From davidswalkabout at gmail.com  Fri Oct  1 05:24:20 2021
From: davidswalkabout at gmail.com (David P)
Date: Fri, 1 Oct 2021 18:24:20 +1300
Subject: [Freeswitch-users] iOS 15 blocking WSS login on port 8082
In-Reply-To: <mailman.0.1633003201.35705.freeswitch-users@lists.freeswitch.org>
References: <mailman.0.1633003201.35705.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAE23trsMrQxbR3jFNzAi1xEHA8m7Pb=wMXtvQ931z-28pPg-HQ@mail.gmail.com>

>
> Good to know, Chris.

It's hard to tell where the official repos are for mod_verto and its js
library in github. Would you mind providing a link?

From: cmrienzo at gmail.com
>
> This is fixed in latest mod_verto.
>
> Chris


>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211001/fb698304/attachment.html>

From davidswalkabout at gmail.com  Fri Oct  1 08:33:03 2021
From: davidswalkabout at gmail.com (David P)
Date: Fri, 1 Oct 2021 21:33:03 +1300
Subject: [Freeswitch-users] iOS 15 blocking WSS login on port 8082
In-Reply-To: <CAE23trsMrQxbR3jFNzAi1xEHA8m7Pb=wMXtvQ931z-28pPg-HQ@mail.gmail.com>
References: <mailman.0.1633003201.35705.freeswitch-users@lists.freeswitch.org>
 <CAE23trsMrQxbR3jFNzAi1xEHA8m7Pb=wMXtvQ931z-28pPg-HQ@mail.gmail.com>
Message-ID: <CAE23trurSL=NotsABemjJ0es6Ga4C52Q_eG1pr2d=_vP_H2gkg@mail.gmail.com>

For example, the most recent commit here was a year ago:
https://github.com/signalwire/freeswitch/commits/master/src/mod/endpoints/mod_verto/mod_verto.c

On Fri, 1 Oct 2021, 6:24 pm David P, <davidswalkabout at gmail.com> wrote:

> Good to know, Chris.
>
> It's hard to tell where the official repos are for mod_verto and its js
> library in github. Would you mind providing a link?
>
> From: cmrienzo at gmail.com
>>
>> This is fixed in latest mod_verto.
>>
>> Chris
>
>
>>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211001/5608f9e7/attachment.html>

From s.safarov at gmail.com  Fri Oct  1 09:40:39 2021
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 1 Oct 2021 12:40:39 +0300
Subject: [Freeswitch-users] iOS 15 blocking WSS login on port 8082
In-Reply-To: <CAE23trsMrQxbR3jFNzAi1xEHA8m7Pb=wMXtvQ931z-28pPg-HQ@mail.gmail.com>
References: <mailman.0.1633003201.35705.freeswitch-users@lists.freeswitch.org>
 <CAE23trsMrQxbR3jFNzAi1xEHA8m7Pb=wMXtvQ931z-28pPg-HQ@mail.gmail.com>
Message-ID: <CAHtxdDfj0LP9qA+hO2kM=Kd5V-zjvmyzo0OFqDsYY-tSRS0Pdg@mail.gmail.com>

You can check here
https://github.com/signalwire/freeswitch/tree/master/src/mod/endpoints/mod_verto

On Fri, Oct 1, 2021 at 10:49 AM David P <davidswalkabout at gmail.com> wrote:

> Good to know, Chris.
>
> It's hard to tell where the official repos are for mod_verto and its js
> library in github. Would you mind providing a link?
>
> From: cmrienzo at gmail.com
>>
>> This is fixed in latest mod_verto.
>>
>> Chris
>
>
>> _________________________________________________________________________
>
> 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/20211001/ef90ae5e/attachment.html>

From m.shirazi at gmail.com  Mon Oct  4 13:36:14 2021
From: m.shirazi at gmail.com (Mehdi Shirazi)
Date: Mon, 4 Oct 2021 17:06:14 +0330
Subject: [Freeswitch-users] system call inside lua
Message-ID: <CAG2rjptS4P0tyY92kcHELecTWQwTWmrSHXs5Vf+u1qFwaiRoSg@mail.gmail.com>

Hi
I can run this query on linux cli:
mssql-cli -S 192.168.69.60 -d IVR_SH -U IVRSH1 -P IVRSH -Q "SELECT * FROM
"dbo"."IVR_REQEST" "
but the same query inside lua script with os.execute does not work.
Itried:
-os.execute(' mssql-cli -S 192.168.69.60 -d IVR_SH -U IVRSH1 -P IVRSH -Q
"SELECT * FROM "dbo"."IVR_REQEST" " ');
-os.execute([[ mssql-cli -S 192.168.69.60 -d IVR_SH -U IVRSH1 -P IVRSH -Q
"SELECT * FROM "dbo"."IVR_REQEST" " ]]);
...
Please help.
Regards
M.Shirazi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211004/44947c30/attachment.html>

From avi at avimarcus.net  Mon Oct  4 14:56:27 2021
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 4 Oct 2021 14:56:27 +0000
Subject: [Freeswitch-users] system call inside lua
In-Reply-To: <CAG2rjptS4P0tyY92kcHELecTWQwTWmrSHXs5Vf+u1qFwaiRoSg@mail.gmail.com>
References: <CAG2rjptS4P0tyY92kcHELecTWQwTWmrSHXs5Vf+u1qFwaiRoSg@mail.gmail.com>
Message-ID: <0100017c4bce5bce-0c542e05-2620-41db-a7dd-eef8e9f5434e-000000@email.amazonses.com>

As far as I know, these are forked system calls. Freeswitch doesn't wait
for a response.

Can you use an odbc handle for a Lua db query?

(I worked around it by having the results written to a file and using an
api to read that file.)


On Mon, Oct 4, 2021, 5:27 PM Mehdi Shirazi <m.shirazi at gmail.com> wrote:

> Hi
> I can run this query on linux cli:
> mssql-cli -S 192.168.69.60 -d IVR_SH -U IVRSH1 -P IVRSH -Q "SELECT * FROM
> "dbo"."IVR_REQEST" "
> but the same query inside lua script with os.execute does not work.
> Itried:
> -os.execute(' mssql-cli -S 192.168.69.60 -d IVR_SH -U IVRSH1 -P IVRSH -Q
> "SELECT * FROM "dbo"."IVR_REQEST" " ');
> -os.execute([[ mssql-cli -S 192.168.69.60 -d IVR_SH -U IVRSH1 -P IVRSH -Q
> "SELECT * FROM "dbo"."IVR_REQEST" " ]]);
> ...
> Please help.
> Regards
> M.Shirazi
>
> _________________________________________________________________________
>
> 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/20211004/77ef4e0d/attachment.html>

From nathan at robotics.net  Tue Oct  5 13:13:04 2021
From: nathan at robotics.net (Nathan Stratton)
Date: Tue, 5 Oct 2021 09:13:04 -0400
Subject: [Freeswitch-users] ios 15 issue
Message-ID: <CAHubkyOu08OwiuLx6ZXmboQijo0GBqaEwAyGhCqaE+H5NeZEvQ@mail.gmail.com>

We recently encountered an issue where all socket connection attempts are
repeatedly disconnected and retried upon attempting to connect to
FreeSWITCH. Freeswitch does not contain logs of these connection attempts
and they seem to occur immediately upon the initial request (likely during
the SSL handshake). A look at the Apple developer forums has yielded a
number of people experiencing the same issue, pointing to a problem with a
transition to a new NSURLSession implementation in ios15. Unfortunately,
there is no expected timeline for a fix to address these issues. This is
the link to the forum thread relevant to the problem:

https://developer.apple.com/forums/thread/685403

><>
nathan stratton
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211005/89c02cb7/attachment.html>

From brians at iptel.co  Tue Oct  5 14:39:10 2021
From: brians at iptel.co (Brian :)
Date: Tue, 5 Oct 2021 15:39:10 +0100
Subject: [Freeswitch-users] ios 15 issue
In-Reply-To: <CAHubkyOu08OwiuLx6ZXmboQijo0GBqaEwAyGhCqaE+H5NeZEvQ@mail.gmail.com>
References: <CAHubkyOu08OwiuLx6ZXmboQijo0GBqaEwAyGhCqaE+H5NeZEvQ@mail.gmail.com>
Message-ID: <CAGPQfi9OH48mE8Ub+PO_ybjtJ9Mrz=hvEODf8X9sgNDbkBQO0g@mail.gmail.com>

There's a thread with a similar issue a few days ago.

The upshot was that latest mod_verto resolves.

On Tuesday, October 5, 2021, Nathan Stratton <nathan at robotics.net> wrote:
> We recently encountered an issue where all socket connection attempts are
repeatedly disconnected and retried upon attempting to connect to
FreeSWITCH. Freeswitch does not contain logs of these connection attempts
and they seem to occur immediately upon the initial request (likely during
the SSL handshake). A look at the Apple developer forums has yielded a
number of people experiencing the same issue, pointing to a problem with a
transition to a new NSURLSession implementation in ios15. Unfortunately,
there is no expected timeline for a fix to address these issues. This is
the link to the forum thread relevant to the problem:
>
> https://developer.apple.com/forums/thread/685403
>
>><>
> nathan stratton
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211005/d05c91ee/attachment.html>

From nathan at robotics.net  Tue Oct  5 15:59:52 2021
From: nathan at robotics.net (Nathan Stratton)
Date: Tue, 5 Oct 2021 11:59:52 -0400
Subject: [Freeswitch-users] ios 15 issue
In-Reply-To: <CAGPQfi9OH48mE8Ub+PO_ybjtJ9Mrz=hvEODf8X9sgNDbkBQO0g@mail.gmail.com>
References: <CAHubkyOu08OwiuLx6ZXmboQijo0GBqaEwAyGhCqaE+H5NeZEvQ@mail.gmail.com>
 <CAGPQfi9OH48mE8Ub+PO_ybjtJ9Mrz=hvEODf8X9sgNDbkBQO0g@mail.gmail.com>
Message-ID: <CAHubkyOMZya9AKY-MhGB+uU8OhjqHjJgYf9g1vigWN=TQZU9oQ@mail.gmail.com>

We don't use mod_verto.... Just SIP over WebSockets.

><>
nathan stratton


On Tue, Oct 5, 2021 at 10:39 AM Brian : <brians at iptel.co> wrote:

> There's a thread with a similar issue a few days ago.
>
> The upshot was that latest mod_verto resolves.
>
> On Tuesday, October 5, 2021, Nathan Stratton <nathan at robotics.net> wrote:
> > We recently encountered an issue where all socket connection attempts
> are repeatedly disconnected and retried upon attempting to connect to
> FreeSWITCH. Freeswitch does not contain logs of these connection attempts
> and they seem to occur immediately upon the initial request (likely during
> the SSL handshake). A look at the Apple developer forums has yielded a
> number of people experiencing the same issue, pointing to a problem with a
> transition to a new NSURLSession implementation in ios15. Unfortunately,
> there is no expected timeline for a fix to address these issues. This is
> the link to the forum thread relevant to the problem:
> >
> > https://developer.apple.com/forums/thread/685403
> >
> >><>
> > 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/20211005/ed34ecb0/attachment.html>

From cmrienzo at gmail.com  Tue Oct  5 16:21:55 2021
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Tue, 5 Oct 2021 12:21:55 -0400
Subject: [Freeswitch-users] ios 15 issue
In-Reply-To: <CAHubkyOMZya9AKY-MhGB+uU8OhjqHjJgYf9g1vigWN=TQZU9oQ@mail.gmail.com>
References: <CAHubkyOu08OwiuLx6ZXmboQijo0GBqaEwAyGhCqaE+H5NeZEvQ@mail.gmail.com>
 <CAGPQfi9OH48mE8Ub+PO_ybjtJ9Mrz=hvEODf8X9sgNDbkBQO0g@mail.gmail.com>
 <CAHubkyOMZya9AKY-MhGB+uU8OhjqHjJgYf9g1vigWN=TQZU9oQ@mail.gmail.com>
Message-ID: <CAD-qRki02Pb+=u7v8_q07nyN2qjVgHggNJr8PYQsD5NFbi1rHQ@mail.gmail.com>

Update to latest sofia-sip

On Tue, Oct 5, 2021 at 12:18 PM Nathan Stratton <nathan at robotics.net> wrote:

> We don't use mod_verto.... Just SIP over WebSockets.
>
> ><>
> nathan stratton
>
>
> On Tue, Oct 5, 2021 at 10:39 AM Brian : <brians at iptel.co> wrote:
>
>> There's a thread with a similar issue a few days ago.
>>
>> The upshot was that latest mod_verto resolves.
>>
>> On Tuesday, October 5, 2021, Nathan Stratton <nathan at robotics.net> wrote:
>> > We recently encountered an issue where all socket connection attempts
>> are repeatedly disconnected and retried upon attempting to connect to
>> FreeSWITCH. Freeswitch does not contain logs of these connection attempts
>> and they seem to occur immediately upon the initial request (likely during
>> the SSL handshake). A look at the Apple developer forums has yielded a
>> number of people experiencing the same issue, pointing to a problem with a
>> transition to a new NSURLSession implementation in ios15. Unfortunately,
>> there is no expected timeline for a fix to address these issues. This is
>> the link to the forum thread relevant to the problem:
>> >
>> > https://developer.apple.com/forums/thread/685403
>> >
>> >><>
>> > 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211005/f8fff012/attachment-0001.html>

From avi at avimarcus.net  Tue Oct  5 19:49:39 2021
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 5 Oct 2021 19:49:39 +0000
Subject: [Freeswitch-users] Multiple api_on_answer?
Message-ID: <0100017c5201244d-d8a3f233-80b4-4c2c-a0db-f26dc9311015-000000@email.amazonses.com>

Docs show that you can have multiple execute_on, specifically:
execute_on_answer_1

Does that also work for api_on_answer?

I searched the code for execute_on_answer_1 and couldn't find that or a
loop for 1-9.

https://freeswitch.org/confluence/pages/viewpage.action?pageId=1048892#id-__ChannelVariables-Theexecute_onfamily

 Thanks,
-Avi Marcus
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211005/39821735/attachment.html>

From gregor at infomedia.si  Tue Oct  5 21:49:50 2021
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 5 Oct 2021 23:49:50 +0200
Subject: [Freeswitch-users] Multiple api_on_answer?
In-Reply-To: <0100017c5201244d-d8a3f233-80b4-4c2c-a0db-f26dc9311015-000000@email.amazonses.com>
References: <0100017c5201244d-d8a3f233-80b4-4c2c-a0db-f26dc9311015-000000@email.amazonses.com>
Message-ID: <CAC7ttFGRmV6C6oB1Labz1LJ4iuqfbLUhHLaLDdxCyECrBGNcbg@mail.gmail.com>

Yes it does work.

On Tue, 5 Oct 2021 at 21:51, Avi Marcus <avi at avimarcus.net> wrote:

> Docs show that you can have multiple execute_on, specifically:
> execute_on_answer_1
>
> Does that also work for api_on_answer?
>
> I searched the code for execute_on_answer_1 and couldn't find that or a
> loop for 1-9.
>
>
> https://freeswitch.org/confluence/pages/viewpage.action?pageId=1048892#id-__ChannelVariables-Theexecute_onfamily
>
>  Thanks,
> -Avi Marcus
> _________________________________________________________________________
>
> 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/20211005/f90bd2a4/attachment.html>

From russow at emtex.de  Tue Oct  5 09:33:42 2021
From: russow at emtex.de (Christoph Russow)
Date: Tue, 5 Oct 2021 11:33:42 +0200
Subject: [Freeswitch-users] Jitter and dropping MOS score
Message-ID: <66af870f-ebcd-ff5c-273b-1f79a1d26e5d@emtex.de>

Hi all,

we are currently experiencing jitter and dropping MOS score on an 
increasing amount of calls on our system.

i can't see any increased load on our system (neither CPU nor I/O wise).

Current stats:

~ 15Mbit/s network traffic (on a 1Gbit/s network)
load average: 1,10, 0,82, 0,91 (16 core machine)
memory used: 2,26GB/16GB

freeswitch at lc1> status
UP 0 years, 12 days, 13 hours, 30 minutes, 24 seconds, 649 milliseconds, 
831 microseconds
FreeSWITCH (Version 1.10.1 -release-12-f9990221e6 64bit) is ready
137634 session(s) since startup
218 session(s) - peak 245, last 5min 222
1 session(s) per Sec out of max 30, peak 8, last 5min 2
1000 session(s) max
min idle cpu 0.00/95.37
Current Stack Size/Max 244K/244K

System ist not virtualized.

the only "strange" thing i see is the freeswitch "main"process that 
consumes (according to htop) >=100% cpu (see attached screenshot).
Can this be the problem? if yes what can i do against it.

if not what are possible bottlenecks i should weed out?

if you need any additional informations to help me sort out this problem 
please don't hesitate to ask.

Thanks in advance and best regards
Christoph



-- 
________________________________________________________________

EMTEX GmbH
Christoph Russow
Software Engineer

Bischof-Otto-Weg 9
D-91086 Aurachtal
                           Geschäftsführer:       Markus Enzinger
Tel. +49 9132 7490 0      Sitz der Gesellschaft: 91086 Aurachtal
Fax. +49 9132 7490 900    Amtsgericht Fürth:     HRB6804
________________________________________________________________
-------------- next part --------------
A non-text attachment was scrubbed...
Name: Bildschirmfoto_2021-10-05_10-07-40.png
Type: image/png
Size: 317055 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211005/b2fe188b/attachment-0001.png>

From cary at ytel.com  Tue Oct  5 17:46:07 2021
From: cary at ytel.com (Cary Shouse)
Date: Tue, 5 Oct 2021 17:46:07 +0000
Subject: [Freeswitch-users] FreeSWITCH Stops Responding
In-Reply-To: <CAEt__=2CYMO_=65xesyBkDFd9v2PO35XusAOyWZHKC1znZ3Chw@mail.gmail.com>
References: <CAFGRPVp4Fx9Q5wYQ63PffGR4xyQL-TZ1VJMx+g6Jjpb4uEL4Vg@mail.gmail.com>
 <F73E7BB3-0415-4091-B668-4CE84565A5DD@freeswitch.org>
 <CAFGRPVqk3CXXhinmZOchy208UsX2L3kW+9vL4LQxGasdST6pKg@mail.gmail.com>
 <CAEt__=2CYMO_=65xesyBkDFd9v2PO35XusAOyWZHKC1znZ3Chw@mail.gmail.com>
Message-ID: <CAEt__=1Twd_K3k8m5YEL-aY=F_A10Rzo=eHCMk3FYG140gTwig@mail.gmail.com>

Hello,

Just sending an update on this. The Debian Freeswitch server we set up had
the same issue as our Gentoo servers. Looking at the backtrace we noticed
the first lowlevellock was occurring when or OurOutbound.luac was being
used. We decided to use a version that was not precompiled. After that
change the issue has only occurred once more. So there is still a problem
we are looking into, but the problems have been massively reduced with that
change.

