From edson at inoutglobal.xyz  Mon Feb  1 11:13:14 2021
From: edson at inoutglobal.xyz (edson at inoutglobal.xyz)
Date: Mon, 1 Feb 2021 08:13:14 -0300
Subject: [Freeswitch-users] Feeeswitch HA with Keepalived database conflict
Message-ID: <000901d6f88b$404ba450$c0e2ecf0$@inoutglobal.xyz>

 

Dears,

 

I am configring FReeswitch HA with Keepalived approach just like this
https://freeswitch.org/confluence/display/FREESWITCH/HA+keepalived

Freeswitch database is shared with Postgresql-BDR replication.

 

When executing fsctl crash on node 1, node 2 tries to recover call from its
recovery table, but it somehow tries to insert it again causing following
conclict.

How to avoid this behaviour?

 

>ERROR:  duplicate key value violates unique constraint "recovery_pkey"

< 2021-01-31 12:48:09.394 -03 >DETAIL:  Key
(uuid)=(66727254-af9d-47ef-ad9f-6f6bbb794eb6) already exists.

< 2021-01-31 12:48:09.394 -03 >STATEMENT:  insert into recovery
(runtime_uuid, technology, profile_name, hostname, uuid, metadata) values
('4a864869-fe7f-4470-9992-20b91d09040f','sofia','internal','freeswitch-bndes
','66727254-af9d-47ef-ad9f-6f6bbb794eb6','<cdr
core-uuid="4a864869-fe7f-4470-9992-20b91d09040f"
switchname="freeswitch-bndes">

 

Regards

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210201/f0d7e102/attachment.html>

From daveh at drachtio.org  Mon Feb  1 15:00:04 2021
From: daveh at drachtio.org (David Horton)
Date: Mon, 1 Feb 2021 10:00:04 -0500
Subject: [Freeswitch-users] Freeswitch call recording
In-Reply-To: <CAKU4o50M3H62Rrw_So2oNxCOedKC+_9Ti7-QOHWoKMxc4th0Rg@mail.gmail.com>
References: <CAKU4o50M3H62Rrw_So2oNxCOedKC+_9Ti7-QOHWoKMxc4th0Rg@mail.gmail.com>
Message-ID: <8372B721-FA3E-4BF7-84B1-FB0D2872CD7F@drachtio.org>

There are some open source modules that do something similar*, including this one I wrote:

https://github.com/drachtio/drachtio-freeswitch-modules/tree/master/modules/mod_audio_fork <https://github.com/drachtio/drachtio-freeswitch-modules/tree/master/modules/mod_audio_fork>

*by “similar” I mean it does not duplicate the RTP stream, headers and all, but rather sends the voice stream in linear16 format to a far-end websocket server that you provide.

Dave

On Jan 30, 2021, at 1:43 PM, Rasheed Kalapurackal <rasheed.kalapurackal at gmail.com> wrote:

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
_________________________________________________________________________

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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210201/c8769768/attachment.html>

From brian at freeswitch.com  Mon Feb  1 16:50:35 2021
From: brian at freeswitch.com (Brian West)
Date: Mon, 1 Feb 2021 10:50:35 -0600
Subject: [Freeswitch-users] Wrong Caller ID on Semi-Attended Transfer
In-Reply-To: <AM0PR10MB2564F1E0D35DFDE3615B1E7F81B79@AM0PR10MB2564.EURPRD10.PROD.OUTLOOK.COM>
References: <AM0PR10MB256469A0C786E6A4CC9BB42081B79@AM0PR10MB2564.EURPRD10.PROD.OUTLOOK.COM>
 <AM0PR10MB2564F1E0D35DFDE3615B1E7F81B79@AM0PR10MB2564.EURPRD10.PROD.OUTLOOK.COM>
Message-ID: <CAEJMVkDc_ea==B0WyrEkkxGfirZm2tdoYuOYYNscDo7zkuuPhw@mail.gmail.com>

There is no such thing as a semi attended transfer, doing that transfer in
early media is NOT valid, its against the RFC... the fact it even works is
a trick we do, technically we should just hang up.  Your best bet is to
disable that transfer in the device.

/b


On Mon, Feb 1, 2021 at 9:14 AM Joshua Rupp <j.rupp at onecue.de> wrote:

> 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
> _________________________________________________________________________
>
> 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
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>

Email: brian at 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>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210201/5d003c23/attachment.html>

From brian at freeswitch.com  Mon Feb  1 16:51:25 2021
From: brian at freeswitch.com (Brian West)
Date: Mon, 1 Feb 2021 10:51:25 -0600
Subject: [Freeswitch-users] Odd RTP skew behavior
In-Reply-To: <14b301d6f650$ffa9c120$fefd4360$@voicemeup.com>
References: <14b301d6f650$ffa9c120$fefd4360$@voicemeup.com>
Message-ID: <CAEJMVkCmCK5jYemysSUmmNH+u=5kU92QVi8aW+W5LqDYtDj17A@mail.gmail.com>

You didn't mention the revision of freeswitch you're using?

On Mon, Feb 1, 2021 at 10:17 AM Marc Bernard <marcb at voicemeup.com> wrote:

> 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
> _________________________________________________________________________
>
> 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
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>

Email: brian at 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>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210201/7a3c8f29/attachment-0001.html>

From brian at freeswitch.com  Mon Feb  1 18:42:34 2021
From: brian at freeswitch.com (Brian West)
Date: Mon, 1 Feb 2021 12:42:34 -0600
Subject: [Freeswitch-users] Scanners and botnet vulnerability
In-Reply-To: <14d401d6f654$427a85e0$c76f91a0$@voicemeup.com>
References: <113d01d6f341$6c93d5e0$45bb81a0$@voicemeup.com>
 <4C9CA5FD-573F-46E8-B7C6-77C08F47A815@freeswitch.org>
 <14d401d6f654$427a85e0$c76f91a0$@voicemeup.com>
Message-ID: <CAEJMVkDWLNv0TQY42PVkJqvymHPGFre8F4_mrHYrK5nk2K1=9A@mail.gmail.com>

It would log it, unless you have it misconfigured.

On Mon, Feb 1, 2021 at 11:02 AM Marc Bernard <marcb at voicemeup.com> wrote:

> 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.
>
>
> _________________________________________________________________________
>
> 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
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>

Email: brian at 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>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210201/9cf00698/attachment.html>

From marcb at voicemeup.com  Mon Feb  1 14:52:21 2021
From: marcb at voicemeup.com (Marc Bernard)
Date: Mon, 1 Feb 2021 09:52:21 -0500
Subject: [Freeswitch-users] users registering from IP or FQDN
In-Reply-To: <CANeteichNNJXPRxt3Wwa8KGm__Y_TL05gu=uU6VbbuUM-HwH0g@mail.gmail.com>
References: <CANeteidVEA4_dPJRE1R0WgoKRMoaYcfLpioZnRE2ctWeRvKNBQ@mail.gmail.com>
 <CAFGRPVru1opsQ-Oe_uwteTAj=pM3cA8+i_g8Wfc4p7FKa2dKSw@mail.gmail.com>
 <CANeteichNNJXPRxt3Wwa8KGm__Y_TL05gu=uU6VbbuUM-HwH0g@mail.gmail.com>
Message-ID: <024201d6f8a9$d877a710$8966f530$@voicemeup.com>

Did you set “force-register-domain” in your sofia profile ?

 

https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files#SofiaConfigurationFiles-force-register-domain

 

 

Sent: Friday, January 22, 2021 13:32
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] users registering from IP or FQDN

 

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,

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210201/75526d29/attachment-0001.html>

From rasheed.kalapurackal at gmail.com  Mon Feb  1 18:36:34 2021
From: rasheed.kalapurackal at gmail.com (Rasheed Kalapurackal)
Date: Tue, 2 Feb 2021 00:06:34 +0530
Subject: [Freeswitch-users] Freeswitch call recording
In-Reply-To: <8372B721-FA3E-4BF7-84B1-FB0D2872CD7F@drachtio.org>
References: <CAKU4o50M3H62Rrw_So2oNxCOedKC+_9Ti7-QOHWoKMxc4th0Rg@mail.gmail.com>
 <8372B721-FA3E-4BF7-84B1-FB0D2872CD7F@drachtio.org>
Message-ID: <CAKU4o536y6WPo+OnpxvEhbc9CcAovtE8kYWHv03eF_27TfHiuA@mail.gmail.com>

Hello Dave,
                   Thanks a lot for the information.  This was exactly what
I was looking for.  i hope that , the freeswitch will not be loaded too
much when more number of concurrent calls are getting streamed using
mod_audio_fork through a websocket.  on the other hand , In the normal
recording option available with Freeswitch , there is a high chance that
the Media processing for recording is taking lot of system resources and
will limit the number of concurrent calls.   Thanks again for the valuable
information you shared with me.

Thanks and regards
Rasheed

On Mon, Feb 1, 2021 at 8:38 PM David Horton <daveh at drachtio.org> wrote:

> There are some open source modules that do something similar*, including
> this one I wrote:
>
>
> https://github.com/drachtio/drachtio-freeswitch-modules/tree/master/modules/mod_audio_fork
>
> *by “similar” I mean it does not duplicate the RTP stream, headers and
> all, but rather sends the voice stream in linear16 format to a far-end
> websocket server that you provide.
>
> Dave
>
> On Jan 30, 2021, at 1:43 PM, Rasheed Kalapurackal <
> rasheed.kalapurackal at gmail.com> wrote:
>
> 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
> _________________________________________________________________________
>
> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210202/1cc726c1/attachment-0001.html>

From brian at freeswitch.com  Mon Feb  1 21:44:43 2021
From: brian at freeswitch.com (Brian West)
Date: Mon, 1 Feb 2021 15:44:43 -0600
Subject: [Freeswitch-users] users registering from IP or FQDN
In-Reply-To: <024201d6f8a9$d877a710$8966f530$@voicemeup.com>
References: <CANeteidVEA4_dPJRE1R0WgoKRMoaYcfLpioZnRE2ctWeRvKNBQ@mail.gmail.com>
 <CAFGRPVru1opsQ-Oe_uwteTAj=pM3cA8+i_g8Wfc4p7FKa2dKSw@mail.gmail.com>
 <CANeteichNNJXPRxt3Wwa8KGm__Y_TL05gu=uU6VbbuUM-HwH0g@mail.gmail.com>
 <024201d6f8a9$d877a710$8966f530$@voicemeup.com>
Message-ID: <CAEJMVkA0hhEVeq0MAa94uZaV-gjh2D5jH6aBh5xRzsbM3e=vag@mail.gmail.com>

You have to use force-register-domain, or you use DNS host names, there is
no split/middle ground.

On Mon, Feb 1, 2021 at 3:06 PM Marc Bernard <marcb at voicemeup.com> wrote:

> Did you set “force-register-domain” in your sofia profile ?
>
>
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files#SofiaConfigurationFiles-force-register-domain
>
>
>
>
>
> *Sent:* Friday, January 22, 2021 13:32
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] users registering from IP or FQDN
>
>
>
> 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,
> _________________________________________________________________________
>
> 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
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>

Email: brian at 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>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210201/4b0214d7/attachment.html>

From nathan at robotics.net  Wed Feb  3 11:29:00 2021
From: nathan at robotics.net (Nathan Stratton)
Date: Wed, 3 Feb 2021 06:29:00 -0500
Subject: [Freeswitch-users] callcenter logout on no registration
Message-ID: <CAHubkyM5WwR141wL6S9WpBnFF1hhH8QizWv5cS47BnBW9F3ahg@mail.gmail.com>

I am playing around with call center, and noticed many times that users
will disconnect (stop registering) but be stuck in queue because they did
not log out. How hard would it be to have something that runs every minute
and log out all agents that are not registered?

><>
nathan stratton
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210203/1b8f12fd/attachment.html>

From heedfeld at gmail.com  Wed Feb  3 13:07:37 2021
From: heedfeld at gmail.com (Henning Heedfeld)
Date: Wed, 3 Feb 2021 14:07:37 +0100
Subject: [Freeswitch-users] callcenter logout on no registration
In-Reply-To: <CAHubkyM5WwR141wL6S9WpBnFF1hhH8QizWv5cS47BnBW9F3ahg@mail.gmail.com>
References: <CAHubkyM5WwR141wL6S9WpBnFF1hhH8QizWv5cS47BnBW9F3ahg@mail.gmail.com>
Message-ID: <C91B59D9-C7D1-4D8C-9CBA-A06A377BF85F@gmail.com>

Hi,

you can create a hook for this:

<configuration name="v8.conf" description="Google V8 JavaScript Plug-Ins">
	<settings>
		<hook event="CUSTOM" subclass="sofia::register" script="event_hook.js"/>
		<hook event="CUSTOM" subclass="sofia::unregister" script="event_hook.js"/>
	</settings>
	<modules>
	</modules>
</configuration>

event_hook.js:

var fromUser = event.getHeader("from-user");
var fromHost = event.getHeader("from-host");
var fromUri = event.getHeader("from-user") + "@" + event.getHeader("from-host");
var addFifo = "add callqueue" + "@" + fromHost + " user/" + fromUri + " 1 20 0";
var delFifo = "del callqueue" + "@" + fromHost + " user/" + fromUri;
var rc;
var regType;

if (event.getHeader("Event-Subclass") == "sofia::register") {
        regType = "Register";
} else if (event.getHeader("Event-Subclass") == "sofia::unregister") {
        regType = "Unregister";
}

if (fromUser >= 1001 && fromUser <= 1008) {
    console_log('info', 'USER ' + fromUri + " " + regType);
    if (regType == "Register") {
        rc = apiExecute("fifo_member", addFifo);
    } else if (regType == "Unregister") {
        rc = apiExecute("fifo_member", delFifo);
    }
    console_log(regType + " " + fromUri + " (User Agent: " + event.getHeader("user-agent") + ")");
    rc = apiExecute("fifo", "count callqueue@" + fromHost);
    console_log("FIFO status: " + rc);
}

This is based on a JS hook, but it will work with Lua, too.

My script does a login / logout for FIFO member if a user between 1001 and 1008 does a SIP login or logout. 
You have to adapt the commands for mod_callcenter.

hth
Henning




> Am 03.02.2021 um 12:29 schrieb Nathan Stratton <nathan at robotics.net>:
> 
> I am playing around with call center, and noticed many times that users will disconnect (stop registering) but be stuck in queue because they did not log out. How hard would it be to have something that runs every minute and log out all agents that are not registered?
> 
> ><>
> nathan stratton
> _________________________________________________________________________
> 
> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210203/dc6f2b1c/attachment.html>

From david.villasmil.work at gmail.com  Wed Feb  3 19:31:16 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 3 Feb 2021 19:31:16 +0000
Subject: [Freeswitch-users] callcenter logout on no registration
In-Reply-To: <C91B59D9-C7D1-4D8C-9CBA-A06A377BF85F@gmail.com>
References: <CAHubkyM5WwR141wL6S9WpBnFF1hhH8QizWv5cS47BnBW9F3ahg@mail.gmail.com>
 <C91B59D9-C7D1-4D8C-9CBA-A06A377BF85F@gmail.com>
Message-ID: <CAFGRPVoQMAFHFe0NDVC2FEkXBvvE4oedwEE2BA6wxqU-kNfLSA@mail.gmail.com>

Where would this file be located and what module loads it?

On Wed, 3 Feb 2021 at 13:08, Henning Heedfeld <heedfeld at gmail.com> wrote:

> Hi,
>
> you can create a hook for this:
>
> <configuration name="v8.conf" description="Google V8 JavaScript Plug-Ins">
> <settings>
> <hook event="CUSTOM" subclass="sofia::register" script="event_hook.js"/>
> <hook event="CUSTOM" subclass="sofia::unregister" script="event_hook.js"/>
> </settings>
> <modules>
> </modules>
> </configuration>
>
> event_hook.js:
>
> var fromUser = event.getHeader("from-user");
> var fromHost = event.getHeader("from-host");
> var fromUri = event.getHeader("from-user")
> + "@" + event.getHeader("from-host");
> var addFifo = "add callqueue" + "@" + fromHost + " user/" + fromUri + " 1
> 20 0";
> var delFifo = "del callqueue" + "@" + fromHost + " user/" + fromUri;
> var rc;
> var regType;
>
> if (event.getHeader("Event-Subclass") == "sofia::register") {
>         regType = "Register";
> } else if (event.getHeader("Event-Subclass") == "sofia::unregister") {
>         regType = "Unregister";
> }
>
> if (fromUser >= 1001 && fromUser <= 1008) {
>     console_log('info', 'USER ' + fromUri + " " + regType);
>     if (regType == "Register") {
>         rc = apiExecute("fifo_member", addFifo);
>     } else if (regType == "Unregister") {
>         rc = apiExecute("fifo_member", delFifo);
>     }
>     console_log(regType + " " + fromUri + " (User Agent:
> " + event.getHeader("user-agent") + ")");
>     rc = apiExecute("fifo", "count callqueue@" + fromHost);
>     console_log("FIFO status: " + rc);
> }
>
> This is based on a JS hook, but it will work with Lua, too.
>
> My script does a login / logout for FIFO member if a user between 1001 and
> 1008 does a SIP login or logout.
> You have to adapt the commands for mod_callcenter.
>
> hth
> Henning
>
>
>
>
> Am 03.02.2021 um 12:29 schrieb Nathan Stratton <nathan at robotics.net>:
>
> I am playing around with call center, and noticed many times that users
> will disconnect (stop registering) but be stuck in queue because they did
> not log out. How hard would it be to have something that runs every minute
> and log out all agents that are not registered?
>
> ><>
> nathan stratton
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210203/b997010b/attachment.html>

From avi at avimarcus.net  Wed Feb  3 20:13:39 2021
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 3 Feb 2021 20:13:39 +0000
Subject: [Freeswitch-users] callcenter logout on no registration
In-Reply-To: <CAFGRPVoQMAFHFe0NDVC2FEkXBvvE4oedwEE2BA6wxqU-kNfLSA@mail.gmail.com>
References: <CAHubkyM5WwR141wL6S9WpBnFF1hhH8QizWv5cS47BnBW9F3ahg@mail.gmail.com>
 <C91B59D9-C7D1-4D8C-9CBA-A06A377BF85F@gmail.com>
 <CAFGRPVoQMAFHFe0NDVC2FEkXBvvE4oedwEE2BA6wxqU-kNfLSA@mail.gmail.com>
Message-ID: <0100017769876e1a-f5e55f26-3a51-4b19-92da-9ff2a65c826e-000000@email.amazonses.com>

At conf/autoload_configs/v8.conf.xml

See:
https://github.com/signalwire/freeswitch/blob/master/conf/vanilla/autoload_configs/v8.conf.xml

I didn't know you could set a startup script there or trigger on events...
really cool.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210203/8d001e4e/attachment-0001.html>

From david.villasmil.work at gmail.com  Wed Feb  3 20:34:31 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 3 Feb 2021 20:34:31 +0000
Subject: [Freeswitch-users] callcenter logout on no registration
In-Reply-To: <0100017769876e1a-f5e55f26-3a51-4b19-92da-9ff2a65c826e-000000@email.amazonses.com>
References: <CAHubkyM5WwR141wL6S9WpBnFF1hhH8QizWv5cS47BnBW9F3ahg@mail.gmail.com>
 <C91B59D9-C7D1-4D8C-9CBA-A06A377BF85F@gmail.com>
 <CAFGRPVoQMAFHFe0NDVC2FEkXBvvE4oedwEE2BA6wxqU-kNfLSA@mail.gmail.com>
 <0100017769876e1a-f5e55f26-3a51-4b19-92da-9ff2a65c826e-000000@email.amazonses.com>
Message-ID: <CAFGRPVqkKJZaJASp8Yc3GBuNa_oh0dACPrg01HEKD3qpFDu-Cg@mail.gmail.com>

Very cool indeed!

On Wed, 3 Feb 2021 at 20:14, Avi Marcus <avi at avimarcus.net> wrote:

> At conf/autoload_configs/v8.conf.xml
>
> See:
> https://github.com/signalwire/freeswitch/blob/master/conf/vanilla/autoload_configs/v8.conf.xml
>
> I didn't know you could set a startup script there or trigger on events...
> really cool.
>
> _________________________________________________________________________
>
> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210203/f0f6d6ed/attachment.html>

From heedfeld at gmail.com  Wed Feb  3 20:56:20 2021
From: heedfeld at gmail.com (Henning Heedfeld)
Date: Wed, 3 Feb 2021 21:56:20 +0100
Subject: [Freeswitch-users] callcenter logout on no registration
In-Reply-To: <CAFGRPVoQMAFHFe0NDVC2FEkXBvvE4oedwEE2BA6wxqU-kNfLSA@mail.gmail.com>
References: <CAHubkyM5WwR141wL6S9WpBnFF1hhH8QizWv5cS47BnBW9F3ahg@mail.gmail.com>
 <C91B59D9-C7D1-4D8C-9CBA-A06A377BF85F@gmail.com>
 <CAFGRPVoQMAFHFe0NDVC2FEkXBvvE4oedwEE2BA6wxqU-kNfLSA@mail.gmail.com>
Message-ID: <11BD936C-84C2-43AA-B0B7-858741FF1269@gmail.com>

The script is located in your FS script directory, check the path in fs_cli:

eval ${script_dir}

mod_v8 does the job in this case.



> Am 03.02.2021 um 20:31 schrieb David Villasmil <david.villasmil.work at gmail.com>:
> 
> Where would this file be located and what module loads it?
> 
> On Wed, 3 Feb 2021 at 13:08, Henning Heedfeld <heedfeld at gmail.com <mailto:heedfeld at gmail.com>> wrote:
> Hi,
> 
> you can create a hook for this:
> 
> <configuration name="v8.conf" description="Google V8 JavaScript Plug-Ins">
> 	<settings>
> 		<hook event="CUSTOM" subclass="sofia::register" script="event_hook.js"/>
> 		<hook event="CUSTOM" subclass="sofia::unregister" script="event_hook.js"/>
> 	</settings>
> 	<modules>
> 	</modules>
> </configuration>
> 
> event_hook.js:
> 
> var fromUser = event.getHeader("from-user");
> var fromHost = event.getHeader("from-host");
> var fromUri = event.getHeader("from-user") + "@" + event.getHeader("from-host");
> var addFifo = "add callqueue" + "@" + fromHost + " user/" + fromUri + " 1 20 0";
> var delFifo = "del callqueue" + "@" + fromHost + " user/" + fromUri;
> var rc;
> var regType;
> 
> if (event.getHeader("Event-Subclass") == "sofia::register") {
>         regType = "Register";
> } else if (event.getHeader("Event-Subclass") == "sofia::unregister") {
>         regType = "Unregister";
> }
> 
> if (fromUser >= 1001 && fromUser <= 1008) {
>     console_log('info', 'USER ' + fromUri + " " + regType);
>     if (regType == "Register") {
>         rc = apiExecute("fifo_member", addFifo);
>     } else if (regType == "Unregister") {
>         rc = apiExecute("fifo_member", delFifo);
>     }
>     console_log(regType + " " + fromUri + " (User Agent: " + event.getHeader("user-agent") + ")");
>     rc = apiExecute("fifo", "count callqueue@" + fromHost);
>     console_log("FIFO status: " + rc);
> }
> 
> This is based on a JS hook, but it will work with Lua, too.
> 
> My script does a login / logout for FIFO member if a user between 1001 and 1008 does a SIP login or logout. 
> You have to adapt the commands for mod_callcenter.
> 
> hth
> Henning
> 
> 
> 
> 
>> Am 03.02.2021 um 12:29 schrieb Nathan Stratton <nathan at robotics.net <mailto:nathan at robotics.net>>:
>> 
>> I am playing around with call center, and noticed many times that users will disconnect (stop registering) but be stuck in queue because they did not log out. How hard would it be to have something that runs every minute and log out all agents that are not registered?
>> 
>> ><>
>> nathan stratton
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <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 <https://signalwire.community/>
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>> https://cluecon.com <https://cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> https://freeswitch.com <https://freeswitch.com/>
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <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 <https://signalwire.community/>
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>-- 
> Regards,
> 
> David Villasmil
> email: david.villasmil.work at gmail.com <mailto: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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210203/a3d4339f/attachment-0001.html>

From girish.dharmaraj at gmail.com  Fri Feb  5 11:25:41 2021
From: girish.dharmaraj at gmail.com (Giri)
Date: Fri, 5 Feb 2021 19:25:41 +0800
Subject: [Freeswitch-users] Mod Python - Failing to Compile
Message-ID: <CALv+5v5QthqKpd-hWuwkfGxq7itc6sttkgzb2KVgf9bt9ovDTQ@mail.gmail.com>

Hello Sir.

mod_python fails to build using the below version . Any ideas why that
happen ?

Python Version

python -V
Python 3.7.3

Debian OS Version:

PRETTY_NAME="Debian GNU/Linux 10 (buster)"
NAME="Debian GNU/Linux"
VERSION_ID="10"
VERSION="10 (buster)"

With Best Regards,
Girish Dharmaraj
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210205/4f86ddb5/attachment.html>

From gidoramothra at gmail.com  Thu Feb  4 14:03:05 2021
From: gidoramothra at gmail.com (Stefan)
Date: Thu, 4 Feb 2021 15:03:05 +0100
Subject: [Freeswitch-users] mod_verto,
 verto communicator working only with chromium-based browsers
Message-ID: <20210204140305.GA1394@localhost.localdomain>

Hello, I asked that before, but perhaps the question was somehow wrong
formulated or I didn't provide the right details. My Problem is the
following:

I have a freeswitch 1.10 installation, you can find the container I
use, including a configuration similar to mine, here:

https://github.com/gidmoth/freeswitch-container

(The readme in the etc-freeswitch folder is the one provided by the
freeswitch-minimal config package in debian, but please look at the files
themselfes, because I took that only as a startingpoint).

I have mod_verto enabled and try to connect clients with verto
communicator, build from the sources in the 1.10 release branch on
signalwire/freeswitch (https://github.com/signalwire/freeswitch).

It works perfectly well with chomium-based browsers, but with firefox it
does not work.

>From the logs it is clear, that firefox seems not to start the
negotiation of a key for srtp, and then mod_verto refuses to execute the
call, because the dialplan requires srtp. Here the relevant log, a
failed attemt to call a conference with firefox/verto communicator as a
client:

```
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"
	}
```

My question is: is there anything I can change to make it work with
firefox. Any hints welcome.

__
s.



From jmiller at wndswp.net  Tue Feb  2 16:49:43 2021
From: jmiller at wndswp.net (Jim Miller)
Date: Tue, 2 Feb 2021 11:49:43 -0500
Subject: [Freeswitch-users] Multi-homed box - strange NAT question
In-Reply-To: <CAEJMVkCT++8SLCn-c95pgJyFpxp8frS-hwW3iBv_-SP1v_YR3w@mail.gmail.com>
References: <1cb2abd5-f114-54c3-9915-a5c274a74dc8@wndswp.net>
 <CAEJMVkBPgw=zOvD=WjBPGLSYFYBGN7gmLkhjWccywmrzpMsZ9A@mail.gmail.com>
 <57b35a9e-564e-0fd1-38bb-6d3675da6897@wndswp.net>
 <CAEJMVkCpxpaQAhUqgBLPzcCLhb0J8ZABoUqNbSdzWr=71bbceQ@mail.gmail.com>
 <7b82bb9d-5c80-fa5a-0a98-7beaa6d46adb@wndswp.net>
 <CAEJMVkCT++8SLCn-c95pgJyFpxp8frS-hwW3iBv_-SP1v_YR3w@mail.gmail.com>
Message-ID: <312ad35f-c3b1-c8af-5c84-99b025630557@wndswp.net>

I was getting ready to implement this but I'm questioning something.  

The PBX itself is not behind a NAT.  Why would I set the ext-sip-ip and
ext-rtp-ip to use autonat? 

Presently thety are set to the below where private_rtp_ip is the RFC
address space IP of the internal interface of the PBX.

<param name="ext-rtp-ip" value="$${private_rtp_ip}"/>
<param name="ext-sip-ip" value="$${private_rtp_ip}"/>

Also, in the profile I have <param name="local-network-acl"
value="localnet.auto"/> already.

thoughts?

On 1/29/21 2:07 PM, Brian West wrote:
> 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 <jmiller at wndswp.net
> <mailto:jmiller at wndswp.net>> wrote:
>
>     Let me try this.
>
>     I have a public network interface connected to the external
>     profile with ip 1.1.1.1/24 <http://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 <http://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 <jmiller at wndswp.net
>>     <mailto:jmiller at 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 at wndswp.net <mailto:jmiller at 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 at freeswitch.com <mailto: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
>>>             <mailto: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
>>>         <mailto:sales at freeswitch.com>
>>>
>>>         FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield,
>>>         WI 53045
>>>         <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>>
>>>         Email: brian at freeswitch.com <mailto:brian at freeswitch.com>
>>>
>>>         Mobile: 918-424-9378
>>>
>>>         Website: https://www.FreeSWITCH.com
>>>         <https://www.freeswitch.com/>
>>>
>>>         https://www.facebook.com/signalwireinc?src=email
>>>         <https://www.facebook.com/freeswitch>
>>>         https://twitter.com/freeswitch <https://twitter.com/freeswitch>
>>>
>>
>>
>>     -- 
>>
>>     Brian West | Co-founder and Developer
>>
>>     Need Commercial support? email sales at freeswitch.com
>>     <mailto:sales at freeswitch.com>
>>
>>     FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI
>>     53045
>>     <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>
>>     Email: brian at freeswitch.com <mailto:brian at freeswitch.com>
>>
>>     Mobile: 918-424-9378
>>
>>     Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>
>>     https://www.facebook.com/signalwireinc?src=email
>>     <https://www.facebook.com/freeswitch>
>>     https://twitter.com/freeswitch <https://twitter.com/freeswitch>
>>
>
>
> -- 
>
> Brian West | Co-founder and Developer
>
> Need Commercial support? email sales at freeswitch.com
> <mailto:sales at freeswitch.com>
>
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>
> Email: brian at freeswitch.com <mailto:brian at freeswitch.com>
>
> Mobile: 918-424-9378
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
> https://www.facebook.com/signalwireinc?src=email
> <https://www.facebook.com/freeswitch> https://twitter.com/freeswitch
> <https://twitter.com/freeswitch>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210202/e530532c/attachment-0001.html>

From gregor at infomedia.si  Fri Feb  5 14:41:56 2021
From: gregor at infomedia.si (Gregor Nanger)
Date: Fri, 5 Feb 2021 15:41:56 +0100
Subject: [Freeswitch-users] mod_verto,
 verto communicator working only with chromium-based browsers
In-Reply-To: <20210204140305.GA1394@localhost.localdomain>
References: <20210204140305.GA1394@localhost.localdomain>
Message-ID: <CAC7ttFEei+sWgzjaieYL81f0=HRN02rqryfuZUNbyO+V1RFweQ@mail.gmail.com>

Just for info. We are using verto and it works in Chrome, Edge and Firefox.
Out of the box.

INCOMPATIBLE_DESTINATION could be because of codecs. Try to allow only
pcma, pcmu in verto config. Just a thought to test it...




On Fri, Feb 5, 2021, 15:34 Stefan <gidoramothra at gmail.com> wrote:

> Hello, I asked that before, but perhaps the question was somehow wrong
> formulated or I didn't provide the right details. My Problem is the
> following:
>
> I have a freeswitch 1.10 installation, you can find the container I
> use, including a configuration similar to mine, here:
>
> https://github.com/gidmoth/freeswitch-container
>
> (The readme in the etc-freeswitch folder is the one provided by the
> freeswitch-minimal config package in debian, but please look at the files
> themselfes, because I took that only as a startingpoint).
>
> I have mod_verto enabled and try to connect clients with verto
> communicator, build from the sources in the 1.10 release branch on
> signalwire/freeswitch (https://github.com/signalwire/freeswitch).
>
> It works perfectly well with chomium-based browsers, but with firefox it
> does not work.
>
> From the logs it is clear, that firefox seems not to start the
> negotiation of a key for srtp, and then mod_verto refuses to execute the
> call, because the dialplan requires srtp. Here the relevant log, a
> failed attemt to call a conference with firefox/verto communicator as a
> client:
>
> ```
> 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"
>         }
> ```
>
> My question is: is there anything I can change to make it work with
> firefox. Any hints welcome.
>
> __
> s.
>
>
> _________________________________________________________________________
>
> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210205/c4f6c513/attachment.html>

From ykalayy7 at gmail.com  Tue Feb  2 14:25:11 2021
From: ykalayy7 at gmail.com (Yusuf KALAY)
Date: Tue, 2 Feb 2021 17:25:11 +0300
Subject: [Freeswitch-users] {mod_http_cache} Is there a way to use
 http_remove_cache command with multiple file
Message-ID: <CANM0f9cQswVwP2-Lexuf4SEi0_w4rHrWvcDEzoQ0h=zN5+swGQ@mail.gmail.com>

Hi everyone,
I want to delete multiple http caches from freeswitch in one command
instead of one command for each file. Is there any way to do that? I tried
a separate the files with '&' or ' ', but it does not work. Like below:

->http_remove_cache
http://example.com/media/hello_world.wav&http_remove_cache
http://example.com/media/hello_world_2.wav
Or
->http_remove_cache
http://example.com/media/hello_world.wav&http://example.com/media/hello_world_2.wav

Regards Yusuf
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210202/c3e6fb0b/attachment.html>

From cmrienzo at gmail.com  Fri Feb  5 14:52:05 2021
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Fri, 5 Feb 2021 09:52:05 -0500
Subject: [Freeswitch-users] mod_verto,
 verto communicator working only with chromium-based browsers
In-Reply-To: <20210204140305.GA1394@localhost.localdomain>
References: <20210204140305.GA1394@localhost.localdomain>
Message-ID: <CAD-qRkhiq2QexZG4N9TjUVoDFYfNFGhEtkHuRh0FijQ9tk_7Tg@mail.gmail.com>

Check the SDP- the negotiation failed due to

2021-01-21 15:36:48.773228 [WARNING] switch_core_media.c:6253 Crypto not
negotiated but required.



On Fri, Feb 5, 2021 at 9:50 AM Stefan <gidoramothra at gmail.com> wrote:

> Hello, I asked that before, but perhaps the question was somehow wrong
> formulated or I didn't provide the right details. My Problem is the
> following:
>
> I have a freeswitch 1.10 installation, you can find the container I
> use, including a configuration similar to mine, here:
>
> https://github.com/gidmoth/freeswitch-container
>
> (The readme in the etc-freeswitch folder is the one provided by the
> freeswitch-minimal config package in debian, but please look at the files
> themselfes, because I took that only as a startingpoint).
>
> I have mod_verto enabled and try to connect clients with verto
> communicator, build from the sources in the 1.10 release branch on
> signalwire/freeswitch (https://github.com/signalwire/freeswitch).
>
> It works perfectly well with chomium-based browsers, but with firefox it
> does not work.
>
> From the logs it is clear, that firefox seems not to start the
> negotiation of a key for srtp, and then mod_verto refuses to execute the
> call, because the dialplan requires srtp. Here the relevant log, a
> failed attemt to call a conference with firefox/verto communicator as a
> client:
>
> ```
> 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"
>         }
> ```
>
> My question is: is there anything I can change to make it work with
> firefox. Any hints welcome.
>
> __
> s.
>
>
> _________________________________________________________________________
>
> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210205/b3e7ce57/attachment-0001.html>

From amit.k.chaudhuri at gmail.com  Fri Feb  5 15:05:08 2021
From: amit.k.chaudhuri at gmail.com (Amit Chaudhuri)
Date: Fri, 5 Feb 2021 15:05:08 +0000
Subject: [Freeswitch-users] Mod Python - Failing to Compile
In-Reply-To: <CALv+5v5QthqKpd-hWuwkfGxq7itc6sttkgzb2KVgf9bt9ovDTQ@mail.gmail.com>
References: <CALv+5v5QthqKpd-hWuwkfGxq7itc6sttkgzb2KVgf9bt9ovDTQ@mail.gmail.com>
Message-ID: <CAH=VbNfGgxgRLPiDQu4d2b9DgopXdM4LAs-6YHoG5PVBYiZzJw@mail.gmail.com>

You should probably post the output from the build.....

