From eschmidbauer at gmail.com Fri Jan 1 17:25:38 2021 From: eschmidbauer at gmail.com (E. Schmidbauer) Date: Fri, 1 Jan 2021 12:25:38 -0500 Subject: [Freeswitch-users] docker image In-Reply-To: References: Message-ID: https://github.com/voipxswitch/freeswitch-docker This repo demonstrates building your own docker image using DIND On Wed, Dec 30, 2020, 4:31 AM Sergey Safarov wrote: > 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 > > _________________________________________________________________________ > > 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 martyn at magiccow.co.uk Tue Jan 5 11:32:29 2021 From: martyn at magiccow.co.uk (Martyn Davies) Date: Tue, 5 Jan 2021 11:32:29 +0000 Subject: [Freeswitch-users] Automatically authenticating client Message-ID: I'm trying to support a downstream client (actually a phone on a VoIP gateway) that must work without sending a registration to Freeswitch. I have a definition in the client directory that looks like this: : So, I'm assuming that if the source address matches 192.168.1.x, an invite from 0101 will automatically be authenticated, and the user context will be set to 'default' to go through the dialplan. However, what seems to happen is that the dialplan gets executed in the public context, 0101 doesn't match anything, and so the call fails. I also tried adding entries to the acl.conf file, e.g. e.g. or or adding an 'allow' line to the existing 'domains' list. When I run 'reloadacl', it looks like the rules are being picked up, e.g. 2021-01-05 11:06:07.631090 [NOTICE] switch_utils.c:642 Adding 192.168.1.0/24 (allow) [] to list localnet.auto 2021-01-05 11:06:07.631090 [NOTICE] switch_utils.c:642 Adding 192.168.1.97/32 (allow) [0101 at 192.168.1.241] to list domains However, none of this works, and I still end up in the public context when a call is placed. I would appreciate any examples or advice that people could offer to get my client authenticated, or could anyone give me pointers on how to debug the authentication process when the invite arrives? Regards, Martyn -------------- next part -------------- An HTML attachment was scrubbed... URL: From botelist at gmail.com Tue Jan 5 20:40:46 2021 From: botelist at gmail.com (Bote Man) Date: Tue, 5 Jan 2021 15:40:46 -0500 Subject: [Freeswitch-users] Automatically authenticating client In-Reply-To: References: Message-ID: <000a01d6e3a3$0bd3e9a0$237bbce0$@gmail.com> You can test against any channel variable in the dialplan, not just the destination number. It sounds like a shady workaround, but you could test for a unique information element sent by the remote VoIP device in the public dialplan, then if it matches transfer it to the desired (default, I imagine?) dialplan along with the destination number. This is the way I handle incoming calls from my VoIP provider on my home FS box, along with a few other tests for security and sanity preservation. Where sip_to_user is *my* number, in other words the number that the caller dialed to reach me. Then I have an extension in the default dialplan that looks for “target-nummer” and routes it to a bunch of phones in my house. In your case I’m guessing you would probably want to change target-nummer to a variable that contains what the remote phone actually dialed. FS is very flexible in this way. Just make sure you test enough conditions that only the desired remote VoIP device can route to the default dialplan this way and you should be fine. FWIW, I have never been able to figure out the ACL system in FS. It has some unexpected interactions, notably with the status subscriptions so I just went back to the conventional challenge/authentication system so that it works as expected. Sometimes I think it helped to stop and restart FS rather than just issue the reloadacl command sometimes. Hope this helps. John Boteler Bote Communications From: Martyn Davies Sent: Tuesday, 5 January, 2021 06:32 To: FreeSWITCH Users Help Subject: [Freeswitch-users] Automatically authenticating client I'm trying to support a downstream client (actually a phone on a VoIP gateway) that must work without sending a registration to Freeswitch. I have a definition in the client directory that looks like this: : So, I'm assuming that if the source address matches 192.168.1.x, an invite from 0101 will automatically be authenticated, and the user context will be set to 'default' to go through the dialplan. However, what seems to happen is that the dialplan gets executed in the public context, 0101 doesn't match anything, and so the call fails. I also tried adding entries to the acl.conf file, e.g. e.g. or or adding an 'allow' line to the existing 'domains' list. When I run 'reloadacl', it looks like the rules are being picked up, e.g. 2021-01-05 11:06:07.631090 [NOTICE] switch_utils.c:642 Adding 192.168.1.0/24 (allow) [] to list localnet.auto 2021-01-05 11:06:07.631090 [NOTICE] switch_utils.c:642 Adding 192.168.1.97/32 (allow) [0101 at 192.168.1.241 ] to list domains However, none of this works, and I still end up in the public context when a call is placed. I would appreciate any examples or advice that people could offer to get my client authenticated, or could anyone give me pointers on how to debug the authentication process when the invite arrives? Regards, Martyn -------------- next part -------------- An HTML attachment was scrubbed... URL: From unai at sysbible.org Tue Jan 5 17:14:04 2021 From: unai at sysbible.org (Unai Rodriguez) Date: Tue, 05 Jan 2021 18:14:04 +0100 Subject: [Freeswitch-users] Inverted FROM/TO headers for BYE requests Message-ID: <43c82bad-70ca-4f7e-b43e-137fb02c0923@www.fastmail.com> Dear List, We are having FreeSWITCH behind AVAYA infrastructure and when FreeSWITCH sends a BYE the From and the To headers are inverted. For example this call was initiated by John Smith however the BYE request has the From and To inverted: send 619 bytes to tcp/[10.125.15.122]:5060 at 16:57:04.079105: ------------------------------------------------------------------------ BYE sip:5188 at 10.125.15.122:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.155.20.104;branch=z9hG4bKa99Q7j7crjK9K Max-Forwards: 70 From: ;tag=Ze68N90tZ1D0c To: "John Smith" ;tag=54578355~d340a1b2-9d48-464d-9060-03cf872440b9-97644820 Call-ID: 2ad6b00-ff419a57-85bc53-66199b0a at 10.125.15.122 CSeq: 30379120 BYE User-Agent: FreeSWITCH-mod_sofia/1.8.6~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 These seems to work for Cisco and other Telephony products but not the particular AVAYA system we're working with. Is there any way to modify this behavior and/or is this a known issue? Thank you so much -- unai -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Tue Jan 5 21:46:38 2021 From: brian at freeswitch.com (Brian West) Date: Tue, 5 Jan 2021 15:46:38 -0600 Subject: [Freeswitch-users] Inverted FROM/TO headers for BYE requests In-Reply-To: <43c82bad-70ca-4f7e-b43e-137fb02c0923@www.fastmail.com> References: <43c82bad-70ca-4f7e-b43e-137fb02c0923@www.fastmail.com> Message-ID: To/From are actually meaningless, what exactly are you trying to fix? and you might wanna try a newer rev. /b On Tue, Jan 5, 2021 at 3:42 PM Unai Rodriguez wrote: > Dear List, > > We are having FreeSWITCH behind AVAYA infrastructure and when FreeSWITCH > sends a BYE the From and the To headers are inverted. For example this call > was initiated by John Smith however the BYE request has the From and To > inverted: > > send 619 bytes to tcp/[10.125.15.122]:5060 at 16:57:04.079105: > ------------------------------------------------------------------------ > BYE sip:5188 at 10.125.15.122:5060;transport=tcp SIP/2.0 > Via: SIP/2.0/TCP 10.155.20.104;branch=z9hG4bKa99Q7j7crjK9K > Max-Forwards: 70 > From: ;tag=Ze68N90tZ1D0c > To: "John Smith" >;tag=54578355~d340a1b2-9d48-464d-9060-03cf872440b9-97644820 > Call-ID: 2ad6b00-ff419a57-85bc53-66199b0a at 10.125.15.122 > CSeq: 30379120 BYE > User-Agent: FreeSWITCH-mod_sofia/1.8.6~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > Reason: Q.850;cause=16;text="NORMAL_CLEARING" > Content-Length: 0 > > These seems to work for Cisco and other Telephony products but not the > particular AVAYA system we're working with. > > Is there any way to modify this behavior and/or is this a known issue? > > Thank you so much > > -- unai > > _________________________________________________________________________ > > 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 davidswalkabout at gmail.com Tue Jan 5 22:02:17 2021 From: davidswalkabout at gmail.com (David P) Date: Wed, 6 Jan 2021 11:02:17 +1300 Subject: [Freeswitch-users] TLS 1.3 with verto Message-ID: We use FS 10.5 on Debian for verto calls with setting sip_tls_version=tlsv1.2 and an apache2 reverse proxy for WebSocket logins so there's no port number in the WSS url so there should be no problem with restrictive firewalls for WSS login. In recent months we've seen some of the login attempts timeout, and there's nothing in the FS log at debug level to indicate why. But after a few minutes, verto's reattempts succeed in logging in. Has anyone else experienced this and found the cause? I thought I found the reason in our apache error.log, because it shows that some access attempts use TLSv1.3. I tried to get apache to reject these by switching its config to... SSLProtocol -all +TLSv1.2 ...but the TLSv1.3 attempts still behave the same. (Also, although these attempts appear in error.log, there's no hint about why they are in this log instead of access.log.) Can someone confirm that FS 10.5 doesn't yet support TLSv1.3? Cheers, David -------------- next part -------------- An HTML attachment was scrubbed... URL: From rich.freeswitch at branham.us Wed Jan 6 00:52:33 2021 From: rich.freeswitch at branham.us (Richard-Freeswitch) Date: Tue, 05 Jan 2021 19:52:33 -0500 Subject: [Freeswitch-users] Error for emails related to faxing Message-ID: <2207e735a000e9a0793bc0e3ce2fac1d@branham.us> We have users who report email failures related to fax notifications. I'm digging for further information on how to reproduce the issue, but so far the only error in the FreeSWITCH log is: EMAIL NOT SENT, error [Cannot open tmp file In spandsp.conf.xml, spool-dir is set to /tmp. The directory exists and is world readable and writeable. fs_cli -x version returns: FreeSWITCH Version 1.10.5-release-17-25569c1631~64bit (-release-17-25569c1631 64bit) Any idea what would cause the EMAIL NOT SENT error? -------------- next part -------------- An HTML attachment was scrubbed... URL: From lexxua at gmail.com Wed Jan 6 10:37:35 2021 From: lexxua at gmail.com (Volodymyr Fedorov) Date: Wed, 6 Jan 2021 11:37:35 +0100 Subject: [Freeswitch-users] Rings Versus Seconds In-Reply-To: References: Message-ID: Hi Ring tone duration depends on the country you live in: https://en.wikipedia.org/wiki/Ringing_tone You may try to do calculations by your own. BR, Vova On Tue, Jan 5, 2021 at 10:57 PM KitchM via FreeSWITCH-users wrote: > > > > > ---------- Forwarded message ---------- > From: KitchM > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Thu, 24 Dec 2020 10:15:31 -0700 (MST) > Subject: Rings Versus Seconds > 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/ > > > > > ---------- Forwarded message ---------- > From: KitchM via FreeSWITCH-users > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Tue, 05 Jan 2021 13:57:28 -0800 (PST) > Subject: [Freeswitch-users] Rings Versus Seconds > _________________________________________________________________________ > > 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, Volodymyr From ak at hejdu.dk Wed Jan 6 18:37:33 2021 From: ak at hejdu.dk (Allan Kristensen) Date: Wed, 6 Jan 2021 19:37:33 +0100 Subject: [Freeswitch-users] rtp-timeout-sec VS media_timeout Message-ID: We had some problems with "hanging channels" for our webrtc clients (via kamailio). To solve the problem I tried to use "media_timeout" setting but it didn't really work. So I tried the deprecated "rtp-timeout-sec" and this actually works fine? Not working: Working: How to reproduce: Make a call using webrtc and just close browser window, after some time freeswitch should close the channel because of missing rtp. Instead it just keeps writing: "switch_rtp.c:853 No audio stun for a long time!" forever... Anyway, it works now....just curious why...a typo or bug? /Allan Using: FreeSWITCH Version 1.10.5-release+git~20200818T185121Z~25569c1631~64bit -------------- next part -------------- An HTML attachment was scrubbed... URL: From cong.wang.itsherpa at gmail.com Thu Jan 7 03:34:48 2021 From: cong.wang.itsherpa at gmail.com (=?utf-8?B?546L6IGh?=) Date: Thu, 7 Jan 2021 12:34:48 +0900 Subject: [Freeswitch-users] Is [say] app removed from freeswitch? Message-ID: Hello all, We’re trying to let freeswitch read some given words in Japanese, and I found it is possible via [say] app from documents: https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+say However, [say] app hadn’t found in version 1.10.5, only [say_string] app was found, but it seems not support Japanese. * module [mod_say_ja] has loaded: [CONSOLE] switch_loadable_module.c:1803 Successfully Loaded [mod_say_ja] [NOTICE] switch_loadable_module.c:680 Adding Say interface ‘ja' * call [say] in dialplan: [ERR] switch_core_state_machine.c:276 Dialplan [say] not found, skipping * call [say_string] in dialplan: [ERR] switch_xml.c:3274 Can't find language ja. I wonder if [say] is no longer available in freeswitch, or there is a mistake on my deployment. Regards. —————————————————— 王聡 cong.wang.itsherpa at gmail.com 〒810−0073 福岡市中央区舞鶴2-3-6 赤坂プライムビル 2F 株式会社アイティーシェルパ -------------- next part -------------- An HTML attachment was scrubbed... URL: From tahir at ictinnovations.com Thu Jan 7 16:40:36 2021 From: tahir at ictinnovations.com (Tahir Almas) Date: Thu, 7 Jan 2021 21:40:36 +0500 Subject: [Freeswitch-users] ICTDialer Version 4.0 released Message-ID: Pleased to announce the release of ICTDialer Version 4.0 https://github.com/ictinnovations/ictdialer . ICTDialer is a freeswitch based unified communications auto dialling front end for mass communications and telemarketing needs. Fore more details , please click following link https://www.ictinnovations.com/ICTDialer-Version-4.0-released-open-source-freeswitch-autodialer Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -------------- next part -------------- An HTML attachment was scrubbed... URL: From tahir at ictinnovations.com Thu Jan 7 16:44:16 2021 From: tahir at ictinnovations.com (Tahir Almas) Date: Thu, 7 Jan 2021 21:44:16 +0500 Subject: [Freeswitch-users] ICTCore Version 1.1.0 Released Message-ID: Pleased to announce the release of ICTCore Version 1.1.0 https://github.com/ictinnovations/ictdialer. ICTCore is unified communciations development framework developed over freeswitch, Following you will find more details