Part of the backtrace
Thread 40 (Thread 0x7f2fe191e700 (LWP 15884)):
#0  __lll_lock_wait_private () at
../sysdeps/unix/sysv/linux/x86_64/lowlevellock.S:63
#1  0x00007f30512cce53 in __GI__IO_link_in (fp=fp at entry=0x7f2fec53e400) at
genops.c:93
#2  0x00007f30512cbb92 in _IO_new_file_init_internal
(fp=fp at entry=0x7f2fec53e400)
at fileops.c:114
#3  0x00007f30512c0148 in __fopen_internal (filename=0x7f2fec4d5920
"/usr/local/freeswitch/scripts/360/V3/lua/OurOutbound.luac",
mode=0x7f2fe37d81b2 "r", is32=1) at iofopen.c:74
#4  0x00007f2fe37cc123 in luaL_loadfilex () from
/lib/x86_64-linux-gnu/liblua5.2.so.0
#5  0x00007f302405796d in lua_parse_and_execute (L=<optimized out>,
input_code=<optimized out>, session=0x7f2fc91217f8) at mod_lua.cpp:194
#6  0x00007f3024058191 in lua_function (session=0x7f2fc91217f8,
data=<optimized out>) at mod_lua.cpp:483
#7  0x00007f30518920df in switch_core_session_exec
(session=session at entry=0x7f2fc91217f8,
application_interface=application_interface at entry=0x55d5e9528150,
arg=arg at entry=0x7f2fc96819c0 "360/V3/lua/OurOutbound.luac") at
src/switch_core_session.c:2900
#8  0x00007f305189276f in switch_core_session_execute_application_get_flags
(session=session at entry=0x7f2fc91217f8, app=0x7f2fc96819b8 "lua",
arg=0x7f2fc96819c0 "360/V3/lua/OurOutbound.luac", flags=flags at entry=0x0) at
src/switch_core_session.c:2765
#9  0x00007f3051896ab2 in switch_core_standard_on_execute
(session=0x7f2fc91217f8) at src/switch_core_state_machine.c:354
#10 switch_core_session_run (session=0x7f2fc91217f8) at
src/switch_core_state_machine.c:651
#11 0x00007f305188face in switch_core_session_thread (thread=<optimized
out>, obj=0x7f2fc91217f8) at src/switch_core_session.c:1719
#12 0x00007f305188b4c3 in switch_core_session_thread_pool_worker
(thread=0x7f3034236c70, obj=<optimized out>) at
src/switch_core_session.c:1782
#13 0x00007f3051bc536c in dummy_worker (opaque=0x7f3034236c70) at
threadproc/unix/thread.c:151
#14 0x00007f3051793fa3 in start_thread (arg=<optimized out>) at
pthread_create.c:486
#15 0x00007f30513494cf in clone () at
../sysdeps/unix/sysv/linux/x86_64/clone.S:95

On Thu, Sep 9, 2021 at 1:37 AM Cary Shouse <cary at ytel.com> wrote:

> We have set up a Debian 10 FreeSWITCH-1.10.6 server for testing/debugging.
>
> On Sat, Sep 4, 2021 at 5:51 PM David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> That is correct. But we who have worked with FS since around 2002 know
>> this, newer people get all worked up until they’re told this :)
>>
>> It’s a learning thing..
>>
>> On Sat, 4 Sep 2021 at 17:31, Ken Rice <krice at freeswitch.org> wrote:
>>
>>> “stuck channels” usually arent stuck once you realize things like “show
>>> calls” and “show channels” are selects against the coredb. this usually
>>> happens on busy systems timing put waiting for db delete’s to run that fail
>>> or get skipped.
>>>
>>> Sent from my iPhone
>>>
>>> On Sep 4, 2021, at 05:22, David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>> 
>>>
>>> Hello all,
>>>
>>> Interesting point about Gentoo, but I highly doubt it’s OS related. I’ve
>>> been living this issue (about those “stuck channels”, not a frozen FS,
>>> that’s indeed weird) for year and years. I’ve seen in on CentOS, Debian,
>>> Ubuntu, etc.
>>>
>>> In my experience this became an issue with the rise of VM. Back in the
>>> day you would see this every now and then, like months apart (this, or a
>>> crashed MySQL database), but wasn’t big enough to be a critical problem.
>>> Today day it’s basically an everyday thing if you have enough nodes.
>>>
>>>
>>> On Sat, 4 Sep 2021 at 00:15, Ken Rice <krice at freeswitch.org> wrote:
>>>
>>>> Having massive experience here, you might try replacing your linux
>>>> distro.
>>>>
>>>>  (Yes i know some people are going to take this as brushing off the
>>>> problem or trying to start a distro war. If this starts a distro war I will
>>>> moderate you and only allow replies are not headed to a flame war.)
>>>>
>>>> However, as someone who has ran an extremely large Gentoo based
>>>> cluster, we had to abandon gentoo due to random crashes, unexplainable bugs
>>>> that could not be reproduced or explained.
>>>>
>>>> If you can not replace the distro I VERY HiGHLY recommend running
>>>> vanilla libc and other libs without the gentoo portage included patches.
>>>> (hint the use flag: vanilla)
>>>>
>>>> Ken
>>>> Sent from my iPhone
>>>>
>>>> On Sep 3, 2021, at 14:49, Cary Shouse <cary at ytel.com> wrote:
>>>>
>>>> 
>>>>
>>>>  We have not been able to reproduce this in testing. It only happens in our production environment.
>>>>  Are there any other debug options that could be enabled, or is there any code in FreeSWITCH that you think may be causing this that we could look into.  We could get you access to a server if that would help.
>>>> Server Details:
>>>> Gentoo-5.10.27
>>>> RAM: 15GB
>>>> CPU: 4
>>>> FreeSWITCH-1-.10.6
>>>> Sqlite-3.34.1
>>>>
>>>>
>>>> On Thu, Sep 2, 2021 at 7:24 PM Brian West <brian at freeswitch.com> wrote:
>>>>
>>>>> And these exhibit the issue on command?
>>>>>
>>>>> On Wed, Sep 1, 2021 at 5:06 PM Cary Shouse <cary at ytel.com> wrote:
>>>>>
>>>>>> Here are two originate samples.
>>>>>>
>>>>>> https://pastebin.com/YKRKrJLu
>>>>>>
>>>>>> https://pastebin.com/TEX3yfX6
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Wed, Sep 1, 2021 at 8:00 PM Brian West <brian at freeswitch.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Can you try to create one where this happens without all this mess?
>>>>>>>
>>>>>>> On Wed, Sep 1, 2021 at 2:41 PM Cary Shouse <cary at ytel.com> wrote:
>>>>>>>
>>>>>>>> Here is the LUA code regarding creating sessions and bridging. It
>>>>>>>> is broken up into 5 sections.
>>>>>>>> The top part is the common functions, and the 4 sections below are
>>>>>>>> different call scenarios.
>>>>>>>> https://pastebin.com/qqGw3tms
>>>>>>>>
>>>>>>>>
>>>>>>>> On Wed, Sep 1, 2021 at 3:53 PM Brian West <brian at freeswitch.com>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>>> I you do sessions from your lua, I need to see how you do that.
>>>>>>>>>
>>>>>>>>> On Tue, Aug 31, 2021 at 4:20 PM Cary Shouse <cary at ytel.com> wrote:
>>>>>>>>>
>>>>>>>>>> Yes, it bridges the inbound call to an outbound one.
>>>>>>>>>>
>>>>>>>>>> (ProxyOutbound.luac)https://pastebin.com/RDUMy473
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Tue, Aug 31, 2021 at 8:39 PM Brian West <brian at freeswitch.com>
>>>>>>>>>> wrote:
>>>>>>>>>>
>>>>>>>>>>> Guessing you create sessions in lua? ProxyOutbound.luac?
>>>>>>>>>>>
>>>>>>>>>>> On Tue, Aug 31, 2021 at 3:33 PM Brian West <brian at freeswitch.com>
>>>>>>>>>>> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> and to make sure when you uuid kill one it says it didn't exist
>>>>>>>>>>>> but show channels had it still listed correct?
>>>>>>>>>>>>
>>>>>>>>>>>> On Tue, Aug 31, 2021 at 2:58 PM Cary Shouse <cary at ytel.com>
>>>>>>>>>>>> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> Hello,
>>>>>>>>>>>>>
>>>>>>>>>>>>> We have got the logs for a few calls that were stuck.
>>>>>>>>>>>>> (Show channels fe8f206c-2bd9-4af2-b329-35f53228a38a)
>>>>>>>>>>>>> https://pastebin.com/NTnte7J7
>>>>>>>>>>>>>
>>>>>>>>>>>>> (Extracted from logs fe8f206c-2bd9-4af2-b329-35f53228a38a)
>>>>>>>>>>>>> https://pastebin.com/sAhFiGjY
>>>>>>>>>>>>>
>>>>>>>>>>>>> (Show channels 46c6e04b-a58e-4b82-9968-fb0e9c2c0fb3)
>>>>>>>>>>>>> https://pastebin.com/bmcdjzAr
>>>>>>>>>>>>>
>>>>>>>>>>>>> (Extracted from logs 46c6e04b-a58e-4b82-9968-fb0e9c2c0fb3)
>>>>>>>>>>>>> https://pastebin.com/aGyY3m9F
>>>>>>>>>>>>>
>>>>>>>>>>>>> (Show channels 624cac90-0a7b-11ec-b7d0-6f543a588268)
>>>>>>>>>>>>> https://pastebin.com/0cnQaRhC
>>>>>>>>>>>>>
>>>>>>>>>>>>> (Extracted from logs 624cac90-0a7b-11ec-b7d0-6f543a588268)
>>>>>>>>>>>>> https://pastebin.com/atX0VXdZ
>>>>>>>>>>>>>
>>>>>>>>>>>>> Please let us know if there is anything else we can provide
>>>>>>>>>>>>> that would be useful. Thank you.
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Mon, Aug 30, 2021 at 8:20 PM Brian West <
>>>>>>>>>>>>> brian at freeswitch.com> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> Check the UUID of the session in question, and extract its
>>>>>>>>>>>>>> logs, and let's diagnose it
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Mon, Aug 30, 2021 at 12:39 PM Cary Shouse <cary at ytel.com>
>>>>>>>>>>>>>> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>  Yes, that is what we are seeing when running uuid_kill.  We
>>>>>>>>>>>>>>> cannot "replicate" the issue, but it happens multiple times a day for us
>>>>>>>>>>>>>>> across our servers.   We have looked through logs across
>>>>>>>>>>>>>>> multiple systems, taken SIP traces to see if that showed anything.
>>>>>>>>>>>>>>> Last we got a Core dump of the running process after the
>>>>>>>>>>>>>>> failure. Nothing seems to stand out as the cause.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>  Are there any other debug steps you can think of or what
>>>>>>>>>>>>>>> other information to look at?  We could capture all traffic on the server,
>>>>>>>>>>>>>>> or recompile with special debug flags whatever it is that might help.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Steps we have taken:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> sofia loglevel all 9
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> sofia global siptrace on
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> fsctl loglevel 7
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> # Set environment variable
>>>>>>>>>>>>>>> TPORT_LOG=1
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Sat, Aug 28, 2021 at 12:54 AM Brian West <
>>>>>>>>>>>>>>> brian at freeswitch.com> wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> It's an issue we've been searching for, if you uuid_kill
>>>>>>>>>>>>>>>> it, and it says no session, it only means the record was NOT removed from
>>>>>>>>>>>>>>>> the sqlite database.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Nobody can seem to replicate the steps required to get into
>>>>>>>>>>>>>>>> that condition, maybe you could look thru the logs and see if you can
>>>>>>>>>>>>>>>> replicate what ever happened on that lingering UUID
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> /b
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> On Fri, Aug 27, 2021 at 5:40 PM Cary Shouse <cary at ytel.com>
>>>>>>>>>>>>>>>> wrote:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>  We recently started having a problem with our FreeSWITCH
>>>>>>>>>>>>>>>>> servers where calls are working well, with about 200 calls operating
>>>>>>>>>>>>>>>>> sometimes for a couple of days, and then something happens and all call
>>>>>>>>>>>>>>>>> progress stops. Then any new inbound calls that come into the server get
>>>>>>>>>>>>>>>>> stuck in initial status, new inbound get stuck at RINGING. This happens
>>>>>>>>>>>>>>>>> until the max sessions are hit and of course nothing else works from there.
>>>>>>>>>>>>>>>>> Our thought is something is creating a lock and stopping anything else from
>>>>>>>>>>>>>>>>> progressing.
>>>>>>>>>>>>>>>>>  We are seeing the stuck sessions using "show channels".
>>>>>>>>>>>>>>>>> Doing a uuid_kill on the calls seems to make no difference to the output of
>>>>>>>>>>>>>>>>> "show channels", but we have noticed in the logs that some events do happen
>>>>>>>>>>>>>>>>> to some of the calls.
>>>>>>>>>>>>>>>>>  We had ran into a similar issue when writing a custom C
>>>>>>>>>>>>>>>>> module and made some mistakes with session_rwlock etc. But in this case,
>>>>>>>>>>>>>>>>> all of this is using LUA.
>>>>>>>>>>>>>>>>> Some general information:
>>>>>>>>>>>>>>>>> FreeSWITCH Versions 1.10.6 and 1.10.1
>>>>>>>>>>>>>>>>> Modules used:
>>>>>>>>>>>>>>>>> LUA, xml_curl(dialplan), event_socket,
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>  Most of this is just application playing audio files and
>>>>>>>>>>>>>>>>> or creating a new call to bridge to.
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>  Really looking for some help on what debugging we can
>>>>>>>>>>>>>>>>> enable to help identify what is getting stuck to identify where a bug might
>>>>>>>>>>>>>>>>> be on our usage or possible issue with FreeSWITCH.
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>  We have captured a core dump of the running process when
>>>>>>>>>>>>>>>>> it got into the bad state as well, if that is helpful, we can send it in.
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Thank you for your help.
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> 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>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> 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>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> --
>>>>>>>>>>>>
>>>>>>>>>>>> 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>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> --
>>>>>>>>>>>
>>>>>>>>>>> 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>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>>
>>>>>>>>> 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>
>>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>>
>>>>>>> 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>
>>>>>>>
>>>>>>
>>>>>
>>>>> --
>>>>>
>>>>> 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
>>>
>>> --
>>> 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
>>
>> --
>> 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/20211005/89901890/attachment-0001.html>

From david.villasmil.work at gmail.com  Tue Oct  5 23:42:46 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 6 Oct 2021 00:42:46 +0100
Subject: [Freeswitch-users] stop channel from entering conference
Message-ID: <CAFGRPVom2RvjFoyoE2mmwLFtDsXyxM_YPn35Fb0J9xmP5xjBTQ@mail.gmail.com>

Hello all,

I have a scenario where:

A calls B (i.e.: +19995551212) and puts it on hold.
A calls C (i.e.: +99911111) and when it answers:
The client sends a refer to B to connect to a conference, and then
transfers C to the conference.

All this works well, but i need to block specific called numbers from
entering the conference.
I can identify the channels as they come in and i'm trying to hangup the
leg i want.

I tried with

<action application="hangup"/>

and also with

<action application="set" data="res=${uuid_kill(${uuid})}"/>

but all channels are disconnected.

Not sure if i'm doing something wrong.

Any help would be appreciated

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/20211006/4d9fc0e4/attachment.html>

From achinthau at gmail.com  Wed Oct  6 02:39:28 2021
From: achinthau at gmail.com (Achintha)
Date: Wed, 6 Oct 2021 08:09:28 +0530
Subject: [Freeswitch-users] jitters generated from freeswitch to trunk in
	conference call
Message-ID: <CA+JwzxttC7AjrrR05Ji9+BX4Jv_UAsoYOR1Gi9Pe4VXV_t2yyw@mail.gmail.com>

Hi all,

I configured 2 freeswitch servers (FreeSWITCH version:
1.10.5-release-17-25569c1631~64bit) on Debian 10.4 as media servers. one
opensips server is located in front of the freeswitch servers and opensips
act as SIP load balancer. media directly connect with sip trunk and
freeswitch.We use dynamic dialplan to generate calls and use g711.
Anyway i generate below mentioned scenario

   1. registered as extension
   2. dial outbound to mobile call through the sip trunk and talk
   3. put hold the first call
   4. dial another outbound call to another mobile through the same sip
trunk and talk.
   5. connect both calls (conference).
in the 5th step jitter buffers generated from freeswitch to trunk. voice
not clear.
I have changed codecs and tests (opus,G711a,G711u)
but the issue is not fixed. how to solve this problem.

Best Regards..
Achintha Udukumbura
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211006/13e727ed/attachment.html>

From dragos at freeswitch.org  Wed Oct  6 10:25:25 2021
From: dragos at freeswitch.org (Dragos Oancea)
Date: Wed, 6 Oct 2021 13:25:25 +0300
Subject: [Freeswitch-users] jitters generated from freeswitch to trunk
 in conference call
In-Reply-To: <CA+JwzxttC7AjrrR05Ji9+BX4Jv_UAsoYOR1Gi9Pe4VXV_t2yyw@mail.gmail.com>
References: <CA+JwzxttC7AjrrR05Ji9+BX4Jv_UAsoYOR1Gi9Pe4VXV_t2yyw@mail.gmail.com>
Message-ID: <CAGvJV4rF8pHJ0joSZ5hMr=i__dyHSUuM-hn28SRe3mTwU33mig@mail.gmail.com>

Make sure you're using a timer.
eg: <param name="rtp-timer-name" value="soft"/>

"jitter buffers generated from freeswitch to trunk" - does this mean you
noticed jitter or you enabled the jitter buffer in FS ?
If you don't use the jitter buffer, start using it.
https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer


On Wed, Oct 6, 2021 at 5:40 AM Achintha <achinthau at gmail.com> wrote:

> Hi all,
>
> I configured 2 freeswitch servers (FreeSWITCH version:
> 1.10.5-release-17-25569c1631~64bit) on Debian 10.4 as media servers. one
> opensips server is located in front of the freeswitch servers and opensips
> act as SIP load balancer. media directly connect with sip trunk and
> freeswitch.We use dynamic dialplan to generate calls and use g711.
> Anyway i generate below mentioned scenario
>
>    1. registered as extension
>    2. dial outbound to mobile call through the sip trunk and talk
>    3. put hold the first call
>    4. dial another outbound call to another mobile through the same sip
> trunk and talk.
>    5. connect both calls (conference).
> in the 5th step jitter buffers generated from freeswitch to trunk. voice
> not clear.
> I have changed codecs and tests (opus,G711a,G711u)
> but the issue is not fixed. how to solve this problem.
>
> Best Regards..
> Achintha Udukumbura
> _________________________________________________________________________
>
> 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/20211006/351439c3/attachment.html>

From dragos at freeswitch.org  Wed Oct  6 10:33:02 2021
From: dragos at freeswitch.org (Dragos Oancea)
Date: Wed, 6 Oct 2021 13:33:02 +0300
Subject: [Freeswitch-users] Jitter and dropping MOS score
In-Reply-To: <66af870f-ebcd-ff5c-273b-1f79a1d26e5d@emtex.de>
References: <66af870f-ebcd-ff5c-273b-1f79a1d26e5d@emtex.de>
Message-ID: <CAGvJV4oueJAr-m8GHn1nid1ATwvpd+fnvOsdj75pk_5mzUpc4g@mail.gmail.com>

if it really consumes 100% CPU with few calls then it's abnormal.
kill it with SIGABRT (make sure it will dump core before that), get a
backtrace with gdb and open a github issue.
before doing this you need to make sure it's really eating 100 % CPU, this
screenshot is not very relevant.
Also, you need to make sure you don't have real packet loss. Perhaps it's a
matter of dimensioning, eg: how many simultaneous calls do you expect to
have on that hardware, what codec, etc.


On Wed, Oct 6, 2021 at 2:26 AM Christoph Russow <russow at emtex.de> wrote:

> Hi all,
>
> we are currently experiencing jitter and dropping MOS score on an
> increasing amount of calls on our system.
>
> i can't see any increased load on our system (neither CPU nor I/O wise).
>
> Current stats:
>
> ~ 15Mbit/s network traffic (on a 1Gbit/s network)
> load average: 1,10, 0,82, 0,91 (16 core machine)
> memory used: 2,26GB/16GB
>
> freeswitch at lc1> status
> UP 0 years, 12 days, 13 hours, 30 minutes, 24 seconds, 649 milliseconds,
> 831 microseconds
> FreeSWITCH (Version 1.10.1 -release-12-f9990221e6 64bit) is ready
> 137634 session(s) since startup
> 218 session(s) - peak 245, last 5min 222
> 1 session(s) per Sec out of max 30, peak 8, last 5min 2
> 1000 session(s) max
> min idle cpu 0.00/95.37
> Current Stack Size/Max 244K/244K
>
> System ist not virtualized.
>
> the only "strange" thing i see is the freeswitch "main"process that
> consumes (according to htop) >=100% cpu (see attached screenshot).
> Can this be the problem? if yes what can i do against it.
>
> if not what are possible bottlenecks i should weed out?
>
> if you need any additional informations to help me sort out this problem
> please don't hesitate to ask.
>
> Thanks in advance and best regards
> Christoph
>
>
>
> --
> ________________________________________________________________
>
> EMTEX GmbH
> Christoph Russow
> Software Engineer
>
> Bischof-Otto-Weg 9
> D-91086 Aurachtal
>                            Geschäftsführer:       Markus Enzinger
> Tel. +49 9132 7490 0      Sitz der Gesellschaft: 91086 Aurachtal
> Fax. +49 9132 7490 900    Amtsgericht Fürth:     HRB6804
>
> _________________________________________________________________________________________________________________________________________
>
> 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/20211006/fbe10e75/attachment-0001.html>

From david.villasmil.work at gmail.com  Wed Oct  6 10:33:09 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 6 Oct 2021 11:33:09 +0100
Subject: [Freeswitch-users] Jitter and dropping MOS score
In-Reply-To: <66af870f-ebcd-ff5c-273b-1f79a1d26e5d@emtex.de>
References: <66af870f-ebcd-ff5c-273b-1f79a1d26e5d@emtex.de>
Message-ID: <CAFGRPVpP_1e4S84bziLJR6f410OnuL=Ryf6TctNw_qzeMUXr6A@mail.gmail.com>

First thing to look at is check your lua scripts if you have any. It is
very strange for fs to take 100% of cpu on such low loads. So check your
call flows.

