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Lua ASR grammar make issue
by Murugan Pandian 04 Aug '17

04 Aug '17

Hi, I try to use ASR in my IVR, Using this Example https://freeswitch.org/confluence/display/FREESWITCH/Lua+ASR+TTS+Directory+… but when i edit my directory.sent file and type make inside grammar folder it shows make: *** No targets specified and no makefile found. Stop. Regard's Murugan Pandian

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1.6 compilation from source on Centos 7
by Wesley Akio 04 Aug '17

04 Aug '17

Hi all, I know confluence says the recommended way is to install from RPMs on Centos 7 but... I do need mod_khomp and they don't have packages so here I am. I followed the step by step on https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 but got the following issue, I'm assuming this could be a version issue but I'm way out of my depth here... [root@fs1 freeswitch]# cpp --version cpp (GCC) 4.8.5 20150623 (Red Hat 4.8.5-11) ... [root@fs1 freeswitch]# make ... make[9]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' LTCOMPILE tport.lo LTCOMPILE tport_logging.lo LTCOMPILE tport_stub_sigcomp.lo LTCOMPILE tport_type_udp.lo LTCOMPILE tport_type_tcp.lo LTCOMPILE tport_type_sctp.lo *tport_type_sctp.c:* In function '*tport_sctp_init_socket*': *tport_type_sctp.c:206:10:* *error: *variable '*initmsg*' has initializer but incomplete type struct sctp_initmsg initmsg = { 0 }; * ^* *tport_type_sctp.c:206:10:* *error: *excess elements in struct initializer [-Werror] *tport_type_sctp.c:206:10:* *error: *(near initialization for '*initmsg*') [-Werror] *tport_type_sctp.c:206:23:* *error: *storage size of '*initmsg*' isn't known struct sctp_initmsg initmsg = { 0 }; * ^* *tport_type_sctp.c:211:36:* *error: *'*SCTP_INITMSG*' undeclared (first use in this function) if (setsockopt(socket, SOL_SCTP, SCTP_INITMSG, &initmsg, sizeof initmsg) < 0) * ^* *tport_type_sctp.c:211:36:* *note: *each undeclared identifier is reported only once for each function it appears in *tport_type_sctp.c:206:23:* *error: *unused variable '*initmsg*' [-Werror=unused-variable] struct sctp_initmsg initmsg = { 0 }; * ^* Thank you all, Wes -- <http://www.tuntscorp.com/> *Wesley Akio* Travessa da Lapa 96 - Sala 24 Centro – Curitiba – 80010-190 : +55 (41) 99950-0033 : +1 (305) 842-7095 : wesley_akio_

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Freeswitch and VoiceModems
by Enrico Mioso 04 Aug '17

04 Aug '17

Hello guys. First of all - hello to you all nice community. I am sorry i started a new thread here, instead of joining the original one at http://lists.freeswitch.org/pipermail/freeswitch-users/2017-August/127153.h… I coincidentally started thinking about this thing some days ago. i enjoyed a lot using FreeSwitch with some Huawei Dongles, and i can say it worked with Huawei E173, Huawei E3531 and maybe others. There where some problems, like DTMF not working properly with all dongles, but voice could be heard. But the most important problem, to me at least, was instability. After some time of operation, the modem did ring but FS didn't notice, or outgoing call did get stuck in some strange way. I decided to let things go at that time, but now something new happened: ModemManager project: https://www.freedesktop.org/wiki/Software/ModemManager/ operated by very nice and cool people, implemented support for voice in modems supporting it. I didn't look deep enough, but I think it's handling only initializzation and preparation tasks in general. I tought it would be cool to let FreeSwitch use ModemManager, for various reasons. 1 - They did all the work of preparing udev rules for different hardware, and they ship them in an elegantly packaged and proper way. 2 - They support a lot of different modems, with related firmware kludges and workarounds. 3 - Most importantly, they offer unified APIs to control the modem they support, over a DBUS interface. I think this would bring FS support for voicemodems to the next level, for various reasons. The first and most important one being robustness: since ModemManager handles tricky devices from long time now, and has undergone a good development and bugfixing time by now. The second thing to consider is the fact that it will allow us to use our modems in a more advanced way. Some modems offer native-level functionalities like SMS handling and other things via protocols like QMI: and using those functionalities this way could be very interesting, especially considering the buggyness some firmwares can exhibit when the modem is interfaced with situations that Windows drivers / Vendor software did not think about. I a very new to FS, both from a user perspective, and even more from a developer one: so i started a cursory look at what's happening in the FS eventing system. I didn't read further. Still, I was thinking about the following steps: 1 - Implementing a DBUS->FS gateway module, allowing for modules inside FS to listen and react fo dbus events. this would be necessary to allow us to talk to ModemManager, which talks via dbus. I don't know if this already exist: but I think it could be useful even for other situations. 2 - Isolating audio processing in one module. i noticed this is already in progress, good job guys! And thanks. 3 - Reimplementing something like mod_gsmopen, that talks via DBUS with ModemManager and interfaces with FS, allowing for call setup/teardown, SMS handling and so on. I know I know: ModemManager is an external process and is a piece of complex software. But following a little bit the development via the repository and playing with different modems, I can tell you there are good reasons for this. And companies building those hardware are not known for standardizzing things so much: and the amount of work to be done isn't small by no mean. I think we should try to do it only once if possible, with incremental develpment / fixes. FS is known for it's reliability and robustness: and building something reliable and robust is fundamental I think. I couldn't convince myself about 3G/4G modems firmwares being robust pieces of software, also due to their complexity. I think we should join our effort. Enrico

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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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