- FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH support for RFC 4733
by Muhammad Naseer Bhatti 26 Jun '17

26 Jun '17

Hi, like the subject says, do we have support for RFC 4733? Can’t seem to find anything references in the code so far. -- Sent with Airmail

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Information: German acoustical social network with fs
by Thomas Hoellriegel 26 Jun '17

26 Jun '17

Hi All, Fs is very nice to handle complex jobs. I have setup a german inclsionproject "dorf-telefonchat". This is a none commercial network. This is a virtual audioworld, with complex sounddesign, and ambiences. Facebook and other social networks are often not useable for handycap users, or users to having problems to get informations as barrierefree. The navigation is digits only (voicemenus). For examle: press 1 for originizer, 2, for messageboards 3 for mailbox on so on. I have setup sip, german landline, iax and skype, connections to call these system. Thanks. --------------- Website: www.dorf-telefonchat.de

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Freeswitch API access
by Deepika Yadav 26 Jun '17

26 Jun '17

Hi, I have a scenario where multiple conferences are established among users. This is achieved through python programs over ESL. For each conference, a new thread is created. So, should I also use locks before firing originate calls APIs? Regards Deepika https://www.iiitd.edu.in/~deepikay/

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SKYPE: mod_skypopen END-OF-LIFE
by Giovanni Maruzzelli 26 Jun '17

26 Jun '17

Hello FreeSWITCHers, Microsoft decided to kill all old Skype clients July the 1st, and new clients do not support APIs to interact with calls. So, no way mod_skypopen can continue to work. Has been a nice trip, a lot has changed in the Real Time Communication world since first chan_skypiax, then mod_skypopen bridged Skype and open standard calls. mod_skypopen sez: "GOODBYE WORLD!" -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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Verto configure with domain name
by Ketan Kothari 26 Jun '17

26 Jun '17

Hello Guys, I setup verto communicator on our server but now we want to run it on our domain name also can anyone help to remove its default port 9001? Please find below image of logs of verto http://prntscr.com/fo7o1c Please help me on this.

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Re: [Freeswitch-users] Refer with replaces - how to configure to support
by Magnus Kelly 25 Jun '17

25 Jun '17

Hello All, Further to prior question on support for Refer with replaces configuration question - the ladder diagram below is the desired sequence - currently fails at line 45 This is one of the call cases required to interop with MS Skype4Biz of which I am curious if this can be handled as it seems the Refer C target is not revealed until after the "parking" Invite is sent Example Refer is below ------------------------------------------------------------------------ REFER sip:gw+s4btrunk@10.113.3.4:5066;transport=udp;gw=s4btrunk SIP/2.0 FROM: <sip:+4419xxxxxxxx@10.113.8.32:5068 ;transport=tcp>;epid=ED60F08DC5;tag=6acfb33460 TO: <sip:+4414xxxxxxxx;transport=udp@193.113.106.226>;tag=r3cp3tF36UmZg CSEQ: 2 REFER CALL-ID: 78dff094-ca55-1235-8c91-000c29a74006 MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 10.113.8.32:5068;branch=z9hG4bK433d335f CONTACT: <sip:skype4biz.xxxxxxxxx.co.uk:5068 ;transport=Tcp;maddr=10.113.8.32;ms-opaque=1befc8e31af6a72f> CONTENT-LENGTH: 0 REFER-TO: <sip:skype4biz.xxxxxxxxx.co.uk:5068 ;transport=Tcp;maddr=10.113.8.32;ms-opaque=1befc8e31af6a72f?REPLACES=d71cf52b-2961-4b3c-b823-73191c204d39%3Bfrom-tag%3D58af409ce4%3Bto-tag%3Dd6814c6feb> USER-AGENT: RTCC/6.0.0.0 MediationServer ------------------------------------------------------------------------ 1PSTN 2 S4B/Mediation Server PSTN/FS1 -IN 2 | | | 3 | | (1) INVITE | 4 | |<----------------------------------------------| 5 | | | 6 | | (2) 100 Trying | 7 | |---------------------------------------------->| 8 | | | 9 | | (3) 183 Session Progress (Dialog 1) | 10 | |---------------------------------------------->| 11 | | | 12 | | (4) PRACK | 13 | |<----------------------------------------------| 14 | | | 15 | | (5) 200 OK (PRACK) | 16 | |---------------------------------------------->| 17 | | | 18 | | Early Media | 19 | |==============================================>| 20 | | | 21 | | (6) 200 OK (Dialog 1) | 22 | |---------------------------------------------->| 23 | | | 24 | | (7) ACK | 25 | |<----------------------------------------------| 26 | | | 27 | | Bidirectional RTP | 28 | |<=============================================>| 29 | | | 30 | | (8) INVITE(Hold) | 31 | |---------------------------------------------->| 32 | | | 33 | | (9) 200 OK (hold) | 34 | |<----------------------------------------------| 35 | | | 36 | | (10) ACK | 37 | |---------------------------------------------->| 38 | | | 39 | | (11) REFER | 40 | |---------------------------------------------->| 41 | | | 42 | | (12) 202 Accepted (OK till here) | 43 | |<----------------------------------------------| 44 | | | 45 | | (13) INVITE(Replaces, Dialog 2) (Not sent)| 46 | |<----------------------------------------------| 47 | | | 48 | | (14) 183 Session Progress(Dialog 2) | 49 | |---------------------------------------------->| 50 | | | 51 | | (15) NOTIFY (Trying) | 52 | |<----------------------------------------------| 53 | | | 54 | | (16) 200 OK (NOTIFY) | 55 | |---------------------------------------------->| 56 | | | 57 | (17) INVITE(Dialog 3) | | 58 |<---------------------------------| | 59 | | | 60 | (18) 183 Session Progress | | 61 |--------------------------------->| | 62 | | | 63 | | (19) 183 Session Progress(Dialog 2) | 64 | |---------------------------------------------->| 65 | | | 66 | | (20) NOTIFY (Trying) | 67 | |<----------------------------------------------| 68 | | | 69 | | (21) 200 OK (NOTIFY) | 70 | |---------------------------------------------->| 71 | | | 72 | (22) 200 OK(Dialog 3) | | 73 |--------------------------------->| | 74 | | | 75 | | (23) 200 OK(Dialog 2) | 76 | |---------------------------------------------->| 77 | | | 78 | | (24) NOTIFY | 79 | |<----------------------------------------------| 80 | | | 81 | | (25) 200 OK (NOTIFY) | 82 | |---------------------------------------------->| 83 | | | 84 | | (26) BYE(Dialog 1) | 85 | |<----------------------------------------------| 86 | | | 87 | | (27) 200 OK (BYE, Dialog 1) | 88 | |---------------------------------------------->| On 25 June 2017 at 17:14, Magnus Kelly <magnus.kelly(a)gmail.com> wrote: > Hello all, > > I am trying to configure the FS dial plan to fully support "Refer with > Replaces" ? > > Call case is FS establishes the call successfully with an upstream server, > but when the user behind the upstream server performs a transfer, a Refer > with replaces is generated and sent to FS, FS accepts the request, but post > the 202 the expected invite (FS to upstream) in support of replaces does > not occur and then the transfer fails. > > My assumption is I am missing the correct configuration for Refer with > Replaces, I have spent some time searching web but not yet found the > correct way to handle this. My assumption is FS fully supports "Refer with > Replaces" but confirmation and any tips on configuration most welcome. > > Appreciate any insight. > > Thanks > Magnus > > >

