January 2021 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch call recording
by Rasheed Kalapurackal 01 Feb '21

01 Feb '21

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

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Wrong Caller ID on Semi-Attended Transfer
by Joshua Rupp 01 Feb '21

01 Feb '21

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

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Re: [Freeswitch-users] Multi-homed box - strange NAT question
by Brian West 29 Jan '21

29 Jan '21

Without a full understanding of your network topology it's difficult to say. On Thu, Jan 28, 2021 at 3:53 PM Jim Miller <jmiller(a)wndswp.net> 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 <jmiller(a)wndswp.net> 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(a)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(a)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(a)freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> > > Email: brian(a)freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> > > [image: https://www.facebook.com/signalwireinc?src=email] > <https://www.facebook.com/freeswitch> [image: > https://twitter.com/freeswitch] <https://twitter.com/freeswitch> > > -- Brian West | Co-founder and Developer Need Commercial support? email sales(a)freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> Email: brian(a)freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch>

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Multi-homed box - strange NAT question
by Jim Miller 28 Jan '21

28 Jan '21

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

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Re: [Freeswitch-users] Rings Versus Seconds
by Robert Fitzjohn 28 Jan '21

28 Jan '21

> 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(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: kitchm(a)tutanota.com > To: Joseph Barrero <joe(a)expert.net> > Cc: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > 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(a)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(a)lists.freeswitch.org> wrote: > > > > > ---------- Forwarded message ---------- > From: KitchM <kitchm(a)tutanota.com> > To: freeswitch-users(a)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(a)lists.freeswitch.org> > To: freeswitch-users(a)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(a)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(a)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 <freeswitch-users(a)lists.freeswitch.org> > To: Joseph Barrero <joe(a)expert.net> > Cc: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > 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(a)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(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com

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Freeswitch Interop issue with MS Teams
by Arslan Saeed 28 Jan '21

28 Jan '21

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 -- <https://overthewire.com.au/email-link/otwbanner>

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Re: [Freeswitch-users] Rings Versus Seconds
by Joseph Barrero 25 Jan '21

25 Jan '21

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(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: KitchM <kitchm(a)tutanota.com> > To: freeswitch-users(a)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(a)lists.freeswitch.org> > To: freeswitch-users(a)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(a)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(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com

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dtls-srtp problem with firefox 84.0.2
by Stefan 22 Jan '21

22 Jan '21

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.

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Limiting Calls Per Second
by Alexander Perkins 21 Jan '21

21 Jan '21

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

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verto js files
by Mundkowsky, Robert 21 Jan '21

21 Jan '21

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. ________________________________

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