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FREESWITCH and Verto Communicator
by Alexey Gorbachev 12 Feb '17

12 Feb '17

Hi, everybody! I'm new in FreeSwitch. I installed it via Quick Start FreeSWITCH Demo With Verto Communicator in VirtualBox (NET mode is NAT mode). The only difference was that I generated letsencrypt certificate manually. I can call from Linphone to my freeswitch server, but I'm getting the "Waiting for server reconnection" error in 50-80% of times when accessing it via verto communicator from localhost. What's the problem? Thanks in advance. -- Best regards, Alexey Gorbachev.

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filter json cdrs params before pass it as post request
by devang nathwani 10 Feb '17

10 Feb '17

Hello, How can i filter many of the parameters from json cdrs before passing it as http request to my web server? Because the default json cdrs string is having hell lot of parameters and i am not using that many so if i get only my required params on the web server side and server side script will have less load to parse unnecessary params. So how to filter out not required params from json cdrs string before pass it as post request to web server.

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limit calls to user callув via group/xxx@domain
by Борисов, Дмитрий / Dmitriy Borisov 10 Feb '17

10 Feb '17

Hi, All! I don`t understood how can I limit number of concurrent calls to user, when I`m not calling them, but calling they call-group. More common question: how to limit calls to users when I`m calling him not by dialplan but by dialstring? Somthing like: {execute_on_ring='limit hash ${domain}-lines ${dialed_user} 1'}user/${dialed_user}%{dialed_domain} looking very bed, because it generate lost call on user`s phone. Any other ways? -- -- С уважением, Борисов Дмитрий Отдел облачных технологий 1С-Рарус Тел. +7 (495) 231-20-02 доб.: 15-94 -- with best regards, Dmitriy Borisov 1C-Rarus Cloud Services tel.: +7 (495) 231-20-02 add.: 15-94

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Video and Slides: SIP and WebRTC Load Balancing and High Availability for FreeSWITCH, FOSDEM
by Giovanni Maruzzelli 10 Feb '17

10 Feb '17

Hello, here you can find both video and slides of my presentation at FOSDEM 2017, Bruxelles: https://fosdem.org/2017/schedule/event/freeswitch/ Enjoy! -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT Cell : +39-347-2665618

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Dest./port unreachable for RTCP and first few RTP packages
by René Weiss 10 Feb '17

10 Feb '17

Hello, While looking at some traces in Wireshark I noticed some packages which generated an "Destination unreachable (Port unreachable)" ICMP answer from my server. These where: - All RTCP packages - At least in one case the first few RTP packages in a new call (6 packages over around 0.12s, after that RTP worked) So my questions are: - Should I be worried about these? - What's the deal with RTCP packages? Can/should I simply ignore them or is there an option in freeswitch that I should set to handle them? Thank you, René

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T38 488 behavior on 1.6.14
by andymurray439＠yahoo.com 10 Feb '17

10 Feb '17

Good evening, I noticed Bryan's thread on "BEHAVIOR CHANGE (YES PAY ATTENTION TO THIS IFYOU USE FAXING)" but still having issues. I'm running FreeSWITCH Version 1.6.14+git~20170130T165359Z~e9cb966d18~64bit (git e9cb966 2017-01-30 16:53:59Z 64bit) When a G711 call comes in and then the reinvite occurs from the calling side for T38; FS sends back a "488 Not Acceptable Here" I have set this in vars.xml per the suggestion: <X-PRE-PROCESS cmd="set" data="fax_enable_t38=true"/> Below is the pastebin of the siptrace: http://pastebin.com/hjnhEqym Below is the dialplan for it: <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="set" data="fax_enable_t38=true"/> <action application="rxfax" data="/tmp/FAX-${uuid}.tif"/> <action application="hangup"/> Am I missing anything else to just make this work? Thanks! andy....

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OpenSSL + Windows build
by Gregor Nanger 09 Feb '17

09 Feb '17

I am using FS with Windows. Due to bug or compatibility in OpenSSL, I get error audio Handshake failure 1 when I try to call from verto client in Chrome. Solution is to get newer OpenSSL dlls (ssleay32.dll and libeay.dll) and replace ones that come with FS installation. Since both dlls are created during Windows build, does anyone knows how to change FS windows build to download newest source files of OpenSLL and create newest version of OpenSSL dll. Best regards, Gregor

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strange behavior of ignore_early_media
by Brandon Johnson 09 Feb '17

09 Feb '17

My freeswitch application is using early media to send a video preview from the caller to the callee before answering. This works, if I am not forking the call to multiple users. If I fork the call, only the first endpoint to receive the early media gets to answer, and the other calls are cancelled (this is expected, according to the documentation). If I set ignore_early_media to true in the dialplan, then the forking issue goes away but the early media stops working. I did some packet captures with ignore_early_media turned on and off, and the signaling doesn’t change, so I’m sure it’s not a problem on the endpoints side of things. Freeswitch just doesn’t send the RTP when it is set to true. Is there some trick I am missing, or a workaround I can try? Here’s my dialplan, it’s very straightforward: <extension name="C4 All Group"> <condition field="destination_number" expression="^All$"> <action application="set" data="ignore_early_media=true"/> <action application="bridge" data="user/000FFF0B001A,user/000FFF803637"/> </condition> </extension>

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Fwd: Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
by Michael Nielsen 09 Feb '17

09 Feb '17

I'm trying to create a voicemail, but are getting error due to: EXECUTE sofia/internal/4412345678(a)sip.nexmo.com bridge({absolute_codec_string='PCMU'}user/4487654321(a)my-domain.com) 2017-02-09 14:44:33.887109 [DEBUG] switch_ivr_originate.c:2138 Parsing global variables 2017-02-09 14:44:33.887109 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f047c0262f0 Connected. 2017-02-09 14:44:33.887109 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f047c0262f0 released. 2017-02-09 14:44:33.887109 [DEBUG] switch_ivr_originate.c:2138 Parsing global variables 2017-02-09 14:44:33.887109 [NOTICE] switch_ivr_originate.c:2841 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2017-02-09 14:44:33.887109 [DEBUG] switch_ivr_originate.c:3829 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2017-02-09 14:44:33.887109 [NOTICE] switch_ivr_originate.c:2841 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2017-02-09 14:44:33.887109 [DEBUG] switch_ivr_originate.c:3829 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2017-02-09 14:44:33.887109 [INFO] mod_dptools.c:3409 Originate Failed. Cause: USER_NOT_REGISTERED My dial plan is as follows: <condition field="${user_exists}" expression="^true$" break="never"> <action application="set" data="call_timeout=20"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="{absolute_codec_string=' PCMU'}user/${dialed_number}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_number}"/> </condition> So I guess the user should not be registered, as it is for voicemail.

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Unable to show avatar/mute image in video confererence
by Muthupandi Govindharaj 09 Feb '17

09 Feb '17

I am using conference in FreeSWITCH. The moderator should be able to join with video+audio. Non-moderators who join the conference should have avatar/image(their video should not be shown in the conference).video-mode is video-mode is mux and video-layout-name is 1x1. I tried the following. But I tried the following. I am getting a blank screen. default-avatar.png is not shown in the video. I have confirmed that the image is available in the location. <action application="answer"/> <action application="set" data="video_mute_png=${images_dir}/default-avatar.png"/> <action application="conference" data="myconf@confname++flags{non_moderator|vmute}"/> Please help me. Regards, Muthu

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