December 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH docs/phrase files
by Brian West 02 Dec '16

02 Dec '16

FreeSWITCHers, I'd like to know if everyone can take a moment to review the phrase XML that you can https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/docs/phrase The purpose is we are in the process of making sure all the sound files are recorded and anything missing is being documented and ordered ASAP before the 1.8 release. So please help me out. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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Verto conference and Yealink T49 video
by Cj B 02 Dec '16

02 Dec '16

Hi all, I have verto working and am able to do video conferences with multiple people. But I was wondering what it takes to get a yealink t49 or other video capable phone to be able to join the conference and share video? I haven’t been able to find anything in documentation or the email list. I can do T49->T49 video right now, but when they join the conference there’s no video. Thanks! Cj B

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Force one way audio?
by Peter Steinbach 01 Dec '16

01 Dec '16

Hello, we want to setup a system where a larger numer of callers (500-1000) are listening to MOH. In order to reduce bandwidth, I would like to force Freeswitch to negociate only one way audio (outbound audio only, MOH). Is there a way to do this in Freeswitch, maybe? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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Re: [Freeswitch-users] Skype
by Russell Treleaven 01 Dec '16

01 Dec '16

How did this land? On Aug 4, 2016 1:35 PM, "Giovanni Maruzzelli" <gmaruzz(a)gmail.com> wrote: About skypopen: I suspect they (M$) will retract support soon for the protocol/API used by mod_skypopen. Don't seriously invest in it until situation will clarify, probably in Autumn... 6 concurrent call limit can be overridden in many ways, but I repeat, I would not counseil to seriously invest in it until Autumn. I'm mod_skypopen author, btw. If someone has more info, please step up and let us know. -giovanni On Thu, Aug 4, 2016 at 5:53 PM, Rick Jarvis <rick(a)magicmail.mooo.com> wrote: > Skype are telling me that it is no longer possible to receive calls to > Skype names / addresses (as appose to Skype numbers) via SIP with Skype > Connect. Does anyone know if this is the case? Their support is awful, but > this would seem a bit of a backward step, and I don’t know whether to trust > what I’m being told. > > The alternative is Skypopen, of course. But I notice that it has a maximum > of 6 concurrent calls. Is this for resource reasons? I’m thinking of > building a server to handle all Skype to SIP calls for our customers, so > would be interested to know if it’s possible to bypass the limit with a > beefy spec…? > > Thanks > R > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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FreeSWITCH Registrar TLS offload
by Vladyslav Zakhozhai 01 Dec '16

01 Dec '16

Hi, I'm trying to understand what is the best or suitable approach to the following use case. Let me simplify thing a little bit. Suppose we have one FreeSWITCH registrar behind SIP proxy (kamailio). I'd like to offload SSL/TLS encryption/decryption to SIP proxy: REGISTER: Request: UAC == SIP/TLS ==> Kamailio == UDP ==> FreeSWITCH:50 Reply: UAC <== SIP/TLS == Kamailio <== UDP == FreeSWITCH INVITE: UAC1 == SIP/TLS ==> Kamailio == UDP == > FreeSWITCH == UDP ==> Kamailio == SIP/TLS ==> UAC2 (FreeSWITCH uses kamailio as outbound proxy with fs_path tag appended in dialplan). The main problem is in Contact header which contains transport=tls and we can see it in FreeSWITCH console: User: user(a)domain.com Contact: "" <sip:user@UAC_IP:57976;transport=tls> Status: Registered(TLS)(unknown) EXP(2016-11-22 10:16:59) EXPSECS(108) IP: SIP_PROXY_IP Port: 5060 When FreeSWITCH sends INVITE to UAC2 (during call) it tries to establish TLS session to UAC2. It fails because there is no TLS-enabled sofia profiles in the config of FreeSWITCH. I have only one solution in my mind: rewrite transport tag in Contact header on SIP proxy (transport=udp to FreeSWITCH, and transport=tls to UAC). I'd like to know it this solution ok or there is more elegant solutions. I've tried appending tag transport=udp in FreeSWITCH's dialplan but no success. Thank you in advance. -- С уважением, Владислав Захожай

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mod_python InputCallback()
by Christoph Blau 01 Dec '16

01 Dec '16

Hi There I want to catch some DTMF and if possibly events from mod_callcenter using a Python script. For now DTMF would be a good start to get me going. Following the examples on the old and new Wiki brought no luck since it is just not executing my callback function I hope it is okay that I dumped dialplan, script and CLI output into on gist https://gist.github.com/cblaupanda/587a07b746f0e6427ceaf013576f29af Am I missing something? Could someone please point me in a direction that I get this working? Many Thanks, Christoph Blau

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