On Wed, 6 Oct 2021 at 00:58, Christoph Russow <russow at emtex.de> wrote:

> Hi all,
>
> we are currently experiencing jitter and dropping MOS score on an
> increasing amount of calls on our system.
>
> i can't see any increased load on our system (neither CPU nor I/O wise).
>
> Current stats:
>
> ~ 15Mbit/s network traffic (on a 1Gbit/s network)
> load average: 1,10, 0,82, 0,91 (16 core machine)
> memory used: 2,26GB/16GB
>
> freeswitch at lc1> status
> UP 0 years, 12 days, 13 hours, 30 minutes, 24 seconds, 649 milliseconds,
> 831 microseconds
> FreeSWITCH (Version 1.10.1 -release-12-f9990221e6 64bit) is ready
> 137634 session(s) since startup
> 218 session(s) - peak 245, last 5min 222
> 1 session(s) per Sec out of max 30, peak 8, last 5min 2
> 1000 session(s) max
> min idle cpu 0.00/95.37
> Current Stack Size/Max 244K/244K
>
> System ist not virtualized.
>
> the only "strange" thing i see is the freeswitch "main"process that
> consumes (according to htop) >=100% cpu (see attached screenshot).
> Can this be the problem? if yes what can i do against it.
>
> if not what are possible bottlenecks i should weed out?
>
> if you need any additional informations to help me sort out this problem
> please don't hesitate to ask.
>
> Thanks in advance and best regards
> Christoph
>
>
>
> --
> ________________________________________________________________
>
> EMTEX GmbH
> Christoph Russow
> Software Engineer
>
> Bischof-Otto-Weg 9
> D-91086 Aurachtal
>                            Geschäftsführer:       Markus Enzinger
> Tel. +49 9132 7490 0      Sitz der Gesellschaft: 91086 Aurachtal
> Fax. +49 9132 7490 900    Amtsgericht Fürth:     HRB6804
>
> _________________________________________________________________________________________________________________________________________
>
> 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/20211006/a615bdda/attachment.html>

From dragos at freeswitch.org  Wed Oct  6 10:36:26 2021
From: dragos at freeswitch.org (Dragos Oancea)
Date: Wed, 6 Oct 2021 13:36:26 +0300
Subject: [Freeswitch-users] FreeSWITCH Stops Responding
In-Reply-To: <CAEt__=1Twd_K3k8m5YEL-aY=F_A10Rzo=eHCMk3FYG140gTwig@mail.gmail.com>
References: <CAFGRPVp4Fx9Q5wYQ63PffGR4xyQL-TZ1VJMx+g6Jjpb4uEL4Vg@mail.gmail.com>
 <F73E7BB3-0415-4091-B668-4CE84565A5DD@freeswitch.org>
 <CAFGRPVqk3CXXhinmZOchy208UsX2L3kW+9vL4LQxGasdST6pKg@mail.gmail.com>
 <CAEt__=2CYMO_=65xesyBkDFd9v2PO35XusAOyWZHKC1znZ3Chw@mail.gmail.com>
 <CAEt__=1Twd_K3k8m5YEL-aY=F_A10Rzo=eHCMk3FYG140gTwig@mail.gmail.com>
Message-ID: <CAGvJV4q1Aq9TJqwxzjYnkVxbCNxBCvf4k+KvuitNmLBXUDHobw@mail.gmail.com>

Thanks for the update.

On Wed, Oct 6, 2021 at 2:28 AM Cary Shouse <cary at ytel.com> wrote:

> Hello,
>
> Just sending an update on this. The Debian Freeswitch server we set up had
> the same issue as our Gentoo servers. Looking at the backtrace we noticed
> the first lowlevellock was occurring when or OurOutbound.luac was being
> used. We decided to use a version that was not precompiled. After that
> change the issue has only occurred once more. So there is still a problem
> we are looking into, but the problems have been massively reduced with that
> change.
>
> Part of the backtrace
> Thread 40 (Thread 0x7f2fe191e700 (LWP 15884)):
> #0  __lll_lock_wait_private () at
> ../sysdeps/unix/sysv/linux/x86_64/lowlevellock.S:63
> #1  0x00007f30512cce53 in __GI__IO_link_in (fp=fp at entry=0x7f2fec53e400)
> at genops.c:93
> #2  0x00007f30512cbb92 in _IO_new_file_init_internal (fp=fp at entry=0x7f2fec53e400)
> at fileops.c:114
> #3  0x00007f30512c0148 in __fopen_internal (filename=0x7f2fec4d5920
> "/usr/local/freeswitch/scripts/360/V3/lua/OurOutbound.luac",
> mode=0x7f2fe37d81b2 "r", is32=1) at iofopen.c:74
> #4  0x00007f2fe37cc123 in luaL_loadfilex () from
> /lib/x86_64-linux-gnu/liblua5.2.so.0
> #5  0x00007f302405796d in lua_parse_and_execute (L=<optimized out>,
> input_code=<optimized out>, session=0x7f2fc91217f8) at mod_lua.cpp:194
> #6  0x00007f3024058191 in lua_function (session=0x7f2fc91217f8,
> data=<optimized out>) at mod_lua.cpp:483
> #7  0x00007f30518920df in switch_core_session_exec (session=session at entry=0x7f2fc91217f8,
> application_interface=application_interface at entry=0x55d5e9528150,
> arg=arg at entry=0x7f2fc96819c0 "360/V3/lua/OurOutbound.luac") at
> src/switch_core_session.c:2900
> #8  0x00007f305189276f in
> switch_core_session_execute_application_get_flags (session=session at entry=0x7f2fc91217f8,
> app=0x7f2fc96819b8 "lua", arg=0x7f2fc96819c0 "360/V3/lua/OurOutbound.luac",
> flags=flags at entry=0x0) at src/switch_core_session.c:2765
> #9  0x00007f3051896ab2 in switch_core_standard_on_execute
> (session=0x7f2fc91217f8) at src/switch_core_state_machine.c:354
> #10 switch_core_session_run (session=0x7f2fc91217f8) at
> src/switch_core_state_machine.c:651
> #11 0x00007f305188face in switch_core_session_thread (thread=<optimized
> out>, obj=0x7f2fc91217f8) at src/switch_core_session.c:1719
> #12 0x00007f305188b4c3 in switch_core_session_thread_pool_worker
> (thread=0x7f3034236c70, obj=<optimized out>) at
> src/switch_core_session.c:1782
> #13 0x00007f3051bc536c in dummy_worker (opaque=0x7f3034236c70) at
> threadproc/unix/thread.c:151
> #14 0x00007f3051793fa3 in start_thread (arg=<optimized out>) at
> pthread_create.c:486
> #15 0x00007f30513494cf in clone () at
> ../sysdeps/unix/sysv/linux/x86_64/clone.S:95
>
> On Thu, Sep 9, 2021 at 1:37 AM Cary Shouse <cary at ytel.com> wrote:
>
>> We have set up a Debian 10 FreeSWITCH-1.10.6 server for
>> testing/debugging.
>>
>> On Sat, Sep 4, 2021 at 5:51 PM David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> That is correct. But we who have worked with FS since around 2002 know
>>> this, newer people get all worked up until they’re told this :)
>>>
>>> It’s a learning thing..
>>>
>>> On Sat, 4 Sep 2021 at 17:31, Ken Rice <krice at freeswitch.org> wrote:
>>>
>>>> “stuck channels” usually arent stuck once you realize things like “show
>>>> calls” and “show channels” are selects against the coredb. this usually
>>>> happens on busy systems timing put waiting for db delete’s to run that fail
>>>> or get skipped.
>>>>
>>>> Sent from my iPhone
>>>>
>>>> On Sep 4, 2021, at 05:22, David Villasmil <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>> 
>>>>
>>>> Hello all,
>>>>
>>>> Interesting point about Gentoo, but I highly doubt it’s OS related.
>>>> I’ve been living this issue (about those “stuck channels”, not a frozen FS,
>>>> that’s indeed weird) for year and years. I’ve seen in on CentOS, Debian,
>>>> Ubuntu, etc.
>>>>
>>>> In my experience this became an issue with the rise of VM. Back in the
>>>> day you would see this every now and then, like months apart (this, or a
>>>> crashed MySQL database), but wasn’t big enough to be a critical problem.
>>>> Today day it’s basically an everyday thing if you have enough nodes.
>>>>
>>>>
>>>> On Sat, 4 Sep 2021 at 00:15, Ken Rice <krice at freeswitch.org> wrote:
>>>>
>>>>> Having massive experience here, you might try replacing your linux
>>>>> distro.
>>>>>
>>>>>  (Yes i know some people are going to take this as brushing off the
>>>>> problem or trying to start a distro war. If this starts a distro war I will
>>>>> moderate you and only allow replies are not headed to a flame war.)
>>>>>
>>>>> However, as someone who has ran an extremely large Gentoo based
>>>>> cluster, we had to abandon gentoo due to random crashes, unexplainable bugs
>>>>> that could not be reproduced or explained.
>>>>>
>>>>> If you can not replace the distro I VERY HiGHLY recommend running
>>>>> vanilla libc and other libs without the gentoo portage included patches.
>>>>> (hint the use flag: vanilla)
>>>>>
>>>>> Ken
>>>>> Sent from my iPhone
>>>>>
>>>>> On Sep 3, 2021, at 14:49, Cary Shouse <cary at ytel.com> wrote:
>>>>>
>>>>> 
>>>>>
>>>>>  We have not been able to reproduce this in testing. It only happens in our production environment.
>>>>>  Are there any other debug options that could be enabled, or is there any code in FreeSWITCH that you think may be causing this that we could look into.  We could get you access to a server if that would help.
>>>>> Server Details:
>>>>> Gentoo-5.10.27
>>>>> RAM: 15GB
>>>>> CPU: 4
>>>>> FreeSWITCH-1-.10.6
>>>>> Sqlite-3.34.1
>>>>>
>>>>>
>>>>> On Thu, Sep 2, 2021 at 7:24 PM Brian West <brian at freeswitch.com>
>>>>> wrote:
>>>>>
>>>>>> And these exhibit the issue on command?
>>>>>>
>>>>>> On Wed, Sep 1, 2021 at 5:06 PM Cary Shouse <cary at ytel.com> wrote:
>>>>>>
>>>>>>> Here are two originate samples.
>>>>>>>
>>>>>>> https://pastebin.com/YKRKrJLu
>>>>>>>
>>>>>>> https://pastebin.com/TEX3yfX6
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Sep 1, 2021 at 8:00 PM Brian West <brian at freeswitch.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Can you try to create one where this happens without all this mess?
>>>>>>>>
>>>>>>>> On Wed, Sep 1, 2021 at 2:41 PM Cary Shouse <cary at ytel.com> wrote:
>>>>>>>>
>>>>>>>>> Here is the LUA code regarding creating sessions and bridging. It
>>>>>>>>> is broken up into 5 sections.
>>>>>>>>> The top part is the common functions, and the 4 sections below are
>>>>>>>>> different call scenarios.
>>>>>>>>> https://pastebin.com/qqGw3tms
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Wed, Sep 1, 2021 at 3:53 PM Brian West <brian at freeswitch.com>
>>>>>>>>> wrote:
>>>>>>>>>
>>>>>>>>>> I you do sessions from your lua, I need to see how you do that.
>>>>>>>>>>
>>>>>>>>>> On Tue, Aug 31, 2021 at 4:20 PM Cary Shouse <cary at ytel.com>
>>>>>>>>>> wrote:
>>>>>>>>>>
>>>>>>>>>>> Yes, it bridges the inbound call to an outbound one.
>>>>>>>>>>>
>>>>>>>>>>> (ProxyOutbound.luac)https://pastebin.com/RDUMy473
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On Tue, Aug 31, 2021 at 8:39 PM Brian West <brian at freeswitch.com>
>>>>>>>>>>> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Guessing you create sessions in lua? ProxyOutbound.luac?
>>>>>>>>>>>>
>>>>>>>>>>>> On Tue, Aug 31, 2021 at 3:33 PM Brian West <
>>>>>>>>>>>> brian at freeswitch.com> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> and to make sure when you uuid kill one it says it didn't
>>>>>>>>>>>>> exist but show channels had it still listed correct?
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Tue, Aug 31, 2021 at 2:58 PM Cary Shouse <cary at ytel.com>
>>>>>>>>>>>>> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> Hello,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> We have got the logs for a few calls that were stuck.
>>>>>>>>>>>>>> (Show channels fe8f206c-2bd9-4af2-b329-35f53228a38a)
>>>>>>>>>>>>>> https://pastebin.com/NTnte7J7
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> (Extracted from logs fe8f206c-2bd9-4af2-b329-35f53228a38a)
>>>>>>>>>>>>>> https://pastebin.com/sAhFiGjY
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> (Show channels 46c6e04b-a58e-4b82-9968-fb0e9c2c0fb3)
>>>>>>>>>>>>>> https://pastebin.com/bmcdjzAr
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> (Extracted from logs 46c6e04b-a58e-4b82-9968-fb0e9c2c0fb3)
>>>>>>>>>>>>>> https://pastebin.com/aGyY3m9F
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> (Show channels 624cac90-0a7b-11ec-b7d0-6f543a588268)
>>>>>>>>>>>>>> https://pastebin.com/0cnQaRhC
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> (Extracted from logs 624cac90-0a7b-11ec-b7d0-6f543a588268)
>>>>>>>>>>>>>> https://pastebin.com/atX0VXdZ
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Please let us know if there is anything else we can provide
>>>>>>>>>>>>>> that would be useful. Thank you.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Mon, Aug 30, 2021 at 8:20 PM Brian West <
>>>>>>>>>>>>>> brian at freeswitch.com> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Check the UUID of the session in question, and extract its
>>>>>>>>>>>>>>> logs, and let's diagnose it
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Mon, Aug 30, 2021 at 12:39 PM Cary Shouse <cary at ytel.com>
>>>>>>>>>>>>>>> wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>  Yes, that is what we are seeing when running uuid_kill.  We
>>>>>>>>>>>>>>>> cannot "replicate" the issue, but it happens multiple times a day for us
>>>>>>>>>>>>>>>> across our servers.   We have looked through logs across
>>>>>>>>>>>>>>>> multiple systems, taken SIP traces to see if that showed anything.
>>>>>>>>>>>>>>>> Last we got a Core dump of the running process after the
>>>>>>>>>>>>>>>> failure. Nothing seems to stand out as the cause.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>  Are there any other debug steps you can think of or what
>>>>>>>>>>>>>>>> other information to look at?  We could capture all traffic on the server,
>>>>>>>>>>>>>>>> or recompile with special debug flags whatever it is that might help.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Steps we have taken:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> sofia loglevel all 9
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> sofia global siptrace on
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> fsctl loglevel 7
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> # Set environment variable
>>>>>>>>>>>>>>>> TPORT_LOG=1
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> On Sat, Aug 28, 2021 at 12:54 AM Brian West <
>>>>>>>>>>>>>>>> brian at freeswitch.com> wrote:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> It's an issue we've been searching for, if you uuid_kill
>>>>>>>>>>>>>>>>> it, and it says no session, it only means the record was NOT removed from
>>>>>>>>>>>>>>>>> the sqlite database.
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Nobody can seem to replicate the steps required to get
>>>>>>>>>>>>>>>>> into that condition, maybe you could look thru the logs and see if you can
>>>>>>>>>>>>>>>>> replicate what ever happened on that lingering UUID
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> /b
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> On Fri, Aug 27, 2021 at 5:40 PM Cary Shouse <cary at ytel.com>
>>>>>>>>>>>>>>>>> wrote:
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>  We recently started having a problem with our FreeSWITCH
>>>>>>>>>>>>>>>>>> servers where calls are working well, with about 200 calls operating
>>>>>>>>>>>>>>>>>> sometimes for a couple of days, and then something happens and all call
>>>>>>>>>>>>>>>>>> progress stops. Then any new inbound calls that come into the server get
>>>>>>>>>>>>>>>>>> stuck in initial status, new inbound get stuck at RINGING. This happens
>>>>>>>>>>>>>>>>>> until the max sessions are hit and of course nothing else works from there.
>>>>>>>>>>>>>>>>>> Our thought is something is creating a lock and stopping anything else from
>>>>>>>>>>>>>>>>>> progressing.
>>>>>>>>>>>>>>>>>>  We are seeing the stuck sessions using "show channels".
>>>>>>>>>>>>>>>>>> Doing a uuid_kill on the calls seems to make no difference to the output of
>>>>>>>>>>>>>>>>>> "show channels", but we have noticed in the logs that some events do happen
>>>>>>>>>>>>>>>>>> to some of the calls.
>>>>>>>>>>>>>>>>>>  We had ran into a similar issue when writing a custom C
>>>>>>>>>>>>>>>>>> module and made some mistakes with session_rwlock etc. But in this case,
>>>>>>>>>>>>>>>>>> all of this is using LUA.
>>>>>>>>>>>>>>>>>> Some general information:
>>>>>>>>>>>>>>>>>> FreeSWITCH Versions 1.10.6 and 1.10.1
>>>>>>>>>>>>>>>>>> Modules used:
>>>>>>>>>>>>>>>>>> LUA, xml_curl(dialplan), event_socket,
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>  Most of this is just application playing audio files and
>>>>>>>>>>>>>>>>>> or creating a new call to bridge to.
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>  Really looking for some help on what debugging we can
>>>>>>>>>>>>>>>>>> enable to help identify what is getting stuck to identify where a bug might
>>>>>>>>>>>>>>>>>> be on our usage or possible issue with FreeSWITCH.
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>  We have captured a core dump of the running process when
>>>>>>>>>>>>>>>>>> it got into the bad state as well, if that is helpful, we can send it in.
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> Thank you for your help.
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> 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>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> 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>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> --
>>>>>>>>>>>>>
>>>>>>>>>>>>> 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>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> --
>>>>>>>>>>>>
>>>>>>>>>>>> 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>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>>
>>>>>>>>>> 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>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>>
>>>>>>>> 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>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>> 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
>>>>
>>>> --
>>>> 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
>>>
>>> --
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.work at gmail.com
>>> phone: +34669448337
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211006/fe3ac35c/attachment-0001.html>

From avi at avimarcus.net  Wed Oct  6 12:32:22 2021
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 6 Oct 2021 12:32:22 +0000
Subject: [Freeswitch-users] Bridge to other FS server has no audio ???
Message-ID: <0100017c55972813-4ae9de8a-588c-46ec-ac70-7e2ce2230bf2-000000@email.amazonses.com>

Any ideas on why a call doesn't have media? It used to work, but I think my
upstream changed his SDP again.

- FreeSWITCH Server A - call comes in and bypass_media bridges to FS server
B. Media works.
- FreeSWITCH Server A - call comes in and bridges to FS server B (not on
bypass). Media works.
- FreeSWITCH Server A - call comes in, gets answered, then bridges to FS
server B. Call looks OK, but no media is flowing (I don't hear anything,
PCAPs just have SIP, and there isn't 80kbps network traffic). All the same
codecs are set in the json cdrs (PCMU).

FS server B is to join a conference if that matters.

I was assuming it had to do with codecs, but setting absolute_codec_string
to PCMU doesn't make any difference in the logs  - it's already always PCMU.

I have NO clue what further could cause this other than codecs, which seem
to be fine. Any ideas please?


