May 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FS6947 - Tuning Opus bandwidth
by Emrah 26 Oct '16

26 Oct '16

Hi list, I re-read FS6947 and don't understand how this problematic was addressed and the issue fixed. The scope is simple. There should be a setting in the dialplan that allows downsampling of Opus for applications that do not require the 48khz / 2 channels framework. I.e.: terminating to the PSTN with Opus to take advantage of low bandwidth and great PLC. There seems to be a lot of confusion around bandwidth in general there. It doesn't matter if the internal clock of the device is always sampling at 48khz / 2ch. There are settings that can facilitate a lower bandwidth consumption for particular use cases, and it seems the reason it is not being implemented in FS is just a matter of being confused about the intent of the 48khz 2ch base. Please revisit this issue. FS should allow tuning of Opus audio / network bandwidth in the dialplan. It would optimize greatly lots of use cases. If I'm calling the PSTN, I'd rather have my client downsample and stream at a lower bandwidth, even if my audio capture would still be at 48khz / 2ch as per the RFC, and save on bandwidth, than transcode the full 48khz spectrum into PCM on my FS and minimize processing power on the client's side. Jira here: https://freeswitch.org/jira/browse/FS-6947 Emrah

8 16

Capturing the Current Call Audio
by Leonardo Lima Ribeiro 13 Oct '16

13 Oct '16

Hi! I’m tying to capture the current call audio but I have no idea on how to do that. My main achievement is to get the current call audio and them stream this audio bytes to an API via HTTP Post. I was doing some tries with mod_shout and mod_vlc, and I was successful in the streaming part - I tested some wavs and mp3 files. But the mainly part is that I need the current call audio, and it has no file in the system to stream right? I was thinking if there’s a command that I can get the audio bytes from the current call but I really don’t know, I could not find any option like that. Could you please help me? Thank you, Leonardo Ribeiro

6 10

Kernel Hyperthreading ON or OFF?
by Raimund Sacherer 06 Oct '16

06 Oct '16

Hi, we have a new Freeswitch cluster with Dell R410. It has 2 CPUs (Xeon E5620, QuadCore. it's 8 cores withouth HT and 16 cores with HT. I googled with hyperthreading and freeswitch but some say turn it on it will give you a boost, others say turn it off, you will gain performance. Does anyone know if it's better to leave it on or turn it off? And some small explanation? I would be very gratefull for your help, best regards Raimund --

2 1

freeswitch 1.6 without video support
by Grant Bagdasarian 19 Aug '16

19 Aug '16

Hello, Is it possible to compile and install fs 1.6 without support for video? Or is the 1.4 version the same as 1.6 with bug fixes and security updates? Regards, Grant

