- FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_verto: subscribe events for specific accounts
by António Silva 29 Apr '20

29 Apr '20

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

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Need help with mod_v8 install
by Pete Kay 29 Apr '20

29 Apr '20

Hi I am getting error when trying to install mod_v8. Could someone pls help? # make make all-am make[1]: Entering directory `/usr/src/freeswitch/src/mod/languages/mod_v8' CXX mod_v8_la-mod_v8.lo In file included from *mod_v8.h:36:0*, from *mod_v8.cpp:68*: *./include/javascript.hpp:35:16:* *fatal error: *v8.h: No such file or directory #include <v8.h> * ^* compilation terminated. make[1]: *** [mod_v8_la-mod_v8.lo] Error 1 make[1]: Leaving directory `/usr/src/freeswitch/src/mod/languages/mod_v8' make: *** [all] Error 2

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Auto Changing RTP concern
by Sonny Lafuente 28 Apr '20

28 Apr '20

Hello Everyone, Does anyone here experiencing the “Auto changing RTP port” in freeswitch? It actually happened when initiating a call from local SIP going to SIP server installed in the CLOUD. There is no Audio both ends after the SIP call setup. Now, when initiating call from cloud going to local SIP I have no experience in Intermittent RTP. Btw, the device that is registered to the SIP server in the cloud is Polycom IP Phone. Basically, I have SIP server in cloud and a SIP server in local. The local SIP server is using VPN to tunnel the RTP because it is behind NAT. Instead of doing port forwarding. I have tried the following: 1. In the freeswitch, I configured the ”auto_adjust_rtp=true” 2. in the local freeswitch, I configured the internal and external profile to ext-rtp-ip=<VPN IP> If anyone who is experiencing this and able to solve the issue, please give me some point on what I need to check. Thank you. SL Sent from my iPhone

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curl zombie_exec
by Mai Duy Quang 28 Apr '20

28 Apr '20

Hi, I call curl in lua script but has this error, been googling all day long but no hope. Any have fix for that? Channel is hungup and application curl does not have the zombie_exec flag. my lua code: -- to allow curl after the call session:execute("set_zombie_exec"); freeswitch.consoleLog("notice", "[curl][ring groups][missed] " .. missed_call_data .. " "); session:execute("curl", missed_call_data .. " phone=" .. caller_id_number .. "&target=" .. ring_group_name); curl_response_code = session:getVariable("curl_response_code") curl_response = session:getVariable("curl_response_data”) Many thanks, Johnathan Mai

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freeswitch video conference recording black screen
by Aidar Kamalov 28 Apr '20

28 Apr '20

Hello, I am trying to record video conference: <param name="auto-record" value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.mp4"/> But I am getting black screen with audio only in recording file. Although I can record video on every single channels: uuid_record UUID start /var/lib/freeswitch/recordings/UUID.mp4 -- Aydar A. Kamalov

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mod_conference and Hebrew-language conferencing (pro-bono production server for school system)
by Yisroel Meir Blumstein 28 Apr '20

28 Apr '20

We want to set up a production system handling at least 1000 users per server, but ideally on the order of 100,000 users on one server. This would temporarily replace a local, or even regional, school system (pro-bono work, but they'd pay server costs; this will obviously cost a lot less than they're going to need to pay the more established company). Most of these people will need to use a Hebrew conference interface, so we also need to know if freeswitch/src/mod/say/mod_say_he/mod_say_he.c is production-ready. In the worst-case-scenario (which is obviously crazy), where everyone is talking at once, and these people are in anywhere from 10-30 person conferences (classes), what are the needed server specs? In a more realistic scenario, where a maximum of 2 people are talking at once per conference: - 20,000 users speaking, 2 per each of the 10,000 conferences - ~6700 users speaking, 2 per each of the ~3300 conferences Or, 3 people at once: - 30,000 users speaking, 3 per each of the 10,000 conferences - ~10,000 users speaking, 3 per each of the ~33,333 conferences What are the variables here? My friend runs Bestfone <https://github.com/signalwire/freeswitch/issues/www.bestfone.com>, and —while he handles tens of thousands of calls a month— he doesn't handle that many simultaneous calls, so we're trying to figure out exactly what level of power is needed here. Please help! We're trying to literally help thousands of school kids (over 124,527, as of last official numbers) if that motivates you. Thanks! <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

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sofia reject MESSAGE method
by Sean Hansen 28 Apr '20

28 Apr '20

Hello, Does anyone know of a way to reject chat/messaging for sofia? Specifically the MESSAGE method? We use FreeSWITCH purely as an SBC/media proxy for voice only, so therefore have no need for directory/chatplan/etc. We have noticed with our current configuration, when our PBX passes a MESSAGE (usually generated on accident...) to freeswitch it sends the MESSAGE right back to the PBX - which creates a loop and jams up the proxies. The log is complaining as such: freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.195480 [DEBUG] sofia_presence.c:225 Can't find registered user XXXXXXXXXXX(a)domain.com freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.195480 [DEBUG] sofia_presence.c:225 Can't find registered user XXXXXXXXXXX(a)domain.com freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.215485 [DEBUG] sofia_presence.c:225 Can't find registered user XXXXXXXXXXX(a)domain.com freeswitch.log.2020-04-21-11-57-04.1:2020-04-21 10:50:43.215485 [DEBUG] sofia_presence.c:225 Can't find registered user XXXXXXXXXXX(a)domain.com where XXXXXXXXXXX is the destination number for the message. We do not load mod_sms, and all of the chatplans are disabled. Thanks.

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Stream File on both leg simultaneously
by Jay Desai 27 Apr '20

27 Apr '20

Hi, Is there a way to stream file to both legs simultaneously? I'm trying to play a consent message to both parties before recording a call. I've tried this https://lists.freeswitch.org/pipermail/freeswitch-users/2014-February/10309… without any luck the mux option seems to make no difference. I'm on FreeSWITCH version (*1.10.2-release+git~20191231T140119Z~f7bdd3845a~64bit*) and using XML dial plan to execute commands. It plays file alternatively while the other leg is muted.

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send message via python script using ESL library
by Youssef ELOUAM 27 Apr '20

27 Apr '20

Hi everyone, I am trying to send SMS via python script but not working, I followed the script in confluence from ESL import * con = ESLconnection("127.0.0.1", "8021", "ClueCon") if con.connected(): event = ESLevent("CUSTOM", "SMS::SEND_MESSAGE") event.addHeader("to", "1000(a)192.168.1.6") event.addHeader("from", "5000(a)192.168.1.6") event.addHeader("dest_proto", "sip") print(event.serialize("plain")) event.addBody("message contents") con.sendEvent(event) as result of print command I get : Event-Name: CUSTOM Event-Subclass: SMS%3A%3ASEND_MESSAGE to: 1000%40192.168.1.6 from: 5000%40192.168.1.6 dest_proto: sip but no message received in 1000 extension. Youssef ELOUAM https://www.linkedin.com/in/elouam/

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bridge two outbound calls but got different sdp
by Leo Wolf 25 Apr '20

25 Apr '20

Hi, I usr freeswitch to call out A & B, and want to bridge calls after A & B responded. After bridging the two call, I found out that the sdp codec negotiated from A and from B are different, e.g., 102 and 111. This will cause A & B codec not sync problem. Is there any way to force clients using specific codec in freeswitch? I have found out some code in switch_core_media.c: /* it could be 98 but chrome reserves 98 and 99 for some internal stuff even though they should not. Everyone expects dtmf to be at 101 and Its not worth the trouble so we'll start at 102 */ smh->payload_space = 102; Seems the codec send to clients always begin from 102? Many Thanks, Leo

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