-Avi Marcus
1-718-989-9485 (USA)
02-372-1570 (Israel)
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211006/8f88249a/attachment.html>

From brian at freeswitch.com  Wed Oct  6 13:33:43 2021
From: brian at freeswitch.com (Brian West)
Date: Wed, 6 Oct 2021 08:33:43 -0500
Subject: [Freeswitch-users] FreeSWITCH Stops Responding
In-Reply-To: <CAEt__=1Twd_K3k8m5YEL-aY=F_A10Rzo=eHCMk3FYG140gTwig@mail.gmail.com>
References: <CAFGRPVp4Fx9Q5wYQ63PffGR4xyQL-TZ1VJMx+g6Jjpb4uEL4Vg@mail.gmail.com>
 <F73E7BB3-0415-4091-B668-4CE84565A5DD@freeswitch.org>
 <CAFGRPVqk3CXXhinmZOchy208UsX2L3kW+9vL4LQxGasdST6pKg@mail.gmail.com>
 <CAEt__=2CYMO_=65xesyBkDFd9v2PO35XusAOyWZHKC1znZ3Chw@mail.gmail.com>
 <CAEt__=1Twd_K3k8m5YEL-aY=F_A10Rzo=eHCMk3FYG140gTwig@mail.gmail.com>
Message-ID: <CAEJMVkCkBqjHTz4G0oCsOd_ekrsqnrb_M9a7gz3DwRE2=tMh4w@mail.gmail.com>

Ah, I see where you might have zigged where thou shall zagged, 'Gentoo
servers', RUN!

Seriously, that's a minefield of WTF.

/b


On Tue, Oct 5, 2021 at 7:11 PM Cary Shouse <cary at ytel.com> wrote:

> Hello,
>
> Just sending an update on this. The Debian Freeswitch server we set up had
> the same issue as our Gentoo servers. Looking at the backtrace we noticed
> the first lowlevellock was occurring when or OurOutbound.luac was being
> used. We decided to use a version that was not precompiled. After that
> change the issue has only occurred once more. So there is still a problem
> we are looking into, but the problems have been massively reduced with that
> change.
>
> Part of the backtrace
> Thread 40 (Thread 0x7f2fe191e700 (LWP 15884)):
> #0  __lll_lock_wait_private () at
> ../sysdeps/unix/sysv/linux/x86_64/lowlevellock.S:63
> #1  0x00007f30512cce53 in __GI__IO_link_in (fp=fp at entry=0x7f2fec53e400)
> at genops.c:93
> #2  0x00007f30512cbb92 in _IO_new_file_init_internal (fp=fp at entry=0x7f2fec53e400)
> at fileops.c:114
> #3  0x00007f30512c0148 in __fopen_internal (filename=0x7f2fec4d5920
> "/usr/local/freeswitch/scripts/360/V3/lua/OurOutbound.luac",
> mode=0x7f2fe37d81b2 "r", is32=1) at iofopen.c:74
> #4  0x00007f2fe37cc123 in luaL_loadfilex () from
> /lib/x86_64-linux-gnu/liblua5.2.so.0
> #5  0x00007f302405796d in lua_parse_and_execute (L=<optimized out>,
> input_code=<optimized out>, session=0x7f2fc91217f8) at mod_lua.cpp:194
> #6  0x00007f3024058191 in lua_function (session=0x7f2fc91217f8,
> data=<optimized out>) at mod_lua.cpp:483
> #7  0x00007f30518920df in switch_core_session_exec (session=session at entry=0x7f2fc91217f8,
> application_interface=application_interface at entry=0x55d5e9528150,
> arg=arg at entry=0x7f2fc96819c0 "360/V3/lua/OurOutbound.luac") at
> src/switch_core_session.c:2900
> #8  0x00007f305189276f in
> switch_core_session_execute_application_get_flags (session=session at entry=0x7f2fc91217f8,
> app=0x7f2fc96819b8 "lua", arg=0x7f2fc96819c0 "360/V3/lua/OurOutbound.luac",
> flags=flags at entry=0x0) at src/switch_core_session.c:2765
> #9  0x00007f3051896ab2 in switch_core_standard_on_execute
> (session=0x7f2fc91217f8) at src/switch_core_state_machine.c:354
> #10 switch_core_session_run (session=0x7f2fc91217f8) at
> src/switch_core_state_machine.c:651
> #11 0x00007f305188face in switch_core_session_thread (thread=<optimized
> out>, obj=0x7f2fc91217f8) at src/switch_core_session.c:1719
> #12 0x00007f305188b4c3 in switch_core_session_thread_pool_worker
> (thread=0x7f3034236c70, obj=<optimized out>) at
> src/switch_core_session.c:1782
> #13 0x00007f3051bc536c in dummy_worker (opaque=0x7f3034236c70) at
> threadproc/unix/thread.c:151
> #14 0x00007f3051793fa3 in start_thread (arg=<optimized out>) at
> pthread_create.c:486
> #15 0x00007f30513494cf in clone () at
> ../sysdeps/unix/sysv/linux/x86_64/clone.S:95
>
> On Thu, Sep 9, 2021 at 1:37 AM Cary Shouse <cary at ytel.com> wrote:
>
>> We have set up a Debian 10 FreeSWITCH-1.10.6 server for
>> testing/debugging.
>>
>> On Sat, Sep 4, 2021 at 5:51 PM David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> That is correct. But we who have worked with FS since around 2002 know
>>> this, newer people get all worked up until they’re told this :)
>>>
>>> It’s a learning thing..
>>>
>>> On Sat, 4 Sep 2021 at 17:31, Ken Rice <krice at freeswitch.org> wrote:
>>>
>>>> “stuck channels” usually arent stuck once you realize things like “show
>>>> calls” and “show channels” are selects against the coredb. this usually
>>>> happens on busy systems timing put waiting for db delete’s to run that fail
>>>> or get skipped.
>>>>
>>>> Sent from my iPhone
>>>>
>>>> On Sep 4, 2021, at 05:22, David Villasmil <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>> 
>>>>
>>>> Hello all,
>>>>
>>>> Interesting point about Gentoo, but I highly doubt it’s OS related.
>>>> I’ve been living this issue (about those “stuck channels”, not a frozen FS,
>>>> that’s indeed weird) for year and years. I’ve seen in on CentOS, Debian,
>>>> Ubuntu, etc.
>>>>
>>>> In my experience this became an issue with the rise of VM. Back in the
>>>> day you would see this every now and then, like months apart (this, or a
>>>> crashed MySQL database), but wasn’t big enough to be a critical problem.
>>>> Today day it’s basically an everyday thing if you have enough nodes.
>>>>
>>>>
>>>> On Sat, 4 Sep 2021 at 00:15, Ken Rice <krice at freeswitch.org> wrote:
>>>>
>>>>> Having massive experience here, you might try replacing your linux
>>>>> distro.
>>>>>
>>>>>  (Yes i know some people are going to take this as brushing off the
>>>>> problem or trying to start a distro war. If this starts a distro war I will
>>>>> moderate you and only allow replies are not headed to a flame war.)
>>>>>
>>>>> However, as someone who has ran an extremely large Gentoo based
>>>>> cluster, we had to abandon gentoo due to random crashes, unexplainable bugs
>>>>> that could not be reproduced or explained.
>>>>>
>>>>> If you can not replace the distro I VERY HiGHLY recommend running
>>>>> vanilla libc and other libs without the gentoo portage included patches.
>>>>> (hint the use flag: vanilla)
>>>>>
>>>>> Ken
>>>>> Sent from my iPhone
>>>>>
>>>>> On Sep 3, 2021, at 14:49, Cary Shouse <cary at ytel.com> wrote:
>>>>>
>>>>> 
>>>>>
>>>>>  We have not been able to reproduce this in testing. It only happens in our production environment.
>>>>>  Are there any other debug options that could be enabled, or is there any code in FreeSWITCH that you think may be causing this that we could look into.  We could get you access to a server if that would help.
>>>>> Server Details:
>>>>> Gentoo-5.10.27
>>>>> RAM: 15GB
>>>>> CPU: 4
>>>>> FreeSWITCH-1-.10.6
>>>>> Sqlite-3.34.1
>>>>>
>>>>>
>>>>> On Thu, Sep 2, 2021 at 7:24 PM Brian West <brian at freeswitch.com>
>>>>> wrote:
>>>>>
>>>>>> And these exhibit the issue on command?
>>>>>>
>>>>>> On Wed, Sep 1, 2021 at 5:06 PM Cary Shouse <cary at ytel.com> wrote:
>>>>>>
>>>>>>> Here are two originate samples.
>>>>>>>
>>>>>>> https://pastebin.com/YKRKrJLu
>>>>>>>
>>>>>>> https://pastebin.com/TEX3yfX6
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Sep 1, 2021 at 8:00 PM Brian West <brian at freeswitch.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Can you try to create one where this happens without all this mess?
>>>>>>>>
>>>>>>>> On Wed, Sep 1, 2021 at 2:41 PM Cary Shouse <cary at ytel.com> wrote:
>>>>>>>>
>>>>>>>>> Here is the LUA code regarding creating sessions and bridging. It
>>>>>>>>> is broken up into 5 sections.
>>>>>>>>> The top part is the common functions, and the 4 sections below are
>>>>>>>>> different call scenarios.
>>>>>>>>> https://pastebin.com/qqGw3tms
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Wed, Sep 1, 2021 at 3:53 PM Brian West <brian at freeswitch.com>
>>>>>>>>> wrote:
>>>>>>>>>
>>>>>>>>>> I you do sessions from your lua, I need to see how you do that.
>>>>>>>>>>
>>>>>>>>>> On Tue, Aug 31, 2021 at 4:20 PM Cary Shouse <cary at ytel.com>
>>>>>>>>>> wrote:
>>>>>>>>>>
>>>>>>>>>>> Yes, it bridges the inbound call to an outbound one.
>>>>>>>>>>>
>>>>>>>>>>> (ProxyOutbound.luac)https://pastebin.com/RDUMy473
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On Tue, Aug 31, 2021 at 8:39 PM Brian West <brian at freeswitch.com>
>>>>>>>>>>> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Guessing you create sessions in lua? ProxyOutbound.luac?
>>>>>>>>>>>>
>>>>>>>>>>>> On Tue, Aug 31, 2021 at 3:33 PM Brian West <
>>>>>>>>>>>> brian at freeswitch.com> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> and to make sure when you uuid kill one it says it didn't
>>>>>>>>>>>>> exist but show channels had it still listed correct?
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Tue, Aug 31, 2021 at 2:58 PM Cary Shouse <cary at ytel.com>
>>>>>>>>>>>>> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> Hello,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> We have got the logs for a few calls that were stuck.
>>>>>>>>>>>>>> (Show channels fe8f206c-2bd9-4af2-b329-35f53228a38a)
>>>>>>>>>>>>>> https://pastebin.com/NTnte7J7
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> (Extracted from logs fe8f206c-2bd9-4af2-b329-35f53228a38a)
>>>>>>>>>>>>>> https://pastebin.com/sAhFiGjY
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> (Show channels 46c6e04b-a58e-4b82-9968-fb0e9c2c0fb3)
>>>>>>>>>>>>>> https://pastebin.com/bmcdjzAr
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> (Extracted from logs 46c6e04b-a58e-4b82-9968-fb0e9c2c0fb3)
>>>>>>>>>>>>>> https://pastebin.com/aGyY3m9F
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> (Show channels 624cac90-0a7b-11ec-b7d0-6f543a588268)
>>>>>>>>>>>>>> https://pastebin.com/0cnQaRhC
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> (Extracted from logs 624cac90-0a7b-11ec-b7d0-6f543a588268)
>>>>>>>>>>>>>> https://pastebin.com/atX0VXdZ
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Please let us know if there is anything else we can provide
>>>>>>>>>>>>>> that would be useful. Thank you.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Mon, Aug 30, 2021 at 8:20 PM Brian West <
>>>>>>>>>>>>>> brian at freeswitch.com> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Check the UUID of the session in question, and extract its
>>>>>>>>>>>>>>> logs, and let's diagnose it
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Mon, Aug 30, 2021 at 12:39 PM Cary Shouse <cary at ytel.com>
>>>>>>>>>>>>>>> wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>  Yes, that is what we are seeing when running uuid_kill.  We
>>>>>>>>>>>>>>>> cannot "replicate" the issue, but it happens multiple times a day for us
>>>>>>>>>>>>>>>> across our servers.   We have looked through logs across
>>>>>>>>>>>>>>>> multiple systems, taken SIP traces to see if that showed anything.
>>>>>>>>>>>>>>>> Last we got a Core dump of the running process after the
>>>>>>>>>>>>>>>> failure. Nothing seems to stand out as the cause.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>  Are there any other debug steps you can think of or what
>>>>>>>>>>>>>>>> other information to look at?  We could capture all traffic on the server,
>>>>>>>>>>>>>>>> or recompile with special debug flags whatever it is that might help.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Steps we have taken:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> sofia loglevel all 9
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> sofia global siptrace on
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> fsctl loglevel 7
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> # Set environment variable
>>>>>>>>>>>>>>>> TPORT_LOG=1
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> On Sat, Aug 28, 2021 at 12:54 AM Brian West <
>>>>>>>>>>>>>>>> brian at freeswitch.com> wrote:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> It's an issue we've been searching for, if you uuid_kill
>>>>>>>>>>>>>>>>> it, and it says no session, it only means the record was NOT removed from
>>>>>>>>>>>>>>>>> the sqlite database.
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Nobody can seem to replicate the steps required to get
>>>>>>>>>>>>>>>>> into that condition, maybe you could look thru the logs and see if you can
>>>>>>>>>>>>>>>>> replicate what ever happened on that lingering UUID
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> /b
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> On Fri, Aug 27, 2021 at 5:40 PM Cary Shouse <cary at ytel.com>
>>>>>>>>>>>>>>>>> wrote:
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>  We recently started having a problem with our FreeSWITCH
>>>>>>>>>>>>>>>>>> servers where calls are working well, with about 200 calls operating
>>>>>>>>>>>>>>>>>> sometimes for a couple of days, and then something happens and all call
>>>>>>>>>>>>>>>>>> progress stops. Then any new inbound calls that come into the server get
>>>>>>>>>>>>>>>>>> stuck in initial status, new inbound get stuck at RINGING. This happens
>>>>>>>>>>>>>>>>>> until the max sessions are hit and of course nothing else works from there.
>>>>>>>>>>>>>>>>>> Our thought is something is creating a lock and stopping anything else from
>>>>>>>>>>>>>>>>>> progressing.
>>>>>>>>>>>>>>>>>>  We are seeing the stuck sessions using "show channels".
>>>>>>>>>>>>>>>>>> Doing a uuid_kill on the calls seems to make no difference to the output of
>>>>>>>>>>>>>>>>>> "show channels", but we have noticed in the logs that some events do happen
>>>>>>>>>>>>>>>>>> to some of the calls.
>>>>>>>>>>>>>>>>>>  We had ran into a similar issue when writing a custom C
>>>>>>>>>>>>>>>>>> module and made some mistakes with session_rwlock etc. But in this case,
>>>>>>>>>>>>>>>>>> all of this is using LUA.
>>>>>>>>>>>>>>>>>> Some general information:
>>>>>>>>>>>>>>>>>> FreeSWITCH Versions 1.10.6 and 1.10.1
>>>>>>>>>>>>>>>>>> Modules used:
>>>>>>>>>>>>>>>>>> LUA, xml_curl(dialplan), event_socket,
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>  Most of this is just application playing audio files and
>>>>>>>>>>>>>>>>>> or creating a new call to bridge to.
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>  Really looking for some help on what debugging we can
>>>>>>>>>>>>>>>>>> enable to help identify what is getting stuck to identify where a bug might
>>>>>>>>>>>>>>>>>> be on our usage or possible issue with FreeSWITCH.
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>  We have captured a core dump of the running process when
>>>>>>>>>>>>>>>>>> it got into the bad state as well, if that is helpful, we can send it in.
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> Thank you for your help.
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> 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>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> 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>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> --
>>>>>>>>>>>>>
>>>>>>>>>>>>> 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>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> --
>>>>>>>>>>>>
>>>>>>>>>>>> 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>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>>
>>>>>>>>>> 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>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>>
>>>>>>>> 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>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>> 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
>>>>
>>>> --
>>>> 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
>>>
>>> --
>>> 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/20211006/12cecf74/attachment-0001.html>

From avi at avimarcus.net  Wed Oct  6 15:37:45 2021
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 6 Oct 2021 15:37:45 +0000
Subject: [Freeswitch-users] Bridge to other FS server has no audio ???
In-Reply-To: <0100017c55972813-4ae9de8a-588c-46ec-ac70-7e2ce2230bf2-000000@email.amazonses.com>
References: <0100017c55972813-4ae9de8a-588c-46ec-ac70-7e2ce2230bf2-000000@email.amazonses.com>
Message-ID: <0100017c5640e271-50912241-c8e6-4c92-bb1d-8fb6b4b337e0-000000@email.amazonses.com>

Apparently it was a network issue - SIP signaling worked but not the RTP!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211006/291fedc7/attachment.html>

From david.villasmil.work at gmail.com  Wed Oct  6 15:46:46 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 6 Oct 2021 16:46:46 +0100
Subject: [Freeswitch-users] Bridge to other FS server has no audio ???
In-Reply-To: <0100017c55972813-4ae9de8a-588c-46ec-ac70-7e2ce2230bf2-000000@email.amazonses.com>
References: <0100017c55972813-4ae9de8a-588c-46ec-ac70-7e2ce2230bf2-000000@email.amazonses.com>
Message-ID: <CAFGRPVoGcNpTm4steoXcZLeK6z_UcF04_9zw5nXg20KbF_=g4Q@mail.gmail.com>

What’s the difference between second and third scenario? Is the third in
bypass?