On Fri, 5 Feb 2021 at 11:55, Giri <girish.dharmaraj at gmail.com> wrote:
>
> Hello Sir.
>
> mod_python fails to build using the below version . Any ideas why that happen ?
>
> Python Version
>
> python -V
> Python 3.7.3
>
> Debian OS Version:
>
> PRETTY_NAME="Debian GNU/Linux 10 (buster)"
> NAME="Debian GNU/Linux"
> VERSION_ID="10"
> VERSION="10 (buster)"
>
> With Best Regards,
> Girish Dharmaraj
> _________________________________________________________________________
>
> 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 girish.dharmaraj at gmail.com  Fri Feb  5 15:25:03 2021
From: girish.dharmaraj at gmail.com (Giri)
Date: Fri, 5 Feb 2021 23:25:03 +0800
Subject: [Freeswitch-users] Mod Python - Failing to Compile
In-Reply-To: <CAH=VbNfGgxgRLPiDQu4d2b9DgopXdM4LAs-6YHoG5PVBYiZzJw@mail.gmail.com>
References: <CALv+5v5QthqKpd-hWuwkfGxq7itc6sttkgzb2KVgf9bt9ovDTQ@mail.gmail.com>
 <CAH=VbNfGgxgRLPiDQu4d2b9DgopXdM4LAs-6YHoG5PVBYiZzJw@mail.gmail.com>
Message-ID: <CALv+5v6Bp_nxrL8_z+HFrPkLA-aJHucfK6YCOj_TsH1ewcdrWA@mail.gmail.com>

 CC       mod_python_la-mod_python.lo
mod_python.c: In function ‘print_python_error’:
mod_python.c:107:66: error: implicit declaration of function
‘PyString_Check’; did you mean ‘PyMapping_Check’?
[-Werror=implicit-function-declaration]
  if (pyType != NULL && (pyString=PyObject_Str(pyType))!=NULL &&
(PyString_Check(pyString))) {

^~~~~~~~~~~~~~

PyMapping_Check
mod_python.c:108:18: error: implicit declaration of function
‘PyString_AsString’; did you mean ‘PyBytes_AsString’?
[-Werror=implicit-function-declaration]
   strcat(buffer, PyString_AsString(pyString));
                  ^~~~~~~~~~~~~~~~~
                  PyBytes_AsString
mod_python.c:108:18: error: passing argument 2 of ‘strcat’ makes pointer
from integer without a cast [-Werror=int-conversion]
   strcat(buffer, PyString_AsString(pyString));
                  ^~~~~~~~~~~~~~~~~~~~~~~~~~~
In file included from /usr/include/python3.7m/Python.h:30,
                 from mod_python.c:35:
/usr/x86_64-linux-gnu/include/string.h:129:14: note: expected ‘const char *
restrict’ but argument is of type ‘int’
 extern char *strcat (char *__restrict __dest, const char *__restrict __src)
              ^~~~~~
mod_python.c:118:18: error: passing argument 2 of ‘strcat’ makes pointer
from integer without a cast [-Werror=int-conversion]
   strcat(buffer, PyString_AsString(pyString));
                  ^~~~~~~~~~~~~~~~~~~~~~~~~~~
In file included from /usr/include/python3.7m/Python.h:30,
                 from mod_python.c:35:
/usr/x86_64-linux-gnu/include/string.h:129:14: note: expected ‘const char *
restrict’ but argument is of type ‘int’
 extern char *strcat (char *__restrict __dest, const char *__restrict __src)
              ^~~~~~
mod_python.c:136:21: error: passing argument 2 of ‘strcat’ makes pointer
from integer without a cast [-Werror=int-conversion]
      strcat(buffer, PyString_AsString(pyResult));
                     ^~~~~~~~~~~~~~~~~~~~~~~~~~~
In file included from /usr/include/python3.7m/Python.h:30,
                 from mod_python.c:35:
/usr/x86_64-linux-gnu/include/string.h:129:14: note: expected ‘const char *
restrict’ but argument is of type ‘int’
 extern char *strcat (char *__restrict __dest, const char *__restrict __src)
              ^~~~~~
mod_python.c:153:40: error: format ‘%s’ expects argument of type ‘char *’,
but argument 3 has type ‘int’ [-Werror=format=]
    sprintf((char*)sTemp, "\n\tFile: \"%s\", line %i, in %s",
                                       ~^
                                       %d
      PyString_AsString(pyTB->tb_frame->f_code->co_filename),
      ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
mod_python.c:153:58: error: format ‘%s’ expects argument of type ‘char *’,
but argument 5 has type ‘int’ [-Werror=format=]
    sprintf((char*)sTemp, "\n\tFile: \"%s\", line %i, in %s",
                                                         ~^
                                                         %d
mod_python.c:156:6:
      PyString_AsString(pyTB->tb_frame->f_code->co_name) );
      ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
mod_python.c: In function ‘eval_some_python’:
mod_python.c:313:18: error: cast to pointer from integer of different size
[-Werror=int-to-pointer-cast]
    *str = strdup((char *) PyString_AsString(result));
                  ^
mod_python.c: In function ‘mod_python_shutdown’:
mod_python.c:608:2: error: ‘PyEval_AcquireLock’ is deprecated
[-Werror=deprecated-declarations]
  PyEval_AcquireLock();
  ^~~~~~~~~~~~~~~~~~
In file included from /usr/include/python3.7m/Python.h:141,
                 from mod_python.c:35:
/usr/include/python3.7m/ceval.h:198:18: note: declared here
 PyAPI_FUNC(void) PyEval_AcquireLock(void) Py_DEPRECATED(3.2);
                  ^~~~~~~~~~~~~~~~~~
mod_python.c:645:2: error: ‘PyEval_AcquireLock’ is deprecated
[-Werror=deprecated-declarations]
  PyEval_AcquireLock();
  ^~~~~~~~~~~~~~~~~~
In file included from /usr/include/python3.7m/Python.h:141,
                 from mod_python.c:35:
/usr/include/python3.7m/ceval.h:198:18: note: declared here
 PyAPI_FUNC(void) PyEval_AcquireLock(void) Py_DEPRECATED(3.2);
                  ^~~~~~~~~~~~~~~~~~
cc1: all warnings being treated as errors
make: *** [Makefile:743: mod_python_la-mod_python.lo] Error 1


With Best Regards,
Girish Dharmaraj


On Fri, Feb 5, 2021 at 11:05 PM Amit Chaudhuri <amit.k.chaudhuri at gmail.com>
wrote:

> You should probably post the output from the build.....
>
> On Fri, 5 Feb 2021 at 11:55, Giri <girish.dharmaraj at gmail.com> wrote:
> >
> > Hello Sir.
> >
> > mod_python fails to build using the below version . Any ideas why that
> happen ?
> >
> > Python Version
> >
> > python -V
> > Python 3.7.3
> >
> > Debian OS Version:
> >
> > PRETTY_NAME="Debian GNU/Linux 10 (buster)"
> > NAME="Debian GNU/Linux"
> > VERSION_ID="10"
> > VERSION="10 (buster)"
> >
> > With Best Regards,
> > Girish Dharmaraj
> > _________________________________________________________________________
> >
> > 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210205/3eb8cd99/attachment.html>

From cmrienzo at gmail.com  Fri Feb  5 15:54:10 2021
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Fri, 5 Feb 2021 10:54:10 -0500
Subject: [Freeswitch-users] {mod_http_cache} Is there a way to use
 http_remove_cache command with multiple file
In-Reply-To: <CANM0f9cQswVwP2-Lexuf4SEi0_w4rHrWvcDEzoQ0h=zN5+swGQ@mail.gmail.com>
References: <CANM0f9cQswVwP2-Lexuf4SEi0_w4rHrWvcDEzoQ0h=zN5+swGQ@mail.gmail.com>
Message-ID: <CAD-qRkgi2FfyNopXrv=rvegwVOr-B_5u9mC_USHakdxG49o+ZA@mail.gmail.com>

No, this isn't possible.  You can clear the whole cache, execute api to
expire an individual URL, prefix the url with {refresh=true}
http://example.com/myfile.wav to force it to discard the cached file,  have
your server set Cache-Control: max-age= to set a reasonable expiration, or
change the default expiration in the config if there is no Cache-Control
set.  Best practice is to rename the file if you want to ensure immediate
recaching.

Chris


On Fri, Feb 5, 2021 at 10:46 AM Yusuf KALAY <ykalayy7 at gmail.com> wrote:

> Hi everyone,
> I want to delete multiple http caches from freeswitch in one command
> instead of one command for each file. Is there any way to do that? I tried
> a separate the files with '&' or ' ', but it does not work. Like below:
>
> ->http_remove_cache
> http://example.com/media/hello_world.wav&http_remove_cache
> http://example.com/media/hello_world_2.wav
> Or
> ->http_remove_cache
> http://example.com/media/hello_world.wav&http://example.com/media/hello_world_2.wav
>
> Regards Yusuf
> _________________________________________________________________________
>
> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210205/711b9b7e/attachment-0001.html>

From amit.k.chaudhuri at gmail.com  Fri Feb  5 17:21:17 2021
From: amit.k.chaudhuri at gmail.com (Amit Chaudhuri)
Date: Fri, 5 Feb 2021 17:21:17 +0000
Subject: [Freeswitch-users] Mod Python - Failing to Compile
In-Reply-To: <CALv+5v6Bp_nxrL8_z+HFrPkLA-aJHucfK6YCOj_TsH1ewcdrWA@mail.gmail.com>
References: <CALv+5v5QthqKpd-hWuwkfGxq7itc6sttkgzb2KVgf9bt9ovDTQ@mail.gmail.com>
 <CAH=VbNfGgxgRLPiDQu4d2b9DgopXdM4LAs-6YHoG5PVBYiZzJw@mail.gmail.com>
 <CALv+5v6Bp_nxrL8_z+HFrPkLA-aJHucfK6YCOj_TsH1ewcdrWA@mail.gmail.com>
Message-ID: <CAH=VbNdNN4F+vXgWjKMAu3tC6O48ZOYr_owzmTpxh_j_3cKAdA@mail.gmail.com>

I'm not into python in a serious way but in general terms "implicit
declaration of function ‘PyString_Check’; " would suggest that the
compiler is looking for that function and cannot find it. Perhaps your
system is missing the provider of that function. I would start by
trying to find out where that is defined (i.e. provided) and whether
the provider is on your system. If it *is* on your system, is it in
the place that the build system is looking?

On Fri, 5 Feb 2021 at 15:42, Giri <girish.dharmaraj at gmail.com> wrote:
>
>  CC       mod_python_la-mod_python.lo
> mod_python.c: In function ‘print_python_error’:
> mod_python.c:107:66: error: implicit declaration of function ‘PyString_Check’; did you mean ‘PyMapping_Check’? [-Werror=implicit-function-declaration]
>   if (pyType != NULL && (pyString=PyObject_Str(pyType))!=NULL && (PyString_Check(pyString))) {
>                                                                   ^~~~~~~~~~~~~~
>                                                                   PyMapping_Check
> mod_python.c:108:18: error: implicit declaration of function ‘PyString_AsString’; did you mean ‘PyBytes_AsString’? [-Werror=implicit-function-declaration]
>    strcat(buffer, PyString_AsString(pyString));
>                   ^~~~~~~~~~~~~~~~~
>                   PyBytes_AsString
> mod_python.c:108:18: error: passing argument 2 of ‘strcat’ makes pointer from integer without a cast [-Werror=int-conversion]
>    strcat(buffer, PyString_AsString(pyString));
>                   ^~~~~~~~~~~~~~~~~~~~~~~~~~~
> In file included from /usr/include/python3.7m/Python.h:30,
>                  from mod_python.c:35:
> /usr/x86_64-linux-gnu/include/string.h:129:14: note: expected ‘const char * restrict’ but argument is of type ‘int’
>  extern char *strcat (char *__restrict __dest, const char *__restrict __src)
>               ^~~~~~
> mod_python.c:118:18: error: passing argument 2 of ‘strcat’ makes pointer from integer without a cast [-Werror=int-conversion]
>    strcat(buffer, PyString_AsString(pyString));
>                   ^~~~~~~~~~~~~~~~~~~~~~~~~~~
> In file included from /usr/include/python3.7m/Python.h:30,
>                  from mod_python.c:35:
> /usr/x86_64-linux-gnu/include/string.h:129:14: note: expected ‘const char * restrict’ but argument is of type ‘int’
>  extern char *strcat (char *__restrict __dest, const char *__restrict __src)
>               ^~~~~~
> mod_python.c:136:21: error: passing argument 2 of ‘strcat’ makes pointer from integer without a cast [-Werror=int-conversion]
>       strcat(buffer, PyString_AsString(pyResult));
>                      ^~~~~~~~~~~~~~~~~~~~~~~~~~~
> In file included from /usr/include/python3.7m/Python.h:30,
>                  from mod_python.c:35:
> /usr/x86_64-linux-gnu/include/string.h:129:14: note: expected ‘const char * restrict’ but argument is of type ‘int’
>  extern char *strcat (char *__restrict __dest, const char *__restrict __src)
>               ^~~~~~
> mod_python.c:153:40: error: format ‘%s’ expects argument of type ‘char *’, but argument 3 has type ‘int’ [-Werror=format=]
>     sprintf((char*)sTemp, "\n\tFile: \"%s\", line %i, in %s",
>                                        ~^
>                                        %d
>       PyString_AsString(pyTB->tb_frame->f_code->co_filename),
>       ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> mod_python.c:153:58: error: format ‘%s’ expects argument of type ‘char *’, but argument 5 has type ‘int’ [-Werror=format=]
>     sprintf((char*)sTemp, "\n\tFile: \"%s\", line %i, in %s",
>                                                          ~^
>                                                          %d
> mod_python.c:156:6:
>       PyString_AsString(pyTB->tb_frame->f_code->co_name) );
>       ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> mod_python.c: In function ‘eval_some_python’:
> mod_python.c:313:18: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
>     *str = strdup((char *) PyString_AsString(result));
>                   ^
> mod_python.c: In function ‘mod_python_shutdown’:
> mod_python.c:608:2: error: ‘PyEval_AcquireLock’ is deprecated [-Werror=deprecated-declarations]
>   PyEval_AcquireLock();
>   ^~~~~~~~~~~~~~~~~~
> In file included from /usr/include/python3.7m/Python.h:141,
>                  from mod_python.c:35:
> /usr/include/python3.7m/ceval.h:198:18: note: declared here
>  PyAPI_FUNC(void) PyEval_AcquireLock(void) Py_DEPRECATED(3.2);
>                   ^~~~~~~~~~~~~~~~~~
> mod_python.c:645:2: error: ‘PyEval_AcquireLock’ is deprecated [-Werror=deprecated-declarations]
>   PyEval_AcquireLock();
>   ^~~~~~~~~~~~~~~~~~
> In file included from /usr/include/python3.7m/Python.h:141,
>                  from mod_python.c:35:
> /usr/include/python3.7m/ceval.h:198:18: note: declared here
>  PyAPI_FUNC(void) PyEval_AcquireLock(void) Py_DEPRECATED(3.2);
>                   ^~~~~~~~~~~~~~~~~~
> cc1: all warnings being treated as errors
> make: *** [Makefile:743: mod_python_la-mod_python.lo] Error 1
>
>
> With Best Regards,
> Girish Dharmaraj
>
>
> On Fri, Feb 5, 2021 at 11:05 PM Amit Chaudhuri <amit.k.chaudhuri at gmail.com> wrote:
>>
>> You should probably post the output from the build.....
>>
>> On Fri, 5 Feb 2021 at 11:55, Giri <girish.dharmaraj at gmail.com> wrote:
>> >
>> > Hello Sir.
>> >
>> > mod_python fails to build using the below version . Any ideas why that happen ?
>> >
>> > Python Version
>> >
>> > python -V
>> > Python 3.7.3
>> >
>> > Debian OS Version:
>> >
>> > PRETTY_NAME="Debian GNU/Linux 10 (buster)"
>> > NAME="Debian GNU/Linux"
>> > VERSION_ID="10"
>> > VERSION="10 (buster)"
>> >
>> > With Best Regards,
>> > Girish Dharmaraj
>> > _________________________________________________________________________
>> >
>> > 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


From mario_fs at mgtech.com  Fri Feb  5 18:57:55 2021
From: mario_fs at mgtech.com (mario_fs)
Date: Fri, 5 Feb 2021 10:57:55 -0800
Subject: [Freeswitch-users] Homebrew now on Mac M1
Message-ID: <B14A130F-6FD0-4E7F-9FE8-D28793D56650@mgtech.com>

An FYI for the people running Freeswitch on MacOS: homebrew announced 3.0.0 today <https://brew.sh/2021/02/05/homebrew-3.0.0/> which runs nativity on Mac M1 computers. I don’t plan to get one for a long time but I would appreciate anyone who tries to build freeswitch on one to let us know how it went. You’ll have to wait though, Freeswitch 1.10.5 GA and 1.10.6 Development does not build or run on Big Sur or Catalina since November. Keep and eye on the macOS status page <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Installation> to check when you can build Freeswitch on macOS again.
Mario G
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210205/a8d7a071/attachment.html>

From brian at freeswitch.com  Fri Feb  5 21:47:57 2021
From: brian at freeswitch.com (Brian West)
Date: Fri, 5 Feb 2021 15:47:57 -0600
Subject: [Freeswitch-users] Homebrew now on Mac M1
In-Reply-To: <B14A130F-6FD0-4E7F-9FE8-D28793D56650@mgtech.com>
References: <B14A130F-6FD0-4E7F-9FE8-D28793D56650@mgtech.com>
Message-ID: <CAEJMVkDnb718_2RxyLAa=RzGjENKeBJGkwJpfsFwNFQO01bPXQ@mail.gmail.com>

Hasn't it been native for a while now?  Mine from two weeks ago is

On Fri, Feb 5, 2021 at 1:07 PM mario_fs <mario_fs at mgtech.com> wrote:

> An FYI for the people running Freeswitch on MacOS: homebrew announced
> 3.0.0 today <https://brew.sh/2021/02/05/homebrew-3.0.0/> which runs
> nativity on Mac M1 computers. I don’t plan to get one for a long time but I
> would appreciate anyone who tries to build freeswitch on one to let us know
> how it went. You’ll have to wait though, Freeswitch 1.10.5 GA and 1.10.6
> Development does not build or run on Big Sur or Catalina since November.
> Keep and eye on the macOS status page
> <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Installation> to
> check when you can build Freeswitch on macOS again.
> Mario G
> _________________________________________________________________________
>
> 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
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>

Email: brian at 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>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210205/81285ff3/attachment.html>

From mario_fs at mgtech.com  Mon Feb  8 06:35:43 2021
From: mario_fs at mgtech.com (mario_fs)
Date: Sun, 7 Feb 2021 22:35:43 -0800
Subject: [Freeswitch-users] Homebrew now on Mac M1
In-Reply-To: <CAEJMVkDnb718_2RxyLAa=RzGjENKeBJGkwJpfsFwNFQO01bPXQ@mail.gmail.com>
References: <B14A130F-6FD0-4E7F-9FE8-D28793D56650@mgtech.com>
 <CAEJMVkDnb718_2RxyLAa=RzGjENKeBJGkwJpfsFwNFQO01bPXQ@mail.gmail.com>
Message-ID: <1304141E-011C-4364-AABC-A98DDDF2573F@mgtech.com>

I don’t have an M1 so HB M1 support may have been in beta for you. M1 official support was announced Feb 5.

To add to the info below: I spent the weekend testing a clean install of the new homebrew 3.0 and freeswitch prerequisites on all macOS versions and found there is no way to build and run Freeswitch on any of the three supported macOS versions. Freeswitch segment faults on startup only on Mojave. I don’t plan on opening an issue since it could be homebrew related. Two other FS issues must be solved for Catalina and Big Sur (build issue is closed but the fix has not made it to master, other startup issue still open). I will keep the macOS status page <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Installation> updated.
Mario G

> On Feb 5, 2021, at 1:47 PM, Brian West <brian at freeswitch.com> wrote:
> 
> Hasn't it been native for a while now?  Mine from two weeks ago is
> 
> On Fri, Feb 5, 2021 at 1:07 PM mario_fs <mario_fs at mgtech.com <mailto:mario_fs at mgtech.com>> wrote:
> An FYI for the people running Freeswitch on MacOS: homebrew announced 3.0.0 today <https://brew.sh/2021/02/05/homebrew-3.0.0/> which runs nativity on Mac M1 computers. I don’t plan to get one for a long time but I would appreciate anyone who tries to build freeswitch on one to let us know how it went. You’ll have to wait though, Freeswitch 1.10.5 GA and 1.10.6 Development does not build or run on Big Sur or Catalina since November. Keep and eye on the macOS status page <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Installation> to check when you can build Freeswitch on macOS again.
> Mario G
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <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 <https://signalwire.community/>
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>
> 
> -- 
> 
> Brian West | Co-founder and Developer
> Need Commercial support? email sales at freeswitch.com <mailto:sales at freeswitch.com> 
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
> Email: brian at freeswitch.com <mailto:brian at freeswitch.com>
> Mobile: 918-424-9378
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>  <https://www.facebook.com/freeswitch>  <https://twitter.com/freeswitch>
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210207/0c50d4a6/attachment.html>

From mayamatakeshi at gmail.com  Mon Feb  8 08:50:03 2021
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Mon, 8 Feb 2021 17:50:03 +0900
Subject: [Freeswitch-users] Prevent conversion from rfc2833 to SIP INFO
Message-ID: <CABaNFCaU2L8XZAnhKcTuKUePfcy=dAh+XNicB9SH6+KdZC5D4g@mail.gmail.com>

Hi,

I have a gateway inviting FS with an SDP without payload telephone-event
(rfc2833).
This channelA is bridged to another channelB that does support rfc2833.
I noticed that in this case FS converts the rfc28333 digit from channelB to
SIP INFO to channelA.
The gateway doesn't support SIP INFO, but does support rfc2833 even if it
doesn't advertise it (at least it does support it for outgoing calls from
what was reported to me).
I tried to use this before the bridge:
       <action application="set" data="pass_rfc2833=true"/>
but behavior didn't change.
Is there anything else I can try?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210208/42cc392d/attachment.html>

From dujinfang at gmail.com  Mon Feb  8 11:40:47 2021
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 8 Feb 2021 19:40:47 +0800
Subject: [Freeswitch-users] Homebrew now on Mac M1
In-Reply-To: <1304141E-011C-4364-AABC-A98DDDF2573F@mgtech.com>
References: <B14A130F-6FD0-4E7F-9FE8-D28793D56650@mgtech.com>
 <CAEJMVkDnb718_2RxyLAa=RzGjENKeBJGkwJpfsFwNFQO01bPXQ@mail.gmail.com>
 <1304141E-011C-4364-AABC-A98DDDF2573F@mgtech.com>
Message-ID: <CAJs+-5qEb2Ye8daBoGrVzKVkvT8UfRDq782ewC-sXVUfRjeEUA@mail.gmail.com>

I'm running on a branch on top of master, works find on Big Sur 11.0.1

On Mon, Feb 8, 2021 at 2:52 PM mario_fs <mario_fs at mgtech.com> wrote:

> I don’t have an M1 so HB M1 support may have been in beta for you. M1
> official support was announced Feb 5.
>
> To add to the info below: I spent the weekend testing a clean install of
> the new homebrew 3.0 and freeswitch prerequisites on all macOS versions and
> found there is no way to build and run Freeswitch on any of the three
> supported macOS versions. Freeswitch segment faults on startup only on
> Mojave. I don’t plan on opening an issue since it could be homebrew
> related. Two other FS issues must be solved for Catalina and Big Sur (build
> issue is closed but the fix has not made it to master, other startup issue
> still open). I will keep the macOS status page
> <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Installation>
>  updated.
> Mario G
>
> On Feb 5, 2021, at 1:47 PM, Brian West <brian at freeswitch.com> wrote:
>
> Hasn't it been native for a while now?  Mine from two weeks ago is
>
> On Fri, Feb 5, 2021 at 1:07 PM mario_fs <mario_fs at mgtech.com> wrote:
>
>> An FYI for the people running Freeswitch on MacOS: homebrew announced
>> 3.0.0 today <https://brew.sh/2021/02/05/homebrew-3.0.0/> which runs
>> nativity on Mac M1 computers. I don’t plan to get one for a long time but I
>> would appreciate anyone who tries to build freeswitch on one to let us know
>> how it went. You’ll have to wait though, Freeswitch 1.10.5 GA and 1.10.6
>> Development does not build or run on Big Sur or Catalina since November.
>> Keep and eye on the macOS status page
>> <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Installation> to
>> check when you can build Freeswitch on macOS again.
>> Mario G
>> _________________________________________________________________________
>>
>> 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
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
> Email: brian at 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>
> _________________________________________________________________________
>
> 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



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210208/cfb16f06/attachment-0001.html>

From gidoramothra at gmail.com  Sun Feb  7 10:44:10 2021
From: gidoramothra at gmail.com (Stefan)
Date: Sun, 7 Feb 2021 11:44:10 +0100
Subject: [Freeswitch-users] mod_verto,
 verto communicator working only with chromium-based browsers
In-Reply-To: <CAC7ttFEei+sWgzjaieYL81f0=HRN02rqryfuZUNbyO+V1RFweQ@mail.gmail.com>
References: <20210204140305.GA1394@localhost.localdomain>
 <CAC7ttFEei+sWgzjaieYL81f0=HRN02rqryfuZUNbyO+V1RFweQ@mail.gmail.com>
Message-ID: <20210207104410.GA2037@localhost.localdomain>

OK, thx, the codec at least doesn't seem to be the problem sice, as you
can see from the log in my previous mail, firefox and freeswitch find a
match in G722, but it can still be the codec that makes the key
negotiation somehow failing, so I'll try as soon as I can.

@ Christopher: The problem seems to be that somehow the Crypto is not
negotiated with firefox, here the SDP from a failed firefox call to a
conference:

```
v=0
o=mozilla...THIS_IS_SDPARTA-85.0 5850506564681204176 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 D7:49:52:46:C7:03:BB:AA:00:51:A6:3A:87:3E:39:87:7B:11:BC:88:48:AB:69:52:1F:38:A7:AF:FE:B9:AB:A5
a=group:BUNDLE 0 1
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 49081 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 93.104.26.144
a=candidate:0 1 UDP 2122252543 192.168.178.102 49081 typ host
a=candidate:2 1 UDP 2122187007 192.168.178.101 38136 typ host
a=candidate:4 1 TCP 2105524479 192.168.178.102 9 typ host tcptype active
a=candidate:5 1 TCP 2105458943 192.168.178.101 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 192.168.178.102 52162 typ host
a=candidate:2 2 UDP 2122187006 192.168.178.101 38972 typ host
a=candidate:4 2 TCP 2105524478 192.168.178.102 9 typ host tcptype active
a=candidate:5 2 TCP 2105458942 192.168.178.101 9 typ host tcptype active
a=candidate:1 1 UDP 1686052863 93.104.26.144 49081 typ srflx raddr 192.168.178.102 rport 49081
a=candidate:3 1 UDP 1685987327 93.104.26.144 38136 typ srflx raddr 192.168.178.101 rport 38136
a=candidate:1 2 UDP 1686052862 93.104.26.144 52162 typ srflx raddr 192.168.178.102 rport 52162
a=candidate:3 2 UDP 1685987326 93.104.26.144 38972 typ srflx raddr 192.168.178.101 rport 38972
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1; stereo=1; sprop-stereo=1
a=fmtp:101 0-15
a=ice-pwd:cb982c1b6798dbb33f8b657af28ac2f2
a=ice-ufrag:1cc0dc33
a=mid:0
a=msid:{285f57af-e709-46c6-9c82-ad85bd0c13fe} {3fbe4a7a-d3a8-4c46-9902-c2375d4c3d25}
a=rtcp:52162 IN IP4 93.104.26.144
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:384906769 cname:{650dc0fe-0f89-4fa7-80c3-ad3a466120db}
m=video 48855 UDP/TLS/RTP/SAVPF 120 124 121 125
c=IN IP4 93.104.26.144
a=candidate:0 1 UDP 2122252543 192.168.178.102 48855 typ host
a=candidate:2 1 UDP 2122187007 192.168.178.101 48484 typ host
a=candidate:4 1 TCP 2105524479 192.168.178.102 9 typ host tcptype active
a=candidate:5 1 TCP 2105458943 192.168.178.101 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 192.168.178.102 33147 typ host
a=candidate:2 2 UDP 2122187006 192.168.178.101 60540 typ host
a=candidate:4 2 TCP 2105524478 192.168.178.102 9 typ host tcptype active
a=candidate:5 2 TCP 2105458942 192.168.178.101 9 typ host tcptype active
a=candidate:1 1 UDP 1686052863 93.104.26.144 48855 typ srflx raddr 192.168.178.102 rport 48855
a=candidate:3 1 UDP 1685987327 93.104.26.144 48484 typ srflx raddr 192.168.178.101 rport 48484
a=candidate:1 2 UDP 1686052862 93.104.26.144 33147 typ srflx raddr 192.168.178.102 rport 33147
a=candidate:3 2 UDP 1685987326 93.104.26.144 60540 typ srflx raddr 192.168.178.101 rport 60540
a=recvonly
a=end-of-candidates
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:4 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:5 urn:ietf:params:rtp-hdrext:toffset
a=extmap:6/recvonly http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:7 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=fmtp:120 max-fs=12288;max-fr=60
a=fmtp:124 apt=120
a=fmtp:121 max-fs=12288;max-fr=60
a=fmtp:125 apt=121
a=ice-pwd:cb982c1b6798dbb33f8b657af28ac2f2
a=ice-ufrag:1cc0dc33
a=mid:1
a=rtcp:33147 IN IP4 93.104.26.144
a=rtcp-fb:120 nack
a=rtcp-fb:120 nack pli
a=rtcp-fb:120 ccm fir
a=rtcp-fb:120 goog-remb
a=rtcp-fb:120 transport-cc
a=rtcp-fb:121 nack
a=rtcp-fb:121 nack pli
a=rtcp-fb:121 ccm fir
a=rtcp-fb:121 goog-remb
a=rtcp-fb:121 transport-cc
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:120 VP8/90000
a=rtpmap:124 rtx/90000
a=rtpmap:121 VP9/90000
a=rtpmap:125 rtx/90000
a=setup:actpass
a=ssrc:1996378200 cname:{650dc0fe-0f89-4fa7-80c3-ad3a466120db}
```

The call failes after that. Here the way to the failure, there is no
change in the SDP description in between:

```
Dialplan: verto.rtc/31000 parsing [team->enable_srtp] continue=true
Dialplan: verto.rtc/31000 Regex (PASS) [enable_srtp] destination_number(31000) =~ /.*/ break=on-false
Dialplan: verto.rtc/31000 Action set(rtp_secure_media=mandatory)
Dialplan: verto.rtc/31000 Action export(rtp_secure_media=mandatory)
Dialplan: verto.rtc/31000 parsing [team->teamusers_extension] continue=false
Dialplan: verto.rtc/31000 Regex (FAIL) [teamusers_extension] destination_number(31000) =~ /^(20[0-9][0-9][0-9])$/ break=on-false

...

EXECUTE [depth=0] verto.rtc/31000 set(rtp_secure_media=mandatory)
2021-02-07 09:33:56.077134 [DEBUG] mod_dptools.c:1672 SET verto.rtc/31000 [rtp_secure_media]=[mandatory]
EXECUTE [depth=0] verto.rtc/31000 export(rtp_secure_media=mandatory)
2021-02-07 09:33:56.077134 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [rtp_secure_media]=[mandatory]
EXECUTE [depth=0] verto.rtc/31000 set(codec_string=G722)
2021-02-07 09:33:56.077134 [DEBUG] mod_dptools.c:1672 SET verto.rtc/31000 [codec_string]=[G722]
EXECUTE [depth=0] verto.rtc/31000 answer()

...

2021-02-07 09:33:56.077134 [WARNING] switch_core_media.c:5667 Crypto not negotiated but required.
2021-02-07 09:33:56.077134 [DEBUG] switch_core_media.c:5911 verto.rtc/31000 Set 2833 dtmf send payload to 101 recv payload to 101
2021-02-07 09:33:56.077134 [DEBUG] switch_core_media.c:6239 No matches with FTMP, fallback to ignoring FMTP
2021-02-07 09:33:56.077134 [DEBUG] switch_core_media.c:6247 No matches with inherit_codec, fallback to ignoring PT
2021-02-07 09:33:56.077134 [WARNING] switch_core_media.c:6253 Crypto not negotiated but required.
2021-02-07 09:33:56.077134 [NOTICE] switch_channel.c:3908 Hangup verto.rtc/31000 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
```

In contrast, the SDP from a successful call with Microsoft edge:

```
v=0
o=- 1501624953028608797 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS UhrKcrZiGH1JculGLi7sf3qZxcpdua5AvC21
m=audio 55788 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 93.104.26.144
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1683339660 1 udp 2122260223 172.26.240.1 55787 typ host generation 0 network-id 1
a=candidate:617392483 1 udp 2122194687 192.168.1.112 55788 typ host generation 0 network-id 2 network-cost 10
a=candidate:2776936407 1 udp 1685987071 93.104.26.144 55788 typ srflx raddr 192.168.1.112 rport 55788 generation 0 network-id 2 network-cost 10
a=candidate:718685564 1 tcp 1518280447 172.26.240.1 9 typ host tcptype active generation 0 network-id 1
a=candidate:1783584147 1 tcp 1518214911 192.168.1.112 9 typ host tcptype active generation 0 network-id 2 network-cost 10
a=ice-ufrag:gbag
a=ice-pwd:SF5HWmdFQndyPQ2lRmuDYEOp
a=ice-options:trickle
a=fingerprint:sha-256 CE:30:A5:80:39:1D:88:B4:82:C8:CB:E3:8D:2D:31:57:35:6E:4A:B0:20:D5:7E:0A:0E:8E:CF:82:B0:DD:D0:0C
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1; stereo=1; sprop-stereo=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:1934233218 cname:rqfJ84fHxVg6HTKI
a=ssrc:1934233218 msid:UhrKcrZiGH1JculGLi7sf3qZxcpdua5AvC21 1896f7b1-ca03-46dd-ae93-3e74dc3baa0f
a=ssrc:1934233218 mslabel:UhrKcrZiGH1JculGLi7sf3qZxcpdua5AvC21
a=ssrc:1934233218 label:1896f7b1-ca03-46dd-ae93-3e74dc3baa0f
m=video 55790 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 121 127 120 125 107 108 109 124 119 123 118 114 115 116
c=IN IP4 93.104.26.144
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1683339660 1 udp 2122260223 172.26.240.1 55789 typ host generation 0 network-id 1
a=candidate:617392483 1 udp 2122194687 192.168.1.112 55790 typ host generation 0 network-id 2 network-cost 10
a=candidate:2776936407 1 udp 1685987071 93.104.26.144 55790 typ srflx raddr 192.168.1.112 rport 55790 generation 0 network-id 2 network-cost 10
a=candidate:718685564 1 tcp 1518280447 172.26.240.1 9 typ host tcptype active generation 0 network-id 1
a=candidate:1783584147 1 tcp 1518214911 192.168.1.112 9 typ host tcptype active generation 0 network-id 2 network-cost 10
a=ice-ufrag:gbag
a=ice-pwd:SF5HWmdFQndyPQ2lRmuDYEOp
a=ice-options:trickle
a=fingerprint:sha-256 CE:30:A5:80:39:1D:88:B4:82:C8:CB:E3:8D:2D:31:57:35:6E:4A:B0:20:D5:7E:0A:0E:8E:CF:82:B0:DD:D0:0C
a=setup:actpass
a=mid:video
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:13 urn:3gpp:video-orientation
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=sendrecv
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=fmtp:98 profile-id=0
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:100 VP9/90000
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=fmtp:100 profile-id=2
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:102 H264/90000
a=rtcp-fb:102 goog-remb
a=rtcp-fb:102 transport-cc
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 nack
a=rtcp-fb:102 nack pli
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:121 rtx/90000
a=fmtp:121 apt=102
a=rtpmap:127 H264/90000
a=rtcp-fb:127 goog-remb
a=rtcp-fb:127 transport-cc
a=rtcp-fb:127 ccm fir
a=rtcp-fb:127 nack
a=rtcp-fb:127 nack pli
a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f
a=rtpmap:120 rtx/90000
a=fmtp:120 apt=127
a=rtpmap:125 H264/90000
a=rtcp-fb:125 goog-remb
a=rtcp-fb:125 transport-cc
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:107 rtx/90000
a=fmtp:107 apt=125
a=rtpmap:108 H264/90000
a=rtcp-fb:108 goog-remb
a=rtcp-fb:108 transport-cc
a=rtcp-fb:108 ccm fir
a=rtcp-fb:108 nack
a=rtcp-fb:108 nack pli
a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f
a=rtpmap:109 rtx/90000
a=fmtp:109 apt=108
a=rtpmap:124 H264/90000
a=rtcp-fb:124 goog-remb
a=rtcp-fb:124 transport-cc
a=rtcp-fb:124 ccm fir
a=rtcp-fb:124 nack
a=rtcp-fb:124 nack pli
a=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d001f
a=rtpmap:119 rtx/90000
a=fmtp:119 apt=124
a=rtpmap:123 H264/90000
a=rtcp-fb:123 goog-remb
a=rtcp-fb:123 transport-cc
a=rtcp-fb:123 ccm fir
a=rtcp-fb:123 nack
a=rtcp-fb:123 nack pli
a=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=64001f
a=rtpmap:118 rtx/90000
a=fmtp:118 apt=123
a=rtpmap:114 red/90000
a=rtpmap:115 rtx/90000
a=fmtp:115 apt=114
a=rtpmap:116 ulpfec/90000
a=ssrc-group:FID 850477237 3405802114
a=ssrc:850477237 cname:rqfJ84fHxVg6HTKI
a=ssrc:850477237 msid:UhrKcrZiGH1JculGLi7sf3qZxcpdua5AvC21 557fc3d7-46b4-4f5b-81ff-7aa9f69022be
a=ssrc:850477237 mslabel:UhrKcrZiGH1JculGLi7sf3qZxcpdua5AvC21
a=ssrc:850477237 label:557fc3d7-46b4-4f5b-81ff-7aa9f69022be
a=ssrc:3405802114 cname:rqfJ84fHxVg6HTKI
a=ssrc:3405802114 msid:UhrKcrZiGH1JculGLi7sf3qZxcpdua5AvC21 557fc3d7-46b4-4f5b-81ff-7aa9f69022be
a=ssrc:3405802114 mslabel:UhrKcrZiGH1JculGLi7sf3qZxcpdua5AvC21
a=ssrc:3405802114 label:557fc3d7-46b4-4f5b-81ff-7aa9f69022be
```

No failure occurs, and without any change in the SDP description the
Crypto seems to work:

```
Dialplan: verto.rtc/31000 parsing [team->enable_srtp] continue=true
Dialplan: verto.rtc/31000 Regex (PASS) [enable_srtp] destination_number(31000) =~ /.*/ break=on-false
Dialplan: verto.rtc/31000 Action set(rtp_secure_media=mandatory)
Dialplan: verto.rtc/31000 Action export(rtp_secure_media=mandatory)
Dialplan: verto.rtc/31000 parsing [team->teamusers_extension] continue=false
Dialplan: verto.rtc/31000 Regex (FAIL) [teamusers_extension] destination_number(31000) =~ /^(20[0-9][0-9][0-9])$/ break=on-false

...

EXECUTE [depth=0] verto.rtc/31000 set(rtp_secure_media=mandatory)
2021-02-07 10:18:42.577108 [DEBUG] mod_dptools.c:1672 SET verto.rtc/31000 [rtp_secure_media]=[mandatory]
EXECUTE [depth=0] verto.rtc/31000 export(rtp_secure_media=mandatory)
2021-02-07 10:18:42.577108 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [rtp_secure_media]=[mandatory]
EXECUTE [depth=0] verto.rtc/31000 set(codec_string=G722)
2021-02-07 10:18:42.577108 [DEBUG] mod_dptools.c:1672 SET verto.rtc/31000 [codec_string]=[G722]
EXECUTE [depth=0] verto.rtc/31000 answer()

...

2021-02-07 10:18:42.577108 [DEBUG] switch_core_media.c:8885 Activating RTCP PORT 55788
2021-02-07 10:18:42.577108 [DEBUG] switch_rtp.c:4848 RTCP send rate is: 1000 and packet rate is: 20000 Remote Port: 55788
2021-02-07 10:18:42.577108 [INFO] switch_core_media.c:8896 Skipping RTCP ICE (Same as RTP)
2021-02-07 10:18:42.577108 [INFO] switch_rtp.c:3764 Activate RTP/RTCP audio DTLS client
2021-02-07 10:18:42.577108 [INFO] switch_rtp.c:3927 Changing audio DTLS state from OFF to HANDSHAKE
2021-02-07 10:18:42.577108 [DEBUG] switch_core_media.c:2554 Setting Jitterbuffer to 20ms (1 frames) (50 max frames)
2021-02-07 10:18:42.577108 [DEBUG] switch_core_media.c:8977 verto.rtc/31000 Set 2833 dtmf send payload to 126
2021-02-07 10:18:42.577108 [DEBUG] switch_core_media.c:8984 verto.rtc/31000 Set 2833 dtmf receive payload to 126
2021-02-07 10:18:42.577108 [DEBUG] switch_core_media.c:8645 Audio params are unchanged for verto.rtc/31000.
2021-02-07 10:18:42.577108 [DEBUG] mod_verto.c:2518 Local SDP verto.rtc/31000:
v=0
o=FreeSWITCH 1612661558 1612661559 IN IP4 46.4.114.220
s=FreeSWITCH
c=IN IP4 46.4.114.220
t=0 0
a=msid-semantic: WMS GnrX0xb28nF71JDGAWtDzqu8kW1fLUQi
m=audio 31564 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:31564 IN IP4 46.4.114.220
a=ice-ufrag:H2Q7ja6tXXyzHfHG
a=ice-pwd:dDuyAcSLpaObWmotf4qo5mHT
a=candidate:4541172879 1 udp 659136 46.4.114.220 31564 typ host generation 0
a=end-of-candidates
a=ssrc:1210358074 cname:875Yv7RukmscWVIm
a=ssrc:1210358074 msid:GnrX0xb28nF71JDGAWtDzqu8kW1fLUQi a0
a=ssrc:1210358074 mslabel:GnrX0xb28nF71JDGAWtDzqu8kW1fLUQi
a=ssrc:1210358074 label:GnrX0xb28nF71JDGAWtDzqu8kW1fLUQia0
m=video 0 UDP/TLS/RTP/SAVPF 19

2021-02-07 10:18:42.577108 [NOTICE] mod_dptools.c:1406 Channel [verto.rtc/31000] has been answered
2021-02-07 10:18:42.596994 [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})
```

Perhaps I'm missing a setting in mod_verto, to force firefox or all
clients to negotiate the srtp-key earlier? I'm not sure, because asd you
can see here:

https://github.com/gidmoth/freeswitch-container/blob/main/etc-freeswitch/dialplan/team.xml

I setup the required srtp in the dialplan, that should be after the
profiles (sofia/verto) configs. Is that the right way to do it?

thx for the help so far,

__
s.




On Fri, Feb 05, 2021 at 03:41:56PM +0100, Gregor Nanger wrote:
> Just for info. We are using verto and it works in Chrome, Edge and Firefox.
> Out of the box.
> 
> INCOMPATIBLE_DESTINATION could be because of codecs. Try to allow only
> pcma, pcmu in verto config. Just a thought to test it...
> 
> 
> 
> 
> On Fri, Feb 5, 2021, 15:34 Stefan <gidoramothra at gmail.com> wrote:
> 
> > Hello, I asked that before, but perhaps the question was somehow wrong
> > formulated or I didn't provide the right details. My Problem is the
> > following:
> >
> > I have a freeswitch 1.10 installation, you can find the container I
> > use, including a configuration similar to mine, here:
> >
> > https://github.com/gidmoth/freeswitch-container
> >
> > (The readme in the etc-freeswitch folder is the one provided by the
> > freeswitch-minimal config package in debian, but please look at the files
> > themselfes, because I took that only as a startingpoint).
> >
> > I have mod_verto enabled and try to connect clients with verto
> > communicator, build from the sources in the 1.10 release branch on
> > signalwire/freeswitch (https://github.com/signalwire/freeswitch).
> >
> > It works perfectly well with chomium-based browsers, but with firefox it
> > does not work.
> >
> > From the logs it is clear, that firefox seems not to start the
> > negotiation of a key for srtp, and then mod_verto refuses to execute the
> > call, because the dialplan requires srtp. Here the relevant log, a
> > failed attemt to call a conference with firefox/verto communicator as a
> > client:
> >
> > ```
> > 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"
> >         }
> > ```
> >
> > My question is: is there anything I can change to make it work with
> > firefox. Any hints welcome.
> >
> > __
> > s.
> >
> >
> > _________________________________________________________________________
> >
> > 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



From sr at flexmind.de  Tue Feb  2 11:21:45 2021
From: sr at flexmind.de (Stephan Reich)
Date: Tue, 2 Feb 2021 12:21:45 +0100
Subject: [Freeswitch-users] Huge Jitter after "Incorrect Timestamp"
In-Reply-To: <fb8f363b-b92f-cbdf-85cb-d34f5c63eb0d@flexmind.de>
References: <fb8f363b-b92f-cbdf-85cb-d34f5c63eb0d@flexmind.de>
Message-ID: <f4d60221-d374-4212-ab35-00e689972571@flexmind.de>

Hello,

we have problems with latency on some calls.

