December 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Ubuntu 14.x and 16.x Packages
by Brian West 13 Dec '16

13 Dec '16

FreeSWITCHers, We have published both Ubuntu repositories: https://freeswitch.org/confluence/display/FREESWITCH/Ubuntu+16.04+Xenial https://freeswitch.org/confluence/display/FREESWITCH/Ubuntu+14.04+Trusty Please review and give us feedback. Also if you're new check out my signature for a discount code for the ebooks, This code applies to FreeSWITCH Cookbook And the new Mastering FreeSWITCH book. If you've not done so check them out, The entire team put in a lot of work writing and proofing the books. It will help you get started with FreeSWITCH faster. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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Re: [Freeswitch-users] SBC
by Valter Nogueira 13 Dec '16

13 Dec '16

Its a little hard to change the mind from asterisk to fs. I will try it tomorrow when users are away and the mice can play. Thanks Em 10 de dez de 2016 11:11 AM, "David Villasmil" < david.villasmil.work(a)gmail.com> escreveu: > Hello, > > You don't HAVE to, but it's good practice. > Configure your provider anywhere you want, internal or external. The names > are not relevant, really. They are just names indicating that the internal > profile will authenticate registers/call (auth_calls parameter) whereas by > default external doesn't, though you can auth via acl.xml.conf. > Once registered with your provider, you just send the call out with, i.e.: > > {origination_caller_id_number=INCOMING_CALLER_ID}sofia/gatew > ay/your-provider/$1 > > hope this helps > > David > > On Sat, Dec 10, 2016 at 1:39 PM Valter Nogueira <valter(a)fastway.com.br> > wrote: > >> Yes, I want to use FS as a registar. >> >> I already have this working in an asterisk server and want to test FS as >> a replacement. >> >> 18 servers share a single sip provider and FS will act as a gateway. >> >> I have seen example.com.xml which is a user and have a gateway inside its >> definition. >> >> What I don't get is why it is in external profile. >> >> By the way, if I won't register in a sip provider, should I declare it as >> a gateway? >> >> Thanks >> >> Em 9 de dez de 2016 20:47, "Vladyslav Zakhozhai" <v.zakhozhai(a)gmail.com> >> escreveu: >> >> Hi Valter, >> >> You are right that you need to use directory rather than sip_profiles. >> Gateways in sip profiles are gateways which freeswitch is registered >> against. >> >> By the way do I correctly understand you that you want Asterisk servers >> to register against freeswitch, i.e. freeswitch is registrar? And also you >> always can see exact problem in freeswitch console and logs. >> >> >> 2016-12-09 20:01 GMT+02:00 Valter Nogueira <valter(a)fastway.com.br>: >> >> I would like to set up FS to run as a small SBC. >> >> Profile "internal" (nic em1) would be source for N asterisks that will >> register in FS. Some Asterisks should be trusted by IP (acl) >> >> Profile "external" (nic em2) would connect in an SIP Provider. >> >> So call coming from internal would routed to external gateway and >> vice-versa in dialplan. >> >> To mimic external profile I added in internal >> >> <gateways> >> <X-PRE-PROCESS cmd="include" data="internal/*.xml"/> >> </gateways> >> >> Asterisks are unable to register. I guess that the problem is internal >> gateway should be in directory instead of in sip_profile. >> >> Thanks, >> >> Valter >> >> _________________________________________________________________________ >> 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 >> >> >> >> >> -- >> С уважением, >> Владислав Захожай >> >> >> _________________________________________________________________________ >> 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 >> >> _________________________________________________________________________ >> 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 > > ᐧ > > _________________________________________________________________________ > 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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mod_hiredis documentation
by Stanislav Sinyagin 12 Dec '16

12 Dec '16

mod_hiredis was not properly documented, and I updated the page to as much as I could dig from the source: https://freeswitch.org/confluence/display/FREESWITCH/mod_hiredis Please review and suggest additions if needed.

