February 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] portaudio and pulsaudio works on FS, but INCOPATIBLE DESTINATION
by walter＠telivo.net 15 Feb '20

15 Feb '20

Check out my presentation https://drive.google.com/uc?id=1O8uWLWYsN3RLQPgWIeIjcWJSYwYMiwpZ&export=dow… Password for archive: 7777 _________________________________________________________________________ > >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

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convert sqlite to mysql
by Joli Martinez 15 Feb '20

15 Feb '20

Hello, We have several freeswitch boxes that are running sqlite. We have outgrown this and need to move the DB to mySQL. Not sure how to import it, or if there is a procedure already in place for this. What would be the best option for this? let me know

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Re: [Freeswitch-users] Limit cps per source IP and port
by Mickael Hubert 14 Feb '20

14 Feb '20

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

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bypass media and session timers
by Sandro Bordacchini 14 Feb '20

14 Feb '20

Hello everyone. I have troubles to make these two features work together. I experienced this on a 1.6 version with NAT at the far-end, but i tried it in a simpler and more up-to-date scenario (LAN and 1.10) and i still have the same problem. I have a simple call between two sip phones, with bypass media set to true. At first, session-timer reINVITEs are sent correctly. If a phone does something like a hold-unhold or a blind transfer, the following session-timer reINVITEs are sent without any SDP from FS (i.e. no sdp in invite from fs, sdp in 200 ok from phone, no sdp in ack from fs). Doing some search on the mailing list, I came across a post on this ml dated Jan 28th, 2011 where Mr. Minessale stated that, since session-timer code is in sofia libraries and bypass media is in freeswitch, it might be not easy to have them working together. Have I understood correctly and is this still true? Unfortunately the proposed workaround did not work for me. Do you have any suggestions for me to have a configuration working with these two features enabled and correctly working? I would be able also to start call with bypass_media = false and then use an uuid_media off command to detach FS, but also this approach did not come out as a working solution. I am able to provide network dump and console log if needed. Thank you a lot in advance. SB

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Multi-Tenant PBX with commercial support
by Andrew Cassidy 14 Feb '20

14 Feb '20

Hi Guys, I have a client looking for a multi-tenant PBX system with commercial support. I have already put forward Kazoo but they're after a number of options. Any and all suggestions are welcome! Kind regards, -- *Andrew Cassidy BSc (Hons) MBCS* Managing Director 0330 44 55 960 <+443304455960> andrew(a)cassidyweb.co.uk www.cassidyweb.co.uk

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Local and Remote endpoints
by Sean Devoy 13 Feb '20

13 Feb '20

I want to configure a FreeSwitch server to run at the main office and allow local phones and phones from remote offices to register. The server and local phones are on a NAT Lan. Do I need separate Directory Domains (multi-tenant)? Do I need separate sip profiles on different ports? Thanks for any tips. Sean

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error in mod_shout build
by kazem mokhtari 12 Feb '20

12 Feb '20

Hi all, I'm working on libmp3lame-dev error! I have installed that but when i configure and make freeswitch i get same error. please help me, thanks making all mod_shout make[4]: Entering directory `/usr/local/src/freeswitch/src/mod/formats/mod_shout' Makefile:924: *** You must install libmp3lame-dev to build mod_shout. Stop. make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod/formats/mod_shout' make[3]: *** [mod_shout-all] Error 1 make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/local/src/freeswitch/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [all] Error 2

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inject srtp source to call
by Aidar Kamalov 12 Feb '20

12 Feb '20

Hello! I have a video call between two endpoints. Is it any way to inject rtsp source from ip video camera to existing bridge as another participant? -- Aydar A. Kamalov

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free switch use high CPU utilization
by Manjeesh Chauhan 12 Feb '20

12 Feb '20

hi team, Free switch use high CPU utilization in linux os. and CPU utilization 100% consumption myserver configuration 32 CPU 64 GB RAM call received per minutus 20 -- Thank You With Regards Manjeesh Chauhan +919818249665

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portaudio callout only PCMA (but not G722)
by Tamer Higazi 12 Feb '20

12 Feb '20

Hi people, I try to figure out why I am only cappable to callout with portaudio with PCMA and not with G722 In the dialplan I have this: <action application="set" data="media_mix_inbound_outbound_codecs=true"/> otherwise I am not capable to call out. Do I need any other modules or codecs which I forgot to compile ? When placing the call I see this: 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[L16:100:48000:20:768000:1] 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5508 Set telephone-event payload to 101@8000 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:3837 Set Codec sofia/external/08000xxxxxx PCMA/8000 20 ms 160 samples 64000 bits 1 channels 2020-02-12 15:02:57.593179 [DEBUG] switch_core_codec.c:111 sofia/external/08000xxxxxx Original read codec set to PCMA:8 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5851 Set telephone-event payload to 101@8000 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5909 sofia/external/08000xxxxxx Set 2833 dtmf send payload to 101 recv payload to 101 When I call from my sip hard phone I see this: 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match any ideas ? best, Tamer

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