https://www.ictinnovations.com/ICTCore-Version-1.1.0-released-an-open-source-communications-and-api-provisioning-platform Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -------------- next part -------------- An HTML attachment was scrubbed... URL: From grant.bagdasarian at cm.com Thu Jan 7 10:49:14 2021 From: grant.bagdasarian at cm.com (Grant Bagdasarian | CM.com) Date: Thu, 7 Jan 2021 10:49:14 +0000 Subject: FROM/P-Asserted-Identity and anonymous Message-ID: Hi, I'm changing the value of the From URI, in the dialplan, based on whether the Privacy: id header is present. If it's present, the value of the From URI is set to sip:anonymous AT anonymous.invalid, else it is set to sip:PHONE_NUMBER AT X.X.X.X. This works fine, but for some reason, during the bridging of the call using a gateway (with cid_type set to pid), Freeswitch sets the host part of the P-Asserted-Identity URI to anonymous.invalid (only during anonymous calls of course). Is there a way to make Freeswitch leave the host part of the P-Asserted-Identity header alone and set it to its IP address, while still utilizing the cid_type variable? Actual: From: Privacy: id P-Asserted-Identity: "1234" Expected: From: Privacy: id P-Asserted-Identity: "1234" Removing the cid_type variable (or set to none), and inserting a P-Asserted-Identity header manually in the required format works fine. Also, when using the cid_type=pid, is it possible to add the user=phone param to the P-Asserted-Identity URI? Or does this also require setting the header manually? Regards, Grant -------------- next part -------------- An HTML attachment was scrubbed... URL: From grcamauer at gmail.com Thu Jan 7 19:54:49 2021 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Thu, 7 Jan 2021 16:54:49 -0300 Subject: [Freeswitch-users] ICTDialer Version 4.0 released In-Reply-To: References: Message-ID: I just looked at the install script and it seems to be using FreeSwitch 1.6. Does it work with the 1.10.x branch? Does it work on Debian? (I only saw instructions for CentOS) Guillermo On Thu, Jan 7, 2021 at 2:28 PM Tahir Almas wrote: > Pleased to announce the release of ICTDialer Version 4.0 > https://github.com/ictinnovations/ictdialer . > ICTDialer is a freeswitch based unified communications auto dialling > front end for mass communications and telemarketing needs. > > Fore more details , please click following link > > https://www.ictinnovations.com/ICTDialer-Version-4.0-released-open-source-freeswitch-autodialer > > Regards > *Tahir Almas* > > Managing Partner > ICT Innovations > http://www.ictinnovations.com > Leveraging open source in ICT > > _________________________________________________________________________ > > 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 -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From tahir at ictinnovations.com Fri Jan 8 11:56:03 2021 From: tahir at ictinnovations.com (Tahir Almas) Date: Fri, 8 Jan 2021 16:56:03 +0500 Subject: [Freeswitch-users] ICTDialer Version 4.0 released In-Reply-To: References: Message-ID: We have not tested with freeswitch 1.10.x however it should work with freeswitch 1.10.x We have plans to write installation instruction for debian in near future , I will update here regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Fri, Jan 8, 2021 at 1:20 AM Guillermo Ruiz Camauer wrote: > I just looked at the install script and it seems to be using FreeSwitch > 1.6. Does it work with the 1.10.x branch? > Does it work on Debian? (I only saw instructions for CentOS) > > Guillermo > > On Thu, Jan 7, 2021 at 2:28 PM Tahir Almas > wrote: > >> Pleased to announce the release of ICTDialer Version 4.0 >> https://github.com/ictinnovations/ictdialer . >> ICTDialer is a freeswitch based unified communications auto dialling >> front end for mass communications and telemarketing needs. >> >> Fore more details , please click following link >> >> https://www.ictinnovations.com/ICTDialer-Version-4.0-released-open-source-freeswitch-autodialer >> >> Regards >> *Tahir Almas* >> >> Managing Partner >> ICT Innovations >> http://www.ictinnovations.com >> Leveraging open source in ICT >> >> _________________________________________________________________________ >> >> 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 > > > > -- > Guillermo Ruiz Camauer > _________________________________________________________________________ > > 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 aryeklt at gmail.com Sun Jan 10 23:39:12 2021 From: aryeklt at gmail.com (=?UTF-8?B?15DXqNeZ15Qg16fXnNeY16g=?=) Date: Mon, 11 Jan 2021 01:39:12 +0200 Subject: [Freeswitch-users] play ringtone until connecting media Message-ID: Hello After play_and_get_digits, in some cases i am bridging the call to external server. The remote server answers with 200 OK after something like 0.3 second But, the remote server starting to send media only after 2.1s Any idea how to play ringtone (or another wav file) until the remote server really sends media? Regards Arye -------------- next part -------------- An HTML attachment was scrubbed... URL: From jolexpert at gmail.com Mon Jan 11 10:15:48 2021 From: jolexpert at gmail.com (Kakiman Expert) Date: Mon, 11 Jan 2021 11:15:48 +0100 Subject: [Freeswitch-users] call / session stats Message-ID: Hello, I am working on freeSWITCH 1.10.2 (with webRTC / OPUS) and looking for some calls statistics. Indeed, I saw that we can have real time stats (show calls/session...), but I don't find any way to have it later For example, having the registration or calls for a daily basis, or for a specific period. I just saw that with "show status" we have such a statistic but without any period/range information. 637 session(s) since startup / 0 session(s) - peak 4, last 5min 0 Is there a solution to have these reports ? thanks Joe -------------- next part -------------- An HTML attachment was scrubbed... URL: From kunaalz at gmail.com Mon Jan 11 20:18:21 2021 From: kunaalz at gmail.com (Kunal Mittal) Date: Mon, 11 Jan 2021 12:18:21 -0800 Subject: [Freeswitch-users] Freeswitch crash reloading mod_callcenter Message-ID: Hi, I have the latest version of the freeswitch 1.10.5 running with debian 10.7. It crashes everytime on reloading the mod_callcenter module after 3-4 minutes. Any suggestions? Is it a known bug? Any suggestions on how to troubleshoot this issue? Freeswitch version - 1.10.5 Debian - 10.7 Kunal -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Tue Jan 12 02:43:47 2021 From: brian at freeswitch.com (Brian West) Date: Mon, 11 Jan 2021 20:43:47 -0600 Subject: [Freeswitch-users] call / session stats In-Reply-To: References: Message-ID: Postprocess the XML CDRs /b On Mon, Jan 11, 2021 at 4:03 PM Kakiman Expert wrote: > Hello, > > I am working on freeSWITCH 1.10.2 (with webRTC / OPUS) and looking for > some calls statistics. > > Indeed, I saw that we can have real time stats (show calls/session...), > but I don't find any way to have it later > For example, having the registration or calls for a daily basis, or for a > specific period. > > I just saw that with "show status" we have such a statistic but without > any period/range information. > > 637 session(s) since startup / 0 session(s) - peak 4, last 5min 0 > > Is there a solution to have these reports ? > thanks > Joe > > > _________________________________________________________________________ > > 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 per at wgtwo.com Tue Jan 12 13:08:47 2021 From: per at wgtwo.com (Per Modin) Date: Tue, 12 Jan 2021 14:08:47 +0100 Subject: [Freeswitch-users] call / session stats In-Reply-To: References: Message-ID: <4f691cf6-1cd8-d381-5d6a-621dcfa24758@wgtwo.com> On 1/11/21 11:15 AM, Kakiman Expert wrote: > Indeed, I saw that we can have real time stats (show calls/session...), > but I don't find any way to have it later  > For example, having the registration or calls for a daily basis, or for > a specific period. We're using an in-house Prometheus exporter using ESL (which is then fed into Grafana), there's some projects around that might work as inspiration[0][1], or as-is. [0]: https://github.com/znerol/prometheus-freeswitch-exporter [1]: https://github.com/friends-of-freeswitch/mod_prometheus Best, Per. From bilaln018 at gmail.com Thu Jan 14 10:42:43 2021 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Thu, 14 Jan 2021 15:42:43 +0500 Subject: [Freeswitch-users] [FreeSWITCH sched_hangup does not execute 'execute_extension'] Message-ID: Hi Users, I want to build a scenario in which if the B-Party hangup the call, A-Party can still be able to process through the dialplan. This works fine if B-Party hangup the call, but if the B-Party is hanup forcefully via sched_hangup (timeout), call gets terminated i-e A-Party is not able to reach 'execute_extension' application. Here is my dialplan snippet https://justpaste.it/2ppra I have attached as an image as well. Regards Bilal Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: Screenshot 2021-01-14 at 3.38.02 PM.png Type: image/png Size: 53710 bytes Desc: not available URL: From brian at freeswitch.com Fri Jan 15 16:56:54 2021 From: brian at freeswitch.com (Brian West) Date: Fri, 15 Jan 2021 10:56:54 -0600 Subject: [Freeswitch-users] [FreeSWITCH sched_hangup does not execute 'execute_extension'] In-Reply-To: References: Message-ID: It's because you're executing it on the wrong leg. /b PS: Don't use nolocal. On Thu, Jan 14, 2021 at 5:25 AM Bilal Abbasi wrote: > Hi Users, > I want to build a scenario in which if the B-Party hangup the call, > A-Party can still be able to process through the dialplan. > This works fine if B-Party hangup the call, but if the B-Party is hanup > forcefully via sched_hangup (timeout), call gets terminated i-e A-Party is > not able to reach 'execute_extension' application. > > Here is my dialplan snippet > https://justpaste.it/2ppra > I have attached as an image as well. > > Regards > Bilal Abbasi > _________________________________________________________________________ > > 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 olle at zaark.com Mon Jan 18 08:27:48 2021 From: olle at zaark.com (olle at zaark.com) Date: Mon, 18 Jan 2021 09:27:48 +0100 Subject: [Freeswitch-users] Using last_bridge_hangup_cause Message-ID: <001b01d6ed73$ce05b1f0$6a1115d0$@zaark.com> Hi, I have a need of using last_bridge_hangup cause in a condition statement to activate a second call leg but I can't get it to work. The log line prints the correct values of last_bridge_hangup_cause, "NO_ANSWER" , but when I use it in the condition it's not working and the same if I try to assign it to another variable. Here is a code snippet I use in the dialplan: .. Do some stuff ... I would prefer using the dialplan without implementing this in lua if possible. Any help appreciated /Olle -------------- next part -------------- An HTML attachment was scrubbed... URL: From jacobgreene1991 at gmail.com Wed Jan 13 22:02:30 2021 From: jacobgreene1991 at gmail.com (Jacob Greene) Date: Wed, 13 Jan 2021 16:02:30 -0600 Subject: [Freeswitch-users] AWS - Audio Delay - rtp timer Message-ID: Hello everyone, Recently, I am running into an issue with some severe audio delay with some freeswitch boxes running on AWS. It gets to the point where audio is delayed up to 10 seconds in both directions. The calls start off fine and slowly get worse. The only way I've been able to fix this is by disabling the rtp timer on the sofia profile(rtp-timer-name=none). Making this change immediately fixes the issue. No obvious system bottlenecks, 1/5/15 min load averages all less 1 on dual core CPUs, tons of free memory, no I/O issues on the nic, etc. I suspect this might have something to do with some weird CPU sharing AWS is doing on their hypervisor, but I'm not sure. I can't find a ton of info on exactly what the rtp timer does, other than disabling it "disables asynchronous rtp" and "makes freeswitch handle media the same way as asterisks". I'm not really sure what this means. I've read it's less effecient and come across a couple very old post on the mailing list of Anthony steering people away from disabling the rtp-timer. Has anyone had a similar problem? Or does anyone have some more information/resources regarding exactly what the function of the rtp timer is? I'm trying to get in front of this. What are the drawbacks of not using an rtp timer? Thanks for reading! -------------- next part -------------- An HTML attachment was scrubbed... URL: From m.hald at outlook.com Thu Jan 14 15:14:34 2021 From: m.hald at outlook.com (Marcel Haldemann) Date: Thu, 14 Jan 2021 15:14:34 +0000 Subject: [Freeswitch-users] [FreeSWITCH sched_hangup does not execute 'execute_extension'] In-Reply-To: References: Message-ID: Since you hangup with an error-code (ALLOTTED_TIMEOUT) you maybe need to set : Before the bridge. https://freeswitch.org/confluence/display/FREESWITCH/continue_on_fail Von: FreeSWITCH-users Im Auftrag von Bilal Abbasi Gesendet: Donnerstag, 14. Januar 2021 11:43 An: FreeSWITCH Users Help Betreff: [Freeswitch-users] [FreeSWITCH sched_hangup does not execute 'execute_extension'] Hi Users, I want to build a scenario in which if the B-Party hangup the call, A-Party can still be able to process through the dialplan. This works fine if B-Party hangup the call, but if the B-Party is hanup forcefully via sched_hangup (timeout), call gets terminated i-e A-Party is not able to reach 'execute_extension' application. Here is my dialplan snippet https://justpaste.it/2ppra I have attached as an image as well. Regards Bilal Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: From jolexpert at gmail.com Fri Jan 15 14:11:17 2021 From: jolexpert at gmail.com (Kakiman Expert) Date: Fri, 15 Jan 2021 15:11:17 +0100 Subject: [Freeswitch-users] external ip and Nat Message-ID: Hello I am working on freeSWITCH 1.10.2 with webRTC. I have a private address on my server, that is translated on a static public IP in order to reach webRTC/FS through Internet. In the vars.xml, i have then indicated this public IP as *external_sip_ip* and *external_rtp_ip *and everything is working. Now, I also want to be able to reach webRTC with its private IP, so what is the value to configure in vars.xml external_sip and external_rtp ? I can not put both public and private IP. thanks for your helpful answers Joe -------------- next part -------------- An HTML attachment was scrubbed... URL: From kunaalz at gmail.com Fri Jan 15 18:42:03 2021 From: kunaalz at gmail.com (Kunal Mittal) Date: Fri, 15 Jan 2021 10:42:03 -0800 Subject: [Freeswitch-users] Freeswitch crash reloading mod_callcenter In-Reply-To: References: Message-ID: Checking to see if anyone has any suggestions on this? On Mon, Jan 11, 2021 at 12:18 PM Kunal Mittal wrote: > Hi, > I have the latest version of the freeswitch 1.10.5 running with debian > 10.7. It crashes everytime on reloading the mod_callcenter module after > 3-4 minutes. Any suggestions? Is it a known bug? Any