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High Availability & Verto Communicator
by Zach Rothy 12 Dec '16

12 Dec '16

Good Morning, A project I am working on is using Freeswitch and Verto for video conferencing. In a previous post to the mailing list I saw someone mention the high availability capability Freeswitch has and thought it would be of use. While reading through the instructions I see it only mentions sip profiles only, and to my understanding Verto is its own signaling protocol separate from the sip/sip profiles on Freeswitch. So I tried to do some research on this, but didn't seem to get any results. I was wondering if the High Availability option also would work properly with Verto. If it does work, does it more or less follow the same set up instructions as it does for SIP? Thanks in advance, Zach R

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detect dtmf tones in early media state
by david escartin 12 Dec '16

12 Dec '16

hello how are you? do you know if there is anyway to detect inbound dtmf tones in early media by doing a session preAnswer, not connecting the call? thanks a lot and regards david

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Freeswitch 1.6 with Nuance Vocalizer 6.2.3 - "Client connection refused"
by Anonim Stefan 12 Dec '16

12 Dec '16

Hi, I unsuccessfully tried to do TTS on Freeswitch using mod_unimrcp (with specific nuance profile), by connecting to nuance-server. I started the Nuance server in mrcp mode: [0] For MRCP v1 I get [1]. For MRCP v2 (using TCP) I get [2]. For nuance nvscmdline I get [3]. The FS unimrcp profiles I used are [4] for v1 and [5] for v2. Am I missing something here?! Have anyone tried this lately? Is Freeswitch 1.6 mod_unimrcp working with latest Nuance? Thank you, Stefan [0] http://pastebin.com/qXixTb5y [1] http://pastebin.com/dSxWNvX5 [2] http://pastebin.com/CPdRcpBY [3] http://pastebin.com/9D5kHuwc [4] http://pastebin.com/Q39PX74Q [5] http://pastebin.com/t2iAr6H1

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Gateway configuration
by jacek burghardt 12 Dec '16

12 Dec '16

I had installed freeswitch on centos 7 and I also installed fusionpbx. How I can get gateway to register. I configured my voip.ms account and it shows stoped and I cant get it started. How I can make sure fusionpbx and freeswitch are working together.

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Re- mod_pocketsphinx working example
by Andrew Keil 12 Dec '16