On Wed, 6 Oct 2021 at 13:40, Avi Marcus <avi at avimarcus.net> wrote:

> Any ideas on why a call doesn't have media? It used to work, but I think
> my upstream changed his SDP again.
>
> - FreeSWITCH Server A - call comes in and bypass_media bridges to FS
> server B. Media works.
> - FreeSWITCH Server A - call comes in and bridges to FS server B (not on
> bypass). Media works.
> - FreeSWITCH Server A - call comes in, gets answered, then bridges to FS
> server B. Call looks OK, but no media is flowing (I don't hear anything,
> PCAPs just have SIP, and there isn't 80kbps network traffic). All the same
> codecs are set in the json cdrs (PCMU).
>
> FS server B is to join a conference if that matters.
>
> I was assuming it had to do with codecs, but setting absolute_codec_string
> to PCMU doesn't make any difference in the logs  - it's already always PCMU.
>
> I have NO clue what further could cause this other than codecs, which seem
> to be fine. Any ideas please?
>
>
> -Avi Marcus
> 1-718-989-9485 (USA)
> 02-372-1570 (Israel)
> _________________________________________________________________________
>
> 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/20211006/bc04e3c1/attachment.html>

From cary at ytel.com  Wed Oct  6 14:52:41 2021
From: cary at ytel.com (Cary Shouse)
Date: Wed, 6 Oct 2021 14:52:41 +0000
Subject: [Freeswitch-users] FreeSWITCH Stops Responding
In-Reply-To: <CAEJMVkCkBqjHTz4G0oCsOd_ekrsqnrb_M9a7gz3DwRE2=tMh4w@mail.gmail.com>
References: <CAFGRPVp4Fx9Q5wYQ63PffGR4xyQL-TZ1VJMx+g6Jjpb4uEL4Vg@mail.gmail.com>
 <F73E7BB3-0415-4091-B668-4CE84565A5DD@freeswitch.org>
 <CAFGRPVqk3CXXhinmZOchy208UsX2L3kW+9vL4LQxGasdST6pKg@mail.gmail.com>
 <CAEt__=2CYMO_=65xesyBkDFd9v2PO35XusAOyWZHKC1znZ3Chw@mail.gmail.com>
 <CAEt__=1Twd_K3k8m5YEL-aY=F_A10Rzo=eHCMk3FYG140gTwig@mail.gmail.com>
 <CAEJMVkCkBqjHTz4G0oCsOd_ekrsqnrb_M9a7gz3DwRE2=tMh4w@mail.gmail.com>
Message-ID: <CAEt__=3awVvyo4zC=7qvX0RS=P10OfOE+56iHAMGAYPQ_jKwtQ@mail.gmail.com>

lol


On Wed, Oct 6, 2021 at 1:58 PM Brian West <brian at freeswitch.com> wrote:

> Ah, I see where you might have zigged where thou shall zagged, 'Gentoo
> servers', RUN!
>
> Seriously, that's a minefield of WTF.
>
> /b
>
>
> On Tue, Oct 5, 2021 at 7:11 PM Cary Shouse <cary at ytel.com> wrote:
>
>> Hello,
>>
>> Just sending an update on this. The Debian Freeswitch server we set up
>> had the same issue as our Gentoo servers. Looking at the backtrace we
>> noticed the first lowlevellock was occurring when or OurOutbound.luac was
>> being used. We decided to use a version that was not precompiled. After
>> that change the issue has only occurred once more. So there is still a
>> problem we are looking into, but the problems have been massively reduced
>> with that change.
>>
>> Part of the backtrace
>> Thread 40 (Thread 0x7f2fe191e700 (LWP 15884)):
>> #0  __lll_lock_wait_private () at
>> ../sysdeps/unix/sysv/linux/x86_64/lowlevellock.S:63
>> #1  0x00007f30512cce53 in __GI__IO_link_in (fp=fp at entry=0x7f2fec53e400)
>> at genops.c:93
>> #2  0x00007f30512cbb92 in _IO_new_file_init_internal (fp=fp at entry=0x7f2fec53e400)
>> at fileops.c:114
>> #3  0x00007f30512c0148 in __fopen_internal (filename=0x7f2fec4d5920
>> "/usr/local/freeswitch/scripts/360/V3/lua/OurOutbound.luac",
>> mode=0x7f2fe37d81b2 "r", is32=1) at iofopen.c:74
>> #4  0x00007f2fe37cc123 in luaL_loadfilex () from
>> /lib/x86_64-linux-gnu/liblua5.2.so.0
>> #5  0x00007f302405796d in lua_parse_and_execute (L=<optimized out>,
>> input_code=<optimized out>, session=0x7f2fc91217f8) at mod_lua.cpp:194
>> #6  0x00007f3024058191 in lua_function (session=0x7f2fc91217f8,
>> data=<optimized out>) at mod_lua.cpp:483
>> #7  0x00007f30518920df in switch_core_session_exec (session=session at entry=0x7f2fc91217f8,
>> application_interface=application_interface at entry=0x55d5e9528150,
>> arg=arg at entry=0x7f2fc96819c0 "360/V3/lua/OurOutbound.luac") at
>> src/switch_core_session.c:2900
>> #8  0x00007f305189276f in
>> switch_core_session_execute_application_get_flags (session=session at entry=0x7f2fc91217f8,
>> app=0x7f2fc96819b8 "lua", arg=0x7f2fc96819c0 "360/V3/lua/OurOutbound.luac",
>> flags=flags at entry=0x0) at src/switch_core_session.c:2765
>> #9  0x00007f3051896ab2 in switch_core_standard_on_execute
>> (session=0x7f2fc91217f8) at src/switch_core_state_machine.c:354
>> #10 switch_core_session_run (session=0x7f2fc91217f8) at
>> src/switch_core_state_machine.c:651
>> #11 0x00007f305188face in switch_core_session_thread (thread=<optimized
>> out>, obj=0x7f2fc91217f8) at src/switch_core_session.c:1719
>> #12 0x00007f305188b4c3 in switch_core_session_thread_pool_worker
>> (thread=0x7f3034236c70, obj=<optimized out>) at
>> src/switch_core_session.c:1782
>> #13 0x00007f3051bc536c in dummy_worker (opaque=0x7f3034236c70) at
>> threadproc/unix/thread.c:151
>> #14 0x00007f3051793fa3 in start_thread (arg=<optimized out>) at
>> pthread_create.c:486
>> #15 0x00007f30513494cf in clone () at
>> ../sysdeps/unix/sysv/linux/x86_64/clone.S:95
>>
>> On Thu, Sep 9, 2021 at 1:37 AM Cary Shouse <cary at ytel.com> wrote:
>>
>>> We have set up a Debian 10 FreeSWITCH-1.10.6 server for
>>> testing/debugging.
>>>
>>> On Sat, Sep 4, 2021 at 5:51 PM David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>>> That is correct. But we who have worked with FS since around 2002 know
>>>> this, newer people get all worked up until they’re told this :)
>>>>
>>>> It’s a learning thing..
>>>>
>>>> On Sat, 4 Sep 2021 at 17:31, Ken Rice <krice at freeswitch.org> wrote:
>>>>
>>>>> “stuck channels” usually arent stuck once you realize things like
>>>>> “show calls” and “show channels” are selects against the coredb. this
>>>>> usually happens on busy systems timing put waiting for db delete’s to run
>>>>> that fail or get skipped.
>>>>>
>>>>> Sent from my iPhone
>>>>>
>>>>> On Sep 4, 2021, at 05:22, David Villasmil <
>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>
>>>>> 
>>>>>
>>>>> Hello all,
>>>>>
>>>>> Interesting point about Gentoo, but I highly doubt it’s OS related.
>>>>> I’ve been living this issue (about those “stuck channels”, not a frozen FS,
>>>>> that’s indeed weird) for year and years. I’ve seen in on CentOS, Debian,
>>>>> Ubuntu, etc.
>>>>>
>>>>> In my experience this became an issue with the rise of VM. Back in the
>>>>> day you would see this every now and then, like months apart (this, or a
>>>>> crashed MySQL database), but wasn’t big enough to be a critical problem.
>>>>> Today day it’s basically an everyday thing if you have enough nodes.
>>>>>
>>>>>
>>>>> On Sat, 4 Sep 2021 at 00:15, Ken Rice <krice at freeswitch.org> wrote:
>>>>>
>>>>>> Having massive experience here, you might try replacing your linux
>>>>>> distro.
>>>>>>
>>>>>>  (Yes i know some people are going to take this as brushing off the
>>>>>> problem or trying to start a distro war. If this starts a distro war I will
>>>>>> moderate you and only allow replies are not headed to a flame war.)
>>>>>>
>>>>>> However, as someone who has ran an extremely large Gentoo based
>>>>>> cluster, we had to abandon gentoo due to random crashes, unexplainable bugs
>>>>>> that could not be reproduced or explained.
>>>>>>
>>>>>> If you can not replace the distro I VERY HiGHLY recommend running
>>>>>> vanilla libc and other libs without the gentoo portage included patches.
>>>>>> (hint the use flag: vanilla)
>>>>>>
>>>>>> Ken
>>>>>> Sent from my iPhone
>>>>>>
>>>>>> On Sep 3, 2021, at 14:49, Cary Shouse <cary at ytel.com> wrote:
>>>>>>
>>>>>> 
>>>>>>
>>>>>>  We have not been able to reproduce this in testing. It only happens in our production environment.
>>>>>>  Are there any other debug options that could be enabled, or is there any code in FreeSWITCH that you think may be causing this that we could look into.  We could get you access to a server if that would help.
>>>>>> Server Details:
>>>>>> Gentoo-5.10.27
>>>>>> RAM: 15GB
>>>>>> CPU: 4
>>>>>> FreeSWITCH-1-.10.6
>>>>>> Sqlite-3.34.1
>>>>>>
>>>>>>
>>>>>> On Thu, Sep 2, 2021 at 7:24 PM Brian West <brian at freeswitch.com>
>>>>>> wrote:
>>>>>>
>>>>>>> And these exhibit the issue on command?
>>>>>>>
>>>>>>> On Wed, Sep 1, 2021 at 5:06 PM Cary Shouse <cary at ytel.com> wrote:
>>>>>>>
>>>>>>>> Here are two originate samples.
>>>>>>>>
>>>>>>>> https://pastebin.com/YKRKrJLu
>>>>>>>>
>>>>>>>> https://pastebin.com/TEX3yfX6
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Wed, Sep 1, 2021 at 8:00 PM Brian West <brian at freeswitch.com>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>>> Can you try to create one where this happens without all this mess?
>>>>>>>>>
>>>>>>>>> On Wed, Sep 1, 2021 at 2:41 PM Cary Shouse <cary at ytel.com> wrote:
>>>>>>>>>
>>>>>>>>>> Here is the LUA code regarding creating sessions and bridging. It
>>>>>>>>>> is broken up into 5 sections.
>>>>>>>>>> The top part is the common functions, and the 4 sections below
>>>>>>>>>> are different call scenarios.
>>>>>>>>>> https://pastebin.com/qqGw3tms
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Wed, Sep 1, 2021 at 3:53 PM Brian West <brian at freeswitch.com>
>>>>>>>>>> wrote:
>>>>>>>>>>
>>>>>>>>>>> I you do sessions from your lua, I need to see how you do that.
>>>>>>>>>>>
>>>>>>>>>>> On Tue, Aug 31, 2021 at 4:20 PM Cary Shouse <cary at ytel.com>
>>>>>>>>>>> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Yes, it bridges the inbound call to an outbound one.
>>>>>>>>>>>>
>>>>>>>>>>>> (ProxyOutbound.luac)https://pastebin.com/RDUMy473
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On Tue, Aug 31, 2021 at 8:39 PM Brian West <
>>>>>>>>>>>> brian at freeswitch.com> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> Guessing you create sessions in lua? ProxyOutbound.luac?
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Tue, Aug 31, 2021 at 3:33 PM Brian West <
>>>>>>>>>>>>> brian at freeswitch.com> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> and to make sure when you uuid kill one it says it didn't
>>>>>>>>>>>>>> exist but show channels had it still listed correct?
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Tue, Aug 31, 2021 at 2:58 PM Cary Shouse <cary at ytel.com>
>>>>>>>>>>>>>> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Hello,
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> We have got the logs for a few calls that were stuck.
>>>>>>>>>>>>>>> (Show channels fe8f206c-2bd9-4af2-b329-35f53228a38a)
>>>>>>>>>>>>>>> https://pastebin.com/NTnte7J7
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> (Extracted from logs fe8f206c-2bd9-4af2-b329-35f53228a38a)
>>>>>>>>>>>>>>> https://pastebin.com/sAhFiGjY
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> (Show channels 46c6e04b-a58e-4b82-9968-fb0e9c2c0fb3)
>>>>>>>>>>>>>>> https://pastebin.com/bmcdjzAr
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> (Extracted from logs 46c6e04b-a58e-4b82-9968-fb0e9c2c0fb3)
>>>>>>>>>>>>>>> https://pastebin.com/aGyY3m9F
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> (Show channels 624cac90-0a7b-11ec-b7d0-6f543a588268)
>>>>>>>>>>>>>>> https://pastebin.com/0cnQaRhC
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> (Extracted from logs 624cac90-0a7b-11ec-b7d0-6f543a588268)
>>>>>>>>>>>>>>> https://pastebin.com/atX0VXdZ
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Please let us know if there is anything else we can provide
>>>>>>>>>>>>>>> that would be useful. Thank you.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Mon, Aug 30, 2021 at 8:20 PM Brian West <
>>>>>>>>>>>>>>> brian at freeswitch.com> wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Check the UUID of the session in question, and extract its
>>>>>>>>>>>>>>>> logs, and let's diagnose it
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> On Mon, Aug 30, 2021 at 12:39 PM Cary Shouse <cary at ytel.com>
>>>>>>>>>>>>>>>> wrote:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>  Yes, that is what we are seeing when running uuid_kill.  We
>>>>>>>>>>>>>>>>> cannot "replicate" the issue, but it happens multiple times a day for us
>>>>>>>>>>>>>>>>> across our servers.   We have looked through logs across
>>>>>>>>>>>>>>>>> multiple systems, taken SIP traces to see if that showed anything.
>>>>>>>>>>>>>>>>> Last we got a Core dump of the running process after the
>>>>>>>>>>>>>>>>> failure. Nothing seems to stand out as the cause.
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>  Are there any other debug steps you can think of or what
>>>>>>>>>>>>>>>>> other information to look at?  We could capture all traffic on the server,
>>>>>>>>>>>>>>>>> or recompile with special debug flags whatever it is that might help.
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Steps we have taken:
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> sofia loglevel all 9
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> sofia global siptrace on
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> fsctl loglevel 7
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> # Set environment variable
>>>>>>>>>>>>>>>>> TPORT_LOG=1
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> On Sat, Aug 28, 2021 at 12:54 AM Brian West <
>>>>>>>>>>>>>>>>> brian at freeswitch.com> wrote:
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> It's an issue we've been searching for, if you uuid_kill
>>>>>>>>>>>>>>>>>> it, and it says no session, it only means the record was NOT removed from
>>>>>>>>>>>>>>>>>> the sqlite database.
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> Nobody can seem to replicate the steps required to get
>>>>>>>>>>>>>>>>>> into that condition, maybe you could look thru the logs and see if you can
>>>>>>>>>>>>>>>>>> replicate what ever happened on that lingering UUID
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> /b
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>> On Fri, Aug 27, 2021 at 5:40 PM Cary Shouse <
>>>>>>>>>>>>>>>>>> cary at ytel.com> wrote:
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>  We recently started having a problem with our
>>>>>>>>>>>>>>>>>>> FreeSWITCH servers where calls are working well, with about 200 calls
>>>>>>>>>>>>>>>>>>> operating sometimes for a couple of days, and then something happens and
>>>>>>>>>>>>>>>>>>> all call progress stops. Then any new inbound calls that come into the
>>>>>>>>>>>>>>>>>>> server get stuck in initial status, new inbound get stuck at RINGING. This
>>>>>>>>>>>>>>>>>>> happens until the max sessions are hit and of course nothing else works
>>>>>>>>>>>>>>>>>>> from there. Our thought is something is creating a lock and stopping
>>>>>>>>>>>>>>>>>>> anything else from progressing.
>>>>>>>>>>>>>>>>>>>  We are seeing the stuck sessions using "show channels".
>>>>>>>>>>>>>>>>>>> Doing a uuid_kill on the calls seems to make no difference to the output of
>>>>>>>>>>>>>>>>>>> "show channels", but we have noticed in the logs that some events do happen
>>>>>>>>>>>>>>>>>>> to some of the calls.
>>>>>>>>>>>>>>>>>>>  We had ran into a similar issue when writing a custom C
>>>>>>>>>>>>>>>>>>> module and made some mistakes with session_rwlock etc. But in this case,
>>>>>>>>>>>>>>>>>>> all of this is using LUA.
>>>>>>>>>>>>>>>>>>> Some general information:
>>>>>>>>>>>>>>>>>>> FreeSWITCH Versions 1.10.6 and 1.10.1
>>>>>>>>>>>>>>>>>>> Modules used:
>>>>>>>>>>>>>>>>>>> LUA, xml_curl(dialplan), event_socket,
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>  Most of this is just application playing audio files
>>>>>>>>>>>>>>>>>>> and or creating a new call to bridge to.
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>  Really looking for some help on what debugging we can
>>>>>>>>>>>>>>>>>>> enable to help identify what is getting stuck to identify where a bug might
>>>>>>>>>>>>>>>>>>> be on our usage or possible issue with FreeSWITCH.
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>>  We have captured a core dump of the running process
>>>>>>>>>>>>>>>>>>> when it got into the bad state as well, if that is helpful, we can send it
>>>>>>>>>>>>>>>>>>> in.
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> Thank you for your help.
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>>> 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>
>>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> 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>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> 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>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> --
>>>>>>>>>>>>>
>>>>>>>>>>>>> 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>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> --
>>>>>>>>>>>
>>>>>>>>>>> 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>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>>
>>>>>>>>> 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>
>>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>>
>>>>>>> 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
>>>>>
>>>>> --
>>>>> 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
>>>>
>>>> --
>>>> 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>
> _________________________________________________________________________
>
> 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/20211006/6a545a49/attachment-0001.html>

From john.hardiman at missionlabs.co.uk  Wed Oct  6 08:36:01 2021
From: john.hardiman at missionlabs.co.uk (John Hardiman)
Date: Wed, 6 Oct 2021 08:36:01 +0000
Subject: [Freeswitch-users] Jitter and dropping MOS score
In-Reply-To: <66af870f-ebcd-ff5c-273b-1f79a1d26e5d@emtex.de>
References: <66af870f-ebcd-ff5c-273b-1f79a1d26e5d@emtex.de>
Message-ID: <CWXP265MB21523259640F26F3F8306F19A1B09@CWXP265MB2152.GBRP265.PROD.OUTLOOK.COM>

Hi Christoph,

For the CPU part: On linux boxes if a process in top goes over 100% or top shows >1.0 in load, this means that more than one core is occupied. Since you have 16 cores this is fine as you are using 1/16 of your CPU for the FreeSWITCH process. So 200% would be 2 cores fully used etc, etc.

For the Jitter part/dropping MoS, we might need a bit more info, like times or patterns that is happens or even what the MoS drops by and how much jitter is seen. I personally use SNGrep and a combination of Homer/VoIPMonitor to get all of the stuff I need. 

Accurate monitoring will point you in the right direction if there is an issue (Homer 7 is free and really good as a side note)

Thanks,

John.

-----Original Message-----
From: Christoph Russow <russow at emtex.de> 
Sent: 05 October 2021 10:34
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Jitter and dropping MOS score

Hi all,

we are currently experiencing jitter and dropping MOS score on an increasing amount of calls on our system.

i can't see any increased load on our system (neither CPU nor I/O wise).

Current stats:

~ 15Mbit/s network traffic (on a 1Gbit/s network) load average: 1,10, 0,82, 0,91 (16 core machine) memory used: 2,26GB/16GB

freeswitch at lc1> status
UP 0 years, 12 days, 13 hours, 30 minutes, 24 seconds, 649 milliseconds,
831 microseconds
FreeSWITCH (Version 1.10.1 -release-12-f9990221e6 64bit) is ready
137634 session(s) since startup
218 session(s) - peak 245, last 5min 222
1 session(s) per Sec out of max 30, peak 8, last 5min 2
1000 session(s) max
min idle cpu 0.00/95.37
Current Stack Size/Max 244K/244K

System ist not virtualized.

the only "strange" thing i see is the freeswitch "main"process that consumes (according to htop) >=100% cpu (see attached screenshot).
Can this be the problem? if yes what can i do against it.

if not what are possible bottlenecks i should weed out?

if you need any additional informations to help me sort out this problem please don't hesitate to ask.

Thanks in advance and best regards
Christoph



--
________________________________________________________________

EMTEX GmbH
Christoph Russow
Software Engineer

Bischof-Otto-Weg 9
D-91086 Aurachtal
                           Geschäftsführer:       Markus Enzinger
Tel. +49 9132 7490 0      Sitz der Gesellschaft: 91086 Aurachtal
Fax. +49 9132 7490 900    Amtsgericht Fürth:     HRB6804
________________________________________________________________

From Jerry.Kendall at BishopHosting.com  Wed Oct  6 02:57:47 2021
From: Jerry.Kendall at BishopHosting.com (Jerry Kendall)
Date: Tue, 5 Oct 2021 22:57:47 -0400
Subject: [Freeswitch-users] Jitter and dropping MOS score
In-Reply-To: <66af870f-ebcd-ff5c-273b-1f79a1d26e5d@emtex.de>
References: <66af870f-ebcd-ff5c-273b-1f79a1d26e5d@emtex.de>
Message-ID: <6a483e1a-da5e-f7f1-9078-ea4209764e15@BishopHosting.com>

Christoph,

If you are running any lua scripts, you might want to look at those...

I had a huge load issue I went through in testing and found out my lua scripts were killing the CPU during "high" usage of 3-5 calls...

Search for my postings on "High CPU usage and Slow BLFs with only 8 channels" or... https://lists.freeswitch.org/pipermail/freeswitch-users/2021-September/135066.html


The issue I had was looking at the EVENTS in the lua script.... Every channel created 20-30 events
I changed the strategy for catching the EVENTS and the load has never been over 0.2. The MOS is now steady at 4+ and ZERO jitter....

While I never commented on the jitter in my postings, there were issues and I knew it was due to the high CPU usage. My issue, why the high load with only a few calls.

