May 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Fail to adjust PTIME with OPUS
by Luciano Augusto Rezende 10 May '16

10 May '16

Hello, I have an environment with debian 8 and freeswitch 1.6.7 installed from the fs repositories. With this env my call were working fine, using opus codec. But recently I deployed a new env, with the same configurations from the previous one but with freeswitch's new release 1.6.8, and my calls are failing after few seconds when adjusting PTIME as the following log shows: [WARNING] switch_core_media.c:2567 [VBR]: Asynchronous PTIME supported, adjusting JB size. Remote PTIME changed from [20] to [40] [DEBUG] switch_core_media.c:2988 Changing Codec from opus@20ms@48000hz to opus@40ms@48000hz [DEBUG] mod_opus.c:661 Opus decoder stats: Frames[0] PLC[0] FEC[0] [DEBUG] mod_opus.c:661 Opus decoder stats: Frames[0] PLC[0] FEC[0] [WARNING] switch_core_codec.c:721 Codec opus Exists but not at the desired implementation. 48000hz 40ms 2ch [ERR] switch_core_media.c:3025 Can't load codec? [NOTICE] switch_core_media.c:3026 Hangup sofia/<profile>/<user>@<domain> [CS_EXCHANGE_MEDIA] [INCOMPATIBLE_DESTINATION] Sometimes it successfully adjust few times before breaking, or breaks on the first try like this one. It also happens with other PTIME combinations, like [40] to [80] or [20] to [60]. Is that an issue/bug? Maybe related to https://freeswitch.org/jira/browse/FS-9062 that came with the last release. Thanks Luciano Rezende

2 1

Verto - call hold
by Gregor Nanger 10 May '16

10 May '16

Can someone confirm me this behaviour regarding verto. I am using call waiting functionality, so user can have many active calls. One active, other are ringing, for example. Then I tried to play with call hold. If I put active call on hold and accept another call, put this one on hold and unhold previous call, sound is lost (mic is working, but speaker is not working). It looks that call hold/unhold is working only on last received call. Anyone else find this behaviour? Will try to reprogram our code so that each call will use its own video tag. Maybe this will solve my problem. Best regards, Gregor

2 2

Cutting out video
by Luca Pradovera 10 May '16

10 May '16

Hello, it's me again with my conference broadcasting application! It has been reported to me that performance is way better if the video flow from the server to the presenter (which does not actually watch the video) is turned off. That is accomplished by choosing a few call-time options on Jitsi, but I would like it to just be the default. Is there a way to effectively do not send the video stream to a conference participant? Best regards, Luca

2 3

Dynamic return call (callback)
by Gabriel Bustos 10 May '16

10 May '16

Hello, I'm trying to implement a callback (for specific numbers) in my project. What I need to do is hangup the incoming call and return to that number. What I tried is this: <action application="set" data="api_hangup_hook=originate user/${redirect_to} &bridge(gsmopen/ANY/${number})"/> Actually this works (sort of), it originates a call in softphone and when I answer the softphone the call to destination number is made. BUT what I need is to call to destination number and bridge to softphone when the call is answered. And here is where I'm having problems. My last attempt was calling a lua script who fires a custom event, like this: <action application="set" data="api_hangup_hook=${lua(websip/teste.lua ${caller_id_number})}"/> teste.lua: function fireEvent() local event = freeswitch.Event("CUSTOM", "Gateway::Callback") event:addHeader("My-Header", "test") event:addBody("EventTest") session:sendEvent(event) --event:fire(); end fireEvent() Ok, the event is fired as I can see in CLI but I think the event hook doesn't work. This is the line from lua.conf: <hook envent="CUSTOM" subclass="Gateway::Callback" script="testecallback.lua"/> Can you help me telling me if I'm the right path or if there's a better way to do that? Thanks in advance.

2 4

(no subject)
by amani mansour 10 May '16

10 May '16

Hi all, How we can configure FreeSWITCH to send DTMF for example :4 after(1 seconde) 8 after(1 second) 9 Best regards Amani

4 4

What you need to know about sponsoring ClueCon 2016!
by Kathleen King 10 May '16

10 May '16

[image: http://cluecon.com] <http://cluecon.com> <http://cluecon.com>

1 0

API break vs uuid_break
by Marek Slivanský - Livispace s.r.o. 10 May '16

10 May '16

Hi there, sorry if I'm not choosing the right way for asking for something, please point me out, this is my first time using the mailing list. I have a question regarding API command "break" has changed to "uuid_break". I really tried to but I've not found any information about that. We are using "bgapi break <uuid>" to stop playing some sound to a call (leg), e.g. moh, and to continue with next dialplan item. We are using v1.2.stable. Probably since last upgrade to the latest commit in this branch (prior we were using some 09/2012 commit), sometimes (rarely) the command "bgapi break <uuid>" gets ignored by FreeSwitch and sound remains playing forever (local_stream://moh). Could anyone please shed some light on this change (and perhaps add some info why break has been deprecated and consequences of continuing to use it, into wiki)? Thanks in advance for any answer, Marek

3 3

One way hold
by Oleg Stolyar 10 May '16

10 May '16

Hi, I am looking for a way to deaf a channel and play some hold music into it while still receiving audio from the channel. I tried uuid_displace but it also mutes the channel, so I don't receive any audio from it. Is there a combination of commands that can do this?

3 4

Bug: FreeSwitch change from TCP to UDP if he reseive a refer
by Alexander Haugg 10 May '16

10 May '16

Hi all, The scenario is: - The registered SIP Client make an outbound call - After a connect the client send a refer to a new destination (Blind Transfer) - The 202 accepted is correct as tcp sended - The Notify (sipfrag 200 OK) and the following Bye will send via UDP If I do the same with an inbound call, it works successfully. Is it a Bug? My Freeswitch version is: FreeSWITCH Version 1.7.0+git~20160502T014035Z~96e963d3eb~64bit (git 96e963d 2016-05-02 01:40:35Z 64bit) Mit freundlichen Grüßen Alexander Haugg C4B Com For Business AG Gabriele-Münter-Straße 3-5 D-82110 Germering Telefon +49 89 840798-170 Fax +49 89 840798-13170 alexander.haugg(a)c4b.de<mailto:alexander.haugg@c4b.de> www.c4b.de<http://www.c4b.de/> Registergericht: Amtsgericht München, HRB 125040 Vorstand: Bernd Woköck, Michael Schreier Aufsichtsratsvorsitzender: Dr. Helmut Maier

3 4

Verto, -32601 Invalid Method, Missing Method or Permission Denied
by Oivvio Polite 10 May '16

10 May '16

I'm getting closer to having Verto working. I now have the wss certs working, I can successfull authenticate against directory configuration fetched via xml_curl. Now after the client sends a "verto.invite" method to the server the server responds with a code -32601, "Invalid Method, Missing Method or Permission Denied". Since verto.js probably isn't outputting an invalid or missing method my assumptions is that for some reason the authenticated user does not have the requisite permissions to have the call move to the next step. I'm unsure what my next troubleshooting step should be. Regards Oivvio Full FreeSwitch log: http://termbin.com/1fap Full Chrome log: http://termbin.com/6igj

2 3