From obelousov at gmail.com Wed Dec 2 12:56:33 2020 From: obelousov at gmail.com (Oleg Belousov) Date: Wed, 2 Dec 2020 13:56:33 +0100 Subject: [Freeswitch-users] Change SDP by SIP UPDATE Message-ID: Hi. Is there a possibility in Freeswitch to update the SDP of a pre-answered call by SIP UPDATE message? Call flow is as following: 1. FS responds to A-party with 183/SDP (SDP IP==FS IP) and performs some checks in preanswer mode 2. If checks on step 1 passed ok, FS initiates a call to B-party, receives from B-party 183/SDP (SDP IP = MGW_B) and sends to A-party 183/SDP with same new SDP IP = MGW_B, so doing change of SDP parameter in the same transaction. That scenario executes with below commands (need in media travel directly between parties after the bridge) session:execute("set","bypass_media=true") second_session = session:execute("bridge",BridgeStr); The question: is there a parameter which allows during the bridge to update SDP by SIP UPDATE, instead of changing SDP using provisional responses? A-party switch does not accept such SDP change and thus there is a problem with audio after the bridge. -- obelousov.tel -------------- next part -------------- An HTML attachment was scrubbed... URL: From adrian.worutowicz at esifrance.net Thu Dec 3 11:32:57 2020 From: adrian.worutowicz at esifrance.net (Adrian Worutowicz) Date: Thu, 3 Dec 2020 11:32:57 +0000 Subject: [Freeswitch-users] Force Inband DTMF Reception (SOLVED) Message-ID: Just for the records : In order to receive DTMS in-band for incoming calls, FS must not send in SDP : a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 This is achieved by : In vars.xml : In sip_profiles\internal.xml and/or external.xml (Settings): Add line : Remove line (if exists) : Big thanks to @seven1240 ! De : FreeSWITCH-users De la part de Piotr Gregor Envoyé : lundi 30 novembre 2020 16:58 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] Force Inband DTMF Reception Cześć Adrian, Tak, to ma sens. Spróbuj może ustawić flagę dtmf_type=none To powinno sprawić, że w SDP ANSWER nie będzie a=rtpmap:101 telephone-event/8000, tylko nie jestem pewien czy detekcja in-band DTMF będzie włączona. pozdrawiam, Piotr On Mon, 30 Nov 2020 at 15:34, Adrian Worutowicz > wrote: Witaj Piotrze, Co za niespodzianka! 😊 Otóż faktycznie INVITE zawiera: v=0 o=BroadWorks 210259115 1 IN IP4 xxx.xxx.xxx.xxx s=- c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 62810 RTP/AVP 8 18 101 a=fmtp:18 annexb=no a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 Zaś SIP/2.0 200 OK zawiera: v=0 o=FreeSWITCH 1606729402 1606729403 IN IP4 xxx.xxx.xxx.xxx s=FreeSWITCH c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 20616 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Gateway to ORANGE, także mam tu niewielkie pole manewru. Powiedzieli mi jednak, że jeśli usunę a=rtpmap:101 telephone-event/8000, to będzie inband – co jest możliwe, bo w tym połączeniu proponują kodek G729 a ja im na to PCMA no i transmisja jest w PCMA (G711A). Także pozostaje pytanie, jak usunąć tę linię z sekcji SDP wysyłanej przez FS w 200 OK. Serdecznie pozdrawiam, Adrian. De : FreeSWITCH-users > De la part de Piotr Gregor Envoyé : lundi 30 novembre 2020 11:58 À : FreeSWITCH Users Help > Objet : Re: [Freeswitch-users] Force Inband DTMF Reception Cześć Adrian, Tryb w którym FS będzie używał DTMF zależy od trybu który jest oferowany w SDP INVITE. Zazwyczaj klient jest konfigurowalny, np. w Bria: Preferences -> Calls -> DTMF: Send in-band. Tak więc upewnij się, że to nie klient żąda RFC 2833. pozdrawiam serdecznie, Piotr [https://drive.google.com/a/dataandsignal.com/uc?id=1lUdzdLjEycxXcGhDz-bKqDX8-AIlZuCq&export=download] Piotr Gregor Software Engineer M: (+44) 07483 866 525 L: (+44) 01256 597 470 www: dataandsignal.com On Thu, Nov 26, 2020 at 6:03 PM Adrian Worutowicz > wrote: I tried in the SIP profile: And in dialplan default.xml / and also I tried > wrote: Hello, For incoming calls, I want to receive DTMF inband. FS receives INVITE and at certain stage sends 200 OK with a SDP section, which always contains : a=rtpmap:101 telephone-event/8000 (what means ‘DTMF in RTP metadata’). Indeed the DTMFs come via RTP. I tried various FS configuration options to remove that line, without success. How to do it? How to tell a gateway to send DTMFs inband ? In advance thank you very, very much, Adrian. _________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com _________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From ceo at teo-en-ming.com Tue Dec 1 15:03:31 2020 From: ceo at teo-en-ming.com (Turritopsis Dohrnii Teo En Ming) Date: Tue, 01 Dec 2020 23:03:31 +0800 Subject: [Freeswitch-users] How to DIY/Setup An Open Source IP PBX Appliance/Server? Message-ID: <8a69f6724de75116bec8413d61251fa0@teo-en-ming.com> Subject: How to DIY/Setup An Open Source IP PBX Appliance/Server? Good day from Singapore, After reading recent reviews, I gather that Asterisk is the gold standard when it comes to open source VoIP systems and it is the most famous open source PBX out there. Article: Compare the Top 10 Best Open Source PBX Software of 2020 Link: https://www.voipreview.org/business-voip/best-open-source-pbx-software Article: Top 10 Free Open Source PBX Software Solutions Link: https://getvoip.com/blog/2016/09/23/best-open-source-pbx-software/ The following is an excerpt from Wikipedia: "Asterisk is a core component in many commercial products and open-source projects. Some of the commercial products are hardware and software bundles, for which the manufacturer supports and releases the software with an open-source distribution model. AskoziaPBX, a fork of the m0n0wall project, uses Asterisk PBX software to realize all telephony functions. AstLinux is a "Network Appliance for Communications" open-source software distribution.[15] FreePBX, an open-source graphical user interface, bundles Asterisk as the core of its FreePBX Distro[16] LinuxMCE bundles Asterisk to provide telephony; there is also an embedded version of Asterisk for OpenWrt routers. PBX in a Flash/Incredible PBX and trixbox are software PBXes based on Asterisk. Elastix previously used Asterisk, HylaFAX, Openfire and Postfix to offer PBX, fax, instant messaging and email functions, respectively, before switching to 3CX. Issabel is an open-source Unified Communications software which uses Asterisk for telephony functions. It was forked from the open-source versions of Elastix when 3CX acquired it. *astTECS uses Asterisk in its VoIP and mobile gateways." Link: https://en.wikipedia.org/wiki/Asterisk_(PBX) I would like to DIY/setup an IP PBX appliance/server using free open source projects. Which free open source project, mentioned in the list and links above, would you recommend to DIY my IP PBX appliance/server? Should I buy a desktop computer or get one of those appliances listed in the link below to serve as my IP PBX appliance/server? Link: https://www.lazada.sg/products/pfsense-iron-metal-case-fanless-intel-celeron-j1800-dual-core-mini-pc-firewall-soft-router-with-ddr3l-msata-ssd-4-gigabit-lan-rj45-com-port-i449270007-s1196780479.html?spm=a2o42.searchlist.list.89.100857d22PjCYx&search=1 Please also refer me to very good, detailed and well explained guides/tutorials/manuals on setting up open source IP PBX appliances/servers. Lastly, please recommend a cheap and affordable IP phone (suggest brand and model) to go along with my DIY open source IP PBX appliance/server. Mr. Turritopsis Dohrnii Teo En Ming, 42 years as of 1st December 2020 Tuesday, is a TARGETED INDIVIDUAL (TI) living in Singapore. Thank you very much. -----BEGIN EMAIL SIGNATURE----- The Gospel for all Targeted Individuals (TIs): [The New York Times] Microwave Weapons Are Prime Suspect in Ills of U.S. Embassy Workers Link: https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html ******************************************************************************************** Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's Academic Qualifications as at 14 Feb 2019 and refugee seeking attempts at the United Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug 2019) and Australia (25 Dec 2019 to 9 Jan 2020): [1] https://tdtemcerts.wordpress.com/ [2] https://tdtemcerts.blogspot.sg/ [3] https://www.scribd.com/user/270125049/Teo-En-Ming -----END EMAIL SIGNATURE----- From hamid2kviii at hotmail.com Thu Dec 3 08:32:44 2020 From: hamid2kviii at hotmail.com (Hamid Hashmi) Date: Thu, 3 Dec 2020 08:32:44 +0000 Subject: [Freeswitch-users] How to relay DTMF during Early Media (Before Call Answered) ? Message-ID: Hello I have been trying to relay DTMF in Early Media (Before Call Answered), I have tried the following option. (Ref) but it didn't generate any DTMF in early media and it caused an issue of Robotic sound in transcoding from AMR-WB to G729 as well. OS: DISTRIB_DESCRIPTION="Ubuntu 18.04.2 LTS" NAME="Ubuntu" VERSION="18.04.2 LTS (Bionic Beaver)" Container: PRETTY_NAME="Debian GNU/Linux 8 (jessie)" NAME="Debian GNU/Linux" VERSION_ID="8" VERSION="8 (jessie)" Freeswitch: FreeSWITCH Version 1.9.0-n20180314T133342Z-1~jessie+1+git~20180305T173609Z~dd0bb0e331~64bit (-n20180314T133342Z-1~jessie+1git dd0bb0e 2018-03-05 17:36:09Z 64bit) PS: DTMF is working fine once the call is answered. Regards Hamid R. Hashmi -------------- next part -------------- An HTML attachment was scrubbed... URL: From larry.hemenway at gmail.com Tue Dec 1 18:12:34 2020 From: larry.hemenway at gmail.com (Larry Hemenway) Date: Tue, 1 Dec 2020 12:12:34 -0600 Subject: [Freeswitch-users] 20ms silent RTP inserted into conference call output (about 1 out of every 100 packets) Message-ID: We're setting up a conference bridge as follows: (canned wav file) --> chrome --(g722)--> FreeSWITCH conference --(linear PCM)--> c++ app What we are seeing is that FreeSWITCH is inserting 20 ms of silence into the linear PCM output. It does not drop any audio - it is just inserting extra silence. On some traces we see ~1 out of every 100 output audio packet as silence. There is no regular cadence for the insertion. We are able to verify this in two ways: 1. When we look at the linear PCM tcpdump trace, we can identify silence because it's all 0xff. 2. We convert both the g722 and linear PCM into wav files. This is how we observe that no audio is dropped, just 20ms of silence added. The linear PCM wav file adds 20ms of audio and falls behind the g722 audio. We look at the pacing from the tcpdump and chrome is pacing every 20ms. FreeSWITCH, likewise, is also producing at 20ms intervals. We did try turning on the jitter buffer in FreeSWITCH and now see mitigated packets instead of silence, so that did not fix the issue. We'll probably turn off PLC next as mitigated packets are not easily identifiable. After reading https://github.com/signalwire/freeswitch/issues/735 I went back and looked at an old trace and it does appear that we see silence in the output linear PCM ~50-70ms after getting an RTCP packet from chrome. It's not 1:1; in the trace I looked at we had 44 rtcp packets, but only 20 silent packets, but for every silent packet we see an RTCP packet 50-70ms earlier. I hope to get more traces to verify this after turning PLC off. Environment: FreeSWITCH version 1.10.3, running inside a container on bare metal. This is reproducible running a single conference with two participants - no loading. We are turning off encryption in Chrome so we can decode the g722 (no dtls/srtp) and feeding chrome a canned wav file, so we can compare audio. Has anyone else seen this issue? Any suggestions on how to start debugging? I'm trying to trace through read_rtp_packets() in switch_rtp.c, but am a bit lost and not sure if that is where I should start. This issue is severely degrading our audio quality measurements so its causing a lot of headaches for us at the moment. From Alexander.Haugg at c4b.de Wed Dec 2 14:06:19 2020 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Wed, 2 Dec 2020 14:06:19 +0000 Subject: [Freeswitch-users] Full Trickle ICE via mod_sofia supported? Message-ID: <11ad1ccd13d34d7cbc22a8a293266987@c4b.de> Hello, I only mean that the client can deliver candidates. The Freeswitch of course sends all candidates in the first SDP. Currently I'm trying to implement the Full Trickle mechanism as described here https://tools.ietf.org/id/draft-ietf-mmusic-trickle-ice-sip-11.html Candidates should be forwarded via SIP Info during the connection setup. Since I haven't succeeded so far, I have the following questions. Is Full Trickle supported via mod_sofia? If so, which settings or sip headers are required? Thanks Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.org Thu Dec 3 22:33:13 2020 From: mike at freeswitch.org (Mike Jerris) Date: Thu, 3 Dec 2020 15:33:13 -0700 Subject: [Freeswitch-users] Full Trickle ICE via mod_sofia supported? In-Reply-To: References: Message-ID: <6090BBFC-7FC2-4AD2-AD38-B2F34A363D7B@freeswitch.org> > On Nov 30, 2020, at 7:56 AM, Alexander Haugg wrote: > Currently I'm trying to implement the Full Trickle mechanism as described here https://tools.ietf.org/id/draft-ietf-mmusic-trickle-ice-sip-11.html > Since I haven't succeeded so far, I have the following questions. > Is Full Trickle supported via mod_sofia? No -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.org Thu Dec 3 22:35:56 2020 From: mike at freeswitch.org (Mike Jerris) Date: Thu, 3 Dec 2020 15:35:56 -0700 Subject: [Freeswitch-users] Masking caller In-Reply-To: <20201127174456.001bb045@pc.home.lkiesow.io> References: <20201127174456.001bb045@pc.home.lkiesow.io> Message-ID: > On Nov 27, 2020, at 9:44 AM, Lars Kiesow wrote: > > Hi everyone, > I'm trying to mask the caller_id_name in a FreeSWITCH dialplan to > prevent the real phone numbers to show up in our conferencing software. > Someone sent me the following lines: > > > > > While this works perfectly and does exactly what I want, I'm unsure > about potential security risks. Its a good thing to be concerned with, yes thats real > > The caller_id_name ends up in a shell command after all and I'm > wondering if someone could send a name like `; rm /*` (you get the > idea). > > Is this safe? Is the caller_id_name sanitized? Is there a better way to > do something like this? > No not safe. Check out https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+regex > Best regards, > Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.org Thu Dec 3 22:36:37 2020 From: mike at freeswitch.org (Mike Jerris) Date: Thu, 3 Dec 2020 15:36:37 -0700 Subject: [Freeswitch-users] Help with tls In-Reply-To: References: Message-ID: > On Nov 28, 2020, at 4:09 PM, Dr. Ogg wrote: > > > I have been looking for some good documentation regarding the configuration of TLS with FreeSWITCH. If anyone has a good resource, I would be grateful https://freeswitch.org/confluence/display/FREESWITCH/SIP+TLS -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.org Thu Dec 3 22:38:40 2020 From: mike at freeswitch.org (Mike Jerris) Date: Thu, 3 Dec 2020 15:38:40 -0700 Subject: [Freeswitch-users] Force Inband DTMF Reception (SOLVED) In-Reply-To: References: Message-ID: <58BB7770-71E6-4738-8789-30735E7A63D5@freeswitch.org> If a device is capable of doing rfc2833 dtmf. You should never want to fall back to inland, it WILL be less reliable > On Dec 3, 2020, at 4:32 AM, Adrian Worutowicz wrote: > > Just for the records : > > In order to receive DTMS in-band for incoming calls, FS must not send in SDP : > > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > This is achieved by : > > In vars.xml : > > > In sip_profiles\internal.xml and/or external.xml (Settings): > > Add line : > > > Remove line (if exists) : > > > Big thanks to @seven1240 ! > > > De : FreeSWITCH-users > De la part de Piotr Gregor > Envoyé : lundi 30 novembre 2020 16:58 > À : FreeSWITCH Users Help > > Objet : Re: [Freeswitch-users] Force Inband DTMF Reception > > Cześć Adrian, > > Tak, to ma sens. Spróbuj może ustawić flagę dtmf_type=none > > > > To powinno sprawić, że w SDP ANSWER nie będzie a=rtpmap:101 telephone-event/8000, tylko nie jestem pewien czy detekcja in-band DTMF będzie włączona. > > pozdrawiam, > Piotr > > > On Mon, 30 Nov 2020 at 15:34, Adrian Worutowicz > wrote: > Witaj Piotrze, > > Co za niespodzianka! 😊 > > Otóż faktycznie INVITE zawiera: > v=0 > o=BroadWorks 210259115 1 IN IP4 xxx.xxx.xxx.xxx > s=- > c=IN IP4 xxx.xxx.xxx.xxx > t=0 0 > m=audio 62810 RTP/AVP 8 18 101 > a=fmtp:18 annexb=no > a=rtpmap:18 G729/8000 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > > > Zaś SIP/2.0 200 OK zawiera: > v=0 > o=FreeSWITCH 1606729402 1606729403 IN IP4 xxx.xxx.xxx.xxx > s=FreeSWITCH > c=IN IP4 xxx.xxx.xxx.xxx > t=0 0 > m=audio 20616 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > > Gateway to ORANGE, także mam tu niewielkie pole manewru. > Powiedzieli mi jednak, że jeśli usunę a=rtpmap:101 telephone-event/8000, to będzie inband – co jest możliwe, bo w tym połączeniu proponują kodek G729 a ja im na to PCMA no i transmisja jest w PCMA (G711A). > > Także pozostaje pytanie, jak usunąć tę linię z sekcji SDP wysyłanej przez FS w 200 OK. > > Serdecznie pozdrawiam, > Adrian. > > > > De : FreeSWITCH-users > De la part de Piotr Gregor > Envoyé : lundi 30 novembre 2020 11:58 > À : FreeSWITCH Users Help > > Objet : Re: [Freeswitch-users] Force Inband DTMF Reception > > Cześć Adrian, > > Tryb w którym FS będzie używał DTMF zależy od trybu który jest oferowany w SDP INVITE. > Zazwyczaj klient jest konfigurowalny, np. w Bria: Preferences -> Calls -> DTMF: Send in-band. > Tak więc upewnij się, że to nie klient żąda RFC 2833. > > pozdrawiam serdecznie, > Piotr > > > Piotr Gregor > Software Engineer > > M: (+44) 07483 866 525 L: (+44) 01256 597 470 www: dataandsignal.com > > > > > > On Thu, Nov 26, 2020 at 6:03 PM Adrian Worutowicz > wrote: > I tried in the SIP profile: > > And in dialplan default.xml / > and also I tried > On Thu, Nov 26, 2020 at 5:59 PM Adrian Worutowicz > wrote: > Hello, > > For incoming calls, I want to receive DTMF inband. > > FS receives INVITE and at certain stage sends 200 OK with a SDP section, which always contains : > > a=rtpmap:101 telephone-event/8000 (what means ‘DTMF in RTP metadata’). Indeed the DTMFs come via RTP. > > I tried various FS configuration options to remove that line, without success. > > How to do it? How to tell a gateway to send DTMFs inband ? > > In advance thank you very, very much, > Adrian. > > > > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From Alexander.Haugg at c4b.de Fri Dec 4 06:47:40 2020 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Fri, 4 Dec 2020 06:47:40 +0000 Subject: [Freeswitch-users] Full Trickle ICE via mod_sofia supported? In-Reply-To: <6090BBFC-7FC2-4AD2-AD38-B2F34A363D7B@freeswitch.org> References: <6090BBFC-7FC2-4AD2-AD38-B2F34A363D7B@freeswitch.org> Message-ID: <5025d661c27446c4a8b7c34e360b9aa0@c4b.de> OK, is it supported via mod_verto? Or is it possible to add candidates with CLI commands? Von: FreeSWITCH-users Im Auftrag von Mike Jerris Gesendet: Donnerstag, 3. Dezember 2020 23:33 An: FreeSWITCH Users Help Cc: Stefan Dietrich Betreff: Re: [Freeswitch-users] Full Trickle ICE via mod_sofia supported? On Nov 30, 2020, at 7:56 AM, Alexander Haugg > wrote: Currently I'm trying to implement the Full Trickle mechanism as described here https://tools.ietf.org/id/draft-ietf-mmusic-trickle-ice-sip-11.html Since I haven't succeeded so far, I have the following questions. Is Full Trickle supported via mod_sofia? No -------------- next part -------------- An HTML attachment was scrubbed... URL: From muintelserver at gmail.com Fri Dec 4 15:15:30 2020 From: muintelserver at gmail.com (Md Ahammad Muin) Date: Fri, 4 Dec 2020 21:15:30 +0600 Subject: [Freeswitch-users] please give user list Message-ID: hello sir please give me a user list -------------- next part -------------- An HTML attachment was scrubbed... URL: From Jazmin.Marino at on24.com Fri Dec 4 17:24:24 2020 From: Jazmin.Marino at on24.com (Jazmin Marina Florez Marino) Date: Fri, 4 Dec 2020 17:24:24 +0000 Subject: [Freeswitch-users] HELP DTLS HANDSHAKE to FAIL chromeV87 Message-ID: Hi I have a verto client trying to connect to FS using Chrome V87 but we have DTLS problems. The openssl version we are using on FS side is OpenSSL 1.0.2k-fips FreeSWITCH (Version 1.8.4 64bit) FS LOG - Chrome V87 FAILED CONNECTION 2020-12-03 11:23:06.762291 [ERR] switch_rtp.c:3199 video Handshake failure 1 2020-12-03 11:23:06.762291 [INFO] switch_rtp.c:3200 Changing video DTLS state from HANDSHAKE to FAIL 2020-12-03 11:23:06.832295 [ERR] switch_rtp.c:3199 audio Handshake failure 1 2020-12-03 11:23:06.832295 [INFO] switch_rtp.c:3200 Changing audio DTLS state from HANDSHAKE to FAIL 2020-12-03 11:23:06.852295 [NOTICE] switch_rtp.c:3181 Hangup verto.rtc/3520 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2020-12-03 11:23:06.852295 [DEBUG] switch_core_media.c:7470 verto.rtc/3520 Video thread ended 2020-12-03 11:23:06.872305 [INFO] conference_loop.c:1670 Channel leaving conference, cause: DESTINATION_OUT_OF_ORDER 2020-12-03 11:23:06.872305 [DEBUG] switch_core_session.c:2639 verto.rtc/3520 skip receive message [DISPLAY] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] mod_conference.c:2467 verto.rtc/3520 skip receive message [TRANSFER] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_media.c:12220 verto.rtc/3520 skip receive message [BITRATE_REQ] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_codec.c:248 verto.rtc/3520 Restore previous codec opus:116. 2020-12-03 11:23:06.872305 [DEBUG] switch_core_session.c:2886 verto.rtc/3520 skip receive message [PHONE_EVENT] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:650 (verto.rtc/3520) State EXECUTE going to sleep 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:584 (verto.rtc/3520) Running State Change CS_HANGUP (Cur 1 Tot 3) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:847 (verto.rtc/3520) Callstate Change ACTIVE -> HANGUP 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:849 (verto.rtc/3520) State HANGUP 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:60 verto.rtc/3520 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:849 (verto.rtc/3520) State HANGUP going to sleep The weird thing is the connection works ok when the version of chrome is V86. Some weeks ago chrome was updated and we detected this issue. FS LOG - ChromeV86 CONNECTION OK 2020-12-03 11:19:19.622294 [INFO] switch_rtp.c:3206 Changing video DTLS state from HANDSHAKE to SETUP 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:3113 video Fingerprint Verified. 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:4104 Activating video Secure RTP SEND 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:4082 Activating video Secure RTP RECV 2020-12-03 11:19:19.672317 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:3155 Changing video DTLS state from SETUP to READY 2020-12-03 11:19:19.672317 [DEBUG] switch_rtp.c:1890 rtcp_stats_init: video ssrc[2719546543] base_seq[1999] 2020-12-03 11:19:19.672317 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3206 Changing audio DTLS state from HANDSHAKE to SETUP 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3113 audio Fingerprint Verified. 