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Javascript FileIO
by Faruq Ahmad 25 Jun '17

25 Jun '17

Hi, I have a FS dialplan that fetches callflow from a JSON file. File is read using var JsonFd = new FileIO(JsonPath, 'r'); JsonFd.read(2048); result = JsonFd.data(); var menu = JSON.parse(result), Is there anyway I can read the whole file in one attempt, i.e. when a smaller size was given to the .read() function it wouldn't read the complete file. I have increased the value of size for read function however my concern is in the long run file size might increase and parse would get an incomplete JSON. Is there anyway I can get filesize from FileIO object or detect EOF from the read buffer to make sure the whole file is loaded? Also if I increase the size for read fucntion buffer way over the estimated filesizes, is there any guarantee that no garbage values will be read from the disk after the EOF?

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postgresql ssl
by Madovsky 25 Jun '17

25 Jun '17

Hi folks, is there any special syntax to connect postgresql in ssl mode from pgsql://? thanks

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att_xfer or other method for mod_callcenter queue to queue transfer
by Jaco Jacobs 24 Jun '17

24 Jun '17

I've been using mod_callcenter in what I can imagine is not the originally intended use case. It's worked so far but now I've run into a problem and being new to FS I've gotten stuck. I originate a call to an extension on another voice system (non FS), and then use Mod_callcenter uuid_standby to keep the agent ready in the queue. I then make an outbound call and put the call into the queue ( like a dialer or click to dial type setup). This all works great, the problem comes when I want to transfer. I've tried uuid-transfer, tried to park the call etc., as soon as I try this the outbound call drops. The closest I've gotten is to use att_xfer directly to an extension on the other non FS voice system, this works fine. The problem is if I want to transfer to another queue, which is a function I definitely want. Any assistance would be greatly appreciated.

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Question: Skype 4.3 support endet! alternate to Fs?
by Thomas Hoellriegel 24 Jun '17

24 Jun '17

Hi all, Microsoft remove all older skypeclients on 07.01.2017. The problem: skype is not working after this date. mod_skypopen support only skype 4.3.37. I don't found a manual to use fs on latest skype versions. Do your can help please? Do your have a way after on Jul 2017? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc

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