January 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Requiring Pin of "1" for all conferences + announcement
by Spencer Angerbauer 01 Feb '20

01 Feb '20

I am trying to get all conferences to say a phrase before entering the pin #1 prior to entering (as we are connecting via outbound dialer from api). I have installed TTS to do a “say” function, and have updated the conf/autoload_configs/conference.conf.xml by making the “pin” required and to require pin as “1”, but every time a caller connects, it just transfers them directly into the conference without announcing that they need to press “1” to enter, nor does it require a “pin” to connect to the conference. Is there somewhere else I need to update the dialing plan to require all my conference calls require a “1” when using the web api outbound dialer to connect everyone? (I am using http://<<servername>>:8080/webapi/conference?<<friendlyname>> to connect multiple calls together). Also, is there another location to add speak to text for the beginning of every conference? Thank you for your help on both these related issues. -Spence

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spandsp detects tone only while sleeping?
by Mitchell Langs 29 Jan '20

29 Jan '20

I want to originate a call and do something as soon as the busy tone is detected. I use this lua script but nothing is detected: session = freeswitch.Session("{codec_string=PCMA}sofia/external/ringbusyde@192.168.2.4:5080"); consumer = freeswitch.EventConsumer("DETECTED_TONE"); session:execute("spandsp_start_tone_detect", "49"); event = consumer:pop(1,6000); However when I sleep after spandsp_start_tone_detect the tone is detected properly: session = freeswitch.Session("{codec_string=PCMA}sofia/external/ringbusyde@192.168.2.4:5080"); consumer = freeswitch.EventConsumer("DETECTED_TONE"); session:execute("spandsp_start_tone_detect", "49"); session:execute("sleep", "2000"); event = consumer:pop(1,6000); I'd like to understand why it is not working in the first case. log output if needed: https://pastebin.com/CnFsKNsJ

1 0

Detecting chosen devices in verto
by David P 29 Jan '20

29 Jan '20

How can a verto-based app report which devices verto has chosen to use, esp the mic? If it's possible that verto changes its choice during a call, perhaps due to a device being just plugged in, how can the app detect this?

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Re: [Freeswitch-users] Limit cps per source IP and port
by David Villasmil 29 Jan '20

29 Jan '20

Can you share your potables? Might be interesting for the the community. On Wed, 29 Jan 2020 at 10:32, António Silva via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: "António Silva" <asilva(a)wirelessmundi.com> > To: Mickael Hubert <mickael(a)winlux.fr>, FreeSWITCH Users Help < > freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Wed, 29 Jan 2020 11:09:37 +0100 > Subject: Re: [Freeswitch-users] Limit cps per source IP and port > > Hi Michael, > > Thanks a lot for the reply. I was searching of something at sofia level, > before it enters the xml routing decision. > > We already have the parameters sessions-per-second and max-sessions, but > is for the entire switch, my idea has to have the same type of parameters > per account/source ip, but i is not implemented. i've a draft on a lab > machine where i'm trying to implement this but for now is not good... > > Right now I'm using iptables to limit the maximum of new invites to FS, > and it keeps me "safe" and i can limit resources per source IP. > > > On 29/01/2020 10:15, Mickael Hubert wrote: > > Maybe it's very late, > but I use this: > https://github.com/Mickaelh51/freeswitch/blob/master/pingcalls_limit.xml > it's not per IP, but it's easy to change this code > > Le lun. 22 juil. 2019 à 23:03, António Silva via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> a écrit : > >> >> >> >> ---------- Forwarded message ---------- >> From: "António Silva" <asilva(a)wirelessmundi.com> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Bcc: >> Date: Thu, 18 Jul 2019 10:52:14 +0200 >> Subject: Limit cps per source IP and port >> Hi, >> >> is it possible to limit the number of calls per source ip and port >> before the dialplan? >> >> I know that this can be done using mod_limit in the dialplan, i was >> thinking of a limit like the sessions-per-second global limit, this way >> less resources are used if some ip/port is flooding our machine. >> >> Another possibility is using the firewall... but i like this option less >> because i wont know of the drop calls. >> >> Thanks, >> >> -- >> Saludos / Regards / Cumprimentos >> António Silva >> >> >> >> >> >> ---------- 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: Mon, 22 Jul 2019 14:03:23 -0700 (PDT) >> Subject: [Freeswitch-users] Limit cps per source IP and port >> _________________________________________________________________________ >> >> 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 > > -- > Saludos / Regards / Cumprimentos > António Silva > > > > > ---------- Forwarded message ---------- > From: "António Silva via FreeSWITCH-users" < > freeswitch-users(a)lists.freeswitch.org> > To: Mickael Hubert <mickael(a)winlux.fr>, FreeSWITCH Users Help < > freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Wed, 29 Jan 2020 02:32:08 -0800 (PST) > Subject: Re: [Freeswitch-users] Limit cps per source IP and port > _________________________________________________________________________ > > 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 -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0

freeswitch call-transfer - inbound call
by Kashif Abbasi 29 Jan '20

29 Jan '20

Hi, I am working on a call flow which includes call transfer (A calls B (B is registered extension at freeswitch), B answers and then transfers the call to C and finally A and C are connected to each other). During call transfer, I want outbound leg to be handled as inbound leg through localhost interface at freeswitch hence creating separate CDR for that (basically this is a requirement at my prepaid billing system to have another inbound leg coming from freeswitch). Can someone please tell how can I configure public dialplan to work as above and route the call to my default dialplan for further routing? My current matching condition in public dialplan is as below where transferred leg with new prefix is simply going to default dialplan as outbound leg. <extension name="xfer_calls"> <condition field="destination_number" expression="^(1122)(\d+)$"> <action application="transfer" data="$1$2 XML default"/> </condition> </extension> I tried to put "fs_path=sip:127.0.0.1" in above transfer application but that doesn't seem to be working. Please suggest. Thanks Regards