In a pcap I see this entry.
"Incorrect Timestamp"
 
After this paket we get several seconds delay in the call.

Are there parameters that fix  this problem ?


Stephan Reich
Leiter Technik

 

foncloud GmbH & Co KG
Hahlweg 2a
36093 Künzell

Tel: +49 661 968990 20 / 
Fax: +49 661 968990-99

 

Email: stephan.reich at foncloud.net
 Web: www.foncloud.net





P.S.: Wussten Sie schon? 

Unter https://www.foncloud.net/wissen  finden Sie zahlreiche Informationen und hilfreiche Artikel rund um unsere Produkte und Services.

 


Registergericht: Amtsgericht Fulda, Persönlich haftende Gesellschafterin der foncloud GmbH&Co.KG: Global Brain Network GmbHGeschäftsführer der Global Brain Network GmbH: Peter Krug Sitz der Gesellschaft: Künzell.

Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet.

This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden.



-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210202/8bbccb1d/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.png
Type: image/png
Size: 45411 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210202/8bbccb1d/attachment-0001.png>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: 0eb5f48c.e33a6fec.jpg
Type: image/jpeg
Size: 56822 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210202/8bbccb1d/attachment-0001.jpg>

From mario_fs at mgtech.com  Mon Feb  8 16:50:08 2021
From: mario_fs at mgtech.com (mario_fs)
Date: Mon, 8 Feb 2021 08:50:08 -0800
Subject: [Freeswitch-users] Homebrew now on Mac M1
In-Reply-To: <CAJs+-5qEb2Ye8daBoGrVzKVkvT8UfRDq782ewC-sXVUfRjeEUA@mail.gmail.com>
References: <B14A130F-6FD0-4E7F-9FE8-D28793D56650@mgtech.com>
 <CAEJMVkDnb718_2RxyLAa=RzGjENKeBJGkwJpfsFwNFQO01bPXQ@mail.gmail.com>
 <1304141E-011C-4364-AABC-A98DDDF2573F@mgtech.com>
 <CAJs+-5qEb2Ye8daBoGrVzKVkvT8UfRDq782ewC-sXVUfRjeEUA@mail.gmail.com>
Message-ID: <83529887-4F75-474E-BA75-E0562B4B7114@mgtech.com>

I don’t see how with  #961 <https://github.com/signalwire/freeswitch/issues/961> outstanding. Affects Big Sur and Catalina. No way to connect to FS or get CLI to work. Also, did you make sure /local was empty and rebuild everything from scratch?
Mario G

> On Feb 8, 2021, at 3:40 AM, Seven Du <dujinfang at gmail.com> wrote:
> 
> I'm running on a branch on top of master, works find on Big Sur 11.0.1
> 
> On Mon, Feb 8, 2021 at 2:52 PM mario_fs <mario_fs at mgtech.com <mailto:mario_fs at mgtech.com>> wrote:
> I don’t have an M1 so HB M1 support may have been in beta for you. M1 official support was announced Feb 5.
> 
> To add to the info below: I spent the weekend testing a clean install of the new homebrew 3.0 and freeswitch prerequisites on all macOS versions and found there is no way to build and run Freeswitch on any of the three supported macOS versions. Freeswitch segment faults on startup only on Mojave. I don’t plan on opening an issue since it could be homebrew related. Two other FS issues must be solved for Catalina and Big Sur (build issue is closed but the fix has not made it to master, other startup issue still open). I will keep the macOS status page <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Installation> updated.
> Mario G
> 
>> On Feb 5, 2021, at 1:47 PM, Brian West <brian at freeswitch.com <mailto:brian at freeswitch.com>> wrote:
>> 
>> Hasn't it been native for a while now?  Mine from two weeks ago is
>> 
>> On Fri, Feb 5, 2021 at 1:07 PM mario_fs <mario_fs at mgtech.com <mailto:mario_fs at mgtech.com>> wrote:
>> An FYI for the people running Freeswitch on MacOS: homebrew announced 3.0.0 today <https://brew.sh/2021/02/05/homebrew-3.0.0/> which runs nativity on Mac M1 computers. I don’t plan to get one for a long time but I would appreciate anyone who tries to build freeswitch on one to let us know how it went. You’ll have to wait though, Freeswitch 1.10.5 GA and 1.10.6 Development does not build or run on Big Sur or Catalina since November. Keep and eye on the macOS status page <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Installation> to check when you can build Freeswitch on macOS again.
>> Mario G
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <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 <https://signalwire.community/>
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>> https://cluecon.com <https://cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> -- 
>> 
>> Brian West | Co-founder and Developer
>> Need Commercial support? email sales at freeswitch.com <mailto:sales at freeswitch.com> 
>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>> Email: brian at freeswitch.com <mailto:brian at freeswitch.com>
>> Mobile: 918-424-9378
>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>  <https://www.facebook.com/freeswitch>  <https://twitter.com/freeswitch>
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <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 <https://signalwire.community/>
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>> https://cluecon.com <https://cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> https://freeswitch.com <https://freeswitch.com/>
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <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 <https://signalwire.community/>
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>
> 
> -- 
> About: http://about.me/dujinfang <http://about.me/dujinfang>
> Blog: http://www.dujinfang.com <http://www.dujinfang.com/>
> Proj:  http://www.freeswitch.org.cn <http://www.freeswitch.org.cn/>_________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210208/07f5feb1/attachment.html>

From mario_fs at mgtech.com  Mon Feb  8 16:58:18 2021
From: mario_fs at mgtech.com (mario_fs)
Date: Mon, 8 Feb 2021 08:58:18 -0800
Subject: [Freeswitch-users] Homebrew now on Mac M1
In-Reply-To: <CAJs+-5qEb2Ye8daBoGrVzKVkvT8UfRDq782ewC-sXVUfRjeEUA@mail.gmail.com>
References: <B14A130F-6FD0-4E7F-9FE8-D28793D56650@mgtech.com>
 <CAEJMVkDnb718_2RxyLAa=RzGjENKeBJGkwJpfsFwNFQO01bPXQ@mail.gmail.com>
 <1304141E-011C-4364-AABC-A98DDDF2573F@mgtech.com>
 <CAJs+-5qEb2Ye8daBoGrVzKVkvT8UfRDq782ewC-sXVUfRjeEUA@mail.gmail.com>
Message-ID: <73757833-66D9-4BE1-8596-042B15059A75@mgtech.com>

I don’t see how with  #961 <https://github.com/signalwire/freeswitch/issues/961> outstanding. Affects Big Sur and Catalina. No way to connect to FS or get CLI to work on a system built from scratch, Big Sur, CTL, etc. Did you make sure:

New install of Big Sur, not upgraded from something
Using CLT V12 - this is when #961 showed up
 /local was empty and built everything from scratch?
Using the default conf directory
Have IPV6 available on the computer, not sure of this is related but when I turned it off it made no difference.
Mario G

> On Feb 8, 2021, at 3:40 AM, Seven Du <dujinfang at gmail.com <mailto:dujinfang at gmail.com>> wrote:
> 
> I'm running on a branch on top of master, works find on Big Sur 11.0.1
> 
> On Mon, Feb 8, 2021 at 2:52 PM mario_fs <mario_fs at mgtech.com <mailto:mario_fs at mgtech.com>> wrote:
> I don’t have an M1 so HB M1 support may have been in beta for you. M1 official support was announced Feb 5.
> 
> To add to the info below: I spent the weekend testing a clean install of the new homebrew 3.0 and freeswitch prerequisites on all macOS versions and found there is no way to build and run Freeswitch on any of the three supported macOS versions. Freeswitch segment faults on startup only on Mojave. I don’t plan on opening an issue since it could be homebrew related. Two other FS issues must be solved for Catalina and Big Sur (build issue is closed but the fix has not made it to master, other startup issue still open). I will keep the macOS status page <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Installation> updated.
> Mario G
> 
>> On Feb 5, 2021, at 1:47 PM, Brian West <brian at freeswitch.com <mailto:brian at freeswitch.com>> wrote:
>> 
>> Hasn't it been native for a while now?  Mine from two weeks ago is
>> 
>> On Fri, Feb 5, 2021 at 1:07 PM mario_fs <mario_fs at mgtech.com <mailto:mario_fs at mgtech.com>> wrote:
>> An FYI for the people running Freeswitch on MacOS: homebrew announced 3.0.0 today <https://brew.sh/2021/02/05/homebrew-3.0.0/> which runs nativity on Mac M1 computers. I don’t plan to get one for a long time but I would appreciate anyone who tries to build freeswitch on one to let us know how it went. You’ll have to wait though, Freeswitch 1.10.5 GA and 1.10.6 Development does not build or run on Big Sur or Catalina since November. Keep and eye on the macOS status page <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Installation> to check when you can build Freeswitch on macOS again.
>> Mario G
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <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 <https://signalwire.community/>
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>> https://cluecon.com <https://cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> -- 
>> 
>> Brian West | Co-founder and Developer
>> Need Commercial support? email sales at freeswitch.com <mailto:sales at freeswitch.com> 
>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>> Email: brian at freeswitch.com <mailto:brian at freeswitch.com>
>> Mobile: 918-424-9378
>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>  <https://www.facebook.com/freeswitch>  <https://twitter.com/freeswitch>
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <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 <https://signalwire.community/>
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>> https://cluecon.com <https://cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> https://freeswitch.com <https://freeswitch.com/>
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <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 <https://signalwire.community/>
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>
> 
> -- 
> About: http://about.me/dujinfang <http://about.me/dujinfang>
> Blog: http://www.dujinfang.com <http://www.dujinfang.com/>
> Proj:  http://www.freeswitch.org.cn <http://www.freeswitch.org.cn/>_________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <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 <https://signalwire.community/>
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto: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


> On Feb 8, 2021, at 3:40 AM, Seven Du <dujinfang at gmail.com> wrote:
> 
> I'm running on a branch on top of master, works find on Big Sur 11.0.1
> 
> On Mon, Feb 8, 2021 at 2:52 PM mario_fs <mario_fs at mgtech.com <mailto:mario_fs at mgtech.com>> wrote:
> I don’t have an M1 so HB M1 support may have been in beta for you. M1 official support was announced Feb 5.
> 
> To add to the info below: I spent the weekend testing a clean install of the new homebrew 3.0 and freeswitch prerequisites on all macOS versions and found there is no way to build and run Freeswitch on any of the three supported macOS versions. Freeswitch segment faults on startup only on Mojave. I don’t plan on opening an issue since it could be homebrew related. Two other FS issues must be solved for Catalina and Big Sur (build issue is closed but the fix has not made it to master, other startup issue still open). I will keep the macOS status page <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Installation> updated.
> Mario G
> 
>> On Feb 5, 2021, at 1:47 PM, Brian West <brian at freeswitch.com <mailto:brian at freeswitch.com>> wrote:
>> 
>> Hasn't it been native for a while now?  Mine from two weeks ago is
>> 
>> On Fri, Feb 5, 2021 at 1:07 PM mario_fs <mario_fs at mgtech.com <mailto:mario_fs at mgtech.com>> wrote:
>> An FYI for the people running Freeswitch on MacOS: homebrew announced 3.0.0 today <https://brew.sh/2021/02/05/homebrew-3.0.0/> which runs nativity on Mac M1 computers. I don’t plan to get one for a long time but I would appreciate anyone who tries to build freeswitch on one to let us know how it went. You’ll have to wait though, Freeswitch 1.10.5 GA and 1.10.6 Development does not build or run on Big Sur or Catalina since November. Keep and eye on the macOS status page <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Installation> to check when you can build Freeswitch on macOS again.
>> Mario G
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <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 <https://signalwire.community/>
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>> https://cluecon.com <https://cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> -- 
>> 
>> Brian West | Co-founder and Developer
>> Need Commercial support? email sales at freeswitch.com <mailto:sales at freeswitch.com> 
>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>> Email: brian at freeswitch.com <mailto:brian at freeswitch.com>
>> Mobile: 918-424-9378
>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>  <https://www.facebook.com/freeswitch>  <https://twitter.com/freeswitch>
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <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 <https://signalwire.community/>
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>> https://cluecon.com <https://cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> https://freeswitch.com <https://freeswitch.com/>
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <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 <https://signalwire.community/>
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>
> 
> -- 
> About: http://about.me/dujinfang <http://about.me/dujinfang>
> Blog: http://www.dujinfang.com <http://www.dujinfang.com/>
> Proj:  http://www.freeswitch.org.cn <http://www.freeswitch.org.cn/>_________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com


-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210208/c4f8d1f6/attachment-0001.html>

From edson at inoutglobal.xyz  Mon Feb  8 20:51:54 2021
From: edson at inoutglobal.xyz (edson at inoutglobal.xyz)
Date: Mon, 8 Feb 2021 17:51:54 -0300
Subject: [Freeswitch-users] Freeswitch High Memory Consumption
Message-ID: <000a01d6fe5c$401eb950$c05c2bf0$@inoutglobal.xyz>

Dears,

 

Is it possible to find out what Freeswitch's  task leads operation system
run out of memory? 

It causes OS to start OOM Killer.

 

1-This is a backtrace Log output. 

https://pastebin.com/SLEPh4p2

 

 

2-dmesg output showing OOM killer in action.

 

[1385455.947689] Out of memory: Kill process 21698 (freeswitch) score 834 or
sacrifice child

[1385455.956673] Killed process 21698 (freeswitch) total-vm:12718944kB,
anon-rss:8202856kB, file-rss:0kB, shmem-rss:0kB

 

Regards,

 

Edson

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210208/261580db/attachment.html>

From mayamatakeshi at gmail.com  Tue Feb  9 00:17:48 2021
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 9 Feb 2021 09:17:48 +0900
Subject: [Freeswitch-users] Prevent conversion from rfc2833 to SIP INFO
In-Reply-To: <CABaNFCaU2L8XZAnhKcTuKUePfcy=dAh+XNicB9SH6+KdZC5D4g@mail.gmail.com>
References: <CABaNFCaU2L8XZAnhKcTuKUePfcy=dAh+XNicB9SH6+KdZC5D4g@mail.gmail.com>
Message-ID: <CABaNFCbNDvnu9jGDv_Gvg_p0mpkvNkzUydDr=V-h7vJjo8wxdw@mail.gmail.com>

On Mon, Feb 8, 2021 at 5:50 PM mayamatakeshi <mayamatakeshi at gmail.com>
wrote:

> Hi,
>
> I have a gateway inviting FS with an SDP without payload telephone-event
> (rfc2833).
> This channelA is bridged to another channelB that does support rfc2833.
> I noticed that in this case FS converts the rfc28333 digit from channelB
> to SIP INFO to channelA.
> The gateway doesn't support SIP INFO, but does support rfc2833 even if it
> doesn't advertise it (at least it does support it for outgoing calls from
> what was reported to me).
> I tried to use this before the bridge:
>        <action application="set" data="pass_rfc2833=true"/>
> but behavior didn't change.
> Is there anything else I can try?
>
>
Thinking again, since the telephone-event was not negotiated at channelA
side, of course FS cannot relay the RFC2833 packet.
So I am thinking in detect rfc2833 digits at channelB and send them as
inband tones at channelA using:
  send_dtmf <dtmf digits>[@<tone_duration>]
But I'm worried with the possibility of overlapping tones at channelA.
Does anyone know if send_dtmf has queueing behavior, meaning if I call it
multiple times in rapid succession, one operation will not interrupt the
previous one?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210209/c4619879/attachment.html>

From dujinfang at gmail.com  Tue Feb  9 01:01:55 2021
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 9 Feb 2021 09:01:55 +0800
Subject: [Freeswitch-users] Homebrew now on Mac M1
In-Reply-To: <83529887-4F75-474E-BA75-E0562B4B7114@mgtech.com>
References: <B14A130F-6FD0-4E7F-9FE8-D28793D56650@mgtech.com>
 <CAEJMVkDnb718_2RxyLAa=RzGjENKeBJGkwJpfsFwNFQO01bPXQ@mail.gmail.com>
 <1304141E-011C-4364-AABC-A98DDDF2573F@mgtech.com>
 <CAJs+-5qEb2Ye8daBoGrVzKVkvT8UfRDq782ewC-sXVUfRjeEUA@mail.gmail.com>
 <83529887-4F75-474E-BA75-E0562B4B7114@mgtech.com>
Message-ID: <CAJs+-5qtqxmZSW3GVm10rzxNniORYmLCFLb2fi1f3pEHj-odQA@mail.gmail.com>

ok, commented on #961

On Tue, Feb 9, 2021 at 1:21 AM mario_fs <mario_fs at mgtech.com> wrote:

> I don’t see how with  #961
> <https://github.com/signalwire/freeswitch/issues/961> outstanding.
> Affects Big Sur and Catalina. No way to connect to FS or get CLI to work.
> Also, did you make sure /local was empty and rebuild everything from
> scratch?
> Mario G
>
> On Feb 8, 2021, at 3:40 AM, Seven Du <dujinfang at gmail.com> wrote:
>
> I'm running on a branch on top of master, works find on Big Sur 11.0.1
>
> On Mon, Feb 8, 2021 at 2:52 PM mario_fs <mario_fs at mgtech.com> wrote:
>
>> I don’t have an M1 so HB M1 support may have been in beta for you. M1
>> official support was announced Feb 5.
>>
>> To add to the info below: I spent the weekend testing a clean install of
>> the new homebrew 3.0 and freeswitch prerequisites on all macOS versions and
>> found there is no way to build and run Freeswitch on any of the three
>> supported macOS versions. Freeswitch segment faults on startup only on
>> Mojave. I don’t plan on opening an issue since it could be homebrew
>> related. Two other FS issues must be solved for Catalina and Big Sur (build
>> issue is closed but the fix has not made it to master, other startup issue
>> still open). I will keep the macOS status page
>> <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Installation>
>>  updated.
>> Mario G
>>
>> On Feb 5, 2021, at 1:47 PM, Brian West <brian at freeswitch.com> wrote:
>>
>> Hasn't it been native for a while now?  Mine from two weeks ago is
>>
>> On Fri, Feb 5, 2021 at 1:07 PM mario_fs <mario_fs at mgtech.com> wrote:
>>
>>> An FYI for the people running Freeswitch on MacOS: homebrew announced
>>> 3.0.0 today <https://brew.sh/2021/02/05/homebrew-3.0.0/> which runs
>>> nativity on Mac M1 computers. I don’t plan to get one for a long time but I
>>> would appreciate anyone who tries to build freeswitch on one to let us know
>>> how it went. You’ll have to wait though, Freeswitch 1.10.5 GA and 1.10.6
>>> Development does not build or run on Big Sur or Catalina since November.
>>> Keep and eye on the macOS status page
>>> <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Installation> to
>>> check when you can build Freeswitch on macOS again.
>>> Mario G
>>> _________________________________________________________________________
>>>
>>> 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
>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>> Email: brian at 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>
>> _________________________________________________________________________
>>
>> 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
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210209/dee4a4de/attachment-0001.html>

From hamid2kviii at hotmail.com  Wed Feb 10 12:28:38 2021
From: hamid2kviii at hotmail.com (hrhashmi)
Date: Wed, 10 Feb 2021 05:28:38 -0700 (MST)
Subject: [Freeswitch-users] Cron-like execution in FS
In-Reply-To: <87f2f3b90903250940s4bf14381m8bb78d69f2f922fe@mail.gmail.com>
References: <4256bf830903250826ta96905dn4c9fee129351208f@mail.gmail.com>
 <87f2f3b90903250940s4bf14381m8bb78d69f2f922fe@mail.gmail.com>
Message-ID: <1612960118441-0.post@n2.nabble.com>

I need Cron-Like Execution in FS to do online Billing. Is there any method to
do that?

When the Call is answered, I need to execute an API after every specific
time of duration to bill consumed second and taking permission to keep the
call answered for the next specific duration of time. 

Regards
Hamid R. Hashmi



--
Sent from: http://freeswitch-users.2379917.n2.nabble.com/


From david.villasmil.work at gmail.com  Wed Feb 10 12:54:28 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 10 Feb 2021 12:54:28 +0000
Subject: [Freeswitch-users] Cron-like execution in FS
In-Reply-To: <1612960118441-0.post@n2.nabble.com>
References: <4256bf830903250826ta96905dn4c9fee129351208f@mail.gmail.com>
 <87f2f3b90903250940s4bf14381m8bb78d69f2f922fe@mail.gmail.com>
 <1612960118441-0.post@n2.nabble.com>
Message-ID: <CAFGRPVr0=CpE6Jo6XS5PD-hyJJjPFZcpwGQT4iQbf5AxHcemMQ@mail.gmail.com>

try

sched_api or sched_broadcast

https://freeswitch.org/confluence/display/FREESWITCH/mod_commands

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337


On Wed, Feb 10, 2021 at 12:28 PM hrhashmi <hamid2kviii at hotmail.com> wrote:

> I need Cron-Like Execution in FS to do online Billing. Is there any method
> to
> do that?
>
> When the Call is answered, I need to execute an API after every specific
> time of duration to bill consumed second and taking permission to keep the
> call answered for the next specific duration of time.
>
> Regards
> Hamid R. Hashmi
>
>
>
> --
> Sent from: http://freeswitch-users.2379917.n2.nabble.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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210210/c9a47480/attachment.html>

From jurijs.ivolga at gmail.com  Wed Feb 10 13:03:57 2021
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Wed, 10 Feb 2021 15:03:57 +0200
Subject: [Freeswitch-users] Cron-like execution in FS
In-Reply-To: <1612960118441-0.post@n2.nabble.com>
References: <4256bf830903250826ta96905dn4c9fee129351208f@mail.gmail.com>
 <87f2f3b90903250940s4bf14381m8bb78d69f2f922fe@mail.gmail.com>
 <1612960118441-0.post@n2.nabble.com>
Message-ID: <CAOrA2UaqR8d4Qx2ttVaH-BBxny2mxXG_xtCSQhZaDFkqDQ6Yng@mail.gmail.com>

Hi,

I do this in following way:

First I check with lua script if a call is allowed and there are credits
and I reserve credits for n seconds. Then I bridge with
"api_on_answer=luarun myscript.lua". This will execute the myscript.lua
when a call is answered and then inside that script every n seconds I try
to reserve more seconds, if not then the call is hung up, because of the
scheduler.

With kind regards,

Jurijs


On Wed, Feb 10, 2021 at 2:42 PM hrhashmi <hamid2kviii at hotmail.com> wrote:

> I need Cron-Like Execution in FS to do online Billing. Is there any method
> to
> do that?
>
> When the Call is answered, I need to execute an API after every specific
> time of duration to bill consumed second and taking permission to keep the
> call answered for the next specific duration of time.
>
> Regards
> Hamid R. Hashmi
>
>
>
> --
> Sent from: http://freeswitch-users.2379917.n2.nabble.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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210210/5deca208/attachment.html>

From david.villasmil.work at gmail.com  Wed Feb 10 13:44:39 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 10 Feb 2021 13:44:39 +0000
Subject: [Freeswitch-users] Cron-like execution in FS
In-Reply-To: <CAOrA2UaqR8d4Qx2ttVaH-BBxny2mxXG_xtCSQhZaDFkqDQ6Yng@mail.gmail.com>
References: <4256bf830903250826ta96905dn4c9fee129351208f@mail.gmail.com>
 <87f2f3b90903250940s4bf14381m8bb78d69f2f922fe@mail.gmail.com>
 <1612960118441-0.post@n2.nabble.com>
 <CAOrA2UaqR8d4Qx2ttVaH-BBxny2mxXG_xtCSQhZaDFkqDQ6Yng@mail.gmail.com>
Message-ID: <CAFGRPVqRcX6qtkTin6_54FWQvN4mcm+x3fb81ADgL68Jj8bexg@mail.gmail.com>

Not sure what you're asking

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337


On Wed, Feb 10, 2021 at 1:04 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
wrote:

> Hi,
>
> I do this in following way:
>
> First I check with lua script if a call is allowed and there are credits
> and I reserve credits for n seconds. Then I bridge with
> "api_on_answer=luarun myscript.lua". This will execute the myscript.lua
> when a call is answered and then inside that script every n seconds I try
> to reserve more seconds, if not then the call is hung up, because of the
> scheduler.
>
> With kind regards,
>
> Jurijs
>
>
> On Wed, Feb 10, 2021 at 2:42 PM hrhashmi <hamid2kviii at hotmail.com> wrote:
>
>> I need Cron-Like Execution in FS to do online Billing. Is there any
>> method to
>> do that?
>>
>> When the Call is answered, I need to execute an API after every specific
>> time of duration to bill consumed second and taking permission to keep the
>> call answered for the next specific duration of time.
>>
>> Regards
>> Hamid R. Hashmi
>>
>>
>>
>> --
>> Sent from: http://freeswitch-users.2379917.n2.nabble.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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210210/9eb24b9d/attachment-0001.html>

From davidswalkabout at gmail.com  Fri Feb 12 00:48:09 2021
From: davidswalkabout at gmail.com (David P)
Date: Fri, 12 Feb 2021 13:48:09 +1300
Subject: [Freeswitch-users] Forward conference via HLS instead of RTMP?
Message-ID: <CAE23truyJwKQ8oKwK1MKiZix2zF2gbdM=EfM2UMKnbqxGUJzoA@mail.gmail.com>

We currently forward conferences via RTMP but we'd like to remain at the
inbound sampling rate of 48kHz rather than resample to 44.1kHz as RTMP
seems to require.

It looks like HLS supports 48kHz and we wonder if FS supports streaming-out
on it. I found a tweet about ClueCon 2018 in which it seems FS might
support this: "Streaming from #WebRTC to RTMP & HLS with @Freeswitch by
@shitlucapsays at @cluecon #ClueCon2018"
https://twitter.com/chadwallacehart/status/1021868541291778048

Is this supported in FS v1.10? I can't find any confluence or other
documentation.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210212/5f6ed9ea/attachment.html>

From brian at freeswitch.com  Fri Feb 12 01:21:01 2021
From: brian at freeswitch.com (Brian West)
Date: Thu, 11 Feb 2021 19:21:01 -0600
Subject: [Freeswitch-users] Forward conference via HLS instead of RTMP?
In-Reply-To: <CAE23truyJwKQ8oKwK1MKiZix2zF2gbdM=EfM2UMKnbqxGUJzoA@mail.gmail.com>
References: <CAE23truyJwKQ8oKwK1MKiZix2zF2gbdM=EfM2UMKnbqxGUJzoA@mail.gmail.com>
Message-ID: <CAEJMVkBxma8D0kZN04EBMCfauuym5w5bVH8pgDsYq64PKMoy8A@mail.gmail.com>

Not currently possible.

On Thu, Feb 11, 2021 at 7:16 PM David P <davidswalkabout at gmail.com> wrote:

> We currently forward conferences via RTMP but we'd like to remain at the
> inbound sampling rate of 48kHz rather than resample to 44.1kHz as RTMP
> seems to require.
>
> It looks like HLS supports 48kHz and we wonder if FS supports
> streaming-out on it. I found a tweet about ClueCon 2018 in which it seems
> FS might support this: "Streaming from #WebRTC to RTMP & HLS with
> @Freeswitch by @shitlucapsays at @cluecon #ClueCon2018"
> https://twitter.com/chadwallacehart/status/1021868541291778048
>
> Is this supported in FS v1.10? I can't find any confluence or other
> documentation.
> _________________________________________________________________________
>
> 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
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>

Email: brian at 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>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210211/4925e386/attachment.html>

From davidswalkabout at gmail.com  Fri Feb 12 22:14:00 2021
From: davidswalkabout at gmail.com (David P)
Date: Sat, 13 Feb 2021 11:14:00 +1300
Subject: [Freeswitch-users] Chrome is deprecating plan-b SDP semantics. Will
	verto be updated?
Message-ID: <CAE23truyfF_0j4UFMkLWexngxbkK6UKRkrjwi-XMxB0dO2PdTA@mail.gmail.com>

Google announced that support in Chrome for "plan-b" SDP semantics will be
removed this August:
https://groups.google.com/g/discuss-webrtc/c/Zrsn2hi8FV0/m/KIbn0EZPBQAJ?authuser=0
As I understand it, this means that FS' verto js library will stop working
unless it's updated to use "unified-plan" semantics.

I searched github and google for "verto" and "jquery.verto.js" to see if
I'd missed an announcement that this change was in-progress or done, but I
don't get a good match at either. Has it been moved out of
https://github.com/signalwire/freeswitch to a community repo? Does it have
a maintainer?

I can't become the maintainer, but this is a serious problem for us if we
don't have a solution by August.

Btw, Brian, thank you for your answer about HLS.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210213/be3131b3/attachment.html>

From carsten at ng-voice.com  Thu Feb 11 10:41:53 2021
From: carsten at ng-voice.com (Carsten Bock)
Date: Thu, 11 Feb 2021 11:41:53 +0100
Subject: [Freeswitch-users] Reply to Via
Message-ID: <CAOCjumEBiQM-UJiNKv7X73HH_PuvHzF79TpsvL+=cdezrkcd0g@mail.gmail.com>

Hi,

I have a quick NAT related question.
Is there a way to disable NAT handling completely?

FreeSwitch always replies to the IP/Port were it received the message from
and not to the host/port in Via....

recv 1166 bytes from udp/[172.17.0.1]:36835 at 10:20:56.652918:
------------------------------------------------------------------------
INVITE sip:ann-early at 167.99.136.226:5080 SIP/2.0
Record-Route: <sip:167.99.136.226:5070;lr>
Via: SIP/2.0/UDP 167.99.136.226:5070;branch=z9hG4bKe305.92946d84.0
Via: SIP/2.0/UDP 192.168.178.52:37350
;received=93.221.23.15;branch=z9hG4bK-524287-1---936bd2b8103acd0c;rport=37350
[...]

send 366 bytes to udp/[172.17.0.1]:5070 at 10:20:56.659313:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 167.99.136.226:5070
;branch=z9hG4bKe305.92946d84.0;received=172.17.0.1
From: <sip:b2bua at 167.99.136.226:5070;transport=UDP>;tag=7793d52b
To: <sip:3 at 167.99.136.226:5070>
Call-ID: akjc99KAbZfHxXHSea_iTQ..
CSeq: 1 INVITE
User-Agent:
FreeSWITCH-mod_sofia/1.10.5-release+git-20201002T121516Z~a1bfb14c04~64bit
Content-Length: 0

Is there a way, to make FreeSwitch simply reply to the Host/Port in the
Via-Header?

My Profile is rather simple and I tried it with setting the local network
ACS and NDLB-force-rport:

    <profile name="local">
   <settings>
   <param name="local-network-acl" value="none"/>
<param name="NDLB-force-rport" value="disabled"/>
   <param name="context" value="local"/>
        <param name="sip-ip" value="$${local_address}"/>
        <param name="ext-sip-ip" value="autonat:167.99.136.226"/>
        <param name="sip-port" value="5080"/>
        <param name="rtp-ip" value="$${local_address}"/>
        <param name="ext-rtp-ip" value="autonat:167.99.136.226"/>
        <param name="auth-calls" value="false"/>
      </settings>
    </profile>

Changing "NDLB-force-rport" to false did not have any impact.

Thanks,
Carsten



--
Carsten Bock I CTO & Founder

ng-voice GmbH

Trostbrücke 1 I 20457 Hamburg I Germany
T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210211/e8003fbe/attachment.html>

From modlibor at gmail.com  Tue Feb  9 14:29:50 2021
From: modlibor at gmail.com (=?utf-8?Q?Grzegorz_Orze=C5=82?=)
Date: Tue, 9 Feb 2021 15:29:50 +0100
Subject: [Freeswitch-users] Fax to extension of B number
Message-ID: <DE6324F4-55AE-42A1-8C61-B404549006F4@gmail.com>

Hi folks,

I have a FreeSwitch serving multiple services. One of them is fax2mail/mail2fax. Logic is implemented as a set of lua scripts and it works fine. I’d like extend mail2fax direction. Currently it (lua script) picks number B, tiff file and sends it directly to B using command originate as follows:

originate {<_bunch_of_headers_>} sofia/service/+4812345678 at some.domain;user=phone &txfax(<_tiff_file_path_>) XML <_some_profile_>

What I’d like to do is to add possibility to dial some other extension number. Lest’s say fax is available on +4812345678 but when the call is aswered there needs to be dialed number 123.

I’ve already reviewed all FreeSwitch books on safaribooks,  googled but I haven't found any example… :-/ 

Maybe it is enough to simply add it to the call url - I mean something like:

originate {<_bunch_of_headers_>} sofia/service/+4812345678,,123 at some.domain;user=phone &txfax(<_tiff_file_path_>) XML <_some_profile_>

Can you give me some hint how to do that?

Thanks in advance,
Grzegorz
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210209/d4112bd4/attachment-0001.html>

From david.villasmil.work at gmail.com  Tue Feb  9 19:12:48 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 9 Feb 2021 19:12:48 +0000
Subject: [Freeswitch-users] G.722 to PCMA choppy audio
Message-ID: <CAFGRPVqWSFa_Qse2BbdQjK5cbDGFYONE168mk22qB+p+4d+nOw@mail.gmail.com>

Hello all,

This is driving me nuts.