suggestions on how > to troubleshoot this issue? > > Freeswitch version - 1.10.5 > Debian - 10.7 > > Kunal > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jurijs.ivolga at gmail.com Mon Jan 18 15:27:10 2021 From: jurijs.ivolga at gmail.com (Jurijs Ivolga) Date: Mon, 18 Jan 2021 17:27:10 +0200 Subject: [Freeswitch-users] AWS - Audio Delay - rtp timer In-Reply-To: References: Message-ID: Hi, I got similar problems. For me solution was to enable jitter buffer, something like this: .... .... In this case I didn't get delay when I set "" in sip profile. Jurijs On Mon, Jan 18, 2021 at 5:10 PM Jacob Greene wrote: > Hello everyone, > > Recently, I am running into an issue with some severe audio delay with > some freeswitch boxes running on AWS. It gets to the point where audio is > delayed up to 10 seconds in both directions. The calls start off fine and > slowly get worse. > > The only way I've been able to fix this is by disabling the rtp timer on > the sofia profile(rtp-timer-name=none). Making this change immediately > fixes the issue. > > No obvious system bottlenecks, 1/5/15 min load averages all less 1 on dual > core CPUs, tons of free memory, no I/O issues on the nic, etc. > > I suspect this might have something to do with some weird CPU sharing AWS > is doing on their hypervisor, but I'm not sure. > > I can't find a ton of info on exactly what the rtp timer does, other than > disabling it "disables asynchronous rtp" and "makes freeswitch handle media > the same way as asterisks". I'm not really sure what this means. I've read > it's less effecient and come across a couple very old post on the mailing > list of Anthony steering people away from disabling the rtp-timer. > > Has anyone had a similar problem? Or does anyone have some more > information/resources regarding exactly what the function of the rtp timer > is? I'm trying to get in front of this. What are the drawbacks of not using > an rtp timer? > > Thanks for reading! > _________________________________________________________________________ > > 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 bilaln018 at gmail.com Mon Jan 18 20:05:20 2021 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Tue, 19 Jan 2021 01:05:20 +0500 Subject: [Freeswitch-users] [FreeSWITCH sched_hangup does not execute 'execute_extension'] In-Reply-To: References: Message-ID: Hi Brian and Marcel, I tried both things but still my dialplan is not going to execute extension. Here is my code snippet. https://justpaste.it/7pwix On Mon, Jan 18, 2021 at 8:38 PM Marcel Haldemann wrote: > Since you hangup with an error-code (*ALLOTTED_TIMEOUT*) you maybe need > to set : > > > > > > > > Before the bridge. > > > > https://freeswitch.org/confluence/display/FREESWITCH/continue_on_fail > > > > > > *Von:* FreeSWITCH-users *Im > Auftrag von *Bilal Abbasi > *Gesendet:* Donnerstag, 14. Januar 2021 11:43 > *An:* FreeSWITCH Users Help > *Betreff:* [Freeswitch-users] [FreeSWITCH sched_hangup does not execute > 'execute_extension'] > > > > Hi Users, > > I want to build a scenario in which if the B-Party hangup the call, > A-Party can still be able to process through the dialplan. > > This works fine if B-Party hangup the call, but if the B-Party is hanup > forcefully via sched_hangup (timeout), call gets terminated i-e A-Party is > not able to reach 'execute_extension' application. > > > > Here is my dialplan snippet > > https://justpaste.it/2ppra > > > I have attached as an image as well. > > > > Regards > > Bilal Abbasi > _________________________________________________________________________ > > 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: -------------- next part -------------- A non-text attachment was scrubbed... Name: Screenshot 2021-01-19 at 1.01.43 AM.png Type: image/png Size: 395012 bytes Desc: not available URL: From gaurang.gohil at ecosmob.com Tue Jan 19 07:00:17 2021 From: gaurang.gohil at ecosmob.com (Gaurang Gohil) Date: Tue, 19 Jan 2021 12:30:17 +0530 Subject: [Freeswitch-users] Freeswitch crash reloading mod_callcenter In-Reply-To: References: Message-ID: hi Kunal, Did you get any core dump while crashing if so please put the gdb logs it can help to find the cause. On Mon, Jan 18, 2021 at 8:41 PM Kunal Mittal wrote: > Checking to see if anyone has any suggestions on this? > > > On Mon, Jan 11, 2021 at 12:18 PM Kunal Mittal wrote: > >> Hi, >> I have the latest version of the freeswitch 1.10.5 running with debian >> 10.7. It crashes everytime on reloading the mod_callcenter module after >> 3-4 minutes. Any suggestions? Is it a known bug? Any suggestions on how >> to troubleshoot this issue? >> >> Freeswitch version - 1.10.5 >> Debian - 10.7 >> >> Kunal >> > _________________________________________________________________________ > > 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 -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From jacobgreene1991 at gmail.com Tue Jan 19 01:23:10 2021 From: jacobgreene1991 at gmail.com (Jacob Greene) Date: Mon, 18 Jan 2021 19:23:10 -0600 Subject: [Freeswitch-users] AWS - Audio Delay - rtp timer In-Reply-To: References: Message-ID: Thanks for the info. That's really interesting. I have the jitter buffer completely disabled like this: "" Disabling the jitter buffer reduced the reports from our customers of audio delay. It seems counter intuitive that turning on an additional buffer would decrease delay. I wonder what is causing this? Any ideas? I'll setup some kind of canary deployment and see if I get similar results with the rtp-timer disabled and the jitter buffer on. Really appreciate your input! On Mon, Jan 18, 2021, 9:46 AM Jurijs Ivolga wrote: > Hi, > > I got similar problems. For me solution was to enable jitter buffer, > something like this: > > .... > > > > data="nolocal:rtp_jitter_buffer_during_bridge=true"/> > > .... > > In this case I didn't get delay when I set " value="soft"/>" in sip profile. > > Jurijs > > > On Mon, Jan 18, 2021 at 5:10 PM Jacob Greene > wrote: > >> Hello everyone, >> >> Recently, I am running into an issue with some severe audio delay with >> some freeswitch boxes running on AWS. It gets to the point where audio is >> delayed up to 10 seconds in both directions. The calls start off fine and >> slowly get worse. >> >> The only way I've been able to fix this is by disabling the rtp timer on >> the sofia profile(rtp-timer-name=none). Making this change immediately >> fixes the issue. >> >> No obvious system bottlenecks, 1/5/15 min load averages all less 1 on >> dual core CPUs, tons of free memory, no I/O issues on the nic, etc. >> >> I suspect this might have something to do with some weird CPU sharing AWS >> is doing on their hypervisor, but I'm not sure. >> >> I can't find a ton of info on exactly what the rtp timer does, other than >> disabling it "disables asynchronous rtp" and "makes freeswitch handle media >> the same way as asterisks". I'm not really sure what this means. I've read >> it's less effecient and come across a couple very old post on the mailing >> list of Anthony steering people away from disabling the rtp-timer. >> >> Has anyone had a similar problem? Or does anyone have some more >> information/resources regarding exactly what the function of the rtp timer >> is? I'm trying to get in front of this. What are the drawbacks of not using >> an rtp timer? >> >> Thanks for reading! >> _________________________________________________________________________ >> >> 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 ehtasham.malik at expertflow.com Tue Jan 19 14:44:55 2021 From: ehtasham.malik at expertflow.com (Malik Ehtasham) Date: Tue, 19 Jan 2021 07:44:55 -0700 (MST) Subject: [Freeswitch-users] AUDIO RTP REPORTS ERROR: [Remote Address Error!] Message-ID: <1611067495693-0.post@n2.nabble.com> hello everyone, I am using sip.js for webRTC with Freeswitch, I am able to initiate a call from sip.js to other sip extension but fail to answer the incoming call on sip.js, and the following error occurs in the fs_cli. 2021-01-18 21:31:15.701411 [DEBUG] switch_rtp.c:4450 Starting timer [soft] 960 bytes per 20ms 2021-01-18 21:31:20.701423 [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS ERROR: [Remote Address Error!] 2021-01-18 21:31:20.701423 [NOTICE] switch_core_media.c:9670 Hangup sofia/internal/5de16hg3 at ccujprafrvr6.invalid [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] 2021-01-18 21:31:20.701423 [ERR] sofia.c:8506 RTP Error! -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/ From mrjoli021 at gmail.com Tue Jan 19 17:39:25 2021 From: mrjoli021 at gmail.com (Joli Martinez) Date: Tue, 19 Jan 2021 12:39:25 -0500 Subject: [Freeswitch-users] register with FQDN or IP Message-ID: Hello, I am running a small freeswitch system with several phones, users register with the IP of the freeswitch box. Since they are now working from home more, they need to be able to connect remotely. I have built a kamailio proxy and exposed it. I created an FQDN sip.mydomain.com and pointed it to the kamailio box. The registration comes in and kamailio sends it to freeswitch. On the freeswitch box my internal sip profile looks like this. I have also reloaded it. Registrations from the public internet still fail, but users registering with the IP work. What am I missing? 2021-01-19 12:19:37.480591 [WARNING] sofia_reg.c:1794 SIP auth challenge (REGISTER) on sofia profile 'internal' for [222 at Sip.mydomain.com] from ip 1.1.1.1 2021-01-19 12:19:37.540512 [WARNING] sofia_reg.c:2930 Can't find user [ 222 at Sip.mydomain.com] from 1.1.1.1 You must define a domain called 'Sip.mydomain.com' in your directory and add a user with the id="222" attribute and you must configure your device to use the proper domain in its authentication credentials. 2021-01-19 12:19:37.540512 [WARNING] sofia_reg.c:1739 SIP auth failure (REGISTER) on sofia profile 'internal' for [222 at Sip.mydomain.com] from ip 1.1.1.1 Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Wed Jan 20 18:29:11 2021 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Wed, 20 Jan 2021 23:29:11 +0500 Subject: [Freeswitch-users] [FreeSWITCH sched_hangup does not execute 'execute_extension'] In-Reply-To: References: Message-ID: Hi Users, I found the solution via doing the following stuff, it might not be the most appropriate way of doing it, but it worked for me. Now i know exactly which leg i am hanging up, by generating the two uuid variables before starting the bridge. Thanks everyone Bilal Abbasi On Tue, Jan 19, 2021 at 1:05 AM Bilal Abbasi wrote: > Hi Brian and Marcel, > I tried both things but still my dialplan is not going to execute > extension. Here is my code snippet. > https://justpaste.it/7pwix > > On Mon, Jan 18, 2021 at 8:38 PM Marcel Haldemann > wrote: > >> Since you hangup with an error-code (*ALLOTTED_TIMEOUT*) you maybe need >> to set : >> >> >> >> >> >> >> >> Before the bridge. >> >> >> >> https://freeswitch.org/confluence/display/FREESWITCH/continue_on_fail >> >> >> >> >> >> *Von:* FreeSWITCH-users *Im >> Auftrag von *Bilal Abbasi >> *Gesendet:* Donnerstag, 14. Januar 2021 11:43 >> *An:* FreeSWITCH Users Help >> *Betreff:* [Freeswitch-users] [FreeSWITCH sched_hangup does not execute >> 'execute_extension'] >> >> >> >> Hi Users, >> >> I want to build a scenario in which if the B-Party hangup the call, >> A-Party can still be able to process through the dialplan. >> >> This works fine if B-Party hangup the call, but if the B-Party is hanup >> forcefully via sched_hangup (timeout), call gets terminated i-e A-Party is >> not able to reach 'execute_extension' application. >> >> >> >> Here is my dialplan snippet >> >> https://justpaste.it/2ppra >> >> >> I have attached as an image as well. >> >> >> >> Regards >> >> Bilal Abbasi >> _________________________________________________________________________ >> >> 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 rmundkowsky at ets.org Wed Jan 20 21:49:48 2021 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Wed, 20 Jan 2021 21:49:48 +0000 Subject: [Freeswitch-users] verto js files Message-ID: Hi, Does Freeswitch still have verto javascript libraries? I looked in github for /js/verto/jquery.FSRTC.js and did not find it. Robert ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From cmrienzo at gmail.com Wed Jan 20 22:10:50 2021 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Wed, 20 Jan 2021 17:10:50 -0500 Subject: [Freeswitch-users] verto js files In-Reply-To: References: Message-ID: Yes, look in https://github.com/freeswitch/verto-client Chris On Wed, Jan 20, 2021 at 5:09 PM Mundkowsky, Robert wrote: > Hi, > > > > Does Freeswitch still have verto javascript libraries? > > > > I looked in github for /js/verto/jquery.FSRTC.js and did not find it. > > > > Robert > > ------------------------------ > > This e-mail and any files transmitted with it may contain privileged or > confidential information. It is solely for use by the individual for whom > it is intended, even if addressed incorrectly. If you received this e-mail > in error, please notify the sender; do not disclose, copy, distribute, or > take any action in reliance on the contents of this information; and delete > it from your system. Any other use of this e-mail is prohibited. > > Thank you for your compliance. > ------------------------------ > _________________________________________________________________________ > > 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 alexanderhenryperkins at gmail.com Thu Jan 21 03:08:25 2021 From: alexanderhenryperkins at gmail.com (Alexander Perkins) Date: Wed, 20 Jan 2021 22:08:25 -0500 Subject: [Freeswitch-users] Limiting Calls Per Second Message-ID: Hi All. I have a client that needs a second Freeswitch server and I need to figure out how to handle her 'calls per second' limit. Currently, although not perfect, it is not a big deal - I simply decrement or increment a column in a database. However, I am concerned that with a second server, I may run into collisions, etc. What is the best practice to have two servers accurately measure calls per second? Is there a module I can use or something without re-inventing the wheel? Thanks! Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: From francesco at delagarda.com Thu Jan 21 09:01:14 2021 From: francesco at delagarda.com (Francesco Facco de Lagarda) Date: Thu, 21 Jan 2021 10:01:14 +0100 Subject: [Freeswitch-users] verto js files In-Reply-To: References: Message-ID: <003801d6efd3$f99ca960$ecd5fc20$@delagarda.com> Great minds think alike! I was looking for this too… I have followed installation instructions, but when I run grunt I get: Warning: Task "Gruntfile.js" not found. Use --force to continue. Aborted due to warnings. From: FreeSWITCH-users On Behalf Of Christopher Rienzo Sent: mercoledì 20 gennaio 2021 23:11 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] verto js files Yes, look in https://github.com/freeswitch/verto-client Chris On Wed, Jan 20, 2021 at 5:09 PM Mundkowsky, Robert > wrote: Hi, Does Freeswitch still have verto javascript libraries? I looked in github for /js/verto/jquery.FSRTC.js and did not find it. Robert _____ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. _____ _________________________________________________________________________ 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 Jan 21 09:15:37 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 21 Jan 2021 09:15:37 +0000 Subject: [Freeswitch-users] Limiting Calls Per Second In-Reply-To: References: Message-ID: https://freeswitch.org/confluence/plugins/servlet/mobile?contentId=3375201#content/view/3375201 is your friend On Thu, 21 Jan 2021 at 03:09, Alexander Perkins < alexanderhenryperkins at gmail.com> wrote: > Hi All. I have a client that needs a second Freeswitch server and I need > to figure out how to handle her 'calls per second' limit. Currently, > although not perfect, it is not a big deal - I simply decrement or > increment a column in a database. However, I am concerned that with a > second server, I may run into collisions, etc. > > What is the best practice to have two servers accurately measure calls per > second? Is there a module I can use or something without re-inventing the > wheel? > > Thanks! > Alex > _________________________________________________________________________ > > 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 abaci64 at gmail.com Thu Jan 21 17:16:00 2021 From: abaci64 at gmail.com (Abaci B) Date: Thu, 21 Jan 2021 12:16:00 -0500 Subject: [Freeswitch-users] Limiting Calls Per Second In-Reply-To: References: Message-ID: and keep in mind that not all backends support the interval, your only option is probably the redis backend On Thu, Jan 21, 2021 at 4:16 AM David Villasmil < david.villasmil.work at gmail.com> wrote: > > https://freeswitch.org/confluence/plugins/servlet/mobile?contentId=3375201#content/view/3375201 > > is your friend > > > On Thu, 21 Jan 2021 at 03:09, Alexander Perkins < > alexanderhenryperkins at gmail.com> wrote: > >> Hi All. I have a client that needs a second Freeswitch server and I need >> to figure out how to handle her 'calls per second' limit. Currently, >> although not perfect, it is not a big deal - I simply decrement or >> increment a column in a database. However, I am concerned that with a >> second server, I may run into collisions, etc. >> >> What is the best practice to have two servers accurately measure calls >> per second? Is there a module I can use or something without re-inventing >> the wheel? >> >> Thanks! >> Alex >> _________________________________________________________________________ >> >> 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 mrjoli021 at gmail.com Fri Jan 22 14:51:26 2021 From: mrjoli021 at gmail.com (Joli Martinez) Date: Fri, 22 Jan 2021 09:51:26 -0500 Subject: [Freeswitch-users] users registering from IP or FQDN Message-ID: Hello, I am running a small freeswitch system with several phones, users register with the IP of the freeswitch box. Since they are now working from home more, they need to be able to connect remotely. I have built a kamailio proxy and exposed it. I created an FQDN sip.mydomain.com and pointed it to the kamailio box. The registration comes in and kamailio sends it to freeswitch. I have added my doemain underneath the internal sip profile like this. I have also reloaded it. Registrations from the public internet still fail, but users registering with the IP work. What am I missing? 2021-01-19 12:19:37.480591 [WARNING] sofia_reg.c:1794 SIP auth challenge (REGISTER) on sofia profile 'internal' for [222 at Sip.mydomain.com] from ip 1.1.1.1 2021-01-19 12:19:37.540512 [WARNING] sofia_reg.c:2930 Can't find user [ 222 at Sip.mydomain.com] from 1.1.1.1 You must define a domain called 'Sip.mydomain.com' in your directory and add a user with the id="222" attribute and you must configure your device to use the proper domain in its authentication credentials. 2021-01-19 12:19:37.540512 [WARNING] sofia_reg.c:1739 SIP auth failure (REGISTER) on sofia profile 'internal' for [222 at Sip.mydomain.com] from ip 1.1.1.1 -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Fri Jan 22 15:05:40 2021 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 22 Jan 2021 15:05:40 +0000 Subject: [Freeswitch-users] users registering from IP or FQDN In-Reply-To: References: Message-ID: Does the user actually exists in an xml file with its domain name? On Fri, 22 Jan 2021 at 14:52, Joli Martinez wrote: > Hello, > > I am running a small freeswitch system with several phones, users register > with the IP of the freeswitch box. Since they are now working from home > more, they need to be able to connect remotely. I have built a kamailio > proxy and exposed it. I created an FQDN sip.mydomain.com and pointed it > to the kamailio box. The registration comes in and kamailio sends it to > freeswitch. > > I have added my doemain underneath the internal sip profile like this. I > have also reloaded it. > > > > Registrations from the public internet still fail, but users registering > with the IP work. What am I missing? > > 2021-01-19 12:19:37.480591 [WARNING] sofia_reg.c:1794 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [222 at Sip.mydomain.com] from ip > 1.1.1.1 > 2021-01-19 12:19:37.540512 [WARNING] sofia_reg.c:2930 Can't find user [ > 222 at Sip.mydomain.com] from 1.1.1.1 > You must define a domain called 'Sip.mydomain.com' in your directory and > add a user with the id="222" attribute > and you must configure your device to use the proper domain in its > authentication credentials. > 2021-01-19 12:19:37.540512 [WARNING] sofia_reg.c:1739 SIP auth failure > (REGISTER) on sofia profile 'internal' for [222 at Sip.mydomain.com] from ip > 1.1.1.1 > _________________________________________________________________________ > > 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 Fri Jan 22 18:32:26 2021 From: mrjoli021 at gmail.com (Joli Martinez) Date: Fri, 22 Jan 2021 13:32:26 -0500 Subject: [Freeswitch-users] users registering from IP or FQDN In-Reply-To: References: Message-ID: Hello NO. I am not trying to have two sets of users. All the users are already in the 1.1.1.1 domain and they are all working. I am trying to figure out how to get the external users to register the same way the internal users are doing without having to mess with the user configs. I was reading that if I put the domain as an alias it would work. thanks, On Fri, Jan 22, 2021 at 10:42 AM David Villasmil < david.villasmil.work at gmail.com> wrote: > Does the user actually exists in an xml file with its domain name? > > On Fri, 22 Jan 2021 at 14:52, Joli Martinez wrote: > >> Hello, >> >> I am running a small freeswitch system with several phones, users >> register with the IP of the freeswitch box. Since they are now working >> from home more, they need to be able to connect remotely. I have built a >> kamailio proxy and exposed it. I created an FQDN sip.mydomain.com and >> pointed it to the kamailio box. The registration comes in and kamailio >> sends it to freeswitch. >> >> I have added my doemain underneath the internal sip profile like this. I >> have also reloaded it. >> >> >> >> Registrations from the public internet still fail, but users registering >> with the IP work. What am I missing? >> >> 2021-01-19 12:19:37.480591 [WARNING] sofia_reg.c:1794 SIP auth challenge >> (REGISTER) on sofia profile 'internal' for [222 at Sip.mydomain.com] from >> ip 1.1.1.1 >> 2021-01-19 12:19:37.540512 [WARNING] sofia_reg.c:2930 Can't find user [ >> 222 at Sip.mydomain.com] from 1.1.1.1 >> You must define a domain called 'Sip.mydomain.com' in your directory and >> add a user with the id="222" attribute >> and you must configure your device to use the proper domain in its >> authentication credentials. >> 2021-01-19 12:19:37.540512 [WARNING] sofia_reg.c:1739 SIP auth failure >> (REGISTER) on sofia profile 'internal' for [222 at Sip.mydomain.com] from >> ip 1.1.1.1 >> _________________________________________________________________________ >> >> 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 gidoramothra at gmail.com Fri Jan 22 13:05:00 2021 From: gidoramothra at gmail.com (Stefan) Date: Fri, 22 Jan 2021 14:05:00 +0100 Subject: [Freeswitch-users] dtls-srtp problem with firefox 84.0.2 Message-ID: <20210122130500.GA1418@localhost.localdomain> Hi, I have a problem getting verto-communicator and/or mod_verto running with firefox. It works with chromium-based browsers without problems, but it looks to me (that means: I don't know exactly, but I suspect) as if the ciphersute to get the key for srtp is not perfect-forward-secure. Maybe that's the reason for firefox to fail, but I couldn't find any documented way to change the ciphersuite, neither in mod_verto, nor in mod_rtc (perhaps I oversee something). What I attach here are two logs. The first is the successful call to a conference witch edge (chromium based), the second is the try to do the same with firefox. Both logs begin at the same point, where verto does the state change CS_ROUTING -> CS_EXECUTE, the whole logs only differ after this point. Can anyone help me? In case it matters: the certs are from letsencrypt. __ s. -------------- next part -------------- 2021-01-22 09:33:32.288018 [DEBUG] switch_core_state_machine.c:287 (verto.rtc/31000) State Change CS_ROUTING -> CS_EXECUTE 2021-01-22 09:33:32.288018 [DEBUG] switch_core_state_machine.c:644 (verto.rtc/31000) State ROUTING going to sleep 2021-01-22 09:33:32.288018 [DEBUG] switch_core_state_machine.c:585 (verto.rtc/31000) Running State Change CS_EXECUTE (Cur 1 Tot 1) 2021-01-22 09:33:32.288018 [DEBUG] switch_core_state_machine.c:651 (verto.rtc/31000) State EXECUTE 2021-01-22 09:33:32.288018 [DEBUG] mod_rtc.c:120 verto.rtc/31000 RTC EXECUTE 2021-01-22 09:33:32.288018 [DEBUG] switch_core_state_machine.c:329 verto.rtc/31000 Standard EXECUTE EXECUTE [depth=0] verto.rtc/31000 set(rtp_secure_media=mandatory) 2021-01-22 09:33:32.288018 [CONSOLE] sofia_presence.c:1619 Event Thread Started 2021-01-22 09:33:32.288018 [DEBUG] mod_dptools.c:1672 SET verto.rtc/31000 [rtp_secure_media]=[mandatory] EXECUTE [depth=0] verto.rtc/31000 export(rtp_secure_media=mandatory) 2021-01-22 09:33:32.288018 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [rtp_secure_media]=[mandatory] EXECUTE [depth=0] verto.rtc/31000 set(codec_string=G722) 2021-01-22 09:33:32.288018 [DEBUG] mod_dptools.c:1672 SET verto.rtc/31000 [codec_string]=[G722] EXECUTE [depth=0] verto.rtc/31000 answer() 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [opus:111:48000:20:0:2]/[G722:9:8000:20:64000:1] 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [ISAC:103:16000:30:32000:1]/[G722:9:8000:20:64000:1] 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [ISAC:104:32000:30:32000:1]/[G722:9:8000:20:64000:1] 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:5649 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [CN:106:32000:20:0:1]/[G722:9:8000:20:64000:1] 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [CN:105:16000:20:0:1]/[G722:9:8000:20:64000:1] 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [CN:13:8000:20:0:1]/[G722:9:8000:20:64000:1] 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:5510 Set telephone-event payload to 110 at 48000 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:3839 Set Codec verto.rtc/31000 G722/8000 20 ms 160 samples 64000 bits 1 channels 2021-01-22 09:33:32.307889 [DEBUG] switch_core_codec.c:111 verto.rtc/31000 Original read codec set to G722:9 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:4284 Save audio Candidate cid: 1 proto: udp type: host addr: 192.168.1.112:58547 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:4284 Save audio Candidate cid: 1 proto: udp type: srflx addr: 93.104.12.213:58547 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:4329 Searching for rtp candidate. 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:4338 Choose rtp candidate, index 1, 93.104.12.213:58547 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:4104 verto.rtc/31000 choosing family v4 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:4349 Choose same candidate, index 0, for rtcp based on rtcp-mux attribute 93.104.12.213:58547 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:4401 setting remote audio ice addr to index 1 93.104.12.213:58547 based on candidate 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:4436 Setting remote rtcp audio addr to 93.104.12.213:58547 based on candidate 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:5853 Set telephone-event payload to 126 at 8000 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:5911 verto.rtc/31000 Set 2833 dtmf send payload to 126 recv payload to 126 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:6239 No matches with FTMP, fallback to ignoring FMTP 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:6247 No matches with inherit_codec, fallback to ignoring PT 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:8663 AUDIO RTP [verto.rtc/31000] 46.4.114.220 port 24164 -> 93.104.12.213 port 58547 codec: 9 ms: 20 2021-01-22 09:33:32.307889 [DEBUG] switch_rtp.c:4450 Starting timer [soft] 160 bytes per 20ms 2021-01-22 09:33:32.307889 [INFO] switch_core_media.c:8845 Activating Audio ICE 2021-01-22 09:33:32.307889 [NOTICE] switch_rtp.c:4952 Activating RTP audio ICE: Aw7j:eTMUc5TXONdQPFuX 93.104.12.213:58547 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:8885 Activating RTCP PORT 58547 2021-01-22 09:33:32.307889 [DEBUG] switch_rtp.c:4848 RTCP send rate is: 1000 and packet rate is: 20000 Remote Port: 58547 2021-01-22 09:33:32.307889 [INFO] switch_core_media.c:8896 Skipping RTCP ICE (Same as RTP) 2021-01-22 09:33:32.307889 [INFO] switch_rtp.c:3764 Activate RTP/RTCP audio DTLS client 2021-01-22 09:33:32.307889 [INFO] switch_rtp.c:3927 Changing audio DTLS state from OFF to HANDSHAKE 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:2554 Setting Jitterbuffer to 20ms (1 frames) (50 max frames) 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:8977 verto.rtc/31000 Set 2833 dtmf send payload to 126 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:8984 verto.rtc/31000 Set 2833 dtmf receive payload to 126 2021-01-22 09:33:32.307889 [DEBUG] switch_core_media.c:8645 Audio params are unchanged for verto.rtc/31000. 