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:4104 Activating audio Secure RTP SEND 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:4082 Activating audio Secure RTP RECV 2020-12-03 11:19:19.702298 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3155 Changing audio DTLS state from SETUP to READY 2020-12-03 11:19:19.702298 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 -------------- next part -------------- An HTML attachment was scrubbed... URL: From kaduww at gmail.com Fri Dec 4 18:39:25 2020 From: kaduww at gmail.com (Carlos Eduardo) Date: Fri, 4 Dec 2020 15:39:25 -0300 Subject: [Freeswitch-users] HELP DTLS HANDSHAKE to FAIL chromeV87 In-Reply-To: References: Message-ID: Update to 1.10.5... I had the same problem and worked fine with the lastest Em sex., 4 de dez. de 2020 às 15:09, Jazmin Marina Florez Marino < Jazmin.Marino at on24.com> escreveu: > Hi > > I have a verto client trying to connect to FS using Chrome V87 but we have > DTLS problems. > > The openssl version we are using on FS side is OpenSSL 1.0.2k-fips > > FreeSWITCH (Version 1.8.4 64bit) > > > > > > FS LOG - Chrome V87 FAILED CONNECTION > > > > 2020-12-03 11:23:06.762291 [ERR] switch_rtp.c:3199 video Handshake failure > 1 > > 2020-12-03 11:23:06.762291 [INFO] switch_rtp.c:3200 Changing video DTLS > state from HANDSHAKE to FAIL > > 2020-12-03 11:23:06.832295 [ERR] switch_rtp.c:3199 audio Handshake failure > 1 > > 2020-12-03 11:23:06.832295 [INFO] switch_rtp.c:3200 Changing audio DTLS > state from HANDSHAKE to FAIL > > 2020-12-03 11:23:06.852295 [NOTICE] switch_rtp.c:3181 Hangup > verto.rtc/3520 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] > > 2020-12-03 11:23:06.852295 [DEBUG] switch_core_media.c:7470 verto.rtc/3520 > Video thread ended > > 2020-12-03 11:23:06.872305 [INFO] conference_loop.c:1670 Channel leaving > conference, cause: DESTINATION_OUT_OF_ORDER > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_session.c:2639 > verto.rtc/3520 skip receive message [DISPLAY] (channel is hungup already) > > 2020-12-03 11:23:06.872305 [DEBUG] mod_conference.c:2467 verto.rtc/3520 > skip receive message [TRANSFER] (channel is hungup already) > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_media.c:12220 > verto.rtc/3520 skip receive message [BITRATE_REQ] (channel is hungup > already) > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_codec.c:248 verto.rtc/3520 > Restore previous codec opus:116. > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_session.c:2886 > verto.rtc/3520 skip receive message [PHONE_EVENT] (channel is hungup > already) > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:650 > (verto.rtc/3520) State EXECUTE going to sleep > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:584 > (verto.rtc/3520) Running State Change CS_HANGUP (Cur 1 Tot 3) > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:847 > (verto.rtc/3520) Callstate Change ACTIVE -> HANGUP > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:849 > (verto.rtc/3520) State HANGUP > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:60 > verto.rtc/3520 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:849 > (verto.rtc/3520) State HANGUP going to sleep > > > > The weird thing is the connection works ok when the version of chrome is > V86. Some weeks ago chrome was updated and we detected this issue. > > > > FS LOG - ChromeV86 CONNECTION OK > > > > 2020-12-03 11:19:19.622294 [INFO] switch_rtp.c:3206 Changing video DTLS > state from HANDSHAKE to SETUP > > 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:3113 video Fingerprint > Verified. > > 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:4104 Activating video > Secure RTP SEND > > 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:4082 Activating video > Secure RTP RECV > > 2020-12-03 11:19:19.672317 [DEBUG] switch_core_sqldb.c:2617 Secure Type: > srtp:dtls:AES_CM_128_HMAC_SHA1_80 > > 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:3155 Changing video DTLS > state from SETUP to READY > > 2020-12-03 11:19:19.672317 [DEBUG] switch_rtp.c:1890 rtcp_stats_init: > video ssrc[2719546543] base_seq[1999] > > 2020-12-03 11:19:19.672317 [DEBUG] switch_core_sqldb.c:2617 Secure Type: > srtp:dtls:AES_CM_128_HMAC_SHA1_80 > > 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3206 Changing audio DTLS > state from HANDSHAKE to SETUP > > 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3113 audio Fingerprint > Verified. > > 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:4104 Activating audio > Secure RTP SEND > > 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:4082 Activating audio > Secure RTP RECV > > 2020-12-03 11:19:19.702298 [DEBUG] switch_core_sqldb.c:2617 Secure Type: > srtp:dtls:AES_CM_128_HMAC_SHA1_80 > > 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3155 Changing audio DTLS > state from SETUP to READY > > 2020-12-03 11:19:19.702298 [DEBUG] switch_core_sqldb.c:2617 Secure Type: > srtp:dtls:AES_CM_128_HMAC_SHA1_80 > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- *Carlos E. Wagner* *Tecnólogo em Telecomunicações, Opensips Certified Professional* *Fone: +55 48 99981-0894* *E-mail:* kaduww at gmail.com *LinkedIn:* https://www.linkedin.com/in/carlos-eduardo-wagner-96bbb433/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From guenther.eberl at besharp.at Fri Dec 4 18:19:08 2020 From: guenther.eberl at besharp.at (Eberl Guenther) Date: Fri, 4 Dec 2020 18:19:08 +0000 Subject: [Freeswitch-users] HELP DTLS HANDSHAKE to FAIL chromeV87 In-Reply-To: References: Message-ID: Hi, I guess it is the freeswitch bug were TLS V 1.0 for audio streams is hardcoded in older versions. You should see it on your data traces on the audio ports. Bye Gunther Von: FreeSWITCH-users Im Auftrag von Jazmin Marina Florez Marino Gesendet: Freitag, 4. Dezember 2020 18:24 An: FreeSWITCH Users Help ; freeswitch-dev at lists.freeswitch.org; freeswitch-users-request at lists.freeswitch.org Betreff: [Freeswitch-users] HELP DTLS HANDSHAKE to FAIL chromeV87 Hi I have a verto client trying to connect to FS using Chrome V87 but we have DTLS problems. The openssl version we are using on FS side is OpenSSL 1.0.2k-fips FreeSWITCH (Version 1.8.4 64bit) FS LOG - Chrome V87 FAILED CONNECTION 2020-12-03 11:23:06.762291 [ERR] switch_rtp.c:3199 video Handshake failure 1 2020-12-03 11:23:06.762291 [INFO] switch_rtp.c:3200 Changing video DTLS state from HANDSHAKE to FAIL 2020-12-03 11:23:06.832295 [ERR] switch_rtp.c:3199 audio Handshake failure 1 2020-12-03 11:23:06.832295 [INFO] switch_rtp.c:3200 Changing audio DTLS state from HANDSHAKE to FAIL 2020-12-03 11:23:06.852295 [NOTICE] switch_rtp.c:3181 Hangup verto.rtc/3520 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2020-12-03 11:23:06.852295 [DEBUG] switch_core_media.c:7470 verto.rtc/3520 Video thread ended 2020-12-03 11:23:06.872305 [INFO] conference_loop.c:1670 Channel leaving conference, cause: DESTINATION_OUT_OF_ORDER 2020-12-03 11:23:06.872305 [DEBUG] switch_core_session.c:2639 verto.rtc/3520 skip receive message [DISPLAY] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] mod_conference.c:2467 verto.rtc/3520 skip receive message [TRANSFER] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_media.c:12220 verto.rtc/3520 skip receive message [BITRATE_REQ] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_codec.c:248 verto.rtc/3520 Restore previous codec opus:116. 2020-12-03 11:23:06.872305 [DEBUG] switch_core_session.c:2886 verto.rtc/3520 skip receive message [PHONE_EVENT] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:650 (verto.rtc/3520) State EXECUTE going to sleep 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:584 (verto.rtc/3520) Running State Change CS_HANGUP (Cur 1 Tot 3) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:847 (verto.rtc/3520) Callstate Change ACTIVE -> HANGUP 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:849 (verto.rtc/3520) State HANGUP 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:60 verto.rtc/3520 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:849 (verto.rtc/3520) State HANGUP going to sleep The weird thing is the connection works ok when the version of chrome is V86. Some weeks ago chrome was updated and we detected this issue. FS LOG - ChromeV86 CONNECTION OK 2020-12-03 11:19:19.622294 [INFO] switch_rtp.c:3206 Changing video DTLS state from HANDSHAKE to SETUP 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:3113 video Fingerprint Verified. 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:4104 Activating video Secure RTP SEND 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:4082 Activating video Secure RTP RECV 2020-12-03 11:19:19.672317 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:3155 Changing video DTLS state from SETUP to READY 2020-12-03 11:19:19.672317 [DEBUG] switch_rtp.c:1890 rtcp_stats_init: video ssrc[2719546543] base_seq[1999] 2020-12-03 11:19:19.672317 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3206 Changing audio DTLS state from HANDSHAKE to SETUP 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3113 audio Fingerprint Verified. 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:4104 Activating audio Secure RTP SEND 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:4082 Activating audio Secure RTP RECV 2020-12-03 11:19:19.702298 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3155 Changing audio DTLS state from SETUP to READY 2020-12-03 11:19:19.702298 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 -------------- next part -------------- An HTML attachment was scrubbed... URL: From mrjoli021 at gmail.com Fri Dec 4 19:48:01 2020 From: mrjoli021 at gmail.com (Joli Martinez) Date: Fri, 4 Dec 2020 14:48:01 -0500 Subject: [Freeswitch-users] freeswitch won't open file Message-ID: Hello, I have the following function that works normally on lua, but when I copied and pasted it into my freeswitch lua script, the file wont open. I am trying to put the passwords I use in a json format outside the script, so that I can share the script. I have installed the correct libraries for lunajson to work. Any suggestions? --Reads json file and parses it function returnJson(option) freeswitch.consoleLog("INFO", "Inside function\n") local file = io.open("password.json", "r") if not file then return nil end local jsonString = file:read("*a") freeswitch.consoleLog("INFO", "File is open\n") file:close() jsonResult = lunajson.decode( jsonString ) freeswitch.consoleLog("INFO", "Parsing json\n") return ( jsonResult[option] ) end --SID = returnJson("SID") --Token = returnJson("TOKEN") ------------------------------------------------------------------------------------------------------------ 2020-12-04 14:38:13.637108 [ERR] mod_lua.cpp:203 /usr/share/freeswitch/scripts/test.lua:67: attempt to concatenate global 'SID' (a nil value) stack traceback: /usr/share/freeswitch/scripts/test.lua:67: in main chunk -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Fri Dec 4 21:35:20 2020 From: brian at freeswitch.com (Brian West) Date: Fri, 4 Dec 2020 15:35:20 -0600 Subject: [Freeswitch-users] freeswitch won't open file In-Reply-To: References: Message-ID: check permissions of the file. On Fri, Dec 4, 2020 at 2:36 PM Joli Martinez wrote: > Hello, > > I have the following function that works normally on lua, but when I > copied and pasted it into my freeswitch lua script, the file wont open. I > am trying to put the passwords I use in a json format outside the script, > so that I can share the script. I have installed the correct libraries for > lunajson to work. > Any suggestions? > > > --Reads json file and parses it > function returnJson(option) > freeswitch.consoleLog("INFO", "Inside function\n") > > local file = io.open("password.json", "r") > if not file then > return nil > end > local jsonString = file:read("*a") > > freeswitch.consoleLog("INFO", "File is open\n") > file:close() > jsonResult = lunajson.decode( jsonString ) > freeswitch.consoleLog("INFO", "Parsing json\n") > return ( jsonResult[option] ) > end > > --SID = returnJson("SID") > --Token = returnJson("TOKEN") > > ------------------------------------------------------------------------------------------------------------ > > 2020-12-04 14:38:13.637108 [ERR] mod_lua.cpp:203 > /usr/share/freeswitch/scripts/test.lua:67: attempt to concatenate global > 'SID' (a nil value) > > stack traceback: > > /usr/share/freeswitch/scripts/test.lua:67: in main chunk > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From ciprian.dosoftei at gmail.com Fri Dec 4 21:45:08 2020 From: ciprian.dosoftei at gmail.com (Ciprian Dosoftei) Date: Fri, 4 Dec 2020 16:45:08 -0500 Subject: [Freeswitch-users] HELP DTLS HANDSHAKE to FAIL chromeV87 In-Reply-To: References: Message-ID: Check the length of the dtls-srtp.pem key, if its length is short (< 4096) you can run into handshake issues. You can also remove the pem file and allow FreeSWITCH to create a fresh one upon restart. On Fri, 4 Dec 2020 at 14:36, Eberl Guenther wrote: > Hi, > > > > I guess it is the freeswitch bug were TLS V 1.0 for audio streams is > hardcoded in older versions. > > > > You should see it on your data traces on the audio ports. > > > > Bye > > Gunther > > > > > > *Von:* FreeSWITCH-users *Im > Auftrag von *Jazmin Marina Florez Marino > *Gesendet:* Freitag, 4. Dezember 2020 18:24 > *An:* FreeSWITCH Users Help ; > freeswitch-dev at lists.freeswitch.org; > freeswitch-users-request at lists.freeswitch.org > *Betreff:* [Freeswitch-users] HELP DTLS HANDSHAKE to FAIL chromeV87 > > > > Hi > > I have a verto client trying to connect to FS using Chrome V87 but we have > DTLS problems. > > The openssl version we are using on FS side is OpenSSL 1.0.2k-fips > > FreeSWITCH (Version 1.8.4 64bit) > > > > > > FS LOG - Chrome V87 FAILED CONNECTION > > > > 2020-12-03 11:23:06.762291 [ERR] switch_rtp.c:3199 video Handshake failure > 1 > > 2020-12-03 11:23:06.762291 [INFO] switch_rtp.c:3200 Changing video DTLS > state from HANDSHAKE to FAIL > > 2020-12-03 11:23:06.832295 [ERR] switch_rtp.c:3199 audio Handshake failure > 1 > > 2020-12-03 11:23:06.832295 [INFO] switch_rtp.c:3200 Changing audio DTLS > state from HANDSHAKE to FAIL > > 2020-12-03 11:23:06.852295 [NOTICE] switch_rtp.c:3181 Hangup > verto.rtc/3520 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] > > 2020-12-03 11:23:06.852295 [DEBUG] switch_core_media.c:7470 verto.rtc/3520 > Video thread ended > > 2020-12-03 11:23:06.872305 [INFO] conference_loop.c:1670 Channel leaving > conference, cause: DESTINATION_OUT_OF_ORDER > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_session.c:2639 > verto.rtc/3520 skip receive message [DISPLAY] (channel is hungup already) > > 2020-12-03 11:23:06.872305 [DEBUG] mod_conference.c:2467 verto.rtc/3520 > skip receive message [TRANSFER] (channel is hungup already) > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_media.c:12220 > verto.rtc/3520 skip receive message [BITRATE_REQ] (channel is hungup > already) > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_codec.c:248 verto.rtc/3520 > Restore previous codec opus:116. > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_session.c:2886 > verto.rtc/3520 skip receive message [PHONE_EVENT] (channel is hungup > already) > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:650 > (verto.rtc/3520) State EXECUTE going to sleep > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:584 > (verto.rtc/3520) Running State Change CS_HANGUP (Cur 1 Tot 3) > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:847 > (verto.rtc/3520) Callstate Change ACTIVE -> HANGUP > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:849 > (verto.rtc/3520) State HANGUP > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:60 > verto.rtc/3520 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:849 > (verto.rtc/3520) State HANGUP going to sleep > > > > The weird thing is the connection works ok when the version of chrome is > V86. Some weeks ago chrome was updated and we detected this issue. > > > > FS LOG - ChromeV86 CONNECTION OK > > > > 2020-12-03 11:19:19.622294 [INFO] switch_rtp.c:3206 Changing video DTLS > state from HANDSHAKE to SETUP > > 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:3113 video Fingerprint > Verified. > > 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:4104 Activating video > Secure RTP SEND > > 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:4082 Activating video > Secure RTP RECV > > 2020-12-03 11:19:19.672317 [DEBUG] switch_core_sqldb.c:2617 Secure Type: > srtp:dtls:AES_CM_128_HMAC_SHA1_80 > > 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:3155 Changing video DTLS > state from SETUP to READY > > 2020-12-03 11:19:19.672317 [DEBUG] switch_rtp.c:1890 rtcp_stats_init: > video ssrc[2719546543] base_seq[1999] > > 2020-12-03 11:19:19.672317 [DEBUG] switch_core_sqldb.c:2617 Secure Type: > srtp:dtls:AES_CM_128_HMAC_SHA1_80 > > 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3206 Changing audio DTLS > state from HANDSHAKE to SETUP > > 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3113 audio Fingerprint > Verified. > > 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:4104 Activating audio > Secure RTP SEND > > 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:4082 Activating audio > Secure RTP RECV > > 2020-12-03 11:19:19.702298 [DEBUG] switch_core_sqldb.c:2617 Secure Type: > srtp:dtls:AES_CM_128_HMAC_SHA1_80 > > 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3155 Changing audio DTLS > state from SETUP to READY > > 2020-12-03 11:19:19.702298 [DEBUG] switch_core_sqldb.c:2617 Secure Type: > srtp:dtls:AES_CM_128_HMAC_SHA1_80 > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Best Regards, Ciprian Dosoftei The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message. Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender. -------------- next part -------------- An HTML attachment was scrubbed... URL: From mrjoli021 at gmail.com Fri Dec 4 22:41:19 2020 From: mrjoli021 at gmail.com (Joli Martinez) Date: Fri, 4 Dec 2020 17:41:19 -0500 Subject: [Freeswitch-users] freeswitch won't open file In-Reply-To: References: Message-ID: Hello, The file permissions are set to rw-r--r-- www-data:www-data same as every other file in the scripts directory. Thanks, On Fri, Dec 4, 2020 at 5:02 PM Brian West wrote: > check permissions of the file. > > On Fri, Dec 4, 2020 at 2:36 PM Joli Martinez wrote: > >> Hello, >> >> I have the following function that works normally on lua, but when I >> copied and pasted it into my freeswitch lua script, the file wont open. I >> am trying to put the passwords I use in a json format outside the script, >> so that I can share the script. I have installed the correct libraries for >> lunajson to work. >> Any suggestions? >> >> >> --Reads json file and parses it >> function returnJson(option) >> freeswitch.consoleLog("INFO", "Inside function\n") >> >> local file = io.open("password.json", "r") >> if not file then >> return nil >> end >> local jsonString = file:read("*a") >> >> freeswitch.consoleLog("INFO", "File is open\n") >> file:close() >> jsonResult = lunajson.decode( jsonString ) >> freeswitch.consoleLog("INFO", "Parsing json\n") >> return ( jsonResult[option] ) >> end >> >> --SID = returnJson("SID") >> --Token = returnJson("TOKEN") >> >> ------------------------------------------------------------------------------------------------------------ >> >> 2020-12-04 14:38:13.637108 [ERR] mod_lua.cpp:203 >> /usr/share/freeswitch/scripts/test.lua:67: attempt to concatenate global >> 'SID' (a nil value) >> >> stack traceback: >> >> /usr/share/freeswitch/scripts/test.lua:67: in main chunk >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/freeswitch] > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sat Dec 5 08:50:30 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 5 Dec 2020 08:50:30 +0000 Subject: [Freeswitch-users] freeswitch won't open file In-Reply-To: References: Message-ID: If you’re running fs as freeswitch, you need to give it permission to open those files. Something like chmod o+rw On Sat, 5 Dec 2020 at 00:22, Joli Martinez wrote: > Hello, > > The file permissions are set to rw-r--r-- www-data:www-data same as every > other file in the scripts directory. > > Thanks, > > On Fri, Dec 4, 2020 at 5:02 PM Brian West wrote: > >> check permissions of the file. >> >> On Fri, Dec 4, 2020 at 2:36 PM Joli Martinez wrote: >> >>> Hello, >>> >>> I have the following function that works normally on lua, but when I >>> copied and pasted it into my freeswitch lua script, the file wont open. I >>> am trying to put the passwords I use in a json format outside the script, >>> so that I can share the script. I have installed the correct libraries for >>> lunajson to work. >>> Any suggestions? >>> >>> >>> --Reads json file and parses it >>> function returnJson(option) >>> freeswitch.consoleLog("INFO", "Inside function\n") >>> >>> local file = io.open("password.json", "r") >>> if not file then >>> return nil >>> end >>> local jsonString = file:read("*a") >>> >>> freeswitch.consoleLog("INFO", "File is open\n") >>> file:close() >>> jsonResult = lunajson.decode( jsonString ) >>> freeswitch.consoleLog("INFO", "Parsing json\n") >>> return ( jsonResult[option] ) >>> end >>> >>> --SID = returnJson("SID") >>> --Token = returnJson("TOKEN") >>> >>> ------------------------------------------------------------------------------------------------------------ >>> >>> 2020-12-04 14:38:13.637108 [ERR] mod_lua.cpp:203 >>> /usr/share/freeswitch/scripts/test.lua:67: attempt to concatenate global >>> 'SID' (a nil value) >>> >>> stack traceback: >>> >>> /usr/share/freeswitch/scripts/test.lua:67: in main chunk >>> _________________________________________________________________________ >>> >>> 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 at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> >> >> -- >> >> Brian West | Co-founder and Developer >> >> Need Commercial support? email sales at freeswitch.com >> >> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >> >> >> Email: brian at freeswitch.com >> >> Mobile: 918-424-9378 >> >> Website: https://www.FreeSWITCH.com >> >> [image: https://www.facebook.com/signalwireinc?src=email] >> [image: >> https://twitter.com/freeswitch] >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From mrjoli021 at gmail.com Sat Dec 5 18:22:56 2020 From: mrjoli021 at gmail.com (Joli Martinez) Date: Sat, 5 Dec 2020 13:22:56 -0500 Subject: [Freeswitch-users] freeswitch won't open file In-Reply-To: References: Message-ID: Hello, It is not a permissions issue. I