Hope it helps,
Jerry

On 10/5/2021 5:33 AM, Christoph Russow wrote:
> Hi all,
>
> we are currently experiencing jitter and dropping MOS score on an increasing amount of calls on our system.
>
> i can't see any increased load on our system (neither CPU nor I/O wise).
>
> Current stats:
>
> ~ 15Mbit/s network traffic (on a 1Gbit/s network)
> load average: 1,10, 0,82, 0,91 (16 core machine)
> memory used: 2,26GB/16GB
>
> freeswitch at lc1> status
> UP 0 years, 12 days, 13 hours, 30 minutes, 24 seconds, 649 milliseconds, 831 microseconds
> FreeSWITCH (Version 1.10.1 -release-12-f9990221e6 64bit) is ready
> 137634 session(s) since startup
> 218 session(s) - peak 245, last 5min 222
> 1 session(s) per Sec out of max 30, peak 8, last 5min 2
> 1000 session(s) max
> min idle cpu 0.00/95.37
> Current Stack Size/Max 244K/244K
>
> System ist not virtualized.
>
> the only "strange" thing i see is the freeswitch "main"process that consumes (according to htop) >=100% cpu (see attached screenshot).
> Can this be the problem? if yes what can i do against it.
>
> if not what are possible bottlenecks i should weed out?
>
> if you need any additional informations to help me sort out this problem please don't hesitate to ask.
>
> Thanks in advance and best regards
> Christoph
>
>
>
>
> _________________________________________________________________________
>
> 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 mitch.capper at gmail.com  Wed Oct  6 16:44:00 2021
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 6 Oct 2021 09:44:00 -0700
Subject: [Freeswitch-users] tls-verify-depth
In-Reply-To: <9e2e6979-7103-f483-d499-4d31543dc120@gmail.com>
References: <9e2e6979-7103-f483-d499-4d31543dc120@gmail.com>
Message-ID: <CALn233pUKs1aKh+hvKqFtov+8=4No_RX-SPxcnqX4+H5OyhOYg@mail.gmail.com>

I don't know if it puts anything in the logs but the theory here is to
prevent a potentially longer certificate chain validation.   In theory a
connection could provide a certificate and then several above that that
would get walked to try and validate to the depth specified (and if no
matching signed cert is found at that point it would throw an error).
Generally having this variable set to the length of the max cert chain-1
should work but you can always set it to more than that without a negative
effect.

Is your issue you think it should be failing and it isnt or?

~mitch (they, them)


On Thu, Aug 19, 2021 at 7:44 AM Victor Bogatyryev <
victor.bogatyryev at gmail.com> wrote:

> Hi!
>
> I cannot understand how the directive <param name="tls-verify-depth"
> value="1" /> works.
>
> The verification policy is set like this
>
> <param name="tls-verify-policy" value="all|subjects_all" />.
>
> The certificate chain has one intermediate CA. But I don't see anything
> in the tport log except for checking the subject of the user's certificate.
>
>
> Regards.
>
> --
>
> V.Bogatyryev
> _________________________________________________________________________
>
> 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/20211006/d871552a/attachment.html>

From jurijs.ivolga at gmail.com  Wed Oct  6 16:51:32 2021
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Wed, 6 Oct 2021 19:51:32 +0300
Subject: [Freeswitch-users] Jitter and dropping MOS score
In-Reply-To: <CWXP265MB21523259640F26F3F8306F19A1B09@CWXP265MB2152.GBRP265.PROD.OUTLOOK.COM>
References: <66af870f-ebcd-ff5c-273b-1f79a1d26e5d@emtex.de>
 <CWXP265MB21523259640F26F3F8306F19A1B09@CWXP265MB2152.GBRP265.PROD.OUTLOOK.COM>
Message-ID: <CAOrA2UZv04634sg-kWxNx-xWAfTiTYBKcausS9DWcNHgfn56xA@mail.gmail.com>

Hi,

I can not tell you about everybody, but usually when I get this problem 99%
of the time the issue is on the provider side. I would try to reproduce
this issue by myself and capture pcap files and analyze those.

With kind regards,

Jurijs


On Wed, Oct 6, 2021 at 7:43 PM John Hardiman <
john.hardiman at missionlabs.co.uk> wrote:

> Hi Christoph,
>
> For the CPU part: On linux boxes if a process in top goes over 100% or top
> shows >1.0 in load, this means that more than one core is occupied. Since
> you have 16 cores this is fine as you are using 1/16 of your CPU for the
> FreeSWITCH process. So 200% would be 2 cores fully used etc, etc.
>
> For the Jitter part/dropping MoS, we might need a bit more info, like
> times or patterns that is happens or even what the MoS drops by and how
> much jitter is seen. I personally use SNGrep and a combination of
> Homer/VoIPMonitor to get all of the stuff I need.
>
> Accurate monitoring will point you in the right direction if there is an
> issue (Homer 7 is free and really good as a side note)
>
> Thanks,
>
> John.
>
> -----Original Message-----
> From: Christoph Russow <russow at emtex.de>
> Sent: 05 October 2021 10:34
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Jitter and dropping MOS score
>
> Hi all,
>
> we are currently experiencing jitter and dropping MOS score on an
> increasing amount of calls on our system.
>
> i can't see any increased load on our system (neither CPU nor I/O wise).
>
> Current stats:
>
> ~ 15Mbit/s network traffic (on a 1Gbit/s network) load average: 1,10,
> 0,82, 0,91 (16 core machine) memory used: 2,26GB/16GB
>
> freeswitch at lc1> status
> UP 0 years, 12 days, 13 hours, 30 minutes, 24 seconds, 649 milliseconds,
> 831 microseconds
> FreeSWITCH (Version 1.10.1 -release-12-f9990221e6 64bit) is ready
> 137634 session(s) since startup
> 218 session(s) - peak 245, last 5min 222
> 1 session(s) per Sec out of max 30, peak 8, last 5min 2
> 1000 session(s) max
> min idle cpu 0.00/95.37
> Current Stack Size/Max 244K/244K
>
> System ist not virtualized.
>
> the only "strange" thing i see is the freeswitch "main"process that
> consumes (according to htop) >=100% cpu (see attached screenshot).
> Can this be the problem? if yes what can i do against it.
>
> if not what are possible bottlenecks i should weed out?
>
> if you need any additional informations to help me sort out this problem
> please don't hesitate to ask.
>
> Thanks in advance and best regards
> Christoph
>
>
>
> --
> ________________________________________________________________
>
> EMTEX GmbH
> Christoph Russow
> Software Engineer
>
> Bischof-Otto-Weg 9
> D-91086 Aurachtal
>                            Geschäftsführer:       Markus Enzinger
> Tel. +49 9132 7490 0      Sitz der Gesellschaft: 91086 Aurachtal
> Fax. +49 9132 7490 900    Amtsgericht Fürth:     HRB6804
> ________________________________________________________________
> _________________________________________________________________________
>
> 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/20211006/a8db4102/attachment-0001.html>

From david.villasmil.work at gmail.com  Wed Oct  6 20:50:57 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 6 Oct 2021 21:50:57 +0100
Subject: [Freeswitch-users] Bridge to other FS server has no audio ???
In-Reply-To: <0100017c5640e271-50912241-c8e6-4c92-bb1d-8fb6b4b337e0-000000@email.amazonses.com>
References: <0100017c55972813-4ae9de8a-588c-46ec-ac70-7e2ce2230bf2-000000@email.amazonses.com>
 <0100017c5640e271-50912241-c8e6-4c92-bb1d-8fb6b4b337e0-000000@email.amazonses.com>
Message-ID: <CAFGRPVqZhkcVPYLKsBW-0cQ+3CL+3ZWrRMW_EaCOr20Ei_nYKg@mail.gmail.com>

Lol, I’m glad it got resolved.

On Wed, 6 Oct 2021 at 17:30, Avi Marcus <avi at avimarcus.net> wrote:

> Apparently it was a network issue - SIP signaling worked but not the RTP!
> _________________________________________________________________________
>
> 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/20211006/89c8268c/attachment.html>

From avi at avimarcus.net  Thu Oct  7 06:35:07 2021
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 7 Oct 2021 06:35:07 +0000
Subject: [Freeswitch-users] Bridge to other FS server has no audio until DTMF
Message-ID: <0100017c59766f67-90aa83ae-7742-464c-9fe2-9fa789725f4a-000000@email.amazonses.com>

I started a new thread in case anyone muted it... it wasn't simply a
network issue.

It seems the bridging occurs and dialplan processes, but no media flows -
until DTMF from the A-leg.
Call flow: PSTN (via carrier) to freeswitch A -> media and IVR ->
freeswitch B.

Calls directly from carrier to Freeswitch B are fine.
Calls from a different carrier to Freeswitch A -> media and IVR ->
Freeswitch B are also fine.
So it sounds like a carrier/unique SIP/RTP issue, but since FS is in the
media path, it's an FS issue...


I actually mcguyvered this right now with a queue_dtmf before the bridge,
to force the audio stream to update.

Here's the log on freeswitch B:

EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
 log(DEBUG class chosen: 1234567)
2021-10-07 09:16:24.343175 [DEBUG] mod_dptools.c:1879 class chosen: 1234567
EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
 javascript(conference/lookupAndJoinConference.js 1234567)
EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
 playback(class/hold-wait-teacher.wav)
2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/
972581234567 at 172.123.123.123 entering state [completed][200]
2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/
972581234567 at 172.123.123.123 entering state [ready][200]
2021-10-07 09:16:24.363379 [DEBUG
] switch_ivr_play_say.c:1486 Codec Activated L16 at 8000hz 1 channels 20ms




2021-10-07 09:16:34.903283 [DEBUG
] switch_rtp.c:7793 Correct audio ip/port confirmed.
2021-10-07 09:16:34.923190 [DEBUG] switch_rtp.c:8038 RTP RECV DTMF 3:2080
2021-10-07 09:16:34.923190 [INFO] switch_channel.c:522 RECV DTMF 3:2080
2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3
2021-10-07 09:16:37.143169 [DEBUG
] switch_ivr_play_say.c:1931 done playing file
/usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav


You can see a 10 second gap between call ready 200 and correct audio/ip and
file done playing (it's a 2 second file), and this doesn't happen
automatically, only when I choose to press something.


Any ideas as to the root cause of this?


-Avi Marcus

---------- Forwarded message ---------
From: Avi Marcus <avi at avimarcus.net>
Date: Wed, Oct 6, 2021 at 3:32 PM
Subject: Bridge to other FS server has no audio ???
To: FreeSWITCH Users Help <FreeSWITCH-users at lists.freeswitch.org>


Any ideas on why a call doesn't have media? It used to work, but I think my
upstream changed his SDP again.

- FreeSWITCH Server A - call comes in and bypass_media bridges to FS server
B. Media works.
- FreeSWITCH Server A - call comes in and bridges to FS server B (not on
bypass). Media works.
- FreeSWITCH Server A - call comes in, gets answered, then bridges to FS
server B. Call looks OK, but no media is flowing (I don't hear anything,
PCAPs just have SIP, and there isn't 80kbps network traffic). All the same
codecs are set in the json cdrs (PCMU).

FS server B is to join a conference if that matters.

I was assuming it had to do with codecs, but setting absolute_codec_string
to PCMU doesn't make any difference in the logs  - it's already always PCMU.

I have NO clue what further could cause this other than codecs, which seem
to be fine. Any ideas please?


-Avi Marcus
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211007/1571ded0/attachment.html>

From russow at emtex.de  Thu Oct  7 09:23:45 2021
From: russow at emtex.de (Christoph Russow)
Date: Thu, 7 Oct 2021 11:23:45 +0200
Subject: [Freeswitch-users] Jitter and dropping MOS score
In-Reply-To: <CAGvJV4oueJAr-m8GHn1nid1ATwvpd+fnvOsdj75pk_5mzUpc4g@mail.gmail.com>
References: <66af870f-ebcd-ff5c-273b-1f79a1d26e5d@emtex.de>
 <CAGvJV4oueJAr-m8GHn1nid1ATwvpd+fnvOsdj75pk_5mzUpc4g@mail.gmail.com>
Message-ID: <19d3dd01-f84d-1672-aa6a-42dabac5a188@emtex.de>

Hi all,

thanks for all the input.

We could sort out the Jitter/MOS problem as a temporary Network problem 
with one of our carriers.

about the CPU:
how should i make sure that it is really eating 100% CPU?

"ps aux" shows the freeswitch process at arround 30% so it seems to not 
really consume 100%cpu as htop still tells me.

USER       PID %CPU %MEM    VSZ   RSS TTY      STAT START   TIME COMMAND
freeswi+ 30074 30.6  3.9 8582604 636456 ?      SNsl Sep22 6431:58 
/usr/bin/freeswitch -ncwait -u freeswitch -g emtex-files -nonat

best regards
Christoph

Am 06.10.21 um 12:33 schrieb Dragos Oancea:
> if it really consumes 100% CPU with few calls then it's abnormal.
> kill it with SIGABRT (make sure it will dump core before that), get a 
> backtrace with gdb and open a github issue.
> before doing this you need to make sure it's really eating 100 % CPU, 
> this screenshot is not very relevant.
> Also, you need to make sure you don't have real packet loss. Perhaps 
> it's a matter of dimensioning, eg: how many simultaneous calls do you 
> expect to have on that hardware, what codec, etc.
> 
> On Wed, Oct 6, 2021 at 2:26 AM Christoph Russow <russow at emtex.de 
> <mailto:russow at emtex.de>> wrote:
> 
>     Hi all,
> 
>     we are currently experiencing jitter and dropping MOS score on an
>     increasing amount of calls on our system.
> 
>     i can't see any increased load on our system (neither CPU nor I/O wise).
> 
>     Current stats:
> 
>     ~ 15Mbit/s network traffic (on a 1Gbit/s network)
>     load average: 1,10, 0,82, 0,91 (16 core machine)
>     memory used: 2,26GB/16GB
> 
>     freeswitch at lc1> status
>     UP 0 years, 12 days, 13 hours, 30 minutes, 24 seconds, 649
>     milliseconds,
>     831 microseconds
>     FreeSWITCH (Version 1.10.1 -release-12-f9990221e6 64bit) is ready
>     137634 session(s) since startup
>     218 session(s) - peak 245, last 5min 222
>     1 session(s) per Sec out of max 30, peak 8, last 5min 2
>     1000 session(s) max
>     min idle cpu 0.00/95.37
>     Current Stack Size/Max 244K/244K
> 
>     System ist not virtualized.
> 
>     the only "strange" thing i see is the freeswitch "main"process that
>     consumes (according to htop) >=100% cpu (see attached screenshot).
>     Can this be the problem? if yes what can i do against it.
> 
>     if not what are possible bottlenecks i should weed out?
> 
>     if you need any additional informations to help me sort out this
>     problem
>     please don't hesitate to ask.
> 
>     Thanks in advance and best regards
>     Christoph
> 
> 
> 
>     -- 
>     ________________________________________________________________
> 
>     EMTEX GmbH
>     Christoph Russow
>     Software Engineer
> 
>     Bischof-Otto-Weg 9
>     D-91086 Aurachtal
>                                 Geschäftsführer:       Markus Enzinger
>     Tel. +49 9132 7490 0      Sitz der Gesellschaft: 91086 Aurachtal
>     Fax. +49 9132 7490 900    Amtsgericht Fürth:     HRB6804
>     _________________________________________________________________________________________________________________________________________
> 
>     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
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
> 


From david.villasmil.work at gmail.com  Thu Oct  7 10:40:33 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 7 Oct 2021 11:40:33 +0100
Subject: [Freeswitch-users] Bridge to other FS server has no audio until
	DTMF
In-Reply-To: <0100017c59766f67-90aa83ae-7742-464c-9fe2-9fa789725f4a-000000@email.amazonses.com>
References: <0100017c59766f67-90aa83ae-7742-464c-9fe2-9fa789725f4a-000000@email.amazonses.com>
Message-ID: <CAFGRPVrbaphJiRb09t2aGRetW34sB2agcy1moKbN0JCpgDpsfg@mail.gmail.com>

I seem to remember Brian saying this was because FS is waiting for the
remote end to send audio before starting itself. I believe he recommended
sending an empty (silence) to force the audio stream to be sent even if fs
hasn’t received anything.

On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi at avimarcus.net> wrote:

> I started a new thread in case anyone muted it... it wasn't simply a
> network issue.
>
> It seems the bridging occurs and dialplan processes, but no media flows -
> until DTMF from the A-leg.
> Call flow: PSTN (via carrier) to freeswitch A -> media and IVR ->
> freeswitch B.
>
> Calls directly from carrier to Freeswitch B are fine.
> Calls from a different carrier to Freeswitch A -> media and IVR ->
> Freeswitch B are also fine.
> So it sounds like a carrier/unique SIP/RTP issue, but since FS is in the
> media path, it's an FS issue...
>
>
> I actually mcguyvered this right now with a queue_dtmf before the bridge,
> to force the audio stream to update.
>
> Here's the log on freeswitch B:
>
> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>  log(DEBUG class chosen: 1234567)
> 2021-10-07 09:16:24.343175 [DEBUG
> ] mod_dptools.c:1879 class chosen: 1234567
> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>  javascript(conference/lookupAndJoinConference.js 1234567)
> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>  playback(class/hold-wait-teacher.wav)
> 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/
> 972581234567 at 172.123.123.123 entering state [completed][200]
> 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/
> 972581234567 at 172.123.123.123 entering state [ready][200]
> 2021-10-07 09:16:24.363379 [DEBUG
> ] switch_ivr_play_say.c:1486 Codec Activated L16 at 8000hz 1 channels 20ms
>
>
>
>
> 2021-10-07 09:16:34.903283 [DEBUG
> ] switch_rtp.c:7793 Correct audio ip/port confirmed.
> 2021-10-07 09:16:34.923190 [DEBUG] switch_rtp.c:8038 RTP RECV DTMF 3:2080
> 2021-10-07 09:16:34.923190 [INFO] switch_channel.c:522 RECV DTMF 3:2080
> 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3
> 2021-10-07 09:16:37.143169 [DEBUG
> ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav
>
>
> You can see a 10 second gap between call ready 200 and correct audio/ip
> and file done playing (it's a 2 second file), and this doesn't happen
> automatically, only when I choose to press something.
>
>
> Any ideas as to the root cause of this?
>
>
> -Avi Marcus
>
> ---------- Forwarded message ---------
> From: Avi Marcus <avi at avimarcus.net>
> Date: Wed, Oct 6, 2021 at 3:32 PM
> Subject: Bridge to other FS server has no audio ???
> To: FreeSWITCH Users Help <FreeSWITCH-users at lists.freeswitch.org>
>
>
> Any ideas on why a call doesn't have media? It used to work, but I think
> my upstream changed his SDP again.
>
> - FreeSWITCH Server A - call comes in and bypass_media bridges to FS
> server B. Media works.
> - FreeSWITCH Server A - call comes in and bridges to FS server B (not on
> bypass). Media works.
> - FreeSWITCH Server A - call comes in, gets answered, then bridges to FS
> server B. Call looks OK, but no media is flowing (I don't hear anything,
> PCAPs just have SIP, and there isn't 80kbps network traffic). All the same
> codecs are set in the json cdrs (PCMU).
>
> FS server B is to join a conference if that matters.
>
> I was assuming it had to do with codecs, but setting absolute_codec_string
> to PCMU doesn't make any difference in the logs  - it's already always PCMU.
>
> I have NO clue what further could cause this other than codecs, which seem
> to be fine. Any ideas please?
>
>
> -Avi Marcus
>
> _________________________________________________________________________
>
> 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/20211007/2d9eff9c/attachment-0001.html>

From avi at avimarcus.net  Thu Oct  7 11:10:42 2021
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 7 Oct 2021 11:10:42 +0000
Subject: [Freeswitch-users] Bridge to other FS server has no audio until
	DTMF
In-Reply-To: <CAFGRPVrbaphJiRb09t2aGRetW34sB2agcy1moKbN0JCpgDpsfg@mail.gmail.com>
References: <0100017c59766f67-90aa83ae-7742-464c-9fe2-9fa789725f4a-000000@email.amazonses.com>
 <CAFGRPVrbaphJiRb09t2aGRetW34sB2agcy1moKbN0JCpgDpsfg@mail.gmail.com>
Message-ID: <0100017c5a72bf6d-7c11a864-673c-437b-8780-b1a9b7cbe2e9-000000@email.amazonses.com>

I'm using dialplan bridge, so then the dialplan is over. How do I send
silence after the bridge...? An api_on_answer with a uuid_broadcast..
seems overly complicated.