I'm originating a call to a voicemail, wich i then use for ASR with a
lumenvox box.
When the outbound leg is PCMA/PCMU, everything's fine, but if transcoding
is needed, the audio being sent to lumenvox is choppy. unimrcp's config
says to send PCMA.

I'm attaching a raw audio file of the choppy audio.

fs is 1.10

Any ideas?

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210209/6f92672e/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: Saved-RTP-Audio.raw
Type: application/octet-stream
Size: 480960 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210209/6f92672e/attachment-0001.obj>

From krishdinesh at yahoo.com  Thu Feb 11 05:59:51 2021
From: krishdinesh at yahoo.com (Dinesh Krishnamurthy)
Date: Thu, 11 Feb 2021 05:59:51 +0000 (UTC)
Subject: Installing FS on RHEL 7
References: <767941273.416266.1613023191302.ref@mail.yahoo.com>
Message-ID: <767941273.416266.1613023191302@mail.yahoo.com>

Hi,
I am trying to install FS on RHEL 7.x but i am facing issues while setting up. I am following the documentation provided on the confluence portal. 
https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7


Snapshot of few error messages, did not want to paste the whole error snapshot.  
Error: Package: freeswitch-lang-pt-1.10.5.release.8-1.el7.x86_64 (freeswitch)           Requires: libspeexdsp.so.1()(64bit)Error: Package: freeswitch-application-redis-1.10.5.release.8-1.el7.x86_64 (freeswitch)           Requires: libspeexdsp.so.1()(64bit)Error: Package: libks-1.2.0-13.el7.centos.x86_64 (freeswitch)           Requires: libuuid-develError: Package: freeswitch-application-lcr-1.10.5.release.8-1.el7.x86_64 (freeswitch)           Requires: libpq.so.5()(64bit)Error: Package: freeswitch-1.10.5.release.8-1.el7.x86_64 (freeswitch)           Requires: liboggError: Package: freeswitch-application-conference-1.10.5.release.8-1.el7.x86_64 (freeswitch)           Requires: libspeex.so.1()(64bit)Error: Package: freeswitch-application-fsv-1.10.5.release.8-1.el7.x86_64 (freeswitch)           Requires: libspeex.so.1()(64bit)Error: Package: freeswitch-application-httapi-1.10.5.release.8-1.el7.x86_64 (freeswitch)           Requires: libpq.so.5()(64bit)Error: Package: freeswitch-lang-de-1.10.5.release.8-1.el7.x86_64 (freeswitch)           Requires: libspeex.so.1()(64bit)

Any help would be appreciated. 
Thank you,DK
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210211/470f033a/attachment.html>

From krishdinesh at yahoo.com  Thu Feb 11 13:45:00 2021
From: krishdinesh at yahoo.com (Dinesh Krishnamurthy)
Date: Thu, 11 Feb 2021 13:45:00 +0000 (UTC)
Subject: Official freeswitch docker image
References: <1515018331.476479.1613051100449.ref@mail.yahoo.com>
Message-ID: <1515018331.476479.1613051100449@mail.yahoo.com>

Hi,
Can you someone help to understand which is the official Freeswitch image on the docker hub ?
Thank you,DK
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210211/525fb452/attachment.html>

From maximilian.dorn at pinguin.ag  Thu Feb 11 12:33:04 2021
From: maximilian.dorn at pinguin.ag (Maximilian Leonhard Dorn)
Date: Thu, 11 Feb 2021 12:33:04 +0000
Subject: [Freeswitch-users] fs_nat=yes on webrtc connection
References: <cc372afe-fcd4-4eae-a7c7-ab0505bbfa4e.c13571e5-0035-44ec-b541-02bbd1b5bf07.8dacbe28-5807-4d31-8734-0b2450011813@emailsignatures365.codetwo.com>
Message-ID: <DB6PR0802MB25976EA84BD35D110CABB78D9A8C9@DB6PR0802MB2597.eurprd08.prod.outlook.com>

Hi everyone,

I am currently working with sip.js and freeswitch which is currently connecting to our freeswitch via the our internal network.
Looking at the registration I am seeing that the contact show fs_nat= yes. Here the registration from the server.

Call-ID:        6ctjv10kv92d3eft9i6lc7
User:           01234678 at example.com<mailto:01234678 at example.com>
Contact:        "01234678" <sip:01234678 at example.com;transport=ws;fs_nat=yes;fs_path=sip%3A01234678%4010.128.130.129%3A51786%3Btransport%3Dwss>
Agent:          SIP.js/0.15.10
Status:         Registered(WSS-NAT)(unknown) EXP(2021-02-11 13:10:07) EXPSECS(658)
Ping-Status:    Reachable
Ping-Time:      0.00
Host:           example
IP:             10.128.130.129
Port:           51786
Auth-User:      unknown
Auth-Realm:     example.com
MWI-Account:    01234678 at example.com<mailto:01234678 at example.com>

Nat detection is disabled for the freeswitch and a normal snom phone registering to the server does not show the fs_nat=yes
Does freeswitch always detect nat for webrtc connections? Or is there possibly something wrong in my configuration?

Kind Regards,
Max

------------------------------------------------------------
Phone:    +491752242485
Fax:
Mail:       maximilian.dorn at pinguin.ag

------------------------------------------------------------
Phone:  +49.30.30 36 63 366
Fax:       +49.30.30 36 63 367
E-Mail: info at pinguin.ag<mailto:info at pinguin.ag>

www.pinguin.ag<http://www.pinguin.ag>

------------------------------------------------------------
[cid:pinguin.ag_f1f2f94b-4088-4a1c-afba-e73816a444e1.png]

Pinguin AG
Hauptstra?e 8 | 10827 Berlin | Germany

USt-ID: DE240701807

Sitz der Gesellschaft: Berlin-Sch?neberg, AG Berlin-Charlottenburg, HRB 101380 B
Vorstand: Roland Becker, Aufsichtsratsvorsitzender: Clemens C. Vogelsberg
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210211/09049adf/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: pinguin.ag_f1f2f94b-4088-4a1c-afba-e73816a444e1.png
Type: image/png
Size: 5020 bytes
Desc: pinguin.ag_f1f2f94b-4088-4a1c-afba-e73816a444e1.png
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210211/09049adf/attachment-0001.png>

From tech at amlotus.edu  Thu Feb 11 12:59:24 2021
From: tech at amlotus.edu (tech amlotus)
Date: Thu, 11 Feb 2021 12:59:24 +0000
Subject: [Freeswitch-users] list
Message-ID: <CAMT64Ux5o7PCPy3Janu7iNPWAkYRmxFPiqrqBRmONq2vQJ0icA@mail.gmail.com>

-- 
*Amlotus Technology Department*


Amlotus LLC
500 8th Ave #909
New York, NY 10018
(212)-912-0100
www.amlotus.edu
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210211/fa9d5b12/attachment.html>

From david.villasmil.work at gmail.com  Tue Feb 16 10:14:43 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 16 Feb 2021 10:14:43 +0000
Subject: [Freeswitch-users] Reply to Via
In-Reply-To: <CAOCjumEBiQM-UJiNKv7X73HH_PuvHzF79TpsvL+=cdezrkcd0g@mail.gmail.com>
References: <CAOCjumEBiQM-UJiNKv7X73HH_PuvHzF79TpsvL+=cdezrkcd0g@mail.gmail.com>
Message-ID: <CAFGRPVoRvu7RthXEUs7TLSA9NE8g3+k=uQjKRQPh0Z1zAaWA+Q@mail.gmail.com>

-nonat ?

On Tue, 16 Feb 2021 at 10:00, Carsten Bock <carsten at ng-voice.com> wrote:

> Hi,
>
> I have a quick NAT related question.
> Is there a way to disable NAT handling completely?
>
> FreeSwitch always replies to the IP/Port were it received the message from
> and not to the host/port in Via....
>
> recv 1166 bytes from udp/[172.17.0.1]:36835 at 10:20:56.652918:
> ------------------------------------------------------------------------
> INVITE sip:ann-early at 167.99.136.226:5080 SIP/2.0
> Record-Route: <sip:167.99.136.226:5070;lr>
> Via: SIP/2.0/UDP 167.99.136.226:5070;branch=z9hG4bKe305.92946d84.0
> Via: SIP/2.0/UDP 192.168.178.52:37350
> ;received=93.221.23.15;branch=z9hG4bK-524287-1---936bd2b8103acd0c;rport=37350
> [...]
>
> send 366 bytes to udp/[172.17.0.1]:5070 at 10:20:56.659313:
> ------------------------------------------------------------------------
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 167.99.136.226:5070
> ;branch=z9hG4bKe305.92946d84.0;received=172.17.0.1
> From: <sip:b2bua at 167.99.136.226:5070;transport=UDP>;tag=7793d52b
> To: <sip:3 at 167.99.136.226:5070>
> Call-ID: akjc99KAbZfHxXHSea_iTQ..
> CSeq: 1 INVITE
> User-Agent:
> FreeSWITCH-mod_sofia/1.10.5-release+git-20201002T121516Z~a1bfb14c04~64bit
> Content-Length: 0
>
> Is there a way, to make FreeSwitch simply reply to the Host/Port in the
> Via-Header?
>
> My Profile is rather simple and I tried it with setting the local network
> ACS and NDLB-force-rport:
>
>     <profile name="local">
>    <settings>
>    <param name="local-network-acl" value="none"/>
> <param name="NDLB-force-rport" value="disabled"/>
>    <param name="context" value="local"/>
>         <param name="sip-ip" value="$${local_address}"/>
>         <param name="ext-sip-ip" value="autonat:167.99.136.226"/>
>         <param name="sip-port" value="5080"/>
>         <param name="rtp-ip" value="$${local_address}"/>
>         <param name="ext-rtp-ip" value="autonat:167.99.136.226"/>
>         <param name="auth-calls" value="false"/>
>       </settings>
>     </profile>
>
> Changing "NDLB-force-rport" to false did not have any impact.
>
> Thanks,
> Carsten
>
>
>
> --
> Carsten Bock I CTO & Founder
>
> ng-voice GmbH
>
> Trostbrücke 1
> <https://www.google.com/maps/search/Trostbr%C3%BCcke+1?entry=gmail&source=g>
> I 20457 Hamburg I Germany
> T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com
>
> Registry Office at Local Court Hamburg, HRB 120189
> Managing Directors: Dr. David Bachmann, Carsten Bock
>
>
>
> _________________________________________________________________________
>
> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210216/e86b9283/attachment.html>

From amit.k.chaudhuri at gmail.com  Tue Feb 16 11:42:56 2021
From: amit.k.chaudhuri at gmail.com (Amit Chaudhuri)
Date: Tue, 16 Feb 2021 11:42:56 +0000
Subject: [Freeswitch-users] Installing FS on RHEL 7
In-Reply-To: <mailman.1237.1613469633.1264.freeswitch-users@lists.freeswitch.org>
References: <767941273.416266.1613023191302.ref@mail.yahoo.com>
 <mailman.1237.1613469633.1264.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAH=VbNeo-Wp0NvLrqfmvv-m_8Cf2V2uqMj2dKc+ezGcJv2Okyw@mail.gmail.com>

Looks like you are missing a various packages that fs requires. If you
scroll down the page you provided a link for, there are sections which
appear to detail what needs to be installed first. Example
libuuid-devel.
A

On Tue, 16 Feb 2021 at 11:12, Dinesh Krishnamurthy via
FreeSWITCH-users <freeswitch-users at lists.freeswitch.org> wrote:
>
>
>
>
> ---------- Forwarded message ----------
> From: Dinesh Krishnamurthy <krishdinesh at yahoo.com>
> To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Thu, 11 Feb 2021 05:59:51 +0000 (UTC)
> Subject: Installing FS on RHEL 7
> Hi,
>
> I am trying to install FS on RHEL 7.x but i am facing issues while setting up. I am following the documentation provided on the confluence portal.
>
> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7
>
>
> Snapshot of few error messages, did not want to paste the whole error snapshot.
>
> Error: Package: freeswitch-lang-pt-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>            Requires: libspeexdsp.so.1()(64bit)
> Error: Package: freeswitch-application-redis-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>            Requires: libspeexdsp.so.1()(64bit)
> Error: Package: libks-1.2.0-13.el7.centos.x86_64 (freeswitch)
>            Requires: libuuid-devel
> Error: Package: freeswitch-application-lcr-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>            Requires: libpq.so.5()(64bit)
> Error: Package: freeswitch-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>            Requires: libogg
> Error: Package: freeswitch-application-conference-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>            Requires: libspeex.so.1()(64bit)
> Error: Package: freeswitch-application-fsv-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>            Requires: libspeex.so.1()(64bit)
> Error: Package: freeswitch-application-httapi-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>            Requires: libpq.so.5()(64bit)
> Error: Package: freeswitch-lang-de-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>            Requires: libspeex.so.1()(64bit)
>
>
> Any help would be appreciated.
>
> Thank you,
> DK
>
>
>
> ---------- Forwarded message ----------
> From: Dinesh Krishnamurthy via FreeSWITCH-users <freeswitch-users at lists.freeswitch.org>
> To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Tue, 16 Feb 2021 03:12:25 -0800 (PST)
> Subject: [Freeswitch-users] Installing FS on RHEL 7
> _________________________________________________________________________
>
> 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 yehavi.bourvine at gmail.com  Tue Feb 16 12:37:35 2021
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 16 Feb 2021 14:37:35 +0200
Subject: [Freeswitch-users] Installing FS on RHEL 7
In-Reply-To: <mailman.1237.1613469633.1264.freeswitch-users@lists.freeswitch.org>
References: <767941273.416266.1613023191302.ref@mail.yahoo.com>
 <mailman.1237.1613469633.1264.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAAtWKeFAQDcB0KSTFayPMRkUVx-5mQa9UV2zJRS932EdYtMvBA@mail.gmail.com>

I used to run Freeswitch on Centos. Version 1.10.3 is the last one that I
managed to compile, as the newer ones need libraries that are not available
on Centos. I moved to Ubuntu...

           __Yehavi:

‫בתאריך יום ג׳, 16 בפבר׳ 2021 ב-12:00 מאת ‪Dinesh Krishnamurthy via
FreeSWITCH-users‬‏ <‪freeswitch-users at lists.freeswitch.org‬‏>:‬

>
>
>
> ---------- Forwarded message ----------
> From: Dinesh Krishnamurthy <krishdinesh at yahoo.com>
> To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Thu, 11 Feb 2021 05:59:51 +0000 (UTC)
> Subject: Installing FS on RHEL 7
> Hi,
>
> I am trying to install FS on RHEL 7.x but i am facing issues while setting
> up. I am following the documentation provided on the confluence portal.
>
> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7
>
>
> Snapshot of few error messages, did not want to paste the whole error
> snapshot.
>
> Error: Package: freeswitch-lang-pt-1.10.5.release.8-1.el7.x86_64
> (freeswitch)
>            Requires: libspeexdsp.so.1()(64bit)
> Error: Package: freeswitch-application-redis-1.10.5.release.8-1.el7.x86_64
> (freeswitch)
>            Requires: libspeexdsp.so.1()(64bit)
> Error: Package: libks-1.2.0-13.el7.centos.x86_64 (freeswitch)
>            Requires: libuuid-devel
> Error: Package: freeswitch-application-lcr-1.10.5.release.8-1.el7.x86_64
> (freeswitch)
>            Requires: libpq.so.5()(64bit)
> Error: Package: freeswitch-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>            Requires: libogg
> Error: Package:
> freeswitch-application-conference-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>            Requires: libspeex.so.1()(64bit)
> Error: Package: freeswitch-application-fsv-1.10.5.release.8-1.el7.x86_64
> (freeswitch)
>            Requires: libspeex.so.1()(64bit)
> Error: Package:
> freeswitch-application-httapi-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>            Requires: libpq.so.5()(64bit)
> Error: Package: freeswitch-lang-de-1.10.5.release.8-1.el7.x86_64
> (freeswitch)
>            Requires: libspeex.so.1()(64bit)
>
>
> Any help would be appreciated.
>
> Thank you,
> DK
>
>
>
> ---------- Forwarded message ----------
> From: Dinesh Krishnamurthy via FreeSWITCH-users <
> freeswitch-users at lists.freeswitch.org>
> To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Tue, 16 Feb 2021 02:00:46 -0800 (PST)
> Subject: [Freeswitch-users] Installing FS on RHEL 7
> _________________________________________________________________________
>
> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210216/acdc0bf8/attachment.html>

From s.safarov at gmail.com  Tue Feb 16 15:56:45 2021
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 16 Feb 2021 18:56:45 +0300
Subject: [Freeswitch-users] Official freeswitch docker image
In-Reply-To: <mailman.1238.1613469633.1264.freeswitch-users@lists.freeswitch.org>
References: <1515018331.476479.1613051100449.ref@mail.yahoo.com>
 <mailman.1238.1613469633.1264.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAHtxdDfVJzxivXEf-35ayUVNHLPzO9TabdCDqU5TcCibN6FF5Q@mail.gmail.com>

I think no official docker image.
I in repo present docker image that may be build official image.
But image not really build


On Tue, Feb 16, 2021, 1:51 PM Dinesh Krishnamurthy via FreeSWITCH-users <
freeswitch-users at lists.freeswitch.org> wrote:

>
>
>
> ---------- Forwarded message ----------
> From: Dinesh Krishnamurthy <krishdinesh at yahoo.com>
> To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Thu, 11 Feb 2021 13:45:00 +0000 (UTC)
> Subject: Official freeswitch docker image
> Hi,
>
> Can you someone help to understand which is the official Freeswitch image
> on the docker hub ?
>
> Thank you,
> DK
>
>
>
> ---------- Forwarded message ----------
> From: Dinesh Krishnamurthy via FreeSWITCH-users <
> freeswitch-users at lists.freeswitch.org>
> To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Tue, 16 Feb 2021 02:51:19 -0800 (PST)
> Subject: [Freeswitch-users] Official freeswitch docker image
> _________________________________________________________________________
>
> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210216/f685fe4f/attachment.html>

From krice at freeswitch.org  Tue Feb 16 17:13:08 2021
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 16 Feb 2021 11:13:08 -0600
Subject: [Freeswitch-users] Reply to Via
In-Reply-To: <CAFGRPVoRvu7RthXEUs7TLSA9NE8g3+k=uQjKRQPh0Z1zAaWA+Q@mail.gmail.com>
References: <CAFGRPVoRvu7RthXEUs7TLSA9NE8g3+k=uQjKRQPh0Z1zAaWA+Q@mail.gmail.com>
Message-ID: <6A38C937-9D7B-45FB-A1E3-0C65ED13E4E6@freeswitch.org>

-nonat only disable trying to find natpmp/upnp stuff at startup. it does not stop protocol specific nat handling (as in mod_sofia) from happening.

Make sure your NAT ACLs and and related settings are correct for your routae networks and you arent forcing some of the NDLB settings on the sofia profiles

Sent from my iPhone

> On Feb 16, 2021, at 04:15, David Villasmil <david.villasmil.work at gmail.com> wrote:
> 
> 
> -nonat ?
> 
>> On Tue, 16 Feb 2021 at 10:00, Carsten Bock <carsten at ng-voice.com> wrote:
>> Hi,
>> 
>> I have a quick NAT related question.
>> Is there a way to disable NAT handling completely?
>> 
>> FreeSwitch always replies to the IP/Port were it received the message from and not to the host/port in Via....
>> 
>> recv 1166 bytes from udp/[172.17.0.1]:36835 at 10:20:56.652918:
>> ------------------------------------------------------------------------
>> INVITE sip:ann-early at 167.99.136.226:5080 SIP/2.0
>> Record-Route: <sip:167.99.136.226:5070;lr>
>> Via: SIP/2.0/UDP 167.99.136.226:5070;branch=z9hG4bKe305.92946d84.0
>> Via: SIP/2.0/UDP 192.168.178.52:37350;received=93.221.23.15;branch=z9hG4bK-524287-1---936bd2b8103acd0c;rport=37350
>> [...]
>> 
>> send 366 bytes to udp/[172.17.0.1]:5070 at 10:20:56.659313:
>> ------------------------------------------------------------------------
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP 167.99.136.226:5070;branch=z9hG4bKe305.92946d84.0;received=172.17.0.1
>> From: <sip:b2bua at 167.99.136.226:5070;transport=UDP>;tag=7793d52b
>> To: <sip:3 at 167.99.136.226:5070>
>> Call-ID: akjc99KAbZfHxXHSea_iTQ..
>> CSeq: 1 INVITE
>> User-Agent: FreeSWITCH-mod_sofia/1.10.5-release+git-20201002T121516Z~a1bfb14c04~64bit
>> Content-Length: 0
>> 
>> Is there a way, to make FreeSwitch simply reply to the Host/Port in the Via-Header?
>> 
>> My Profile is rather simple and I tried it with setting the local network ACS and NDLB-force-rport:
>> 
>>     <profile name="local">
>>    <settings>
>>    <param name="local-network-acl" value="none"/>
>> <param name="NDLB-force-rport" value="disabled"/>    
>>    <param name="context" value="local"/>
>>         <param name="sip-ip" value="$${local_address}"/>
>>         <param name="ext-sip-ip" value="autonat:167.99.136.226"/>
>>         <param name="sip-port" value="5080"/>
>>         <param name="rtp-ip" value="$${local_address}"/>
>>         <param name="ext-rtp-ip" value="autonat:167.99.136.226"/>
>>         <param name="auth-calls" value="false"/>
>>       </settings>
>>     </profile>
>> 
>> Changing "NDLB-force-rport" to false did not have any impact.
>> 
>> Thanks,
>> Carsten
>> 
>> 
>> 
>> --
>> Carsten Bock I CTO & Founder
>> 
>> ng-voice GmbH
>> Trostbrücke 1 I 20457 Hamburg I Germany
>> T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com
>> 
>> Registry Office at Local Court Hamburg, HRB 120189
>> Managing Directors: Dr. David Bachmann, Carsten Bock
>> 
>> 
>> _________________________________________________________________________
>> 
>> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210216/9c6eb909/attachment-0001.html>

From krice at freeswitch.org  Tue Feb 16 17:17:36 2021
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 16 Feb 2021 11:17:36 -0600
Subject: [Freeswitch-users] Installing FS on RHEL 7
In-Reply-To: <CAH=VbNeo-Wp0NvLrqfmvv-m_8Cf2V2uqMj2dKc+ezGcJv2Okyw@mail.gmail.com>
References: <CAH=VbNeo-Wp0NvLrqfmvv-m_8Cf2V2uqMj2dKc+ezGcJv2Okyw@mail.gmail.com>
Message-ID: <017B559B-5E2D-4A1E-8911-0B901A9829A9@freeswitch.org>

check the confluence page for installing on centos. there are several libs there. also make sure you have epel enabled and available.


However, keep in mind that the core FS abandoned  rhel/centos ages ago for Debian as their primary dev platform. This was done for a laundry list of reason, one of which was simply not having libs or having old and broken libs in the distro repo

You’ll get the best feature list out of debian

Sent from my iPhone

> On Feb 16, 2021, at 05:43, Amit Chaudhuri <amit.k.chaudhuri at gmail.com> wrote:
> 
> Looks like you are missing a various packages that fs requires. If you
> scroll down the page you provided a link for, there are sections which
> appear to detail what needs to be installed first. Example
> libuuid-devel.
> A
> 
>> On Tue, 16 Feb 2021 at 11:12, Dinesh Krishnamurthy via
>> FreeSWITCH-users <freeswitch-users at lists.freeswitch.org> wrote:
>> 
>> 
>> 
>> 
>> ---------- Forwarded message ----------
>> From: Dinesh Krishnamurthy <krishdinesh at yahoo.com>
>> To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Bcc:
>> Date: Thu, 11 Feb 2021 05:59:51 +0000 (UTC)
>> Subject: Installing FS on RHEL 7
>> Hi,
>> 
>> I am trying to install FS on RHEL 7.x but i am facing issues while setting up. I am following the documentation provided on the confluence portal.
>> 
>> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7
>> 
>> 
>> Snapshot of few error messages, did not want to paste the whole error snapshot.
>> 
>> Error: Package: freeswitch-lang-pt-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>>           Requires: libspeexdsp.so.1()(64bit)
>> Error: Package: freeswitch-application-redis-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>>           Requires: libspeexdsp.so.1()(64bit)
>> Error: Package: libks-1.2.0-13.el7.centos.x86_64 (freeswitch)
>>           Requires: libuuid-devel
>> Error: Package: freeswitch-application-lcr-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>>           Requires: libpq.so.5()(64bit)
>> Error: Package: freeswitch-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>>           Requires: libogg
>> Error: Package: freeswitch-application-conference-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>>           Requires: libspeex.so.1()(64bit)
>> Error: Package: freeswitch-application-fsv-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>>           Requires: libspeex.so.1()(64bit)
>> Error: Package: freeswitch-application-httapi-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>>           Requires: libpq.so.5()(64bit)
>> Error: Package: freeswitch-lang-de-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>>           Requires: libspeex.so.1()(64bit)
>> 
>> 
>> Any help would be appreciated.
>> 
>> Thank you,
>> DK
>> 
>> 
>> 
>> ---------- Forwarded message ----------
>> From: Dinesh Krishnamurthy via FreeSWITCH-users <freeswitch-users at lists.freeswitch.org>
>> To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Bcc:
>> Date: Tue, 16 Feb 2021 03:12:25 -0800 (PST)
>> Subject: [Freeswitch-users] Installing FS on RHEL 7
>> _________________________________________________________________________
>> 
>> 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


From carsten at ng-voice.com  Tue Feb 16 12:08:12 2021
From: carsten at ng-voice.com (Carsten Bock)
Date: Tue, 16 Feb 2021 13:08:12 +0100
Subject: [Freeswitch-users] Reply to Via
In-Reply-To: <CAFGRPVoRvu7RthXEUs7TLSA9NE8g3+k=uQjKRQPh0Z1zAaWA+Q@mail.gmail.com>
References: <CAOCjumEBiQM-UJiNKv7X73HH_PuvHzF79TpsvL+=cdezrkcd0g@mail.gmail.com>
 <CAFGRPVoRvu7RthXEUs7TLSA9NE8g3+k=uQjKRQPh0Z1zAaWA+Q@mail.gmail.com>
Message-ID: <CAOCjumGxOWnFs_RW6DuuFXzah-9tFzx_6rqDxcbQJv5AgPxGWw@mail.gmail.com>

Yes, FreeSwitch is started with "-nonat"...

Thanks,
Carsten
--
Carsten Bock I CTO & Founder

ng-voice GmbH

Trostbrücke 1 I 20457 Hamburg I Germany
T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock


Am Di., 16. Feb. 2021 um 11:51 Uhr schrieb David Villasmil <
david.villasmil.work at gmail.com>:

> -nonat ?
>
> On Tue, 16 Feb 2021 at 10:00, Carsten Bock <carsten at ng-voice.com> wrote:
>
>> Hi,
>>
>> I have a quick NAT related question.
>> Is there a way to disable NAT handling completely?
>>
>> FreeSwitch always replies to the IP/Port were it received the message
>> from and not to the host/port in Via....
>>
>> recv 1166 bytes from udp/[172.17.0.1]:36835 at 10:20:56.652918:
>> ------------------------------------------------------------------------
>> INVITE sip:ann-early at 167.99.136.226:5080 SIP/2.0
>> Record-Route: <sip:167.99.136.226:5070;lr>
>> Via: SIP/2.0/UDP 167.99.136.226:5070;branch=z9hG4bKe305.92946d84.0
>> Via: SIP/2.0/UDP 192.168.178.52:37350
>> ;received=93.221.23.15;branch=z9hG4bK-524287-1---936bd2b8103acd0c;rport=37350
>> [...]
>>
>> send 366 bytes to udp/[172.17.0.1]:5070 at 10:20:56.659313:
>> ------------------------------------------------------------------------
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP 167.99.136.226:5070
>> ;branch=z9hG4bKe305.92946d84.0;received=172.17.0.1
>> From: <sip:b2bua at 167.99.136.226:5070;transport=UDP>;tag=7793d52b
>> To: <sip:3 at 167.99.136.226:5070>
>> Call-ID: akjc99KAbZfHxXHSea_iTQ..
>> CSeq: 1 INVITE
>> User-Agent:
>> FreeSWITCH-mod_sofia/1.10.5-release+git-20201002T121516Z~a1bfb14c04~64bit
>> Content-Length: 0
>>
>> Is there a way, to make FreeSwitch simply reply to the Host/Port in the
>> Via-Header?
>>
>> My Profile is rather simple and I tried it with setting the local network
>> ACS and NDLB-force-rport:
>>
>>     <profile name="local">
>>    <settings>
>>    <param name="local-network-acl" value="none"/>
>> <param name="NDLB-force-rport" value="disabled"/>
>>    <param name="context" value="local"/>
>>         <param name="sip-ip" value="$${local_address}"/>
>>         <param name="ext-sip-ip" value="autonat:167.99.136.226"/>
>>         <param name="sip-port" value="5080"/>
>>         <param name="rtp-ip" value="$${local_address}"/>
>>         <param name="ext-rtp-ip" value="autonat:167.99.136.226"/>
>>         <param name="auth-calls" value="false"/>
>>       </settings>
>>     </profile>
>>
>> Changing "NDLB-force-rport" to false did not have any impact.
>>
>> Thanks,
>> Carsten
>>
>>
>>
>> --
>> Carsten Bock I CTO & Founder
>>
>> ng-voice GmbH
>>
>> Trostbrücke 1
>> <https://www.google.com/maps/search/Trostbr%C3%BCcke+1?entry=gmail&source=g>
>> I 20457 Hamburg I Germany
>> T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com
>>
>> Registry Office at Local Court Hamburg, HRB 120189
>> Managing Directors: Dr. David Bachmann, Carsten Bock
>>
>>
>>
>> _________________________________________________________________________
>>
>> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210216/19a4e6a5/attachment.html>

From botelist at gmail.com  Wed Feb 17 17:36:54 2021
From: botelist at gmail.com (Bote Man)
Date: Wed, 17 Feb 2021 12:36:54 -0500
Subject: [Freeswitch-users] Reply to Via
In-Reply-To: <CAOCjumGxOWnFs_RW6DuuFXzah-9tFzx_6rqDxcbQJv5AgPxGWw@mail.gmail.com>
References: <CAOCjumEBiQM-UJiNKv7X73HH_PuvHzF79TpsvL+=cdezrkcd0g@mail.gmail.com>
 <CAFGRPVoRvu7RthXEUs7TLSA9NE8g3+k=uQjKRQPh0Z1zAaWA+Q@mail.gmail.com>
 <CAOCjumGxOWnFs_RW6DuuFXzah-9tFzx_6rqDxcbQJv5AgPxGWw@mail.gmail.com>
Message-ID: <004e01d70553$7c51a970$74f4fc50$@gmail.com>

I think the -nonat switch only stops FS from detecting NAT during startup. FreeSWITCH can be told to be very aggressive about working through NAT using various settings.

 

Also, I was recently trying to solve what I believed to be a NAT problem and discovered that the various NAT settings do not do exactly what you might think. It is worth reading the wiki about them.

 

https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal

 

That is all the help I can provide at this point, I’m afraid. 

 

 

John Boteler

Bote Communications

 

 

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Carsten Bock
Sent: Tuesday, 16 February, 2021 07:08
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Reply to Via

 

Yes, FreeSwitch is started with "-nonat"...

 

Thanks,

Carsten


--

Carsten Bock I CTO & Founder





ng-voice GmbH

Trostbrücke 1 I 20457 Hamburg I Germany
T +49 40 524 75 93-40 | M +49 179 2021244 I  <http://www.ng-voice.com/> www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock

 

 

Am Di., 16. Feb. 2021 um 11:51 Uhr schrieb David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com> >:

-nonat ?

 

On Tue, 16 Feb 2021 at 10:00, Carsten Bock <carsten at ng-voice.com <mailto:carsten at ng-voice.com> > wrote:

Hi,

 

I have a quick NAT related question.

Is there a way to disable NAT handling completely?

 

FreeSwitch always replies to the IP/Port were it received the message from and not to the host/port in Via....

 

recv 1166 bytes from udp/[172.17.0.1]:36835 at 10:20:56.652918:
------------------------------------------------------------------------
INVITE sip:ann-early at 167.99.136.226:5080 <http://sip:ann-early at 167.99.136.226:5080>  SIP/2.0
Record-Route: <sip:167.99.136.226:5070;lr>
Via: SIP/2.0/UDP 167.99.136.226:5070;branch=z9hG4bKe305.92946d84.0
Via: SIP/2.0/UDP 192.168.178.52:37350;received=93.221.23.15;branch=z9hG4bK-524287-1---936bd2b8103acd0c;rport=37350

[...]

 

send 366 bytes to udp/[172.17.0.1]:5070 at 10:20:56.659313:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 167.99.136.226:5070;branch=z9hG4bKe305.92946d84.0;received=172.17.0.1
From: <sip:b2bua at 167.99.136.226:5070;transport=UDP>;tag=7793d52b
To: <sip:3 at 167.99.136.226:5070 <http://sip:3 at 167.99.136.226:5070> >
Call-ID: akjc99KAbZfHxXHSea_iTQ..
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release+git-20201002T121516Z~a1bfb14c04~64bit
Content-Length: 0

 

Is there a way, to make FreeSwitch simply reply to the Host/Port in the Via-Header?

 

My Profile is rather simple and I tried it with setting the local network ACS and NDLB-force-rport:

 

    <profile name="local">
   <settings>
   <param name="local-network-acl" value="none"/>
<param name="NDLB-force-rport" value="disabled"/>    
   <param name="context" value="local"/>
        <param name="sip-ip" value="$${local_address}"/>
        <param name="ext-sip-ip" value="autonat:167.99.136.226"/>
        <param name="sip-port" value="5080"/>
        <param name="rtp-ip" value="$${local_address}"/>
        <param name="ext-rtp-ip" value="autonat:167.99.136.226"/>
        <param name="auth-calls" value="false"/>
      </settings>
    </profile>

 

Changing "NDLB-force-rport" to false did not have any impact.

 

Thanks,

Carsten

 

 

 

--

Carsten Bock I CTO & Founder





ng-voice GmbH

Trostbrücke 1 <https://www.google.com/maps/search/Trostbr%C3%BCcke+1?entry=gmail&source=g>  I 20457 Hamburg I Germany
T +49 40 524 75 93-40 | M +49 179 2021244 I  <http://www.ng-voice.com/> www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock

 

 

_________________________________________________________________________

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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210217/6957e105/attachment.html>

From brian at freeswitch.com  Wed Feb 17 21:12:48 2021
From: brian at freeswitch.com (Brian West)
Date: Wed, 17 Feb 2021 15:12:48 -0600
Subject: [Freeswitch-users] Reply to Via
In-Reply-To: <004e01d70553$7c51a970$74f4fc50$@gmail.com>
References: <CAOCjumEBiQM-UJiNKv7X73HH_PuvHzF79TpsvL+=cdezrkcd0g@mail.gmail.com>
 <CAFGRPVoRvu7RthXEUs7TLSA9NE8g3+k=uQjKRQPh0Z1zAaWA+Q@mail.gmail.com>
 <CAOCjumGxOWnFs_RW6DuuFXzah-9tFzx_6rqDxcbQJv5AgPxGWw@mail.gmail.com>
 <004e01d70553$7c51a970$74f4fc50$@gmail.com>
Message-ID: <CAEJMVkA5xB4_W+YRA9H6Nn5HamT9UV0YOhuvKmHotJauNX78bA@mail.gmail.com>

don't set ext-sip-ip or ext-rtp-ip or the local-network-acl and you
shouldn't have any processing anymore.

On Wed, Feb 17, 2021 at 12:08 PM Bote Man <botelist at gmail.com> wrote:

> I think the -nonat switch only stops FS from detecting NAT during startup.
> FreeSWITCH can be told to be very aggressive about working through NAT
> using various settings.
>
>
>
> Also, I was recently trying to solve what I believed to be a NAT problem
> and discovered that the various NAT settings do not do exactly what you
> might think. It is worth reading the wiki about them.
>
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal
>
>
>
> That is all the help I can provide at this point, I’m afraid.
>
>
>
>
>
> John Boteler
>
> Bote Communications
>
>
>
>
>
>
>
>
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Carsten Bock
> *Sent:* Tuesday, 16 February, 2021 07:08
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Reply to Via
>
>
>
> Yes, FreeSwitch is started with "-nonat"...
>
>
>
> Thanks,
>
> Carsten
>
> --
>
> Carsten Bock I CTO & Founder
>
>
>
> ng-voice GmbH
>
> Trostbrücke 1 I 20457 Hamburg I Germany
> T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com
>
> Registry Office at Local Court Hamburg, HRB 120189
> Managing Directors: Dr. David Bachmann, Carsten Bock
>
>
>
>
>
> Am Di., 16. Feb. 2021 um 11:51 Uhr schrieb David Villasmil <
> david.villasmil.work at gmail.com>:
>
> -nonat ?
>
>
>
> On Tue, 16 Feb 2021 at 10:00, Carsten Bock <carsten at ng-voice.com> wrote:
>
> Hi,
>
>
>
> I have a quick NAT related question.
>
> Is there a way to disable NAT handling completely?
>
>
>
> FreeSwitch always replies to the IP/Port were it received the message from
> and not to the host/port in Via....
>
>
>
> recv 1166 bytes from udp/[172.17.0.1]:36835 at 10:20:56.652918:
> ------------------------------------------------------------------------
> INVITE sip:ann-early at 167.99.136.226:5080 SIP/2.0
> Record-Route: <sip:167.99.136.226:5070;lr>
> Via: SIP/2.0/UDP 167.99.136.226:5070;branch=z9hG4bKe305.92946d84.0
> Via: SIP/2.0/UDP 192.168.178.52:37350
> ;received=93.221.23.15;branch=z9hG4bK-524287-1---936bd2b8103acd0c;rport=37350
>
> [...]
>
>
>
> send 366 bytes to udp/[172.17.0.1]:5070 at 10:20:56.659313:
> ------------------------------------------------------------------------
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 167.99.136.226:5070
> ;branch=z9hG4bKe305.92946d84.0;received=172.17.0.1
> From: <sip:b2bua at 167.99.136.226:5070;transport=UDP>;tag=7793d52b
> To: <sip:3 at 167.99.136.226:5070>
> Call-ID: akjc99KAbZfHxXHSea_iTQ..
> CSeq: 1 INVITE
> User-Agent:
> FreeSWITCH-mod_sofia/1.10.5-release+git-20201002T121516Z~a1bfb14c04~64bit
> Content-Length: 0
>
>
>
> Is there a way, to make FreeSwitch simply reply to the Host/Port in the
> Via-Header?
>
>
>
> My Profile is rather simple and I tried it with setting the local network
> ACS and NDLB-force-rport:
>
>
>
>     <profile name="local">
>    <settings>
>    <param name="local-network-acl" value="none"/>
> <param name="NDLB-force-rport" value="disabled"/>
>    <param name="context" value="local"/>
>         <param name="sip-ip" value="$${local_address}"/>
>         <param name="ext-sip-ip" value="autonat:167.99.136.226"/>
>         <param name="sip-port" value="5080"/>
>         <param name="rtp-ip" value="$${local_address}"/>
>         <param name="ext-rtp-ip" value="autonat:167.99.136.226"/>
>         <param name="auth-calls" value="false"/>
>       </settings>
>     </profile>
>
>
>
> Changing "NDLB-force-rport" to false did not have any impact.
>
>
>
> Thanks,
>
> Carsten
>
>
>
>
>
>
>
> --
>
> Carsten Bock I CTO & Founder
>
>
>
> ng-voice GmbH
>
> Trostbrücke 1
> <https://www.google.com/maps/search/Trostbr%C3%BCcke+1?entry=gmail&source=g>
> I 20457 Hamburg I Germany
> T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com
>
> Registry Office at Local Court Hamburg, HRB 120189
> Managing Directors: Dr. David Bachmann, Carsten Bock
>
>
>
>
>
> _________________________________________________________________________
>
> 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



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>

Email: brian at 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>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210217/8c6e99e4/attachment-0001.html>

From chad at apartmentlines.com  Thu Feb 18 20:00:02 2021
From: chad at apartmentlines.com (Chad Phillips)
Date: Thu, 18 Feb 2021 13:00:02 -0700
Subject: [Freeswitch-users] Realtime audio translation English <--> Spanish
Message-ID: <CAJ0sghSaYX2XbeuKZZgu9uVWKnAg0n2o-+hmmnBt5aUEU1Vc4A@mail.gmail.com>

Google's translation service seems like it could support realtime
translation of a conversation between an English and Spanish speaker.
Granted, there would be delays in translation, but that would be no
different than a live interpreter.

