April 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Acl to ignore mdns ice candidates generated by browsers
by Babak Yakhchali 13 May '20

13 May '20

Hi I'm trying to use sipjs with freeswitch but the problem is freeswitch is accepting candidates like this: a=candidate:2131708102 1 udp 2113937151 ad7f0d0b-7245-4c33-84af-fda3f2921d18.local 54385 typ host generation 0 network-cost 999 these are generated by mdns from browser and cause errors like AUDIO RTP REPORTS ERROR: [Remote Address Error!] and then INCOMPATIBLE_DESTINATION I tried wan_v4.auto as "apply-candidate-acl" but it is accepting those candidate too is there any builtin acl that I can use to make fs accept only public ipv4 addresses? thanks

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asr_handle codec = (null)
by Bernd Krueger-Knauber 12 May '20

12 May '20

Hi, I'm testing unimrcp but get empty rtp packets from fs to unimrcp server when I want detect speech. The support from unimrcp told me that with my unimrcp setup is everything fine. He pointed me to the following log entry: [INFO] mod_unimrcp.c:3134 asr_handle: name = unimrcp, codec = (null), rate = 8000, grammar = (null), param = uni2 codec = (null) is wrong. But I have no idea what I can do. Everything else is working in this fs installation. Also speech generation via unimrcp is working. I use FreeSwitch 1.8.7. Any hints?

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mod call center tier behaviour
by abubacker 04 May '20

04 May '20

I am working in mod_callcenter Before my call center configuration is like this /referene : callcenter.conf.xml/ /<agents> <agent name="1000@default" type="callback" contact="[call_timeout=10]user/1000" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1001@default" type="callback" contact="[call_timeout=10]user/1001" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1003@default" type="callback" contact="[call_timeout=10]user/1006" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> </agents> <tiers>  <tier agent="1000@default" queue="support@default" level="1" position="1"/> <tier agent="1001@default" queue="support@default" level="1" position="2"/> <tier agent="1003@default" queue="support@default" level="1" position="3"/>  <tier agent="1003@default" queue="secondary" level="1" position="3"/> <tier agent="1001@default" queue="secondary" level="1" position="2"/> <tier agent="1000@default" queue="secondary" level="1" position="1"/> </tiers> </configuration> / Tier list : /freeswitch@FMS-Queue> callcenter_config tier list support@default queue|agent|state|level|position support@default|1000@default|Ready|1|1 support@default|1001@default|Ready|1|2 support@default|1003@default|Ready|1|3 +OK/ When I renamed the 1003@default as 1006@default / <agents> <agent name="1000@default" type="callback" contact="[call_timeout=10]user/1000" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1001@default" type="callback" contact="[call_timeout=10]user/1001" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1006@default" type="callback" contact="[call_timeout=10]user/1006" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> </agents> <tiers>  <tier agent="1000@default" queue="support@default" level="1" position="1"/> <tier agent="1001@default" queue="support@default" level="1" position="2"/> <tier agent="1006@default" queue="support@default" level="1" position="3"/>  <tier agent="1006@default" queue="secondary" level="1" position="3"/> <tier agent="1001@default" queue="secondary" level="1" position="2"/> <tier agent="1000@default" queue="secondary" level="1" position="1"/> </tiers>/ After that I have reload the mod_callcenter http://pastebin.freeswitch.org/14810 The 1003@default is not overwritten , but 1006@default has added newly why 1003@default is still exist /freeswitch@FMS-Queue> callcenter_config tier list support@default queue|agent|state|level|position support@default|1000@default|Ready|1|1 support@default|1001@default|Ready|1|2 support@default|1003@default|Ready|1|3 support@default|1006@default|Ready|1|3 +OK/ Is it a bug or Is it a default behaviour ? -- Best regards, N.Abubacker , Associate system engineer , bk systems pvt ltd , Ph : 9144-43902701 Disclaimer:http://www.bksystems.co.in/email-policy

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Announcing SaraPhone, SIP WebRTC Open Source business phone
by Giovanni Maruzzelli 01 May '20

01 May '20

My fellow VoIPers, I am pleased to announce the early availability of: SaraPhone ------------------ SaraPhone is a bare bone SIP WebRTC voice phone, complete with most features real companies want to use in real world: HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer, Notifications, running on all Browsers both on Desktop and SmartPhone. SaraPhone is fully integrated with FusionPBX, the full-featured domain based multi-tenant PBX and voice switch for FreeSwitch. Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies, gateways, and servers (Asterisk, OpenSIPS, Kamailio, etc). Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from Giovanni's wife, Sara Hosseini. In addition to providing all of the usual DeskPhone functionality, SaraPhone got: - Desktop Notification for Incoming Calls - Live MWI update - Real Time BLFs status update - BLF click to call - Caller Name and Number Display - Call Error Cause Display - AutoAnswer - Network Disconnect Reload - Show and Set Caller-ID (incoming-outbound) You an find it in GitHub ( https://github.com/gmaruzz/saraphone ). Anyone interested can play with it :). Have fun, giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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ICE TCP to reduce TURN cost
by David P 30 Apr '20

30 Apr '20

Hey Brian, I think I muddied the waters by referring to "TURN TCP" when I meant to be asking about "ICE TCP" Can FS be configured to support RFC 6544 ? https://tools.ietf.org/html/rfc6544 Cheers, David