<action application="bridge" data="sofia/external/number at yyy.bestfone.com
"/>


(And I don't know why there isn't audio - I had to set up an audio to get
to this options in the IVR... so there's already audio. And Server B also
started a file playback so should have initiated audio.)

-Avi Marcus

On Thu, Oct 7, 2021 at 1:41 PM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> I seem to remember Brian saying this was because FS is waiting for the
> remote end to send audio before starting itself. I believe he recommended
> sending an empty (silence) to force the audio stream to be sent even if fs
> hasn’t received anything.
>
> On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi at avimarcus.net> wrote:
>
>> I started a new thread in case anyone muted it... it wasn't simply a
>> network issue.
>>
>> It seems the bridging occurs and dialplan processes, but no media flows -
>> until DTMF from the A-leg.
>> Call flow: PSTN (via carrier) to freeswitch A -> media and IVR ->
>> freeswitch B.
>>
>> Calls directly from carrier to Freeswitch B are fine.
>> Calls from a different carrier to Freeswitch A -> media and IVR ->
>> Freeswitch B are also fine.
>> So it sounds like a carrier/unique SIP/RTP issue, but since FS is in the
>> media path, it's an FS issue...
>>
>>
>> I actually mcguyvered this right now with a queue_dtmf before the bridge,
>> to force the audio stream to update.
>>
>> Here's the log on freeswitch B:
>>
>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>  log(DEBUG class chosen: 1234567)
>> 2021-10-07 09:16:24.343175 [DEBUG
>> ] mod_dptools.c:1879 class chosen: 1234567
>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>  javascript(conference/lookupAndJoinConference.js 1234567)
>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>  playback(class/hold-wait-teacher.wav)
>> 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/
>> 972581234567 at 172.123.123.123 entering state [completed][200]
>> 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/
>> 972581234567 at 172.123.123.123 entering state [ready][200]
>> 2021-10-07 09:16:24.363379 [DEBUG
>> ] switch_ivr_play_say.c:1486 Codec Activated L16 at 8000hz 1 channels 20ms
>>
>>
>>
>>
>> 2021-10-07 09:16:34.903283 [DEBUG
>> ] switch_rtp.c:7793 Correct audio ip/port confirmed.
>> 2021-10-07 09:16:34.923190 [DEBUG] switch_rtp.c:8038 RTP RECV DTMF 3:2080
>> 2021-10-07 09:16:34.923190 [INFO] switch_channel.c:522 RECV DTMF 3:2080
>> 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3
>> 2021-10-07 09:16:37.143169 [DEBUG
>> ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav
>>
>>
>> You can see a 10 second gap between call ready 200 and correct audio/ip
>> and file done playing (it's a 2 second file), and this doesn't happen
>> automatically, only when I choose to press something.
>>
>>
>> Any ideas as to the root cause of this?
>>
>>
>> -Avi Marcus
>>
>> ---------- Forwarded message ---------
>> From: Avi Marcus <avi at avimarcus.net>
>> Date: Wed, Oct 6, 2021 at 3:32 PM
>> Subject: Bridge to other FS server has no audio ???
>> To: FreeSWITCH Users Help <FreeSWITCH-users at lists.freeswitch.org>
>>
>>
>> Any ideas on why a call doesn't have media? It used to work, but I think
>> my upstream changed his SDP again.
>>
>> - FreeSWITCH Server A - call comes in and bypass_media bridges to FS
>> server B. Media works.
>> - FreeSWITCH Server A - call comes in and bridges to FS server B (not on
>> bypass). Media works.
>> - FreeSWITCH Server A - call comes in, gets answered, then bridges to FS
>> server B. Call looks OK, but no media is flowing (I don't hear anything,
>> PCAPs just have SIP, and there isn't 80kbps network traffic). All the same
>> codecs are set in the json cdrs (PCMU).
>>
>> FS server B is to join a conference if that matters.
>>
>> I was assuming it had to do with codecs, but setting
>> absolute_codec_string to PCMU doesn't make any difference in the logs  -
>> it's already always PCMU.
>>
>> I have NO clue what further could cause this other than codecs, which
>> seem to be fine. Any ideas please?
>>
>>
>> -Avi Marcus
>>
>> _________________________________________________________________________
>>
>> 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/20211007/18749e8a/attachment.html>

From david.villasmil.work at gmail.com  Thu Oct  7 12:18:40 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 7 Oct 2021 13:18:40 +0100
Subject: [Freeswitch-users] Bridge to other FS server has no audio until
	DTMF
In-Reply-To: <0100017c5a72bf6d-7c11a864-673c-437b-8780-b1a9b7cbe2e9-000000@email.amazonses.com>
References: <0100017c59766f67-90aa83ae-7742-464c-9fe2-9fa789725f4a-000000@email.amazonses.com>
 <CAFGRPVrbaphJiRb09t2aGRetW34sB2agcy1moKbN0JCpgDpsfg@mail.gmail.com>
 <0100017c5a72bf6d-7c11a864-673c-437b-8780-b1a9b7cbe2e9-000000@email.amazonses.com>
Message-ID: <CAFGRPVouJaxa+0t0N38b-BpwGU5maEC7Bv7NF2f9wjHwg3XLAA@mail.gmail.com>

If you see rtp glowing both ways, then this is not the stalemate I was
talking about. The scenario I’m referring to is about FS not starting
sending rtp waiting for the other side to start sending, and the other side
doing the same thing, thus going into a stalemate. This is solved by
injecting a silence (I would do api_on_answer).

What you’re describing seems different to me.

On Thu, 7 Oct 2021 at 12:36, Avi Marcus <avi at avimarcus.net> wrote:

> I'm using dialplan bridge, so then the dialplan is over. How do I send
> silence after the bridge...? An api_on_answer with a uuid_broadcast..
> seems overly complicated.
>
> <action application="bridge" data="sofia/external/number at yyy.bestfone.com
> "/>
>
>
> (And I don't know why there isn't audio - I had to set up an audio to get
> to this options in the IVR... so there's already audio. And Server B also
> started a file playback so should have initiated audio.)
>
>
> -Avi Marcus
>
> On Thu, Oct 7, 2021 at 1:41 PM David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> I seem to remember Brian saying this was because FS is waiting for the
>> remote end to send audio before starting itself. I believe he recommended
>> sending an empty (silence) to force the audio stream to be sent even if fs
>> hasn’t received anything.
>>
>> On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> I started a new thread in case anyone muted it... it wasn't simply a
>>> network issue.
>>>
>>> It seems the bridging occurs and dialplan processes, but no media flows
>>> - until DTMF from the A-leg.
>>> Call flow: PSTN (via carrier) to freeswitch A -> media and IVR ->
>>> freeswitch B.
>>>
>>> Calls directly from carrier to Freeswitch B are fine.
>>> Calls from a different carrier to Freeswitch A -> media and IVR ->
>>> Freeswitch B are also fine.
>>> So it sounds like a carrier/unique SIP/RTP issue, but since FS is in the
>>> media path, it's an FS issue...
>>>
>>>
>>> I actually mcguyvered this right now with a queue_dtmf before the
>>> bridge, to force the audio stream to update.
>>>
>>> Here's the log on freeswitch B:
>>>
>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>  log(DEBUG class chosen: 1234567)
>>> 2021-10-07 09:16:24.343175 [DEBUG
>>> ] mod_dptools.c:1879 class chosen: 1234567
>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>  javascript(conference/lookupAndJoinConference.js 1234567)
>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>  playback(class/hold-wait-teacher.wav)
>>> 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/
>>> 972581234567 at 172.123.123.123 entering state [completed][200]
>>> 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/
>>> 972581234567 at 172.123.123.123 entering state [ready][200]
>>> 2021-10-07 09:16:24.363379 [DEBUG
>>> ] switch_ivr_play_say.c:1486 Codec Activated L16 at 8000hz 1 channels 20ms
>>>
>>>
>>>
>>>
>>> 2021-10-07 09:16:34.903283 [DEBUG
>>> ] switch_rtp.c:7793 Correct audio ip/port confirmed.
>>> 2021-10-07 09:16:34.923190 [DEBUG
>>> ] switch_rtp.c:8038 RTP RECV DTMF 3:2080
>>> 2021-10-07 09:16:34.923190 [INFO] switch_channel.c:522 RECV DTMF 3:2080
>>> 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3
>>> 2021-10-07 09:16:37.143169 [DEBUG
>>> ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav
>>>
>>>
>>> You can see a 10 second gap between call ready 200 and correct audio/ip
>>> and file done playing (it's a 2 second file), and this doesn't happen
>>> automatically, only when I choose to press something.
>>>
>>>
>>> Any ideas as to the root cause of this?
>>>
>>>
>>> -Avi Marcus
>>>
>>> ---------- Forwarded message ---------
>>> From: Avi Marcus <avi at avimarcus.net>
>>> Date: Wed, Oct 6, 2021 at 3:32 PM
>>> Subject: Bridge to other FS server has no audio ???
>>> To: FreeSWITCH Users Help <FreeSWITCH-users at lists.freeswitch.org>
>>>
>>>
>>> Any ideas on why a call doesn't have media? It used to work, but I think
>>> my upstream changed his SDP again.
>>>
>>> - FreeSWITCH Server A - call comes in and bypass_media bridges to FS
>>> server B. Media works.
>>> - FreeSWITCH Server A - call comes in and bridges to FS server B (not on
>>> bypass). Media works.
>>> - FreeSWITCH Server A - call comes in, gets answered, then bridges to FS
>>> server B. Call looks OK, but no media is flowing (I don't hear anything,
>>> PCAPs just have SIP, and there isn't 80kbps network traffic). All the same
>>> codecs are set in the json cdrs (PCMU).
>>>
>>> FS server B is to join a conference if that matters.
>>>
>>> I was assuming it had to do with codecs, but setting
>>> absolute_codec_string to PCMU doesn't make any difference in the logs  -
>>> it's already always PCMU.
>>>
>>> I have NO clue what further could cause this other than codecs, which
>>> seem to be fine. Any ideas please?
>>>
>>>
>>> -Avi Marcus
>>>
>>> _________________________________________________________________________
>>>
>>> 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

-- 
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/20211007/f7bda4e0/attachment.html>

From avi at avimarcus.net  Thu Oct  7 12:30:09 2021
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 7 Oct 2021 12:30:09 +0000
Subject: [Freeswitch-users] Bridge to other FS server has no audio until
	DTMF
In-Reply-To: <CAFGRPVouJaxa+0t0N38b-BpwGU5maEC7Bv7NF2f9wjHwg3XLAA@mail.gmail.com>
References: <0100017c59766f67-90aa83ae-7742-464c-9fe2-9fa789725f4a-000000@email.amazonses.com>
 <CAFGRPVrbaphJiRb09t2aGRetW34sB2agcy1moKbN0JCpgDpsfg@mail.gmail.com>
 <0100017c5a72bf6d-7c11a864-673c-437b-8780-b1a9b7cbe2e9-000000@email.amazonses.com>
 <CAFGRPVouJaxa+0t0N38b-BpwGU5maEC7Bv7NF2f9wjHwg3XLAA@mail.gmail.com>
Message-ID: <0100017c5abb7cb1-07d9dc17-acc2-4dac-95bb-024af41aca0e-000000@email.amazonses.com>

I meant there's audio from pstn to fs1, but indeed I'm observing no audio
between fs1 and fs2.

What api should I call with api on answer..?

On Thu, Oct 7, 2021, 3:19 PM David Villasmil <david.villasmil.work at gmail.com>
wrote:

> If you see rtp glowing both ways, then this is not the stalemate I was
> talking about. The scenario I’m referring to is about FS not starting
> sending rtp waiting for the other side to start sending, and the other side
> doing the same thing, thus going into a stalemate. This is solved by
> injecting a silence (I would do api_on_answer).
>
> What you’re describing seems different to me.
>
> On Thu, 7 Oct 2021 at 12:36, Avi Marcus <avi at avimarcus.net> wrote:
>
>> I'm using dialplan bridge, so then the dialplan is over. How do I send
>> silence after the bridge...? An api_on_answer with a uuid_broadcast..
>> seems overly complicated.
>>
>> <action application="bridge" data="sofia/external/number at yyy.bestfone.com
>> "/>
>>
>>
>> (And I don't know why there isn't audio - I had to set up an audio to get
>> to this options in the IVR... so there's already audio. And Server B also
>> started a file playback so should have initiated audio.)
>>
>>
>> -Avi Marcus
>>
>> On Thu, Oct 7, 2021 at 1:41 PM David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> I seem to remember Brian saying this was because FS is waiting for the
>>> remote end to send audio before starting itself. I believe he recommended
>>> sending an empty (silence) to force the audio stream to be sent even if fs
>>> hasn’t received anything.
>>>
>>> On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>>> I started a new thread in case anyone muted it... it wasn't simply a
>>>> network issue.
>>>>
>>>> It seems the bridging occurs and dialplan processes, but no media flows
>>>> - until DTMF from the A-leg.
>>>> Call flow: PSTN (via carrier) to freeswitch A -> media and IVR ->
>>>> freeswitch B.
>>>>
>>>> Calls directly from carrier to Freeswitch B are fine.
>>>> Calls from a different carrier to Freeswitch A -> media and IVR ->
>>>> Freeswitch B are also fine.
>>>> So it sounds like a carrier/unique SIP/RTP issue, but since FS is in
>>>> the media path, it's an FS issue...
>>>>
>>>>
>>>> I actually mcguyvered this right now with a queue_dtmf before the
>>>> bridge, to force the audio stream to update.
>>>>
>>>> Here's the log on freeswitch B:
>>>>
>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>>  log(DEBUG class chosen: 1234567)
>>>> 2021-10-07 09:16:24.343175 [DEBUG
>>>> ] mod_dptools.c:1879 class chosen: 1234567
>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>>  javascript(conference/lookupAndJoinConference.js 1234567)
>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>>  playback(class/hold-wait-teacher.wav)
>>>> 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/
>>>> 972581234567 at 172.123.123.123 entering state [completed][200]
>>>> 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/
>>>> 972581234567 at 172.123.123.123 entering state [ready][200]
>>>> 2021-10-07 09:16:24.363379 [DEBUG
>>>> ] switch_ivr_play_say.c:1486 Codec Activated L16 at 8000hz 1 channels 20ms
>>>>
>>>>
>>>>
>>>>
>>>> 2021-10-07 09:16:34.903283 [DEBUG
>>>> ] switch_rtp.c:7793 Correct audio ip/port confirmed.
>>>> 2021-10-07 09:16:34.923190 [DEBUG
>>>> ] switch_rtp.c:8038 RTP RECV DTMF 3:2080
>>>> 2021-10-07 09:16:34.923190 [INFO] switch_channel.c:522 RECV DTMF 3:2080
>>>> 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3
>>>> 2021-10-07 09:16:37.143169 [DEBUG
>>>> ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav
>>>>
>>>>
>>>> You can see a 10 second gap between call ready 200 and correct audio/ip
>>>> and file done playing (it's a 2 second file), and this doesn't happen
>>>> automatically, only when I choose to press something.
>>>>
>>>>
>>>> Any ideas as to the root cause of this?
>>>>
>>>>
>>>> -Avi Marcus
>>>>
>>>> ---------- Forwarded message ---------
>>>> From: Avi Marcus <avi at avimarcus.net>
>>>> Date: Wed, Oct 6, 2021 at 3:32 PM
>>>> Subject: Bridge to other FS server has no audio ???
>>>> To: FreeSWITCH Users Help <FreeSWITCH-users at lists.freeswitch.org>
>>>>
>>>>
>>>> Any ideas on why a call doesn't have media? It used to work, but I
>>>> think my upstream changed his SDP again.
>>>>
>>>> - FreeSWITCH Server A - call comes in and bypass_media bridges to FS
>>>> server B. Media works.
>>>> - FreeSWITCH Server A - call comes in and bridges to FS server B (not
>>>> on bypass). Media works.
>>>> - FreeSWITCH Server A - call comes in, gets answered, then bridges to
>>>> FS server B. Call looks OK, but no media is flowing (I don't hear anything,
>>>> PCAPs just have SIP, and there isn't 80kbps network traffic). All the same
>>>> codecs are set in the json cdrs (PCMU).
>>>>
>>>> FS server B is to join a conference if that matters.
>>>>
>>>> I was assuming it had to do with codecs, but setting
>>>> absolute_codec_string to PCMU doesn't make any difference in the logs  -
>>>> it's already always PCMU.
>>>>
>>>> I have NO clue what further could cause this other than codecs, which
>>>> seem to be fine. Any ideas please?
>>>>
>>>>
>>>> -Avi Marcus
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> 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
>
> --
> 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/20211007/499a1e09/attachment-0001.html>

From brian at freeswitch.com  Thu Oct  7 15:01:32 2021
From: brian at freeswitch.com (Brian West)
Date: Thu, 7 Oct 2021 10:01:32 -0500
Subject: [Freeswitch-users] Bridge to other FS server has no audio until
	DTMF
In-Reply-To: <0100017c5abb7cb1-07d9dc17-acc2-4dac-95bb-024af41aca0e-000000@email.amazonses.com>
References: <0100017c59766f67-90aa83ae-7742-464c-9fe2-9fa789725f4a-000000@email.amazonses.com>
 <CAFGRPVrbaphJiRb09t2aGRetW34sB2agcy1moKbN0JCpgDpsfg@mail.gmail.com>
 <0100017c5a72bf6d-7c11a864-673c-437b-8780-b1a9b7cbe2e9-000000@email.amazonses.com>
 <CAFGRPVouJaxa+0t0N38b-BpwGU5maEC7Bv7NF2f9wjHwg3XLAA@mail.gmail.com>
 <0100017c5abb7cb1-07d9dc17-acc2-4dac-95bb-024af41aca0e-000000@email.amazonses.com>
Message-ID: <CAEJMVkC4PgeTWQ=MMVDmSCac1UVObRsTKNCirbnHoBqedY3tFQ@mail.gmail.com>

execure_on_answer=playback::silence_stream://100 should solve it.