I'm imagining:

Spanish <--> FreeSWITCH <--> Google Translate <--> FreeSWITCH <--> English

...such that the audio of both speakers would not go directly to the other
party, but would instead be passed to Google Translate, with the returned
audio being played to the other party.

Has anybody taken a crack at this?

Chad
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210218/03c20064/attachment.html>

From dinesh.krishnamurthy at teleapps.com  Wed Feb 17 04:42:39 2021
From: dinesh.krishnamurthy at teleapps.com (Dinesh Krishnamurthy)
Date: Wed, 17 Feb 2021 10:12:39 +0530
Subject: [Freeswitch-users] Installing FS on RHEL 7
In-Reply-To: <017B559B-5E2D-4A1E-8911-0B901A9829A9@freeswitch.org>
References: <CAH=VbNeo-Wp0NvLrqfmvv-m_8Cf2V2uqMj2dKc+ezGcJv2Okyw@mail.gmail.com>
 <017B559B-5E2D-4A1E-8911-0B901A9829A9@freeswitch.org>
Message-ID: <000c01d704e7$5421d9a0$fc658ce0$@teleapps.com>

Thanks. Some of the customers specific demand RHEL so we must be ready. Though our setup has been with Debian we will have to understand the limitations if any on RHEL or other distributions and make decisions. 

-----Original Message-----
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Ken Rice
Sent: 16 February 2021 10:48 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Installing FS on RHEL 7

check the confluence page for installing on centos. there are several libs there. also make sure you have epel enabled and available.


However, keep in mind that the core FS abandoned  rhel/centos ages ago for Debian as their primary dev platform. This was done for a laundry list of reason, one of which was simply not having libs or having old and broken libs in the distro repo

You’ll get the best feature list out of debian

Sent from my iPhone

> On Feb 16, 2021, at 05:43, Amit Chaudhuri <amit.k.chaudhuri at gmail.com> wrote:
> 
> Looks like you are missing a various packages that fs requires. If 
> you scroll down the page you provided a link for, there are sections 
> which appear to detail what needs to be installed first. Example 
> libuuid-devel.
> A
> 
>> On Tue, 16 Feb 2021 at 11:12, Dinesh Krishnamurthy via 
>> FreeSWITCH-users <freeswitch-users at lists.freeswitch.org> wrote:
>> 
>> 
>> 
>> 
>> ---------- Forwarded message ----------
>> From: Dinesh Krishnamurthy <krishdinesh at yahoo.com>
>> To: "freeswitch-users at lists.freeswitch.org" 
>> <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Bcc:
>> Date: Thu, 11 Feb 2021 05:59:51 +0000 (UTC)
>> Subject: Installing FS on RHEL 7
>> Hi,
>> 
>> I am trying to install FS on RHEL 7.x but i am facing issues while setting up. I am following the documentation provided on the confluence portal.
>> 
>> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHE
>> L+7
>> 
>> 
>> Snapshot of few error messages, did not want to paste the whole error snapshot.
>> 
>> Error: Package: freeswitch-lang-pt-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>>           Requires: libspeexdsp.so.1()(64bit)
>> Error: Package: freeswitch-application-redis-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>>           Requires: libspeexdsp.so.1()(64bit)
>> Error: Package: libks-1.2.0-13.el7.centos.x86_64 (freeswitch)
>>           Requires: libuuid-devel
>> Error: Package: freeswitch-application-lcr-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>>           Requires: libpq.so.5()(64bit)
>> Error: Package: freeswitch-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>>           Requires: libogg
>> Error: Package: freeswitch-application-conference-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>>           Requires: libspeex.so.1()(64bit)
>> Error: Package: freeswitch-application-fsv-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>>           Requires: libspeex.so.1()(64bit)
>> Error: Package: freeswitch-application-httapi-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>>           Requires: libpq.so.5()(64bit)
>> Error: Package: freeswitch-lang-de-1.10.5.release.8-1.el7.x86_64 (freeswitch)
>>           Requires: libspeex.so.1()(64bit)
>> 
>> 
>> Any help would be appreciated.
>> 
>> Thank you,
>> DK
>> 
>> 
>> 
>> ---------- Forwarded message ----------
>> From: Dinesh Krishnamurthy via FreeSWITCH-users 
>> <freeswitch-users at lists.freeswitch.org>
>> To: "freeswitch-users at lists.freeswitch.org" 
>> <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Bcc:
>> Date: Tue, 16 Feb 2021 03:12:25 -0800 (PST)
>> Subject: [Freeswitch-users] Installing FS on RHEL 7 
>> _____________________________________________________________________
>> ____
>> 
>> 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-us
>> ers
>> 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-use
> rs
> 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



From praveenkumar.t at teleapps.com  Thu Feb 18 14:41:16 2021
From: praveenkumar.t at teleapps.com (Praveenkumar T)
Date: Thu, 18 Feb 2021 20:11:16 +0530
Subject: [Freeswitch-users] Hardware Specification and Performance
Message-ID: <007d01d70604$2134c020$639e4060$@teleapps.com>

Dear, 

 

I have been going through the below link and It would be helpful if you
could let me know the below two points. 

 

https://freeswitch.org/confluence/display/FREESWITCH/Specifications

 



 

1.	like to know more about the maximum concurrent call the free switch
can support with the recommended system requirement. 
2.	Does it need any upgrade in the system hardware
requirements/additional packages if one tries to bring SBC before the
Freeswitch in case of external SIP trunk configuration.  

 

 

Thanks & Regards

Praveen

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210218/4f1426e3/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 14648 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210218/4f1426e3/attachment-0001.jpg>

From botelist at gmail.com  Thu Feb 18 20:44:10 2021
From: botelist at gmail.com (Bote Man)
Date: Thu, 18 Feb 2021 15:44:10 -0500
Subject: [Freeswitch-users] FreeSWITCH to Shoretel Connect
Message-ID: <001301d70636$cf9e55e0$6edb01a0$@gmail.com>

I am trying to get FreeSWITCH to place outbound calls through a Shoretel
Connect ST100 (now labeled Mitel).

 

What is the best approach? 

 

For now FreeSWITCH has a gateway definition that registers as an extension
on the Shoretel.

 

Another company manages the Shoretel so I have no direct access to any
programming screens. So far they have tried the "system" profile on their
end which works sometimes, other times it doesn't.

 

Sometimes a call to another extension completes immediately, most times it
takes 20-30 seconds before successful call progress, and still other times
it drops after timing out. My best guess is that it is trying the 4 other
"SIP Phone Profiles" that are set to higher priorities until it finally
times out and uses the lowest priority profile named "System" with a user
agent string of ".*" ( dot star )

 

Is there any magic setting that I could be missing?

 

Thanks.

 

 

--- 

John Boteler 

BnC Group U.S.A. 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210218/17fc2c71/attachment.html>

From avi at avimarcus.net  Thu Feb 18 20:46:13 2021
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 18 Feb 2021 20:46:13 +0000
Subject: [Freeswitch-users] Installing FS on RHEL 7
In-Reply-To: <000c01d704e7$5421d9a0$fc658ce0$@teleapps.com>
References: <CAH=VbNeo-Wp0NvLrqfmvv-m_8Cf2V2uqMj2dKc+ezGcJv2Okyw@mail.gmail.com>
 <017B559B-5E2D-4A1E-8911-0B901A9829A9@freeswitch.org>
 <000c01d704e7$5421d9a0$fc658ce0$@teleapps.com>
Message-ID: <01000177b6e4a4ca-41a0524f-d482-403f-a8fe-8f77d5f7a930-000000@email.amazonses.com>

To mention an option: people are running freeswitch in docker, which seems
to work on RHEL 7.

On Thu, Feb 18, 2021, 10:22 PM Dinesh Krishnamurthy <
dinesh.krishnamurthy at teleapps.com> wrote:

> Thanks. Some of the customers specific demand RHEL so we must be ready.
> Though our setup has been with Debian we will have to understand the
> limitations if any on RHEL or other distributions and make decisions.
>
> -----Original Message-----
> From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On
> Behalf Of Ken Rice
> Sent: 16 February 2021 10:48 PM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Installing FS on RHEL 7
>
> check the confluence page for installing on centos. there are several libs
> there. also make sure you have epel enabled and available.
>
>
> However, keep in mind that the core FS abandoned  rhel/centos ages ago for
> Debian as their primary dev platform. This was done for a laundry list of
> reason, one of which was simply not having libs or having old and broken
> libs in the distro repo
>
> You’ll get the best feature list out of debian
>
> Sent from my iPhone
>
> > On Feb 16, 2021, at 05:43, Amit Chaudhuri <amit.k.chaudhuri at gmail.com>
> wrote:
> >
> > Looks like you are missing a various packages that fs requires. If
> > you scroll down the page you provided a link for, there are sections
> > which appear to detail what needs to be installed first. Example
> > libuuid-devel.
> > A
> >
> >> On Tue, 16 Feb 2021 at 11:12, Dinesh Krishnamurthy via
> >> FreeSWITCH-users <freeswitch-users at lists.freeswitch.org> wrote:
> >>
> >>
> >>
> >>
> >> ---------- Forwarded message ----------
> >> From: Dinesh Krishnamurthy <krishdinesh at yahoo.com>
> >> To: "freeswitch-users at lists.freeswitch.org"
> >> <freeswitch-users at lists.freeswitch.org>
> >> Cc:
> >> Bcc:
> >> Date: Thu, 11 Feb 2021 05:59:51 +0000 (UTC)
> >> Subject: Installing FS on RHEL 7
> >> Hi,
> >>
> >> I am trying to install FS on RHEL 7.x but i am facing issues while
> setting up. I am following the documentation provided on the confluence
> portal.
> >>
> >> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHE
> >> L+7
> >>
> >>
> >> Snapshot of few error messages, did not want to paste the whole error
> snapshot.
> >>
> >> Error: Package: freeswitch-lang-pt-1.10.5.release.8-1.el7.x86_64
> (freeswitch)
> >>           Requires: libspeexdsp.so.1()(64bit)
> >> Error: Package:
> freeswitch-application-redis-1.10.5.release.8-1.el7.x86_64 (freeswitch)
> >>           Requires: libspeexdsp.so.1()(64bit)
> >> Error: Package: libks-1.2.0-13.el7.centos.x86_64 (freeswitch)
> >>           Requires: libuuid-devel
> >> Error: Package:
> freeswitch-application-lcr-1.10.5.release.8-1.el7.x86_64 (freeswitch)
> >>           Requires: libpq.so.5()(64bit)
> >> Error: Package: freeswitch-1.10.5.release.8-1.el7.x86_64 (freeswitch)
> >>           Requires: libogg
> >> Error: Package:
> freeswitch-application-conference-1.10.5.release.8-1.el7.x86_64 (freeswitch)
> >>           Requires: libspeex.so.1()(64bit)
> >> Error: Package:
> freeswitch-application-fsv-1.10.5.release.8-1.el7.x86_64 (freeswitch)
> >>           Requires: libspeex.so.1()(64bit)
> >> Error: Package:
> freeswitch-application-httapi-1.10.5.release.8-1.el7.x86_64 (freeswitch)
> >>           Requires: libpq.so.5()(64bit)
> >> Error: Package: freeswitch-lang-de-1.10.5.release.8-1.el7.x86_64
> (freeswitch)
> >>           Requires: libspeex.so.1()(64bit)
> >>
> >>
> >> Any help would be appreciated.
> >>
> >> Thank you,
> >> DK
> >>
> >>
> >>
> >> ---------- Forwarded message ----------
> >> From: Dinesh Krishnamurthy via FreeSWITCH-users
> >> <freeswitch-users at lists.freeswitch.org>
> >> To: "freeswitch-users at lists.freeswitch.org"
> >> <freeswitch-users at lists.freeswitch.org>
> >> Cc:
> >> Bcc:
> >> Date: Tue, 16 Feb 2021 03:12:25 -0800 (PST)
> >> Subject: [Freeswitch-users] Installing FS on RHEL 7
> >> _____________________________________________________________________
> >> ____
> >>
> >> 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-us
> >> ers
> >> 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-use
> > rs
> > 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210218/992f7846/attachment-0001.html>

From krice at freeswitch.org  Thu Feb 18 23:06:34 2021
From: krice at freeswitch.org (krice at freeswitch.org)
Date: Thu, 18 Feb 2021 17:06:34 -0600
Subject: [Freeswitch-users] Hardware Specification and Performance
In-Reply-To: <007d01d70604$2134c020$639e4060$@teleapps.com>
References: <007d01d70604$2134c020$639e4060$@teleapps.com>
Message-ID: <066601d7064a$b4b953b0$1e2bfb10$@freeswitch.org>

There is no specific recommendation for scaling FreeSWITCH this is because
single setting changes can cause an order of magnitude or more differences
in concurrent call handing capabilities on the same exact hardware and
software builds. 

 

You will have to test your specific deployment scenarios directly.

 

Example, a box running G.711 to  G.711 calls is simply proxying media and
may easily handle 1000 calls, however make those G729 to G711 calls and that
number may be cut in half to 500 calls, now change that to OPUS Stereo  32K
to G.729 may only do 10% of that capacity due to the additional load
handling required to decode opus to SLIN, then resample it to 8K and mux the
2 stereo channels down to 1 channel, then recompress it all the way down to
G.729. And we havent even talked about voicemail, conferencing/3way calling,
etc etc etc etc etc etc

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On
Behalf Of Praveenkumar T
Sent: Thursday, February 18, 2021 8:41 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Hardware Specification and Performance

 

Dear, 

 

I have been going through the below link and It would be helpful if you
could let me know the below two points. 

 

https://freeswitch.org/confluence/display/FREESWITCH/Specifications

 



 

1.	like to know more about the maximum concurrent call the free switch
can support with the recommended system requirement. 
2.	Does it need any upgrade in the system hardware
requirements/additional packages if one tries to bring SBC before the
Freeswitch in case of external SIP trunk configuration.  

 

 

Thanks & Regards

Praveen

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210218/fe5d5a11/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 8188 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210218/fe5d5a11/attachment.jpg>

From s.kainz at wnt.at  Thu Feb 18 23:12:27 2021
From: s.kainz at wnt.at (Stefan Kainz)
Date: Thu, 18 Feb 2021 23:12:27 +0000
Subject: [Freeswitch-users] Realtime audio translation English <-->
 Spanish
In-Reply-To: <CAJ0sghSaYX2XbeuKZZgu9uVWKnAg0n2o-+hmmnBt5aUEU1Vc4A@mail.gmail.com>
References: <CAJ0sghSaYX2XbeuKZZgu9uVWKnAg0n2o-+hmmnBt5aUEU1Vc4A@mail.gmail.com>
Message-ID: <57584BF3-8514-4372-804F-0481794F74F4@wnt.at>

I haven’t tried, but I have repeatedly thought about it … 
My findings were that you have to pay on a per channel basis for the mrcp google translate module. 

It would be cool if the participants can’t hear each other, but simply hear the translation.

Google TTS even has a phone-quality setting, which I guess uses 8000Mhz as Frequency.

Regards, 

> On 18.02.2021, at 21:00, Chad Phillips <chad at apartmentlines.com> wrote:
> 
> Google's translation service seems like it could support realtime translation of a conversation between an English and Spanish speaker. Granted, there would be delays in translation, but that would be no different than a live interpreter.
> 
> I'm imagining:
> 
> Spanish <--> FreeSWITCH <--> Google Translate <--> FreeSWITCH <--> English
> 
> ...such that the audio of both speakers would not go directly to the other party, but would instead be passed to Google Translate, with the returned audio being played to the other party.
> 
> Has anybody taken a crack at this?
> 
> Chad
> _________________________________________________________________________
> 
> 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 botelist at gmail.com  Fri Feb 19 16:06:46 2021
From: botelist at gmail.com (Bote Man)
Date: Fri, 19 Feb 2021 11:06:46 -0500
Subject: [Freeswitch-users] Realtime audio translation English <-->
	Spanish
In-Reply-To: <57584BF3-8514-4372-804F-0481794F74F4@wnt.at>
References: <CAJ0sghSaYX2XbeuKZZgu9uVWKnAg0n2o-+hmmnBt5aUEU1Vc4A@mail.gmail.com>
 <57584BF3-8514-4372-804F-0481794F74F4@wnt.at>
Message-ID: <007801d706d9$3929f580$ab7de080$@gmail.com>

Be careful, that high frequency could cause R.F. burns! OUCH!

Unless you mean 8000Hz, which is much safer :-)


 --- 
John Boteler 
BnC Group U.S.A. 



-----Original Message-----
From: Stefan Kainz
Sent: Thursday, 18 February, 2021 18:12
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Realtime audio translation English <--> Spanish

Google TTS even has a phone-quality setting, which I guess uses 8000Mhz as Frequency.

Regards, 




From grcamauer at gmail.com  Fri Feb 19 17:39:12 2021
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Fri, 19 Feb 2021 14:39:12 -0300
Subject: [Freeswitch-users] Realtime audio translation English <-->
	Spanish
In-Reply-To: <007801d706d9$3929f580$ab7de080$@gmail.com>
References: <CAJ0sghSaYX2XbeuKZZgu9uVWKnAg0n2o-+hmmnBt5aUEU1Vc4A@mail.gmail.com>
 <57584BF3-8514-4372-804F-0481794F74F4@wnt.at>
 <007801d706d9$3929f580$ab7de080$@gmail.com>
Message-ID: <CAPNTnnrL6U5A2kXgFT4eC1BEs8PnvYjLPpxKYChnp0uSYouLiw@mail.gmail.com>

These modules might help you along:
https://github.com/drachtio/drachtio-freeswitch-modules

There is one to fork audio and another to interface with GoogleTTS.

Guillermo

On Fri, Feb 19, 2021 at 1:41 PM Bote Man <botelist at gmail.com> wrote:

> Be careful, that high frequency could cause R.F. burns! OUCH!
>
> Unless you mean 8000Hz, which is much safer :-)
>
>
>  ---
> John Boteler
> BnC Group U.S.A.
>
>
>
> -----Original Message-----
> From: Stefan Kainz
> Sent: Thursday, 18 February, 2021 18:12
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Realtime audio translation English <-->
> Spanish
>
> Google TTS even has a phone-quality setting, which I guess uses 8000Mhz as
> Frequency.
>
> Regards,
>
>
>
> _________________________________________________________________________
>
> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210219/1f5208ff/attachment.html>

From dragos at freeswitch.org  Sat Feb 20 10:48:33 2021
From: dragos at freeswitch.org (Dragos Oancea)
Date: Sat, 20 Feb 2021 12:48:33 +0200
Subject: [Freeswitch-users] Freeswitch High Memory Consumption
In-Reply-To: <000a01d6fe5c$401eb950$c05c2bf0$@inoutglobal.xyz>
References: <000a01d6fe5c$401eb950$c05c2bf0$@inoutglobal.xyz>
Message-ID: <CAGvJV4qsEYF=2NCzfpKVK5JPzKXg84C30qk+BZqTn6Bcvk5WPA@mail.gmail.com>

When it comes to memory leaks you need to use ASAN (or valgrind).

apt-get install libasan5
cd freeswitch
./configure --enable-address-sanitizer --enable-pool-sanitizer && make
install
export LD_PRELOAD=/usr/lib/gcc/x86_64-linux-gnu/8/libasan.so
export ASAN_OPTIONS=verbosity=1
/usr/local/freeswitch/bin/freeswitch -nf -vg -core

Run some call scenarios you suspect lead to mem leaks and then shut down
FS, you'll see the ASAN output at shutdown.

On Mon, Feb 8, 2021 at 10:54 PM <edson at inoutglobal.xyz> wrote:

> Dears,
>
>
>
> Is it possible to find out what Freeswitch’s  task leads operation system
>  run out of memory?
>
> It causes OS to start OOM Killer.
>
>
>
> 1-This is a backtrace Log output.
>
> https://pastebin.com/SLEPh4p2
>
>
>
>
>
> 2-dmesg output showing OOM killer in action.
>
>
>
> [1385455.947689] Out of memory: Kill process 21698 (freeswitch) score 834
> or sacrifice child
>
> [1385455.956673] Killed process 21698 (freeswitch) total-vm:12718944kB,
> anon-rss:8202856kB, file-rss:0kB, shmem-rss:0kB
>
>
>
> Regards,
>
>
>
> Edson
> _________________________________________________________________________
>
> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210220/ffb57506/attachment.html>

From dragos at freeswitch.org  Sat Feb 20 10:57:19 2021
From: dragos at freeswitch.org (Dragos Oancea)
Date: Sat, 20 Feb 2021 12:57:19 +0200
Subject: [Freeswitch-users] Hardware Specification and Performance
In-Reply-To: <066601d7064a$b4b953b0$1e2bfb10$@freeswitch.org>
References: <007d01d70604$2134c020$639e4060$@teleapps.com>
 <066601d7064a$b4b953b0$1e2bfb10$@freeswitch.org>
Message-ID: <CAGvJV4oJrDOS8vhUjxauzyhoS5wKiqb6MpPuW3BppFxNVYViXQ@mail.gmail.com>

A rather old page but likely still useful:
https://freeswitch.org/confluence/display/FREESWITCH/Performance+Testing+and+Configurations

Stress-test the box with sipp (include RTP in the sipp scenario).



On Fri, Feb 19, 2021 at 1:07 AM <krice at freeswitch.org> wrote:

> There is no specific recommendation for scaling FreeSWITCH this is because
> single setting changes can cause an order of magnitude or more differences
> in concurrent call handing capabilities on the same exact hardware and
> software builds.
>
>
>
> You will have to test your specific deployment scenarios directly.
>
>
>
> Example, a box running G.711 to  G.711 calls is simply proxying media and
> may easily handle 1000 calls, however make those G729 to G711 calls and
> that number may be cut in half to 500 calls, now change that to OPUS
> Stereo  32K to G.729 may only do 10% of that capacity due to the additional
> load handling required to decode opus to SLIN, then resample it to 8K and
> mux the 2 stereo channels down to 1 channel, then recompress it all the way
> down to G.729. And we havent even talked about voicemail, conferencing/3way
> calling, etc etc etc etc etc etc
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Praveenkumar T
> *Sent:* Thursday, February 18, 2021 8:41 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Hardware Specification and Performance
>
>
>
> Dear,
>
>
>
> I have been going through the below link and It would be helpful if you
> could let me know the below two points.
>
>
>
> *https://freeswitch.org/confluence/display/FREESWITCH/Specifications
> <https://freeswitch.org/confluence/display/FREESWITCH/Specifications>*
>
>
>
>
>
>    1. like to know more about the maximum concurrent call the free switch
>    can support with the recommended system requirement.
>    2. Does it need any upgrade in the system hardware
>    requirements/additional packages if one tries to bring SBC before the
>    Freeswitch in case of external SIP trunk configuration.
>
>
>
>
>
> Thanks & Regards
>
> Praveen
> _________________________________________________________________________
>
> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210220/23765d19/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: image/jpeg
Size: 8188 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210220/23765d19/attachment-0001.jpg>

From davidswalkabout at gmail.com  Sun Feb 21 00:27:11 2021
From: davidswalkabout at gmail.com (David P)
Date: Sun, 21 Feb 2021 13:27:11 +1300
Subject: [Freeswitch-users] Chrome is deprecating plan-b SDP semantics.
	Will verto be updated?
In-Reply-To: <mailman.0.1613217602.3637.freeswitch-users@lists.freeswitch.org>
References: <mailman.0.1613217602.3637.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAE23trvpUS8CAhhHVfcOnPBigeB8654Mzwc034acijVJrO1TDw@mail.gmail.com>

I've dipped into a few of the ClueCon 2021 youtube videos in search of an
announcement, but I haven't found any yet. I did see Anthony express
excitement about the WebRTC spec being finalized.

Has there been an announcement about verto getting unified-plan support?

On Sun, 14 Feb 2021, 1:09 am , <
freeswitch-users-request at lists.freeswitch.org> wrote:

Google announced that support in Chrome for "plan-b" SDP semantics will be
> removed this August:
>
> https://groups.google.com/g/discuss-webrtc/c/Zrsn2hi8FV0/m/KIbn0EZPBQAJ?authuser=0
> As I understand it, this means that FS' verto js library will stop working
> unless it's updated to use "unified-plan" semantics.
>
> I searched github and google for "verto" and "jquery.verto.js" to see if
> I'd missed an announcement that this change was in-progress or done, but I
> don't get a good match at either. Has it been moved out of
> https://github.com/signalwire/freeswitch to a community repo? Does it
> have a maintainer?
>
> I can't become the maintainer, but this is a serious problem for us if we
> don't have a solution by August.
>
> Btw, Brian, thank you for your answer about HLS.
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210221/3b56e117/attachment.html>

From matt at doics.co  Sat Feb 20 15:52:09 2021
From: matt at doics.co (Matthew Macdonald-Wallace)
Date: Sat, 20 Feb 2021 15:52:09 +0000
Subject: [Freeswitch-users] Working with an ancient SPA-3000 in the UK,
 can't get gateway registration working?
Message-ID: <6afc2464ba1c378e960b5a59a9135d8bdc279f59@hey.com>

Hi all,

In an attempt to add VoIP to my home network, I'm working with
Freeswitch and FusionPBX.

My outbound connection to the PSTN is via an aging SPA-3000, and whilst
I can get the SPA-3000 to register correctly as a "user", I can't get
the "gateway" side of things in FusionPBX/Freeswitch to register with
the SPA-3000 for outgoing calls.

All of the articles I can find online relate to the SPA-3201 or similar,
which is a very different beast to the SPA-3000.

The error in the logs is as follows:

2021-02-20 08:53:57.680948 [NOTICE] sofia_reg.c:454 Registering
b1039375-091b-478c-86e4-11a25d3c9d00
2021-02-20 08:54:00.820948 [ERR] sofia_reg.c:2470 b1039375-091b-478c-
86e4-11a25d3c9d00 Failed Registration with status Service Unavailable
[503]. failure #35
2021-02-20 08:54:01.700955 [WARNING] sofia_reg.c:511 b1039375-091b-478c-
86e4-11a25d3c9d00 Failed Registration [503], setting retry to 30
seconds.

and my settings look like this:

Gateway: outbound
username: 1900
password: 1234
>From User: <my landline number>
>From Domain: <BLANK>
Proxy: <IP ADDRESS OF SPA-3000>:5060
Realm: <IP ADDRESS OF SPA-3000>
Expire Seconds: 800
Retry Seconds: 30
Distinct To: <BLANK>
Auth Username: user
Extension <Blank>
Register Transport: <BLANK>
Register Proxy: <IP ADDRESS OF SPA-3000>:5060
Outbound Proxy: <IP ADDRESS OF SPA-3000>
Caller ID in From: True
Suppress CNG: <BLANK>
Sip CID Type: <BLANK>
Codec Preferences: <BLANK>
Extension in Contact: <BLANK>
Ping: <Blank>
Ping Min: <Blank>
Ping Max: <Blank>
Channels: 1
Hostname: <Blank>
Domain: Global
Context: Public
Profile: External
Enabled: True
Description: Blank


I'm sure this is something really obvious that I'm missing in the
settings, so all help is appreciated!

Thanks in advance,

Matt
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210220/836c8649/attachment.html>

From keithxcroxford at gmail.com  Sat Feb 20 18:48:31 2021
From: keithxcroxford at gmail.com (Keith Croxford)
Date: Sat, 20 Feb 2021 19:48:31 +0100
Subject: [Freeswitch-users] DTMF not passing from the A Leg to the B Leg
Message-ID: <CABb=kMcK7jeMgAMpHR=sPe3vJpgj_Rp5BttaSLj3xappyTWE_g@mail.gmail.com>

I'm coming across a strange, intermittent behavior with RFC2833 DTMF
packets. This is occurring in a production environment, and I've been able
to duplicate this with a very minimal FreeSWITCH installation in a virtual
machine.

Details of the problem:

A customer occasionally has a problem report where their callers are unable
to use the IVR menu. From the user's standpoint, the digits are not
detected. Upon further review, we do receive DTMF event packets from the
carrier, however, they do not traverse the bridge from the A leg to the B
leg.

>From the console logs, I see the following with the first digit.
--
2021-02-20 11:28:30.842108 [DEBUG] mod_sofia.c:645 SOFIA EXCHANGE_MEDIA
2021-02-20 11:28:30.881612 [DEBUG] switch_rtp.c:1890 rtcp_stats_init: audio
ssrc[1488669269] base_seq[26830]
2021-02-20 11:28:30.933002 [DEBUG] switch_rtp.c:7550 Correct audio ip/port
confirmed.
2021-02-20 11:28:34.881612 [DEBUG] switch_rtp.c:7550 Correct audio ip/port
confirmed.
2021-02-20 11:28:34.923555 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 1:400
2021-02-20 11:28:34.923555 [INFO] switch_channel.c:515 RECV DTMF 1:400
2021-02-20 11:28:34.923555 [DEBUG] switch_rtp.c:5424 Send start packet for
[1] ts=160 dur=160/160/400 seq=51938 lw=160
2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5323 Send middle packet for
[1] ts=160 dur=320/320/400 seq=51939 lw=320
2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5323 Send end packet for
[1] ts=160 dur=480/480/400 seq=51940 lw=320
2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5323 Send end packet for
[1] ts=160 dur=480/480/400 seq=51941 lw=320
2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5323 Send end packet for
[1] ts=160 dur=480/480/400 seq=51942 lw=320
2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5271 Queue digit delay of
40ms
2021-02-20 11:28:35.421487 [DEBUG] switch_rtp.c:6985 Correct audio RTCP
ip/port confirmed.
--

However, on the following series of digits, they are detected as seen in
the logs from switch_rtp.c and switch_channel.c. However, they do not
proceed to the "Send start packet" step as seen with the previous digit.
This mirrors what I see on the B leg client, as it only detects the first
digit.

--
2021-02-20 11:28:37.521489 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 1:400
2021-02-20 11:28:37.521489 [INFO] switch_channel.c:515 RECV DTMF 1:400
2021-02-20 11:28:46.082031 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 3:400
2021-02-20 11:28:46.082031 [INFO] switch_channel.c:515 RECV DTMF 3:400
2021-02-20 11:28:46.481377 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 6:400
2021-02-20 11:28:46.481377 [INFO] switch_channel.c:515 RECV DTMF 6:400
2021-02-20 11:28:47.061795 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 4:400
2021-02-20 11:28:47.061795 [INFO] switch_channel.c:515 RECV DTMF 4:400
2021-02-20 11:28:47.461504 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 5:400
2021-02-20 11:28:47.461504 [INFO] switch_channel.c:515 RECV DTMF 5:400
2021-02-20 11:28:47.881346 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 0:400
2021-02-20 11:28:47.881346 [INFO] switch_channel.c:515 RECV DTMF 0:400
2021-02-20 11:28:48.281220 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 0:400
2021-02-20 11:28:48.281220 [INFO] switch_channel.c:515 RECV DTMF 0:400
2021-02-20 11:28:48.805572 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF #:400
2021-02-20 11:28:48.805572 [INFO] switch_channel.c:515 RECV DTMF #:400
2021-02-20 11:29:04.881294 [NOTICE] sofia.c:1079 Hangup sofia/external/
nobody at 192.168.1.71 [CS_EXECUTE] [NORMAL_CLEARING]
--

Yet, they do arrive on the Freeswitch instances. I've attempted to
add <action application="set"
data="rtp_manual_rtp_bugs=ignore_dtmf_duration"/>  to my inbound dialplan
xml. I'm leaning toward these being a "weird" DTMF packet and not a
FreeSWITCH issue, but I'm curious if anyone has encountered a similar
problem, or has advice of what we could test on the FS side.
--
Supporting Details:
Test Environment:
I took a packet capture from production and filtered it down to the A leg's
RFC2833 packets, and saved this to a new file. I then used this pcap file
in a SIPP UAC scenario which is acting as the A Leg.

Freeswitch sits in the middle and I've configured it following this guide (
https://freeswitch.org/confluence/display/FREESWITCH/SBC+FreeSWITCH+Configuration+Example+2)
as a barebones SBC. Freeswitch is version 1.8.3

The B Leg is a simple pjsua client configured to answer an inbound call.
--
Best Regards,

KC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210220/9ec2e556/attachment.html>

From covici at ccs.covici.com  Mon Feb 22 19:39:31 2021
From: covici at ccs.covici.com (John Covici)
Date: Mon, 22 Feb 2021 14:39:31 -0500
Subject: [Freeswitch-users] Working with an ancient SPA-3000 in the UK,
	can't get gateway registration working?
In-Reply-To: <6afc2464ba1c378e960b5a59a9135d8bdc279f59@hey.com>
References: <6afc2464ba1c378e960b5a59a9135d8bdc279f59@hey.com>
Message-ID: <m3blcbdh8s.wl-covici@ccs.covici.com>

I am pretty sure that the port for outbound has to be different, at
this is the way I remember it from a few years ago when I had to do
this.

On Sat, 20 Feb 2021 10:52:09 -0500,
Matthew Macdonald-Wallace wrote:
> 
> [1  <multipart/alternative; UTF-8 (7bit)>]
> [1.1  <text/plain; UTF-8 (7bit)>]
> Hi all,
> 
> In an attempt to add VoIP to my home network, I'm working with
> Freeswitch and FusionPBX.
> 
> My outbound connection to the PSTN is via an aging SPA-3000, and whilst
> I can get the SPA-3000 to register correctly as a "user", I can't get
> the "gateway" side of things in FusionPBX/Freeswitch to register with
> the SPA-3000 for outgoing calls.
> 
> All of the articles I can find online relate to the SPA-3201 or similar,
> which is a very different beast to the SPA-3000.
> 
> The error in the logs is as follows:
> 
> 2021-02-20 08:53:57.680948 [NOTICE] sofia_reg.c:454 Registering
> b1039375-091b-478c-86e4-11a25d3c9d00
> 2021-02-20 08:54:00.820948 [ERR] sofia_reg.c:2470 b1039375-091b-478c-
> 86e4-11a25d3c9d00 Failed Registration with status Service Unavailable
> [503]. failure #35
> 2021-02-20 08:54:01.700955 [WARNING] sofia_reg.c:511 b1039375-091b-478c-
> 86e4-11a25d3c9d00 Failed Registration [503], setting retry to 30
> seconds.
> 
> and my settings look like this:
> 
> Gateway: outbound
> username: 1900
> password: 1234
> From User: <my landline number>
> From Domain: <BLANK>
> Proxy: <IP ADDRESS OF SPA-3000>:5060
> Realm: <IP ADDRESS OF SPA-3000>
> Expire Seconds: 800
> Retry Seconds: 30
> Distinct To: <BLANK>
> Auth Username: user
> Extension <Blank>
> Register Transport: <BLANK>
> Register Proxy: <IP ADDRESS OF SPA-3000>:5060
> Outbound Proxy: <IP ADDRESS OF SPA-3000>
> Caller ID in From: True
> Suppress CNG: <BLANK>
> Sip CID Type: <BLANK>
> Codec Preferences: <BLANK>
> Extension in Contact: <BLANK>
> Ping: <Blank>
> Ping Min: <Blank>
> Ping Max: <Blank>
> Channels: 1
> Hostname: <Blank>
> Domain: Global
> Context: Public
> Profile: External
> Enabled: True
> Description: Blank
> 
> 
> I'm sure this is something really obvious that I'm missing in the
> settings, so all help is appreciated!
> 
> Thanks in advance,
> 
> Matt
> [1.2  <text/html; UTF-8 (quoted-printable)>]
> [2  <text/plain; utf-8 (base64)>]
> _________________________________________________________________________
> 
> 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

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici wb2una
         covici at ccs.covici.com


From lmorley at neny.cslimits.net  Mon Feb 22 23:56:53 2021
From: lmorley at neny.cslimits.net (Larry Morley)
Date: Mon, 22 Feb 2021 18:56:53 -0500
Subject: [Freeswitch-users] Working with an ancient SPA-3000 in the UK,
	can't get gateway registration working?
In-Reply-To: <6afc2464ba1c378e960b5a59a9135d8bdc279f59@hey.com>
References: <6afc2464ba1c378e960b5a59a9135d8bdc279f59@hey.com>
Message-ID: <d4724bff-a580-41b9-a592-ef4bd0d0fc84@neny.cslimits.net>

If it 'twas me, I'd be using
"*" for "Realm" (or leave it blank); and, I'd use the address of my proxy host for "Proxy" - if all I had was an FS server, I'd try "sip:fssrvr.com:5060".