12 Dec '16

To FreeSWITCH users, I am about to experiment with Speech Recognition inside FreeSWITCH and seem to have hit a wall. After building FreeSWITCH 1.6.13 with asr_tts/mod_pocketsphinx uncommented successfully. Then uncommenting from autoload_configs/modules.conf.xml <load module="mod_pocketsphinx/>. Then trying to run the pizza demo from the old Wiki: https://wiki.freeswitch.org/wiki/Mod_pocketsphinx I get the following: ... EXECUTE sofia/internal/1000@192.168.15.50:5080 detect_speech(pocketsphinx pizza_order undefined) 2016-12-12 14:07:16.345674 [DEBUG] switch_core_media_bug.c:828 Attaching BUG to sofia/internal/1000@192.168.15.50:5080 2016-12-12 14:07:16.345674 [WARNING] mod_pocketsphinx.c:172 Can't open speech model /usr/local/freeswitch/grammar/model/wsj1. 2016-12-12 14:07:16.345674 [DEBUG] switch_ivr_async.c:4800 Error loading Grammar 2016-12-12 14:07:16.345674 [INFO] mod_pocketsphinx.c:257 Port Closed. 2016-12-12 14:07:16.345674 [DEBUG] switch_core_media_bug.c:1190 Removing BUG from sofia/internal/1000@192.168.15.50:5080 2016-12-12 14:07:16.366177 [CRIT] SpeechTools.jm:213 ----ERROR: ReferenceError: XML is not defined EXECUTE sofia/internal/1000@192.168.15.50:5080 detect_speech(resume) 2016-12-12 14:07:16.865609 [DEBUG] switch_core_file.c:334 File /usr/local/freeswitch/sounds/en/us/callie/pizza/GP-DeliveryorTakeout.wav sample rate 8000 doesn't match requested rate 16000 2016-12-12 14:07:16.865609 [DEBUG] switch_ivr_play_say.c:1468 Codec Activated L16@16000hz 1 channels 20ms 2016-12-12 14:07:19.585569 [DEBUG] switch_ivr_play_say.c:1911 done playing file /usr/local/freeswitch/sounds/en/us/callie/pizza/GP-DeliveryorTakeout.wav EXECUTE sofia/internal/1000@192.168.15.50:5080 detect_speech(resume) 2016-12-12 14:07:21.245871 [NOTICE] sofia.c:1011 Hangup sofia/internal/1000@192.168.15.50:5080 [CS_EXECUTE] [NORMAL_CLEARING] 2016-12-12 14:07:21.245871 [ERR] SpeechTools.jm:472 Exception: Session is not active! (near: " rv = this.asr.session.collectInput(this.asr.onInput, this.asr, 1000);") ... I noticed that there is no grammar model "wsj1" inside /usr/local/freeswitch/grammar/model Also I cannot find wsj1 anywhere from the src or freeswitch files online. Does anyone have a tar.gz version of this? So at least I can get past this issue. Then I decided to follow the Lua directory example: https://freeswitch.org/confluence/display/FREESWITCH/Lua+ASR+TTS+Directory+… About half way down it states: ========================================== Next we create our dictionary and language model file. Change your directory to /usr/local/freeswitch/grammar and create a file called directory.sent We are using Matthew Williams and Joe Smith as our examples. <s> MATTHEW_WILLIAMS </s> <s> MATT_WILLIAMS </s> <s> JOSEPH_SMITH </s> <s> JOE_SMITH </s> Type make in the /usr/local/freeswitch/grammar directory. This should give you the following files. ls -la directory -rw-r--r-- 1 root root 82 2008-08-13 10:59 directory.dic -rw-r--r-- 1 root root 727 2008-08-13 10:59 directory.lm This does not work since the "make" does not work here! Any ideas? So far not much luck. I understand that pocketsphinx is not really a production product, however my client would like to make use of it for internal testing only. Any assistance would be great. Thanks in advance, Andrew Keil

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no audio on DTLS-SRTP call from Chrome to Freeswitch
by Dave Horton 12 Dec '16

12 Dec '16

I’m having an issue trying to set up a simple audio test from a Chrome browser to a Freeswitch using DTLS-SRTP. Everything seems to connect ok (in terms of the SIP signaling and the handshake negotiation), but when I try to play a file from Freeswitch I hear nothing in my browser. I’m looking for any tips on how to troubleshoot this (understanding that the problem may not be on the Freeswitch side, but any advice on how to troubleshoot SRTP would be useful since my usual approach of getting a wireshark and stripping the audio to listen to it doesn’t work). There is a pastebin showing the freeswitch debug and sip logs at http://pastebin.com/X2k0amY2 However, first some details on my setup, since it is has some custom pieces. My web application is using Sip.js (http://sipjs.com/) to establish a secure web socket connection to my edge proxy (which is not using Freeswitch, or verto). My edge proxy is connecting to Freeswitch using event socket, and upon receiving an INVITE from Chrome using wss transport, then generates an INVITE over UDP to Freeswitch. The call is answered and a file is played, using event socket to transmit those commands. I can see from the above trace that Freeswitch answers the call and negotiates the handshake for DTLS-SRTP ok. The file is played and I can see from a wireshark trace that packets are flowing. However, I hear no audio. Any thoughts on next steps for troubleshooting? Dave

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SBC
by Valter Nogueira 10 Dec '16

10 Dec '16

I would like to set up FS to run as a small SBC. Profile "internal" (nic em1) would be source for N asterisks that will register in FS. Some Asterisks should be trusted by IP (acl) Profile "external" (nic em2) would connect in an SIP Provider. So call coming from internal would routed to external gateway and vice-versa in dialplan. To mimic external profile I added in internal <gateways> <X-PRE-PROCESS cmd="include" data="internal/*.xml"/> </gateways> Asterisks are unable to register. I guess that the problem is internal gateway should be in directory instead of in sip_profile. Thanks, Valter

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