have changed the permissions to rw for everyone and it will still not read the file. The password.json file is in the same directory as the script /usr/share/freeswitch/script. I am using lua 5.2.3 and again, the same code works in my home directory. I just copied the password.json file over and pasted the function into my code. When I place the call freeswitch prints "Inside the function", so I know it gets there, but then freeswitch spits out "attempt to index local 'file' (a nil value)" What else could I be missing? ------------------------------- --Reads json file and parses it function returnJson(option) freeswitch.consoleLog("INFO", "Inside function\n") f = "password.json" local file = io.open(f, "r") if not file then return nil end local jsonString = file:read("*a") freeswitch.consoleLog("INFO", "File is open\n") local file:close() local jsonResult = lunajson.decode( jsonString ) freeswitch.consoleLog("INFO", "Parsing json\n") return ( jsonResult[option] ) end SID = returnJson("SID") Token = returnJson("TOKEN") Thanks, On Sat, Dec 5, 2020 at 4:19 AM David Villasmil < david.villasmil.work at gmail.com> wrote: > If you’re running fs as freeswitch, you need to give it permission to open > those files. Something like chmod o+rw > > On Sat, 5 Dec 2020 at 00:22, Joli Martinez wrote: > >> Hello, >> >> The file permissions are set to rw-r--r-- www-data:www-data same as every >> other file in the scripts directory. >> >> Thanks, >> >> On Fri, Dec 4, 2020 at 5:02 PM Brian West wrote: >> >>> check permissions of the file. >>> >>> On Fri, Dec 4, 2020 at 2:36 PM Joli Martinez >>> wrote: >>> >>>> Hello, >>>> >>>> I have the following function that works normally on lua, but when I >>>> copied and pasted it into my freeswitch lua script, the file wont open. I >>>> am trying to put the passwords I use in a json format outside the script, >>>> so that I can share the script. I have installed the correct libraries for >>>> lunajson to work. >>>> Any suggestions? >>>> >>>> >>>> --Reads json file and parses it >>>> function returnJson(option) >>>> freeswitch.consoleLog("INFO", "Inside function\n") >>>> >>>> local file = io.open("password.json", "r") >>>> if not file then >>>> return nil >>>> end >>>> local jsonString = file:read("*a") >>>> >>>> freeswitch.consoleLog("INFO", "File is open\n") >>>> file:close() >>>> jsonResult = lunajson.decode( jsonString ) >>>> freeswitch.consoleLog("INFO", "Parsing json\n") >>>> return ( jsonResult[option] ) >>>> end >>>> >>>> --SID = returnJson("SID") >>>> --Token = returnJson("TOKEN") >>>> >>>> ------------------------------------------------------------------------------------------------------------ >>>> >>>> 2020-12-04 14:38:13.637108 [ERR] mod_lua.cpp:203 >>>> /usr/share/freeswitch/scripts/test.lua:67: attempt to concatenate global >>>> 'SID' (a nil value) >>>> >>>> stack traceback: >>>> >>>> /usr/share/freeswitch/scripts/test.lua:67: in main chunk >>>> >>>> _________________________________________________________________________ >>>> >>>> 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 at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> >>> >>> -- >>> >>> Brian West | Co-founder and Developer >>> >>> Need Commercial support? email sales at freeswitch.com >>> >>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >>> >>> >>> Email: brian at freeswitch.com >>> >>> Mobile: 918-424-9378 >>> >>> Website: https://www.FreeSWITCH.com >>> >>> [image: https://www.facebook.com/signalwireinc?src=email] >>> [image: >>> https://twitter.com/freeswitch] >>> _________________________________________________________________________ >>> >>> 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 at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From mayamatakeshi at gmail.com Sat Dec 5 22:20:06 2020 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Sun, 6 Dec 2020 07:20:06 +0900 Subject: [Freeswitch-users] freeswitch won't open file In-Reply-To: References: Message-ID: On Sun, Dec 6, 2020 at 3:42 AM Joli Martinez wrote: > Hello, > > It is not a permissions issue. I have changed the permissions to rw for > everyone and it will still not read the file. > > The password.json file is in the same directory as the script > /usr/share/freeswitch/script. I am using lua 5.2.3 and again, the same > code works in my home directory. I just copied the password.json file over > and pasted the function into my code. When I place the call freeswitch > prints "Inside the function", so I know it gets there, but then freeswitch > spits out "attempt to index local 'file' (a nil value)" > > What else could I be missing? > Hi, when I check the current folder inside the fs cli I get: *freeswitch at lab002> system pwd* */* So, if you use: *f = "password.json"local file = io.open(f, "r")* freeswitch would try to open "/password.json" So I think you should pass the full file path: f = "/usr/share/freeswitch/script/password.json" -------------- next part -------------- An HTML attachment was scrubbed... URL: From mrjoli021 at gmail.com Sat Dec 5 23:41:57 2020 From: mrjoli021 at gmail.com (Joli Martinez) Date: Sat, 5 Dec 2020 18:41:57 -0500 Subject: [Freeswitch-users] freeswitch won't open file In-Reply-To: References: Message-ID: Hello, Putting the complete path worked. I honestly didn't think of that because they are in the same directory. Thanks. On Sat, Dec 5, 2020 at 5:37 PM mayamatakeshi wrote: > > > On Sun, Dec 6, 2020 at 3:42 AM Joli Martinez wrote: > >> Hello, >> >> It is not a permissions issue. I have changed the permissions to rw for >> everyone and it will still not read the file. >> >> The password.json file is in the same directory as the script >> /usr/share/freeswitch/script. I am using lua 5.2.3 and again, the same >> code works in my home directory. I just copied the password.json file over >> and pasted the function into my code. When I place the call freeswitch >> prints "Inside the function", so I know it gets there, but then freeswitch >> spits out "attempt to index local 'file' (a nil value)" >> >> What else could I be missing? >> > > Hi, > when I check the current folder inside the fs cli I get: > > *freeswitch at lab002> system pwd* > */* > > So, if you use: > > > > *f = "password.json"local file = io.open(f, "r")* > > freeswitch would try to open "/password.json" > So I think you should pass the full file path: > f = "/usr/share/freeswitch/script/password.json" > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From Alexander.Haugg at c4b.de Mon Dec 7 06:11:53 2020 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Mon, 7 Dec 2020 06:11:53 +0000 Subject: [Freeswitch-users] HELP DTLS HANDSHAKE to FAIL chromeV87 In-Reply-To: References: Message-ID: <658d2309a931479b8cbde2db918147ca@c4b.de> Hi, I think your problem is solved if your freeswitch works with DTLS version 1.2 instead of 1.0 https://freeswitch.org/jira/browse/FS-11730 Regards Alex Von: FreeSWITCH-users Im Auftrag von Jazmin Marina Florez Marino Gesendet: Freitag, 4. Dezember 2020 18:24 An: FreeSWITCH Users Help ; freeswitch-dev at lists.freeswitch.org; freeswitch-users-request at lists.freeswitch.org Betreff: [Freeswitch-users] HELP DTLS HANDSHAKE to FAIL chromeV87 Hi I have a verto client trying to connect to FS using Chrome V87 but we have DTLS problems. The openssl version we are using on FS side is OpenSSL 1.0.2k-fips FreeSWITCH (Version 1.8.4 64bit) FS LOG - Chrome V87 FAILED CONNECTION 2020-12-03 11:23:06.762291 [ERR] switch_rtp.c:3199 video Handshake failure 1 2020-12-03 11:23:06.762291 [INFO] switch_rtp.c:3200 Changing video DTLS state from HANDSHAKE to FAIL 2020-12-03 11:23:06.832295 [ERR] switch_rtp.c:3199 audio Handshake failure 1 2020-12-03 11:23:06.832295 [INFO] switch_rtp.c:3200 Changing audio DTLS state from HANDSHAKE to FAIL 2020-12-03 11:23:06.852295 [NOTICE] switch_rtp.c:3181 Hangup verto.rtc/3520 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2020-12-03 11:23:06.852295 [DEBUG] switch_core_media.c:7470 verto.rtc/3520 Video thread ended 2020-12-03 11:23:06.872305 [INFO] conference_loop.c:1670 Channel leaving conference, cause: DESTINATION_OUT_OF_ORDER 2020-12-03 11:23:06.872305 [DEBUG] switch_core_session.c:2639 verto.rtc/3520 skip receive message [DISPLAY] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] mod_conference.c:2467 verto.rtc/3520 skip receive message [TRANSFER] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_media.c:12220 verto.rtc/3520 skip receive message [BITRATE_REQ] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_codec.c:248 verto.rtc/3520 Restore previous codec opus:116. 2020-12-03 11:23:06.872305 [DEBUG] switch_core_session.c:2886 verto.rtc/3520 skip receive message [PHONE_EVENT] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:650 (verto.rtc/3520) State EXECUTE going to sleep 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:584 (verto.rtc/3520) Running State Change CS_HANGUP (Cur 1 Tot 3) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:847 (verto.rtc/3520) Callstate Change ACTIVE -> HANGUP 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:849 (verto.rtc/3520) State HANGUP 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:60 verto.rtc/3520 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:849 (verto.rtc/3520) State HANGUP going to sleep The weird thing is the connection works ok when the version of chrome is V86. Some weeks ago chrome was updated and we detected this issue. FS LOG - ChromeV86 CONNECTION OK 2020-12-03 11:19:19.622294 [INFO] switch_rtp.c:3206 Changing video DTLS state from HANDSHAKE to SETUP 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:3113 video Fingerprint Verified. 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:4104 Activating video Secure RTP SEND 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:4082 Activating video Secure RTP RECV 2020-12-03 11:19:19.672317 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:3155 Changing video DTLS state from SETUP to READY 2020-12-03 11:19:19.672317 [DEBUG] switch_rtp.c:1890 rtcp_stats_init: video ssrc[2719546543] base_seq[1999] 2020-12-03 11:19:19.672317 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3206 Changing audio DTLS state from HANDSHAKE to SETUP 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3113 audio Fingerprint Verified. 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:4104 Activating audio Secure RTP SEND 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:4082 Activating audio Secure RTP RECV 2020-12-03 11:19:19.702298 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3155 Changing audio DTLS state from SETUP to READY 2020-12-03 11:19:19.702298 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 -------------- next part -------------- An HTML attachment was scrubbed... URL: From Alexander.Haugg at c4b.de Mon Dec 7 06:42:57 2020 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Mon, 7 Dec 2020 06:42:57 +0000 Subject: [Freeswitch-users] Full Trickle ICE via mod_sofia supported? In-Reply-To: <5025d661c27446c4a8b7c34e360b9aa0@c4b.de> References: <6090BBFC-7FC2-4AD2-AD38-B2F34A363D7B@freeswitch.org> <5025d661c27446c4a8b7c34e360b9aa0@c4b.de> Message-ID: OK, is it supported via mod_verto? Or is it possible to add candidates with CLI commands? Addition: I only mean that the client can deliver candidates. The Freeswitch of course sends all candidates in the first SDP. Von: FreeSWITCH-users > Im Auftrag von Mike Jerris Gesendet: Donnerstag, 3. Dezember 2020 23:33 An: FreeSWITCH Users Help > Cc: Stefan Dietrich > Betreff: Re: [Freeswitch-users] Full Trickle ICE via mod_sofia supported? On Nov 30, 2020, at 7:56 AM, Alexander Haugg > wrote: Currently I'm trying to implement the Full Trickle mechanism as described here https://tools.ietf.org/id/draft-ietf-mmusic-trickle-ice-sip-11.html Since I haven't succeeded so far, I have the following questions. Is Full Trickle supported via mod_sofia? No -------------- next part -------------- An HTML attachment was scrubbed... URL: From sven at coviu.com Mon Dec 7 06:19:56 2020 From: sven at coviu.com (Sven Peters) Date: Mon, 7 Dec 2020 17:19:56 +1100 Subject: AWS EC2 setup using sometimes internal, sometimes external IP for host candidates Message-ID: <3F576505-35A6-4B40-BF8B-92D7FA44E6A6@coviu.com> Hi, I have followed Amazon EC2 to setup latest Freeswitch on an EC2 instance with an internal and public IP. When I dial out to an external SIP endpoint I can see sometimes the internal IP in the candidates list, sometimes the external IP - same configuration, just retrying the same endpoint. Internal IP: a=candidate:5446536444 1 udp 659136 172.25.36.61 29668 typ host generation External IP: a=candidate:1181434720 1 udp 659136 13.238.218.223 20132 typ host generation 0 How can I debug this and does anybody have an idea why this is inconsistent with the same config when redialing same endpoint? Thanks for your help Sven -------------- next part -------------- An HTML attachment was scrubbed... URL: From aidar.kamalov at gmail.com Mon Dec 7 19:41:09 2020 From: aidar.kamalov at gmail.com (Aidar Kamalov) Date: Mon, 7 Dec 2020 22:41:09 +0300 Subject: [Freeswitch-users] disable NOTIFY message Message-ID: Hello, is it possible to disable NOTIFY messages from freeswitch to external sip profile? NOTIFY sip:aaaa at aaaaaaa;intercom=true SIP/2.0 .... Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Event: talk Allow-Events: talk, hold, conference, refer Subscription-State: terminated;reason=noresource Content-Length: 0 -- Aydar A. Kamalov -------------- next part -------------- An HTML attachment was scrubbed... URL: From Zvonimir.Buzanic at asseco-see.hr Wed Dec 9 08:34:46 2020 From: Zvonimir.Buzanic at asseco-see.hr (=?utf-8?B?WnZvbmltaXIgQnXFvmFuacSH?=) Date: Wed, 9 Dec 2020 08:34:46 +0000 Subject: [Freeswitch-users] disable NOTIFY message In-Reply-To: References: Message-ID: <9c9c3c26c4a34042a98fb241764eac2a@asseco-see.hr> Try to disable mod_voicemail and see if that helps. unload mod_voicemail Br, Zvonimir From: FreeSWITCH-users On Behalf Of Aidar Kamalov Sent: Monday, December 7, 2020 8:41 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] disable NOTIFY message Hello, is it possible to disable NOTIFY messages from freeswitch to external sip profile? NOTIFY sip:aaaa at aaaaaaa;intercom=true SIP/2.0 .... Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Event: talk Allow-Events: talk, hold, conference, refer Subscription-State: terminated;reason=noresource Content-Length: 0 -- Aydar A. Kamalov This communication is for informational purposes only. All market prices, data and other information are not warranted as to completeness or accuracy and are subject to change without notice. Present message and any attached files may be or contain privileged information and is the property exclusive of ASSECO SEE CAPITAL GROUP. This transmission may contain information that is privileged, confidential, legally privileged, and/or exempt from disclosure under applicable law. The information contained in this message is solely intended for the physical or legal person to whom it is addressed and to the authorized persons for receiving it. In the case you are not the intended recipient or the authorized person to receive this message, we inform that disclosure, duplicate, distribution or taking up any actions on information contained in this message are strictly forbidden and are under civil and legal responsibility. In case you received it by error, you are requested to notify the sender and to destroy the original e-mail message from your system. Opinions, conclusions or any other information contained into this message, which are not related to ASSECO SEE CAPITAL GROUP activity must not be understood to be expressed or should be endorsed by ASSECO SEE CAPITAL GROUP. The interpretation expressed in the present message did not reflect ASSECO SEE CAPITAL GROUP opinion. -------------- next part -------------- An HTML attachment was scrubbed... URL: From adrian.worutowicz at esifrance.net Wed Dec 9 13:20:57 2020 From: adrian.worutowicz at esifrance.net (Adrian Worutowicz) Date: Wed, 9 Dec 2020 13:20:57 +0000 Subject: [Freeswitch-users] How to modify 'Allow:' in SIP header Message-ID: <10f1dd3d2a2143abb0079ad87d42986f@SRVEXCHANGE01.esifrance.net> FS sends : [Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, PRACK, NOTIFY] Is it possible to stop FS sending INFO ? If so, how ? In advance thank you for any tips. -------------- next part -------------- An HTML attachment was scrubbed... URL: From outrunner at live.fr Thu Dec 10 20:12:07 2020 From: outrunner at live.fr (jean rouquet) Date: Thu, 10 Dec 2020 20:12:07 +0000 Subject: [Freeswitch-users] ERROR [file is encrypted or is not a database] when connecting with ws or wss In-Reply-To: References: Message-ID: Hello, The problem appears after connecting few users using sipjs with wss or ws protocol. The only way to solve it for the moment is to stop freeswitch, remove sqlite db and restart freeswitch. I saw several bugs about this issue: https://freeswitch.org/jira/browse/FS-10676 https://freeswitch.org/jira/browse/FS-10474 however no workaround or patch is proposed. Does anyone know how to fix the issue? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Thu Dec 10 21:06:32 2020 From: brian at freeswitch.com (Brian West) Date: Thu, 10 Dec 2020 15:06:32 -0600 Subject: [Freeswitch-users] How to modify 'Allow:' in SIP header In-Reply-To: <10f1dd3d2a2143abb0079ad87d42986f@SRVEXCHANGE01.esifrance.net> References: <10f1dd3d2a2143abb0079ad87d42986f@SRVEXCHANGE01.esifrance.net> Message-ID: You would need to patch sofia.c, look for NUTAG_APPL_METHOD, It would need a config option to toggle it. You could do that and issue a PR on github? /b On Wed, Dec 9, 2020 at 7:42 AM Adrian Worutowicz < adrian.worutowicz at esifrance.net> wrote: > FS sends : [Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, *INFO*, > UPDATE, REGISTER, REFER, PRACK, NOTIFY] > > > > Is it possible to stop FS sending *INFO *? If so, how ? > > > > In advance thank you for any tips. > > > > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From adrian.worutowicz at esifrance.net Fri Dec 11 09:11:22 2020 From: adrian.worutowicz at esifrance.net (Adrian Worutowicz) Date: Fri, 11 Dec 2020 09:11:22 +0000 Subject: [Freeswitch-users] How to modify 'Allow:' in SIP header In-Reply-To: References: <10f1dd3d2a2143abb0079ad87d42986f@SRVEXCHANGE01.esifrance.net> Message-ID: <352227d23b1f455fbd6f9b604c5dd6eb@SRVEXCHANGE02.esifrance.net> I believe that it should function as follows : If rtp_liberal_dtmf=true, FS should accept all methods of DTMF transmission (In-band, RTP and SIP INFO), but it should respect the dtmf_type setting. If dtmf_type=none or dtmf_type=rfc2833, INFO should not be sent in Allow. If rtp_liberal_dtmf=false, FS should accept only a method determined by the dtmf_type setting. Thanks. Adrian. De : FreeSWITCH-users De la part de Brian West Envoyé : jeudi 10 décembre 2020 22:07 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] How to modify 'Allow:' in SIP header You would need to patch sofia.c, look for NUTAG_APPL_METHOD, It would need a config option to toggle it. You could do that and issue a PR on github? /b On Wed, Dec 9, 2020 at 7:42 AM Adrian Worutowicz > wrote: FS sends : [Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, PRACK, NOTIFY] Is it possible to stop FS sending INFO ? If so, how ? In advance thank you for any tips. _________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [https://lh6.googleusercontent.com/AYfRoSNaDNtMPRMevPn_GqcVEMd5NDRFi0GlluGUWzV6I5TAY_3T2-Tt0IuIXeUtEdYsgNsM8DOYKRKhjmrG_-n2Ga-LCnoNk46sO8VyEma1sBFYdiGJcLRUvkrD1CYHN79qimeg][https://lh3.googleusercontent.com/W4SqXyybH2qdAozvtoKjcz736qOjk9LHDwldvs1ahc-WVU0putVMSsUH474KDrJ32jsqi6JDjyUWxqeEkN5I1xSlC5ShYrd1b8NIMUkDzDrtbWQfa6A_90UcygqesBtRLgeFirKa] -------------- next part -------------- An HTML attachment was scrubbed... URL: From heller at relix.de Fri Dec 11 09:34:41 2020 From: heller at relix.de (Markus Heller) Date: Fri, 11 Dec 2020 10:34:41 +0100 Subject: [Freeswitch-users] SIP dialin not working Message-ID: <8f1c7193031d3df5d84a9dc8b3c119b099672876.camel@relix.de> Dear list, I am trying to configure SIP dialin. I can call my freeswitch/BBB server. It does execute the "play_and_get_digits", I can enter the PIN number of my conference which I can now see in my BBB GUI, but then I get a HANGUP. fs_clibbb tells me, when the call comes in: After entering the PIN number (DTMF decode works fine): EXECUTE [depth=0] sofia/external/[callernumber]@sip.[xyz].com transfer(SEND_TO_CONFERENCE XML default) ... switch_ivr.c:2243 ... State Change CS_EXECUTE -> CS_ROUTING switch_ivr.c:2250 Transfer sofia/external/[callernumber]@sip.[xyz].com to XML[SEND_TO_CONFERENCE at default] ... some more lines that don't look conspicuous... mod_dialplan_xml.c:637 Processing [callernumber] <[callernumber]>- >SEND_TO_CONFERENCE in context default Then there is a Dialplan block that does not terminate properly: Dialplan: sofia/external/[callernumber]@sip.[xyz].com parsing [default- >unloop] continue=false Dialplan: sofia/external/[callernumber]@sip.[xyz].com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/[callernumber]@sip.[xyz].com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/[callernumber]@sip.[xyz].com parsing [default- >bbb_conferences_ws] continue=false Dialplan: sofia/external/[callernumber]@sip.[xyz].com Regex (FAIL) [bbb_conferences_ws] ${bbb_authorized}() =~ /true/ break=on-false Dialplan: sofia/external/[callernumber]@sip.[xyz].com parsing [default- >bbb_conferences] continue=false Dialplan: sofia/external/[callernumber]@sip.[xyz].com Regex (FAIL) [bbb_conferences] ${bbb_authorized}() =~ /true/ break=on-false Dialplan: sofia/external/[callernumber]@sip.[xyz].com parsing [default- >bbb_echo_test_direct] continue=false Dialplan: sofia/external/[callernumber]@sip.[xyz].com Regex (FAIL) [bbb_echo_test_direct] ${bbb_authorized}() =~ /true/ break=on-false Dialplan: sofia/external/[callernumber]@sip.[xyz].com parsing [default- >ECHO_TO_CONFERENCE] continue=false Dialplan: sofia/external/[callernumber]@sip.[xyz].com Regex (FAIL) [ECHO_TO_CONFERENCE] ${bbb_from_echo}() =~ /true/ break=on-false Ending up in: 2020-12-11 10:16:01.332329 [INFO] switch_core_state_machine.c:312 No Route, Aborting With a final statement: 2020-12-11 10:16:01.332329 [NOTICE] switch_core_state_machine.c:313 Hangup sofia/external/[callernumber]@sip.