See if that won't at least get the thing to register.

tcpdump and wireshark are your friends. Search for "sip dialog" and you'll find all sorts of info on what should be going between your SPA-3000 and  FS.

- Larry 


⁣Get BlueMail for Android ​

On Feb 22, 2021, 12:17, at 12:17, Matthew Macdonald-Wallace <matt at doics.co> wrote:
>Hi all,
>
>In an attempt to add VoIP to my home network, I'm working with
>Freeswitch and FusionPBX.
>
>My outbound connection to the PSTN is via an aging SPA-3000, and whilst
>I can get the SPA-3000 to register correctly as a "user", I can't get
>the "gateway" side of things in FusionPBX/Freeswitch to register with
>the SPA-3000 for outgoing calls.
>
>All of the articles I can find online relate to the SPA-3201 or
>similar,
>which is a very different beast to the SPA-3000.
>
>The error in the logs is as follows:
>
>2021-02-20 08:53:57.680948 [NOTICE] sofia_reg.c:454 Registering
>b1039375-091b-478c-86e4-11a25d3c9d00
>2021-02-20 08:54:00.820948 [ERR] sofia_reg.c:2470 b1039375-091b-478c-
>86e4-11a25d3c9d00 Failed Registration with status Service Unavailable
>[503]. failure #35
>2021-02-20 08:54:01.700955 [WARNING] sofia_reg.c:511
>b1039375-091b-478c-
>86e4-11a25d3c9d00 Failed Registration [503], setting retry to 30
>seconds.
>
>and my settings look like this:
>
>Gateway: outbound
>username: 1900
>password: 1234
>From User: <my landline number>
>From Domain: <BLANK>
>Proxy: <IP ADDRESS OF SPA-3000>:5060
>Realm: <IP ADDRESS OF SPA-3000>
>Expire Seconds: 800
>Retry Seconds: 30
>Distinct To: <BLANK>
>Auth Username: user
>Extension <Blank>
>Register Transport: <BLANK>
>Register Proxy: <IP ADDRESS OF SPA-3000>:5060
>Outbound Proxy: <IP ADDRESS OF SPA-3000>
>Caller ID in From: True
>Suppress CNG: <BLANK>
>Sip CID Type: <BLANK>
>Codec Preferences: <BLANK>
>Extension in Contact: <BLANK>
>Ping: <Blank>
>Ping Min: <Blank>
>Ping Max: <Blank>
>Channels: 1
>Hostname: <Blank>
>Domain: Global
>Context: Public
>Profile: External
>Enabled: True
>Description: Blank
>
>
>I'm sure this is something really obvious that I'm missing in the
>settings, so all help is appreciated!
>
>Thanks in advance,
>
>Matt
>
>
>------------------------------------------------------------------------
>
>_________________________________________________________________________
>
>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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210222/593e89df/attachment-0001.html>

From schoch+freeswitch.org at xwin32.com  Tue Feb 23 21:01:03 2021
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Tue, 23 Feb 2021 13:01:03 -0800
Subject: [Freeswitch-users] New system: sofia is on wrong IP address
Message-ID: <CAOkuKdfqc658HHbhyE=tnQYHhApSdp5PR3eTw4cJ2s4OHGNSgg@mail.gmail.com>

The last time I played with FreeSWITCH was 10 years ago, and now I'm
helping a local nonprofit retire their obsolete analog phone system.

I have a pbx system with 2 network cards: One dedicated to the phones,
plugs into a PoE switch. On that network, the pbx has a DHCP server for the
phones, as well as an FTP server for the Polycom SoundPoint IP 320 phones.
The IP address on that network is 192.168.3.2.

Since I'm setting up the system here at home, the other network card is on
my Xfinity LAN, with an IP address of 10.0.0.167 (via DHCP from the Comcast
modem).

I understand the "domain" should be fairly fixed, so I set that to my
"phone" network:

  <X-PRE-PROCESS cmd="set" data="phone_ip_v4=192.168.3.2"/>

  <X-PRE-PROCESS cmd="set" data="domain=$${phone_ip_v4}"/>

  <X-PRE-PROCESS cmd="set" data="domain_name=eastwest"/>


I defined an sofia profile named "phone" with these lines:

    <param name="rtp-ip" value="$${phone_ip_v4}"/>

    <param name="sip-ip" value="$${phone_ip_v4}"/>

I also used the default "internal" profile, in case an SIP device is hooked
up to the main LAN, but that's not important.
However, a "sofia status" shows this:

                     Name   Type                                       Data
State

=================================================================================================

            external-ipv6 profile   sip:mod_sofia@
[2601:647:4802:9220:c816:30ae:6a9a:d191]:5080 RUNNING (0)

              192.168.3.2   alias                                   internal
ALIASED

                 external profile
sip:mod_sofia at 67.164.101.201:5080 RUNNING
(0)

        external::sip2sip gateway                 sip:eastwest at sip2sip.info
REGED

      external::flowroute gateway
sip:6509889800 at us-west-or.sip-flowroute.com NOREG

                    phone profile
sip:mod_sofia at 10.0.0.167:5060 RUNNING
(0)

            internal-ipv6 profile   sip:mod_sofia@
[2601:647:4802:9220:c816:30ae:6a9a:d191]:5060 RUNNING (0)

                 internal profile
sip:mod_sofia at 67.164.101.201:5060 RUNNING
(0)

=================================================================================================

5 profiles 1 alias

Why does the "phone" profile have "sip:mod_sofia at 10.0.0.167:5060".
Shouldn't that have the 192.168.3.2 address? And why isn't my phone
registering?

-- 
Steve
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210223/85b2e09a/attachment.html>

From botelist at gmail.com  Tue Feb 23 21:35:37 2021
From: botelist at gmail.com (Bote Man)
Date: Tue, 23 Feb 2021 16:35:37 -0500
Subject: [Freeswitch-users] New system: sofia is on wrong IP address
In-Reply-To: <CAOkuKdfqc658HHbhyE=tnQYHhApSdp5PR3eTw4cJ2s4OHGNSgg@mail.gmail.com>
References: <CAOkuKdfqc658HHbhyE=tnQYHhApSdp5PR3eTw4cJ2s4OHGNSgg@mail.gmail.com>
Message-ID: <002e01d70a2b$d3539260$79fab720$@gmail.com>

HO BOY. By “pbx system” may we assume that you mean the linux box running FreeSWITCH?

 

1) First and foremost, it appears that you (inadvertently?) have FreeSWITCH listening on a public interface since sofia status shows both an ipv4 and an ipv6 address. I strongly suggest you eliminate that before attackers attack from those interfaces. Since the FS “Vanilla” configuration files are intended to get you up and running as quickly as possible, they anticipate the most common scenario which is an internal SIP profile to service your phones (named default) and an external SIP profile (named external) that you may or may not need to listen on a different ip:port pair. 

 

I usually rename those files from external.xml to external.hold so they remain as references, but won’t be picked up by FS when it starts since they no longer have the .xml suffix.

 

2) Those two gateway definitions are picked up by an include statement at the top of one of the SIP profiles. I’ve never questioned how or why, but my gateway is included in my internal profile and it works just fine. This is why I say that you might not even need that external profile at all, but you’ll need to investigate that further.

 

3) The SIP profiles as well as all the other XML definitions depend on the “name=” tag at the top, so the filename is irrelevant to FS. The Vanilla configs name the internal profile “default” inside and from your sofia status table it appears that you’ve copied that and named it “phone” so now you have 3 SIP profiles listening, with the internal profile listening on your public IP address.

 

4) The NAT stuff can be tricky, but you probably want to look into that, as well. Here’s a starting point:

https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal

 

I’m not sure I’ve gotten everything right in here, but my advice is to trim it down to keep it simple and then build it back to more complexity as you master each part. 

 

Hope this helps.

 

 

--- 

John Boteler 

BnC Group U.S.A. 

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Steven Schoch
Sent: Tuesday, 23 February, 2021 16:01
To: freeswitch-users <FreeSWITCH-users at lists.freeswitch.org>
Subject: [Freeswitch-users] New system: sofia is on wrong IP address

 

The last time I played with FreeSWITCH was 10 years ago, and now I'm helping a local nonprofit retire their obsolete analog phone system.

 

I have a pbx system with 2 network cards: One dedicated to the phones, plugs into a PoE switch. On that network, the pbx has a DHCP server for the phones, as well as an FTP server for the Polycom SoundPoint IP 320 phones. The IP address on that network is 192.168.3.2.

 

Since I'm setting up the system here at home, the other network card is on my Xfinity LAN, with an IP address of 10.0.0.167 (via DHCP from the Comcast modem).

 

I understand the "domain" should be fairly fixed, so I set that to my "phone" network:

 

  <X-PRE-PROCESS cmd="set" data="phone_ip_v4=192.168.3.2"/>

  <X-PRE-PROCESS cmd="set" data="domain=$${phone_ip_v4}"/>

  <X-PRE-PROCESS cmd="set" data="domain_name=eastwest"/>

 

I defined an sofia profile named "phone" with these lines:

 

    <param name="rtp-ip" value="$${phone_ip_v4}"/>

    <param name="sip-ip" value="$${phone_ip_v4}"/>

 

I also used the default "internal" profile, in case an SIP device is hooked up to the main LAN, but that's not important.

However, a "sofia status" shows this:

                     Name                 Type                                         Data                State

=================================================================================================

            external-ipv6        profile     sip:mod_sofia@[2601:647:4802:9220:c816:30ae:6a9a:d191]:5080        RUNNING (0)

              192.168.3.2           alias                                       internal             ALIASED

                 external              profile           sip:mod_sofia at 67.164.101.201:5080 <http://sip:mod_sofia at 67.164.101.201:5080>             RUNNING (0)

        external::sip2sip      gateway                               sip:eastwest at sip2sip.info <mailto:sip%3Aeastwest at sip2sip.info>           REGED

      external::flowroute   gateway                 sip:6509889800 at us-west-or.sip-flowroute.com <mailto:sip%3A6509889800 at us-west-or.sip-flowroute.com>              NOREG

                    phone               profile               sip:mod_sofia at 10.0.0.167:5060 <http://sip:mod_sofia at 10.0.0.167:5060>   RUNNING (0)

            internal-ipv6         profile     sip:mod_sofia@[2601:647:4802:9220:c816:30ae:6a9a:d191]:5060        RUNNING (0)

                 internal              profile           sip:mod_sofia at 67.164.101.201:5060 <http://sip:mod_sofia at 67.164.101.201:5060>             RUNNING (0)

=================================================================================================

5 profiles 1 alias

 

Why does the "phone" profile have "sip:mod_sofia at 10.0.0.167:5060 <http://sip:mod_sofia at 10.0.0.167:5060> ". Shouldn't that have the 192.168.3.2 address? And why isn't my phone registering?

 

-- 

Steve

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210223/9ae7554d/attachment-0001.html>

From schoch+freeswitch.org at xwin32.com  Tue Feb 23 22:37:04 2021
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Tue, 23 Feb 2021 14:37:04 -0800
Subject: [Freeswitch-users] New system: sofia is on wrong IP address
In-Reply-To: <002e01d70a2b$d3539260$79fab720$@gmail.com>
References: <CAOkuKdfqc658HHbhyE=tnQYHhApSdp5PR3eTw4cJ2s4OHGNSgg@mail.gmail.com>
 <002e01d70a2b$d3539260$79fab720$@gmail.com>
Message-ID: <CAOkuKddpvmtjOUWk4viTf9=Zi+eO0Sxv_2tBVWr8prUviQ25Xw@mail.gmail.com>

Thanks, John!

The system does not have a "public" interface. I assume it gets the 67.*
address from auto-nat. Its 2 IP addresses are 10.0.0.167 (DHCP)
(local_ip_v4), and 192.168.3.2 (phone_ip_v4). I disabled the "internal"
profile and the internal-ipv6 profile to avoid confusion. Now "ss -n -a
sport = 5060" shows this:

Netid         State          Recv-Q          Send-Q                   Local
Address:Port                   Peer Address:Port

udp           UNCONN         0               0
192.168.3.2:5060                        0.0.0.0:*

tcp           LISTEN         0               64
192.168.3.2:5060                        0.0.0.0:*

...which means it's listening on the "phone" network, as it should be.
However "sofia status" still gives me this:

                     Name   Type                                       Data
State

=================================================================================================

            external-ipv6 profile   sip:mod_sofia@
[2601:647:4802:9220:c816:30ae:6a9a:d191]:5080 RUNNING (0)

                 external profile
sip:mod_sofia at 67.164.101.201:5080 RUNNING
(0)

      external::flowroute gateway
sip:6509889800 at us-west-or.sip-flowroute.com NOREG

                    phone profile
sip:mod_sofia at 10.0.0.167:5060 RUNNING
(0)

=================================================================================================

3 profiles 0 aliases

The IP address after "sip:mod_sofia@" is my home LAN. Why isn't it the IP
of the "phone" network?

-- 
Steve


On Tue, Feb 23, 2021 at 1:36 PM Bote Man <botelist at gmail.com> wrote:

> HO BOY. By “pbx system” may we assume that you mean the linux box running
> FreeSWITCH?
>
>
>
> 1) First and foremost, it appears that you (inadvertently?) have
> FreeSWITCH listening on a public interface since sofia status shows both an
> ipv4 and an ipv6 address. I strongly suggest you eliminate that before
> attackers attack from those interfaces. Since the FS “Vanilla”
> configuration files are intended to get you up and running as quickly as
> possible, they anticipate the most common scenario which is an internal SIP
> profile to service your phones (named default) and an external SIP profile
> (named external) that you may or may not need to listen on a different
> ip:port pair.
>
>
>
> I usually rename those files from external.xml to external.hold so they
> remain as references, but won’t be picked up by FS when it starts since
> they no longer have the .xml suffix.
>
>
>
> 2) Those two gateway definitions are picked up by an include statement at
> the top of one of the SIP profiles. I’ve never questioned how or why, but
> my gateway is included in my internal profile and it works just fine. This
> is why I say that you might not even need that external profile at all, but
> you’ll need to investigate that further.
>
>
>
> 3) The SIP profiles as well as all the other XML definitions depend on the
> “name=” tag at the top, so the filename is irrelevant to FS. The Vanilla
> configs name the internal profile “default” inside and from your sofia
> status table it appears that you’ve copied that and named it “phone” so now
> you have 3 SIP profiles listening, with the internal profile listening on
> your public IP address.
>
>
>
> 4) The NAT stuff can be tricky, but you probably want to look into that,
> as well. Here’s a starting point:
>
> https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal
>
>
>
> I’m not sure I’ve gotten everything right in here, but my advice is to
> trim it down to keep it simple and then build it back to more complexity as
> you master each part.
>
>
>
> Hope this helps.
>
>
>
>
>
> ---
>
> John Boteler
>
> BnC Group U.S.A.
>
>
>
>
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Steven Schoch
> *Sent:* Tuesday, 23 February, 2021 16:01
> *To:* freeswitch-users <FreeSWITCH-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] New system: sofia is on wrong IP address
>
>
>
> The last time I played with FreeSWITCH was 10 years ago, and now I'm
> helping a local nonprofit retire their obsolete analog phone system.
>
>
>
> I have a pbx system with 2 network cards: One dedicated to the phones,
> plugs into a PoE switch. On that network, the pbx has a DHCP server for the
> phones, as well as an FTP server for the Polycom SoundPoint IP 320 phones.
> The IP address on that network is 192.168.3.2.
>
>
>
> Since I'm setting up the system here at home, the other network card is on
> my Xfinity LAN, with an IP address of 10.0.0.167 (via DHCP from the Comcast
> modem).
>
>
>
> I understand the "domain" should be fairly fixed, so I set that to my
> "phone" network:
>
>
>
>   <X-PRE-PROCESS cmd="set" data="phone_ip_v4=192.168.3.2"/>
>
>   <X-PRE-PROCESS cmd="set" data="domain=$${phone_ip_v4}"/>
>
>   <X-PRE-PROCESS cmd="set" data="domain_name=eastwest"/>
>
>
>
> I defined an sofia profile named "phone" with these lines:
>
>
>
>     <param name="rtp-ip" value="$${phone_ip_v4}"/>
>
>     <param name="sip-ip" value="$${phone_ip_v4}"/>
>
>
>
> I also used the default "internal" profile, in case an SIP device is
> hooked up to the main LAN, but that's not important.
>
> However, a "sofia status" shows this:
>
>                      Name                 Type
>               Data                State
>
>
> =================================================================================================
>
>             external-ipv6        profile
> sip:mod_sofia@[2601:647:4802:9220:c816:30ae:6a9a:d191]:5080        RUNNING
> (0)
>
>               192.168.3.2           alias
>       internal             ALIASED
>
>                  external              profile
> sip:mod_sofia at 67.164.101.201:5080            RUNNING (0)
>
>         external::sip2sip      gateway
> sip:eastwest at sip2sip.info          REGED
>
>       external::flowroute   gateway
> sip:6509889800 at us-west-or.sip-flowroute.com             NOREG
>
>                     phone               profile
> sip:mod_sofia at 10.0.0.167:5060  RUNNING (0)
>
>             internal-ipv6         profile
> sip:mod_sofia@[2601:647:4802:9220:c816:30ae:6a9a:d191]:5060        RUNNING
> (0)
>
>                  internal              profile
> sip:mod_sofia at 67.164.101.201:5060            RUNNING (0)
>
>
> =================================================================================================
>
> 5 profiles 1 alias
>
>
>
> Why does the "phone" profile have "sip:mod_sofia at 10.0.0.167:5060".
> Shouldn't that have the 192.168.3.2 address? And why isn't my phone
> registering?
>
>
>
> --
>
> Steve
> _________________________________________________________________________
>
> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210223/e5231f80/attachment-0001.html>

From schoch+freeswitch.org at xwin32.com  Wed Feb 24 01:15:27 2021
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Tue, 23 Feb 2021 17:15:27 -0800
Subject: [Freeswitch-users] New system: sofia is on wrong IP address
In-Reply-To: <CAOkuKddpvmtjOUWk4viTf9=Zi+eO0Sxv_2tBVWr8prUviQ25Xw@mail.gmail.com>
References: <CAOkuKdfqc658HHbhyE=tnQYHhApSdp5PR3eTw4cJ2s4OHGNSgg@mail.gmail.com>
 <002e01d70a2b$d3539260$79fab720$@gmail.com>
 <CAOkuKddpvmtjOUWk4viTf9=Zi+eO0Sxv_2tBVWr8prUviQ25Xw@mail.gmail.com>
Message-ID: <CAOkuKdfZWSrz6sSjOJ6Q=4kpaVTbedDkg=H6O_Tr+kPSU1BOSg@mail.gmail.com>

So I tried going to the factory settings. I managed to get a phone to dial
another extension and leave voicemail, but only after pulling the phone off
the PoE network, putting it on the LAN (10.0.0.*), and using a power
supply. This configuration will not be acceptable.

The issue is that I can't get the phone to register to the network on the
2nd network (192.168.3.*). What am I doing wrong?

-- 
Steve

On Tue, Feb 23, 2021 at 2:37 PM Steven Schoch <
schoch+freeswitch.org at xwin32.com> wrote:

> Thanks, John!
>
> The system does not have a "public" interface. I assume it gets the 67.*
> address from auto-nat. Its 2 IP addresses are 10.0.0.167 (DHCP)
> (local_ip_v4), and 192.168.3.2 (phone_ip_v4). I disabled the "internal"
> profile and the internal-ipv6 profile to avoid confusion. Now "ss -n -a
> sport = 5060" shows this:
>
> Netid         State          Recv-Q          Send-Q                   Local
> Address:Port                   Peer Address:Port
>
> udp           UNCONN         0               0
> 192.168.3.2:5060                        0.0.0.0:*
>
> tcp           LISTEN         0               64
> 192.168.3.2:5060                        0.0.0.0:*
>
> ...which means it's listening on the "phone" network, as it should be.
> However "sofia status" still gives me this:
>
>                      Name   Type
> Data State
>
>
> =================================================================================================
>
>             external-ipv6 profile   sip:mod_sofia@
> [2601:647:4802:9220:c816:30ae:6a9a:d191]:5080 RUNNING (0)
>
>                  external profile
> sip:mod_sofia at 67.164.101.201:5080 RUNNING (0)
>
>       external::flowroute gateway
> sip:6509889800 at us-west-or.sip-flowroute.com NOREG
>
>                     phone profile
> sip:mod_sofia at 10.0.0.167:5060 RUNNING (0)
>
>
> =================================================================================================
>
> 3 profiles 0 aliases
>
> The IP address after "sip:mod_sofia@" is my home LAN. Why isn't it the IP
> of the "phone" network?
>
> --
> Steve
>
>
> On Tue, Feb 23, 2021 at 1:36 PM Bote Man <botelist at gmail.com> wrote:
>
>> HO BOY. By “pbx system” may we assume that you mean the linux box running
>> FreeSWITCH?
>>
>>
>>
>> 1) First and foremost, it appears that you (inadvertently?) have
>> FreeSWITCH listening on a public interface since sofia status shows both an
>> ipv4 and an ipv6 address. I strongly suggest you eliminate that before
>> attackers attack from those interfaces. Since the FS “Vanilla”
>> configuration files are intended to get you up and running as quickly as
>> possible, they anticipate the most common scenario which is an internal SIP
>> profile to service your phones (named default) and an external SIP profile
>> (named external) that you may or may not need to listen on a different
>> ip:port pair.
>>
>>
>>
>> I usually rename those files from external.xml to external.hold so they
>> remain as references, but won’t be picked up by FS when it starts since
>> they no longer have the .xml suffix.
>>
>>
>>
>> 2) Those two gateway definitions are picked up by an include statement at
>> the top of one of the SIP profiles. I’ve never questioned how or why, but
>> my gateway is included in my internal profile and it works just fine. This
>> is why I say that you might not even need that external profile at all, but
>> you’ll need to investigate that further.
>>
>>
>>
>> 3) The SIP profiles as well as all the other XML definitions depend on
>> the “name=” tag at the top, so the filename is irrelevant to FS. The
>> Vanilla configs name the internal profile “default” inside and from your
>> sofia status table it appears that you’ve copied that and named it “phone”
>> so now you have 3 SIP profiles listening, with the internal profile
>> listening on your public IP address.
>>
>>
>>
>> 4) The NAT stuff can be tricky, but you probably want to look into that,
>> as well. Here’s a starting point:
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal
>>
>>
>>
>> I’m not sure I’ve gotten everything right in here, but my advice is to
>> trim it down to keep it simple and then build it back to more complexity as
>> you master each part.
>>
>>
>>
>> Hope this helps.
>>
>>
>>
>>
>>
>> ---
>>
>> John Boteler
>>
>> BnC Group U.S.A.
>>
>>
>>
>>
>>
>>
>>
>> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
>> Behalf Of *Steven Schoch
>> *Sent:* Tuesday, 23 February, 2021 16:01
>> *To:* freeswitch-users <FreeSWITCH-users at lists.freeswitch.org>
>> *Subject:* [Freeswitch-users] New system: sofia is on wrong IP address
>>
>>
>>
>> The last time I played with FreeSWITCH was 10 years ago, and now I'm
>> helping a local nonprofit retire their obsolete analog phone system.
>>
>>
>>
>> I have a pbx system with 2 network cards: One dedicated to the phones,
>> plugs into a PoE switch. On that network, the pbx has a DHCP server for the
>> phones, as well as an FTP server for the Polycom SoundPoint IP 320 phones.
>> The IP address on that network is 192.168.3.2.
>>
>>
>>
>> Since I'm setting up the system here at home, the other network card is
>> on my Xfinity LAN, with an IP address of 10.0.0.167 (via DHCP from the
>> Comcast modem).
>>
>>
>>
>> I understand the "domain" should be fairly fixed, so I set that to my
>> "phone" network:
>>
>>
>>
>>   <X-PRE-PROCESS cmd="set" data="phone_ip_v4=192.168.3.2"/>
>>
>>   <X-PRE-PROCESS cmd="set" data="domain=$${phone_ip_v4}"/>
>>
>>   <X-PRE-PROCESS cmd="set" data="domain_name=eastwest"/>
>>
>>
>>
>> I defined an sofia profile named "phone" with these lines:
>>
>>
>>
>>     <param name="rtp-ip" value="$${phone_ip_v4}"/>
>>
>>     <param name="sip-ip" value="$${phone_ip_v4}"/>
>>
>>
>>
>> I also used the default "internal" profile, in case an SIP device is
>> hooked up to the main LAN, but that's not important.
>>
>> However, a "sofia status" shows this:
>>
>>                      Name                 Type
>>               Data                State
>>
>>
>> =================================================================================================
>>
>>             external-ipv6        profile
>> sip:mod_sofia@[2601:647:4802:9220:c816:30ae:6a9a:d191]:5080        RUNNING
>> (0)
>>
>>               192.168.3.2           alias
>>         internal             ALIASED
>>
>>                  external              profile
>> sip:mod_sofia at 67.164.101.201:5080            RUNNING (0)
>>
>>         external::sip2sip      gateway
>> sip:eastwest at sip2sip.info          REGED
>>
>>       external::flowroute   gateway
>> sip:6509889800 at us-west-or.sip-flowroute.com             NOREG
>>
>>                     phone               profile
>> sip:mod_sofia at 10.0.0.167:5060  RUNNING (0)
>>
>>             internal-ipv6         profile
>> sip:mod_sofia@[2601:647:4802:9220:c816:30ae:6a9a:d191]:5060        RUNNING
>> (0)
>>
>>                  internal              profile
>> sip:mod_sofia at 67.164.101.201:5060            RUNNING (0)
>>
>>
>> =================================================================================================
>>
>> 5 profiles 1 alias
>>
>>
>>
>> Why does the "phone" profile have "sip:mod_sofia at 10.0.0.167:5060".
>> Shouldn't that have the 192.168.3.2 address? And why isn't my phone
>> registering?
>>
>>
>>
>> --
>>
>> Steve
>> _________________________________________________________________________
>>
>> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210223/68e48624/attachment-0001.html>

From botelist at gmail.com  Wed Feb 24 01:49:45 2021
From: botelist at gmail.com (Bote Man)
Date: Tue, 23 Feb 2021 20:49:45 -0500
Subject: [Freeswitch-users] New system: sofia is on wrong IP address
In-Reply-To: <CAOkuKddpvmtjOUWk4viTf9=Zi+eO0Sxv_2tBVWr8prUviQ25Xw@mail.gmail.com>
References: <CAOkuKdfqc658HHbhyE=tnQYHhApSdp5PR3eTw4cJ2s4OHGNSgg@mail.gmail.com>
 <002e01d70a2b$d3539260$79fab720$@gmail.com>
 <CAOkuKddpvmtjOUWk4viTf9=Zi+eO0Sxv_2tBVWr8prUviQ25Xw@mail.gmail.com>
Message-ID: <004201d70a4f$5400ed40$fc02c7c0$@gmail.com>

It looks like you still have 3 SIP profiles listening, so I would rename 

external-ipv6.xml -> external-ipv6.hold

external.xml -> external.hold 

just to get them out of the way for now. You can always bring them back in later if necessary.

 

All of these files are merely serving suggestions, you should change things to suit your needs.

 

I would work with internal.xml as your “phone” profile just because it’s intuitive.

 

So, in each SIP profile I would set the desired interface address explicitly.

 

<profile name="phone">

…

    <param name="rtp-ip" value="192.168.3.2"/>

    <param name="sip-ip" value="192.168.3.2"/>

<!--    <param name="ext-rtp-ip" value="auto-nat"/>-->

<!--    <param name="ext-sip-ip" value="auto-nat"/>-->

 

In my own vars.xml it’s using STUN to determine my public IP address, but sofia status only shows FS listening on the private RFC1918 address. It registers just fine with both CallCentric and InterMedia via gateway definitions. If you don’t need FS to listen for remote registrations from outside I don’t think you need the external SIP profile.

 

Anyway, try it just with the one profile and see what sofia says. Once you get the phones to register with FS locally you can expand your horizons to the other network.

 

Bote

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Steven Schoch
Sent: Tuesday, 23 February, 2021 17:37
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] New system: sofia is on wrong IP address

 

Thanks, John!

 

The system does not have a "public" interface. I assume it gets the 67.* address from auto-nat. Its 2 IP addresses are 10.0.0.167 (DHCP) (local_ip_v4), and 192.168.3.2 (phone_ip_v4). I disabled the "internal" profile and the internal-ipv6 profile to avoid confusion. Now "ss -n -a sport = 5060" shows this:

 

Netid         State          Recv-Q          Send-Q                   Local Address:Port                   Peer Address:Port         

udp           UNCONN         0               0                          192.168.3.2:5060 <http://192.168.3.2:5060>                         0.0.0.0:*            

tcp           LISTEN         0               64                         192.168.3.2:5060 <http://192.168.3.2:5060>                         0.0.0.0:*            

 

...which means it's listening on the "phone" network, as it should be. However "sofia status" still gives me this:

 

                     Name                 Type                                         Data                State

=================================================================================================

            external-ipv6        profile     sip:mod_sofia@[2601:647:4802:9220:c816:30ae:6a9a:d191]:5080        RUNNING (0)

                 external              profile           sip:mod_sofia at 67.164.101.201:5080 <http://sip:mod_sofia at 67.164.101.201:5080>             RUNNING (0)

      external::flowroute   gateway                 sip:6509889800 at us-west-or.sip-flowroute.com <mailto:sip%3A6509889800 at us-west-or.sip-flowroute.com>              NOREG

                    phone               profile               sip:mod_sofia at 10.0.0.167:5060 <http://sip:mod_sofia at 10.0.0.167:5060>   RUNNING (0)

=================================================================================================

3 profiles 0 aliases

 

The IP address after "sip:mod_sofia@" is my home LAN. Why isn't it the IP of the "phone" network?

 

-- 

Steve

 

 

On Tue, Feb 23, 2021 at 1:36 PM Bote Man <botelist at gmail.com <mailto:botelist at gmail.com> > wrote:

HO BOY. By “pbx system” may we assume that you mean the linux box running FreeSWITCH?

 

1) First and foremost, it appears that you (inadvertently?) have FreeSWITCH listening on a public interface since sofia status shows both an ipv4 and an ipv6 address. I strongly suggest you eliminate that before attackers attack from those interfaces. Since the FS “Vanilla” configuration files are intended to get you up and running as quickly as possible, they anticipate the most common scenario which is an internal SIP profile to service your phones (named default) and an external SIP profile (named external) that you may or may not need to listen on a different ip:port pair. 

 

I usually rename those files from external.xml to external.hold so they remain as references, but won’t be picked up by FS when it starts since they no longer have the .xml suffix.

 

2) Those two gateway definitions are picked up by an include statement at the top of one of the SIP profiles. I’ve never questioned how or why, but my gateway is included in my internal profile and it works just fine. This is why I say that you might not even need that external profile at all, but you’ll need to investigate that further.

 

3) The SIP profiles as well as all the other XML definitions depend on the “name=” tag at the top, so the filename is irrelevant to FS. The Vanilla configs name the internal profile “default” inside and from your sofia status table it appears that you’ve copied that and named it “phone” so now you have 3 SIP profiles listening, with the internal profile listening on your public IP address.

 

4) The NAT stuff can be tricky, but you probably want to look into that, as well. Here’s a starting point:

https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal

 

I’m not sure I’ve gotten everything right in here, but my advice is to trim it down to keep it simple and then build it back to more complexity as you master each part. 

 

Hope this helps.

 

 

--- 

John Boteler 

BnC Group U.S.A. 

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> > On Behalf Of Steven Schoch
Sent: Tuesday, 23 February, 2021 16:01
To: freeswitch-users <FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org> >
Subject: [Freeswitch-users] New system: sofia is on wrong IP address

 

The last time I played with FreeSWITCH was 10 years ago, and now I'm helping a local nonprofit retire their obsolete analog phone system.

 

I have a pbx system with 2 network cards: One dedicated to the phones, plugs into a PoE switch. On that network, the pbx has a DHCP server for the phones, as well as an FTP server for the Polycom SoundPoint IP 320 phones. The IP address on that network is 192.168.3.2.

 

Since I'm setting up the system here at home, the other network card is on my Xfinity LAN, with an IP address of 10.0.0.167 (via DHCP from the Comcast modem).

 

I understand the "domain" should be fairly fixed, so I set that to my "phone" network:

 

  <X-PRE-PROCESS cmd="set" data="phone_ip_v4=192.168.3.2"/>

  <X-PRE-PROCESS cmd="set" data="domain=$${phone_ip_v4}"/>

  <X-PRE-PROCESS cmd="set" data="domain_name=eastwest"/>

 

I defined an sofia profile named "phone" with these lines:

 

    <param name="rtp-ip" value="$${phone_ip_v4}"/>

    <param name="sip-ip" value="$${phone_ip_v4}"/>

 

I also used the default "internal" profile, in case an SIP device is hooked up to the main LAN, but that's not important.

However, a "sofia status" shows this:

                     Name                 Type                                         Data                State

=================================================================================================

            external-ipv6        profile     sip:mod_sofia@[2601:647:4802:9220:c816:30ae:6a9a:d191]:5080        RUNNING (0)

              192.168.3.2           alias                                       internal             ALIASED

                 external              profile           sip:mod_sofia at 67.164.101.201:5080 <http://sip:mod_sofia at 67.164.101.201:5080>             RUNNING (0)

        external::sip2sip      gateway                               sip:eastwest at sip2sip.info <mailto:sip%3Aeastwest at sip2sip.info>           REGED

      external::flowroute   gateway                 sip:6509889800 at us-west-or.sip-flowroute.com <mailto:sip%3A6509889800 at us-west-or.sip-flowroute.com>              NOREG

                    phone               profile               sip:mod_sofia at 10.0.0.167:5060 <http://sip:mod_sofia at 10.0.0.167:5060>   RUNNING (0)

            internal-ipv6         profile     sip:mod_sofia@[2601:647:4802:9220:c816:30ae:6a9a:d191]:5060        RUNNING (0)

                 internal              profile           sip:mod_sofia at 67.164.101.201:5060 <http://sip:mod_sofia at 67.164.101.201:5060>             RUNNING (0)

=================================================================================================

5 profiles 1 alias

 

Why does the "phone" profile have "sip:mod_sofia at 10.0.0.167:5060 <http://sip:mod_sofia at 10.0.0.167:5060> ". Shouldn't that have the 192.168.3.2 address? And why isn't my phone registering?

 

-- 

Steve

_________________________________________________________________________

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 <mailto: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 <mailto: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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210223/d8736b8d/attachment-0001.html>

From covici at ccs.covici.com  Wed Feb 24 02:38:04 2021
From: covici at ccs.covici.com (John Covici)
Date: Tue, 23 Feb 2021 21:38:04 -0500
Subject: [Freeswitch-users] New system: sofia is on wrong IP address
In-Reply-To: <CAOkuKdfZWSrz6sSjOJ6Q=4kpaVTbedDkg=H6O_Tr+kPSU1BOSg@mail.gmail.com>
References: <CAOkuKdfqc658HHbhyE=tnQYHhApSdp5PR3eTw4cJ2s4OHGNSgg@mail.gmail.com>
 <002e01d70a2b$d3539260$79fab720$@gmail.com>
 <CAOkuKddpvmtjOUWk4viTf9=Zi+eO0Sxv_2tBVWr8prUviQ25Xw@mail.gmail.com>
 <CAOkuKdfZWSrz6sSjOJ6Q=4kpaVTbedDkg=H6O_Tr+kPSU1BOSg@mail.gmail.com>
Message-ID: <m3pn0qb377.wl-covici@ccs.covici.com>

Do you even have connectivity to that network?  Can a packet go from
your phones network to the network where you want the phone to
register?

