- FreeSWITCH-users - FreeSWITCH Mailing Lists

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/

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[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

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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

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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/

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Announcing conference participant after PIN entry
by Thilo-Alexander Ginkel 27 Jan '20

27 Jan '20

Hello everyone, let's say, I have a FreeSWITCH conference that is PIN-protected and where I want to record each participant's name. How do I announce the name of a participant *after* the caller's PIN has been validated? So far I tried something like this: <extension name="Record Name and schedule conf announce"> <condition field="destination_number" expression="^34(\d{2})$"> <action application="answer"/> <action application="set" data="namefile=/tmp/${uuid}-name.wav" inline="true"/> <action application="sleep" data="1000"/> <action application="playback" data="voicemail/vm-record_name1.wav"/> <action application="playback" data="tone_stream://%(500,0,500)"/> <action application="record" data="${namefile} 5"/> <action application="playback" data="ivr/ivr-call_being_transferred.wav"/> <action application="set" data="res=${sched_api +1 none conference 31$1-${domain_name} play file_string://${namefile}!conference/conf-has_joined.wav}"/> <action application="transfer" data="31$1 XML default"/> </condition> </extension> <extension name="wb_conferences"> <condition field="destination_number" expression="^(31\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@wideband${conference_pin}"/> </condition> </extension> Participant dials 34xx, name is recorded, playback scheduled, call transferred to 31xx, which initiates the conference, but sched_api plays the join message before the pin has been validated. Is there a way to hook into the conference so that the join message is only scheduled after the PIN has been validated successfully? Thanks, Thilo

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Grandstream ATA702 - dropped calls
by William Colls 25 Jan '20

25 Jan '20

When I make an outgoing call through this device, the call is often disconnected after about 30-40 seconds. Incoming calls are fine. I have SIP transport set to UDP, NAT Transversal set to NO, Enable SIP Options Keel Alive set to YES, SIP Options Keep Alive Interval set at 30, and SIP Options Keep Alive Max Lost at 3. Freeswitch version 1.10.1 was started with -nonat. Obviously I have missed, or miss-understood, something. Any pointers or suggestions will be most welcome. Thanks for your time. William.

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Cannot force transcoding between PCMU and PCMA
by Kliment Toshkov, Netfinity JSC 23 Jan '20

23 Jan '20

Hello, I have to force transcoding between PCMU and PCMA in the following scenario: Customer [PCMA, PCMU, G729] > Kamailio > Freeswitch 1..N > termination gateway [PCMA] I need transcoding from PCMU and G729 to take place on Freeswitch boxes. Currently it works for G729 but for PCMU calls they are originated with PCMU. Inbound profile: <param name="inbound-codec-prefs" value="$${global_codec_prefs}”/> <param name="disable-transcoding" value="false"/> Outbound profile: <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="disable-transcoding" value="false"/> vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,PCMU,G729"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA"/> In xml dial plan: <action application="set" data="inherit_codec=false"/> <action application="set" data="media_mix_inbound_outbound_codecs=true"/> But the calls are still not transcoded between PCMU and PCMA. Only one thing worked - forcing the codec in bridge app: <action application="bridge" data="{absolute_codec_string='PCMA'} ..."/> But I really need to know what I am missing and how to fix it using config files only. Brute force is not preferred. Thank you! Kliment Toshkov Key Accounts 24/7: 0700 30000 w: www.netfinity.bg <http://www.netfinity.bg/> t: +359 2 4918888 f: +359 2 4815555 <http://www.netfinity.bg/> This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited.

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No audio for conference call
by Md,Mehedi Hasan Kabir(Tanim) 23 Jan '20

23 Jan '20

Hi I have tried to test FreeSWITCH conference by dialing 3000 from two zoiper clients. When I call the conference number, I can hear hold music. when the second call dial the conference number hold music stops, but no voice between legs. what would be the reason for this? Audio is fine when I call other users directly. Regards Tanim

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FreeSWITCH uniMRCP dynamic speech context
by Arjunstl 23 Jan '20

23 Jan '20

Hi, I'm working on ASR application with uniMRCP in lua. but when doing with dynamic speech context mentioned in unimrcp usage manual. i'm unable to get exact results. my aim is to recognize only those words which i want to define in phrases(my defined speech context). most of the times, recognizer recognizes random words, that are not defined in speech context. any help or suggestion would be highly appreciated. Best Regards -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

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