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Invitation for OpenSIPS Summit 2020 Call for Paper
by Bogdan-Andrei Iancu 29 Jan '20

29 Jan '20

Hello fellows VOIPer, If you want to share with the rest of the VoIP & RTC community some news, interesting or breaking through ideas, or even more, some experience you had in terms of designing, integrating or operating various solutions or platform based on Open Source Softwares, then you should consider submitting a paper for the OpenSIPS Summit 2020 in May, Amsterdam. https://www.papercall.io/opensips-summit-2020 We welcome anyone ready to share knowledge! Best regards, Bogdan -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ OpenSIPS Bootcamp, Miami, March 2020 https://opensips.org/training/OpenSIPS_Bootcamp_2020/

1 0

[Issue with setting variable while bridge/transfer]
by Bilal Abbasi 28 Jan '20

28 Jan '20

Hi Users, I have a switch where people are calling and they press 1 to get transferred to client's DID. I need to have stats for live total calls on my switch along with calls that are transferred(realtime stats). I am doing following commands to get the live total calls.(basically increasing variable ${CampaignId}_live on getting call and on hangup i decrement that variable). <action application="set" data="ignore=${memcache(increment ${CampaignId}_live 1)}"/> <action application="set" data="api_hangup_hook=memcache decrement ${CampaignId}_live 1"/> I am doing this while i transfer the calls to client's number(basically increasing variable ${CampaignId}_transfer on when somebody presses transfer button and on hangup i decrement that variable). <action application="bridge" data="{api_hangup_hook='memcache decrement ${CampaignId}_transfer 1 both',ignore='${memcache(increment ${CampaignId}_transfer 1)}'transfer_trigger=1}sofia/external/${TransferTo}"/> Now i have live calls stats coming correctly. But in case of transfer calls, when i have low number of calls(like 3-4 calls) it works good, but when i have 50-60 calls the stats are not presented correctly(it does not increase/decrease memcache variable). Can someone guide me how to do this in a better way?, or what mistake i am making in doing it. Regards Abbasi

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Re: [Freeswitch-users] Freeswitch Performance issue while RPC request
by Seven Du 28 Jan '20

28 Jan '20

I had this before and my solution was to fork mod_xml_rpc to mod_xml_rpc2 ... 3 .. 4 .. and proxy_pass with nginx at front. It does help. You could also try https://freeswitch.org/confluence/display/FREESWITCH/RESTful+Verto see if it performs better. On Mon, Jan 27, 2020 at 11:01 PM vineet via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: vineet <vineet.verma(a)bics.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Mon, 27 Jan 2020 07:50:49 -0700 (MST) > Subject: Freeswitch Performance issue while RPC request > Dear All, > > we are facing freeswitch performance issue > we are sending https RPC request to originate calls from freeswitch. > it seems like after 35 caps/ per second. > freeswitch is choking . > delay keeps on growing and we start receiving Http time out . > some time response is coming after 60 seconds. > > Can you please help ? > > Thanks, > vineet > > > > -- > Sent from: http://freeswitch-users.2379917.n2.nabble.com/ > > > > > ---------- Forwarded message ---------- > From: vineet via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Mon, 27 Jan 2020 07:01:53 -0800 (PST) > Subject: [Freeswitch-users] Freeswitch Performance issue while RPC request > _________________________________________________________________________ > > 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 -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

1 0

mod_portaudio: pa devlist returns ERR unless FS running under root
by Tom Lynn 28 Jan '20

28 Jan '20

I'm having some trouble understanding why the CLI pa command "pa devlist" returns ERR unless I'm running it under root. I installed on an ARM single board computer over the weekend from the ARM repository following the instructions for raspberry pi. I can use aplay from the command line to play the music on hold wav files included with FS. I've plugged in a USB sound card device and made it the default sound device and rebooted. Same result. The OS image is dietpi and there's no other running processes running that use sound. I saw alsactl running and killed it, restarted FS service, but still no devlist. Looking back through emails here in the list, there seem to be similar questions over the years, but with no solve that I found. Does anybody have first hand experience with this? I'm away from home, but I can follow this up with some console output it if helps. I'm trying to set up a loudspeaker paging device. Tom

1 0

Freeswitch Performance issue while RPC request
by vineet 27 Jan '20

27 Jan '20

Dear All, we are facing freeswitch performance issue we are sending https RPC request to originate calls from freeswitch. it seems like after 35 caps/ per second. freeswitch is choking . delay keeps on growing and we start receiving Http time out . some time response is coming after 60 seconds. Can you please help ? Thanks, vineet -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

1 0