November 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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TLS configuration
by Agustí Ubalde 30 Nov '16

30 Nov '16

Hi all, I've an Asterisk Server with TLS enabled (wildcard certificate). I'm trying to connect from FreeSWITCH but I don't know how. I need specified CA file? Thanks, PRESENCE TECHNOLOGY An ENGHOUSE INTERACTIVE Company Agustí Ubalde Bellot Chief Developer C/ Comte Urgell 240 3º-A Barcelona 08036 aubalde(a)presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com <http://www.presenceco.com/>*

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Problem with CALL PICKUP (intercept) in QUEUES (mod_callcenter)
by Ewgeny 29 Nov '16

29 Nov '16

Hi ! First about terminology: *Call Pickup*the ability to pull a ringing call to the phone you are currently on. Call Pickup = Call Intercept = Call group pickup. We're using FreeSWITCH version: 1.6.8~64bit in our complex telephony system with Kamailio and other SIP services. The problem with call pickup (intercept) when using a Queue (mod_callcenter). The group pickup scheme we are using described here: http://www.tech-invite.com/fo-sip/tinv-fo-sip-service-16.html. We do SUBSCRIBE (3 see link) to some our service - that return information about the call (call legs) in NOTIFY with XML body (5). Then we do an INVITE with REPLACES (7) that actually do the Intercept. This works on normal calls, but doesn't work with Queues. In the queue a few agents simultaneously calling, and when INVITE w Replace headers comes it didn't intercept the call. The question is: how to implement Call Group Pickup (Call Intercept) with mod_callcenter ? If it necessary i can add call sip traces for more details. Thanks in advance for any help. Regards Ewgeny.

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freeswitch logs A-leg and B-leg json_cdr in different directories
by Aqs Younas 29 Nov '16

29 Nov '16

Hi, I am using mod_json_cdr to log files in a directory. I am setting *json_cdr_base* in my javascript to dynamically set log directory. I have observed freeswitch is writing only A-leg files in directory set by *json_cdr_base *and whereas B-leg files always goes to *log-dir *set in json_cdr.conf.xml If I do not set *json_cdr_base in *script both A-leg and B-leg goes to the same directory set by *log-dir *in json_cdr.conf.xml I need to set directory dynamically for json_cdr that is why I am using *json_cdr_base*. I tried to use log-dir like this but freeswitch is not evaluating the expression. Instead, it is treating this as a string. param name="log-dir" value="/tmp/${strftime(%Y-%m-%d)}"/> I need to set directory dyonically having both A-leg and B-leg json_cdrs. Any suggestion is much appreciated. Best Regards.

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absolute_codec_string not working
by Lợi Đặng 29 Nov '16

29 Nov '16

Hi List, I got some trouble with using `absolute_codec_string` param. My call scenario is pretty simple: caller <--> FS <--> callee. My caller compose `m=audio 7078 RTP/AVP 8 0 101` in its INVITE, and I'm doing `<action application="bridge" data="{absolute_codec_string=PCMU,GSM}sofia/gateway/callee/$1"/>` in the dialplan. But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the callee. not sure what I'm missing, helps would be appreciated. Note that when I'm using `originate` application in fs_cli, things are good. `originate {absolute_codec_string=PCMU}sofia/gateway/caller/100 &bridge({absolute_codec_string=PCMA}sofia/gateway/callee/100`. I have FS with proper behavior in transcoding, caller has `m=audio 31184 RTP/AVP 0 101` received, and callee has `m=audio 21922 RTP/AVP 8 101` received. rgds, Loi Dang Thanh Phone : 84.1224.735.448 Email : loi.dangthanh(a)gmail.com

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Best Way To Chat / Presence / File Xfer
by Adrian Thompson 29 Nov '16

29 Nov '16

Hello, Been a user and fan of Freeswitch for two years now - thank you for the great software! I've done days of research and would like to reach out to see if anyone else has some ideas here: We are looking into giving all of our techs a softphone for their mobile device so they can communicate back to base and to each other while out in the field. Our desired features are as follows: - Video (for troubleshooting equipment with other remote techs) - Chat (for quick updates) - Group Chat (for group reminders) - Page (reach out to a busy individual with an important message) - Group Page (reach out to a busy group of workers with an important message) - File Transfer (send each other marketing material and tech manuals on the fly) - Presence (so employees can let everyone else know if they are available) Video - Check, Working Chat - Check, SIMPLE, Working Group Chat - Not an option with SIMPLE?, Not working Page - Check, Working Group Page - Check, Working File Transfer - Not an option with SIMPLE?, Not working Presence - I have BLF working great, but what about extended presence such as 'busy' or 'I'm a leprechaun' Do I have to start using XMPP for these things? It could complicate the setup three-fold if I have to go that route and I would rather not.. But if I have to, is there documents supporting a configuration that includes Freeswitch for voice/video and XMPP for everything else? Also, does anyone have a suggestion for a mobile softphone with video, chat, file xfer and can be provisioned server side? (I've tried the Grandstream Wave, but it cannot use an HTTPS server with authentication for some reason so I'm looking for more options) Thanks in advance for advice! Adrian

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