7 14

Presence issue. How packet should look?
by Igor Olhovskiy 26 Jul '16

26 Jul '16

Hi! I’m trying to subscribe to Freeswitch with this packet SUBSCRIBE sip:*55@master.rufan.at SIP/2.0 Record-Route: <sip:10.0.20.70:5060;ftag=jz8i26dy0f;lr=on> Via: SIP/2.0/UDP 212.232.26.232:5060 ;branch=z9hG4bKbb4c.eafe0e8f9ec4b28fb6e4056a64c702ed.0 Via: SIP/2.0/UDP 192.168.88.59:37600 ;received=176.37.91.15;branch=z9hG4bK-rzfuku5tc99o;rport=37600 Route: <sip:10.0.20.70:5060;ftag=jz8i26dy0f;lr=on> From: <sip:607@master.rufan.at>;tag=jz8i26dy0f To: <sip:*55@master.rufan.at;user=phone>;tag=qBuxIeW1MNWF Call-ID: 313435373534383931393431313632-94bfcwdyrzz2 CSeq: 610 SUBSCRIBE Max-Forwards: 69 User-Agent: snom760/8.7.5.35 Contact: <sip:607@176.37.91.15:37600;line=ukipitb6>;reg-id=1 Event: dialog Accept: application/dialog-info+xml Expires: 80 Content-Length: 0 Path: <sip:212.232.26.232:5060;lr;received=sip:176.37.91.15:37600> X-AUTH-IP: 176.37.91.15 Actually I get Accepted (202) But with NOTIFY from freeswitch I got this NOTIFY sip:607@176.37.91.15:37600;line=ukipitb6 SIP/2.0 Via: SIP/2.0/UDP 10.0.20.71;rport;branch=z9hG4bKS5ZKag4N6XXFK Route: <sip:10.0.20.70:5060>;ftag=jz8i26dy0f;lr=on Record-Route: <sip:10.0.20.70:5060>;ftag=jz8i26dy0f;lr=on Max-Forwards: 70 From: <sip:*55@master.rufan.at;user=phone>;tag=qBuxIeW1MNWF To: <sip:607@master.rufan.at>;tag=jz8i26dy0f Call-ID: 313435373534383931393431313632-94bfcwdyrzz2 CSeq: 297327351 NOTIFY Contact: <sip:*55@10.0.20.71:5060> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Event: dialog Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=80 Content-Type: application/dialog-info+xml Content-Length: 152 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="8" state="full" entity="sip:*55@master.rufan.at"> </dialog-info> Problem is, XML data is empty, cause actually there is no registrations for *55 extension. Why it’s not sending «Unavail» or smth. like this? What is the correct packet for SUBSCRIBE? Or internal profile? Internal profile is nonce-ttl [60] auth-calls [true] inbound-reg-force-matching-username [true] auth-all-packets [false] ext-rtp-ip [10.0.20.71] ext-sip-ip [10.0.20.71] rtp-timeout-sec [300] rtp-hold-timeout-sec [1800] tls-verify-policy [all] multiple-registrations [contact] enable-timer [false] dbname [share_presence] send-presence-on-register [true] inbound-codec-negotiation [greedy] NDLB-force-rport [safe] challenge-realm [auto_to] outbound-proxy [10.0.20.70] track-calls [true] nat-options-ping [true] liberal-dtmf [true] all-reg-options-ping [true] force-publish-expires [true] unregister-on-options-fail [true] user-agent-string [FreeSWITCH] log-auth-failures [true] forward-unsolicited-mwi-notify [false] context [public] rfc2833-pt [101] sip-port [5060] dialplan [XML] dtmf-duration [2000] inbound-codec-prefs [G7221@32000h,G7221@16000h,G722,PCMU,PCMA,OPUS,SILK] outbound-codec-prefs [G7221@32000h,G7221@16000h,G722,PCMU,PCMA,OPUS,SILK] rtp-timer-name [soft] rtp-ip [10.0.20.71] sip-ip [10.0.20.71] hold-music [local_stream://default] apply-nat-acl [nat.auto] aggressive-nat-detection [true] apply-inbound-acl [domains] local-network-acl [localnet.auto] record-path [/usr/local/freeswitch/recordings] record-template [${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}] manage-presence [true] presence-probe-on-register [true] manage-shared-appearance [true] tls [false] -- Best regards, Igor

2 14

Update RTP Address On Re-Invite
by Colin Morelli 08 Jul '16

08 Jul '16

I have mobile applications running behind NAT. When their reachability changes (and the device's local route updates), I want to automatically switch the RTP stream to the new address/port combination. I've tried using the RTP ALWAYS_AUTO_ADJUST bug, but that results in *very* choppy audio. I also don't really need it, as the only time I care to perform RTP auto-adjust is after an invite session. For this case I can safely assume that the client *will* send a re-invite when its address changes. Is there any way to perform RTP audio adjust only on a re-invite? Using STUN doesn't seem right for two reasons: 1) on its own it can't solve this problem (even if it determines the external IP, the port is still wrong). 2) it results in issues on FS "Invalid STUN/ICE packet received 20 bytes" Would appreciate any help. Best, Colin

2 7

Re- End Lua script after HangupHook handled without all the extra code to handle the return to the function
by Andrew Keil 08 Jul '16

08 Jul '16

To FreeSWITCH Users, See below for a sample template for a Lua Service Script running inside FreeSWITCH. The issue I have is fairly straightforward. I need a function to run when hangup is detected (ie. at the end of the call) however I understand this must not delay ending the script. This function is CleanUp(). Then I would like the service to end. The problem I am having is if the caller hangs up during the playback of "intro.wav" (as shown inside the MainService() function below), then the code jumps to the myHangupHook which calls CleanUp() perfectly, the issue is once CleanUp() is complete I would like the Lua script to end there and then (ie. at the bottom of CleanUp()). What actually happens is it returns to MainService() and continues to try and play "info.wav", unless I either check for session:ready() everywhere or add a goto as shown below under each streamFile() function call. My aim is to reduce extra code and to make the Lua script simpler and easier to read. Also I would like to try and avoid goto statements, which I know can be done with if (session:ready()) etc.... So is there a way to stop a Lua script running inside FreeSWITCH cleanly? I have tried the os.exit() this is barred from use by FreeSWITCH. I have also tried session:destroy() which crashes FreeSWITCH (version 1.6.5 on CentOS 6.7, CentOS 7 and windows) 100% of the time! I could look further into the Lua additions done by the FreeSWITCH team in the source code, however if someone has already solved this then that would be the best solution. FYI: Obviously the script below is simple, however I am sure that you understand if the script was complicated having to use "if (session:ready()) then ...." or "if (not session:ready()) then goto HANGUPEXIT end" makes the code ugly. Thanks in advance, Andrew Keil Visytel Pty Ltd ------------------------------------------------------------------------------ Sample Lua Service ----------------------------------------------------------------------- -- Lua template for FreeSWITCH service -- By: Andrew Keil (Visytel Pty Ltd) -- Email: support(a)visytel.com -- Setup script wide variables here function PreAnswer() freeswitch.consoleLog("INFO", "PRE ANSWER SECTION\n"); -- Add your pre answer code from here -- End of your pre answer code freeswitch.consoleLog("INFO", "PRE ANSWER SECTION COMPLETE\n"); end function AnswerCaller() session:answer() session:sleep(1000) end function MainService() freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION\n"); if (session:ready()) then -- Note (1): If you wish to end the call then simply use: goto ENDSERVICE -- Note (2): To terminate the service sooner when HANGUP is detected use: if (not session:ready()) then goto HANGUPEXIT end -- Add your main service code from here (caller would have been answered) session:streamFile("intro.wav") if (not session:ready()) then goto HANGUPEXIT end session:streamFile("info.wav") if (not session:ready()) then goto HANGUPEXIT end session:streamFile("outro.wav") if (not session:ready()) then goto HANGUPEXIT end -- End of your main service code end ::ENDSERVICE:: if (session:ready()) then -- End of service so hangup session:hangup() -- Should automatically jump to CleanUp() via hangup handler if caller still online at this stage end goto END ::HANGUPEXIT:: freeswitch.consoleLog("INFO", "END OF SERVICE (HANGUP DETECTED)\n"); ::END:: freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION COMPLETE\n"); end function CleanUp() freeswitch.consoleLog("INFO", "CLEANUP SECTION\n"); -- Add your cleanup code from here (caller would have been disconnected) -- End of your cleanup code freeswitch.consoleLog("INFO", "CLEANUP SECTION COMPLETE\n"); end function myHangupHook(s, status, arg) session:hangup() CleanUp() -- Run CleanUp function now since the caller has disconnected end -- Setup Hangup event handler here v_hangup = "HANGUP" session:setHangupHook("myHangupHook", "v_hangup") -- Call service functions in order PreAnswer() AnswerCaller() MainService() -- End of Lua service

5 16

calling all native speakers of languages other than english. We need your help, and this will be easy.
by Michael Jerris 20 Jun '16

20 Jun '16

Just committed into tree is internationalization of Verto Communicator, so far, just English and Italian. If you speak another language, we need your help to translate these into other languages. The file that needs to be translated is: https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/html5/vert… <https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/html5/vert…> If you can assist with this, Please make a new file named locale-language.json (where "language" is the 2 letter code for your language) and translate all the text in quotes to the right thing for your language and send them back our way. Lets see how quickly we can crowd source this Thanks!

13 21

Run a LUA script when voicemail received
by Steven Schoch 16 Jun '16

16 Jun '16

This sounds like a FAQ, but my searches found nothing. I'd like to run a LUA script when a message is left on voicemail. My alternative is to write my own mailer-app script that would scan the message to see if it's from mod_voicemail, do what I want to do, or call exim otherwise, but this seems like something that could be done from the dialplan. Is there a way in the dialplan to run an action with parameters from the voicemail message left? The reason I'm asking this is that I want to use the new Flowroute SMS API to send a text message when a voicemail is received, with the caller-ID info from the message. -- Steve

5 13

ReInvte in bypass/proxy mode | a > a+v
by Mirko Brankovic 13 Jun '16

13 Jun '16

Hi, Can someone help me figure out the problem i am hitting. Im trying to upgrade call from audio to audio + video (opus 48k and vp8- but this one is only in passtrough mode). So set up is browser (sip.js/node.js) >> kamailio wss >> freswitch (bypass mode bridge without answer) >> kamailio >> browser (node.js) Problem is that freeswitch receives reInvite on existing audio call with new sdp containing also video codecs, but directly from client so no proxy (kamailio) in path, and exiting invite towards kamailio for outgoing leg is missing video in a=group BUNDLE, so only audio and a=sendrcv is also being stripped (that is the difference to 1 on 1 video call). I understood that bypass mode shouldn't change/touch sdp. Also passtrough codec shouldn't be the problem i guess since both clients can decode it. Any help is appreciated. Thanks, Mirko

2 2