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by Ignacio Durli 18 Nov '16

18 Nov '16

Thanks for your reply Joel. Here's the output, I see that the FS process is binding 10.X interface with port 5060, but I don't know why that doesn't match what sofia status is showing: root@osboxes:~# iptables -L -n Chain INPUT (policy ACCEPT) target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination root@osboxes:~# netstat -putan | grep -i freeswitch tcp 0 0 10.0.3.15:5080 0.0.0.0:* LISTEN 3519/freeswitch tcp 0 0 10.0.3.15:5081 0.0.0.0:* LISTEN 3519/freeswitch tcp 0 0 10.0.3.15:5060 0.0.0.0:* LISTEN 3519/freeswitch tcp 0 0 10.0.3.15:5061 0.0.0.0:* LISTEN 3519/freeswitch tcp 0 0 0.0.0.0:8080 0.0.0.0:* LISTEN 3519/freeswitch tcp 0 0 0.0.0.0:8081 0.0.0.0:* LISTEN 3519/freeswitch tcp 0 0 0.0.0.0:8082 0.0.0.0:* LISTEN 3519/freeswitch tcp 0 0 10.0.3.15:7443 0.0.0.0:* LISTEN 3519/freeswitch tcp 0 0 192.168.99.103:59607 192.168.99.103:3306 ESTABLISHED 3519/freeswitch tcp 0 0 192.168.99.103:59606 192.168.99.103:3306 ESTABLISHED 3519/freeswitch tcp 0 0 192.168.99.103:59611 192.168.99.103:3306 ESTABLISHED 3519/freeswitch tcp6 0 0 ::1:5080 :::* LISTEN 3519/freeswitch tcp6 0 0 ::1:5081 :::* LISTEN 3519/freeswitch tcp6 0 0 ::1:5060 :::* LISTEN 3519/freeswitch tcp6 0 0 ::1:5061 :::* LISTEN 3519/freeswitch tcp6 0 0 ::1:8081 :::* LISTEN 3519/freeswitch tcp6 0 0 ::1:8082 :::* LISTEN 3519/freeswitch tcp6 0 0 :::8021 :::* LISTEN 3519/freeswitch udp 0 0 10.0.3.15:5060 0.0.0.0:* 3519/freeswitch udp 0 0 10.0.3.15:5080 0.0.0.0:* 3519/freeswitch udp 0 0 0.0.0.0:1337 0.0.0.0:* 3519/freeswitch udp6 0 0 ::1:5060 :::* 3519/freeswitch udp6 0 0 ::1:5080 :::* 3519/freeswitch

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Video conference with screensharing. Good quality without high CPU usage.
by Александр Дружилов 18 Nov '16

18 Nov '16

Hello everybody! My case is: make video conference (with layout 1x1 only) and with possibility to share screen or video from camera. The main problem for me is a high CPU usage for good quality video (for example screen sharing 1920x1280. I use latest version of Freeswitch and mod_sofia. My config in mod_conference is: <param name="video-mode" value="mux"/> <param name="video-layout-name" value="1x1"/> <param name="video-fps" value="5"/> <param name="video-codec-bandwidth" value="1mb"/> <param name="video-canvas-size" value="1920x1080"/> <param name="video-quality" value="1"/> <param name="interval" value="40"/> <param name="conference-flags" value="wait-mod|audio-always|manage-inbound-video-bitrate"/> In case of "mux" with this configuration if somebody tries to send his video from camera (800x600) then everybody will see him bordered by black canvas. But if I will set video-canvas-size = 800x600, then if somebody tries to share his screen in resolution 1920x1280 then everybody will see only 800x600 and quality will be not really good. The same situation with the video-fps parameter. My questions are: 1) If I have only 1x1 layout, so what video-mode should I use for best quality and lowest CPU usage? transcode or mux? 2) How to make screen sharing video stream and camera video stream to have different resolution/bitrate/fps? 3) Should I use mod_verto for better quality/performance? 4) What kind of optimizations could you recommend me for better quality/performance?

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Allow only one register
by Agustí Ubalde 18 Nov '16

18 Nov '16

Hi all, I'm using FreeSWITCH with WebRTC and I'm very interested on allow only one user per user. Is it possible? Thanks, PRESENCE TECHNOLOGY An ENGHOUSE INTERACTIVE Company Agustí Ubalde Bellot Chief Developer C/ Comte Urgell 240 3º-A Barcelona 08036 aubalde(a)presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com <http://www.presenceco.com/>*

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Only one bug of this type allowed!
by Murat 18 Nov '16

18 Nov '16

Hey, occasionally, we're getting the following error log from FS. [ERR] switch_core_media_bug.c:432 Only one bug of this type allowed! I'd like to understand what's happening. I've googled and searched around a but can't find anything useful aside from the code [1]. The code seems to indicate that multi-threading problems are recognized, I assume from a faulty plugin? Our version is 1.5.8b. Since it happens very rarely, I'm not sure what the implications are. Any pointers, thoughts, ideas? Thanks, murat [1] https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/switch…

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Lua freeswitch.Dbh() question
by Abaci B 17 Nov '16

17 Nov '16

when runned a query using freeswitch.Dbh dbh:query, if there is results it a function for each returned row where you can see the data. however, if there is an error you can see the database returned error in the console on level ERR but you don't get it back in lua is someone aware of a way to get the rror back into lua?

