August 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

create user
by mrjoli021 04 Aug '17

04 Aug '17

Hello, Is there a way to create a user from the FS API so that we can automate user creation we will need to create an extension with the following extention number Password outbound callerid We cant put it on an xml cause freeswitch is running on a database. thanks, -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/create-user-tp7596342.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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Verto Communicator - White-labeling options
by Jonathan Hunter 04 Aug '17

04 Aug '17

Hi Guys, I am working with mod_verto and developing our own client, however I really like Verto Communicator. Are there white-labeling options available with this client? For example in the top left hand corner can the Verto Communicator text be changed and images added to the tool bar? I know we can put images/labeling within the conferencing configuration but just wondered what can be achieved with the Verto Communicator client if anything? (I am also clear on the url changes to define own domain its more around the client that Im interested in). Many thanks Jon

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payload type mismatch with bridge command
by John Narduchi 03 Aug '17

03 Aug '17

Hello, question about a 2833 DTMF situation I'm running into. We run a IVR in node.js using the outbound event socket. The node IVR answers the call, captures some digits from caller and then bridges with bypass_media=true the b leg out to a conferencing server, taking freeswitch out of the media path and all works well except.. for one scenario. When we receive telephone-events/8000 on a attribute other than 101 we end up with a payload type mismatch on a and b leg. With the bridge command the b leg always seems to offer what is specified in the rfc2833-pt setting of corresponding sip_profile. My question is, is there a way to configure FS to offer telephone-event on the same attribute in the b leg that came in on initial invite of a leg with bridge command on a answered call? Or possibly renegotiate the payload type if needed at time of bridge command? If so any examples for that. Sorry my SIP knowledge is not the best so please forgive if i am missing something here. Configs are vanilla and tested on latest FS version 1.6.19. Thanks!

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Click2Call
by Igor Olhovskiy 03 Aug '17

03 Aug '17

Hi! Is there any programs (or browser extensions) exists to support click2call for freeswitch (via Event Socket)? I see many of those for Asterisk, but couldn’t find any for FS. -- Best regards, Igor

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Ring Groups
by Tilo Schmidt 03 Aug '17

03 Aug '17

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question about session:setInputCallback
by Abaci B 03 Aug '17

03 Aug '17

Hi, >From what I understand based on the documentation at https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaA… you can pass an arg to the callback function in the second argument of session:setInputCallback, however in my callback if I have function my_cb(s, type, obj, arg) the value of arg is always nil, does anybody have this working or can confirm if the documentation is correct? Thanks

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Re: [Freeswitch-users] FS AWS does not call between mobile phones connected to the same WiFi
by Robert Mundkowsky 02 Aug '17

02 Aug '17

What does SIP headers look like from the two? -------- Original message -------- From: Jose David Jurado Alonso <josedavid(a)zennio.com> Date: 7/31/17 2:36 AM (GMT-05:00) To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] FS AWS does not call between mobile phones connected to the same WiFi Hi Giovanni, I have tried with the change you mention, changing the domain by the public IP, but it still does not work. Both devices are connected to the same WiFi, on a TP-Link Archer C1200 router, and in the ALG section I do not see the SIP option. By connecting one of the devices to this router and the other to another network, the call is successful. The problem is when the two are connected to the same network (I have tested with several different networks). It's as if FS does not find the destination number and stays waiting. Both devices are correctly registered and can be called or called by any device as long as it is not on the same network. 2017-07-28 18:05 GMT+02:00 Giovanni Maruzzelli <gmaruzz(a)gmail.com>: On 28 July 2017 at 20:21, Jose David Jurado Alonso <josedavid(a)zennio.com> wrote: My current configuration is the next: vars.xml: <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=host:mydomain.us"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=host:mydomain.us"/> try making it: <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=135.44.55.123"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=135.44.55.123"/> Stop and start FreeSeWITCH and see if it helps Also, maybe something is blocked by some ALG in the clients' router? Try to connect via TLS... Hope this help, -giovanni _________________________________________________________________________ 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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Debian 8 installation
by Tom Lynn 02 Aug '17

02 Aug '17

I'm encountering a bit of catch 22 in as much as Freeswitch won't yet build on Debian 9, but the Debian folks seem to have pulled Debian 8. Does anybody know where installation media for 8 may be had while we're waiting on FS1.8? Thank you. Tom

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mod_verto and Verto communicator URL creation and invitation
by Jonathan Hunter 02 Aug '17

02 Aug '17

Hi Guys, I am playing with mod_verto and Verto communicator as I am looking for a jitsi meet replacement, as I need moderator commands. Its not clear from the documentation, and maybe I am missing something and need to develop something of my own, but is it possible to create unique URL's for each conference created (I assume so) and then send the URL as an invitation to a user, so they just need to click on it to access the conference without the need for username/password? Thanks! Jon

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DTMF filter (RFC2833)
by Mickael Hubert 02 Aug '17

02 Aug '17

Hi list, I'm looking for a way to filter ingress DTMF on A leg. I receive "bad" DTMFs from SIP provider (Signal A,B,C, etc ...) I want to filter only these DTMF, not all others. I receive only RFC2833 DTMF. Ex: Sip Provider -- (DTMF 1,A,2,C on A leg) --> FS --> (DTMF 1,2 on B leg) --> Do you have an idea for me please ? Thanks in adavance FS version: FreeSWITCH Version 1.3.13b+git~20130213T200819Z~cd7d1efc2a (git cd7d1ef 2013-02-13 20:08:19Z) -- Mickaël HUBERT Responsable du service VoIP Tel: (+33) 03.26.79.30.05

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