[xyz].com [CS_ROUTING] [NO_ROUTE_DESTINATION] I understand that the Regex checks try to match the extension names in the three files in /opt/freeswitch/conf/dialplan/default. Probably, the call should match with "bbb_conferences", where the Regex fails with "not authorized". This is strange, because I am explicitly using an "authorized" statement in my dialin.xml config file: >>>>> <<<<< I think I fail to understand something right in front of me :-) Please help. Thanks a lot in advance! BR Markus From henry1 at bossdio.com Fri Dec 11 11:13:01 2020 From: henry1 at bossdio.com (Henry S) Date: Fri, 11 Dec 2020 22:13:01 +1100 Subject: INCOMPATIBLE_DESTINATION sip softphone <-> jssip webrtc Message-ID: <2d4afdfb-8b8f-c444-5ecd-e1eadb4c4adb@bossdio.com> Hi FS users I am stuck with INCOMPATIBLE reason of "Not Acceptable Here" 488 message. Calling from sipphone to sipphone works, calling from webrtc to webrtc works, calling a conference room from webrtc and sipphone (mixed) works, but not when directly crossing over. Sipphones tried are microsip and linphone I use the "proxy_media=true" for the webrtc calls to work. I have checked the codecs in the SIP bodies from both sides' INVITE message, they seem to have common codecs in different orders (not sure if this matters) Websocket endpoint listens to 127.0.0.1 7443 behind apache ws_tunnel. SIP 5060 listens on public IP Not sure why either side won't cross over. Text messages work fine from webrtc to sipphone. Any ideas? From aidar.kamalov at gmail.com Fri Dec 11 08:15:47 2020 From: aidar.kamalov at gmail.com (Aidar Kamalov) Date: Fri, 11 Dec 2020 11:15:47 +0300 Subject: [Freeswitch-users] disable NOTIFY message In-Reply-To: <9c9c3c26c4a34042a98fb241764eac2a@asseco-see.hr> References: <9c9c3c26c4a34042a98fb241764eac2a@asseco-see.hr> Message-ID: thanks for reply, but it doesn't help :( [image: image.png] ср, 9 дек. 2020 г. в 11:46, Zvonimir Bužanić : > Try to disable mod_voicemail and see if that helps. > > > > unload mod_voicemail > > > > Br, Zvonimir > > > > *From:* FreeSWITCH-users *On > Behalf Of *Aidar Kamalov > *Sent:* Monday, December 7, 2020 8:41 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] disable NOTIFY message > > > > Hello, > > > > is it possible to disable NOTIFY messages from freeswitch to external sip > profile? > > > > NOTIFY sip:aaaa at aaaaaaa;intercom=true SIP/2.0 > > .... > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, > REFER, NOTIFY > Supported: timer, path, replaces > Event: talk > Allow-Events: talk, hold, conference, refer > Subscription-State: terminated;reason=noresource > Content-Length: 0 > > > > > > -- > > Aydar A. Kamalov > This communication is for informational purposes only. All market prices, > data and other information are not warranted as to completeness or accuracy > and are subject to change without notice. Present message and any attached > files may be or contain privileged information and is the property > exclusive of ASSECO SEE CAPITAL GROUP. This transmission may contain > information that is privileged, confidential, legally privileged, and/or > exempt from disclosure under applicable law. The information contained in > this message is solely intended for the physical or legal person to whom it > is addressed and to the authorized persons for receiving it. In the case > you are not the intended recipient or the authorized person to receive this > message, we inform that disclosure, duplicate, distribution or taking up > any actions on information contained in this message are strictly forbidden > and are under civil and legal responsibility. In case you received it by > error, you are requested to notify the sender and to destroy the original > e-mail message from your system. Opinions, conclusions or any other > information contained into this message, which are not related to ASSECO > SEE CAPITAL GROUP activity must not be understood to be expressed or should > be endorsed by ASSECO SEE CAPITAL GROUP. The interpretation expressed in > the present message did not reflect ASSECO SEE CAPITAL GROUP opinion. > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Aydar A. Kamalov -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 12303 bytes Desc: not available URL: From rehfjo at gmail.com Fri Dec 11 14:31:03 2020 From: rehfjo at gmail.com (Robert Fitzjohn) Date: Fri, 11 Dec 2020 14:31:03 +0000 Subject: [Freeswitch-users] Reasons for call dropped/RFC2543 incompatible destination Message-ID: Hello, FreeSwitch v. 1.6.20 I'm calling my own mobile no. via my localphone gateway, and while this used to work when I last tested it a while ago, recently the call no longer goes through, with freeswitch dropping the call with the RFC2543 warning and then 'incompatible destination'. Codecs on both sides look good & calling other mobile numbers on other networks via the same gateway works fine, for what it's worth. The fuller/anonymized paste is at: https://pastebin.freeswitch.org/view/ddafee03 but the bits which I think matter are below. Could someone give me any hints as to whether this is a FS issue or whether the problem lies with my gateway or otherwise. Thanks - recv 1205 bytes from udp/[94.0.0.0]:5060 at 10:21:18.751270: - ------------------------------------------------------------------------ - SIP/2.0 183 Session Progress - Via: SIP/2.0/UDP 45.0.0.0:5080;rport=5080;branch=z9hG4bKHNUQK4208H7Br - Record-Route: ,, - To: ;tag=lp-2k9-5ee88476-00002a55-0005d2b3R2d70439b.b - From: "Robert" ;tag=B3N7U1NHeem0F - Call-ID: 45818239-b574-1239-8a8e-5600002a6bbb - CSeq: 29244045 INVITE - Allow: PUBLISH,MESSAGE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,OPTIONS,BYE,INVITE,ACK,CANCEL - Contact: - Content-Type: application/sdp - Content-Length: 474 - - v=0 - o=UK1-SBC-15-1 70259913 70259914 IN IP4 0.0.0.0 - s=sip call - c=IN IP4 0.0.0.0 - t=0 0 - m=audio 0 RTP/SAVP 18 8 0 101 - a=rtpmap:18 G729/8000 - a=fmtp:18 annexb=no - a=rtpmap:8 PCMA/8000 - a=rtpmap:0 PCMU/8000 - a=rtpmap:101 telephone-event/8000 - a=fmtp:101 0-15 - a=ptime:20 - m=audio 0 RTP/AVP 18 8 0 101 - a=rtpmap:18 G729/8000 - a=fmtp:18 annexb=no - a=rtpmap:8 PCMA/8000 - a=rtpmap:0 PCMU/8000 - a=rtpmap:101 telephone-event/8000 - a=fmtp:101 0-15 - a=ptime:20 - a=nortpproxy:yes - ------------------------------------------------------------------------ - 2020-12-10 10:21:18.738771 [DEBUG] sofia.c:7084 Channel sofia/external/447970000000 entering state [proceeding][183] - 2020-12-10 10:21:18.738771 [NOTICE] sofia.c:7192 Ring-Ready sofia/external/447970000000! - 2020-12-10 10:21:18.738771 [DEBUG] switch_channel.c:3346 (sofia/external/447970000000) Callstate Change DOWN -> RINGING - 2020-12-10 10:21:18.738771 [DEBUG] sofia.c:7084 Channel sofia/external/447970000000 entering state [proceeding][183] - 2020-12-10 10:21:18.738771 [DEBUG] sofia.c:7094 Remote SDP: - v=0 - o=UK1-SBC-15-1 70259913 70259914 IN IP4 0.0.0.0 - s=sip call - c=IN IP4 0.0.0.0 - t=0 0 - m=audio 0 RTP/SAVP 18 8 0 101 - m=audio 0 RTP/AVP 18 8 0 101 - - 2020-12-10 10:21:18.738771 [WARNING] switch_core_media.c:3951 RFC2543 from March 1999 called; They want their 0.0.0.0 hold method back..... - 2020-12-10 10:21:18.738771 [NOTICE] sofia.c:7273 Hangup sofia/external/447970000000 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] - 2020-12-10 10:21:18.738771 [DEBUG] switch_core_state_machine.c:584 (sofia/external/447970000000) Running State Change CS_HANGUP (Cur 4 Tot 65) - 2020-12-10 10:21:18.738771 [DEBUG] switch_core_state_machine.c:850 (sofia/external/447970000000) Callstate Change RINGING -> HANGUP - 2020-12-10 10:21:18.738771 [DEBUG] switch_core_state_machine.c:852 (sofia/external/447970000000) State HANGUP - 2020-12-10 10:21:18.738771 [DEBUG] mod_sofia.c:438 Channel sofia/external/447970000000 hanging up, cause: INCOMPATIBLE_DESTINATION -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Fri Dec 11 15:43:51 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 11 Dec 2020 15:43:51 +0000 Subject: [Freeswitch-users] Can't test for empty variable Message-ID: Hello all, I'm trying to test the play_and_get_digits variable from here: Like this: But it's always false... How can I test for an empty (null?) variable? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From kaduww at gmail.com Fri Dec 11 20:10:25 2020 From: kaduww at gmail.com (Carlos Eduardo) Date: Fri, 11 Dec 2020 17:10:25 -0300 Subject: [Freeswitch-users] Can't test for empty variable In-Reply-To: References: Message-ID: Hey David, Test with with this expression="^.?$" Regards... Em sex., 11 de dez. de 2020 às 12:44, David Villasmil < david.villasmil.work at gmail.com> escreveu: > Hello all, > > I'm trying to test the play_and_get_digits variable from here: > > > > Like this: > > > > But it's always false... > > How can I test for an empty (null?) variable? > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- *Carlos E. Wagner* *Tecnólogo em Telecomunicações, Opensips Certified Professional* *Fone: +55 48 99981-0894* *E-mail:* kaduww at gmail.com *LinkedIn:* https://www.linkedin.com/in/carlos-eduardo-wagner-96bbb433/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From borik.internet at gmail.com Fri Dec 11 20:52:04 2020 From: borik.internet at gmail.com (Dmitriy Borisov) Date: Fri, 11 Dec 2020 23:52:04 +0300 Subject: [Freeswitch-users] Can't test for empty variable In-Reply-To: References: Message-ID: As I remember, the empty value of a variable is equivalent of unset variable. So, I think you can't test variable if it is empty at all пт, 11 дек. 2020 г. в 23:46, Carlos Eduardo : > Hey David, > > Test with with this > > expression="^.?$" > > Regards... > > > > Em sex., 11 de dez. de 2020 às 12:44, David Villasmil < > david.villasmil.work at gmail.com> escreveu: > >> Hello all, >> >> I'm trying to test the play_and_get_digits variable from here: >> >> >> >> Like this: >> >> >> >> But it's always false... >> >> How can I test for an empty (null?) variable? >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > *Carlos E. Wagner* > *Tecnólogo em Telecomunicações, Opensips Certified Professional* > > *Fone: +55 48 99981-0894* > *E-mail:* kaduww at gmail.com > *LinkedIn:* https://www.linkedin.com/in/carlos-eduardo-wagner-96bbb433/ > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- With best regards Dmitry Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragic.dusan at gmail.com Sun Dec 13 19:38:31 2020 From: dragic.dusan at gmail.com (=?UTF-8?B?RHXFoWFuIERyYWdpxIc=?=) Date: Sun, 13 Dec 2020 20:38:31 +0100 Subject: [Freeswitch-users] Freeswitch doesn't run on Raspberry Pi 1 Message-ID: Hello all, I just installed Freeswitch 1.10.5 (1.10.5~release~6~25569c1631~buster-1~buster+1) from the offical freeswitch rpi repo on an old Raspberry Pi 1B, but trying to run freeswitch I'm getting "Illegal instruction". The OS is latest Raspberry Pi OS Lite, fully updated. If a take the SD card, plug it in my PC, mount it and run it in a chroot with qemu user arm emulation (qemu-arm-static) I can start freeswitch just fine, it only doesn't run on the rpi 1 hardware. I haven't installed any debug symbols so this isn't really useful, but gdb says: Program received signal SIGILL, Illegal instruction. 0xb68db728 in ?? () from /lib/arm-linux-gnueabihf/libcrypto.so.1.1 (gdb) bt full #0 0xb68db728 in ?? () from /lib/arm-linux-gnueabihf/libcrypto.so.1.1 No symbol table info available. #1 0xb68d518c in ?? () from /lib/arm-linux-gnueabihf/libcrypto.so.1.1 No symbol table info available. Backtrace stopped: previous frame identical to this frame (corrupt stack?) Does anyone know if the old rpi 1 is still supported by the packages from the repo or do they only work on newer pi models (armv7 cpus)? Regards, -- Dušan Dragić -------------- next part -------------- An HTML attachment was scrubbed... URL: From hawkins at hawkinsegroup.com Sun Dec 13 22:08:30 2020 From: hawkins at hawkinsegroup.com (Don Hawkins) Date: Sun, 13 Dec 2020 16:08:30 -0600 Subject: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE Suddenly Message-ID: Hi everyone! Suddenly all outbound calls (ALL of our providers) are returning NORMAL_TEMPORARY_FAILURE on our FS server. I've been working on this for roughly four hours so I wanted to reach out. It does seem that at least with one provider the calls are getting to their servers since they show as "Failed" in their management portal. We've changed nothing, suddenly origination just stopped working. sofia/gateway/*Provider*/12145551212%20&park() Any input is appreciated, I know it's something simple, just need to be guided in the right direction. Thanks! Don -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sun Dec 13 23:11:08 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sun, 13 Dec 2020 23:11:08 +0000 Subject: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE Suddenly In-Reply-To: References: Message-ID: What does the log say? On Sun, 13 Dec 2020 at 22:39, Don Hawkins wrote: > Hi everyone! > > Suddenly all outbound calls (ALL of our providers) are returning > NORMAL_TEMPORARY_FAILURE on our FS server. I've been working on this for > roughly four hours so I wanted to reach out. > > It does seem that at least with one provider the calls are getting to > their servers since they show as "Failed" in their management portal. > > We've changed nothing, suddenly origination just stopped working. > > sofia/gateway/*Provider*/12145551212%20&park() > > Any input is appreciated, I know it's something simple, just need to be > guided in the right direction. > > Thanks! > Don > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sun Dec 13 23:12:20 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sun, 13 Dec 2020 23:12:20 +0000 Subject: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE Suddenly In-Reply-To: References: Message-ID: console loglevel debug sofia global siptrace on <- if you want to the traces On Sun, 13 Dec 2020 at 23:11, David Villasmil < david.villasmil.work at gmail.com> wrote: > What does the log say? > > On Sun, 13 Dec 2020 at 22:39, Don Hawkins > wrote: > >> Hi everyone! >> >> Suddenly all outbound calls (ALL of our providers) are returning >> NORMAL_TEMPORARY_FAILURE on our FS server. I've been working on this for >> roughly four hours so I wanted to reach out. >> >> It does seem that at least with one provider the calls are getting to >> their servers since they show as "Failed" in their management portal. >> >> We've changed nothing, suddenly origination just stopped working. >> >> sofia/gateway/*Provider*/12145551212%20&park() >> >> Any input is appreciated, I know it's something simple, just need to be >> guided in the right direction. >> >> Thanks! >> Don >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From chad at apartmentlines.com Sun Dec 13 23:52:55 2020 From: chad at apartmentlines.com (Chad Phillips) Date: Sun, 13 Dec 2020 15:52:55 -0800 Subject: [Freeswitch-users] FreeSWITCH call recording missing header information Message-ID: I have a dialplan setup that records a bridged call in FreeSWITCH, which sets the 'api_hangup_hook' variable to call a lua script when the call hangs up, and the called lua script moves the recorded .wav file from it's temporary location to a final destination. This works, but recently I noticed that some of my .wav recordings have no header information, which results in many players choking when trying to play it. I eventually figured out that the header information for the .wav file appears to be written *after* the call is hung up, and file move operation in the api_hangup_hook moves the file before the header information has been written. This occurs sporadically. I've found that introducing a 1 second sleep in my lua script fixes this race condition, but that seems a bit of an ugly hack. Curious if anybody else has run into this, and/or can offer me a cleaner solution than my sleep hack. I'm attaching a brief summary of the dialplan and lua script for anyone that wants to see the code. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- --[[ Moves a completed operator recording from operator_recordings_in_progress to operator_recordings. This prevents the cron-based script that processes these recordings to customer directories from processing recordings in progress. ]] function process_finished_operator_recording() local AL_RECORDING_TEMP_DIR = get_value("AL_RECORDING_TEMP_DIR") local OPERATOR_RECORDINGS = get_value("OPERATOR_RECORDINGS") local operator_recording_filename = get_value("OPERATOR_RECORDING_FILENAME") local in_progress_recording = AL_RECORDING_TEMP_DIR .. "/operator_recordings_in_progress/" .. operator_recording_filename local recording = OPERATOR_RECORDINGS .. "/" .. operator_recording_filename if file_exists(in_progress_recording) then local sleep_secs = 1 debug_print(string.format([[Sleeping %d seconds to allow recording to finish]], sleep_secs)) os.execute(string.format([[sleep %d]], sleep_secs)) local ok = os.execute(string.format([[mv '%s' '%s']], in_progress_recording, recording)) if ok then debug_print(in_progress_recording .. " -> " .. recording) --[[ This can be used to dump recordings to a temp directory for inspection. local tmpdir = '/tmp/' local ok = os.execute(string.format("cp -a '%s' '%s'", recording, tmpdir)) if ok then debug_print(recording .. " copied to " .. tmpdir) end ]] end end end From hawkins at hawkinsegroup.com Mon Dec 14 00:09:21 2020 From: hawkins at hawkinsegroup.com (Don Hawkins) Date: Sun, 13 Dec 2020 18:09:21 -0600 Subject: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE Suddenly In-Reply-To: References: Message-ID: Here's the SIP trace. The only thing that catches my eye is the 503 Service Unavailable... freeswitch at heg_sip> /exit root at heg_sip:~# screen c=IN IP4 198.58.99.250 2020-12-14 00:04:16.141587 [DEBUG] switch_core_state_machine.c:621 (sofia/external/12145551212) State CONSUME_MEDIA going to sleep t=0 0 m=audio 22602 RTP/AVP 0 8 102 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:102 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 2020-12-14 00:04:16.141587 [DEBUG] sofia.c:6933 Channel sofia/external/12145551212 entering state [calling][0] recv 355 bytes from udp/[216.53.4.1]:5060 at 00:04:16.447428: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 198.58.99.250:5080 ;branch=z9hG4bK9DQSS9r0QS78c;received=198.58.99.250;rport=5080 From: "" ;tag=52X35SSgpQtrB To: Call-ID: bf3b677f-b842-1239-b3a8-da7d13be054f Contact: CSeq: 29398336 INVITE Server: sbc_5 Content-Length: 0 ------------------------------------------------------------------------ recv 419 bytes from udp/[216.53.4.1]:5060 at 00:04:16.501784: ------------------------------------------------------------------------ SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 198.58.99.250:5080 ;branch=z9hG4bK9DQSS9r0QS78c;received=198.58.99.250;rport=5080 From: "" ;tag=52X35SSgpQtrB To: ;tag=sbcsipuas_1_C20736_20201213190416427_ucs02sb04 Call-ID: bf3b677f-b842-1239-b3a8-da7d13be054f Contact: CSeq: 29398336 INVITE Server: sbc_5 Content-Length: 0 ------------------------------------------------------------------------ send 371 bytes to udp/[216.53.4.1]:5060 at 00:04:16.501891: ------------------------------------------------------------------------ ACK sip:12145551212 at 216.53.4.1:5060 SIP/2.0 Via: SIP/2.0/UDP 198.58.99.250:5080;rport;branch=z9hG4bK9DQSS9r0QS78c Max-Forwards: 70 From: "" ;tag=52X35SSgpQtrB To: ;tag=sbcsipuas_1_C20736_20201213190416427_ucs02sb04 Call-ID: bf3b677f-b842-1239-b3a8-da7d13be054f CSeq: 29398336 ACK Content-Length: 0 ------------------------------------------------------------------------ 2020-12-14 00:04:16.241578 [DEBUG] sofia.c:6933 Channel sofia/external/12145551212 entering state [terminated][503] 2020-12-14 00:04:16.241578 [NOTICE] sofia.c:7961 Hangup sofia/external/12145551212 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:543 (sofia/external/12145551212) Running State Change CS_HANGUP 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:809 (sofia/external/12145551212) Callstate Change DOWN -> HANGUP 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:811 (sofia/external/12145551212) State HANGUP 2020-12-14 00:04:16.241578 [DEBUG] mod_sofia.c:437 Channel sofia/external/12145551212 hanging up, cause: NORMAL_TEMPORARY_FAILURE 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:60 sofia/external/12145551212 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:811 (sofia/external/12145551212) State HANGUP going to sleep 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:578 (sofia/external/12145551212) State Change CS_HANGUP -> CS_REPORTING 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:543 (sofia/external/12145551212) Running State Change CS_REPORTING 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:897 (sofia/external/12145551212) State REPORTING 2020-12-14 00:04:16.241578 [DEBUG] switch_ivr_originate.c:3750 Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] On Sun, Dec 13, 2020 at 5:13 PM David Villasmil < david.villasmil.work at gmail.com> wrote: > > console loglevel debug > sofia global siptrace on <- if you want to the traces > > On Sun, 13 Dec 2020 at 23:11, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> What does the log say? >> >> On Sun, 13 Dec 2020 at 22:39, Don Hawkins >> wrote: >> >>> Hi everyone! >>> >>> Suddenly all outbound calls (ALL of our providers) are returning >>> NORMAL_TEMPORARY_FAILURE on our FS server. I've been working on this for >>> roughly four hours so I wanted to reach out. >>> >>> It does seem that at least with one provider the calls are getting to >>> their servers since they show as "Failed" in their management portal. >>> >>> We've changed nothing, suddenly origination just stopped working. >>> >>> sofia/gateway/*Provider*/12145551212%20&park() >>> >>> Any input is appreciated, I know it's something simple, just need to be >>> guided in the right direction. >>> >>> Thanks! >>> Don >>> _________________________________________________________________________ >>> >>> 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 at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- *Sincerely,* Don Hawkins CEO Hawkins Enterprise Group LLC https://hawkinsegroup.com Zello PTT : push2don P: 469-214-5044 -------------- next part -------------- An HTML attachment was scrubbed... URL: From kaiduanx at yahoo.ca Mon Dec 14 14:25:32 2020 From: kaiduanx at yahoo.ca (kaiduan xie) Date: Mon, 14 Dec 2020 14:25:32 +0000 (UTC) Subject: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE Suddenly In-Reply-To: References: Message-ID: <1231727172.328597.1607955932896@mail.yahoo.com> Your FS received 503 from 216.53.4.1, what is 216.53.4.1? Best regards, /Kaiduan On Sunday, December 13, 2020, 07:10:14 p.m. EST, Don Hawkins wrote: Here's the SIP trace. The only thing that catches my eye is the 503 Service Unavailable... freeswitch at heg_sip> /exit root at heg_sip:~# screen    c=IN IP4 198.58.99.250 2020-12-14 00:04:16.141587 [DEBUG] switch_core_state_machine.c:621 (sofia/external/12145551212) State CONSUME_MEDIA going to sleep    t=0 0    m=audio 22602 RTP/AVP 0 8 102 101    a=rtpmap:0 PCMU/8000    a=rtpmap:8 PCMA/8000    a=rtpmap:102 SPEEX/8000    a=rtpmap:101 telephone-event/8000    a=fmtp:101 0-16    a=ptime:20    ------------------------------------------------------------------------ 2020-12-14 00:04:16.141587 [DEBUG] sofia.c:6933 Channel