2021-01-22 09:33:32.307889 [DEBUG] mod_verto.c:2518 Local SDP verto.rtc/31000: v=0 o=FreeSWITCH 1611283848 1611283849 IN IP4 46.4.114.220 s=FreeSWITCH c=IN IP4 46.4.114.220 t=0 0 a=msid-semantic: WMS Ngpso7aRbsoTpMLXCeAoPst8US4qbbBO m=audio 24164 UDP/TLS/RTP/SAVPF 9 126 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=fingerprint:sha-256 5B:B0:37:08:CA:B3:02:48:CB:B6:1C:9C:A0:C9:66:1C:1D:A2:5F:25:95:59:25:6D:92:A5:80:6C:FB:79:EA:72 a=setup:active a=rtcp-mux a=rtcp:24164 IN IP4 46.4.114.220 a=ice-ufrag:eTMUc5TXONdQPFuX a=ice-pwd:mrk0poGFG5bB6uZbdCVgxF3c a=candidate:4690159198 1 udp 659136 46.4.114.220 24164 typ host generation 0 a=end-of-candidates a=ssrc:2953804084 cname:iR7wO7ePqszF4asx a=ssrc:2953804084 msid:Ngpso7aRbsoTpMLXCeAoPst8US4qbbBO a0 a=ssrc:2953804084 mslabel:Ngpso7aRbsoTpMLXCeAoPst8US4qbbBO a=ssrc:2953804084 label:Ngpso7aRbsoTpMLXCeAoPst8US4qbbBOa0 m=video 0 UDP/TLS/RTP/SAVPF 19 2021-01-22 09:33:32.307889 [NOTICE] mod_dptools.c:1406 Channel [verto.rtc/31000] has been answered 2021-01-22 09:33:32.307889 [DEBUG] switch_channel.c:3865 (verto.rtc/31000) Callstate Change RINGING -> ACTIVE EXECUTE [depth=0] verto.rtc/31000 conference(friends_16kHz at 16kHz-novideo+2357+flags{moderator|mute-detect}) 2021-01-22 09:33:32.307889 [DEBUG] mod_conference.c:3414 using channel sound prefix: /usr/share/freeswitch/sounds/en/us/callie 2021-01-22 09:33:32.307889 [DEBUG] mod_conference.c:228 Setup timer success interval: 20 samples: 320 2021-01-22 09:33:32.348029 [DEBUG] mod_verto.c:607 WRITE 93.104.12.213:50500 [{ "jsonrpc": "2.0", "id": 2, "method": "verto.answer", "params": { "callID": "87c1011e-026b-df48-300f-fa07399f1f65", "sdp": "v=0\r\no=FreeSWITCH 1611283848 1611283849 IN IP4 46.4.114.220\r\ns=FreeSWITCH\r\nc=IN IP4 46.4.114.220\r\nt=0 0\r\na=msid-semantic: WMS Ngpso7aRbsoTpMLXCeAoPst8US4qbbBO\r\nm=audio 24164 UDP/TLS/RTP/SAVPF 9 126\r\na=rtpmap:9 G722/8000\r\na=rtpmap:126 telephone-event/8000\r\na=silenceSupp:off - - - -\r\na=ptime:20\r\na=sendrecv\r\na=fingerprint:sha-256 5B:B0:37:08:CA:B3:02:48:CB:B6:1C:9C:A0:C9:66:1C:1D:A2:5F:25:95:59:25:6D:92:A5:80:6C:FB:79:EA:72\r\na=setup:active\r\na=rtcp-mux\r\na=rtcp:24164 IN IP4 46.4.114.220\r\na=ice-ufrag:eTMUc5TXONdQPFuX\r\na=ice-pwd:mrk0poGFG5bB6uZbdCVgxF3c\r\na=candidate:4690159198 1 udp 659136 46.4.114.220 24164 typ host generation 0\r\na=end-of-candidates\r\na=ssrc:2953804084 cname:iR7wO7ePqszF4asx\r\na=ssrc:2953804084 msid:Ngpso7aRbsoTpMLXCeAoPst8US4qbbBO a0\r\na=ssrc:2953804084 mslabel:Ngpso7aRbsoTpMLXCeAoPst8US4qbbBO\r\na=ssrc:2953804084 label:Ngpso7aRbsoTpMLXCeAoPst8US4qbbBOa0\r\nm=video 0 UDP/TLS/RTP/SAVPF 19\r\n" } }] -------------- next part -------------- 2021-01-21 15:36:48.773228 [DEBUG] switch_core_state_machine.c:287 (verto.rtc/31000) State Change CS_ROUTING -> CS_EXECUTE 2021-01-21 15:36:48.773228 [DEBUG] switch_core_state_machine.c:644 (verto.rtc/31000) State ROUTING going to sleep 2021-01-21 15:36:48.773228 [DEBUG] switch_core_state_machine.c:585 (verto.rtc/31000) Running State Change CS_EXECUTE (Cur 1 Tot 1) 2021-01-21 15:36:48.773228 [DEBUG] switch_core_state_machine.c:651 (verto.rtc/31000) State EXECUTE 2021-01-21 15:36:48.773228 [DEBUG] mod_rtc.c:120 verto.rtc/31000 RTC EXECUTE 2021-01-21 15:36:48.773228 [DEBUG] switch_core_state_machine.c:329 verto.rtc/31000 Standard EXECUTE EXECUTE [depth=0] verto.rtc/31000 set(rtp_secure_media=mandatory) 2021-01-21 15:36:48.773228 [DEBUG] mod_dptools.c:1672 SET verto.rtc/31000 [rtp_secure_media]=[mandatory] 2021-01-21 15:36:48.773228 [CONSOLE] sofia_presence.c:1619 Event Thread Started EXECUTE [depth=0] verto.rtc/31000 export(rtp_secure_media=mandatory) 2021-01-21 15:36:48.773228 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [rtp_secure_media]=[mandatory] EXECUTE [depth=0] verto.rtc/31000 set(codec_string=G722) 2021-01-21 15:36:48.773228 [DEBUG] mod_dptools.c:1672 SET verto.rtc/31000 [codec_string]=[G722] EXECUTE [depth=0] verto.rtc/31000 answer() 2021-01-21 15:36:48.773228 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [opus:109:48000:20:0:2]/[G722:9:8000:20:64000:1] 2021-01-21 15:36:48.773228 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2021-01-21 15:36:48.773228 [DEBUG] switch_core_media.c:5649 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match 2021-01-21 15:36:48.773228 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2021-01-21 15:36:48.773228 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2021-01-21 15:36:48.773228 [DEBUG] switch_core_media.c:5510 Set telephone-event payload to 101 at 8000 2021-01-21 15:36:48.773228 [WARNING] switch_core_media.c:5667 Crypto not negotiated but required. 2021-01-21 15:36:48.773228 [DEBUG] switch_core_media.c:5911 verto.rtc/31000 Set 2833 dtmf send payload to 101 recv payload to 101 2021-01-21 15:36:48.773228 [DEBUG] switch_core_media.c:6239 No matches with FTMP, fallback to ignoring FMTP 2021-01-21 15:36:48.773228 [DEBUG] switch_core_media.c:6247 No matches with inherit_codec, fallback to ignoring PT 2021-01-21 15:36:48.773228 [WARNING] switch_core_media.c:6253 Crypto not negotiated but required. 2021-01-21 15:36:48.773228 [NOTICE] switch_channel.c:3908 Hangup verto.rtc/31000 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2021-01-21 15:36:48.773228 [DEBUG] switch_core_session.c:2905 verto.rtc/31000 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2021-01-21 15:36:48.773228 [DEBUG] switch_core_state_machine.c:651 (verto.rtc/31000) State EXECUTE going to sleep 2021-01-21 15:36:48.773228 [DEBUG] switch_core_state_machine.c:585 (verto.rtc/31000) Running State Change CS_HANGUP (Cur 1 Tot 1) 2021-01-21 15:36:48.773228 [DEBUG] switch_core_state_machine.c:848 (verto.rtc/31000) Callstate Change RINGING -> HANGUP 2021-01-21 15:36:48.773228 [DEBUG] switch_core_state_machine.c:850 (verto.rtc/31000) State HANGUP 2021-01-21 15:36:48.793250 [DEBUG] switch_core_state_machine.c:60 verto.rtc/31000 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2021-01-21 15:36:48.793250 [DEBUG] switch_core_state_machine.c:850 (verto.rtc/31000) State HANGUP going to sleep 2021-01-21 15:36:48.793250 [DEBUG] switch_core_state_machine.c:620 (verto.rtc/31000) State Change CS_HANGUP -> CS_REPORTING 2021-01-21 15:36:48.793250 [DEBUG] switch_core_state_machine.c:585 (verto.rtc/31000) Running State Change CS_REPORTING (Cur 1 Tot 1) 2021-01-21 15:36:48.793250 [DEBUG] switch_core_state_machine.c:936 (verto.rtc/31000) State REPORTING 2021-01-21 15:36:48.793250 [DEBUG] switch_core_state_machine.c:174 verto.rtc/31000 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2021-01-21 15:36:48.793250 [DEBUG] switch_core_state_machine.c:936 (verto.rtc/31000) State REPORTING going to sleep 2021-01-21 15:36:48.793250 [DEBUG] switch_core_state_machine.c:611 (verto.rtc/31000) State Change CS_REPORTING -> CS_DESTROY 2021-01-21 15:36:48.793250 [DEBUG] switch_core_session.c:1726 Session 1 (verto.rtc/31000) Locked, Waiting on external entities 2021-01-21 15:36:48.793250 [NOTICE] switch_core_session.c:1744 Session 1 (verto.rtc/31000) Ended 2021-01-21 15:36:48.793250 [NOTICE] switch_core_session.c:1748 Close Channel verto.rtc/31000 [CS_DESTROY] 2021-01-21 15:36:48.793250 [DEBUG] switch_core_state_machine.c:739 (verto.rtc/31000) Running State Change CS_DESTROY (Cur 0 Tot 1) 2021-01-21 15:36:48.793250 [DEBUG] switch_core_state_machine.c:749 (verto.rtc/31000) State DESTROY 2021-01-21 15:36:48.793250 [DEBUG] mod_rtc.c:132 verto.rtc/31000 RTC DESTROY 2021-01-21 15:36:48.793250 [DEBUG] switch_core_state_machine.c:181 verto.rtc/31000 Standard DESTROY 2021-01-21 15:36:48.793250 [DEBUG] switch_core_state_machine.c:749 (verto.rtc/31000) State DESTROY going to sleep 2021-01-21 15:36:48.833246 [DEBUG] mod_verto.c:607 WRITE 93.104.1.138:42084 [{ "jsonrpc": "2.0", "id": 2, "method": "verto.bye", "params": { "callID": "360828a2-046a-0c70-8e20-8e23a5418cdf", "causeCode": 88, "cause": "INCOMPATIBLE_DESTINATION" } From marcb at voicemeup.com Mon Jan 25 17:42:14 2021 From: marcb at voicemeup.com (Marc Bernard) Date: Mon, 25 Jan 2021 12:42:14 -0500 Subject: [Freeswitch-users] Scanners and botnet vulnerability Message-ID: <113d01d6f341$6c93d5e0$45bb81a0$@voicemeup.com> Hello All, Is anyone else noticing that there is more and more scanners attempting brute force with no reply to auth request resulting in logging a lot of abandoned calls ? Scenario: - A scanner send an INVITE|REGISTER with no credentials - Freeswitch responds with authentication request and a challenge is send to logs; " 2021-01-25 12:27:39.306075 [WARNING] sofia_reg.c:1792 SIP auth challenge (REGISTER) on sofia profile 'public' for [1730 at 1.2.3.4] from ip 5.6.7.8" - Scanner does not respond - After a while, Freeswitch logs the following: 2ae23e93-c929-4089-a594-8e7af633ca88 2021-01-25 12:28:37.506078 [WARNING] switch_core_state_machine.c:687 2ae23e93-c929-4089-a594-8e7af633ca88 sofia/public/1730 at 1.2.3.4 Abandoned -- In our case, we made fail2ban more sensitive to auth failures logs which does not get triggered because of the scanner not even trying to send credentials. Wouldn't it make more sense for this log to include the IP of sip client that abandoned the call (5.6.7.8) instead of only the IP of the sip profile (1.2.3.4) ? This would allow us to have Fail2ban block this scenario more aggressively. Thoughts ? From krice at freeswitch.org Mon Jan 25 21:01:22 2021 From: krice at freeswitch.org (Ken Rice) Date: Mon, 25 Jan 2021 15:01:22 -0600 Subject: [Freeswitch-users] Scanners and botnet vulnerability In-Reply-To: <113d01d6f341$6c93d5e0$45bb81a0$@voicemeup.com> References: <113d01d6f341$6c93d5e0$45bb81a0$@voicemeup.com> Message-ID: <4C9CA5FD-573F-46E8-B7C6-77C08F47A815@freeswitch.org> this is super common. this is more likely a recon attack than an actual brute force attempt. Eother that they are looking for something with auth turned off. we see tons of these things regularly. Fail to ban helps some but using a SIP RBL and dropping traffic via prefixes associated with regions and bad actor hosts seems to be the best course of action these days. I wont name the company, but a mjor european hosting company i drop their entire AS as its not worth the hassle. Sent from my iPhone > On Jan 25, 2021, at 14:49, Marc Bernard wrote: > > Hello All, > > Is anyone else noticing that there is more and more scanners attempting > brute force with no reply to auth request resulting in logging a lot of > abandoned calls ? > > Scenario: > > - A scanner send an INVITE|REGISTER with no credentials > - Freeswitch responds with authentication request and a challenge is send to > logs; > " > 2021-01-25 12:27:39.306075 [WARNING] sofia_reg.c:1792 SIP auth challenge > (REGISTER) on sofia profile 'public' for [1730 at 1.2.3.4] from ip 5.6.7.8" > - Scanner does not respond > - After a while, Freeswitch logs the following: > 2ae23e93-c929-4089-a594-8e7af633ca88 2021-01-25 12:28:37.506078 [WARNING] > switch_core_state_machine.c:687 2ae23e93-c929-4089-a594-8e7af633ca88 > sofia/public/1730 at 1.2.3.4 Abandoned > > -- > > In our case, we made fail2ban more sensitive to auth failures logs which > does not get triggered because of the scanner not even trying to send > credentials. > > Wouldn't it make more sense for this log to include the IP of sip client > that abandoned the call (5.6.7.8) instead of only the IP of the sip profile > (1.2.3.4) ? > > This would allow us to have Fail2ban block this scenario more aggressively. > > Thoughts ? > > > > > _________________________________________________________________________ > > 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 rbetancor at gmail.com Mon Jan 25 22:24:01 2021 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Mon, 25 Jan 2021 22:24:01 +0000 Subject: [Freeswitch-users] Scanners and botnet vulnerability In-Reply-To: <4C9CA5FD-573F-46E8-B7C6-77C08F47A815@freeswitch.org> References: <113d01d6f341$6c93d5e0$45bb81a0$@voicemeup.com> <4C9CA5FD-573F-46E8-B7C6-77C08F47A815@freeswitch.org> Message-ID: You could tell the name, SAS on France and OVH, they are both nest of bots. On Mon, Jan 25, 2021 at 9:31 PM Ken Rice wrote: > this is super common. this is more likely a recon attack than an actual > brute force attempt. Eother that they are looking for something with auth > turned off. we see tons of these things regularly. Fail to ban helps some > but using a SIP RBL and dropping traffic via prefixes associated with > regions and bad actor hosts seems to be the best course of action these > days. > > I wont name the company, but a mjor european hosting company i drop their > entire AS as its not worth the hassle. > > Sent from my iPhone > > > On Jan 25, 2021, at 14:49, Marc Bernard wrote: > > > > Hello All, > > > > Is anyone else noticing that there is more and more scanners attempting > > brute force with no reply to auth request resulting in logging a lot of > > abandoned calls ? > > > > Scenario: > > > > - A scanner send an INVITE|REGISTER with no credentials > > - Freeswitch responds with authentication request and a challenge is > send to > > logs; > > " > > 2021-01-25 12:27:39.306075 [WARNING] sofia_reg.c:1792 SIP auth challenge > > (REGISTER) on sofia profile 'public' for [1730 at 1.2.3.4] from ip 5.6.7.8" > > - Scanner does not respond > > - After a while, Freeswitch logs the following: > > 2ae23e93-c929-4089-a594-8e7af633ca88 2021-01-25 12:28:37.506078 [WARNING] > > switch_core_state_machine.c:687 2ae23e93-c929-4089-a594-8e7af633ca88 > > sofia/public/1730 at 1.2.3.4 Abandoned > > > > -- > > > > In our case, we made fail2ban more sensitive to auth failures logs which > > does not get triggered because of the scanner not even trying to send > > credentials. > > > > Wouldn't it make more sense for this log to include the IP of sip client > > that abandoned the call (5.6.7.8) instead of only the IP of the sip > profile > > (1.2.3.4) ? > > > > This would allow us to have Fail2ban block this scenario more > aggressively. > > > > Thoughts ? > > > > > > > > > > _________________________________________________________________________ > > > > 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 joe at expert.net Mon Jan 25 22:53:05 2021 From: joe at expert.net (Joseph Barrero) Date: Mon, 25 Jan 2021 16:53:05 -0600 Subject: [Freeswitch-users] Rings Versus Seconds In-Reply-To: References: Message-ID: The issue is that the length of a ring can be different as heard by the caller depending on the country/location of the recipient. Also, the ringing on the recipients' device/phone can also be variable based on their ring-tone preferences and the make and model of the device/phone. In other words, 3 rings can consist of 2 different lengths as heard by the caller and the callee. As such, duration in seconds is really the only consistent way I know how to measure the timing of a ring. On Tue, Jan 5, 2021 at 3:27 PM KitchM via FreeSWITCH-users < freeswitch-users at lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: KitchM > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Thu, 24 Dec 2020 10:15:31 -0700 (MST) > Subject: Rings Versus Seconds > 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/ > > > > > ---------- Forwarded message ---------- > From: KitchM via FreeSWITCH-users > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Tue, 05 Jan 2021 13:27:42 -0800 (PST) > Subject: [Freeswitch-users] Rings Versus Seconds > _________________________________________________________________________ > > 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 krice at freeswitch.org Tue Jan 26 03:04:55 2021 From: krice at freeswitch.org (Ken Rice) Date: Mon, 25 Jan 2021 21:04:55 -0600 Subject: [Freeswitch-users] Scanners and botnet vulnerability In-Reply-To: References: Message-ID: exactly those 2 lol Sent from my iPhone > On Jan 25, 2021, at 16:24, Raúl Alexis Betancor Santana wrote: > >  > You could tell the name, SAS on France and OVH, they are both nest of bots. > >> On Mon, Jan 25, 2021 at 9:31 PM Ken Rice wrote: >> this is super common. this is more likely a recon attack than an actual brute force attempt. Eother that they are looking for something with auth turned off. we see tons of these things regularly. Fail to ban helps some but using a SIP RBL and dropping traffic via prefixes associated with regions and bad actor hosts seems to be the best course of action these days. >> >> I wont name the company, but a mjor european hosting company i drop their entire AS as its not worth the hassle. >> >> Sent from my iPhone >> >> > On Jan 25, 2021, at 14:49, Marc Bernard wrote: >> > >> > Hello All, >> > >> > Is anyone else noticing that there is more and more scanners attempting >> > brute force with no reply to auth request resulting in logging a lot of >> > abandoned calls ? >> > >> > Scenario: >> > >> > - A scanner send an INVITE|REGISTER with no credentials >> > - Freeswitch responds with authentication request and a challenge is send to >> > logs; >> > " >> > 2021-01-25 12:27:39.306075 [WARNING] sofia_reg.c:1792 SIP auth challenge >> > (REGISTER) on sofia profile 'public' for [1730 at 1.2.3.4] from ip 5.6.7.8" >> > - Scanner does not respond >> > - After a while, Freeswitch logs the following: >> > 2ae23e93-c929-4089-a594-8e7af633ca88 2021-01-25 12:28:37.506078 [WARNING] >> > switch_core_state_machine.c:687 2ae23e93-c929-4089-a594-8e7af633ca88 >> > sofia/public/1730 at 1.2.3.4 Abandoned >> > >> > -- >> > >> > In our case, we made fail2ban more sensitive to auth failures logs which >> > does not get triggered because of the scanner not even trying to send >> > credentials. >> > >> > Wouldn't it make more sense for this log to include the IP of sip client >> > that abandoned the call (5.6.7.8) instead of only the IP of the sip profile >> > (1.2.3.4) ? >> > >> > This would allow us to have Fail2ban block this scenario more aggressively. >> > >> > Thoughts ? >> > >> > >> > >> > >> > _________________________________________________________________________ >> > >> > 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 gregor at infomedia.si Tue Jan 26 04:05:05 2021 From: gregor at infomedia.si (Gregor Nanger) Date: Tue, 26 Jan 2021 05:05:05 +0100 Subject: [Freeswitch-users] Scanners and botnet vulnerability In-Reply-To: References: Message-ID: 😁 On Tue, Jan 26, 2021, 04:05 Ken Rice wrote: > exactly those 2 lol > > Sent from my iPhone > > On Jan 25, 2021, at 16:24, Raúl Alexis Betancor Santana < > rbetancor at gmail.com> wrote: > >  > You could tell the name, SAS on France and OVH, they are both nest of bots. > > On Mon, Jan 25, 2021 at 9:31 PM Ken Rice wrote: > >> this is super common. this is more likely a recon attack than an actual >> brute force attempt. Eother that they are looking for something with auth >> turned off. we see tons of these things regularly. Fail to ban helps some >> but using a SIP RBL and dropping traffic via prefixes associated with >> regions and bad actor hosts seems to be the best course of action these >> days. >> >> I wont name the company, but a mjor european hosting company i drop their >> entire AS as its not worth the hassle. >> >> Sent from my iPhone >> >> > On Jan 25, 2021, at 14:49, Marc Bernard wrote: >> > >> > Hello All, >> > >> > Is anyone else noticing that there is more and more scanners attempting >> > brute force with no reply to auth request resulting in logging a lot of >> > abandoned calls ? >> > >> > Scenario: >> > >> > - A scanner send an INVITE|REGISTER with no credentials >> > - Freeswitch responds with authentication request and a challenge is >> send to >> > logs; >> > " >> > 2021-01-25 12:27:39.306075 [WARNING] sofia_reg.c:1792 SIP auth challenge >> > (REGISTER) on sofia profile 'public' for [1730 at 1.2.3.4] from ip >> 5.6.7.8" >> > - Scanner does not respond >> > - After a while, Freeswitch logs the following: >> > 2ae23e93-c929-4089-a594-8e7af633ca88 2021-01-25 12:28:37.506078 >> [WARNING] >> > switch_core_state_machine.c:687 2ae23e93-c929-4089-a594-8e7af633ca88 >> > sofia/public/1730 at 1.2.3.4 Abandoned >> > >> > -- >> > >> > In our case, we made fail2ban more sensitive to auth failures logs which >> > does not get triggered because of the scanner not even trying to send >> > credentials. >> > >> > Wouldn't it make more sense for this log to include the IP of sip client >> > that abandoned the call (5.6.7.8) instead of only the IP of the sip >> profile >> > (1.2.3.4) ? >> > >> > This would allow us to have Fail2ban block this scenario more >> aggressively. >> > >> > Thoughts ? >> > >> > >> > >> > >> > >> _________________________________________________________________________ >> > >> > 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 lloyd.aloysius at gmail.com Tue Jan 26 04:20:17 2021 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Mon, 25 Jan 2021 23:20:17 -0500 Subject: [Freeswitch-users] Scanners and botnet vulnerability In-Reply-To: References: Message-ID: Ken, thank you for the information. Can you please let me know how to block AS numbers from IPTables? On Mon, Jan 25, 2021 at 10:06 PM Ken Rice wrote: > exactly those 2 lol > > Sent from my iPhone > > On Jan 25, 2021, at 16:24, Raúl Alexis Betancor Santana < > rbetancor at gmail.com> wrote: > >  > You could tell the name, SAS on France and OVH, they are both nest of bots. > > On Mon, Jan 25, 2021 at 9:31 PM Ken Rice wrote: > >> this is super common. this is more likely a recon attack than an actual >> brute force attempt. Eother that they are looking for something with auth >> turned off. we see tons of these things regularly. Fail to ban helps some >> but using a SIP RBL and dropping traffic via prefixes associated with >> regions and bad actor hosts seems to be the best course of action these >> days. >> >> I wont name the company, but a mjor european hosting company i drop their >> entire AS as its not worth the hassle. >> >> Sent from my iPhone >> >> > On Jan 25, 2021, at 14:49, Marc Bernard wrote: >> > >> > Hello All, >> > >> > Is anyone else noticing that there is more and more scanners attempting >> > brute force with no reply to auth request resulting in logging a lot of >> > abandoned calls ? >> > >> > Scenario: >> > >> > - A scanner send an INVITE|REGISTER with no credentials >> > - Freeswitch responds with authentication request and a challenge is >> send to >> > logs; >> > " >> > 2021-01-25 12:27:39.306075 [WARNING] sofia_reg.c:1792 SIP auth challenge >> > (REGISTER) on sofia profile 'public' for [1730 at 1.2.3.4] from ip >> 5.6.7.8" >> > - Scanner does not respond >> > - After a while, Freeswitch logs the following: >> > 2ae23e93-c929-4089-a594-8e7af633ca88 2021-01-25 12:28:37.506078 >> [WARNING] >> > switch_core_state_machine.c:687 2ae23e93-c929-4089-a594-8e7af633ca88 >> > sofia/public/1730 at 1.2.3.4 Abandoned >> > >> > -- >> > >> > In our case, we made fail2ban more sensitive to auth failures logs which >> > does not get triggered because of the scanner not even trying to send >> > credentials. >> > >> > Wouldn't it make more sense for this log to include the IP of sip client >> > that abandoned the call (5.6.7.8) instead of only the IP of the sip >> profile >> > (1.2.3.4) ? >> > >> > This would allow us to have Fail2ban block this scenario more >> aggressively. >> > >> > Thoughts ? >> > >> > >> > >> > >> > >> _________________________________________________________________________ >> > >> > 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 rbetancor at gmail.com Tue Jan 26 07:48:17 2021 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Tue, 26 Jan 2021 07:48:17 +0000 Subject: [Freeswitch-users] Scanners and botnet vulnerability In-Reply-To: References: Message-ID: You could not block an AS from iptables, you should get the IP ranges that belongs to that AS and block them. There are scripts/extensions for shorewall (linux firewalling suite), that allow you to do geoip/AS based rules. On Tue, Jan 26, 2021 at 4:47 AM Lloyd Aloysius wrote: > Ken, thank you for the information. Can you please let me know how to > block AS numbers from IPTables? > > > On Mon, Jan 25, 2021 at 10:06 PM Ken Rice wrote: > >> exactly those 2 lol >> >> Sent from my iPhone >> >> On Jan 25, 2021, at 16:24, Raúl Alexis Betancor Santana < >> rbetancor at gmail.com> wrote: >> >>  >> You could tell the name, SAS on France and OVH, they are both nest of >> bots. >> >> On Mon, Jan 25, 2021 at 9:31 PM Ken Rice wrote: >> >>> this is super common. this is more likely a recon attack than an actual >>> brute force attempt. Eother that they are looking for something with auth >>> turned off. we see tons of these things regularly. Fail to ban helps some >>> but using a SIP RBL and dropping traffic via prefixes associated with >>> regions and bad actor hosts seems to be the best course of action these >>> days. >>> >>> I wont name the company, but a mjor european hosting company i drop >>> their entire AS as its not worth the hassle. >>> >>> Sent from my iPhone >>> >>> > On Jan 25, 2021, at 14:49, Marc Bernard wrote: >>> > >>> > Hello All, >>> > >>> > Is anyone else noticing that there is more and more scanners attempting >>> > brute force with no reply to auth request resulting in logging a lot of >>> > abandoned calls ? >>> > >>> > Scenario: >>> > >>> > - A scanner send an INVITE|REGISTER with no credentials >>> > - Freeswitch responds with authentication request and a challenge is >>> send to >>> > logs; >>> > " >>> > 2021-01-25 12:27:39.306075 [WARNING] sofia_reg.c:1792 SIP auth >>> challenge >>> > (REGISTER) on sofia profile 'public' for [1730 at 1.2.3.4] from ip >>> 5.6.7.8" >>> > - Scanner does not respond >>> > - After a while, Freeswitch logs the following: >>> > 2ae23e93-c929-4089-a594-8e7af633ca88 2021-01-25 12:28:37.506078 >>> [WARNING] >>> > switch_core_state_machine.c:687 2ae23e93-c929-4089-a594-8e7af633ca88 >>> > sofia/public/1730 at 1.2.3.4 Abandoned >>> > >>> > -- >>> > >>> > In our case, we made fail2ban more sensitive to auth failures logs >>> which >>> > does not get triggered because of the scanner not even trying to send >>> > credentials. >>> > >>> > Wouldn't it make more sense for this log to include the IP of sip >>> client >>> > that abandoned the call (5.6.7.8) instead of only the IP of the sip >>> profile >>> > (1.2.3.4) ? >>> > >>> > This would allow us to have Fail2ban block this scenario more >>> aggressively. >>> > >>> > Thoughts ? >>> > >>> > >>> > >>> > >>> > >>> _________________________________________________________________________ >>> > >>> > 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 > > _________________________________________________________________________ > > 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 rbetancor at gmail.com Tue Jan 26 07:51:07 2021 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Tue, 26 Jan 2021 07:51:07 +0000 Subject: [Freeswitch-users] Scanners and botnet vulnerability In-Reply-To: References: Message-ID: And the worst thing, is that they fully ignore all the abuse claims, we also ended blacklisting their full ASs and when some of their customers or ours claims not able to access some service/company that are under their umbrella or ours, we just raise up the flag of "they are just a nest of bots and crackers, we do no talk to them". On Tue, Jan 26, 2021 at 3:15 AM Ken Rice wrote: > exactly those 2 lol > > Sent from my iPhone > > On Jan 25, 2021, at 16:24, Raúl Alexis Betancor Santana < > rbetancor at gmail.com> wrote: > >  > You could tell the name, SAS on France and OVH, they are both nest of bots. > > On Mon, Jan 25, 2021 at 9:31 PM Ken Rice wrote: > >> this is super common. this is more likely a recon attack than an actual >> brute force attempt. Eother that they are looking for something with auth >> turned off. we see tons of these things regularly. Fail to ban helps some >> but using a SIP RBL and dropping traffic via prefixes associated with >> regions and bad actor hosts seems to be the best course of action these >> days. >> >> I wont name the company, but a mjor european hosting company i drop their >> entire AS as its not worth the hassle. >> >> Sent from my iPhone >> >> > On Jan 25, 2021, at 14:49, Marc Bernard wrote: >> > >> > Hello All, >> > >> > Is anyone else noticing that there is more and more scanners attempting >> > brute force with no reply to auth request resulting in logging a lot of >> > abandoned calls ? >> > >> > Scenario: >> > >> > - A scanner send an INVITE|REGISTER with no credentials >> > - Freeswitch responds with authentication request and a challenge is >> send to >> > logs; >> > " >> > 2021-01-25 12:27:39.306075 [WARNING] sofia_reg.c:1792 SIP auth challenge >> > (REGISTER) on sofia profile 'public' for [1730 at 1.2.3.4] from ip >> 5.6.7.8" >> > - Scanner does not respond >> > - After a while, Freeswitch logs the following: >> > 2ae23e93-c929-4089-a594-8e7af633ca88 2021-01-25 12:28:37.506078 >> [WARNING] >> > switch_core_state_machine.c:687 2ae23e93-c929-4089-a594-8e7af633ca88 >> > sofia/public/1730 at 1.2.3.4 Abandoned >> > >> > -- >> > >> > In our case, we made fail2ban more sensitive to auth failures logs which >> > does not get triggered because of the scanner not even trying to send >> > credentials. >> > >> > Wouldn't it make more sense for this log to include the IP of sip client >> > that abandoned the call (5.6.7.8) instead of only the IP of the sip >> profile >> > (1.2.3.4) ? >> > >> > This would allow us to have Fail2ban block this scenario more >> aggressively. >> > >> > Thoughts ? >> > >> > >> > >> > >> > >> _________________________________________________________________________ >> > >> > 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 krice at freeswitch.org Tue Jan 26 17:48:51 2021 From: krice at freeswitch.org (Ken Rice) Date: Tue, 26 Jan 2021 11:48:51 -0600 Subject: [Freeswitch-users] Scanners and botnet vulnerability In-Reply-To: References: Message-ID: <106B4B79-BF3E-43E8-B54A-F289B12A581E@freeswitch.org> There are various tools like https://www.countryipblocks.net/acl.php (and more) that will create the datasets you need to feed iptables. There is also things like https://www.apiban.org/ that you might want to look at or proactively blocking bad actors. APIBAN is like a good old RBL you’d use to combat spam but collects data on SIP bad actors. K From: FreeSWITCH-users on behalf of Lloyd Aloysius Reply-To: FreeSWITCH Users Help Date: Monday, January 25, 2021 