July 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_v8 no rpm package for centos7
by Jason Moran 08 Jul '16

08 Jul '16

As noted in this JIRA https://freeswitch.org/jira/browse/FS-6748 when installing rpms via yum (yum install freeswitch-v8) no mod_v8.so gets created, so mod_v8 cannot be used in an rpm-installed environment. Besides reverting to the git/bootstrap/make/make install method, is there another way to get mod_v8.so ?

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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

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mobile SIP provisioning with QR codes, TLS
by Daniel Pocock 07 Jul '16

07 Jul '16

We've got some GSoC students working on SIP and RTC related projects[1] this year under Debian. One of them will be looking at creating a library that can be used for provisioning apps like Lumicall[2] and CSipSimple[3] to work with servers like FreeSWITCH If anybody would like to have some input on how the QR code or TLS certificates should be crafted, please come and join us on the free-rtc list: https://lists.fsfe.org/mailman/listinfo/free-rtc Regards, Daniel 1. https://wiki.debian.org/SummerOfCode2016/RTCProjects 2. https://lumicall.org 3. https://play.google.com/store/apps/details?id=com.csipsimple&hl=en

1 0

multiple FS presence
by Assaf Dahary 07 Jul '16

07 Jul '16

Hi, I have multiple FS nodes where each of them maintains locally its own MySql DB/tables (created automatically on FS installation with dsn setup). Now I'm looking for a way to share clients presence between the FS nodes. I read the latest discussion on this issue which suggested to use common DB for all FS nodes with the same switchname in the switch.xml. But this will not work for me as in my case each FS node needs to maintains locally its own Freeswitch DB tables (there is a reason why it is setup like that). In each FS node I see that the sip_presence table is being updated with each sip client event (reg, unreg, etc). I already have a script that already handles each FS events so I can log all register/unregister/expire events. So my question is, how can I reflect sip client presence events from one FS node to another? Regards Assaf

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webRTC on firefox
by 小原崇寛 07 Jul '16

07 Jul '16

I want to let webrtc communicate using freeswitch. But I cannot let you communicate well in firefox The version of freeswitch is 1.6 The version of firefox is 47 Probably I think that it is caused by codec. So I put setting of codec of vars <X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,VP8,speex@8000h@20i,speex@16000h@20i,H264,H263-1998"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=OPUS,G722,PCMU,PCMA,VP8,speex@8000h@20i,speex@16000h@20i,H264,H263-1998"/> please help me

1 0

(no subject)
by 小原崇寛 07 Jul '16

07 Jul '16

I want to let webrtc communicate using freeswitch. But I cannot let you communicate well in firefox The version of freeswitch is 1.6 The version of firefox is 47 Probably I think that it is caused by codec. So I put setting of codec of vars <X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,VP8,speex@8000h@20i,speex@16000h@20i,H264,H263-1998"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=OPUS,G722,PCMU,PCMA,VP8,speex@8000h@20i,speex@16000h@20i,H264,H263-1998"/> please help me

1 0

variable_sip_outgoing_contact_uri question
by Anonim Stefan 06 Jul '16

06 Jul '16

Hi, In my scenario, that variable looks like this: variable_sip_outgoing_contact_uri: [<sip:gw+mygw1@10.1.1.1:5060 ;transport=udp;gw=mygw1>] Is it possible, to make it look like this? : variable_sip_outgoing_contact_uri: [<sip:10.1.1.1:5060>] So far, the things I've found for the gateway configuration are: <param name="extension-in-contact" value="true"/> <param name="extension" value="contact_username_part"/> ...which only replaces the username part, and: <param name="contact-params" value="new_contact_parameter"/> ... which only adds to the contact parameters. I tried "" and "_undef_" combinations without success. Can you give me some clues about how to trim those contact header parts, if possible? Thanks, Stefan

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Security vs compatibility / NAT etc
by Rick Jarvis 06 Jul '16

06 Jul '16

I’d be interested to hear what different people use to provide some level of security for remote end-users such as homeworkers, and to get round NAT issues. We currently use OpenVPN, as this is built into the firmware of Yealink handsets (it’s a great feature, I’m not sure why more handset manufacturers don’t do this?!). The pros are that not only is it secure, but it also removes any problems with NAT for the RTP streams. The downsides are that it is complicated (and downright frustrating sometimes) to set up, and there are additional things to consider such as the server configuration and overheads. TLS/SSL with SRTP is another option, but my understanding of this is that it can cause NAT problems, with FreeSWITCH trying to initiate control channels back to the phone for inbound calls. In fact, I’ve always had problems with getting phones to work when behind NAT anyway, even without SSL/TLS. STUN can be used to ascertain the IP, but how do you handle situations where multiple handsets are behind NAT - you can’t open all RTP ports to all handsets at once?!! Would be very interested to hear thoughts and methods on these points. Thanks R

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Allow/Supported/Allow-Events remove?!
by Anonim Stefan 06 Jul '16

06 Jul '16

Hi, Because I have a scenario in which needs to send some X- headers in the SIP originated by Freeswitch, the packets become too big and fragmentation occurs. Firewalls don't like fragmentation and won't let my SIP communication pass. The plausible headers from where I could gain some bytes, besides X-FS-Support and Remote-Party-ID, are Allow/Supported/Allow-Events. Any way to remove them? Googling didn't help. I tried unseting sip_h_Allow/Supported/Allow-Events in dialplan, without success. As a temporary solution I am deleting all those headers on the SIP server side. It would be nice if I could remove them on Freeswitch side, because I would remove them only for this scenario and keep them for all others. Thanks, Stefan

3 3

Have same UUID in all call legs
by Victor Chukalovskiy 06 Jul '16

06 Jul '16

Good day, the scenario is that I have one a-leg and numerous b-legs for each call in my CDR. I need a simple way of selecting all leg for the same call at once Previously I've relied on things like ${bleg_uuid}, ${last_bridge_to} etc. But it in the end it involves a 2-step process or inner join to put a- and b- legs together. What would really help is if there is a way to have an extra variabvle which will have the same value for every leg of the same call. Can someone tell me if such var already exists, or if I need to generate it myself and export to both a- and b- legs? Thank you,

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