sofia/external/12145551212 entering state [calling][0] recv 355 bytes from udp/[216.53.4.1]:5060 at 00:04:16.447428:    ------------------------------------------------------------------------    SIP/2.0 100 Trying    Via: SIP/2.0/UDP 198.58.99.250:5080;branch=z9hG4bK9DQSS9r0QS78c;received=198.58.99.250;rport=5080    From: "" ;tag=52X35SSgpQtrB    To:    Call-ID: bf3b677f-b842-1239-b3a8-da7d13be054f    Contact:    CSeq: 29398336 INVITE    Server: sbc_5    Content-Length: 0    ------------------------------------------------------------------------ recv 419 bytes from udp/[216.53.4.1]:5060 at 00:04:16.501784:    ------------------------------------------------------------------------    SIP/2.0 503 Service Unavailable    Via: SIP/2.0/UDP 198.58.99.250:5080;branch=z9hG4bK9DQSS9r0QS78c;received=198.58.99.250;rport=5080    From: "" ;tag=52X35SSgpQtrB    To: ;tag=sbcsipuas_1_C20736_20201213190416427_ucs02sb04    Call-ID: bf3b677f-b842-1239-b3a8-da7d13be054f    Contact:    CSeq: 29398336 INVITE    Server: sbc_5    Content-Length: 0    ------------------------------------------------------------------------ send 371 bytes to udp/[216.53.4.1]:5060 at 00:04:16.501891:    ------------------------------------------------------------------------    ACK sip:12145551212 at 216.53.4.1:5060 SIP/2.0    Via: SIP/2.0/UDP 198.58.99.250:5080;rport;branch=z9hG4bK9DQSS9r0QS78c    Max-Forwards: 70    From: "" ;tag=52X35SSgpQtrB    To: ;tag=sbcsipuas_1_C20736_20201213190416427_ucs02sb04    Call-ID: bf3b677f-b842-1239-b3a8-da7d13be054f    CSeq: 29398336 ACK    Content-Length: 0    ------------------------------------------------------------------------ 2020-12-14 00:04:16.241578 [DEBUG] sofia.c:6933 Channel sofia/external/12145551212 entering state [terminated][503] 2020-12-14 00:04:16.241578 [NOTICE] sofia.c:7961 Hangup sofia/external/12145551212 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:543 (sofia/external/12145551212) Running State Change CS_HANGUP 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:809 (sofia/external/12145551212) Callstate Change DOWN -> HANGUP 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:811 (sofia/external/12145551212) State HANGUP 2020-12-14 00:04:16.241578 [DEBUG] mod_sofia.c:437 Channel sofia/external/12145551212 hanging up, cause: NORMAL_TEMPORARY_FAILURE 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:60 sofia/external/12145551212 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:811 (sofia/external/12145551212) State HANGUP going to sleep 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:578 (sofia/external/12145551212) State Change CS_HANGUP -> CS_REPORTING 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:543 (sofia/external/12145551212) Running State Change CS_REPORTING 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:897 (sofia/external/12145551212) State REPORTING 2020-12-14 00:04:16.241578 [DEBUG] switch_ivr_originate.c:3750 Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] On Sun, Dec 13, 2020 at 5:13 PM David Villasmil wrote: console loglevel debugsofia global siptrace on <- if you want to the traces On Sun, 13 Dec 2020 at 23:11, David Villasmil wrote: What does the log say? On Sun, 13 Dec 2020 at 22:39, Don Hawkins wrote: Hi everyone! Suddenly all outbound calls (ALL of our providers) are returning NORMAL_TEMPORARY_FAILURE on our FS server. I've been working on this for roughly four hours so I wanted to reach out. It does seem that at least with one provider the calls are getting to their servers since they show as "Failed" in their management portal. We've changed nothing, suddenly origination just stopped working. sofia/gateway/Provider/12145551212%20&park() Any input is appreciated, I know it's something simple, just need to be guided in the right direction. Thanks! Don_________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- Regards, David Villasmilemail: david.villasmil.work at gmail.comphone: +34669448337 -- Regards, David Villasmilemail: david.villasmil.work at gmail.comphone: +34669448337_________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- Sincerely, Don Hawkins CEO Hawkins Enterprise Group LLC https://hawkinsegroup.com Zello PTT: push2don P: 469-214-5044 _________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From hawkins at hawkinsegroup.com Mon Dec 14 15:04:26 2020 From: hawkins at hawkinsegroup.com (Don Hawkins) Date: Mon, 14 Dec 2020 09:04:26 -0600 Subject: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE Suddenly In-Reply-To: References: Message-ID: Kaiduan, That's the termination provider's IP address. -------------- next part -------------- An HTML attachment was scrubbed... URL: From kaiduanx at yahoo.ca Mon Dec 14 15:08:52 2020 From: kaiduanx at yahoo.ca (kaiduan xie) Date: Mon, 14 Dec 2020 15:08:52 +0000 (UTC) Subject: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE Suddenly In-Reply-To: References: Message-ID: <166099219.339884.1607958532660@mail.yahoo.com> I think you need to ask them why they returned 503. On Monday, December 14, 2020, 10:04:38 a.m. EST, Don Hawkins wrote: Kaiduan, That's the termination provider's IP address. -------------- next part -------------- An HTML attachment was scrubbed... URL: From hawkins at hawkinsegroup.com Mon Dec 14 15:18:09 2020 From: hawkins at hawkinsegroup.com (Don Hawkins) Date: Mon, 14 Dec 2020 09:18:09 -0600 Subject: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE Suddenly In-Reply-To: <166099219.339884.1607958532660@mail.yahoo.com> References: <166099219.339884.1607958532660@mail.yahoo.com> Message-ID: I think you're right! This morning I added in new termination providers/gateways and they are working fine, so it seems to be an issue with our main provider. Thanks! On Mon, Dec 14, 2020 at 9:08 AM kaiduan xie wrote: > I think you need to ask them why they returned 503. > > On Monday, December 14, 2020, 10:04:38 a.m. EST, Don Hawkins < > hawkins at hawkinsegroup.com> wrote: > > > Kaiduan, > > > That's the termination provider's IP address. > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Dec 14 15:30:01 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 14 Dec 2020 15:30:01 +0000 Subject: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE Suddenly In-Reply-To: References: Message-ID: You should ask them why they’re replying with 503 On Mon, 14 Dec 2020 at 15:16, Don Hawkins wrote: > Kaiduan, > > That's the termination provider's IP address. > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From lconroy at insensate.co.uk Mon Dec 14 15:53:50 2020 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Mon, 14 Dec 2020 15:53:50 +0000 Subject: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE Suddenly In-Reply-To: References: Message-ID: <458B6EF2-E21C-4F58-BB3D-31889F30EFA6@insensate.co.uk> Hi there, This may be a stupid comment; if so please ignore, but ... Whilst it shouldn't matter, you are telling your outgoing provider that the call is from 0000000000. Do they reject calls with duff caller ID? best regards, Lawrence On 14 Dec 2020, at 00:09, Don Hawkins wrote: > Here's the SIP trace. The only thing that catches my eye is the 503 Service Unavailable... > > > freeswitch at heg_sip> /exit > root at heg_sip:~# screen > c=IN IP4 198.58.99.250 > 2020-12-14 00:04:16.141587 [DEBUG] switch_core_state_machine.c:621 (sofia/external/12145551212) State CONSUME_MEDIA going to sleep > t=0 0 > m=audio 22602 RTP/AVP 0 8 102 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:102 SPEEX/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > ------------------------------------------------------------------------ > 2020-12-14 00:04:16.141587 [DEBUG] sofia.c:6933 Channel sofia/external/12145551212 entering state [calling][0] > recv 355 bytes from udp/[216.53.4.1]:5060 at 00:04:16.447428: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 198.58.99.250:5080;branch=z9hG4bK9DQSS9r0QS78c;received=198.58.99.250;rport=5080 > From: "" ;tag=52X35SSgpQtrB > To: > Call-ID: bf3b677f-b842-1239-b3a8-da7d13be054f > Contact: > CSeq: 29398336 INVITE > Server: sbc_5 > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 419 bytes from udp/[216.53.4.1]:5060 at 00:04:16.501784: > ------------------------------------------------------------------------ > SIP/2.0 503 Service Unavailable > Via: SIP/2.0/UDP 198.58.99.250:5080;branch=z9hG4bK9DQSS9r0QS78c;received=198.58.99.250;rport=5080 > From: "" ;tag=52X35SSgpQtrB > To: ;tag=sbcsipuas_1_C20736_20201213190416427_ucs02sb04 > Call-ID: bf3b677f-b842-1239-b3a8-da7d13be054f > Contact: > CSeq: 29398336 INVITE > Server: sbc_5 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 371 bytes to udp/[216.53.4.1]:5060 at 00:04:16.501891: > ------------------------------------------------------------------------ > ACK sip:12145551212 at 216.53.4.1:5060 SIP/2.0 > Via: SIP/2.0/UDP 198.58.99.250:5080;rport;branch=z9hG4bK9DQSS9r0QS78c > Max-Forwards: 70 > From: "" ;tag=52X35SSgpQtrB > To: ;tag=sbcsipuas_1_C20736_20201213190416427_ucs02sb04 > Call-ID: bf3b677f-b842-1239-b3a8-da7d13be054f > CSeq: 29398336 ACK > Content-Length: 0 > > ------------------------------------------------------------------------ > 2020-12-14 00:04:16.241578 [DEBUG] sofia.c:6933 Channel sofia/external/12145551212 entering state [terminated][503] > 2020-12-14 00:04:16.241578 [NOTICE] sofia.c:7961 Hangup sofia/external/12145551212 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] > 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:543 (sofia/external/12145551212) Running State Change CS_HANGUP > 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:809 (sofia/external/12145551212) Callstate Change DOWN -> HANGUP > 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:811 (sofia/external/12145551212) State HANGUP > 2020-12-14 00:04:16.241578 [DEBUG] mod_sofia.c:437 Channel sofia/external/12145551212 hanging up, cause: NORMAL_TEMPORARY_FAILURE > 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:60 sofia/external/12145551212 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE > 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:811 (sofia/external/12145551212) State HANGUP going to sleep > 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:578 (sofia/external/12145551212) State Change CS_HANGUP -> CS_REPORTING > 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:543 (sofia/external/12145551212) Running State Change CS_REPORTING > 2020-12-14 00:04:16.241578 [DEBUG] switch_core_state_machine.c:897 (sofia/external/12145551212) State REPORTING > 2020-12-14 00:04:16.241578 [DEBUG] switch_ivr_originate.c:3750 Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] > > On Sun, Dec 13, 2020 at 5:13 PM David Villasmil wrote: > > console loglevel debug > sofia global siptrace on <- if you want to the traces > > On Sun, 13 Dec 2020 at 23:11, David Villasmil wrote: > What does the log say? > > On Sun, 13 Dec 2020 at 22:39, Don Hawkins wrote: > Hi everyone! > > Suddenly all outbound calls (ALL of our providers) are returning NORMAL_TEMPORARY_FAILURE on our FS server. I've been working on this for roughly four hours so I wanted to reach out. > > It does seem that at least with one provider the calls are getting to their servers since they show as "Failed" in their management portal. > > We've changed nothing, suddenly origination just stopped working. > > sofia/gateway/Provider/12145551212%20&park() > > Any input is appreciated, I know it's something simple, just need to be guided in the right direction. > > Thanks! > Don > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > -- > > > Sincerely, > Don Hawkins > CEO > Hawkins Enterprise Group LLC > https://hawkinsegroup.com > Zello PTT: push2don > P: 469-214-5044 > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com From alexandr.krasnov at exridge.com Mon Dec 14 12:28:51 2020 From: alexandr.krasnov at exridge.com (AlexKr) Date: Mon, 14 Dec 2020 05:28:51 -0700 (MST) Subject: [Freeswitch-users] Rearranging menus within retrieve voicemail app (mod_voicemail_ivr) Message-ID: <1607948931649-0.post@n2.nabble.com> Hello Everybody, I'm trying to remove some lines/menus from the 'retrieve voicemail' IVR. I'm using mod_voicemail_ivr for that. The issue is that changes to profile in the voicemail_ivr.conf.xml like commenting out any line in results in error: config === === log === [ERR] config.c:262 Missing api definition for profile 'default' === Commenting out any of menus does not prevent them from playing. Is there any trick that I'm missing with making changes to the mod_voicemail_ivr default config? Thank you in advance for any advice on the matter Alex -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/ From mitachundkrach at gmail.com Mon Dec 14 11:40:38 2020 From: mitachundkrach at gmail.com (R G) Date: Mon, 14 Dec 2020 12:40:38 +0100 Subject: [Freeswitch-users] Copy incoming audio from participants for ASR Message-ID: Hello everyone, i have a technical question about freeswitch and would like to know if my idea is technically possible. I've got a project at the university and my goal is to improve automatic speech recognition for bigbluebutton. In the summer semester a group of students of my university made a subtitling plugin[1] for bigbluebutton. This plugin adds a participant into the meeting and grabs his audiostream to send it to the kaldi ASR Software. When more than one person speaks at a time the detection decreases. My idea is to copy the incoming freeswitch audiostreams from every participant and send them directly as a copy to the ASR (kaldi can handle multiple audiostreams at once). In the Confluence-Wiki i searched for something in this direction but only found things like Dialplan to ring multiple telephones at once. I hope my idea is formulated understandably. Kind regards, Robert [1] https://github.com/3wille/bbb-kaldi-connector From piotr at dataandsignal.com Tue Dec 15 14:37:22 2020 From: piotr at dataandsignal.com (Piotr Gregor) Date: Tue, 15 Dec 2020 14:37:22 +0000 Subject: [Freeswitch-users] Copy incoming audio from participants for ASR In-Reply-To: References: Message-ID: Hi Robert, FreeSWITCH can be extended by writing custom modules. Modules can register a callback to get every audio frame passed by the core FS to them. You can find a lot of examples in src/mod (best look into src/mod/applications, e.g. mod_avmd works with audio frames). kind regards, Piotr Gregor Software Engineer M: (+44) 07483 866 525 L: (+44) 01256 597 470 www: dataandsignal.com On Mon, Dec 14, 2020 at 4:56 PM R G wrote: > Hello everyone, > > i have a technical question about freeswitch and would like to know if > my idea is technically possible. > > I've got a project at the university and my goal is to improve automatic > speech recognition for bigbluebutton. > > In the summer semester a group of students of my university made a > subtitling plugin[1] for bigbluebutton. This plugin adds a participant > into the meeting and grabs his audiostream to send it to the kaldi ASR > Software. When more than one person speaks at a time the detection > decreases. > > My idea is to copy the incoming freeswitch audiostreams from every > participant and send them directly as a copy to the ASR (kaldi can > handle multiple audiostreams at once). > > In the Confluence-Wiki i searched for something in this direction but > only found things like Dialplan to ring multiple telephones at once. > > I hope my idea is formulated understandably. > > > Kind regards, > > Robert > > [1] https://github.com/3wille/bbb-kaldi-connector > > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From davidswalkabout at gmail.com Tue Dec 15 21:35:44 2020 From: davidswalkabout at gmail.com (David P) Date: Wed, 16 Dec 2020 10:35:44 +1300 Subject: [Freeswitch-users] Copy incoming audio from participants for ASR In-Reply-To: References: Message-ID: How do you currently forward audio to kaldi? If you use an FS conference, I think there is no way to separate the audio per-user. I agree it would be very useful if there were a way to process each user's audio separately. After a few years of using FS, I haven't come across a way to do it. --------- Forwarded message ---------- > From: R G > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Mon, 14 Dec 2020 12:40:38 +0100 > Subject: [Freeswitch-users] Copy incoming audio from participants for ASR > Hello everyone, > > i have a technical question about freeswitch and would like to know if > my idea is technically possible. > > I've got a project at the university and my goal is to improve automatic > speech recognition for bigbluebutton. > > In the summer semester a group of students of my university made a > subtitling plugin[1] for bigbluebutton. This plugin adds a participant > into the meeting and grabs his audiostream to send it to the kaldi ASR > Software. When more than one person speaks at a time the detection > decreases. > > My idea is to copy the incoming freeswitch audiostreams from every > participant and send them directly as a copy to the ASR (kaldi can > handle multiple audiostreams at once). > > In the Confluence-Wiki i searched for something in this direction but > only found things like Dialplan to ring multiple telephones at once. > > I hope my idea is formulated understandably. > > > Kind regards, > > Robert > > [1] https://github.com/3wille/bbb-kaldi-connector > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From per at wgtwo.com Wed Dec 16 10:07:40 2020 From: per at wgtwo.com (Per Modin) Date: Wed, 16 Dec 2020 11:07:40 +0100 Subject: [Freeswitch-users] Can't test for empty variable In-Reply-To: References: Message-ID: <20201216100740.3e5b2bqwbltocvmi@wg2modin> On 2020-12-11 15:43 (Fri), David Villasmil wrote: > > > But it's always false... > > How can I test for an empty (null?) variable? Hi David, try checking for any characters (`".+"`), then you create an `` clause that'll be executed on empty (null?) variable. https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan#XMLDialplan-Anti-Actions Best, Per. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 833 bytes Desc: not available URL: From dujinfang at gmail.com Wed Dec 16 13:21:22 2020 From: dujinfang at gmail.com (Seven Du) Date: Wed, 16 Dec 2020 21:21:22 +0800 Subject: [Freeswitch-users] Copy incoming audio from participants for ASR In-Reply-To: References: Message-ID: I believe directly run the `detect_speech` app on each channel works. On Wed, Dec 16, 2020 at 6:21 AM David P wrote: > How do you currently forward audio to kaldi? > > If you use an FS conference, I think there is no way to separate the audio > per-user. > > I agree it would be very useful if there were a way to process each user's > audio separately. After a few years of using FS, I haven't come across a > way to do it. > > --------- Forwarded message ---------- >> From: R G >> To: freeswitch-users at lists.freeswitch.org >> Cc: >> Bcc: >> Date: Mon, 14 Dec 2020 12:40:38 +0100 >> Subject: [Freeswitch-users] Copy incoming audio from participants for ASR >> Hello everyone, >> >> i have a technical question about freeswitch and would like to know if >> my idea is technically possible. >> >> I've got a project at the university and my goal is to improve automatic >> speech recognition for bigbluebutton. >> >> In the summer semester a group of students of my university made a >> subtitling plugin[1] for bigbluebutton. This plugin adds a participant >> into the meeting and grabs his audiostream to send it to the kaldi ASR >> Software. When more than one person speaks at a time the detection >> decreases. >> >> My idea is to copy the incoming freeswitch audiostreams from every >> participant and send them directly as a copy to the ASR (kaldi can >> handle multiple audiostreams at once). >> >> In the Confluence-Wiki i searched for something in this direction but >> only found things like Dialplan to ring multiple telephones at once. >> >> I hope my idea is formulated understandably. >> >> >> Kind regards, >> >> Robert >> >> [1] https://github.com/3wille/bbb-kaldi-connector >> >> _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: From dujinfang at gmail.com Wed Dec 16 13:23:50 2020 From: dujinfang at gmail.com (Seven Du) Date: Wed, 16 Dec 2020 21:23:50 +0800 Subject: [Freeswitch-users] FreeSWITCH call recording missing header information In-Reply-To: References: Message-ID: maybe listen to the RECORD_STOP event instead of the hangup hook? also inotify on OS worked for me well. On Mon, Dec 14, 2020 at 8:14 AM Chad Phillips wrote: > I have a dialplan setup that records a bridged call in FreeSWITCH, > which sets the 'api_hangup_hook' variable to call a lua script when the > call hangs up, and the called lua script moves the recorded .wav file from > it's temporary location to a final destination. > > This works, but recently I noticed that some of my .wav recordings have no > header information, which results in many players choking when trying to > play it. > > I eventually figured out that the header information for the .wav file > appears to be written *after* the call is hung up, and file move operation > in the api_hangup_hook moves the file before the header information has > been written. This occurs sporadically. > > I've found that introducing a 1 second sleep in my lua script fixes this > race condition, but that seems a bit of an ugly hack. > > Curious if anybody else has run into this, and/or can offer me a cleaner > solution than my sleep hack. > > I'm attaching a brief summary of the dialplan and lua script for anyone > that wants to see the code. > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: From dujinfang at gmail.com Wed Dec 16 13:24:34 2020 From: dujinfang at gmail.com (Seven Du) Date: Wed, 16 Dec 2020 21:24:34 +0800 Subject: [Freeswitch-users] disable NOTIFY message In-Reply-To: References: <9c9c3c26c4a34042a98fb241764eac2a@asseco-see.hr> Message-ID: hack the code. On Sat, Dec 12, 2020 at 12:27 AM Aidar Kamalov wrote: > thanks for reply, but it doesn't help :( > > [image: image.png] > > ср, 9 дек. 2020 г. в 11:46, Zvonimir Bužanić < > Zvonimir.Buzanic at asseco-see.hr>: > >> Try to disable mod_voicemail and see if that helps. >> >> >> >> unload mod_voicemail >> >> >> >> Br, Zvonimir >> >> >> >> *From:* FreeSWITCH-users *On >> Behalf Of *Aidar Kamalov >> *Sent:* Monday, December 7, 2020 8:41 PM >> *To:* FreeSWITCH Users Help >> *Subject:* [Freeswitch-users] disable NOTIFY message >> >> >> >> Hello, >> >> >> >> is it possible to disable NOTIFY messages from freeswitch to external sip >> profile? >> >> >> >> NOTIFY sip:aaaa at aaaaaaa;intercom=true SIP/2.0 >> >> .... >> >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY >> Supported: timer, path, replaces >> Event: talk >> Allow-Events: talk, hold, conference, refer >> Subscription-State: terminated;reason=noresource >> Content-Length: 0 >> >> >> >> >> >> -- >> >> Aydar A. Kamalov >> This communication is for informational purposes