at 10:21 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Scanners and botnet vulnerability Ken, thank you for the information. Can you please let me know how to block AS numbers from IPTables? On Mon, Jan 25, 2021 at 10:06 PM Ken Rice wrote: exactly those 2 lol Sent from my iPhone On Jan 25, 2021, at 16:24, Raúl Alexis Betancor Santana wrote:  You could tell the name, SAS on France and OVH, they are both nest of bots. On Mon, Jan 25, 2021 at 9:31 PM Ken Rice wrote: this is super common. this is more likely a recon attack than an actual brute force attempt. Eother that they are looking for something with auth turned off. we see tons of these things regularly. Fail to ban helps some but using a SIP RBL and dropping traffic via prefixes associated with regions and bad actor hosts seems to be the best course of action these days. I wont name the company, but a mjor european hosting company i drop their entire AS as its not worth the hassle. Sent from my iPhone > On Jan 25, 2021, at 14:49, Marc Bernard wrote: > > Hello All, > > Is anyone else noticing that there is more and more scanners attempting > brute force with no reply to auth request resulting in logging a lot of > abandoned calls ? > > Scenario: > > - A scanner send an INVITE|REGISTER with no credentials > - Freeswitch responds with authentication request and a challenge is send to > logs; > " > 2021-01-25 12:27:39.306075 [WARNING] sofia_reg.c:1792 SIP auth challenge > (REGISTER) on sofia profile 'public' for [1730 at 1.2.3.4] from ip 5.6.7.8" > - Scanner does not respond > - After a while, Freeswitch logs the following: > 2ae23e93-c929-4089-a594-8e7af633ca88 2021-01-25 12:28:37.506078 [WARNING] > switch_core_state_machine.c:687 2ae23e93-c929-4089-a594-8e7af633ca88 > sofia/public/1730 at 1.2.3.4 Abandoned > > -- > > In our case, we made fail2ban more sensitive to auth failures logs which > does not get triggered because of the scanner not even trying to send > credentials. > > Wouldn't it make more sense for this log to include the IP of sip client > that abandoned the call (5.6.7.8) instead of only the IP of the sip profile > (1.2.3.4) ? > > This would allow us to have Fail2ban block this scenario more aggressively. > > Thoughts ? > > > > > _________________________________________________________________________ > > 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 _________________________________________________________________________ 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 kitchm at tutanota.com Mon Jan 25 23:24:41 2021 From: kitchm at tutanota.com (kitchm at tutanota.com) Date: Tue, 26 Jan 2021 00:24:41 +0100 (CET) Subject: [Freeswitch-users] Rings Versus Seconds In-Reply-To: References: Message-ID: Thank you.  That clarifies things. So as I now understand it, the industry uses seconds to define the length of time the callee will be hearing any sounds made during the ringing process of the phone call before he/she picks up the call.  Further, the auto-attendent and/or routing of calls will be based upon counts of seconds at each step Do I understand that correctly? That is odd to the new users because we are used to setting the number of rings before our answering machine picks up. -- Sent with Tutanota, the secure & ad-free mailbox: https://tutanota.com Jan 25, 2021, 22:53 by joe at expert.net: > The issue is that the length of a ring can be different as heard by the caller depending on the country/location of the recipient.  Also, the ringing on the recipients' device/phone can also be variable based on their ring-tone preferences and the make and model of the device/phone.  In other words, 3 rings can consist of 2 different lengths as heard by the caller and the callee.  As such, duration in seconds is really the only consistent way I know how to measure the timing of a ring. > > On Tue, Jan 5, 2021 at 3:27 PM KitchM via FreeSWITCH-users <> freeswitch-users at lists.freeswitch.org> > wrote: > >> >> >> >> ---------- Forwarded message ---------- >> From: KitchM <>> kitchm at tutanota.com>> > >> To: >> freeswitch-users at lists.freeswitch.org >> Cc:  >> Bcc:  >> Date: Thu, 24 Dec 2020 10:15:31 -0700 (MST) >> Subject: Rings Versus Seconds >> 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/ >> >> >> >> >> ---------- Forwarded message ---------- >> From: KitchM via FreeSWITCH-users <>> freeswitch-users at lists.freeswitch.org>> > >> To: >> freeswitch-users at lists.freeswitch.org >> Cc:  >> Bcc:  >> Date: Tue, 05 Jan 2021 13:27:42 -0800 (PST) >> Subject: [Freeswitch-users] Rings Versus Seconds >> _________________________________________________________________________ >> >> 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 arslan.saeed at overthewire.com.au Wed Jan 27 22:18:01 2021 From: arslan.saeed at overthewire.com.au (Arslan Saeed) Date: Thu, 28 Jan 2021 09:18:01 +1100 Subject: [Freeswitch-users] Freeswitch Interop issue with MS Teams Message-ID: Hi All. Has anyone seen any weird signalling behaviour of Freeswitch when interworking with MS Teams server? We have seen a few weird things where freeswitch is apparently not passing the signaling messages from one leg to the other in B2BUA mode. In one of the weird cases, freeswitch decides to not relay the SIP Bye message from the called party to the caller and just consumes it (resulting in the call to stay up from the caller's side). This has happened randomly as well as in one particular call scenario when a feature is activated on ms teams side. However SIP signaling wise we cannot see any difference between a good call (where SIP bye is relayed to caller by freeswitch and the one where it decides to stomp on it). Our freeswitch version is "FreeSWITCH (Version 1.10.5 -release-17-25569c1631 64bit)" I am not including any PCAPS or detailed configuration of freeswitch in the post yet and can provide it if required. Any ideas or link to some useful resources please? Thanks Arslan -- -------------- next part -------------- An HTML attachment was scrubbed... URL: From jmiller at wndswp.net Wed Jan 27 02:46:58 2021 From: jmiller at wndswp.net (Jim Miller) Date: Tue, 26 Jan 2021 21:46:58 -0500 Subject: [Freeswitch-users] Multi-homed box - strange NAT question Message-ID: <1cb2abd5-f114-54c3-9915-a5c274a74dc8@wndswp.net> Hi Folks I'm running FreeSWITCH Version 1.10.3-release~64bit (-release 64bit) on a FreeBSD 12.1 box. The issue I'm having is related to NAT, I'm sure no one has ever seen a post on this topic.... My configuration is a box that is multi homed with an Internet facing interface and a private IP LAN interface.  The clients (Polycoms) are on the private LAN interface but behind a NAT (pfsense) on this subnet.  If I have the clients route directly to the FS box's private LAN without NAT I can make this work but as soon as I NAT them (which I need to for other reasons) I don't see the registrations show up with fs_path or the other variables like I might expect. I've fiddled with the apply-nat-acl variable to no avail.  Thoughts? Thanks Jim From s.kainz at wnt.at Thu Jan 28 13:27:25 2021 From: s.kainz at wnt.at (Stefan Kainz) Date: Thu, 28 Jan 2021 13:27:25 +0000 Subject: [Freeswitch-users] Freeswitch Interop issue with MS Teams In-Reply-To: References: Message-ID: Hi Arslan, i have come across quite a few weird things regarding Freeswitch – MS Teams Interconnection. Maybe you can provide your sofia-profile xml that connects to MS Teams. That would help. Regards, Von: FreeSWITCH-users Im Auftrag von Arslan Saeed Gesendet: Mittwoch, 27. Januar 2021 23:18 An: freeswitch-users at lists.freeswitch.org Betreff: [Freeswitch-users] Freeswitch Interop issue with MS Teams Hi All. Has anyone seen any weird signalling behaviour of Freeswitch when interworking with MS Teams server? We have seen a few weird things where freeswitch is apparently not passing the signaling messages from one leg to the other in B2BUA mode. In one of the weird cases, freeswitch decides to not relay the SIP Bye message from the called party to the caller and just consumes it (resulting in the call to stay up from the caller's side). This has happened randomly as well as in one particular call scenario when a feature is activated on ms teams side. However SIP signaling wise we cannot see any difference between a good call (where SIP bye is relayed to caller by freeswitch and the one where it decides to stomp on it). Our freeswitch version is "FreeSWITCH (Version 1.10.5 -release-17-25569c1631 64bit)" I am not including any PCAPS or detailed configuration of freeswitch in the post yet and can provide it if required. Any ideas or link to some useful resources please? Thanks Arslan -- -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Thu Jan 28 15:51:17 2021 From: brian at freeswitch.com (Brian West) Date: Thu, 28 Jan 2021 09:51:17 -0600 Subject: [Freeswitch-users] Multi-homed box - strange NAT question In-Reply-To: <1cb2abd5-f114-54c3-9915-a5c274a74dc8@wndswp.net> References: <1cb2abd5-f114-54c3-9915-a5c274a74dc8@wndswp.net> Message-ID: You will require one profile per nat interface, you can't cross profiles between transit providers without it. /b On Thu, Jan 28, 2021 at 7:25 AM Jim Miller wrote: > Hi Folks > > I'm running FreeSWITCH Version 1.10.3-release~64bit (-release 64bit) on > a FreeBSD 12.1 box. > > The issue I'm having is related to NAT, I'm sure no one has ever seen a > post on this topic.... > > My configuration is a box that is multi homed with an Internet facing > interface and a private IP LAN interface. The clients (Polycoms) are on > the private LAN interface but behind a NAT (pfsense) on this subnet. If > I have the clients route directly to the FS box's private LAN without > NAT I can make this work but as soon as I NAT them (which I need to for > other reasons) I don't see the registrations show up with fs_path or the > other variables like I might expect. > > I've fiddled with the apply-nat-acl variable to no avail. > > Thoughts? > > Thanks > > Jim > > > _________________________________________________________________________ > > 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 rehfjo at gmail.com Thu Jan 28 13:37:21 2021 From: rehfjo at gmail.com (Robert Fitzjohn) Date: Thu, 28 Jan 2021 13:37:21 +0000 Subject: [Freeswitch-users] Rings Versus Seconds In-Reply-To: References: Message-ID: > That is odd to the new users because we are used to setting the number of rings before our answering machine picks up Likely cos the phone is localized/sold in a certain country, the manufacturer has already done the required calculation. On Thu, 28 Jan 2021 at 12:37, KitchM via FreeSWITCH-users < freeswitch-users at lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: kitchm at tutanota.com > To: Joseph Barrero > Cc: FreeSWITCH Users Help > Bcc: > Date: Tue, 26 Jan 2021 00:24:41 +0100 (CET) > Subject: Re: [Freeswitch-users] Rings Versus Seconds > Thank you. That clarifies things. > > So as I now understand it, the industry uses seconds to define the length > of time the callee will be hearing any sounds made during the ringing > process of the phone call before he/she picks up the call. Further, the > auto-attendent and/or routing of calls will be based upon counts of seconds > at each step > > Do I understand that correctly? > > That is odd to the new users because we are used to setting the number of > rings before our answering machine picks up. > > -- > Sent with Tutanota, the secure & ad-free mailbox: > https://tutanota.com > > > Jan 25, 2021, 22:53 by joe at expert.net: > > The issue is that the length of a ring can be different as heard by the > caller depending on the country/location of the recipient. Also, the > ringing on the recipients' device/phone can also be variable based on their > ring-tone preferences and the make and model of the device/phone. In other > words, 3 rings can consist of 2 different lengths as heard by the caller > and the callee. As such, duration in seconds is really the only > consistent way I know how to measure the timing of a ring. > > On Tue, Jan 5, 2021 at 3:27 PM KitchM via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> wrote: > > > > > ---------- Forwarded message ---------- > From: KitchM > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Thu, 24 Dec 2020 10:15:31 -0700 (MST) > Subject: Rings Versus Seconds > 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/ > > > > > ---------- Forwarded message ---------- > From: KitchM via FreeSWITCH-users > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Tue, 05 Jan 2021 13:27:42 -0800 (PST) > Subject: [Freeswitch-users] Rings Versus Seconds > _________________________________________________________________________ > > 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 > > > > > > ---------- Forwarded message ---------- > From: KitchM via FreeSWITCH-users > To: Joseph Barrero > Cc: FreeSWITCH Users Help > Bcc: > Date: Thu, 28 Jan 2021 04:37:26 -0800 (PST) > Subject: Re: [Freeswitch-users] Rings Versus Seconds > _________________________________________________________________________ > > 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 Thu Jan 28 22:36:38 2021 From: brian at freeswitch.com (Brian West) Date: Thu, 28 Jan 2021 16:36:38 -0600 Subject: [Freeswitch-users] Multi-homed box - strange NAT question In-Reply-To: <57b35a9e-564e-0fd1-38bb-6d3675da6897@wndswp.net> References: <1cb2abd5-f114-54c3-9915-a5c274a74dc8@wndswp.net> <57b35a9e-564e-0fd1-38bb-6d3675da6897@wndswp.net> Message-ID: Without a full understanding of your network topology it's difficult to say. On Thu, Jan 28, 2021 at 3:53 PM Jim Miller wrote: > Brian > > Not sure I 100% follow. The clients are on the same /24 as the "internal" > profile interface is on. The only thing is they are behind a NAT. > > What led me to this was I had a previous configuration whereby the > internal and external profiles were on the same interface IP. When the > clients connected to the internal profile via an totally different public > IP, but also behind a NAT it worked (registrations showed fs_nat and a > fs_path properly). However, for this configuration when I put the clients > on a NAT that was on the same subnet as the internal and external shared IP > it wouldn't work. I thought maybe this was an issue with the profiles > sharing the same IP. Thus I split it to the configuration I documented > below. It makes me think that the NAT issue is related to the fact that > the profile IP is on the same subnet as the NAT. > > Jim > On 1/28/21 10:51 AM, Brian West wrote: > > You will require one profile per nat interface, you can't cross profiles > between transit providers without it. > > /b > > > On Thu, Jan 28, 2021 at 7:25 AM Jim Miller wrote: > >> Hi Folks >> >> I'm running FreeSWITCH Version 1.10.3-release~64bit (-release 64bit) on >> a FreeBSD 12.1 box. >> >> The issue I'm having is related to NAT, I'm sure no one has ever seen a >> post on