- FreeSWITCH-users - FreeSWITCH Mailing Lists

syntax to use TURN server
by Mundkowsky, Robert 17 Sep '19

17 Sep '19

Hi, What is the Javascript syntax to use a TURN server with Verto? With a username and password. Robert ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________

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Re: [Freeswitch-users] Two calls with same call-id
by Dominique Jeannerod 17 Sep '19

17 Sep '19

Hi all, currently working to understand some calls without RTP (one-way audio, or no audio at all), I noticed a strange case in our traces : we send an outgoing call on a trunk. The destination number is translated to another number, and the call comes back to our SBC (the new dest is on our network). What i noticed is that the incoming call has the same call-id as our outgoing call (I suppose that the other operator uses proxies rather than SBCs), and it comes to the *same *freeswitch (has 2 IP addresses) So, wondering what could be the impact of that on our Freeswitch SBC. Can those 2 calls be managed as one and only call by freeswitch ? Thanks with anticipation Best regards, Dominique Jeannerod *Calls traces : * *First call (outgoing) :* Our SBC IP1 -> External IP 1 INVITE sip:+<Dest number 1>@<External IP 1> SIP/2.0 Via: SIP/2.0/UDP <Our SBC IP 1>;rport;branch=z9hG4bKg6y1BXQ5ZvreF Max-Forwards: 68 From: "Source num" <sip:+<Source Num>@< Our SBC IP 1>;tag=r8p71798metHN To: <sip:+ <Dest number 1>@<External IP 1> *Call-ID: d3b11cd5-4a87-1238-068f-5254000db631*CSeq: 9292108 INVITE Contact: <sip:gw+gwname@ <Our SBC IP 1>:5060;transport=udp;gw=gwname> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Session-Expires: 14400;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 248 X-FS-Support: update_display,send_info P-Asserted-Identity: "<PAI Display>" <sip:+<PAI Number>@ <Our SBC IP 1> > v=0 o=FreeSWITCH 1567668560 1567668561 IN IP4 <Our SBC IP 1> s=FreeSWITCH c=IN IP4 <Our SBC IP 1> t=0 0 m=audio 23240 RTP/AVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 SIP/2.0 200 OK Via: SIP/2.0/UDP <Our SBC IP 1> ;received= <Our SBC IP 1> ;rport=5060;branch=z9hG4bKg6y1BXQ5ZvreF From: "Source num" <sip:+<Source Num>@< Our SBC IP 1>>;tag=r8p71798metHN To: <sip:+3 Dest number 1>@<External IP 1>;tag=t25KQD4evZ2DF Call-ID: d3b11cd5-4a87-1238-068f-5254000db631 CSeq: 9292108 INVITE Contact: <sip:+<Dest number 1>@ <External IP 1>;did=4a5.72bd7e82> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Require: timer Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 247 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip: <Dest number 1>@<domain> X-FS-Support: update_display,send_info v=0 o=FreeSWITCH 1567660096 1567660097 IN IP4 <Our SBC IP 2> s=FreeSWITCH c=IN IP4 <External RTP IP 1> t=0 0 m=audio 58152 RTP/AVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 *Second call (Incoming) :* External IP2 -> Our SBC IP 2 INVITE sip:+<Dest number 2>@<Our SBC IP 2>:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP <External IP 2>:5060;branch=z9hG4bK446.1e47ee86.0 Max-Forwards: 62 From: "Source num" <sip:+<Source Num>@<domain>;user=phone>;tag=r8p71798metHN To: <sip:+<Dest number 2>@<domain>;user=phone> *Call-ID: d3b11cd5-4a87-1238-068f-5254000db631*CSeq: 9292108 INVITE Contact: <sip:gw+<gwname>@<External IP 2>;did=4a5.e4c51b72> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Session-Expires: 14400;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 247 X-FS-Support: update_display,send_info P-Asserted-Identity: "<PAI Display>" <sip:+<PAI Number>@<domain>;user=phone> v=0 o=FreeSWITCH 1567668560 1567668561 IN IP4 <Our SBC IP 1> s=FreeSWITCH c=IN IP4 <External RTP IP 2> t=0 0 m=audio 56204 RTP/AVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 SIP/2.0 200 OK Via: SIP/2.0/UDP <External IP 2>:5060;branch=z9hG4bK446.1e47ee86.0 From: "Source num" <sip:+<Source Num>@<domain>;user=phone>;tag=r8p71798metHN To: <sip:+<Dest number 2>@<domain>;user=phone>;tag=t25KQD4evZ2DF Call-ID: d3b11cd5-4a87-1238-068f-5254000db631 CSeq: 9292108 INVITE Contact: <sip:+<Dest num 2>@>Our SBC IP 2>:5060;transport=udp> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Require: timer Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Session-Expires: 14400;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 248 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:<Dest number 2>@keyyo.net X-FS-Support: update_display,send_info P-Asserted-Identity: "Outbound Call" <sip:<Dest number 2>@<domain>> v=0 o=FreeSWITCH 1567660096 1567660097 IN IP4 <Our SBC IP 2> s=FreeSWITCH c=IN IP4 <Our SBC IP 2> t=0 0 m=audio 31704 RTP/AVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 -- Ce message et toutes les pièces jointes (ci-après le “message”) sont établis à l’intention exclusive de ses destinataires et sont confidentiels. Si vous recevez ce message par erreur, merci de le détruire et d’en avertir immédiatement l’expéditeur. Toute utilisation de ce message non conforme a sa destination, toute diffusion ou toute publication, totale ou partielle, est interdite, sauf autorisation expresse. L’internet ne permettant pas d’assurer l’intégrité de ce message . Interactiv-group (et ses filiales) décline(nt) toute responsabilité au titre de ce message, dans l’hypothèse ou il aurait été modifié. IT, ES, UK. <https://interactiv-group.com/disclaimer.html>

