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Stumped by mod_portaudio permissions issue
by Mark Haun 02 Feb '20

02 Feb '20

I built freeswitch version 1.7.0+git~20150730T192909Z~ab7f83c654~32bit for my ARM-based home server (an NVIDIA Jetson board), with the intention of using mod_portaudio with a wireless USB headset for my main home phone. It works well when running freeswitch directly as root, but when I configured freeswitch to run at system startup, mod_portaudio could no longer find any audio devices. No problem, I thought, it's a simple permissions issue. I am using the example debian init script from the wiki (this is a Jessie install with SysV init rather than systemd). It runs freeswitch as user freeswitch, group daemon using the -u and -g command-line options. So I added "freeswitch" to the audio group, then used sudo to verify that I could play sound files with "aplay" as the freeswitch user. That worked, but it did not solve the mod_portaudio failure. Thinking there must be an issue with portaudio separate from ALSA, I fetched and built the "pa_devs" utility which enumerates and prints the available audio devices. (This is basically what mod_portaudio is doing on startup.) Running as the freeswitch user, that worked too. Hmmmm. I tried running freeswitch from the command line using sudo rather than the -u and -g options, i.e. $ sudo freeswitch /usr/local/freeswitch/bin/freeswitch -nc and that worked as well! Then I started experimenting with -u and -g and discovered that $ /usr/local/freeswitch/bin/freeswitch -nc -u freeswitch allows mod_portaudio to load correctly, whereas $ /usr/local/freeswitch/bin/freeswitch -nc -u freeswitch -g daemon causes mod_portaudio to fail. Can anyone explain what's happening here? In both cases the process UID, GID, EUID, EGID, FUID, and FGID are the same (freeswitch:daemon) as displayed with "ps xao pid,uid,gid,euid,egid,fuid,fgid". Could this be a bug in freeswitch? There are a couple of threads from 2-3 years ago in the archives, where mod_portaudio permissions issues were never resolved and the solution remained a mystery. For the time being I guess this (not using -g) is a workaround, but I'd sure like to understand what is going on. Mark

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Angular Vero
by Gregor Nanger 02 Feb '20

02 Feb '20

Hi! Anyone ported angularjs verto service to Angular? Best regards, Gregor

2 2

Calls repeat at 80 seconds?
by Avi Marcus 02 Feb '20

02 Feb '20

FreeSWITCH Version 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit) Is anyone having an issue with calls suddenly "restarting" at about 80 seconds? I thought it was 1 carrier on 1 route, but I'm getting reports now across various destinations using different carriers. It's not on every call, just some. I've upgraded to FreeSWITCH Version 1.10.2-release-13-f7bdd3845a~64bit (-release-13-f7bdd3845a 64bit) and we'll see if I still get the issue. Thanks, -Avi Marcus

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Restricting In bound calls
by William Colls 02 Feb '20

02 Feb '20

I am sure that I read somewhere that there is a parameter that i can set so that only incoming calls from specific IP address are accepted, but I can't find the reference, remember the parameter name, or where it is located. Could some kind sole point me in the right direction? Thanks for you time. William.

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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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MOD_SMS and ChatPlan - sofia_presence.c:225 Can't find registered user
by Aingaran Thirunadarajah 01 Feb '20

01 Feb '20

I am trying to send text between 2 registered extensions using Acrobits Groundwire. Extensions 1000 and 2000. Calls work in & out between the 2 extensions. I created a default.xml chat plan with the following, and reply is successfully sent to the sender. <?xml version="1.0" encoding="utf-8"?> <include> <context name="public"> <extension name="demo"> <condition field="to" expression="^(.*)$"> <action application="info"/> <action application="reply" data="BODY: ${_body}. FROM: ${from_user} TO: ${to_user}"/> <action application="set" data="to=${to_user}@${to_host}"/> <action application="set" data="dest_proto=sip"/> <action application="set" data="skip_global_process=true"/> <action application="set" data="from=${from_user}"/> <action application="set" data="final_delivery=true"/> <action application="set" data="from_full=<sip:${from_user}@${from_host}>"/> </condition> </extension> </context> </include> I am trying to forward, redirect the message from the sender to recipient. >From what I understand you can only use the SEND application once, so what is the correct application I must use to redirect the message from 1000 to 2000 or visa versa? Here is the log: (1001 is the sender). 2020-01-31 21:15:31.585237 [INFO] mod_sms.c:368 Processing text message 1001->1002 in context public Chatplan: 1002 parsing [public->demo] continue=false Chatplan: 1002(a)192.168.2.100 Regex (PASS) [demo] to(1002(a)192.168.2.100) =~ /^(.*)$/ break=on-false Chatplan: 1002(a)192.168.2.100 Action info() Chatplan: 1002(a)192.168.2.100 Action reply(BODY: ${_body}. FROM: ${from_user} TO: ${to_user}) Chatplan: 1002(a)192.168.2.100 Action set(to=${to_user}@${to_host}) Chatplan: 1002(a)192.168.2.100 Action set(dest_proto=sip) Chatplan: 1002(a)192.168.2.100 Action set(skip_global_process=true) Chatplan: 1002(a)192.168.2.100 Action set(from=${from_user}) Chatplan: 1002(a)192.168.2.100 Action set(final_delivery=true) Chatplan: 1002(a)192.168.2.100 Action set(from_full=<sip:${from_user}@${from_host}>) 2020-01-31 21:15:31.605242 [INFO] mod_sms.c:495 CHANNEL_DATA: Event-Name: [MESSAGE] Core-UUID: [87adf9e7-05fa-4a0e-979c-3a95c8a5dcfb] FreeSWITCH-Hostname: [localhost] FreeSWITCH-Switchname: [localhost] FreeSWITCH-IPv4: [198.98.50.94] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2020-01-31 21:15:31] Event-Date-GMT: [Sat, 01 Feb 2020 05:15:31 GMT] Event-Date-Timestamp: [1580534131585237] Event-Calling-File: [sofia_presence.c] Event-Calling-Function: [sofia_presence_handle_sip_i_message] Event-Calling-Line-Number: [4884] Event-Sequence: [14024] login: [sip:mod_sofia@198.98.50.94:8700] proto: [sip] to_proto: [sip] from: [1001(a)192.168.2.100] from_user: [1001] from_host: [192.168.2.100] to_user: [1002] to_host: [192.168.2.100] from_sip_ip: [99.244.253.120] from_sip_port: [2833] to: [1002(a)192.168.2.100] subject: [SIMPLE MESSAGE] context: [public] type: [text/plain] from_full: [<sip:1001@192.168.2.100:8700 >;tag=AABB1326546BE30A8DB5A5B4E1AA398A] sip_profile: [internal] dest_proto: [sip] max_forwards: [70] DP_MATCH: [1002(a)192.168.2.100] Content-Length: 4 Hi. 2020-01-31 21:15:31.605242 [DEBUG] sofia_presence.c:225 Can't find registered user 1001(a)192.168.2.100 Thanks

1 0

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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