^^:G729:PCMU:PCMA}sofia/gateway/carrierGW/$1"/> inbound-late-negotiation is set to true. The logs show that it is exporting: [DEBUG] switch_channel.c:1267 EXPORT (export_vars) [sip_h_Diversion]=[<sip:XXXXXXXXXX@10.10.X.X>;reason=unavailable] EXECUTE sofia/internal/1000(a)10.10.X.X bridge(sofia/gateway/lo7f/XXXXXXXXXX) [DEBUG] switch_channel.c:1221 sofia/internal/1000(a)10.10.X.X EXPORTING[export_vars] [sip_append_audio_sdp]=[a=fmtp:18 annexb=no] to event [DEBUG] switch_channel.c:1221 sofia/internal/1000(a)10.10.X.X EXPORTING[export_vars] [sip_h_Diversion]=[<sip:XXXXXXXXXX@10.10.X.X>;reason=unavailable] to event The SDP does not reflect: Local SDP: v=0 o=FreeSWITCH IN IP4 10.10.X.X s=FreeSWITCH c=IN IP4 10.10.X.X t=0 0 m=audio 20022 RTP/AVP 18 0 8 101 13 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv Any help would be greatly appreciated. Thanks!

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freeswitch crashes
by Nduwayezu, Joselyne 19 May '16

19 May '16

Hello, Why freeswitch crashes when i type fsctl shutdown restart in CLI? it telles: Socket Interrupted, Bye How to resolve the problem ? How to restart freeswitch without having this kind of error? Thank you NDUWAYEZU Joselyne -- Haute École de Bruxelles

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cannot use h264 with firefox and verto demo page
by Antonio Silva 19 May '16

19 May '16

Hi, I'm trying to use H264 with verto demo page with firefox but no luck making it work... using firefox it never detects or send the h264 codec in the SDP. Is it possible to use this configuration? did anyone try to make video calls with verto and h264? If i use VP8 it works ok but with transcoding. My idea is to avoid unnecessary transcoding between local sip endpoints that only support h264. I've enabled the h264 in about:config, media.peerconnection.video.h264_enabled;true In the endpoint i've set the codecs: <variable name="absolute_codec_string" value="opus,h264"/> <variable name="codec_string" value="opus,h264"/> I see FS send the offer with only h264 to firefox but firefox always reply with only vp8. The call flow is from FS to Firefox. You can find the logs for the sdp part in the pastebin: https://pastebin.freeswitch.org/view/b111a3de#cmX8ZzjHVyq3OiM3yiHYE9tQmjnXU… Thanks for the help! Regards, António

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Call-Info header
by Andrew Cassidy 18 May '16

18 May '16

Hi Guys, So I've been playing with some bits and pieces with the Call-Info header, there are 3 main uses for this: - SLA - Auto Answer - Display Pictures Now, the issue is whilst all 3 can be present in a single copy of the header, having multiple Call-Info headers can lead to them being ignored by the phone or crashing it entirely. My question is is there a way to do what I'm trying to do? I notice there's an extra data field that in the source that is prepended to the string so is there perhaps a way to set that data? Or is a better approach to abstract this header entirely, making it a hash or list we can add to at will? Kind regards, -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director 03303 880 960 andrew(a)cassidyweb.co.uk <andrew(a)cassidywebservices.co.uk> www.cassidyweb.co.uk <http://www.cassidywebservices.co.uk>

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wss and nginx
by Philipp Zeitschel 18 May '16

18 May '16

Hi, i've googled for hours but i don't get it to work i have a nginx proxy at my public ip and it forwards the traffic through 10.1.71.5 to my freeswitch box at 10.1.71.10 i've set ext-rtp-ip and ext-sip-ip to my external IP adress and created the acl.conf.xml in autoload_configs with following content: <configuration name="acl.conf" description="Network Lists"> <network-lists> <list name="lan" default="deny"> <node type="allow" cidr="10.1.1.1/32" /> </list> </network-lists> </configuration> In fs_cli i get the message 2016-05-18 08:13:43.245201 [WARNING] switch_core_media.c:3458 NO candidate ACL defined, Defaulting to wan.auto So i think the acl is working. My nginx config looks like this: server { listen 0.0.0.0:443; server_name hostname; keepalive_timeout 70; ssl on; ssl_certificate /etc/ssl/my.ct; ssl_certificate_key /etc/ssl/my.key; ssl_dhparam /etc/ssl/dhparam.pem; ssl_protocols TLSv1 TLSv1.1 TLSv1.2; ssl_prefer_server_ciphers on; ssl_session_cache shared:SSL:10m; ssl_ciphers 'EECDH+AESGCM:EDH+AESGCM:AES256+EECDH:AES256+EDH'; location / { proxy_set_header Upgrade $http_upgrade; proxy_set_header Connection $connection_upgrade; proxy_set_header X-Real-IP $remote_addr; proxy_set_header Host $host; proxy_set_header X-Forwarded-For $proxy_add_x_forwarded_for; proxy_read_timeout 3600; proxy_http_version 1.1; proxy_pass https://10.1.71.10:7443; } } I can login and establish calls but i don't hear voice so i think the rtp packages are not ok Because of that i've enabled sip trace and get this: send 1668 bytes to wss/[10.1.71.5]:32890 at 08:13:43.266640: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/WSS keenaijeb5h6.invalid;branch=z9hG4bK779005;received=10.1.71.5;rport=32890 From: <sip:user@domain>;tag=n8m38teo73 To: <sip:conf@domain>;tag=gX23c0rey78vH Call-ID: toghv9ifikpaeib85mh8 CSeq: 6099 INVITE Contact: <sip:conf@138.201.x.x:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.6.8-15-99de0ad~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 901 Remote-Party-ID: "conf" <sip:conf@domain>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1463528835 1463528836 IN IP4 10.1.71.10 s=FreeSWITCH c=IN IP4 10.1.71.10 t=0 0 a=msid-semantic: WMS 7y98fmZXWMapBpCUcRfJdKQGM9JU5t83 a=end-of-candidates m=audio 30388 UDP/TLS/RTP/SAVPF 111 101 106 a=rtpmap:111 opus/48000/2 a=fmtp:111 useinbandfec=1; minptime=10 a=rtpmap:101 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 5E:AE:56:B1:0B:39:38:7B:00:6F:8A:84:F6:0F:DC:86:8D:24:9A:7D:4D:59:F8:95:8F:CC:ED:A2:6F:90:84:2C a=setup:active a=rtcp-mux a=rtcp:30388 IN IP4 10.1.71.10 a=ice-ufrag:wLDAlj6YjGJfZH8H a=ice-pwd:F0vQH6N0H0LyFhs1SVr45oSe a=candidate:0546395930 1 udp 659136 10.1.71.10 30388 typ host generation 0 a=ssrc:2671785415 cname:R71XkQK7tLmU0PF3 a=ssrc:2671785415 msid:7y98fmZXWMapBpCUcRfJdKQGM9JU5t83 a0 a=ssrc:2671785415 mslabel:7y98fmZXWMapBpCUcRfJdKQGM9JU5t83 a=ssrc:2671785415 label:7y98fmZXWMapBpCUcRfJdKQGM9JU5t83a0 ------------------------------------------------------------------------ My freeswitch (running on debian jessie) version is FreeSWITCH Version 1.6.8-15-99de0ad~64bit (-15-99de0ad 64bit) What am i doing wrong? What else can i do to force freeswitch to use my external IP in the RTP Packages? Regards Philipp

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