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Calls per gateway
by Francesco Facco de Lagarda 30 Apr '20

30 Apr '20

Hello everyone .. I’ve been dredging the ‘net for a week trying to find a solution but can’t find one. I have a complicated system with 3 2n gateways with sim cards. Two have been dedicated to national calls and the last to international calls (the SIMS have been purchased with special deals according to the destination) I just need a simple method to find out how many current calls are on each gateway so I can route my calls accordingly. I use ESL to route my calls. I’ve tried all the variations of sofia status, show calls, show channels, but I cant find any that just tell me how many calls each gateway is handling at present (in and out)

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Re: [Freeswitch-users] mod_verto: subscribe events for specific accounts
by Gregor Nanger 30 Apr '20

30 Apr '20

Thanx for sharing. Could be usefull. One question. Is this right syntax: vertoHandle.subscribe(“presene”, { Is it presene or presence? Best regards, Gregor On Wed, 29 Apr 2020 at 12:08, António Silva via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: "António Silva" <asilva(a)wirelessmundi.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Wed, 29 Apr 2020 12:07:29 +0200 > Subject: Re: mod_verto: subscribe events for specific accounts > Reppling to myself, just found the right syntax, it is posible from > verto.js: > > To get all events: > vertoHandle.subscribe(“presene”, { > handler: function(v, e) { > log.debug("***presence event e:", e); > } > }); > > But to get specific events per account: > vertoHandle.subscribe([“presene.ACCOUNT1”, “presene.ACCOUNT2”], { > handler: function(v, e) { > log.debug("***presence event e:", e); > } > }); > > Hope it help others :) > > > > On 17 Mar 2020, at 13:53, António Silva <asilva(a)wirelessmundi.com> > wrote: > > > > Hi, > > > > Is it possible to subscribe using mod_verto? > > > > Right now I’ve enable in mod_verto the presence events but I get all the > events for all the accounts, and is fine but i was wondering if it’s is > possible to just do a single subscribe like in sip. > > > > Thanks for the help. > > > > > > Regards, > > António > > > > > > ---------- Forwarded message ---------- > From: "António Silva via FreeSWITCH-users" < > freeswitch-users(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Wed, 29 Apr 2020 03:08:12 -0700 (PDT) > Subject: Re: [Freeswitch-users] mod_verto: subscribe events for specific > accounts > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://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 > https://freeswitch.com -- 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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How to play a file to conference moderator?
by Avi Marcus 30 Apr '20

30 Apr '20

I'm setting up a conference system for a school, and a feature is instead of simply unmuting, the student raises their hand. I hacked together something with execute_application javascript that saves to coredb sqlite, so we can get a timestamp. (modifying to add a flag wouldn't be good enough, we want to save the order). Then I want to play a notice only to the moderator that someone has a question. I set up something super-hacky, and was wondering if there's a more robust way. Or a simple code change to add this functionality (I'm not a C coder, but it wasn't obvious to add "saymember moderator X" without a bunch more code) Since I need the moderator's member id to the conference, which isn't set until after he joins, I did this: <action application="set" data="api_result=${jsrun conference/setModerator.js myConference ${uuid}}" /> <action application="conference" data=" myConference@school ++flags{endconf|moderator}"/> Then setModerator.js does a uuid get var to get his id: var memberID = apiExecute("uuid_getvar", argv[1] + " conference_member_id"); console_log("DEBUG","Got ID @ "+i+": " + memberID); memberID = parseInt(memberID,10); if(memberID && memberID>0){ //NaN is falsy, we're good. var saveString="insert/conference_"+argv[0]+"/moderator/"+memberID; console_log("DEBUG","Done, saving: " + memberID + ", string:"+saveString); apiExecute("hash",saveString); I put in a wait and repeat, but I guess starting up v8 takes long enough that the variable is set by then. This seems very hacky, is there a better way to do this? Thanks, -Avi Marcus 1-718-989-9485 (USA) 02-372-1570 (Israel)

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FYI - new Amazon Web Services TTS option
by Nathan Neulinger 30 Apr '20

30 Apr '20

https://aws.amazon.com/blogs/aws/polly-text-to-speech-in-47-voices-and-24-l… The quality blows away everything I've tried out previously... Highly relevant option for voice prompts. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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TURN TCP to reduce TURN cost
by David P 30 Apr '20

30 Apr '20

Brian West <brian(a)freeswitch.com> wrote: > Subject: Re: [Freeswitch-users] ADD 2 ICE CANDIDATES mod_verto > set the variable include_external_ip=true > Hey Brian, Your tip about include_external_ip seemed relevant to my recent question whether FS supports TURN TCP; that is, can FS be configured to provide a tcp candidate in its answer sdp, which would save the cost of sending calls through TURN for users whose firewalls block needed udp ports. I searched for your variable in the source and found https://github.com/signalwire/freeswitch/blob/53cd06900c5ce827b2c0ac7448560… which shows that the transport is not frozen as "udp" but is the value of ice_out->cands[0][0].transport Is it possible for this transport member to be "tcp"? How could one configure FS to provide both udp and tcp candidates? Cheers, David

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