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Disable verto's auto-relogin?
by David P 17 Sep '19

17 Sep '19

When there's a network break, verto automatically attempts to relogin. How can this be disabled? It may be the cause of some of our 2-caller conference calls showing 'conference already full'.

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how get billserc in dialplan?
by Jorge Cadena 16 Sep '19

16 Sep '19

Hi how get billsec in dialplan. i am try a condition after sched_transfer_time. <condition field="${cond( ${answered_time} >= ${duration} ? YES : NO )}" expression="^YES$"> i need check two time in seconds, without hangup the call example <condition field="${cond( 60 >= 30 ? YES : NO )}" expression="^YES$"> Thank

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lua calling bash script not working
by Joli Martinez 16 Sep '19

16 Sep '19

I have a batch script that needs to run when a call comes in. I created a lua script that gathers all the variables needed and sends it to the bash script. If I run the lua script from the linux cli the script calls the bash script without any issues and it executes correctly. Now when I make a call the bash script does not get called. I have outputted messages before and after the call to the script and they show up on the fs_cli, but again my bash script which is the middle of both messages does not get called. I am running Debian 8 on FS 1.6. Both my scripts are in /usr/share/freeswitch/scripts/ I have validated that all the permissions are the same on both scripts. What am I missing?

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building mod_java on windows
by Babak Yakhchali 14 Sep '19

14 Sep '19

Hi has anyone tested mod_java on windows? thanks

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What are these log entries?
by Allan Kristensen 13 Sep '19

13 Sep '19

Hello, Out of nowhere we have started to get a lot of these: [WARNING] switch_core_file.c:408 File has 1 channels, muxing to 2 channels will occur. [WARNING] switch_core_file.c:408 File has 1 channels, muxing to 2 channels will occur. [WARNING] switch_core_file.c:408 File has 1 channels, muxing to 2 channels will occur. I've tried to look in the code for an explanation, without succes. Does anyone know why this is appearing ? Thank you, Allan *Running: FreeSWITCH Version 1.8.7+git~20190801T142138Z~b4a4ac33f8*

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FYI: DEB packages compiling trick
by Giovanni Maruzzelli 12 Sep '19

12 Sep '19

Hello fellow FreeSWITCHers, I just added to documentation in confluence, but I repeat here, for the defense of your hair: ***after installing cowbuilder*** and before building packages, you must, from command line: # echo "ALLOWUNTRUSTED=yes" >> /etc/pbuilderrc because nonstandard packages from freeswitch repo are not trusted by pbuilder. -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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Verto call waiting
by Jose Figueroa 12 Sep '19

12 Sep '19

Hello, I'm trying to implement a verto call waiting (2nd call). I mean when I'm in a call and there's a second call to the same extension I need that verto client can take the second call and leave the first call on waiting. But I cannot complete this, I see on old mail https://lists.freeswitch.org/pipermail/freeswitch-users/2016-May/120422.html which says that on the same video tag dynamically is set. Do you have any examples of this? Any help would be appreciated. Thank you. Jose Figueroa

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ETA 1.10.2? you cannot use DB with mod_fifo and mod_directory in 1.10.1
by Giovanni Maruzzelli 10 Sep '19

10 Sep '19

Hello fellow FreeSWITCHers, there is already an ETA for 1.10.2? Atm, with packages from 1.10.1, you cannot use DB in mod_fifo and mod_directory, due to a bug fixed in master... ( https://freeswitch.org/jira/browse/FS-11995 ) Obviously, if you compile from git master, is already fixed, thanks to Andrey Volk -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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