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Re: [Freeswitch-users] FAX issues/questions
by Alexandru Covalschi 17 Nov '16

17 Nov '16

First of all thanks for answer! It's ok to buy several licenses, but why then it works at all right now? To the same destination 8/10 are successful, 2/10 have that issue with codec. Also, can you point me to any material I can read to understand the T.38 encapsulation? Is that a usual practice or it's just the way FS does that? I'm very "weak" at faxing and maybe I'm asking silly questions, but I really need to understand how things work... Cheers, Alex

2 1

FS-9113, still experiencing TLS crashes
by Emrah 17 Nov '16

17 Nov '16

Hello List, Thanks to the help provided by Stanislav, I learned of issue #9113, https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-9113/FS-9113.h…, which seems to be related to the issues I have been experiencing with FreeSWITCH, TLS and failed call setups. Coincidentally, or not, the fix pushed on that issue was aligned with whole months where I did not experience any TLS issues. Calls were going through fine, until all of a sudden they started failing again. This is on 2 distinct servers running a load balanced FS setup, and using Yealink phones. To sum up, here is what is going on. From the Yealink, calls with TLS work if I don't use SRTP. From the Yealink, calls crash if I use TLS and SRTP. From my laptop softphone, calls only crash sometimes if I use TLS and SRTP. How can I debug the TLS session on the FreeSWITCH side to see what happens with the TLS thread? I don't mean packet capture. I have a feeling that the packet size is too large and doesn't make it to the FS box intact after the 407 Proxy Required is received by the client. Here is the log for the Yealink: http://pastebin.com/smKP286x Your lights would be so appreciated, I'm losing my mind over this. Best, Emrah

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FAX issues/questions
by Alexandru Covalschi 16 Nov '16

16 Nov '16

Hello list, In my set up Freeswitch acts as fax originator, so an image is being processed and sent by FS. I do not receive any fax, so all my questions/issues are related to sending it. All fax is over IP. First a question - how to understand if Freeswitch is sending in T.38? Why do I always see T.30 FLOW in log even if in SDP is mentioned T.38? I do see some "T.38 rxfax set to 0/txfax set to N", I understand that rx is receive and tx is transmit, but why any other log strings shows T.30 FLOW? And the issue - sometimes when I initiate fax I get ----------------- Codec RAW Signed Linear (16 bit) encoder error mod_g729.c:102 This codec is only usable in passthrough mode! ----------------- I can't understand why it's happening as it occurs from time to time, but it always happens before Freeswitch makes re-invite to initiate a T.38 session. I suppose that's because I have no license for g729, but the supplier side negotiates it before FS has chance to re-invite. But why does it work with same SDP in other times? Does it have sense to exclude g729 for the initial codec negotiation, or g729 is essential to keep T.38 working? What shall I do in case of Asia where everything is on g729,? Maybe it has sense to purchase a license - but why then it works unstable right now, I mean, why does it even work if it requires mod_g729_com? Or it doesn't? I'm really confused on all that. Thanks in advance!

2 1

Freeswitch with shared MySQL over ODBC
by Joel Serrano 16 Nov '16

16 Nov '16

Hi, I'm preparing 2 servers with shared MySQL database. I'm not sure if the doc are outdated... are the following 3 settings required in switch.conf.xml: <param name="core-db-dsn" value="odbc://freeswitch:fs:password"/> <param name="core-recovery-db-dsn" value="odbc://freeswitch:fs:password"/> <param name="odbc-dsn" value="odbc://freeswitch:fs:password"/> Or is it enough only with "core-db-dsn" ? Should any of them go in the profile xml rather than the switch.conf.xml? What is the difference between "core-db-dsn" and "odbc-dsn" parameters? (in case "odbc-dsn" is still required) The idea is to run "sofia recover" on second node on the event of a failure on the first node, and to continue with call that are in progress. Thanks. Joel.

1 0

No DTMF in conference
by Jude Mukundane 16 Nov '16

16 Nov '16

Hello All, I have two servers - one running 1.5.15 and another running 1.6.12. In both servers if I issue the originate command on fs_cli "originate user/1000 &conference(h)", I do see DTMF digits on the console for 1.5 but not 1.6. Am using zoiper on the same phone for both servers and the DTMF settings (duration etc) seem to be the same on both servers. I can get DTMF for one to one calls, but not conferences. I googled... Has any one run into this and know how to solve it? Jude

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