only. All market prices, >> data and other information are not warranted as to completeness or accuracy >> and are subject to change without notice. Present message and any attached >> files may be or contain privileged information and is the property >> exclusive of ASSECO SEE CAPITAL GROUP. This transmission may contain >> information that is privileged, confidential, legally privileged, and/or >> exempt from disclosure under applicable law. The information contained in >> this message is solely intended for the physical or legal person to whom it >> is addressed and to the authorized persons for receiving it. In the case >> you are not the intended recipient or the authorized person to receive this >> message, we inform that disclosure, duplicate, distribution or taking up >> any actions on information contained in this message are strictly forbidden >> and are under civil and legal responsibility. In case you received it by >> error, you are requested to notify the sender and to destroy the original >> e-mail message from your system. Opinions, conclusions or any other >> information contained into this message, which are not related to ASSECO >> SEE CAPITAL GROUP activity must not be understood to be expressed or should >> be endorsed by ASSECO SEE CAPITAL GROUP. The interpretation expressed in >> the present message did not reflect ASSECO SEE CAPITAL GROUP opinion. >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > Aydar A. Kamalov > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 12303 bytes Desc: not available URL: From dujinfang at gmail.com Wed Dec 16 13:26:37 2020 From: dujinfang at gmail.com (Seven Du) Date: Wed, 16 Dec 2020 21:26:37 +0800 Subject: [Freeswitch-users] Full Trickle ICE via mod_sofia supported? In-Reply-To: References: <6090BBFC-7FC2-4AD2-AD38-B2F34A363D7B@freeswitch.org> <5025d661c27446c4a8b7c34e360b9aa0@c4b.de> Message-ID: you can add candidates in ACL but FS is ICE-Lite . On Mon, Dec 7, 2020 at 3:14 PM Alexander Haugg wrote: > OK, is it supported via mod_verto? Or is it possible to add candidates > with CLI commands? > > Addition: > > I only mean that the client can deliver candidates. The Freeswitch of > course sends all candidates in the first SDP. > > > > > > *Von:* FreeSWITCH-users *Im > Auftrag von *Mike Jerris > *Gesendet:* Donnerstag, 3. Dezember 2020 23:33 > *An:* FreeSWITCH Users Help > *Cc:* Stefan Dietrich > *Betreff:* Re: [Freeswitch-users] Full Trickle ICE via mod_sofia > supported? > > > > > > > > On Nov 30, 2020, at 7:56 AM, Alexander Haugg > wrote: > > Currently I'm trying to implement the Full Trickle mechanism as described > here https://tools.ietf.org/id/draft-ietf-mmusic-trickle-ice-sip-11.html > > Since I haven't succeeded so far, I have the following questions. > > Is Full Trickle supported via mod_sofia? > > > > No > > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: From joel at textplus.com Wed Dec 16 16:36:26 2020 From: joel at textplus.com (Joel Serrano) Date: Wed, 16 Dec 2020 08:36:26 -0800 Subject: [Freeswitch-users] FreeSWITCH call recording missing header information In-Reply-To: References: Message-ID: Another option is to just rely on CDRs, you can have FS post them remotely AND save them to local disk... 1- record 2- end call 3- when local cdr is generated parse it (json/xml) 4- search for the var that has the recording location 5- do stuff.... Cheers, Joel. On Wed, Dec 16, 2020 at 05:24 Seven Du wrote: > maybe listen to the RECORD_STOP event instead of the hangup hook? also > inotify on OS worked for me well. > > > On Mon, Dec 14, 2020 at 8:14 AM Chad Phillips > wrote: > >> I have a dialplan setup that records a bridged call in FreeSWITCH, >> which sets the 'api_hangup_hook' variable to call a lua script when the >> call hangs up, and the called lua script moves the recorded .wav file from >> it's temporary location to a final destination. >> >> This works, but recently I noticed that some of my .wav recordings have >> no header information, which results in many players choking when trying to >> play it. >> >> I eventually figured out that the header information for the .wav file >> appears to be written *after* the call is hung up, and file move operation >> in the api_hangup_hook moves the file before the header information has >> been written. This occurs sporadically. >> >> I've found that introducing a 1 second sleep in my lua script fixes this >> race condition, but that seems a bit of an ugly hack. >> >> Curious if anybody else has run into this, and/or can offer me a cleaner >> solution than my sleep hack. >> >> I'm attaching a brief summary of the dialplan and lua script for anyone >> that wants to see the code. >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Thu Dec 17 15:38:04 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 17 Dec 2020 15:38:04 +0000 Subject: [Freeswitch-users] reINVITE Message-ID: Hello all, Is there any way to know which commands trigger a reINVITE? I.e.: out of these: - uuid_break - uuid_bridge - uuid_broadcast - uuid_deflect - uuid_displace - uuid_record - uuid_transfer Anyone knows off the top of their head? Thanks David -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From mrjoli021 at gmail.com Sat Dec 19 01:36:05 2020 From: mrjoli021 at gmail.com (Joli Martinez) Date: Fri, 18 Dec 2020 20:36:05 -0500 Subject: [Freeswitch-users] forward registration Message-ID: Hello, We have a several FQDN's that are pointing to a Freeswitch box and users are registering to it. I need forward those registrations for one domain only to a new freeswitch box I am building. Not sure how to do this. -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sat Dec 19 08:07:20 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 19 Dec 2020 08:07:20 +0000 Subject: [Freeswitch-users] forward registration In-Reply-To: References: Message-ID: Change the FQDN :) On Sat, 19 Dec 2020 at 02:11, Joli Martinez wrote: > Hello, > > We have a several FQDN's that are pointing to a Freeswitch box and users > are registering to it. I need forward those registrations for one domain > only to a new freeswitch box I am building. Not sure how to do this. > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Sat Dec 19 09:54:14 2020 From: krice at freeswitch.org (Ken Rice) Date: Sat, 19 Dec 2020 03:54:14 -0600 Subject: [Freeswitch-users] forward registration In-Reply-To: References: Message-ID: put a real proxy in front of FreeSwitch. FreeSwitch is a B2BUA not a proxy. It terminates registrations. You can do a bad upper registration hack but the that domain would still have to live on the first freeswitch box Sent from my iPhone > On Dec 18, 2020, at 19:36, Joli Martinez wrote: > >  > Hello, > > We have a several FQDN's that are pointing to a Freeswitch box and users are registering to it. I need forward those registrations for one domain only to a new freeswitch box I am building. Not sure how to do this. > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com From david.villasmil.work at gmail.com Sat Dec 19 13:59:00 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 19 Dec 2020 13:59:00 +0000 Subject: [Freeswitch-users] forward registration In-Reply-To: References: Message-ID: That’s always the best choice, of course. On Sat, 19 Dec 2020 at 10:27, Ken Rice wrote: > put a real proxy in front of FreeSwitch. FreeSwitch is a B2BUA not a > proxy. It terminates registrations. > > You can do a bad upper registration hack but the that domain would still > have to live on the first freeswitch box > > Sent from my iPhone > > > On Dec 18, 2020, at 19:36, Joli Martinez wrote: > > > >  > > Hello, > > > > We have a several FQDN's that are pointing to a Freeswitch box and users > are registering to it. I need forward those registrations for one domain > only to a new freeswitch box I am building. Not sure how to do this. > > _________________________________________________________________________ > > > > 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 at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From nathan at robotics.net Sat Dec 19 16:41:11 2020 From: nathan at robotics.net (Nathan Stratton) Date: Sat, 19 Dec 2020 11:41:11 -0500 Subject: [Freeswitch-users] Tracking Call Leg through Callcenter Message-ID: On calls flowing them my FreeSWITCH box I use: Before bridge, to correlate my call legs between my public and private networks. The problem is if I put that before application callcenter it is not added. Is there any way to make correlate legs going into and then out from callcenter? ><> nathan stratton -------------- next part -------------- An HTML attachment was scrubbed... URL: From mitachundkrach at gmail.com Wed Dec 16 14:54:53 2020 From: mitachundkrach at gmail.com (R G) Date: Wed, 16 Dec 2020 15:54:53 +0100 Subject: [Freeswitch-users] Copy incoming audio from participants for ASR In-Reply-To: References: Message-ID: <00646f85-3b72-4bc5-0791-2ce1a37b384e@gmail.com> Hi all, currently we add an participant into the conference and take his audio for speech recognition. I think bbb uses the build FS conferences. So the mod_detect_speech app works on single channels in a conference? I hoped i can catch dailers before joining a conference and copy their streams as an alternative. Kind regards, Robert Am 16.12.2020 um 14:21 schrieb Seven Du: > I believe directly run the `detect_speech` app on each channel works. > > On Wed, Dec 16, 2020 at 6:21 AM David P > wrote: > > How do you currently forward audio to kaldi? > > If you use an FS conference, I think there is no way to separate > the audio per-user. > > I agree it would be very useful if there were a way to process > each user's audio separately. After a few years of using FS, I > haven't come across a way to do it. > > --------- Forwarded message ---------- > From: R G > > To: freeswitch-users at lists.freeswitch.org > > Cc: > Bcc: > Date: Mon, 14 Dec 2020 12:40:38 +0100 > Subject: [Freeswitch-users] Copy incoming audio from > participants for ASR > Hello everyone, > > i have a technical question about freeswitch and would like to > know if > my idea is technically possible. > > I've got a project at the university and my goal is to improve > automatic > speech recognition for bigbluebutton. > > In the summer semester a group of students of my university > made a > subtitling plugin[1] for bigbluebutton. This plugin adds a > participant > into the meeting and grabs his audiostream to send it to the > kaldi ASR > Software. When more than one person speaks at a time the > detection > decreases. > > My idea is to copy the incoming freeswitch audiostreams from > every > participant and send them directly as a copy to the ASR (kaldi > can > handle multiple audiostreams at once). > > In the Confluence-Wiki i searched for something in this > direction but > only found things like Dialplan to ring multiple telephones at > once. > > I hope my idea is formulated understandably. > > > Kind regards, > > Robert > > [1] https://github.com/3wille/bbb-kaldi-connector > > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com > > > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From rehfjo at gmail.com Sat Dec 19 15:50:01 2020 From: rehfjo at gmail.com (Robert Fitzjohn) Date: Sat, 19 Dec 2020 15:50:01 +0000 Subject: [Freeswitch-users] Reasons for call dropped/RFC2543 incompatible destination In-Reply-To: References: Message-ID: Hello, Replying to my own message in case it helps anyone else. Updated to FS 1.10.5 - issue persists. Digging a little, it possibly looks like FS doesn't like the m=...SAVP without any crypto lines? Not sure... I don't think it was related to the RFC2543 warning. setting rtp_secure_media_outbound=forbidden on the bridge call to the gateway, meaning FS sends only RTP/AVP, I get back a more normal looking 183, and the call proceeds as normal. I don't know enough about the SIP spec to be able to tell if my gateway returning an m= ..SAVP line without any accompanying crypto lines is valid or not - or even if this was the definite cause of the problem, but it's solved. Regards RF. On Fri, 11 Dec 2020 at 14:31, Robert Fitzjohn wrote: > > Hello, > FreeSwitch v. 1.6.20 > I'm calling my own mobile no. via my localphone gateway, and while this > used to work when I last tested it a while ago, recently the call no longer > goes through, with freeswitch dropping the call with the RFC2543 warning > and then 'incompatible destination'. Codecs on both sides look good & > calling other mobile numbers on other networks via the same gateway works > fine, for what it's worth. > > The fuller/anonymized paste is at: > https://pastebin.freeswitch.org/view/ddafee03 but the bits which I think > matter are below. > Could someone give me any hints as to whether this is a FS issue or > whether the problem lies with my gateway or otherwise. > > Thanks > > > - recv 1205 bytes from udp/[94.0.0.0]:5060 at 10:21:18.751270: > - > ------------------------------------------------------------------------ > - SIP/2.0 183 Session Progress > - Via: SIP/2.0/UDP 45.0.0.0:5080 > ;rport=5080;branch=z9hG4bKHNUQK4208H7Br > - Record-Route: ;lr;ftag=lp-2k9-5ee88476-00002a55-0005d2b3R2d70439b.a>,, > - To: >;tag=lp-2k9-5ee88476-00002a55-0005d2b3R2d70439b.b > - From: "Robert" ;tag=B3N7U1NHeem0F > - Call-ID: 45818239-b574-1239-8a8e-5600002a6bbb > - CSeq: 29244045 INVITE > - Allow: > PUBLISH,MESSAGE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,OPTIONS,BYE,INVITE,ACK,CANCEL > - Contact: > - Content-Type: application/sdp > - Content-Length: 474 > - > - v=0 > - o=UK1-SBC-15-1 70259913 70259914 IN IP4 0.0.0.0 > - s=sip call > - c=IN IP4 0.0.0.0 > - t=0 0 > - m=audio 0 RTP/SAVP 18 8 0 101 > - a=rtpmap:18 G729/8000 > - a=fmtp:18 annexb=no > - a=rtpmap:8 PCMA/8000 > - a=rtpmap:0 PCMU/8000 > - a=rtpmap:101 telephone-event/8000 > - a=fmtp:101 0-15 > - a=ptime:20 > - m=audio 0 RTP/AVP 18 8 0 101 > - a=rtpmap:18 G729/8000 > - a=fmtp:18 annexb=no > - a=rtpmap:8 PCMA/8000 > - a=rtpmap:0 PCMU/8000 > - a=rtpmap:101 telephone-event/8000 > - a=fmtp:101 0-15 > - a=ptime:20 > - a=nortpproxy:yes > - > ------------------------------------------------------------------------ > - 2020-12-10 10:21:18.738771 [DEBUG] sofia.c:7084 Channel > sofia/external/447970000000 entering state [proceeding][183] > - 2020-12-10 10:21:18.738771 [NOTICE] sofia.c:7192 Ring-Ready > sofia/external/447970000000! > - 2020-12-10 10:21:18.738771 [DEBUG] switch_channel.c:3346 > (sofia/external/447970000000) Callstate Change DOWN -> RINGING > - 2020-12-10 10:21:18.738771 [DEBUG] sofia.c:7084 Channel > sofia/external/447970000000 entering state [proceeding][183] > - 2020-12-10 10:21:18.738771 [DEBUG] sofia.c:7094 Remote SDP: > - v=0 > > - o=UK1-SBC-15-1 70259913 70259914 IN IP4 0.0.0.0 > > - s=sip call > - c=IN IP4 0.0.0.0 > - t=0 0 > > - m=audio 0 RTP/SAVP 18 8 0 101 > - m=audio 0 RTP/AVP 18 8 0 101 > > > > - > - 2020-12-10 10:21:18.738771 [WARNING] switch_core_media.c:3951 > RFC2543 from March 1999 called; They want their 0.0.0.0 hold method > back..... > - 2020-12-10 10:21:18.738771 [NOTICE] sofia.c:7273 Hangup > sofia/external/447970000000 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] > - 2020-12-10 10:21:18.738771 [DEBUG] switch_core_state_machine.c:584 > (sofia/external/447970000000) Running State Change CS_HANGUP (Cur 4 Tot 65) > - 2020-12-10 10:21:18.738771 [DEBUG] switch_core_state_machine.c:850 > (sofia/external/447970000000) Callstate Change RINGING -> HANGUP > - 2020-12-10 10:21:18.738771 [DEBUG] switch_core_state_machine.c:852 > (sofia/external/447970000000) State HANGUP > - 2020-12-10 10:21:18.738771 [DEBUG] mod_sofia.c:438 Channel > sofia/external/447970000000 hanging up, cause: INCOMPATIBLE_DESTINATION > > > > > > -- Robert Fitzjohn +44 7971 291 238 -------------- next part -------------- An HTML attachment was scrubbed... URL: From mitachundkrach at gmail.com Sun Dec 20 11:59:23 2020 From: mitachundkrach at gmail.com (R G) Date: Sun, 20 Dec 2020 12:59:23 +0100 Subject: [Freeswitch-users] VLC Errormessage Message-ID: <53840fc6-f71e-db04-0df3-b79e884b4bd5@gmail.com> Hi all, i have a problem with the mod_vlc Plugin. Regardless of whether i try to record a conference to stream or a file i get the following error message: 2020-12-19 22:08:09.523574 [INFO] mod_conference.c:433 Auto recording file: vlc://#standard{access=file,mux=raw,dst=/var/freeswitch/meetings/test.wav} 2020-12-19 22:08:09.523574 [DEBUG] conference_loop.c:1338 Setup timer soft success interval: 20  samples: 960 from codec opus 2020-12-19 22:08:09.523574 [DEBUG] conference_record.c:304 Setup timer success interval: 20  samples: 960 2020-12-19 22:08:09.523574 [ERR] mod_vlc.c:772 VLC error: cannot initialise VLC handle 2020-12-19 22:08:09.523574 [ERR] conference_record.c:277 Error Opening File [vlc://#standard{access=file,mux=raw,dst=/var/freeswitch/meetings/test.wav}] 2020-12-19 22:08:09.523574 [ERR] switch_core_timer.c:117 Timer is not properly configured. 2020-12-19 22:08:09.523574 [INFO] conference_record.c:425 Recording of vlc://#standard{access=file,mux=raw,dst=/var/freeswitch/meetings/test.wav} Stopped This is my conference.conf.xml: I use a Ubuntu 16.04 and libVLC is installed: libvlc-dev/xenial-updates,xenial-security,now 2.2.2-5ubuntu0.16.04.4 amd64 [installed] libvlc5/xenial-updates,xenial-security,now 2.2.2-5ubuntu0.16.04.4 amd64 [installed,automatic] libvlccore-dev/xenial-updates,xenial-security,now 2.2.2-5ubuntu0.16.04.4 amd64 [installed] libvlccore8/xenial-updates,xenial-security,now 2.2.2-5ubuntu0.16.04.4 amd64 [installed,automatic] vlc-data/xenial-updates,xenial-updates,xenial-security,xenial-security,now 2.2.2-5ubuntu0.16.04.4 all [installed,automatic] Kind regards, Robert From mitachundkrach at gmail.com Sun Dec 20 21:37:57 2020 From: mitachundkrach at gmail.com (R G) Date: Sun, 20 Dec 2020 22:37:57 +0100 Subject: [Freeswitch-users] VLC Errormessage In-Reply-To: <53840fc6-f71e-db04-0df3-b79e884b4bd5@gmail.com> References: <53840fc6-f71e-db04-0df3-b79e884b4bd5@gmail.com> Message-ID: <307154ca-a968-b75c-5996-82ef91d2833b@gmail.com> Hi all, i figured out by myself. I was missing vlc and pulseaudio. Kind regards, Robert Am 20.12.2020 um 12:59 schrieb R G: > Hi all, > > i have a problem with the mod_vlc Plugin. Regardless of whether i try > to record a conference to stream or a file i get the following error > message: > > 2020-12-19 22:08:09.523574 [INFO] mod_conference.c:433 Auto recording > file: > vlc://#standard{access=file,mux=raw,dst=/var/freeswitch/meetings/test.wav} > 2020-12-19 22:08:09.523574 [DEBUG] conference_loop.c:1338 Setup timer > soft success interval: 20  samples: 960 from codec opus > 2020-12-19 22:08:09.523574 [DEBUG] conference_record.c:304 Setup timer > success interval: 20  samples: 960 > 2020-12-19 22:08:09.523574 [ERR] mod_vlc.c:772 VLC error: cannot > initialise VLC handle > 2020-12-19 22:08:09.523574 [ERR] conference_record.c:277 Error Opening > File > [vlc://#standard{access=file,mux=raw,dst=/var/freeswitch/meetings/test.wav}] > 2020-12-19 22:08:09.523574 [ERR] switch_core_timer.c:117 Timer is not > properly configured. > 2020-12-19 22:08:09.523574 [INFO] conference_record.c:425 Recording of > vlc://#standard{access=file,mux=raw,dst=/var/freeswitch/meetings/test.wav} > Stopped > > This is my conference.conf.xml: > > value="vlc://#standard{access=file,mux=raw,dst=/var/freeswitch/meetings/test.wav}"/> > > I use a Ubuntu 16.04 and libVLC is installed: > > libvlc-dev/xenial-updates,xenial-security,now 2.2.2-5ubuntu0.16.04.4 > amd64 [installed] > libvlc5/xenial-updates,xenial-security,now 2.2.2-5ubuntu0.16.04.4 > amd64 [installed,automatic] > libvlccore-dev/xenial-updates,xenial-security,now > 2.2.2-5ubuntu0.16.04.4 amd64 [installed] > libvlccore8/xenial-updates,xenial-security,now 2.2.2-5ubuntu0.16.04.4 > amd64 [installed,automatic] > vlc-data/xenial-updates,xenial-updates,xenial-security,xenial-security,now > 2.2.2-5ubuntu0.16.04.4 all [installed,automatic] > > > Kind regards, > > Robert > > From nico at vthadden.de Sat Dec 26 16:56:59 2020 From: nico at vthadden.de (Nicola von Thadden) Date: Sat, 26 Dec 2020 17:56:59 +0100 Subject: [Freeswitch-users] Calls dropping due to SDP change Message-ID: <5ffc1416-8c70-ff43-3ac8-0fe6532efec2@vthadden.de> Hi, I'm currently investigating reproducable call drops when calling mobile-phone numbers from Deutsche Telekom (T-Mobile Germany). My provider has a NGN (SIP) Interconnection with Deutsche Telekom and is transparend for the messages involved here (although IP rewriting is happening). The calls contain following SDP in the invite from my FreeSwitch towards my provider: v=0                                                                                                                                                                                                                                                                                                                          o=FreeSWITCH 1608967203 1608967204 IN IP4 *redacted*                                                                                                                                                                                                                                                                      s=FreeSWITCH                                                                                                                                                                                                                                                                                                                 c=IN IP4  *redacted*                                                                                                                                                                                                                                                                                                    t=0 0                                                                                                                                                                                                                                                                                                                        m=audio 32256 RTP/AVP 9 8 0 101                                                                                                                                                                                                                                                                                              a=rtpmap:9 G722/8000                                                                                                                                                                                                                                                                                                         a=rtpmap:8 