August 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FS AWS does not call between mobile phones connected to the same WiFi
by Jose David Jurado Alonso 07 Aug '17

07 Aug '17

Hi, I have an FS server running on an AWS instance and it does not call between mobiles connected to the same WiFi. The call is successful if each of the mobiles is connected to different Wifi or by mobile network (4G). Obviously FS is always in a different network and I use a domain/URL to routing the calls. I also do not see anything strange in the logs. I think that call does not occur because the server does not find the destination number as registered and hangup/abandoned the call by timeout. It always works except when the two mobiles are connected to the same wifi. 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"/>  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/> <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/> internal.xml: <param name="apply-nat-acl" value="rfc1918.auto"/> <param name="local-network-acl" value="rfc1918.auto"/> <param name="ext-rtp-ip" value="autonat:$${external_rtp_ip}"/> <param name="ext-sip-ip" value="autonat:$${external_sip_ip}"/> internal-ipv6.xml:    external.xml: <param name="local-network-acl" value="rfc1918.auto"/> <param name="ext-rtp-ip" value="autonat:$${external_rtp_ip}"/> <param name="ext-sip-ip" value="autonat:$${external_sip_ip}"/> external-ipv6.xml: <param name="local-network-acl" value="rfc1918.auto"/>    One call example is "sip:1009@mydomain.us:5080"

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World network tones in TGML
by Allan Kristensen 06 Aug '17

06 Aug '17

Hello, Does anyone have/need a complete database of network tones in TGML ? These: https://www.itu.int/dms_pub/itu-t/opb/sp/T-SP-E.180-2010-PDF-E.pdf I'm thinking a lot of WebRTC clients doesn't have any built in support for generating network tones and must have these served as early media? (at least the ones I've worked with ;-). We need it for different reasons, but before we start building it, I just wanted to check if it was already existing out there? Best Regards, Allan Kristensen

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DNS SRV failover
by Sergey Safarov 06 Aug '17

06 Aug '17

I configured two DNS SRV like "_sip._udp.example.com" and pointed to two FS servers. Than i originate call from third FS server and use "fs_path=sip:example.com" Looks like FS 1.6.18 not detect fail of one server and always routes to only one server of two in SRV records. Is this bug and need to report or i do some thing wrong?

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Httpapi and xml dialplan
by Gregor Nanger 06 Aug '17

06 Aug '17

Can someone please advice me. Would like to develop something with httpapi and need to know about concept. Can I mix httpapi and xml dialplan plan? For example, can I execute answer, call httpapi for some work and then proceed with xml dialplan or httpapi continuous loop for asking what's next? Thank you, Gregor -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si

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TURN Configuration under CGN (RFC6598)
by Nandy Dagondon 05 Aug '17

05 Aug '17

Hi folks, I just learned that our Internet provider implemented CGN when I'm about to experiment WebRTC/mod_verto. I already signed up a free TURN server account at http://numb.viagenie.ca/ to proceed. Where do I place turn: numb.viagenie.ca in the config files? Is this correct? sip_profiles/external.xml: <param name="ext-rtp-ip" value="turn:numb.viagenie.ca"/> <param name="ext-sip-ip" value="turn:numb.viagenie.ca"/> /Nandy

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SIM7100 LTE modem with USB audio
by Stanislav Sinyagin 05 Aug '17

05 Aug '17

Simcom has recently released a new 4G/LTE modem, and it has USB audio support. You can find sim7100_usb_audio_application_note_v0.01.pdf with details at the vendor site, or at techship.com after registration. It transmits 8kHZ, 16-bit PCM audio via a USB UART simulation. So, in theory, gsmopen module may be adapted to it (or maybe a new module is worth starting). I ordered a sample, will check it out soon. cheers, stanislav

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ASR with Google speech API
by Murugan Pandian 04 Aug '17

04 Aug '17

Hi, Is there any way i can use Google Speech API with freeswitch ASR Regard's Murugan Pandian

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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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