this topic.... >> >> My configuration is a box that is multi homed with an Internet facing >> interface and a private IP LAN interface. The clients (Polycoms) are on >> the private LAN interface but behind a NAT (pfsense) on this subnet. If >> I have the clients route directly to the FS box's private LAN without >> NAT I can make this work but as soon as I NAT them (which I need to for >> other reasons) I don't see the registrations show up with fs_path or the >> other variables like I might expect. >> >> I've fiddled with the apply-nat-acl variable to no avail. >> >> Thoughts? >> >> Thanks >> >> Jim >> >> >> _________________________________________________________________________ >> >> 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] > > -- 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 brian at freeswitch.com Fri Jan 29 19:07:29 2021 From: brian at freeswitch.com (Brian West) Date: Fri, 29 Jan 2021 13:07:29 -0600 Subject: [Freeswitch-users] Multi-homed box - strange NAT question In-Reply-To: <7b82bb9d-5c80-fa5a-0a98-7beaa6d46adb@wndswp.net> References: <1cb2abd5-f114-54c3-9915-a5c274a74dc8@wndswp.net> <57b35a9e-564e-0fd1-38bb-6d3675da6897@wndswp.net> <7b82bb9d-5c80-fa5a-0a98-7beaa6d46adb@wndswp.net> Message-ID: see local-network-acl and make sure to set the ext-rtp-ip and ext-sip-ip to the prefix of autonat:x.x.x.x On Fri, Jan 29, 2021 at 1:06 PM Jim Miller wrote: > Let me try this. > > I have a public network interface connected to the external profile with > ip 1.1.1.1/24 (e.g. of course) I have a private subnet attached to the > internal profile on 192.168.0.2/24. I've got polycoms registering to > 192.168.0.2 using TLS that show up as 192.168.0.1 given they are NAT'd > behind this firewall. It seems that if the devices try to register to .2 > via an ip on the same subnet that NAT detection is not happy. When the > clients come from something totally different it works. Any way to force > this to work? > > Jim > On 1/28/21 5:36 PM, Brian West wrote: > > Without a full understanding of your network topology it's difficult to > say. > > > On Thu, Jan 28, 2021 at 3:53 PM Jim Miller wrote: > >> Brian >> >> Not sure I 100% follow. The clients are on the same /24 as the >> "internal" profile interface is on. The only thing is they are behind a >> NAT. >> >> What led me to this was I had a previous configuration whereby the >> internal and external profiles were on the same interface IP. When the >> clients connected to the internal profile via an totally different public >> IP, but also behind a NAT it worked (registrations showed fs_nat and a >> fs_path properly). However, for this configuration when I put the clients >> on a NAT that was on the same subnet as the internal and external shared IP >> it wouldn't work. I thought maybe this was an issue with the profiles >> sharing the same IP. Thus I split it to the configuration I documented >> below. It makes me think that the NAT issue is related to the fact that >> the profile IP is on the same subnet as the NAT. >> >> Jim >> On 1/28/21 10:51 AM, Brian West wrote: >> >> You will require one profile per nat interface, you can't cross profiles >> between transit providers without it. >> >> /b >> >> >> On Thu, Jan 28, 2021 at 7:25 AM Jim Miller wrote: >> >>> Hi Folks >>> >>> I'm running FreeSWITCH Version 1.10.3-release~64bit (-release 64bit) on >>> a FreeBSD 12.1 box. >>> >>> The issue I'm having is related to NAT, I'm sure no one has ever seen a >>> post on this topic.... >>> >>> My configuration is a box that is multi homed with an Internet facing >>> interface and a private IP LAN interface. The clients (Polycoms) are on >>> the private LAN interface but behind a NAT (pfsense) on this subnet. If >>> I have the clients route directly to the FS box's private LAN without >>> NAT I can make this work but as soon as I NAT them (which I need to for >>> other reasons) I don't see the registrations show up with fs_path or the >>> other variables like I might expect. >>> >>> I've fiddled with the apply-nat-acl variable to no avail. >>> >>> Thoughts? >>> >>> Thanks >>> >>> Jim >>> >>> >>> _________________________________________________________________________ >>> >>> 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] >> >> > > -- > > 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] > > -- 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 j.rupp at onecue.de Sun Jan 31 01:25:24 2021 From: j.rupp at onecue.de (Joshua Rupp) Date: Sun, 31 Jan 2021 01:25:24 +0000 Subject: [Freeswitch-users] Wrong Caller ID on Semi-Attended Transfer In-Reply-To: References: Message-ID: Hello, We've a problem with the caller id in one transfer scenario: In our Example Party A calls Party B, Party B answers the call and transfers it to Party C. We're talking about the Caller ID in the phone of Party C now. If B is doing an attended or unattended transfer, the caller id ist updating correctly as you would expect it. But there's a third scenario, which is very common for users of older PBX systems: A so called "semi-attended transfer". This means, that Party B is initiating A attended transfer via BLF Key, but hangs up, before Party C has answered the call. In this case, the Caller ID should update to Party A, as soon as B hangs up his phone. But this is not happening: The Caller ID remains on Party B until C answers the call. Only then the caller id is updated. Why is the caller id not updated after B hangs up and before C answers the call? Is there any possibility to change this behaiviour? Thanks for your help Joshua -------------- next part -------------- An HTML attachment was scrubbed... URL: From jmiller at wndswp.net Fri Jan 29 19:06:40 2021 From: jmiller at wndswp.net (Jim Miller) Date: Fri, 29 Jan 2021 14:06:40 -0500 Subject: [Freeswitch-users] Multi-homed box - strange NAT question In-Reply-To: References: <1cb2abd5-f114-54c3-9915-a5c274a74dc8@wndswp.net> <57b35a9e-564e-0fd1-38bb-6d3675da6897@wndswp.net> Message-ID: <7b82bb9d-5c80-fa5a-0a98-7beaa6d46adb@wndswp.net> Let me try this. I have a public network interface connected to the external profile with ip 1.1.1.1/24  (e.g. of course)  I have a private subnet attached to the internal profile on 192.168.0.2/24.   I've got polycoms registering to 192.168.0.2 using TLS that show up as 192.168.0.1 given they are NAT'd behind this firewall.  It seems that if the devices try to register to .2 via an ip on the same subnet that NAT detection is not happy.  When the clients come from something totally different it works.  Any way to force this to work? Jim On 1/28/21 5:36 PM, Brian West wrote: > Without a full understanding of your network topology it's difficult > to say. > > > On Thu, Jan 28, 2021 at 3:53 PM Jim Miller > wrote: > > Brian > > Not sure I 100% follow.  The clients are on the same /24 as the > "internal" profile interface is on.  The only thing is they are > behind a NAT.  > > What led me to this was I had a previous configuration whereby the > internal and external profiles were on the same interface IP. When > the clients connected to the internal profile via an totally > different public IP, but also behind a NAT it worked > (registrations showed fs_nat and a fs_path properly).  However, > for this configuration when I put the clients on a NAT that was on > the same subnet as the internal and external shared IP it wouldn't > work.  I thought maybe this was an issue with the profiles sharing > the same IP.  Thus I split it to the configuration I documented > below.  It makes me think that the NAT issue is related to the > fact that the profile IP is on the same subnet as the NAT.   > > Jim > > On 1/28/21 10:51 AM, Brian West wrote: >> You will require one profile per nat interface, you can't cross >> profiles between transit providers without it. >> >> /b >> >> >> On Thu, Jan 28, 2021 at 7:25 AM Jim Miller > > wrote: >> >> Hi Folks >> >> I'm running FreeSWITCH Version 1.10.3-release~64bit (-release >> 64bit) on >> a FreeBSD 12.1 box. >> >> The issue I'm having is related to NAT, I'm sure no one has >> ever seen a >> post on this topic.... >> >> My configuration is a box that is multi homed with an >> Internet facing >> interface and a private IP LAN interface.  The clients >> (Polycoms) are on >> the private LAN interface but behind a NAT (pfsense) on this >> subnet.  If >> I have the clients route directly to the FS box's private LAN >> without >> NAT I can make this work but as soon as I NAT them (which I >> need to for >> other reasons) I don't see the registrations show up with >> fs_path or the >> other variables like I might expect. >> >> I've fiddled with the apply-nat-acl variable to no avail.  >> >> Thoughts? >> >> Thanks >> >> Jim >> >> >> _________________________________________________________________________ >> >> 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://www.facebook.com/signalwireinc?src=email >> >> https://twitter.com/freeswitch >> > > > -- > > 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://www.facebook.com/signalwireinc?src=email > https://twitter.com/freeswitch > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jmiller at wndswp.net Thu Jan 28 21:53:21 2021 From: jmiller at wndswp.net (Jim Miller) Date: Thu, 28 Jan 2021 16:53:21 -0500 Subject: [Freeswitch-users] Multi-homed box - strange NAT question In-Reply-To: References: <1cb2abd5-f114-54c3-9915-a5c274a74dc8@wndswp.net> Message-ID: <57b35a9e-564e-0fd1-38bb-6d3675da6897@wndswp.net> Brian Not sure I 100% follow.  The clients are on the same /24 as the "internal" profile interface is on.  The only thing is they are behind a NAT.  What led me to this was I had a previous configuration whereby the internal and external profiles were on the same interface IP. When the clients connected to the internal profile via an totally different public IP, but also behind a NAT it worked (registrations showed fs_nat and a fs_path properly).  However, for this configuration when I put the clients on a NAT that was on the same subnet as the internal and external shared IP it wouldn't work.  I thought maybe this was an issue with the profiles sharing the same IP.  Thus I split it to the configuration I documented below.  It makes me think that the NAT issue is related to the fact that the profile IP is on the same subnet as the NAT.   Jim On 1/28/21 10:51 AM, Brian West wrote: > You will require one profile per nat interface, you can't cross > profiles between transit providers without it. > > /b > > > On Thu, Jan 28, 2021 at 7:25 AM Jim Miller > wrote: > > Hi Folks > > I'm running FreeSWITCH Version 1.10.3-release~64bit (-release > 64bit) on > a FreeBSD 12.1 box. > > The issue I'm having is related to NAT, I'm sure no one has ever > seen a > post on this topic.... > > My configuration is a box that is multi homed with an Internet facing > interface and a private IP LAN interface.  The clients (Polycoms) > are on > the private LAN interface but behind a NAT (pfsense) on this > subnet.  If > I have the clients route directly to the FS box's private LAN without > NAT I can make this work but as soon as I NAT them (which I need > to for > other reasons) I don't see the registrations show up with fs_path > or the > other variables like I might expect. > > I've fiddled with the apply-nat-acl variable to no avail.  > > Thoughts? > > Thanks > > Jim > > > _________________________________________________________________________ > > 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://www.facebook.com/signalwireinc?src=email > https://twitter.com/freeswitch > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From marcb at voicemeup.com Fri Jan 29 15:11:19 2021 From: marcb at voicemeup.com (Marc Bernard) Date: Fri, 29 Jan 2021 10:11:19 -0500 Subject: [Freeswitch-users] Odd RTP skew behavior Message-ID: <14b301d6f650$ffa9c120$fefd4360$@voicemeup.com> Hello Peeps, I often notice the following behavior on calls between two FreeSwitch servers: https://contattafiles.s3.us-west-1.amazonaws.com/getinfinity/4WUXPK9ibKr018c /Pasted%20Image%3A%20Jan%2029%2C%202021%20-%2010%3A05%3A07am Telco <==> FS Node 1 <== X ==> FS Node 2 <==> Callee UA Caller UA <==> FS Node 2 <== X ==> FS Node 1 <==> Telco We're not doing proxy media or bypass media. There are other calls at the same time that does not have this behavior, so I doubt it is caused by network issues. Does anyone have an idea of what could cause this ? Appreciate any help. Most Kindly, Marc -------------- next part -------------- An HTML attachment was scrubbed... URL: From marcb at voicemeup.com Fri Jan 29 15:34:39 2021 From: marcb at voicemeup.com (Marc Bernard) Date: Fri, 29 Jan 2021 10:34:39 -0500 Subject: [Freeswitch-users] Scanners and botnet vulnerability In-Reply-To: <4C9CA5FD-573F-46E8-B7C6-77C08F47A815@freeswitch.org> References: <113d01d6f341$6c93d5e0$45bb81a0$@voicemeup.com> <4C9CA5FD-573F-46E8-B7C6-77C08F47A815@freeswitch.org> Message-ID: <14d401d6f654$427a85e0$c76f91a0$@voicemeup.com> Hi Ken, >> Wouldn't it make more sense for this log to include the IP of sip client that abandoned the call (5.6.7.8) instead of only the IP of the sip profile (1.2.3.4) ? What about my suggestion though, which would allow us to block IPs when there is a lot of abandoned calls ? This could also be added to fail2ban by default with a more aggressive ban. Cheers, -----Original Message----- this is super common. this is more likely a recon attack than an actual brute force attempt. Eother that they are looking for something with auth turned off. we see tons of these things regularly. Fail to ban helps some but using a SIP RBL and dropping traffic via prefixes associated with regions and bad actor hosts seems to be the best course of action these days. I wont name the company, but a mjor european hosting company i drop their entire AS as its not worth the hassle. From rasheed.kalapurackal at gmail.com Sat Jan 30 18:43:33 2021 From: rasheed.kalapurackal at gmail.com (Rasheed Kalapurackal) Date: Sun, 31 Jan 2021 00:13:33 +0530 Subject: [Freeswitch-users] Freeswitch call recording Message-ID: Dear All , i am a newbie just started trying and learning Freeswitch platform. Until now my area of focus was call recording for various telephony platforms like Avaya, cisco , and many more.. i would like to know if there is any API available in Freeswitch platform for recording call by a third party application , by streaming the voice as a duplicate RTP stream to the IP of Call Recording application . what I understood that there is Call Recording API like session recordFile available in the ESL for recording the calls in the system itself. But i couldnt find an option to record from a 3rd party application by getting the duplicate RTP stream. please let me know if any information is available on this query. Thanks and regards Abdul Rasheed -------------- next part -------------- An HTML attachment was scrubbed... URL: