February 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Voicemail via web interface
by John 24 Apr '18

24 Apr '18

Hello, Can you point me to any additional information about the voice mail via web interface? I have it up and running; however if you click the play button there is no playback, if you click download it will play in MS media player. Thanks John

5 5

mod_conference scalability
by Brian 19 Apr '18

19 Apr '18

Hi, I'm new to FreeSWITCH and I'm testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn't scale nearly as well as I'd hoped (based on what I've read on how FreeSWITCH is supposed to be generally very scalable). Here's my server setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I've set file logging to "notice" level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from "top", it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian.

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Send RTP to external server
by Mickael Hubert 18 Apr '18

18 Apr '18

Hi list, I want to record each call through freeswitch. But i want record only caller (SSRC 1) OR callee (SSRC 2) voice (not both). I read about SIPREC, Jack, etc ... not interesting Do you have a idea for me please ? Thanks in advance

5 8

start_dtmf false positives
by Roman Dissauer 14 Mar '18

14 Mar '18

Hi Guys, we have troubles reliably recognizing DTMF tones by using dialplan application start_dtmf. From our carriers we get some calls with inband DTMF and some with RFC2833. To our customers we need to always do RFC2833. Therefore we check SDP information and enable start_dtmf when RFC2833 is not available on the incoming call. When start_dtmf is executed we have a lot of false positives mostly at the start of audio but also during the call. Is there a way to prevent false DTMF detection? Does anyone have same issues and a solution for it? Kind regards Roman Dissauer RDI SOLUTIONS e.U. Rosasgasse 13/24 1120 Wien, Österreich T: +43 1 3530349 - 10 F: +43 1 3530349 - 99 roman.dissauer(a)rdi.at www.rdi.at

4 7

How to bridge a call to an extension without the display updating?
by Michael Avers 13 Mar '18

13 Mar '18

Hello, I'm trying to create a random dialplan extension (say 5050) that when dialed into by a registered extension would bridge the call to another registered user (say 3000), but not change the display or P-Asserted-Identity. Right now on the bridge it sends to the calling extension *P-Asserted-Identity: "5050" <sip:3000@sip.domain>.* I tried setting all the possible values for sip_cid_type and setting ignore_display_updates to false, but it still sends the P-Asserted update as shown above. The bridge itself I tried both with user/3000(a)sip.domain and ${sofia_contact(566055(a)rsi.solidphone.net)} - same result. This is a problem because A. the display on the phone changes to "3000" and more importantly, B. if the calling user later hits redial it will dial 3000 instead of 5050. Any way to achieve what I'm trying to do? Thanks Mike

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NAT / UDP hole punching issue
by Tihomir Culjaga 07 Mar '18

07 Mar '18

hi, I have "no audio" issue with TLS and i hope someone could help as Im getting crazy ... literally :( my setup is like this: Phone <> NAT <> INTERNET <> NAT <FreeSWITCH> FreeSWITCH version: 1.6.12~64bit ( 64bit) I have a separate profile configured for TLS: <param name="rtp-ip" value="192.168.100.60"/> <param name="sip-ip" value="192.168.100.60"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="ext-sip-ip" value="stun:stun.freeswitch.org"/> <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/>  <param name="tls-only" value="true"/> <param name="tls-sip-port" value="15061"/> ================================================================================================= Name tls-public Domain Name N/A Auto-NAT false DBName sofia_reg_tls-public Pres Hosts 192.168.100.60,192.168.100.60 Dialplan XML Context public Challenge Realm auto_from RTP-IP 192.168.100.60 Ext-RTP-IP stun:stun.freeswitch.org SIP-IP 192.168.100.60 Ext-SIP-IP 85.114.41.180 TLS-URL sip:mod_sofia@85.114.41.180:15061 TLS-BIND-URL sips:mod_sofia@85.114.41.180:15061 ;maddr=192.168.100.60;transport=tls WS-BIND-URL sip:mod_sofia@192.168.100.60:5066;transport=ws WSS-BIND-URL sips:mod_sofia@192.168.100.60:7443;transport=wss HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMA CODECS OUT PCMA TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU false AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 2 FAILED-CALLS-OUT 2 REGISTRATIONS 0 i manage to register the phone with no problems but when i call the phone i get no audio; bgapi expand originate ${sofia_contact(tls-profile/agent2/ nexios(a)192.168.100.60)} &echo() FS sends the invite as: SDP in INVITE message from FS v=0 o=FreeSWITCH 1519708899 1519708900 IN IP4 *85.114.41.180* s=FreeSWITCH c=IN IP4 *85.114.41.180* t=0 0 m=audio *17480* RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 SIP Client responds with: SDP in 200 OK from the client v=0 o=- 3728718779 3728718780 IN IP4 *213.147.96.240* s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio *4002 *RTP/AVP 8 101 c=IN IP4 213.147.96.240 b=TIAS:64000 a=rtcp:4003 IN IP4 *213.147.96.240* a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 So the UDP stream is: client( *4002 * ) <> ( *17480* )FS when i sniff the traffic (on both sides client/FS) using wireshark, i see RTP packets ( src:4002, dst:17480 ) leaving the client towards FS, but I don't see RTP packets ( src:17480, dst:4002 ) from FS side leaving towards the client. so my question, of course, is why FS is not sending RTP packets to the IP:PORT notified in 200 OK SDP from the client ? Did i miss some needed configuration ? in FS logs i see *192.168.100.60 port 17480 -> 213.147.96.240 port 4002* but nothing is actually being sent out from FS 2018-02-27 11:13:02.733376 [DEBUG] sofia.c:6965 Channel sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 entering state [ready][200] 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4311 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4366 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4227 Set telephone-event payload to 101@8000 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:3021 Set Codec sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 PCMA/8000 20 ms 160 samples 64000 bits 1 channels 2018-02-27 11:13:02.733376 [DEBUG] switch_core_codec.c:111 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 Original read codec set to PCMA:8 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4572 Set telephone-event payload to 101@8000 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4631 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 Set 2833 dtmf send payload to 101 recv payload to 101 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6588 AUDIO RTP [sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551] *192.168.100.60 port 17480 -> 213.147.96.240 port 4002 *codec: 8 ms: 20 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:3875 Starting timer [soft] 160 bytes per 20ms 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6806 Activating RTCP PORT 4003 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:4261 RTCP send rate is: 5000 and packet rate is: 20000 Remote Port: 4003 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:2534 Setting RTCP remote addr to 213.147.96.240:4003 2 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6887 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 Set 2833 dtmf send payload to 101 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6894 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 Set 2833 dtmf receive payload to 101 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6917 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 Set rtp dtmf delay to 40 2018-02-27 11:13:02.733376 [NOTICE] sofia.c:8023 Channel [sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551] has been answered 2018-02-27 11:13:02.733376 [DEBUG] switch_channel.c:3770 (sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551) Callstate Change RINGING -> ACTIVE 2018-02-27 11:13:02.753388 [DEBUG] switch_ivr_originate.c:3686 Originate Resulted in Success: [sofia/tls-public/sip:agent2/ nexios@213.147.96.240:10551] 2018-02-27 11:13:02.753388 [INFO] switch_channel.c:3127 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 Flipping CID from "" <0000000000> to "Outbound Call" <nexios> 2018-02-27 11:13:02.753388 [DEBUG] mod_commands.c:4788 (sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551) State Change CS_CONSUME_MEDIA -> CS_EXECUTE 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:584 (sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551) Running State Change CS_EXECUTE 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:650 (sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551) State EXECUTE 2018-02-27 11:13:02.753388 [DEBUG] mod_sofia.c:198 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 SOFIA EXECUTE 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:328 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 Standard EXECUTE EXECUTE sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 echo() Regards, Tihomir.

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Conference mux mode video flickering
by Dom Rumsey 03 Mar '18

03 Mar '18

Hi guys Thanks for your help previously. I'm now using MUX mode for video and I'm getting a problem whereby a conference with 3 or more participants keeps flickering video streams on the floor (we are using video-muxing-personal-canvas parameter). It's as if it can't decide who to show. Below is my conference profile: <profile name="cp"> <param name="domain" value="$${domain}"/> <param name="rate" value="8000"/> <param name="video-mode" value="mux"/> <param name="video-layout-name" value="1x1"/> <param name="interval" value="20"/> <param name="caller-controls" value="default"/> <param name="energy-level" value="0"/> <param name="video-auto-floor-msec" value="3000" /> <param name="video_no_video_avatar_png" value="/usr/local/freeswitch/conf/images/video-muted.png" /> <param name="video_mute_png" value="/usr/local/freeswitch/conf/images/video-muted.png" /> <param name="conference-flags" value="audio-always|livearray-sync|livearray-json-status|video-mute-exit-canvas|video-muxing-personal-canvas"/>  <param name="max-members" value="25"/> <param name="sound-prefix" value="/usr/local/freeswitch/conf/sounds/"/> <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/> <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/> <param name="inbound-late-negotiation" value="false"/> </profile> Any pointers on where I'm going wrong would be really appreciated. Thank you

5 11

javascript scheduling multiple events excecute_on_answer
by Francesco Facco de Lagarda 01 Mar '18

01 Mar '18

I am developing a calling platform using javascript. I have code that calculates how long THAT user is allowed to call THAT number for. I am trying to schedule 2 events, 1. 1 min before time ends, that plays a message "you have 1 minute left .." 2. The actual hangup when the time expires.. Despite a zillion tests I have been unable to schedule BOTH events.. This is my code: for simplicity's sake I have set call time to 120 secs, with warning at 60: if (session.ready()) { /*** Get user, number, etc. code omitted for simplicity **/ var sessOut = new Session("sofia/gateway/ht503/" + dialedNum + "@192.168.0.201:5062"); var totTime = 60; sessOut.execute("set", "execute_on_answer=sched_hangup +120 alloted_timeout") sessOut.execute("set", "execute_on_answer=sched_broadcast +60 playback::" + soundDir + "one_min_left.wav both"); if (sessOut.ready()) { bridge(session, sessOut); } sessOut.hangup(); session.hangup(); }

2 2

Fax Transmit Rate - force to 9600?
by Andrew 01 Mar '18

01 Mar '18

Hi all, We've been trying to get our fax implementation as reliable as possible using FS; we've had some great results with inbound faxes with the following items set in the incoming dial-plan: action application ="answer"/> action application="playback" data="silence_stream://2000"/> action set ignore_early_media=true action set absolute_codec_string='PCMU,PCMA' action set fax_enable_t38=true action set fax_verbose=true action set fax_use_ecm=true action set disable-v17=true action set fax_v17_disabled=true action set fax_disable_v17=true action set fax_enable_t38_request=true application rxfax /tmp/FAX-IN-${uuid}.tif application hangup The one thing that we have identified with this configuration is that 99% of inbound faxes come through fine when the fax transfer rate is negotiated at 9600 bps (as a result of the fax_v17_disabled=true setting). However, we still see quite a few incoming faxes reported as coming in with a fax transfer rate of 14400 bps, which have a 100% fail rate. I understand that the two ends need to negotiate a rate; but how do we force (or only offer) the rate of 9600 or lower? My understanding is that the fax_v17_disabled=true setting should cover that? My first thought was that the remote end was forcing the rate at 14400, regardless of what we were sending back in terms of fax capabilities. Not sure if it helps, but the two error messages we receive when faxes fail due to the 14400 rate are: - Unexpected DCN while waiting for DCS or DIS, and - The call dropped prematurely My thought was, If we could find a way to get all faxes coming in at 9600, we'd be able to improve the reliability. Looking through the wiki and previous list messages, the fax_disable_v17=true setting seems to be the documented approach, but I'm not sure why we're still seeing faxes come in at a higher rate. Any advice on this or are we missing something with regards to locking in the lower speed? cheers, A

4 6

Cdr files not using correct template in master.csv but does if use separate files
by Joseph Waite 28 Feb '18

28 Feb '18

Hi Guys Im wondering if I’m missing something or if this is expected behaviour? I have 3 different templates and i’m using account code variables in my dial plan to tell FS to use a specific template for certain calls. If I have master-only set to true then it ignores the templates and all cdr’s use the default template. If I disable master-only, all cdr’s are written to master.csv and all use the default template, however I also get a file for each account code and this contains only the cdr’s for that account code and they use the correct template. Surely all cdr’s should use the template that matches the account code if there in the master or not? Im just wanting to check this before I file a Jira. FreeSWITCH Version 1.6.19+git~20171025T192126Z~86184a2f73~64bit (git 86184a2 2017-10-25 19:21:26Z 64bit) I have not tried on latest Master. Regards

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