On Tue, 23 Feb 2021 20:15:27 -0500,
Steven Schoch wrote:
> 
> [1  <multipart/alternative (7bit)>]
> [1.1  <text/plain; UTF-8 (quoted-printable)>]
> So I tried going to the factory settings. I managed to get a phone to dial
> another extension and leave voicemail, but only after pulling the phone off
> the PoE network, putting it on the LAN (10.0.0.*), and using a power
> supply. This configuration will not be acceptable.
> 
> The issue is that I can't get the phone to register to the network on the
> 2nd network (192.168.3.*). What am I doing wrong?
> 
> -- 
> Steve
> 
> On Tue, Feb 23, 2021 at 2:37 PM Steven Schoch <
> schoch+freeswitch.org at xwin32.com> wrote:
> 
> > Thanks, John!
> >
> > The system does not have a "public" interface. I assume it gets the 67.*
> > address from auto-nat. Its 2 IP addresses are 10.0.0.167 (DHCP)
> > (local_ip_v4), and 192.168.3.2 (phone_ip_v4). I disabled the "internal"
> > profile and the internal-ipv6 profile to avoid confusion. Now "ss -n -a
> > sport = 5060" shows this:
> >
> > Netid         State          Recv-Q          Send-Q                   Local
> > Address:Port                   Peer Address:Port
> >
> > udp           UNCONN         0               0
> > 192.168.3.2:5060                        0.0.0.0:*
> >
> > tcp           LISTEN         0               64
> > 192.168.3.2:5060                        0.0.0.0:*
> >
> > ...which means it's listening on the "phone" network, as it should be.
> > However "sofia status" still gives me this:
> >
> >                      Name   Type
> > Data State
> >
> >
> > =================================================================================================
> >
> >             external-ipv6 profile   sip:mod_sofia@
> > [2601:647:4802:9220:c816:30ae:6a9a:d191]:5080 RUNNING (0)
> >
> >                  external profile
> > sip:mod_sofia at 67.164.101.201:5080 RUNNING (0)
> >
> >       external::flowroute gateway
> > sip:6509889800 at us-west-or.sip-flowroute.com NOREG
> >
> >                     phone profile
> > sip:mod_sofia at 10.0.0.167:5060 RUNNING (0)
> >
> >
> > =================================================================================================
> >
> > 3 profiles 0 aliases
> >
> > The IP address after "sip:mod_sofia@" is my home LAN. Why isn't it the IP
> > of the "phone" network?
> >
> > --
> > Steve
> >
> >
> > On Tue, Feb 23, 2021 at 1:36 PM Bote Man <botelist at gmail.com> wrote:
> >
> >> HO BOY. By “pbx system” may we assume that you mean the linux box running
> >> FreeSWITCH?
> >>
> >>
> >>
> >> 1) First and foremost, it appears that you (inadvertently?) have
> >> FreeSWITCH listening on a public interface since sofia status shows both an
> >> ipv4 and an ipv6 address. I strongly suggest you eliminate that before
> >> attackers attack from those interfaces. Since the FS “Vanilla”
> >> configuration files are intended to get you up and running as quickly as
> >> possible, they anticipate the most common scenario which is an internal SIP
> >> profile to service your phones (named default) and an external SIP profile
> >> (named external) that you may or may not need to listen on a different
> >> ip:port pair.
> >>
> >>
> >>
> >> I usually rename those files from external.xml to external.hold so they
> >> remain as references, but won’t be picked up by FS when it starts since
> >> they no longer have the .xml suffix.
> >>
> >>
> >>
> >> 2) Those two gateway definitions are picked up by an include statement at
> >> the top of one of the SIP profiles. I’ve never questioned how or why, but
> >> my gateway is included in my internal profile and it works just fine. This
> >> is why I say that you might not even need that external profile at all, but
> >> you’ll need to investigate that further.
> >>
> >>
> >>
> >> 3) The SIP profiles as well as all the other XML definitions depend on
> >> the “name=” tag at the top, so the filename is irrelevant to FS. The
> >> Vanilla configs name the internal profile “default” inside and from your
> >> sofia status table it appears that you’ve copied that and named it “phone”
> >> so now you have 3 SIP profiles listening, with the internal profile
> >> listening on your public IP address.
> >>
> >>
> >>
> >> 4) The NAT stuff can be tricky, but you probably want to look into that,
> >> as well. Here’s a starting point:
> >>
> >> https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal
> >>
> >>
> >>
> >> I’m not sure I’ve gotten everything right in here, but my advice is to
> >> trim it down to keep it simple and then build it back to more complexity as
> >> you master each part.
> >>
> >>
> >>
> >> Hope this helps.
> >>
> >>
> >>
> >>
> >>
> >> ---
> >>
> >> John Boteler
> >>
> >> BnC Group U.S.A.
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> >> Behalf Of *Steven Schoch
> >> *Sent:* Tuesday, 23 February, 2021 16:01
> >> *To:* freeswitch-users <FreeSWITCH-users at lists.freeswitch.org>
> >> *Subject:* [Freeswitch-users] New system: sofia is on wrong IP address
> >>
> >>
> >>
> >> The last time I played with FreeSWITCH was 10 years ago, and now I'm
> >> helping a local nonprofit retire their obsolete analog phone system.
> >>
> >>
> >>
> >> I have a pbx system with 2 network cards: One dedicated to the phones,
> >> plugs into a PoE switch. On that network, the pbx has a DHCP server for the
> >> phones, as well as an FTP server for the Polycom SoundPoint IP 320 phones.
> >> The IP address on that network is 192.168.3.2.
> >>
> >>
> >>
> >> Since I'm setting up the system here at home, the other network card is
> >> on my Xfinity LAN, with an IP address of 10.0.0.167 (via DHCP from the
> >> Comcast modem).
> >>
> >>
> >>
> >> I understand the "domain" should be fairly fixed, so I set that to my
> >> "phone" network:
> >>
> >>
> >>
> >>   <X-PRE-PROCESS cmd="set" data="phone_ip_v4=192.168.3.2"/>
> >>
> >>   <X-PRE-PROCESS cmd="set" data="domain=$${phone_ip_v4}"/>
> >>
> >>   <X-PRE-PROCESS cmd="set" data="domain_name=eastwest"/>
> >>
> >>
> >>
> >> I defined an sofia profile named "phone" with these lines:
> >>
> >>
> >>
> >>     <param name="rtp-ip" value="$${phone_ip_v4}"/>
> >>
> >>     <param name="sip-ip" value="$${phone_ip_v4}"/>
> >>
> >>
> >>
> >> I also used the default "internal" profile, in case an SIP device is
> >> hooked up to the main LAN, but that's not important.
> >>
> >> However, a "sofia status" shows this:
> >>
> >>                      Name                 Type
> >>               Data                State
> >>
> >>
> >> =================================================================================================
> >>
> >>             external-ipv6        profile
> >> sip:mod_sofia@[2601:647:4802:9220:c816:30ae:6a9a:d191]:5080        RUNNING
> >> (0)
> >>
> >>               192.168.3.2           alias
> >>         internal             ALIASED
> >>
> >>                  external              profile
> >> sip:mod_sofia at 67.164.101.201:5080            RUNNING (0)
> >>
> >>         external::sip2sip      gateway
> >> sip:eastwest at sip2sip.info          REGED
> >>
> >>       external::flowroute   gateway
> >> sip:6509889800 at us-west-or.sip-flowroute.com             NOREG
> >>
> >>                     phone               profile
> >> sip:mod_sofia at 10.0.0.167:5060  RUNNING (0)
> >>
> >>             internal-ipv6         profile
> >> sip:mod_sofia@[2601:647:4802:9220:c816:30ae:6a9a:d191]:5060        RUNNING
> >> (0)
> >>
> >>                  internal              profile
> >> sip:mod_sofia at 67.164.101.201:5060            RUNNING (0)
> >>
> >>
> >> =================================================================================================
> >>
> >> 5 profiles 1 alias
> >>
> >>
> >>
> >> Why does the "phone" profile have "sip:mod_sofia at 10.0.0.167:5060".
> >> Shouldn't that have the 192.168.3.2 address? And why isn't my phone
> >> registering?
> >>
> >>
> >>
> >> --
> >>
> >> Steve
> >> _________________________________________________________________________
> >>
> >> 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
> >
> >
> [1.2  <text/html; UTF-8 (quoted-printable)>]
> [2  <text/plain; utf-8 (base64)>]
> _________________________________________________________________________
> 
> 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

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici wb2una
         covici at ccs.covici.com


From schoch+freeswitch.org at xwin32.com  Wed Feb 24 07:11:34 2021
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Tue, 23 Feb 2021 23:11:34 -0800
Subject: [Freeswitch-users] New system: sofia is on wrong IP address
In-Reply-To: <004201d70a4f$5400ed40$fc02c7c0$@gmail.com>
References: <CAOkuKdfqc658HHbhyE=tnQYHhApSdp5PR3eTw4cJ2s4OHGNSgg@mail.gmail.com>
 <002e01d70a2b$d3539260$79fab720$@gmail.com>
 <CAOkuKddpvmtjOUWk4viTf9=Zi+eO0Sxv_2tBVWr8prUviQ25Xw@mail.gmail.com>
 <004201d70a4f$5400ed40$fc02c7c0$@gmail.com>
Message-ID: <CAOkuKdcKrKwTk6DKH3cPx0fEfSGSrypb6SNUfZJw16bWoxwzOg@mail.gmail.com>

Thanks, that helped.
My problem was I had set:

<param name="ext-rtp-ip" value="auto"/>

<param name="ext-sip-ip" value="auto"/>


This was grabbing the IP address 10.0.0.167. After removing those lines,
sofia status now includes:

                     Name   Type                                       Data
State

=================================================================================================

                    phone profile
sip:mod_sofia at 192.168.3.2:5060 RUNNING
(0)


... which is what I expected.


To answer John's question:


Interface enp2s0 of the Freeswitch system has IP address 192.168.3.2. That
interface plugs into the PoE switch that all the phones will connect. The
Freeswitch system also runs DHCP and FTP services on that interface in
order to provision the SoundPoint IP 320 phones. So yes, they connect
directly.


I just got the phone to register. Now I need to figure out how to get it to
dial out through Flowroute.


-- 

Steve



On Tue, Feb 23, 2021 at 5:50 PM Bote Man <botelist at gmail.com> wrote:

> It looks like you still have 3 SIP profiles listening, so I would rename
>
> external-ipv6.xml -> external-ipv6.hold
>
> external.xml -> external.hold
>
> just to get them out of the way for now. You can always bring them back in
> later if necessary.
>
>
>
> All of these files are merely serving suggestions, you should change
> things to suit your needs.
>
>
>
> I would work with internal.xml as your “phone” profile just because it’s
> intuitive.
>
>
>
> So, in each SIP profile I would set the desired interface address
> explicitly.
>
>
>
> <profile name="phone">
>
> …
>
>     <param name="rtp-ip" value="192.168.3.2"/>
>
>     <param name="sip-ip" value="192.168.3.2"/>
>
> <!--    <param name="ext-rtp-ip" value="auto-nat"/>-->
>
> <!--    <param name="ext-sip-ip" value="auto-nat"/>-->
>
>
>
> In my own vars.xml it’s using STUN to determine my public IP address, but
> sofia status only shows FS listening on the private RFC1918 address. It
> registers just fine with both CallCentric and InterMedia via gateway
> definitions. If you don’t need FS to listen for remote registrations from
> outside I don’t think you need the external SIP profile.
>
>
>
> Anyway, try it just with the one profile and see what sofia says. Once you
> get the phones to register with FS locally you can expand your horizons to
> the other network.
>
>
>
> Bote
>
>
>
>
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Steven Schoch
> *Sent:* Tuesday, 23 February, 2021 17:37
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] New system: sofia is on wrong IP address
>
>
>
> Thanks, John!
>
>
>
> The system does not have a "public" interface. I assume it gets the 67.*
> address from auto-nat. Its 2 IP addresses are 10.0.0.167 (DHCP)
> (local_ip_v4), and 192.168.3.2 (phone_ip_v4). I disabled the "internal"
> profile and the internal-ipv6 profile to avoid confusion. Now "ss -n -a
> sport = 5060" shows this:
>
>
>
> Netid         State          Recv-Q          Send-Q                   Local
> Address:Port                   Peer Address:Port
>
> udp           UNCONN         0               0
> 192.168.3.2:5060                        0.0.0.0:*
>
> tcp           LISTEN         0               64
> 192.168.3.2:5060                        0.0.0.0:*
>
>
>
> ...which means it's listening on the "phone" network, as it should be.
> However "sofia status" still gives me this:
>
>
>
>                      Name                 Type
>               Data                State
>
>
> =================================================================================================
>
>             external-ipv6        profile
> sip:mod_sofia@[2601:647:4802:9220:c816:30ae:6a9a:d191]:5080        RUNNING
> (0)
>
>                  external              profile
> sip:mod_sofia at 67.164.101.201:5080            RUNNING (0)
>
>       external::flowroute   gateway
> sip:6509889800 at us-west-or.sip-flowroute.com             NOREG
>
>                     phone               profile
> sip:mod_sofia at 10.0.0.167:5060  RUNNING (0)
>
>
> =================================================================================================
>
> 3 profiles 0 aliases
>
>
>
> The IP address after "sip:mod_sofia@" is my home LAN. Why isn't it the IP
> of the "phone" network?
>
>
>
> --
>
> Steve
>
>
>
>
>
> On Tue, Feb 23, 2021 at 1:36 PM Bote Man <botelist at gmail.com> wrote:
>
> HO BOY. By “pbx system” may we assume that you mean the linux box running
> FreeSWITCH?
>
>
>
> 1) First and foremost, it appears that you (inadvertently?) have
> FreeSWITCH listening on a public interface since sofia status shows both an
> ipv4 and an ipv6 address. I strongly suggest you eliminate that before
> attackers attack from those interfaces. Since the FS “Vanilla”
> configuration files are intended to get you up and running as quickly as
> possible, they anticipate the most common scenario which is an internal SIP
> profile to service your phones (named default) and an external SIP profile
> (named external) that you may or may not need to listen on a different
> ip:port pair.
>
>
>
> I usually rename those files from external.xml to external.hold so they
> remain as references, but won’t be picked up by FS when it starts since
> they no longer have the .xml suffix.
>
>
>
> 2) Those two gateway definitions are picked up by an include statement at
> the top of one of the SIP profiles. I’ve never questioned how or why, but
> my gateway is included in my internal profile and it works just fine. This
> is why I say that you might not even need that external profile at all, but
> you’ll need to investigate that further.
>
>
>
> 3) The SIP profiles as well as all the other XML definitions depend on the
> “name=” tag at the top, so the filename is irrelevant to FS. The Vanilla
> configs name the internal profile “default” inside and from your sofia
> status table it appears that you’ve copied that and named it “phone” so now
> you have 3 SIP profiles listening, with the internal profile listening on
> your public IP address.
>
>
>
> 4) The NAT stuff can be tricky, but you probably want to look into that,
> as well. Here’s a starting point:
>
> https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal
>
>
>
> I’m not sure I’ve gotten everything right in here, but my advice is to
> trim it down to keep it simple and then build it back to more complexity as
> you master each part.
>
>
>
> Hope this helps.
>
>
>
>
>
> ---
>
> John Boteler
>
> BnC Group U.S.A.
>
>
>
>
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Steven Schoch
> *Sent:* Tuesday, 23 February, 2021 16:01
> *To:* freeswitch-users <FreeSWITCH-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] New system: sofia is on wrong IP address
>
>
>
> The last time I played with FreeSWITCH was 10 years ago, and now I'm
> helping a local nonprofit retire their obsolete analog phone system.
>
>
>
> I have a pbx system with 2 network cards: One dedicated to the phones,
> plugs into a PoE switch. On that network, the pbx has a DHCP server for the
> phones, as well as an FTP server for the Polycom SoundPoint IP 320 phones.
> The IP address on that network is 192.168.3.2.
>
>
>
> Since I'm setting up the system here at home, the other network card is on
> my Xfinity LAN, with an IP address of 10.0.0.167 (via DHCP from the Comcast
> modem).
>
>
>
> I understand the "domain" should be fairly fixed, so I set that to my
> "phone" network:
>
>
>
>   <X-PRE-PROCESS cmd="set" data="phone_ip_v4=192.168.3.2"/>
>
>   <X-PRE-PROCESS cmd="set" data="domain=$${phone_ip_v4}"/>
>
>   <X-PRE-PROCESS cmd="set" data="domain_name=eastwest"/>
>
>
>
> I defined an sofia profile named "phone" with these lines:
>
>
>
>     <param name="rtp-ip" value="$${phone_ip_v4}"/>
>
>     <param name="sip-ip" value="$${phone_ip_v4}"/>
>
>
>
> I also used the default "internal" profile, in case an SIP device is
> hooked up to the main LAN, but that's not important.
>
> However, a "sofia status" shows this:
>
>                      Name                 Type
>               Data                State
>
>
> =================================================================================================
>
>             external-ipv6        profile
> sip:mod_sofia@[2601:647:4802:9220:c816:30ae:6a9a:d191]:5080        RUNNING
> (0)
>
>               192.168.3.2           alias
>       internal             ALIASED
>
>                  external              profile
> sip:mod_sofia at 67.164.101.201:5080            RUNNING (0)
>
>         external::sip2sip      gateway
> sip:eastwest at sip2sip.info          REGED
>
>       external::flowroute   gateway
> sip:6509889800 at us-west-or.sip-flowroute.com             NOREG
>
>                     phone               profile
> sip:mod_sofia at 10.0.0.167:5060  RUNNING (0)
>
>             internal-ipv6         profile
> sip:mod_sofia@[2601:647:4802:9220:c816:30ae:6a9a:d191]:5060        RUNNING
> (0)
>
>                  internal              profile
> sip:mod_sofia at 67.164.101.201:5060            RUNNING (0)
>
>
> =================================================================================================
>
> 5 profiles 1 alias
>
>
>
> Why does the "phone" profile have "sip:mod_sofia at 10.0.0.167:5060".
> Shouldn't that have the 192.168.3.2 address? And why isn't my phone
> registering?
>
>
>
> --
>
> Steve
>
> _________________________________________________________________________
>
> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210223/29041277/attachment-0001.html>

From dragos at freeswitch.org  Wed Feb 24 08:31:21 2021
From: dragos at freeswitch.org (Dragos Oancea)
Date: Wed, 24 Feb 2021 10:31:21 +0200
Subject: [Freeswitch-users] DTMF not passing from the A Leg to the B Leg
In-Reply-To: <CABb=kMcK7jeMgAMpHR=sPe3vJpgj_Rp5BttaSLj3xappyTWE_g@mail.gmail.com>
References: <CABb=kMcK7jeMgAMpHR=sPe3vJpgj_Rp5BttaSLj3xappyTWE_g@mail.gmail.com>
Message-ID: <CAGvJV4qw3Wdm2fCfNKWKx0rrbQq4DNLb56zc5vmfdUqA4VR5FQ@mail.gmail.com>

Try setting channel variable "sensitive_dtmf" to true.

On Mon, Feb 22, 2021 at 7:17 PM Keith Croxford <keithxcroxford at gmail.com>
wrote:

> I'm coming across a strange, intermittent behavior with RFC2833 DTMF
> packets. This is occurring in a production environment, and I've been able
> to duplicate this with a very minimal FreeSWITCH installation in a virtual
> machine.
>
> Details of the problem:
>
> A customer occasionally has a problem report where their callers are
> unable to use the IVR menu. From the user's standpoint, the digits are not
> detected. Upon further review, we do receive DTMF event packets from the
> carrier, however, they do not traverse the bridge from the A leg to the B
> leg.
>
> From the console logs, I see the following with the first digit.
> --
> 2021-02-20 11:28:30.842108 [DEBUG] mod_sofia.c:645 SOFIA EXCHANGE_MEDIA
> 2021-02-20 11:28:30.881612 [DEBUG] switch_rtp.c:1890 rtcp_stats_init:
> audio ssrc[1488669269] base_seq[26830]
> 2021-02-20 11:28:30.933002 [DEBUG] switch_rtp.c:7550 Correct audio ip/port
> confirmed.
> 2021-02-20 11:28:34.881612 [DEBUG] switch_rtp.c:7550 Correct audio ip/port
> confirmed.
> 2021-02-20 11:28:34.923555 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 1:400
> 2021-02-20 11:28:34.923555 [INFO] switch_channel.c:515 RECV DTMF 1:400
> 2021-02-20 11:28:34.923555 [DEBUG] switch_rtp.c:5424 Send start packet for
> [1] ts=160 dur=160/160/400 seq=51938 lw=160
> 2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5323 Send middle packet
> for [1] ts=160 dur=320/320/400 seq=51939 lw=320
> 2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5323 Send end packet for
> [1] ts=160 dur=480/480/400 seq=51940 lw=320
> 2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5323 Send end packet for
> [1] ts=160 dur=480/480/400 seq=51941 lw=320
> 2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5323 Send end packet for
> [1] ts=160 dur=480/480/400 seq=51942 lw=320
> 2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5271 Queue digit delay of
> 40ms
> 2021-02-20 11:28:35.421487 [DEBUG] switch_rtp.c:6985 Correct audio RTCP
> ip/port confirmed.
> --
>
> However, on the following series of digits, they are detected as seen in
> the logs from switch_rtp.c and switch_channel.c. However, they do not
> proceed to the "Send start packet" step as seen with the previous digit.
> This mirrors what I see on the B leg client, as it only detects the first
> digit.
>
> --
> 2021-02-20 11:28:37.521489 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 1:400
> 2021-02-20 11:28:37.521489 [INFO] switch_channel.c:515 RECV DTMF 1:400
> 2021-02-20 11:28:46.082031 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 3:400
> 2021-02-20 11:28:46.082031 [INFO] switch_channel.c:515 RECV DTMF 3:400
> 2021-02-20 11:28:46.481377 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 6:400
> 2021-02-20 11:28:46.481377 [INFO] switch_channel.c:515 RECV DTMF 6:400
> 2021-02-20 11:28:47.061795 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 4:400
> 2021-02-20 11:28:47.061795 [INFO] switch_channel.c:515 RECV DTMF 4:400
> 2021-02-20 11:28:47.461504 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 5:400
> 2021-02-20 11:28:47.461504 [INFO] switch_channel.c:515 RECV DTMF 5:400
> 2021-02-20 11:28:47.881346 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 0:400
> 2021-02-20 11:28:47.881346 [INFO] switch_channel.c:515 RECV DTMF 0:400
> 2021-02-20 11:28:48.281220 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 0:400
> 2021-02-20 11:28:48.281220 [INFO] switch_channel.c:515 RECV DTMF 0:400
> 2021-02-20 11:28:48.805572 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF #:400
> 2021-02-20 11:28:48.805572 [INFO] switch_channel.c:515 RECV DTMF #:400
> 2021-02-20 11:29:04.881294 [NOTICE] sofia.c:1079 Hangup sofia/external/
> nobody at 192.168.1.71 [CS_EXECUTE] [NORMAL_CLEARING]
> --
>
> Yet, they do arrive on the Freeswitch instances. I've attempted to
> add <action application="set"
> data="rtp_manual_rtp_bugs=ignore_dtmf_duration"/>  to my inbound dialplan
> xml. I'm leaning toward these being a "weird" DTMF packet and not a
> FreeSWITCH issue, but I'm curious if anyone has encountered a similar
> problem, or has advice of what we could test on the FS side.
> --
> Supporting Details:
> Test Environment:
> I took a packet capture from production and filtered it down to the A
> leg's RFC2833 packets, and saved this to a new file. I then used this pcap
> file in a SIPP UAC scenario which is acting as the A Leg.
>
> Freeswitch sits in the middle and I've configured it following this guide (
> https://freeswitch.org/confluence/display/FREESWITCH/SBC+FreeSWITCH+Configuration+Example+2)
> as a barebones SBC. Freeswitch is version 1.8.3
>
> The B Leg is a simple pjsua client configured to answer an inbound call.
> --
> Best Regards,
>
> KC
>
>
>
>
>
> _________________________________________________________________________
>
> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210224/c856aaa0/attachment.html>

From botelist at gmail.com  Wed Feb 24 15:56:32 2021
From: botelist at gmail.com (Bote Man)
Date: Wed, 24 Feb 2021 10:56:32 -0500
Subject: [Freeswitch-users] DTMF not passing from the A Leg to the B Leg
In-Reply-To: <CAGvJV4qw3Wdm2fCfNKWKx0rrbQq4DNLb56zc5vmfdUqA4VR5FQ@mail.gmail.com>
References: <CABb=kMcK7jeMgAMpHR=sPe3vJpgj_Rp5BttaSLj3xappyTWE_g@mail.gmail.com>
 <CAGvJV4qw3Wdm2fCfNKWKx0rrbQq4DNLb56zc5vmfdUqA4VR5FQ@mail.gmail.com>
Message-ID: <001601d70ac5$9f5d83e0$de188ba0$@gmail.com>

Well, that’s a new one!

 

I’d be happy to document it in the wiki if someone can provide a useful description of what it controls.

 

TNX.

 

 

--- 

John Boteler 

BnC Group U.S.A. 

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Dragos Oancea
Sent: Wednesday, 24 February, 2021 03:31
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] DTMF not passing from the A Leg to the B Leg

 

Try setting channel variable "sensitive_dtmf" to true.

 

On Mon, Feb 22, 2021 at 7:17 PM Keith Croxford <keithxcroxford at gmail.com <mailto:keithxcroxford at gmail.com> > wrote:

I'm coming across a strange, intermittent behavior with RFC2833 DTMF packets. This is occurring in a production environment, and I've been able to duplicate this with a very minimal FreeSWITCH installation in a virtual machine. 

Details of the problem: 


A customer occasionally has a problem report where their callers are unable to use the IVR menu. From the user's standpoint, the digits are not detected. Upon further review, we do receive DTMF event packets from the carrier, however, they do not traverse the bridge from the A leg to the B leg.  

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210224/23145d47/attachment.html>

From botelist at gmail.com  Wed Feb 24 16:07:32 2021
From: botelist at gmail.com (Bote Man)
Date: Wed, 24 Feb 2021 11:07:32 -0500
Subject: [Freeswitch-users] New system: sofia is on wrong IP address
In-Reply-To: <CAOkuKdcKrKwTk6DKH3cPx0fEfSGSrypb6SNUfZJw16bWoxwzOg@mail.gmail.com>
References: <CAOkuKdfqc658HHbhyE=tnQYHhApSdp5PR3eTw4cJ2s4OHGNSgg@mail.gmail.com>
 <002e01d70a2b$d3539260$79fab720$@gmail.com>
 <CAOkuKddpvmtjOUWk4viTf9=Zi+eO0Sxv_2tBVWr8prUviQ25Xw@mail.gmail.com>
 <004201d70a4f$5400ed40$fc02c7c0$@gmail.com>
 <CAOkuKdcKrKwTk6DKH3cPx0fEfSGSrypb6SNUfZJw16bWoxwzOg@mail.gmail.com>
Message-ID: <001b01d70ac7$290545a0$7b0fd0e0$@gmail.com>

If the include statement lies in your external.xml profile definition, then that profile will pick up the gateway definitions in the external subdirectory. They seemed to work before so they should work again. If they register it’s just a matter of sending them the digits that keep them happy and you should be close to a working system.

 

<profile name="external">

  <gateways>

    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>

  </gateways> 

 

Just be sure that your external.xml SIP profile doesn’t gum up the works again with other undesired settings. Change one thing at a time and observe the results so that you know what change caused what behavior.

 

 

--- 

John Boteler 

BnC Group U.S.A. 

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Steven Schoch
Sent: Wednesday, 24 February, 2021 02:12
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] New system: sofia is on wrong IP address

 

Thanks, that helped.

My problem was I had set:

 

<param name="ext-rtp-ip" value="auto"/>

<param name="ext-sip-ip" value="auto"/>

 

This was grabbing the IP address 10.0.0.167. After removing those lines, sofia status now includes:

                     Name                 Type                                         Data                State

=================================================================================================

                    phone               profile               sip:mod_sofia at 192.168.3.2:5060 <http://sip:mod_sofia at 192.168.3.2:5060>  RUNNING (0)

 

... which is what I expected.

 

To answer John's question:

 

Interface enp2s0 of the Freeswitch system has IP address 192.168.3.2. That interface plugs into the PoE switch that all the phones will connect. The Freeswitch system also runs DHCP and FTP services on that interface in order to provision the SoundPoint IP 320 phones. So yes, they connect directly.

 

I just got the phone to register. Now I need to figure out how to get it to dial out through Flowroute.

 

-- 

Steve

 

 

On Tue, Feb 23, 2021 at 5:50 PM Bote Man <botelist at gmail.com <mailto:botelist at gmail.com> > wrote:

It looks like you still have 3 SIP profiles listening, so I would rename 

external-ipv6.xml -> external-ipv6.hold

external.xml -> external.hold 

just to get them out of the way for now. You can always bring them back in later if necessary.

 

All of these files are merely serving suggestions, you should change things to suit your needs.

 

I would work with internal.xml as your “phone” profile just because it’s intuitive.

 

So, in each SIP profile I would set the desired interface address explicitly.

 

<profile name="phone">

…

    <param name="rtp-ip" value="192.168.3.2"/>

    <param name="sip-ip" value="192.168.3.2"/>

<!--    <param name="ext-rtp-ip" value="auto-nat"/>-->

<!--    <param name="ext-sip-ip" value="auto-nat"/>-->

 

In my own vars.xml it’s using STUN to determine my public IP address, but sofia status only shows FS listening on the private RFC1918 address. It registers just fine with both CallCentric and InterMedia via gateway definitions. If you don’t need FS to listen for remote registrations from outside I don’t think you need the external SIP profile.

 

Anyway, try it just with the one profile and see what sofia says. Once you get the phones to register with FS locally you can expand your horizons to the other network.

 

Bote

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> > On Behalf Of Steven Schoch
Sent: Tuesday, 23 February, 2021 17:37
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: Re: [Freeswitch-users] New system: sofia is on wrong IP address

 

Thanks, John!

 

The system does not have a "public" interface. I assume it gets the 67.* address from auto-nat. Its 2 IP addresses are 10.0.0.167 (DHCP) (local_ip_v4), and 192.168.3.2 (phone_ip_v4). I disabled the "internal" profile and the internal-ipv6 profile to avoid confusion. Now "ss -n -a sport = 5060" shows this:

 

Netid         State          Recv-Q          Send-Q                   Local Address:Port                   Peer Address:Port         

udp           UNCONN         0               0                          192.168.3.2:5060 <http://192.168.3.2:5060>                         0.0.0.0:*            

tcp           LISTEN         0               64                         192.168.3.2:5060 <http://192.168.3.2:5060>                         0.0.0.0:*            

 

...which means it's listening on the "phone" network, as it should be. However "sofia status" still gives me this:

 

                     Name                 Type                                         Data                State

=================================================================================================

            external-ipv6        profile     sip:mod_sofia@[2601:647:4802:9220:c816:30ae:6a9a:d191]:5080        RUNNING (0)

                 external              profile           sip:mod_sofia at 67.164.101.201:5080 <http://sip:mod_sofia at 67.164.101.201:5080>             RUNNING (0)

      external::flowroute   gateway                 sip:6509889800 at us-west-or.sip-flowroute.com <mailto:sip%3A6509889800 at us-west-or.sip-flowroute.com>              NOREG

                    phone               profile               sip:mod_sofia at 10.0.0.167:5060 <http://sip:mod_sofia at 10.0.0.167:5060>   RUNNING (0)

=================================================================================================

3 profiles 0 aliases

 

The IP address after "sip:mod_sofia@" is my home LAN. Why isn't it the IP of the "phone" network?

 

-- 

Steve

 

 

On Tue, Feb 23, 2021 at 1:36 PM Bote Man <botelist at gmail.com <mailto:botelist at gmail.com> > wrote:

HO BOY. By “pbx system” may we assume that you mean the linux box running FreeSWITCH?

 

1) First and foremost, it appears that you (inadvertently?) have FreeSWITCH listening on a public interface since sofia status shows both an ipv4 and an ipv6 address. I strongly suggest you eliminate that before attackers attack from those interfaces. Since the FS “Vanilla” configuration files are intended to get you up and running as quickly as possible, they anticipate the most common scenario which is an internal SIP profile to service your phones (named default) and an external SIP profile (named external) that you may or may not need to listen on a different ip:port pair. 

 

I usually rename those files from external.xml to external.hold so they remain as references, but won’t be picked up by FS when it starts since they no longer have the .xml suffix.

 

2) Those two gateway definitions are picked up by an include statement at the top of one of the SIP profiles. I’ve never questioned how or why, but my gateway is included in my internal profile and it works just fine. This is why I say that you might not even need that external profile at all, but you’ll need to investigate that further.

 

3) The SIP profiles as well as all the other XML definitions depend on the “name=” tag at the top, so the filename is irrelevant to FS. The Vanilla configs name the internal profile “default” inside and from your sofia status table it appears that you’ve copied that and named it “phone” so now you have 3 SIP profiles listening, with the internal profile listening on your public IP address.

 

4) The NAT stuff can be tricky, but you probably want to look into that, as well. Here’s a starting point:

https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal

 

I’m not sure I’ve gotten everything right in here, but my advice is to trim it down to keep it simple and then build it back to more complexity as you master each part. 

 

Hope this helps.

 

 

--- 

John Boteler 

BnC Group U.S.A. 

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> > On Behalf Of Steven Schoch
Sent: Tuesday, 23 February, 2021 16:01
To: freeswitch-users <FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org> >
Subject: [Freeswitch-users] New system: sofia is on wrong IP address

 

The last time I played with FreeSWITCH was 10 years ago, and now I'm helping a local nonprofit retire their obsolete analog phone system.

 

I have a pbx system with 2 network cards: One dedicated to the phones, plugs into a PoE switch. On that network, the pbx has a DHCP server for the phones, as well as an FTP server for the Polycom SoundPoint IP 320 phones. The IP address on that network is 192.168.3.2.

 

Since I'm setting up the system here at home, the other network card is on my Xfinity LAN, with an IP address of 10.0.0.167 (via DHCP from the Comcast modem).

 

I understand the "domain" should be fairly fixed, so I set that to my "phone" network:

 

  <X-PRE-PROCESS cmd="set" data="phone_ip_v4=192.168.3.2"/>

  <X-PRE-PROCESS cmd="set" data="domain=$${phone_ip_v4}"/>

  <X-PRE-PROCESS cmd="set" data="domain_name=eastwest"/>

 

I defined an sofia profile named "phone" with these lines:

 

    <param name="rtp-ip" value="$${phone_ip_v4}"/>

    <param name="sip-ip" value="$${phone_ip_v4}"/>

 

I also used the default "internal" profile, in case an SIP device is hooked up to the main LAN, but that's not important.

However, a "sofia status" shows this:

                     Name                 Type                                         Data                State

=================================================================================================

            external-ipv6        profile     sip:mod_sofia@[2601:647:4802:9220:c816:30ae:6a9a:d191]:5080        RUNNING (0)

              192.168.3.2           alias                                       internal             ALIASED

                 external              profile           sip:mod_sofia at 67.164.101.201:5080 <http://sip:mod_sofia at 67.164.101.201:5080>             RUNNING (0)

        external::sip2sip      gateway                               sip:eastwest at sip2sip.info <mailto:sip%3Aeastwest at sip2sip.info>           REGED

      external::flowroute   gateway                 sip:6509889800 at us-west-or.sip-flowroute.com <mailto:sip%3A6509889800 at us-west-or.sip-flowroute.com>              NOREG

                    phone               profile               sip:mod_sofia at 10.0.0.167:5060 <http://sip:mod_sofia at 10.0.0.167:5060>   RUNNING (0)

            internal-ipv6         profile     sip:mod_sofia@[2601:647:4802:9220:c816:30ae:6a9a:d191]:5060        RUNNING (0)

                 internal              profile           sip:mod_sofia at 67.164.101.201:5060 <http://sip:mod_sofia at 67.164.101.201:5060>             RUNNING (0)

=================================================================================================

5 profiles 1 alias

 

Why does the "phone" profile have "sip:mod_sofia at 10.0.0.167:5060 <http://sip:mod_sofia at 10.0.0.167:5060> ". Shouldn't that have the 192.168.3.2 address? And why isn't my phone registering?

 

-- 

Steve

_________________________________________________________________________

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 <mailto: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 <mailto: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 <mailto: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 <mailto: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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210224/0448cca5/attachment-0001.html>

From keithxcroxford at gmail.com  Wed Feb 24 17:50:05 2021
From: keithxcroxford at gmail.com (Keith Croxford)
Date: Wed, 24 Feb 2021 18:50:05 +0100
Subject: [Freeswitch-users] DTMF not passing from the A Leg to the B Leg
In-Reply-To: <CAGvJV4qw3Wdm2fCfNKWKx0rrbQq4DNLb56zc5vmfdUqA4VR5FQ@mail.gmail.com>
References: <CABb=kMcK7jeMgAMpHR=sPe3vJpgj_Rp5BttaSLj3xappyTWE_g@mail.gmail.com>
 <CAGvJV4qw3Wdm2fCfNKWKx0rrbQq4DNLb56zc5vmfdUqA4VR5FQ@mail.gmail.com>
Message-ID: <CABb=kMeUtgpo-_j3G7qg+3WaTTNMtcuMTsFFSFCVsM8pg37SJg@mail.gmail.com>

I ended up testing with the latest version of FREESWITCH, and the issue is
no longer there.

Something that I did find interesting though: if I played a sequence of
rfc2833 events without any g711 packets (9from the same SSRC, only the
first digit would be sent to the B leg.

If add a few g711 packets (white noise) from the same SSRC between the 2833
packets, all DTMF makes it to the B leg.

I tried to find something in the release notes to explain this, but I
didn't have any luck.