PCMA/8000                                                                                                                                                                                                                                                                                                         a=rtpmap:0 PCMU/8000                                                                                                                                                                                                                                                                                                         a=rtpmap:101 telephone-event/8000                                                                                                                                                                                                                                                                                            a=fmtp:101 0-16                                                                                                                                                                                                                                                                                                              a=ptime:20   Once the call is established, FS sends a re-invite after 50% of the expiration timer is elapsed, 15 minutes in this case. The re-invite contains a slightly modified SDP: v=0                                                                                                                                                                                                                                                                                                                          o=FreeSWITCH 1608967203 1608967204 IN IP4 *redacted*                                                                                                                                                                                                                                                                      s=FreeSWITCH                                                                                                                                                                                                                                                                                                                 c=IN IP4 *redacted*                                                                                                                                                                                                                                                                                                       t=0 0                                                                                                                                                                                                                                                                                                                        m=audio 32256 RTP/AVP 8 101 9 0                                                                                                                                                                                                                                                                                              a=rtpmap:8 PCMA/8000                                                                                                                                                                                                                                                                                                         a=rtpmap:101 telephone-event/8000                                                                                                                                                                                                                                                                                            a=fmtp:101 0-16                                                                                                                                                                                                                                                                                                              a=rtpmap:9 G722/8000                                                                                                                                                                                                                                                                                                         a=rtpmap:0 PCMU/8000                                                                                                                                                                                                                                                                                                         a=ptime:20    The new codec on position 1 (PCMA) is not necessary the chosen one for the session, that call was using G.722 (verified via 'show channels'). Telekom does not like my SDP change and responds with: SIP/2.0 488 SDP Parameter Error In SIP Request                                                                                                                                                                                                                                                                               The freeswitch console only logs: 2020-12-26 16:32:49.578925 [DEBUG] sofia.c:7326 Channel sofia/external/*redacted*entering state [calling][0] 2020-12-26 16:32:49.618925 [DEBUG] sofia.c:7319 Channel sofia/external/*redacted* skipping state [ready][488] The call is disconnected 15 minutes later because the session timer has expired: 2020-12-26 16:47:53.838925 [NOTICE] sofia.c:1089 Hangup sofia/external/*redacted* [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] This does not happen when I only allow PCMA (or any other codec) since the SDP can't get mixed up. I have reproduced that behaviour with 1.10.5~release~6~25569c1631~buster-1~buster+1 and 1.6.20~37~987c9b9-1~jessie+1. Do you have any idea why FS changes the SDP (without reason?) and what I can do about it? Thanks Nico From mrjoli021 at gmail.com Sun Dec 27 19:08:31 2020 From: mrjoli021 at gmail.com (Joli Martinez) Date: Sun, 27 Dec 2020 14:08:31 -0500 Subject: [Freeswitch-users] docker image Message-ID: Hello, Our company is pushing toward Docker now. That is the new focus for next year. Is there a Freeswitch Docker image already available? I searched on Dockerhub and could not find one. If there is not one avail. I was thinking of building one. I was planning on using the latest Debian base image and creating a Dockerfile with the installation of FreeSwitch. Is that the best approach or is there a better alternative? Also is there any reason not to Dockerize Freeswitch? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sun Dec 27 22:12:17 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sun, 27 Dec 2020 23:12:17 +0100 Subject: [Freeswitch-users] docker image In-Reply-To: References: Message-ID: This might give you a starting point https://github.com/davidcsi/docker-freeswitch Regards On Sun, 27 Dec 2020 at 20:46, Joli Martinez wrote: > Hello, > > Our company is pushing toward Docker now. That is the new focus for next > year. Is there a Freeswitch Docker image already available? I searched on > Dockerhub and could not find one. If there is not one avail. I was > thinking of building one. I was planning on using the latest Debian base > image and creating a Dockerfile with the installation of FreeSwitch. Is > that the best approach or is there a better alternative? > > Also is there any reason not to Dockerize Freeswitch? > > Thanks, > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From brians at iptel.co Sun Dec 27 23:04:52 2020 From: brians at iptel.co (Brian :) Date: Sun, 27 Dec 2020 23:04:52 +0000 Subject: [Freeswitch-users] Calls dropping due to SDP change In-Reply-To: <5ffc1416-8c70-ff43-3ac8-0fe6532efec2@vthadden.de> References: <5ffc1416-8c70-ff43-3ac8-0fe6532efec2@vthadden.de> Message-ID: Unless you're doing hd audio on these calls your solution is to just use g711 on calls into this provider. On Saturday, December 26, 2020, Nicola von Thadden wrote: > Hi, > > I'm currently investigating reproducable call drops when calling > mobile-phone numbers from Deutsche Telekom (T-Mobile Germany). > My provider has a NGN (SIP) Interconnection with Deutsche Telekom and is > transparend for the messages involved here (although IP rewriting is > happening). > > The calls contain following SDP in the invite from my FreeSwitch towards > my provider: > v=0 > > o=FreeSWITCH 1608967203 1608967204 IN IP4 *redacted* > > > s=FreeSWITCH > > c=IN IP4 *redacted* > > > t=0 > 0 > > m=audio 32256 RTP/AVP 9 8 0 > 101 > > a=rtpmap:9 > G722/8000 > > a=rtpmap:8 > PCMA/8000 > > a=rtpmap:0 > PCMU/8000 > > a=rtpmap:101 > telephone-event/8000 > > a=fmtp:101 > 0-16 > > a=ptime:20 > > Once the call is established, FS sends a re-invite after 50% of the > expiration timer is elapsed, 15 minutes in this case. The re-invite > contains a slightly modified SDP: > v=0 > > o=FreeSWITCH 1608967203 1608967204 IN IP4 > *redacted* > > s=FreeSWITCH > > c=IN IP4 > *redacted* > > t=0 > 0 > > m=audio 32256 RTP/AVP 8 101 9 > 0 > > a=rtpmap:8 > PCMA/8000 > > a=rtpmap:101 > telephone-event/8000 > > a=fmtp:101 > 0-16 > > a=rtpmap:9 > G722/8000 > > a=rtpmap:0 > PCMU/8000 > > a=ptime:20 > > The new codec on position 1 (PCMA) is not necessary the chosen one for > the session, that call was using G.722 (verified via 'show channels'). > > Telekom does not like my SDP change and responds with: > SIP/2.0 488 SDP Parameter Error In SIP > Request > > > The freeswitch console only logs: > 2020-12-26 16:32:49.578925 [DEBUG] sofia.c:7326 Channel > sofia/external/*redacted*entering state [calling][0] > 2020-12-26 16:32:49.618925 [DEBUG] sofia.c:7319 Channel > sofia/external/*redacted* skipping state [ready][488] > > The call is disconnected 15 minutes later because the session timer has > expired: > 2020-12-26 16:47:53.838925 [NOTICE] sofia.c:1089 Hangup > sofia/external/*redacted* [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > > This does not happen when I only allow PCMA (or any other codec) since > the SDP can't get mixed up. > > I have reproduced that behaviour with > 1.10.5~release~6~25569c1631~buster-1~buster+1 and > 1.6.20~37~987c9b9-1~jessie+1. > > Do you have any idea why FS changes the SDP (without reason?) and what I > can do about it? > > Thanks > Nico > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From joel at textplus.com Mon Dec 28 03:37:05 2020 From: joel at textplus.com (Joel Serrano) Date: Sun, 27 Dec 2020 19:37:05 -0800 Subject: [Freeswitch-users] docker image In-Reply-To: References: Message-ID: Hi David, Out of curiosity, why do you need supervisord in your dockerfile? On Sun, Dec 27, 2020 at 14:13 David Villasmil < david.villasmil.work at gmail.com> wrote: > This might give you a starting point > > https://github.com/davidcsi/docker-freeswitch > > Regards > > On Sun, 27 Dec 2020 at 20:46, Joli Martinez wrote: > >> Hello, >> >> Our company is pushing toward Docker now. That is the new focus for next >> year. Is there a Freeswitch Docker image already available? I searched on >> Dockerhub and could not find one. If there is not one avail. I was >> thinking of building one. I was planning on using the latest Debian base >> image and creating a Dockerfile with the installation of FreeSwitch. Is >> that the best approach or is there a better alternative? >> >> Also is there any reason not to Dockerize Freeswitch? >> >> Thanks, >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Mon Dec 28 08:48:47 2020 From: s.safarov at gmail.com (Sergey Safarov) Date: Mon, 28 Dec 2020 11:48:47 +0300 Subject: [Freeswitch-users] docker image In-Reply-To: References: Message-ID: You can check https://hub.docker.com/repository/docker/safarov/freeswitch About 100 Mb size, based on Debian On Mon, Dec 28, 2020 at 7:07 AM Joel Serrano wrote: > Hi David, > > Out of curiosity, why do you need supervisord in your dockerfile? > > > On Sun, Dec 27, 2020 at 14:13 David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> This might give you a starting point >> >> https://github.com/davidcsi/docker-freeswitch >> >> Regards >> >> On Sun, 27 Dec 2020 at 20:46, Joli Martinez wrote: >> >>> Hello, >>> >>> Our company is pushing toward Docker now. That is the new focus for >>> next year. Is there a Freeswitch Docker image already available? I >>> searched on Dockerhub and could not find one. If there is not one avail. >>> I was thinking of building one. I was planning on using the latest Debian >>> base image and creating a Dockerfile with the installation of FreeSwitch. >>> Is that the best approach or is there a better alternative? >>> >>> Also is there any reason not to Dockerize Freeswitch? >>> >>> Thanks, >>> >>> _________________________________________________________________________ >>> >>> 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 at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From nico at vthadden.de Mon Dec 28 14:56:02 2020 From: nico at vthadden.de (Nicola von Thadden) Date: Mon, 28 Dec 2020 15:56:02 +0100 Subject: [Freeswitch-users] Calls dropping due to SDP change In-Reply-To: References: <5ffc1416-8c70-ff43-3ac8-0fe6532efec2@vthadden.de> Message-ID: Hi, yes, we are using HD audio there. Deutsche Telekom is codec transparent, they even pass opus and other codecs since a while. So disabling any codecs but G.711 is not really a solution, especially since DT is the biggest provider in Germany and that issue might not only affect my setup but in theory all other Freeswitch in Germany here which might want to call a DT landline or mobile number. Is there a way to make freeswitch stop mixing up the SDP? Nico On 12/28/20 12:04 AM, Brian : wrote: > Unless you're doing hd audio on these calls your solution is to just > use g711 on calls into this provider. > > On Saturday, December 26, 2020, Nicola von Thadden > wrote: > > Hi, > > > > I'm currently investigating reproducable call drops when calling > > mobile-phone numbers from Deutsche Telekom (T-Mobile Germany). > > My provider has a NGN (SIP) Interconnection with Deutsche Telekom and is > > transparend for the messages involved here (although IP rewriting is > > happening). > > > > The calls contain following SDP in the invite from my FreeSwitch towards > > my provider: > > > v=0                                                                                                                                                                                                                                                                                                                          > > > > o=FreeSWITCH 1608967203 1608967204 IN IP4 *redacted* > > >                                                                                                                                                                                                                                                                      > > > > > s=FreeSWITCH                                                                                                                                                                                                                                                                                                                 > > > > c=IN IP4  *redacted* > > >                                                                                                                                                                                                                                                                                                    > > > > t=0 > > > 0                                                                                                                                                                                                                                                                                                                        > > > > m=audio 32256 RTP/AVP 9 8 0 > > > 101                                                                                                                                                                                                                                                                                              > > > > a=rtpmap:9 > > > G722/8000                                                                                                                                                                                                                                                                                                         > > > > a=rtpmap:8 > > > PCMA/8000                                                                                                                                                                                                                                                                                                         > > > > a=rtpmap:0 > > > PCMU/8000                                                                                                                                                                                                                                                                                                         > > > > a=rtpmap:101 > > > telephone-event/8000                                                                                                                                                                                                                                                                                            > > > > a=fmtp:101 > > > 0-16                                                                                                                                                                                                                                                                                                              > > > > a=ptime:20   > > > > Once the call is established, FS sends a re-invite after 50% of the > > expiration timer is elapsed, 15 minutes in this case. The re-invite > > contains a slightly modified SDP: > > > v=0                                                                                                                                                                                                                                                                                                                          > > > > o=FreeSWITCH 1608967203 1608967204 IN IP4 > > > *redacted*                                                                                                                                                                                                                                                                      > > > > > s=FreeSWITCH                                                                                                                                                                                                                                                                                                                 > > > > c=IN IP4 > > > *redacted*                                                                                                                                                                                                                                                                                                       > > > > t=0 > > > 0                                                                                                                                                                                                                                                                                                                        > > > > m=audio 32256 RTP/AVP 8 101 9 > > > 0                                                                                                                                                                                                                                                                                              > > > > a=rtpmap:8 > > > PCMA/8000                                                                                                                                                                                                                                                                                                         > > > > a=rtpmap:101 > > > telephone-event/8000                                                                                                                                                                                                                                                                                            > > > > a=fmtp:101 > > > 0-16                                                                                                                                                                                                                                                                                                              > > > > a=rtpmap:9 > > > G722/8000                                                                                                                                                                                                                                                                                                         > > > > a=rtpmap:0 > > > PCMU/8000                                                                                                                                                                                                                                                                                                         > > > > a=ptime:20    > > > > The new codec on position 1 (PCMA) is not necessary the chosen one for > > the session, that call was using G.722 (verified via 'show channels'). > > > > Telekom does not like my SDP change and responds with: > > SIP/2.0 488 SDP Parameter Error In SIP > > > Request                                                                                                                                                                                                                                                                               > > > > > > The freeswitch console only logs: > > 2020-12-26 16:32:49.578925 [DEBUG] sofia.c:7326 Channel > > sofia/external/*redacted*entering state [calling][0] > > 2020-12-26 16:32:49.618925 [DEBUG] sofia.c:7319 Channel > > sofia/external/*redacted* skipping state [ready][488] > > > > The call is disconnected 15 minutes later because the session timer has > > expired: > > 2020-12-26 16:47:53.838925 [NOTICE] sofia.c:1089 Hangup > > sofia/external/*redacted* [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > > > > This does not happen when I only allow PCMA (or any other codec) since > > the SDP can't get mixed up. > > > > I have reproduced that behaviour with > > 1.10.5~release~6~25569c1631~buster-1~buster+1 and > > 1.6.20~37~987c9b9-1~jessie+1. > > > > Do you have any idea why FS changes the SDP (without reason?) and what I > > can do about it? > > > > Thanks > > Nico > > > > > _________________________________________________________________________ > > > > 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 at 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 at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > https://freeswitch.com > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Mon Dec 28 18:19:26 2020 From: brian at freeswitch.com (Brian West) Date: Mon, 28 Dec 2020 12:19:26 -0600 Subject: [Freeswitch-users] Calls dropping due to SDP change In-Reply-To: References: <5ffc1416-8c70-ff43-3ac8-0fe6532efec2@vthadden.de> Message-ID: I've seen this problem before: the remote end is preferring PCMA over PCMU when it does the session timer, If you're talking to any providers in the EU, disallow PMCU always. Can you provide a full sip trace of this? On Mon, Dec 28, 2020 at 9:16 AM Nicola von Thadden wrote: > Hi, > > yes, we are using HD audio there. Deutsche Telekom is codec transparent, > they even pass opus and other codecs since a while. > So disabling any codecs but G.711 is not really a solution, especially > since DT is the biggest provider in Germany and that issue might not only > affect my setup but in theory all other Freeswitch in Germany here which > might want to call a DT landline or mobile number. > > Is there a way to make freeswitch stop mixing up the SDP? > > Nico > > On 12/28/20 12:04 AM, Brian : wrote: > > Unless you're doing hd audio on these calls your solution is to just use > g711 on calls into this provider. > > On Saturday, December 26, 2020, Nicola von Thadden > wrote: > > Hi, > > > > I'm currently investigating reproducable call drops when calling > > mobile-phone numbers from Deutsche Telekom (T-Mobile Germany). > > My provider has a NGN (SIP) Interconnection with Deutsche Telekom and is > > transparend for the messages involved here (although IP rewriting is > > happening). > > > > The calls contain following SDP in the invite from my FreeSwitch towards > > my provider: > > > v=0 > > > > o=FreeSWITCH 1608967203 1608967204 IN IP4 *redacted* > > > > > > > > s=FreeSWITCH > > > > c=IN IP4 *redacted* > > > > > > > t=0 > > > 0 > > > > m=audio 32256 RTP/AVP 9 8 0 > > > 101 > > > > a=rtpmap:9 > > > G722/8000 > > > > a=rtpmap:8 > > > PCMA/8000 > > > > a=rtpmap:0 > > > PCMU/8000 > > > > a=rtpmap:101 > > > telephone-event/8000 > > > > a=fmtp:101 > > > 0-16 > > > > a=ptime:20 > > > > Once the call is established, FS sends a re-invite after 50% of the > > expiration timer is elapsed, 15 minutes in this case. The re-invite > > contains a slightly modified SDP: > > > v=0 > > > > o=FreeSWITCH 1608967203 1608967204 IN IP4 > > > *redacted* > > > > > s=FreeSWITCH > > > > c=IN IP4 > > > *redacted* > > > > t=0 > > > 0 > > > > m=audio 32256 RTP/AVP 8 101 9 > > > 0 > > > > a=rtpmap:8 > > > PCMA/8000 > > > > a=rtpmap:101 > > > telephone-event/8000 > > > > a=fmtp:101 > > > 0-16 > > > > a=rtpmap:9 > > > G722/8000 > > > > a=rtpmap:0 > > > PCMU/8000 > > > > a=ptime:20 > > > > The new codec on position 1 (PCMA) is not necessary the chosen one for > > the session, that call was using G.722 (verified via 'show channels'). > > > > Telekom does not like my SDP change and responds with: > > SIP/2.0 488 SDP Parameter Error In SIP > > > Request > > > > > > The freeswitch console only logs: > > 2020-12-26 16:32:49.578925 [DEBUG] sofia.c:7326 Channel > > sofia/external/*redacted*entering state [calling][0] > > 2020-12-26 16:32:49.618925 [DEBUG] sofia.c:7319 Channel > > sofia/external/*redacted* skipping state [ready][488] > > > > The call is disconnected 15 minutes later because the session timer has > > expired: > > 2020-12-26 16:47:53.838925 [NOTICE] sofia.c:1089 Hangup > > sofia/external/*redacted* [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > > > > This does not happen when I only allow PCMA (or any other codec) since > > the SDP can't get mixed up. > > > > I have reproduced that behaviour with > > 1.10.5~release~6~25569c1631~buster-1~buster+1 and > > 1.6.20~37~987c9b9-1~jessie+1. > > > > Do you have any idea why FS changes the SDP (without reason?) and what I > > can do about it? > > > > Thanks > > Nico > > > > _________________________________________________________________________ > > > > 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 at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com > > _________________________________________________________________________ > > 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 Servicessales at freeswitch.comhttps://freeswitch.com > > Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com > > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From asilva at wirelessmundi.com Tue Dec 29 11:44:15 2020 From: asilva at wirelessmundi.com (=?utf-8?Q?Ant=C3=B3nio_Silva?=) Date: Tue, 29 Dec 2020 11:44:15 +0000 Subject: Using turn relay in verto client Message-ID: Hi, I want to share my experience with verto client and nat. I’m facing a delay issue with verto clients behind some types of nat (symmetric and port restricted cone nat), the port in the SDP is not the same for media and that introduces 3 seconds delay at the begging of the call. It only occurs in the direction Client -> Freeswitch. From the capture I notice that this delay is cause in the dlls negotiation, the Client Hello is sent to a port that is closed at the client side, after 3seconds FS detects that the rtp traffic is arriving in a natted port and it now send the Client Hello to that port. FS —> Client (pkt flow): 1s. Client Hello ——> :49897 <— (port unreached) 2s. Client Hello ——> :49897 <— (port unreached) 3s. Client Hello ——> :1025 <— Server Hello Certificate —> :1025 (Normal audio) Currently I solve it using turn server in the client and forcing to present only relay candidate to FS, DTLS is negotiated immediately, the only issue I see is that FS is not using the relay server (my configuration), it communicates directly to the client so it detects the change of address during media rtp: 1s. Client Hello ——> relay:49897 ——> client:49897 <— Server Hello:49897 FS RTP —> Relay: 49897 ——> client:49897 FS <— RTP from client address: 49897 (FS detects address change, stop send it to relay address and with to client address) FS RTP —> client:49897 The address change detection is perfect in this situation, we only load the relay server for dtls negotiation, then the media flows directly between FS and Client. What do you think of this approach? Do you have a better solution for this problem? -- Saludos / Regards / Cumprimentos António Silva -------------- next part -------------- An HTML attachment was scrubbed... URL: From mrjoli021 at gmail.com Tue Dec 29 23:59:08 2020 From: mrjoli021 at gmail.com (Joli Martinez) Date: Tue, 29 Dec 2020 18:59:08 -0500 Subject: [Freeswitch-users] docker image In-Reply-To: References: Message-ID: Hello, I am trying to download the image from DockerHub but get the following error message? Has the image been moved or is there something that I am missing? docker run --net=host --name freeswitch \ > -e SOUND_RATES=8000:16000 \ > -e SOUND_TYPES=music:en-us-callie \ > -v freeswitch-sounds:/usr/share/freeswitch/sounds \ > -v /etc/freeswitch/:/etc/freeswitch \ > safarov/freeswitch Unable to find image 'safarov/freeswitch:latest' locally docker: Error response from daemon: manifest for safarov/freeswitch:latest not found: manifest unknown: manifest unknown. See 'docker run --help'. On Mon, Dec 28, 2020 at 4:07 AM Sergey Safarov wrote: > You can check > https://hub.docker.com/repository/docker/safarov/freeswitch > > About 100 Mb size, based on Debian > > On Mon, Dec 28, 2020 at 7:07 AM Joel Serrano wrote: > >> Hi David, >> >> Out of curiosity, why do you need supervisord in your dockerfile? >> >> >> On Sun, Dec 27, 2020 at 14:13 David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> This might give you a starting point >>> >>> https://github.com/davidcsi/docker-freeswitch >>> >>> Regards >>> >>> On Sun, 27 Dec 2020 at 20:46, Joli Martinez wrote: >>> >>>> Hello, >>>> >>>> Our company is pushing toward Docker now. That is the new focus for >>>> next year. Is there a Freeswitch Docker image already available? I >>>> searched on Dockerhub and could not find one. If there is not one avail. >>>> I was thinking of building one. I was planning on using the latest Debian >>>> base image and creating a Dockerfile with the installation of FreeSwitch. >>>> Is that the best approach or is there a better alternative? >>>> >>>> Also is there any reason not to Dockerize Freeswitch? >>>> >>>> Thanks, >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> 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 at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> -- >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 >>> _________________________________________________________________________ >>> >>> 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 at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From nneul at mst.edu Wed Dec 30 02:27:42 2020 From: nneul at mst.edu (Nathan Neulinger) Date: Tue, 29 Dec 2020 20:27:42 -0600 Subject: [Freeswitch-users] docker image In-Reply-To: References: Message-ID: There is no latest tag. Go view the repo at the provided link and pick appropriate version/tag in your docker command. i.e. ... safarov/freeswitch:1.10.3 -- Nathan ------------------------------------------------------------------------------------------------------------------------ *From:* Joli Martinez [mailto:mrjoli021 at gmail.com] *Sent:* Tuesday, December 29, 2020, 5:59 PM *To:* FreeSWITCH Users Help *Subject:* [Freeswitch-users] docker image > Hello, > > I am trying to download the image from DockerHub but get the following error message?  Has the image been moved or is > there something that I am missing? > > docker run --net=host --name freeswitch \ > >   -e SOUND_RATES=8000:16000 \ > >   -e SOUND_TYPES=music:en-us-callie \ > >   -v freeswitch-sounds:/usr/share/freeswitch/sounds \ > >   -v /etc/freeswitch/:/etc/freeswitch \ > >   safarov/freeswitch > > Unable to find image 'safarov/freeswitch:latest' locally > > docker: Error response from daemon: manifest for safarov/freeswitch:latest not found: manifest unknown: manifest unknown. > > See 'docker run --help'. > > > > > On Mon, Dec 28, 2020 at 4:07 AM Sergey Safarov > wrote: > > You can check > https://hub.docker.com/repository/docker/safarov/freeswitch > > > About 100 Mb size, based on Debian > > On Mon, Dec 28, 2020 at 7:07 AM Joel Serrano > wrote: > > Hi David, > > Out of curiosity, why do you need supervisord in your dockerfile? > > > On Sun, Dec 27, 2020 at 14:13 David Villasmil > wrote: > > This might give you a starting point > > https://github.com/davidcsi/docker-freeswitch > > Regards > > On Sun, 27 Dec 2020 at 20:46, Joli Martinez > wrote: > > Hello, > > Our company is pushing toward Docker now.  That is the new focus for next year.  Is there a Freeswitch > Docker image already available? I searched on Dockerhub and could not find one.  If there is not one > avail.  I was thinking of building one.  I was planning on using the latest Debian base image and > creating a Dockerfile with the installation of FreeSwitch.  Is that the best approach or is there a > better alternative? > > Also is there any reason not to Dockerize Freeswitch? > > Thanks, > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com > > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 341-6679 System Administrator - Architect (573) 612-1412 System and Desktop Infrastructure Team Manager -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Wed Dec 30 09:31:10 2020 From: s.safarov at gmail.com (Sergey Safarov) Date: Wed, 30 Dec 2020 12:31:10 +0300 Subject: [Freeswitch-users] docker image In-Reply-To: References: Message-ID: Yes, Nathan right. No "latest" tag in repo. I have just added this tag. On Wed, Dec 30, 2020 at 5:38 AM Nathan Neulinger wrote: > There is no latest tag. Go view the repo at the provided link and pick > appropriate version/tag in your docker command. > > i.e. ... safarov/freeswitch:1.10.3 > > -- Nathan > > ------------------------------ > *From:* Joli Martinez [mailto:mrjoli021 at gmail.com ] > *Sent:* Tuesday, December 29, 2020, 5:59 PM > *To:* FreeSWITCH Users Help > > *Subject:* [Freeswitch-users] docker image > > Hello, > > I am trying to download the image from DockerHub but get the following > error message? Has the image been moved or is there something that I > am missing? > > docker run --net=host --name freeswitch \ > > -e SOUND_RATES=8000:16000 \ > > -e SOUND_TYPES=music:en-us-callie \ > > -v freeswitch-sounds:/usr/share/freeswitch/sounds \ > > -v /etc/freeswitch/:/etc/freeswitch \ > > safarov/freeswitch > > Unable to find image 'safarov/freeswitch:latest' locally > > docker: Error response from daemon: manifest for safarov/freeswitch:latest > not found: manifest unknown: manifest unknown. > > See 'docker run --help'. > > > > > On Mon, Dec 28, 2020 at 4:07 AM Sergey Safarov > wrote: > >> You can check >> https://hub.docker.com/repository/docker/safarov/freeswitch >> >> About 100 Mb size, based on Debian >> >> On Mon, Dec 28, 2020 at 7:07 AM Joel Serrano wrote: >> >>> Hi David, >>> >>> Out of curiosity, why do you need supervisord in your dockerfile? >>> >>> >>> On Sun, Dec 27, 2020 at 14:13 David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> This might give you a starting point >>>> >>>> https://github.com/davidcsi/docker-freeswitch >>>> >>>> Regards >>>> >>>> On Sun, 27 Dec 2020 at 20:46, Joli Martinez >>>> wrote: >>>> >>>>> Hello, >>>>> >>>>> Our company is pushing toward Docker now. That is the new focus for >>>>> next year. Is there a Freeswitch Docker image already available? I >>>>> searched on Dockerhub and could not find one. If there is not one avail. >>>>> I was thinking of building one. I was planning on using the latest Debian >>>>> base image and creating a Dockerfile with the installation of FreeSwitch. >>>>> Is that the best approach or is there a better alternative? >>>>> >>>>> Also is there any reason not to Dockerize Freeswitch? >>>>> >>>>> Thanks, >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> 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 at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> https://freeswitch.com >>>> >>>> -- >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work at gmail.com >>>> phone: +34669448337 >>>> >>>> _________________________________________________________________________ >>>> >>>> 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 at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> _________________________________________________________________________ >>> >>> 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 at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > _________________________________________________________________________ > > 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 Servicessales at freeswitch.comhttps://freeswitch.com > > Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com > > -- > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 341-6679 > System Administrator - Architect (573) 612-1412 > System and Desktop Infrastructure Team Manager > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From arsperger at gmail.com Mon Dec 28 10:08:37 2020 From: arsperger at gmail.com (Arsen Semenov) Date: Mon, 28 Dec 2020 15:08:37 +0500 Subject: [Freeswitch-users] docker image In-Reply-To: References: Message-ID: here is my attempt to make freeswitch dockerized https://github.com/arsperger/docker-freeswitch On Mon, Dec 28, 2020 at 1:49 PM Sergey Safarov wrote: > You can check > https://hub.docker.com/repository/docker/safarov/freeswitch > > About 100 Mb size, based on Debian > > On Mon, Dec 28, 2020 at 7:07 AM Joel Serrano wrote: > >> Hi David, >> >> Out of curiosity, why do you need supervisord in your dockerfile? >> >> >> On Sun, Dec 27, 2020 at 14:13 David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> This might give you a starting point >>> >>> https://github.com/davidcsi/docker-freeswitch >>> >>> Regards >>> >>> On Sun, 27 Dec 2020 at 20:46, Joli Martinez wrote: >>> >>>> Hello, >>>> >>>> Our company is pushing toward Docker now. That is the new focus for >>>> next year. Is there a Freeswitch Docker image already available? I >>>> searched on Dockerhub and could not find one. If there is not one avail. >>>> I was thinking of building one. I was planning on using the latest Debian >>>> base image and creating a Dockerfile with the installation of FreeSwitch. >>>> Is that the best approach or is there a better alternative? >>>> >>>> Also is there any reason not to Dockerize Freeswitch? >>>> >>>> Thanks, >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> 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 at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> -- >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 >>> _________________________________________________________________________ >>> >>> 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 at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> >> 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Arsen Semenov -------------- next part -------------- An HTML attachment was scrubbed... URL: From fosdem-rtc-admin at freertc.org Wed Dec 23 19:53:35 2020 From: fosdem-rtc-admin at freertc.org (FOSDEM RTC Team) Date: Wed, 23 Dec 2020 20:53:35 +0100 (CET) Subject: [Freeswitch-users] Fwd: [CFP] FOSDEM 2021, RTC devroom, speakers, volunteers neeeded References: <8a889518-4558-11eb-8a0e-7b0d1f2b0241@freertc.org> Message-ID: <20201223195335.0EE2118046CA@ws3.office.readytechnology.co.uk> FOSDEM - Real Time Communications devroom CfP ============================================= NOTE: we have extended the deadline but we will give preference to people who submitted earlier when we set the time for each talk. Please submit your proposal ASAP so that you can get your preferred time slot. Overview -------- [FOSDEM](https://fosdem.org) is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2021 takes place 6-7 February 2021 and for the very first time, it will be online. This document contains information about: - Real-Time Communications developer room (devroom) - speaking opportunities - volunteering New rules for the online edition -------------------------------- This year FOSDEM will be fully online instead of being held in Brussels, here are the most important things to know about this (quite significant) change: - The reference time will be Brussels local time (CET) - Talks will be pre-recorded in advance, and streamed during the event - The Q/A session will be live - A facility will be provided for people watching to chat between themselves - A facility will be provided for people watching to submit questions Call for participation - Real Time Communications (RTC) ------------------------------------------------------- The Real-Time devroom is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. **We are looking for speakers for the devroom and volunteers who can help manage the scheduling and live Q&A sessions.** The devroom is only on Saturday, 6th of February 2021. To discuss the devroom, volunteer or ask questions, please join the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). ### Key dates - 20th Dec: Submission deadline (extended to 8 January) - 24th Dec: Announcement of selected talks - 15th Jan: Presentations upload deadline - 6th & 7th Feb: Conference dates (online) - 6th Feb: RTC devroom date (online) ### Speaking opportunities Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications devroom: deadline 23:59 UTC on 20th of December. Please use the [Pentabarf](https://penta.fosdem.org/submission/FOSDEM21/) system to submit a talk proposal for the devroom. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". ### First-time speaking? FOSDEM devrooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the devroom administrators personally if you would like to ask any questions about it. This year this is more true than ever, being able to record your presentation offline without an audience in front can greatly help build up one's confidence! ### Submission guidelines The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - The purpose of your talk - Any other talk applications (devrooms, lightning talks, main track) - Availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and devroom themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the devroom administrators based on the number of received proposals. As the two previous devrooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. ### Recording help The devroom organization is able to provide help with recording your session. The recording would be performed at a scheduled time with one of us, so you won't be alone giving your presentation. Minimal edits will be possible, but the ideal plan is to record it in one shot. Thanks Dan Jenkins for providing us with the means to do this! Volunteers needed ----------------- To make the devroom run successfully, we are looking for volunteers. This year many things be done for the first time, so all the help we can get is more than welcome. Spread the word and discuss --------------------------- If you know of any mailing lists where this CfP would be relevant, please forward this document. If this devroom excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: - All projects https://planet.freertc.org planet at freertc.org - XMPP https://planet.jabber.org ralphm at ik.nu - SIP https://planet.sip5060.net planet at sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ------- For any private queries, contact us directly using the address **fosdem-rtc-admin at freertc.org** and for any other queries please ask on the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). The devroom administration team: - Saúl Ibarra Corretgé - Ralph Meijer - Daniel-Constantin Mierla - Daniel Pocock - Guus der Kinderen From kitchm at tutanota.com Thu Dec 24 17:15:31 2020 From: kitchm at tutanota.com (KitchM) Date: Thu, 24 Dec 2020 10:15:31 -0700 (MST) Subject: Rings Versus Seconds Message-ID: <1608830131591-0.post@n2.nabble.com> The system appears to use seconds instead of rings to measure incoming call timing. Is that correct? If so, is there a way to change things to use number or rings instead? -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/