-KC

Op wo 24 feb. 2021 09:37 schreef Dragos Oancea <dragos at freeswitch.org>:

> Try setting channel variable "sensitive_dtmf" to true.
>
> On Mon, Feb 22, 2021 at 7:17 PM Keith Croxford <keithxcroxford at gmail.com>
> wrote:
>
>> I'm coming across a strange, intermittent behavior with RFC2833 DTMF
>> packets. This is occurring in a production environment, and I've been able
>> to duplicate this with a very minimal FreeSWITCH installation in a virtual
>> machine.
>>
>> Details of the problem:
>>
>> A customer occasionally has a problem report where their callers are
>> unable to use the IVR menu. From the user's standpoint, the digits are not
>> detected. Upon further review, we do receive DTMF event packets from the
>> carrier, however, they do not traverse the bridge from the A leg to the B
>> leg.
>>
>> From the console logs, I see the following with the first digit.
>> --
>> 2021-02-20 11:28:30.842108 [DEBUG] mod_sofia.c:645 SOFIA EXCHANGE_MEDIA
>> 2021-02-20 11:28:30.881612 [DEBUG] switch_rtp.c:1890 rtcp_stats_init:
>> audio ssrc[1488669269] base_seq[26830]
>> 2021-02-20 11:28:30.933002 [DEBUG] switch_rtp.c:7550 Correct audio
>> ip/port confirmed.
>> 2021-02-20 11:28:34.881612 [DEBUG] switch_rtp.c:7550 Correct audio
>> ip/port confirmed.
>> 2021-02-20 11:28:34.923555 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 1:400
>> 2021-02-20 11:28:34.923555 [INFO] switch_channel.c:515 RECV DTMF 1:400
>> 2021-02-20 11:28:34.923555 [DEBUG] switch_rtp.c:5424 Send start packet
>> for [1] ts=160 dur=160/160/400 seq=51938 lw=160
>> 2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5323 Send middle packet
>> for [1] ts=160 dur=320/320/400 seq=51939 lw=320
>> 2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5323 Send end packet for
>> [1] ts=160 dur=480/480/400 seq=51940 lw=320
>> 2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5323 Send end packet for
>> [1] ts=160 dur=480/480/400 seq=51941 lw=320
>> 2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5323 Send end packet for
>> [1] ts=160 dur=480/480/400 seq=51942 lw=320
>> 2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5271 Queue digit delay of
>> 40ms
>> 2021-02-20 11:28:35.421487 [DEBUG] switch_rtp.c:6985 Correct audio RTCP
>> ip/port confirmed.
>> --
>>
>> However, on the following series of digits, they are detected as seen in
>> the logs from switch_rtp.c and switch_channel.c. However, they do not
>> proceed to the "Send start packet" step as seen with the previous digit.
>> This mirrors what I see on the B leg client, as it only detects the first
>> digit.
>>
>> --
>> 2021-02-20 11:28:37.521489 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 1:400
>> 2021-02-20 11:28:37.521489 [INFO] switch_channel.c:515 RECV DTMF 1:400
>> 2021-02-20 11:28:46.082031 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 3:400
>> 2021-02-20 11:28:46.082031 [INFO] switch_channel.c:515 RECV DTMF 3:400
>> 2021-02-20 11:28:46.481377 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 6:400
>> 2021-02-20 11:28:46.481377 [INFO] switch_channel.c:515 RECV DTMF 6:400
>> 2021-02-20 11:28:47.061795 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 4:400
>> 2021-02-20 11:28:47.061795 [INFO] switch_channel.c:515 RECV DTMF 4:400
>> 2021-02-20 11:28:47.461504 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 5:400
>> 2021-02-20 11:28:47.461504 [INFO] switch_channel.c:515 RECV DTMF 5:400
>> 2021-02-20 11:28:47.881346 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 0:400
>> 2021-02-20 11:28:47.881346 [INFO] switch_channel.c:515 RECV DTMF 0:400
>> 2021-02-20 11:28:48.281220 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 0:400
>> 2021-02-20 11:28:48.281220 [INFO] switch_channel.c:515 RECV DTMF 0:400
>> 2021-02-20 11:28:48.805572 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF #:400
>> 2021-02-20 11:28:48.805572 [INFO] switch_channel.c:515 RECV DTMF #:400
>> 2021-02-20 11:29:04.881294 [NOTICE] sofia.c:1079 Hangup sofia/external/
>> nobody at 192.168.1.71 [CS_EXECUTE] [NORMAL_CLEARING]
>> --
>>
>> Yet, they do arrive on the Freeswitch instances. I've attempted to
>> add <action application="set"
>> data="rtp_manual_rtp_bugs=ignore_dtmf_duration"/>  to my inbound dialplan
>> xml. I'm leaning toward these being a "weird" DTMF packet and not a
>> FreeSWITCH issue, but I'm curious if anyone has encountered a similar
>> problem, or has advice of what we could test on the FS side.
>> --
>> Supporting Details:
>> Test Environment:
>> I took a packet capture from production and filtered it down to the A
>> leg's RFC2833 packets, and saved this to a new file. I then used this pcap
>> file in a SIPP UAC scenario which is acting as the A Leg.
>>
>> Freeswitch sits in the middle and I've configured it following this guide
>> (
>> https://freeswitch.org/confluence/display/FREESWITCH/SBC+FreeSWITCH+Configuration+Example+2)
>> as a barebones SBC. Freeswitch is version 1.8.3
>>
>> The B Leg is a simple pjsua client configured to answer an inbound call.
>> --
>> Best Regards,
>>
>> KC
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>>
>> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210224/8bd50a1f/attachment.html>

From vma at sip.solutions  Wed Feb 24 20:44:44 2021
From: vma at sip.solutions (Valli A. Vallimamod - SIP Solutions)
Date: Wed, 24 Feb 2021 21:44:44 +0100
Subject: [Freeswitch-users] DTMF not passing from the A Leg to the B Leg
In-Reply-To: <CABb=kMeUtgpo-_j3G7qg+3WaTTNMtcuMTsFFSFCVsM8pg37SJg@mail.gmail.com>
References: <CABb=kMcK7jeMgAMpHR=sPe3vJpgj_Rp5BttaSLj3xappyTWE_g@mail.gmail.com>
 <CAGvJV4qw3Wdm2fCfNKWKx0rrbQq4DNLb56zc5vmfdUqA4VR5FQ@mail.gmail.com>
 <CABb=kMeUtgpo-_j3G7qg+3WaTTNMtcuMTsFFSFCVsM8pg37SJg@mail.gmail.com>
Message-ID: <6D8052E8-F2AC-47E5-8704-51B952524173@sip.solutions>

Hi,

What RTP timer are you using? I remember having a similar issue with timerfd some time ago, it disappeared when I changed to soft timer.
The pass_rfc2833 option also helped.


Best Regards,
-- 
Valli A. Vallimamod
SIP Solutions
vma at sip.solutions
linkedin.com/in/vallimamod
.


> On 24 Feb 2021, at 18:50, Keith Croxford <keithxcroxford at gmail.com> wrote:
> 
> I ended up testing with the latest version of FREESWITCH, and the issue is no longer there.
> 
> Something that I did find interesting though: if I played a sequence of rfc2833 events without any g711 packets (9from the same SSRC, only the first digit would be sent to the B leg.
> 
> If add a few g711 packets (white noise) from the same SSRC between the 2833 packets, all DTMF makes it to the B leg.
> 
> I tried to find something in the release notes to explain this, but I didn't have any luck. 
> 
> -KC
> 
> Op wo 24 feb. 2021 09:37 schreef Dragos Oancea <dragos at freeswitch.org>:
> Try setting channel variable "sensitive_dtmf" to true.
> 
> On Mon, Feb 22, 2021 at 7:17 PM Keith Croxford <keithxcroxford at gmail.com> wrote:
> I'm coming across a strange, intermittent behavior with RFC2833 DTMF packets. This is occurring in a production environment, and I've been able to duplicate this with a very minimal FreeSWITCH installation in a virtual machine. 
> 
> Details of the problem: 
> 
> A customer occasionally has a problem report where their callers are unable to use the IVR menu. From the user's standpoint, the digits are not detected. Upon further review, we do receive DTMF event packets from the carrier, however, they do not traverse the bridge from the A leg to the B leg.  
> 
> From the console logs, I see the following with the first digit. 
> --
> 2021-02-20 11:28:30.842108 [DEBUG] mod_sofia.c:645 SOFIA EXCHANGE_MEDIA
> 2021-02-20 11:28:30.881612 [DEBUG] switch_rtp.c:1890 rtcp_stats_init: audio ssrc[1488669269] base_seq[26830]
> 2021-02-20 11:28:30.933002 [DEBUG] switch_rtp.c:7550 Correct audio ip/port confirmed.
> 2021-02-20 11:28:34.881612 [DEBUG] switch_rtp.c:7550 Correct audio ip/port confirmed.
> 2021-02-20 11:28:34.923555 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 1:400
> 2021-02-20 11:28:34.923555 [INFO] switch_channel.c:515 RECV DTMF 1:400
> 2021-02-20 11:28:34.923555 [DEBUG] switch_rtp.c:5424 Send start packet for [1] ts=160 dur=160/160/400 seq=51938 lw=160
> 2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5323 Send middle packet for [1] ts=160 dur=320/320/400 seq=51939 lw=320
> 2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5323 Send end packet for [1] ts=160 dur=480/480/400 seq=51940 lw=320
> 2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5323 Send end packet for [1] ts=160 dur=480/480/400 seq=51941 lw=320
> 2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5323 Send end packet for [1] ts=160 dur=480/480/400 seq=51942 lw=320
> 2021-02-20 11:28:34.942196 [DEBUG] switch_rtp.c:5271 Queue digit delay of 40ms
> 2021-02-20 11:28:35.421487 [DEBUG] switch_rtp.c:6985 Correct audio RTCP ip/port confirmed.
> --
> 
> However, on the following series of digits, they are detected as seen in the logs from switch_rtp.c and switch_channel.c. However, they do not proceed to the "Send start packet" step as seen with the previous digit. This mirrors what I see on the B leg client, as it only detects the first digit. 
> 
> --
> 2021-02-20 11:28:37.521489 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 1:400
> 2021-02-20 11:28:37.521489 [INFO] switch_channel.c:515 RECV DTMF 1:400
> 2021-02-20 11:28:46.082031 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 3:400
> 2021-02-20 11:28:46.082031 [INFO] switch_channel.c:515 RECV DTMF 3:400
> 2021-02-20 11:28:46.481377 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 6:400
> 2021-02-20 11:28:46.481377 [INFO] switch_channel.c:515 RECV DTMF 6:400
> 2021-02-20 11:28:47.061795 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 4:400
> 2021-02-20 11:28:47.061795 [INFO] switch_channel.c:515 RECV DTMF 4:400
> 2021-02-20 11:28:47.461504 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 5:400
> 2021-02-20 11:28:47.461504 [INFO] switch_channel.c:515 RECV DTMF 5:400
> 2021-02-20 11:28:47.881346 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 0:400
> 2021-02-20 11:28:47.881346 [INFO] switch_channel.c:515 RECV DTMF 0:400
> 2021-02-20 11:28:48.281220 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 0:400
> 2021-02-20 11:28:48.281220 [INFO] switch_channel.c:515 RECV DTMF 0:400
> 2021-02-20 11:28:48.805572 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF #:400
> 2021-02-20 11:28:48.805572 [INFO] switch_channel.c:515 RECV DTMF #:400
> 2021-02-20 11:29:04.881294 [NOTICE] sofia.c:1079 Hangup sofia/external/nobody at 192.168.1.71 [CS_EXECUTE] [NORMAL_CLEARING]
> --
> 
> Yet, they do arrive on the Freeswitch instances. I've attempted to add <action application="set" data="rtp_manual_rtp_bugs=ignore_dtmf_duration"/>  to my inbound dialplan xml. I'm leaning toward these being a "weird" DTMF packet and not a FreeSWITCH issue, but I'm curious if anyone has encountered a similar problem, or has advice of what we could test on the FS side. 
> --
> Supporting Details: 
> Test Environment: 
> I took a packet capture from production and filtered it down to the A leg's RFC2833 packets, and saved this to a new file. I then used this pcap file in a SIPP UAC scenario which is acting as the A Leg. 
> 
> Freeswitch sits in the middle and I've configured it following this guide (https://freeswitch.org/confluence/display/FREESWITCH/SBC+FreeSWITCH+Configuration+Example+2) as a barebones SBC. Freeswitch is version 1.8.3
> 
> The B Leg is a simple pjsua client configured to answer an inbound call. 
> --
> Best Regards, 
> 
> KC
> 
> 
> 
> 
> 
> _________________________________________________________________________
> 
> 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



From ryandelgrosso at gmail.com  Thu Feb 25 18:09:15 2021
From: ryandelgrosso at gmail.com (Ryan Delgrosso)
Date: Thu, 25 Feb 2021 10:09:15 -0800
Subject: [Freeswitch-users] Orphaned channels
Message-ID: <4f112385-ce08-b96f-cdad-0f13aacb9182@gmail.com>

Hi All,

Ive stumbled into a situation where I see orphaned channels accumulate 
on FS occasionally.

If from the API (not cli) i dump channels, i find some entries with a 
UUID and all other values null.

If i do this from the CLI it seems to go to the DB which doesnt show 
these orphans

The FS 'show status' command shows call counts which include the 
orphaned channels

If I attempt to run any 'uuid_' commands on them I just get "-ERR No 
such channel!"

Only hupall seems to tangibly impact these but thats a bit of a nuclear 
option.

Any way to prevent/clean these or identify and prevent them from getting 
orphaned in the first place?

FS Ver is 1.10.2 on CentOS using DB via odbc

Thanks in advance

-Ryan





From svanherwaarden at precisionag.org  Fri Feb 26 12:57:35 2021
From: svanherwaarden at precisionag.org (Sam van Herwaarden)
Date: Fri, 26 Feb 2021 13:57:35 +0100
Subject: [Freeswitch-users] "Shadow calls" - duplicate calls executing in
	parallel
Message-ID: <CAAtWTemObULf7tQUcy+V+h+VCen_O0rXaorZC3AoK_noPDCL8g@mail.gmail.com>

Hi all,

I was just wondering if others have experienced this: on a few of the
FreeSWITCH systems I've interacted with, I've observed situations where
incoming phone calls seem to be duplicated. So there seem to be two call
processes for the same phone number, also both receiving things like DTMF
events. The timestamps of events tend to be very close (sub-second) but not
necessarily identical.

It's not very common, less than 1 per 1000 calls has this happening. It
does tend to throw off some of our tooling (both in call handling and
analysis). Curious to know if there could be an obvious cause/easy fix.

Kind regards,
Sam
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210226/4a53473f/attachment.html>

From grzegorz at orzel.io  Thu Feb 25 10:23:17 2021
From: grzegorz at orzel.io (=?utf-8?Q?Grzegorz_Orze=C5=82?=)
Date: Thu, 25 Feb 2021 11:23:17 +0100
Subject: [Freeswitch-users] Fax to an extension of B leg
Message-ID: <431B6E46-AA66-418F-8526-D175F0B8C305@orzel.io>

Hi folks,

I have a FreeSwitch serving multiple services. One of them is fax2mail/mail2fax. Logic is implemented as a set of lua scripts and it works fine. I’d like extend mail2fax direction. Currently it (lua script) picks number B, tiff file and sends it directly to B using command originate as follows:

originate {<_bunch_of_headers_>} sofia/service/+4812345678 at some.domain <mailto:sofia/service/+4812345678 at some.domain>;user=phone &txfax(<_tiff_file_path_>) XML <_some_profile_>

What I’d like to do is to add possibility to dial some other extension number. Lest’s say fax is available on +4812345678 but when the call is aswered there needs to be dialed number 123.

I’ve already reviewed all FreeSwitch books on safaribooks,  googled but I haven't found any example… :-/ 

Maybe it is enough to simply add it to the call url - I mean something like:

originate {<_bunch_of_headers_>} sofia/service/+4812345678,,123 at some.domain <mailto:123 at some.domain>;user=phone &txfax(<_tiff_file_path_>) XML <_some_profile_>

Can you give me some hint how to do that?

Thanks in advance,
Grzegorz
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210225/50019b68/attachment.html>

From sitdikovt at gmail.com  Thu Feb 25 18:44:02 2021
From: sitdikovt at gmail.com (=?UTF-8?B?0KLQuNC80YPRgCDQodC40YLQtNC40LrQvtCy?=)
Date: Fri, 26 Feb 2021 00:44:02 +0600
Subject: [Freeswitch-users] B-leg ringing after A-leg hangs-up. LUA.
Message-ID: <CABGrKasLSksOF_2zD5i-FqBYN2VsTJ101-ONiWEsqW3samakSQ@mail.gmail.com>

Hi all. Need help with a simple call script.
I'm got this in dialplan
<extension name="call_lua_home">
      <condition field="destination_number" expression="^(1007)$">
        <!-- <action application="set" data="session_in_hangup_hook=true"/> -->
For hangup hook try
        <action application="lua" data="fork.lua"/>
      </condition>
</extension>
And this in fork.lua
if session:ready() then
api = freeswitch.API()
contact = api:execute("sofia_contact", "*/1007 at compA.com");
caller = session:getVariable("caller_id_number");
session1 = freeswitch.Session("{origination_caller_id_name="..caller..
"}[leg_timeout=30]"..contact.."");
-- session:setHangupHook("HangupHook");
if session1:ready() then
    freeswitch.bridge(session, session1);
end
freeswitch.consoleLog("WARNING","=====SCRIPT END\n");
session:hangup()
end

When Caller is hanging up before B-leg answers - Bleg is continuing to
ring. There are still 2 channels in 'show channels' after A-leg hangs up.
I want to drop B-leg when A-leg hangs up. Is it possible?

UPD Tried to use hangup hook. It works with stramFile, but no luck with
session.
function HangupHook(s, status)
    freeswitch.consoleLog("WARNING","Event fired breaking out\n");
    return exit;
    -- return die;
end

session:answer();
session:setHangupHook("HangupHook");
while (session:ready() == true) do

    session:streamFile(
"/usr/share/freeswitch/sounds/en/us/callie/ivr/16000/ivr-on_hold_indefinitely.wav"
);

-- session1 = freeswitch.Session("{origination_caller_id_name="..caller.."}[leg_timeout=30]"..contact.."");
end
freeswitch.consoleLog("WARNING","=====SCRIPT END\n");
session:hangup()
end

Can anyone help me with this?

Thanks! Regards,Tim
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210226/870a78d9/attachment.html>

From brian at freeswitch.com  Fri Feb 26 19:38:37 2021
From: brian at freeswitch.com (Brian West)
Date: Fri, 26 Feb 2021 13:38:37 -0600
Subject: [Freeswitch-users] Fax to an extension of B leg
In-Reply-To: <431B6E46-AA66-418F-8526-D175F0B8C305@orzel.io>
References: <431B6E46-AA66-418F-8526-D175F0B8C305@orzel.io>
Message-ID: <CAEJMVkAFFaPY0cVuMgnjMfqvqK9g5sqoYmuvOahWQVwz19FnnA@mail.gmail.com>

Review this, it may give you a road map
https://raw.githubusercontent.com/signalwire/freeswitch/master/scripts/perl/fax.cgi

The key here is ignore_early_media=true, you don't want to be in the txfax
app until the remote answers, because you are in a chicken egg scenario
where the fax will fail or t.38 will fail as it will attempt to do so
before the other party answers.

/b

On Fri, Feb 26, 2021 at 12:52 PM Grzegorz Orzeł <grzegorz at orzel.io> wrote:

> Hi folks,
>
> I have a FreeSwitch serving multiple services. One of them is
> fax2mail/mail2fax. Logic is implemented as a set of lua scripts and it
> works fine. I’d like extend mail2fax direction. Currently it (lua script)
> picks number B, tiff file and sends it directly to B using command
> originate as follows:
>
> originate {<_bunch_of_headers_>} sofia/service/+4812345678 at some.domain;user=phone
> &txfax(<_tiff_file_path_>) XML <_some_profile_>
>
> What I’d like to do is to add possibility to dial some other extension
> number. Lest’s say fax is available on +4812345678 but when the call is
> aswered there needs to be dialed number 123.
>
> I’ve already reviewed all FreeSwitch books on safaribooks,  googled but I
> haven't found any example… :-/
>
> Maybe it is enough to simply add it to the call url - I mean something
> like:
>
> originate {<_bunch_of_headers_>} sofia/service/+4812345678,,
> 123 at some.domain;user=phone &txfax(<_tiff_file_path_>) XML <_some_profile_>
>
> Can you give me some hint how to do that?
>
> Thanks in advance,
> Grzegorz
> _________________________________________________________________________
>
> 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
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>

Email: brian at 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>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210226/afb59832/attachment-0001.html>

From brian at freeswitch.com  Fri Feb 26 19:45:23 2021
From: brian at freeswitch.com (Brian West)
Date: Fri, 26 Feb 2021 13:45:23 -0600
Subject: [Freeswitch-users] B-leg ringing after A-leg hangs-up. LUA.
In-Reply-To: <CABGrKasLSksOF_2zD5i-FqBYN2VsTJ101-ONiWEsqW3samakSQ@mail.gmail.com>
References: <CABGrKasLSksOF_2zD5i-FqBYN2VsTJ101-ONiWEsqW3samakSQ@mail.gmail.com>
Message-ID: <CAEJMVkDg8V7DLt0nJaqtSN+B7MB47Cp769wr2vgXoEXHZZ3e-w@mail.gmail.com>

Tim,

Based on your description, I would guess that the B-Leg hangs up when they
answer after the a-leg is gone??

/b




On Fri, Feb 26, 2021 at 12:42 PM Тимур Ситдиков <sitdikovt at gmail.com> wrote:

> Hi all. Need help with a simple call script.
> I'm got this in dialplan
> <extension name="call_lua_home">
>       <condition field="destination_number" expression="^(1007)$">
>         <!-- <action application="set" data="session_in_hangup_hook=true"/> -->
> For hangup hook try
>         <action application="lua" data="fork.lua"/>
>       </condition>
> </extension>
> And this in fork.lua
> if session:ready() then
> api = freeswitch.API()
> contact = api:execute("sofia_contact", "*/1007 at compA.com");
> caller = session:getVariable("caller_id_number");
> session1 = freeswitch.Session("{origination_caller_id_name="..caller..
> "}[leg_timeout=30]"..contact.."");
> -- session:setHangupHook("HangupHook");
> if session1:ready() then
>     freeswitch.bridge(session, session1);
> end
> freeswitch.consoleLog("WARNING","=====SCRIPT END\n");
> session:hangup()
> end
>
> When Caller is hanging up before B-leg answers - Bleg is continuing to
> ring. There are still 2 channels in 'show channels' after A-leg hangs up.
> I want to drop B-leg when A-leg hangs up. Is it possible?
>
> UPD Tried to use hangup hook. It works with stramFile, but no luck with
> session.
> function HangupHook(s, status)
>     freeswitch.consoleLog("WARNING","Event fired breaking out\n");
>     return exit;
>     -- return die;
> end
>
> session:answer();
> session:setHangupHook("HangupHook");
> while (session:ready() == true) do
>
>     session:streamFile(
> "/usr/share/freeswitch/sounds/en/us/callie/ivr/16000/ivr-on_hold_indefinitely.wav"
> );
>
> -- session1 = freeswitch.Session("{origination_caller_id_name="..caller.."}[leg_timeout=30]"..contact.."");
> end
> freeswitch.consoleLog("WARNING","=====SCRIPT END\n");
> session:hangup()
> end
>
> Can anyone help me with this?
>
> Thanks! Regards,Tim
> _________________________________________________________________________
>
> 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
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>

Email: brian at 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>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210226/0bb2b801/attachment.html>

From sitdikovt at gmail.com  Fri Feb 26 20:16:26 2021
From: sitdikovt at gmail.com (=?UTF-8?B?0KLQuNC80YPRgCDQodC40YLQtNC40LrQvtCy?=)
Date: Sat, 27 Feb 2021 02:16:26 +0600
Subject: [Freeswitch-users] B-leg ringing after A-leg hangs-up. LUA.
In-Reply-To: <CAEJMVkDg8V7DLt0nJaqtSN+B7MB47Cp769wr2vgXoEXHZZ3e-w@mail.gmail.com>
References: <CABGrKasLSksOF_2zD5i-FqBYN2VsTJ101-ONiWEsqW3samakSQ@mail.gmail.com>
 <CAEJMVkDg8V7DLt0nJaqtSN+B7MB47Cp769wr2vgXoEXHZZ3e-w@mail.gmail.com>
Message-ID: <CABGrKasrUzAMdUmP2yk_-5KBeCpnzXzdF84Sns6gVE6iH3M2Fw@mail.gmail.com>

Hi Brian!
Yes, B hangs up right after pick up. Also there's an error about non
available channels in console.



сб, 27 февр. 2021 г., 02:07 Brian West <brian at freeswitch.com>:

> Tim,
>
> Based on your description, I would guess that the B-Leg hangs up when they
> answer after the a-leg is gone??
>
> /b
>
>
>
>
> On Fri, Feb 26, 2021 at 12:42 PM Тимур Ситдиков <sitdikovt at gmail.com>
> wrote:
>
>> Hi all. Need help with a simple call script.
>> I'm got this in dialplan
>> <extension name="call_lua_home">
>>       <condition field="destination_number" expression="^(1007)$">
>>         <!-- <action application="set" data="session_in_hangup_hook=true"/> -->
>> For hangup hook try
>>         <action application="lua" data="fork.lua"/>
>>       </condition>
>> </extension>
>> And this in fork.lua
>> if session:ready() then
>> api = freeswitch.API()
>> contact = api:execute("sofia_contact", "*/1007 at compA.com");
>> caller = session:getVariable("caller_id_number");
>> session1 = freeswitch.Session("{origination_caller_id_name="..caller..
>> "}[leg_timeout=30]"..contact.."");
>> -- session:setHangupHook("HangupHook");
>> if session1:ready() then
>>     freeswitch.bridge(session, session1);
>> end
>> freeswitch.consoleLog("WARNING","=====SCRIPT END\n");
>> session:hangup()
>> end
>>
>> When Caller is hanging up before B-leg answers - Bleg is continuing to
>> ring. There are still 2 channels in 'show channels' after A-leg hangs up.
>> I want to drop B-leg when A-leg hangs up. Is it possible?
>>
>> UPD Tried to use hangup hook. It works with stramFile, but no luck with
>> session.
>> function HangupHook(s, status)
>>     freeswitch.consoleLog("WARNING","Event fired breaking out\n");
>>     return exit;
>>     -- return die;
>> end
>>
>> session:answer();
>> session:setHangupHook("HangupHook");
>> while (session:ready() == true) do
>>
>>     session:streamFile(
>> "/usr/share/freeswitch/sounds/en/us/callie/ivr/16000/ivr-on_hold_indefinitely.wav"
>> );
>>
>> -- session1 = freeswitch.Session("{origination_caller_id_name="..caller.."}[leg_timeout=30]"..contact.."");
>> end
>> freeswitch.consoleLog("WARNING","=====SCRIPT END\n");
>> session:hangup()
>> end
>>
>> Can anyone help me with this?
>>
>> Thanks! Regards,Tim
>> _________________________________________________________________________
>>
>> 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
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>
> Email: brian at 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>
> _________________________________________________________________________
>
> 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: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210227/54454bcd/attachment-0001.html>

From davidswalkabout at gmail.com  Sat Feb 27 02:34:33 2021
From: davidswalkabout at gmail.com (David P)
Date: Sat, 27 Feb 2021 15:34:33 +1300
Subject: [Freeswitch-users] "Shadow calls" - duplicate calls executing
	in parallel
In-Reply-To: <mailman.1388.1614368359.1264.freeswitch-users@lists.freeswitch.org>
References: <mailman.1388.1614368359.1264.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAE23truFiM4GGceRerdzcQt8Oq9EPMQWHN8Bmi1BNzcgiMTJ9Q@mail.gmail.com>

I think the last time we experienced that was with 10.4 (we're on 10.5 now).

It happened infrequently as you note. I assumed it was an error in the way
our WebRTC app invoked verto. (I didn't have any browser console log to
confirm that, though.)

On Sat, 27 Feb 2021, 9:07 am , <
freeswitch-users-request at lists.freeswitch.org> wrote:

>
>
> ---------- Forwarded message ----------
> From: Sam van Herwaarden <svanherwaarden at precisionag.org>
>
> Hi all,
>
> I was just wondering if others have experienced this: on a few of the
> FreeSWITCH systems I've interacted with, I've observed situations where
> incoming phone calls seem to be duplicated. So there seem to be two call
> processes for the same phone number, also both receiving things like DTMF
> events. The timestamps of events tend to be very close (sub-second) but not
> necessarily identical.
>
> It's not very common, less than 1 per 1000 calls has this happening. It
> does tend to throw off some of our tooling (both in call handling and
> analysis). Curious to know if there could be an obvious cause/easy fix.
>
> Kind regards,
> Sam
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210227/ab51954b/attachment.html>

From lewis.bergman at gmail.com  Fri Feb 26 21:41:24 2021
From: lewis.bergman at gmail.com (Lewis Bergman)
Date: Fri, 26 Feb 2021 15:41:24 -0600
Subject: [Freeswitch-users] mod_sndfile error called from mod_tts_commandline
Message-ID: <CAD2CnaRbYws4FwfgwDVpcv8fnS5NB_r72nv-BjW_SsZMo1KJuA@mail.gmail.com>

Hello all. I am pretty new to Freeswitch and have an experimental container
on Proxmox 6.3 Debian 10 to play with. The goal is to get mod_unimrcp
working. To that end, I have mod_tts_commandline (installed from binary
with FusionPBX as part of the default install) listed in modules.conf.xml
/etc/freeswitch/autoload_configs/modules.conf.xml
                <!-- Speech Recognition / Text to Speech -->
                <load module="mod_unimrcp"/>
                <load module="mod_tts_commandline"/>
                             <load module="mod_flite"/>
I have tried it with and without mod_flite enabled.
Mod_tts_commandline refuses to pull a sound file and the debug log seems to
blame mod_sndfile. This is using the example from the mod_tts_commandline
page
<https://freeswitch.org/confluence/display/FREESWITCH/mod_tts_commandline#mod_tts_commandline-Configuringmod_tts_commandline>
as dialplan shows following the debug log output below:

2021-02-26 15:19:24.713040 [NOTICE] mod_dptools.c:1406 Channel
[sofia/internal/102 at 66.151.243.45] has been answered
2021-02-26 15:19:24.713040 [DEBUG] switch_channel.c:3865 (sofia/internal/
102 at 66.151.243.45) Callstate Change RINGING -> ACTIVE
2021-02-26 15:19:24.713040 [DEBUG] sofia.c:7326 Channel sofia/internal/
102 at 66.151.243.45 entering state [completed][200]
EXECUTE [depth=0] sofia/internal/102 at 66.151.243.45
speak(tts_commandline|pico|This is an example of using tts_commandline)
2021-02-26 15:19:24.713040 [DEBUG] switch_ivr_play_say.c:3023 OPEN TTS
tts_commandline
2021-02-26 15:19:24.713040 [DEBUG] switch_ivr_play_say.c:3033 Raw Codec
Activated
2021-02-26 15:19:24.713040 [DEBUG] mod_tts_commandline.c:160 Executing:
echo 'This is an example of using tts_commandline' | text2wave -f 8000 >
'/tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav'
2021-02-26 15:19:25.433013 [WARNING] mod_sndfile.c:281 Error Opening File
[/tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav] [File contains data in
an unknown format.]
2021-02-26 15:19:25.433013 [ERR] mod_tts_commandline.c:170 Failed to open
file: /tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav
2021-02-26 15:19:25.433013 [DEBUG] switch_ivr_play_say.c:2741 Speaking
text: This is an example of using tts_commandline
2021-02-26 15:19:25.433013 [DEBUG] sofia.c:7326 Channel sofia/internal/
102 at 66.151.243.45 entering state [ready][200]
2021-02-26 15:19:25.433013 [DEBUG] switch_ivr_play_say.c:2905 done speaking
text
2021-02-26 15:19:25.433013 [NOTICE] switch_core_state_machine.c:386
sofia/internal/102 at 66.151.243.45 has executed the last dialplan
instruction, hanging up.
2021-02-26 15:19:25.433013 [NOTICE] switch_core_state_machine.c:388 Hangup
sofia/internal/102 at 66.151.243.45 [CS_EXECUTE] [NORMAL_CLEARING]

The dialplan for extension I am testing looks like this:
<extension name="tts_commandline_test">
   <condition field="destination_number" expression="^561$">
      <action application="speak" data="tts_commandline|pico|This is an
example of using tts_commandline"/>
   </condition>
</extension>

The code lines mentioned in debug seem to be pretty mundane, not that I
really know what I am looking for:
>From mod_tts_commnandline line 160:
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Executing: %s\n",
message);
I tried setting the /tmp directory to 777 but that didn't help and the
error output was identical. mod_tts_commandline never reports an error on
creation and it looks like the next part of of mod_tts_commandline would
print an error if it did like this:
if (switch_system(message, SWITCH_TRUE) < 0) {
    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Failed to
execute command: %s\n", message);
    ret = SWITCH_STATUS_FALSE; goto done;
}
if (switch_core_file_open(info->fh, info->file, 0, //number_of_channels,
    info->rate, //samples_per_second,
    SWITCH_FILE_FLAG_READ | SWITCH_FILE_DATA_SHORT, NULL) !=
SWITCH_STATUS_SUCCESS) {
    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Failed to open
file: %s\n", info->file);
    ret = SWITCH_STATUS_FALSE; goto done;
}

Starting at line 279 in mod_sndfile (sorry for the space formatting):

if (!context->handle) {
if (sndfile_perform_open(context, path, mode, handle) !=
SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Error Opening
File [%s] [%s]\n", path, sf_strerror(context->handle));
status = SWITCH_STATUS_GENERR;
goto end;
}
}

According to the dptools docs
<https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+playback#mod_dptools:playback-3.1>
and
libsndfile wav has been supported forever and it would seem highly unlikely
that is an issue. I check the /tmp directory and no files are present but I
don't know if they are being deleted by some cleanup process or just not
created. Not knowing C, it seems like mod_sndfile never received a file
handle but that is just a guess. Other Freeswitch apps are writing to the
/tmp directory (at least there are some files in there). Maybe someone sees
an error from mod_tts_commandline I don't see or knows I am doing something
stupid. Any help would be appreciated. I only found 2 references to the
error in the mail history and neither had anything to do with a
similar issue.

Freeswitch version reports: FreeSWITCH Version
1.10.5-release-17-25569c1631~64bit (-release-17-25569c1631 64bit)
FusionPBX version, in case it matters, is : 4.5.21
OS is Debian 10 Buster
-- 
Lewis Bergman
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210226/06a4ba8f/attachment.html>

From lewis.bergman at gmail.com  Sat Feb 27 14:06:43 2021
From: lewis.bergman at gmail.com (Lewis Bergman)
Date: Sat, 27 Feb 2021 08:06:43 -0600
Subject: [Freeswitch-users] mod_sndfile error called from
 mod_tts_commandline - [WARNING] mod_sndfile.c:281 Error Opening File
Message-ID: <CAD2CnaQp7Vb9YEq=9w2+tTHbOdMKhuhnA2p+Y+4TbKveGFB_6Q@mail.gmail.com>

Hello all. I am pretty new to Freeswitch and have an experimental container
on Proxmox 6.3 Debian 10 to play with. The goal is to get mod_unimrcp
working. To that end, I have mod_tts_commandline (installed from binary
with FusionPBX as part of the default install) listed in modules.conf.xml
/etc/freeswitch/autoload_configs/modules.conf.xml
                <!-- Speech Recognition / Text to Speech -->
                <load module="mod_unimrcp"/>
                <load module="mod_tts_commandline"/>
                <load module="mod_flite"/>
I have tried it with and without mod_flite enabled.
Mod_tts_commandline refuses to pull a sound file and the debug log seems to
blame mod_sndfile. This is using the example from the mod_tts_commandline
page
<https://freeswitch.org/confluence/display/FREESWITCH/mod_tts_commandline#mod_tts_commandline-Configuringmod_tts_commandline>
as dialplan shows following the debug log output below:

2021-02-26 15:19:24.713040 [NOTICE] mod_dptools.c:1406 Channel
[sofia/internal/102 at 66.151.243.45] has been answered
2021-02-26 15:19:24.713040 [DEBUG] switch_channel.c:3865 (sofia/internal/
102 at 66.151.243.45) Callstate Change RINGING -> ACTIVE
2021-02-26 15:19:24.713040 [DEBUG] sofia.c:7326 Channel sofia/internal/
102 at 66.151.243.45 entering state [completed][200]
EXECUTE [depth=0] sofia/internal/102 at 66.151.243.45
speak(tts_commandline|pico|This is an example of using tts_commandline)
2021-02-26 15:19:24.713040 [DEBUG] switch_ivr_play_say.c:3023 OPEN TTS
tts_commandline
2021-02-26 15:19:24.713040 [DEBUG] switch_ivr_play_say.c:3033 Raw Codec
Activated
2021-02-26 15:19:24.713040 [DEBUG] mod_tts_commandline.c:160 Executing:
echo 'This is an example of using tts_commandline' | text2wave -f 8000 >
'/tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav'
2021-02-26 15:19:25.433013 [WARNING] mod_sndfile.c:281 Error Opening File
[/tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav] [File contains data in
an unknown format.]
2021-02-26 15:19:25.433013 [ERR] mod_tts_commandline.c:170 Failed to open
file: /tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav
2021-02-26 15:19:25.433013 [DEBUG] switch_ivr_play_say.c:2741 Speaking
text: This is an example of using tts_commandline
2021-02-26 15:19:25.433013 [DEBUG] sofia.c:7326 Channel sofia/internal/
102 at 66.151.243.45 entering state [ready][200]
2021-02-26 15:19:25.433013 [DEBUG] switch_ivr_play_say.c:2905 done speaking
text
2021-02-26 15:19:25.433013 [NOTICE] switch_core_state_machine.c:386
sofia/internal/102 at 66.151.243.45 has executed the last dialplan
instruction, hanging up.
2021-02-26 15:19:25.433013 [NOTICE] switch_core_state_machine.c:388 Hangup
sofia/internal/102 at 66.151.243.45 [CS_EXECUTE] [NORMAL_CLEARING]

The dialplan for extension I am testing looks like this:
<extension name="tts_commandline_test">
   <condition field="destination_number" expression="^561$">
      <action application="speak" data="tts_commandline|pico|This is an
example of using tts_commandline"/>
   </condition>
</extension>

The code lines mentioned in debug seem to be pretty mundane, not that I
really know what I am looking for:
>From mod_tts_commnandline line 160:
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Executing: %s\n",
message);
I tried setting the /tmp directory to 777 but that didn't help and the
error output was identical. mod_tts_commandline never reports an error on
creation and it looks like the next part of of mod_tts_commandline would
print an error if it did like this:
if (switch_system(message, SWITCH_TRUE) < 0) {
    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Failed to
execute command: %s\n", message);
    ret = SWITCH_STATUS_FALSE; goto done;
}
if (switch_core_file_open(info->fh, info->file, 0, //number_of_channels,
    info->rate, //samples_per_second,
    SWITCH_FILE_FLAG_READ | SWITCH_FILE_DATA_SHORT, NULL) !=
SWITCH_STATUS_SUCCESS) {
    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Failed to open
file: %s\n", info->file);
    ret = SWITCH_STATUS_FALSE; goto done;
}

Starting at line 279 in mod_sndfile (sorry for the space formatting):

if (!context->handle) {
if (sndfile_perform_open(context, path, mode, handle) !=
SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Error Opening
File [%s] [%s]\n", path, sf_strerror(context->handle));
status = SWITCH_STATUS_GENERR;
goto end;
}
}

According to the dptools docs
<https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+playback#mod_dptools:playback-3.1>
and
libsndfile wav has been supported forever and it would seem highly unlikely
that is an issue. I check the /tmp directory and no files are present but I
don't know if they are being deleted by some cleanup process or just not
created. Not knowing C, it seems like mod_sndfile never received a file
handle but that is just a guess. Other Freeswitch apps are writing to the
/tmp directory (at least there are some files in there). Maybe someone sees
an error from mod_tts_commandline I don't see or knows I am doing something
stupid. Any help would be appreciated. I only found 2 references to the
error in the mail history and neither had anything to do with a
similar issue.

Freeswitch version reports: FreeSWITCH Version
1.10.5-release-17-25569c1631~64bit (-release-17-25569c1631 64bit)
FusionPBX version, in case it matters, is : 4.5.21
OS is Debian 10 Buster

-- 
Lewis Bergman