/b
PS, the non pc term that this has been said to be is
https://en.wikipedia.org/wiki/Mexican_standoff

On Thu, Oct 7, 2021 at 7:39 AM Avi Marcus <avi at avimarcus.net> wrote:

> I meant there's audio from pstn to fs1, but indeed I'm observing no audio
> between fs1 and fs2.
>
> What api should I call with api on answer..?
>
> On Thu, Oct 7, 2021, 3:19 PM David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> If you see rtp glowing both ways, then this is not the stalemate I was
>> talking about. The scenario I’m referring to is about FS not starting
>> sending rtp waiting for the other side to start sending, and the other side
>> doing the same thing, thus going into a stalemate. This is solved by
>> injecting a silence (I would do api_on_answer).
>>
>> What you’re describing seems different to me.
>>
>> On Thu, 7 Oct 2021 at 12:36, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> I'm using dialplan bridge, so then the dialplan is over. How do I send
>>> silence after the bridge...? An api_on_answer with a uuid_broadcast..
>>> seems overly complicated.
>>>
>>> <action application="bridge" data="sofia/external/
>>> number at yyy.bestfone.com"/>
>>>
>>>
>>> (And I don't know why there isn't audio - I had to set up an audio to
>>> get to this options in the IVR... so there's already audio. And Server B
>>> also started a file playback so should have initiated audio.)
>>>
>>>
>>> -Avi Marcus
>>>
>>> On Thu, Oct 7, 2021 at 1:41 PM David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>>> I seem to remember Brian saying this was because FS is waiting for the
>>>> remote end to send audio before starting itself. I believe he recommended
>>>> sending an empty (silence) to force the audio stream to be sent even if fs
>>>> hasn’t received anything.
>>>>
>>>> On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi at avimarcus.net> wrote:
>>>>
>>>>> I started a new thread in case anyone muted it... it wasn't simply a
>>>>> network issue.
>>>>>
>>>>> It seems the bridging occurs and dialplan processes, but no media
>>>>> flows - until DTMF from the A-leg.
>>>>> Call flow: PSTN (via carrier) to freeswitch A -> media and IVR ->
>>>>> freeswitch B.
>>>>>
>>>>> Calls directly from carrier to Freeswitch B are fine.
>>>>> Calls from a different carrier to Freeswitch A -> media and IVR ->
>>>>> Freeswitch B are also fine.
>>>>> So it sounds like a carrier/unique SIP/RTP issue, but since FS is in
>>>>> the media path, it's an FS issue...
>>>>>
>>>>>
>>>>> I actually mcguyvered this right now with a queue_dtmf before the
>>>>> bridge, to force the audio stream to update.
>>>>>
>>>>> Here's the log on freeswitch B:
>>>>>
>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>>>  log(DEBUG class chosen: 1234567)
>>>>> 2021-10-07 09:16:24.343175 [DEBUG
>>>>> ] mod_dptools.c:1879 class chosen: 1234567
>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>>>  javascript(conference/lookupAndJoinConference.js 1234567)
>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>>>  playback(class/hold-wait-teacher.wav)
>>>>> 2021-10-07 09:16:24.363379 [DEBUG
>>>>> ] sofia.c:7406 Channel sofia/external/972581234567 at 172.123.123.123
>>>>>  entering state [completed][200]
>>>>> 2021-10-07 09:16:24.363379 [DEBUG
>>>>> ] sofia.c:7406 Channel sofia/external/972581234567 at 172.123.123.123
>>>>>  entering state [ready][200]
>>>>> 2021-10-07 09:16:24.363379 [DEBUG
>>>>> ] switch_ivr_play_say.c:1486 Codec Activated L16 at 8000hz
>>>>>  1 channels 20ms
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> 2021-10-07 09:16:34.903283 [DEBUG
>>>>> ] switch_rtp.c:7793 Correct audio ip/port confirmed.
>>>>> 2021-10-07 09:16:34.923190 [DEBUG
>>>>> ] switch_rtp.c:8038 RTP RECV DTMF 3:2080
>>>>> 2021-10-07 09:16:34.923190 [INFO
>>>>> ] switch_channel.c:522 RECV DTMF 3:2080
>>>>> 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3
>>>>> 2021-10-07 09:16:37.143169 [DEBUG
>>>>> ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav
>>>>>
>>>>>
>>>>> You can see a 10 second gap between call ready 200 and correct
>>>>> audio/ip and file done playing (it's a 2 second file), and this doesn't
>>>>> happen automatically, only when I choose to press something.
>>>>>
>>>>>
>>>>> Any ideas as to the root cause of this?
>>>>>
>>>>>
>>>>> -Avi Marcus
>>>>>
>>>>> ---------- Forwarded message ---------
>>>>> From: Avi Marcus <avi at avimarcus.net>
>>>>> Date: Wed, Oct 6, 2021 at 3:32 PM
>>>>> Subject: Bridge to other FS server has no audio ???
>>>>> To: FreeSWITCH Users Help <FreeSWITCH-users at lists.freeswitch.org>
>>>>>
>>>>>
>>>>> Any ideas on why a call doesn't have media? It used to work, but I
>>>>> think my upstream changed his SDP again.
>>>>>
>>>>> - FreeSWITCH Server A - call comes in and bypass_media bridges to FS
>>>>> server B. Media works.
>>>>> - FreeSWITCH Server A - call comes in and bridges to FS server B (not
>>>>> on bypass). Media works.
>>>>> - FreeSWITCH Server A - call comes in, gets answered, then bridges to
>>>>> FS server B. Call looks OK, but no media is flowing (I don't hear anything,
>>>>> PCAPs just have SIP, and there isn't 80kbps network traffic). All the same
>>>>> codecs are set in the json cdrs (PCMU).
>>>>>
>>>>> FS server B is to join a conference if that matters.
>>>>>
>>>>> I was assuming it had to do with codecs, but setting
>>>>> absolute_codec_string to PCMU doesn't make any difference in the logs  -
>>>>> it's already always PCMU.
>>>>>
>>>>> I have NO clue what further could cause this other than codecs, which
>>>>> seem to be fine. Any ideas please?
>>>>>
>>>>>
>>>>> -Avi Marcus
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> 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
>>
>> --
>> 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/20211007/6cd1e309/attachment-0001.html>

From david.villasmil.work at gmail.com  Thu Oct  7 16:12:41 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 7 Oct 2021 17:12:41 +0100
Subject: [Freeswitch-users] Bridge to other FS server has no audio until
	DTMF
In-Reply-To: <CAEJMVkC4PgeTWQ=MMVDmSCac1UVObRsTKNCirbnHoBqedY3tFQ@mail.gmail.com>
References: <0100017c59766f67-90aa83ae-7742-464c-9fe2-9fa789725f4a-000000@email.amazonses.com>
 <CAFGRPVrbaphJiRb09t2aGRetW34sB2agcy1moKbN0JCpgDpsfg@mail.gmail.com>
 <0100017c5a72bf6d-7c11a864-673c-437b-8780-b1a9b7cbe2e9-000000@email.amazonses.com>
 <CAFGRPVouJaxa+0t0N38b-BpwGU5maEC7Bv7NF2f9wjHwg3XLAA@mail.gmail.com>
 <0100017c5abb7cb1-07d9dc17-acc2-4dac-95bb-024af41aca0e-000000@email.amazonses.com>
 <CAEJMVkC4PgeTWQ=MMVDmSCac1UVObRsTKNCirbnHoBqedY3tFQ@mail.gmail.com>
Message-ID: <CAFGRPVpDJxrvWxKqP4xR+_DTR=+Q-Exz3NhH9=H5C4y1E42TQQ@mail.gmail.com>

That’s the one!

On Thu, 7 Oct 2021 at 16:11, Brian West <brian at freeswitch.com> wrote:

> execure_on_answer=playback::silence_stream://100 should solve it.
>
> /b
> PS, the non pc term that this has been said to be is
> https://en.wikipedia.org/wiki/Mexican_standoff
>
> On Thu, Oct 7, 2021 at 7:39 AM Avi Marcus <avi at avimarcus.net> wrote:
>
>> I meant there's audio from pstn to fs1, but indeed I'm observing no audio
>> between fs1 and fs2.
>>
>> What api should I call with api on answer..?
>>
>> On Thu, Oct 7, 2021, 3:19 PM David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> If you see rtp glowing both ways, then this is not the stalemate I was
>>> talking about. The scenario I’m referring to is about FS not starting
>>> sending rtp waiting for the other side to start sending, and the other side
>>> doing the same thing, thus going into a stalemate. This is solved by
>>> injecting a silence (I would do api_on_answer).
>>>
>>> What you’re describing seems different to me.
>>>
>>> On Thu, 7 Oct 2021 at 12:36, Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>>> I'm using dialplan bridge, so then the dialplan is over. How do I send
>>>> silence after the bridge...? An api_on_answer with a uuid_broadcast..
>>>> seems overly complicated.
>>>>
>>>> <action application="bridge" data="sofia/external/
>>>> number at yyy.bestfone.com"/>
>>>>
>>>>
>>>> (And I don't know why there isn't audio - I had to set up an audio to
>>>> get to this options in the IVR... so there's already audio. And Server B
>>>> also started a file playback so should have initiated audio.)
>>>>
>>>>
>>>> -Avi Marcus
>>>>
>>>> On Thu, Oct 7, 2021 at 1:41 PM David Villasmil <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>>> I seem to remember Brian saying this was because FS is waiting for the
>>>>> remote end to send audio before starting itself. I believe he recommended
>>>>> sending an empty (silence) to force the audio stream to be sent even if fs
>>>>> hasn’t received anything.
>>>>>
>>>>> On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi at avimarcus.net> wrote:
>>>>>
>>>>>> I started a new thread in case anyone muted it... it wasn't simply a
>>>>>> network issue.
>>>>>>
>>>>>> It seems the bridging occurs and dialplan processes, but no media
>>>>>> flows - until DTMF from the A-leg.
>>>>>> Call flow: PSTN (via carrier) to freeswitch A -> media and IVR ->
>>>>>> freeswitch B.
>>>>>>
>>>>>> Calls directly from carrier to Freeswitch B are fine.
>>>>>> Calls from a different carrier to Freeswitch A -> media and IVR ->
>>>>>> Freeswitch B are also fine.
>>>>>> So it sounds like a carrier/unique SIP/RTP issue, but since FS is in
>>>>>> the media path, it's an FS issue...
>>>>>>
>>>>>>
>>>>>> I actually mcguyvered this right now with a queue_dtmf before the
>>>>>> bridge, to force the audio stream to update.
>>>>>>
>>>>>> Here's the log on freeswitch B:
>>>>>>
>>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>>>>  log(DEBUG class chosen: 1234567)
>>>>>> 2021-10-07 09:16:24.343175 [DEBUG
>>>>>> ] mod_dptools.c:1879 class chosen: 1234567
>>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>>>>  javascript(conference/lookupAndJoinConference.js 1234567)
>>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>>>>  playback(class/hold-wait-teacher.wav)
>>>>>> 2021-10-07 09:16:24.363379 [DEBUG
>>>>>> ] sofia.c:7406 Channel sofia/external/972581234567 at 172.123.123.123
>>>>>>  entering state [completed][200]
>>>>>> 2021-10-07 09:16:24.363379 [DEBUG
>>>>>> ] sofia.c:7406 Channel sofia/external/972581234567 at 172.123.123.123
>>>>>>  entering state [ready][200]
>>>>>> 2021-10-07 09:16:24.363379 [DEBUG
>>>>>> ] switch_ivr_play_say.c:1486 Codec Activated L16 at 8000hz
>>>>>>  1 channels 20ms
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> 2021-10-07 09:16:34.903283 [DEBUG
>>>>>> ] switch_rtp.c:7793 Correct audio ip/port confirmed.
>>>>>> 2021-10-07 09:16:34.923190 [DEBUG
>>>>>> ] switch_rtp.c:8038 RTP RECV DTMF 3:2080
>>>>>> 2021-10-07 09:16:34.923190 [INFO
>>>>>> ] switch_channel.c:522 RECV DTMF 3:2080
>>>>>> 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3
>>>>>> 2021-10-07 09:16:37.143169 [DEBUG
>>>>>> ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav
>>>>>>
>>>>>>
>>>>>> You can see a 10 second gap between call ready 200 and correct
>>>>>> audio/ip and file done playing (it's a 2 second file), and this doesn't
>>>>>> happen automatically, only when I choose to press something.
>>>>>>
>>>>>>
>>>>>> Any ideas as to the root cause of this?
>>>>>>
>>>>>>
>>>>>> -Avi Marcus
>>>>>>
>>>>>> ---------- Forwarded message ---------
>>>>>> From: Avi Marcus <avi at avimarcus.net>
>>>>>> Date: Wed, Oct 6, 2021 at 3:32 PM
>>>>>> Subject: Bridge to other FS server has no audio ???
>>>>>> To: FreeSWITCH Users Help <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>
>>>>>>
>>>>>> Any ideas on why a call doesn't have media? It used to work, but I
>>>>>> think my upstream changed his SDP again.
>>>>>>
>>>>>> - FreeSWITCH Server A - call comes in and bypass_media bridges to FS
>>>>>> server B. Media works.
>>>>>> - FreeSWITCH Server A - call comes in and bridges to FS server B (not
>>>>>> on bypass). Media works.
>>>>>> - FreeSWITCH Server A - call comes in, gets answered, then bridges to
>>>>>> FS server B. Call looks OK, but no media is flowing (I don't hear anything,
>>>>>> PCAPs just have SIP, and there isn't 80kbps network traffic). All the same
>>>>>> codecs are set in the json cdrs (PCMU).
>>>>>>
>>>>>> FS server B is to join a conference if that matters.
>>>>>>
>>>>>> I was assuming it had to do with codecs, but setting
>>>>>> absolute_codec_string to PCMU doesn't make any difference in the logs  -
>>>>>> it's already always PCMU.
>>>>>>
>>>>>> I have NO clue what further could cause this other than codecs, which
>>>>>> seem to be fine. Any ideas please?
>>>>>>
>>>>>>
>>>>>> -Avi Marcus
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>
>>>>>> 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
>>>
>>> --
>>> 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>
> _________________________________________________________________________
>
> 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/20211007/d86a0312/attachment-0001.html>

From asilva at wirelessmundi.com  Thu Oct  7 16:47:11 2021
From: asilva at wirelessmundi.com (=?utf-8?Q?Ant=C3=B3nio_Silva?=)
Date: Thu, 7 Oct 2021 17:47:11 +0100
Subject: Sofia SIP/503 System Busy
Message-ID: <5BD34AAD-18C6-4714-B98F-DD81FA145DE4@wirelessmundi.com>

Hi,


Recently update a server to Debian 10  with freeswitch git Version 1.10.7-dev git 432bfc0 2021-07-27, and after a few days I start to see lot of expired registers and new invites rejected with “ 503 System Busy”. I restart FS and everything is ok. 
Also no log print in FS console that could point me to the root cause of the issue. 
 
The server before the update was running debian 8  with freeswitch Version 1.10.4-dev git 546bebb 2020-08-03 and never see this issue.


The hardware of the server is a ProLiant DL165 G7: 
2 x AMD Opteron(tm) Processor 6128,  Cores por CPU: 8,  Cores totales: 16     
26 GB RAM

The system is not with high load at the time the issue was occurring, fs status report:

freeswitch at srvone> status
UP 0 years, 2 days, 21 hours, 50 minutes, 23 seconds, 264 milliseconds, 655 microseconds
FreeSWITCH (Version 1.10.7-dev git 432bfc0 2021-07-27 19:25:13Z 64bit) is ready
147010 session(s) since startup
86 session(s) - peak 251, last 5min 91 
0 session(s) per Sec out of max 50, peak 26, last 5min 5 
2000 session(s) max
min idle cpu 5.00/66.57
Current Stack Size/Max 240K/8192K


From the source code, Sofia.c:2496
			if (switch_queue_size(mod_sofia_globals.msg_queue) > (unsigned int)critical) {
				nua_respond(nh, 503, "System Busy", SIPTAG_RETRY_AFTER_STR("300"), NUTAG_WITH_THIS(nua), TAG_END());
				goto end;
			}


So the trigger here is the msg_queue hitting the critical value, for what I could check in the source code, is the value of 7200. 
This value is the maximum number of sip dialogs that FS can handle?

I’ve some endpoints registered using TCP, can they affect the msg_queue?  

I can’t think of a route cause for this issue… any hint? 

Thanks for the help. 


--
Saludos / Regards / Cumprimentos
António Silva




-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211007/449e0117/attachment.html>

From avi at avimarcus.net  Thu Oct  7 16:48:40 2021
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 7 Oct 2021 16:48:40 +0000
Subject: [Freeswitch-users] Bridge to other FS server has no audio until
	DTMF
In-Reply-To: <CAEJMVkC4PgeTWQ=MMVDmSCac1UVObRsTKNCirbnHoBqedY3tFQ@mail.gmail.com>
References: <0100017c59766f67-90aa83ae-7742-464c-9fe2-9fa789725f4a-000000@email.amazonses.com>
 <CAFGRPVrbaphJiRb09t2aGRetW34sB2agcy1moKbN0JCpgDpsfg@mail.gmail.com>
 <0100017c5a72bf6d-7c11a864-673c-437b-8780-b1a9b7cbe2e9-000000@email.amazonses.com>
 <CAFGRPVouJaxa+0t0N38b-BpwGU5maEC7Bv7NF2f9wjHwg3XLAA@mail.gmail.com>
 <0100017c5abb7cb1-07d9dc17-acc2-4dac-95bb-024af41aca0e-000000@email.amazonses.com>
 <CAEJMVkC4PgeTWQ=MMVDmSCac1UVObRsTKNCirbnHoBqedY3tFQ@mail.gmail.com>
Message-ID: <0100017c5ba82bd9-711151e4-7603-4eea-a6d9-a9029c06383c-000000@email.amazonses.com>

I had to do this to get it to execute on the B leg:
<action application="export" data="nolocal:execute_on_answer=playback
silence_stream://100"/>

... but it didn't help. Only DTMF worked... either manually dialed or via
queue_dtmf
Freeswitch A waited for my DTMF to actually send the silence.
Version 1.10.6 -release-18-1ff9d0a60e 64bit


 2021-10-07 16:37:10.523346 [DEBUG] switch_core_media.c:9025 Set comfort
noise payload to 13
 2021-10-07 16:37:10.523346 [NOTICE] sofia.c:8586 Channel [sofia/external/
JOIN_CLASS_7229999 at voip.bestfone.com] has been answered
 EXECUTE [depth=1] sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com
playback(silence_stream://100)
 2021-10-07 16:37:10.523346 [DEBUG] switch_ivr_play_say.c:1486 Codec
Activated L16 at 8000hz 1 channels 20ms

 -- 20 seconds later when I pressed a button --

 2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_play_say.c:1931 done playing
file silence_stream://100
 2021-10-07 16:37:30.563357 [DEBUG] switch_channel.c:3865 (sofia/external/
JOIN_CLASS_7229999 at voip.bestfone.com) Callstate Change DOWN -> ACTIVE
 2021-10-07 16:37:30.563357 [DEBUG] switch_ivr_bridge.c:1793
(sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
 2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:585
(sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com) Running State Change
CS_EXCHANGE_MEDIA (Cur 12 Tot 351090)
 2021-10-07 16:37:30.563357 [DEBUG] switch_core_state_machine.c:654
(sofia/external/JOIN_CLASS_7229999 at voip.bestfone.com) State EXCHANGE_MEDIA
 2021-10-07 16:37:30.563357 [DEBUG] mod_sofia.c:656 SOFIA EXCHANGE_MEDIA
 2021-10-07 16:37:30.583346 [DEBUG] switch_rtp.c:5619 Send start packet for
[5] ts=960 dur=160/160/2000 seq=26795 lw=960



This seemingly shouldn't be an issue. FS1 already has active media from the
A leg, so it should initiate to the B leg. The B leg has been instructed to
play a file, so it should initiate to the A leg...
But if this is somehow unavoidable, perhaps we need a workaround config,
where we have a simple variable in the bridge string to avoid the standoff?

-Avi Marcus



On Thu, Oct 7, 2021 at 6:01 PM Brian West <brian at freeswitch.com> wrote:

> execure_on_answer=playback::silence_stream://100 should solve it.
>
> /b
> PS, the non pc term that this has been said to be is
> https://en.wikipedia.org/wiki/Mexican_standoff
>
> On Thu, Oct 7, 2021 at 7:39 AM Avi Marcus <avi at avimarcus.net> wrote:
>
>> I meant there's audio from pstn to fs1, but indeed I'm observing no audio
>> between fs1 and fs2.
>>
>> What api should I call with api on answer..?
>>
>> On Thu, Oct 7, 2021, 3:19 PM David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> If you see rtp glowing both ways, then this is not the stalemate I was
>>> talking about. The scenario I’m referring to is about FS not starting
>>> sending rtp waiting for the other side to start sending, and the other side
>>> doing the same thing, thus going into a stalemate. This is solved by
>>> injecting a silence (I would do api_on_answer).
>>>
>>> What you’re describing seems different to me.
>>>
>>> On Thu, 7 Oct 2021 at 12:36, Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>>> I'm using dialplan bridge, so then the dialplan is over. How do I send
>>>> silence after the bridge...? An api_on_answer with a uuid_broadcast..
>>>> seems overly complicated.
>>>>
>>>> <action application="bridge" data="sofia/external/
>>>> number at yyy.bestfone.com"/>
>>>>
>>>>
>>>> (And I don't know why there isn't audio - I had to set up an audio to
>>>> get to this options in the IVR... so there's already audio. And Server B
>>>> also started a file playback so should have initiated audio.)
>>>>
>>>>
>>>> -Avi Marcus
>>>>
>>>> On Thu, Oct 7, 2021 at 1:41 PM David Villasmil <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>>> I seem to remember Brian saying this was because FS is waiting for the
>>>>> remote end to send audio before starting itself. I believe he recommended
>>>>> sending an empty (silence) to force the audio stream to be sent even if fs
>>>>> hasn’t received anything.
>>>>>
>>>>> On Thu, 7 Oct 2021 at 07:50, Avi Marcus <avi at avimarcus.net> wrote:
>>>>>
>>>>>> I started a new thread in case anyone muted it... it wasn't simply a
>>>>>> network issue.
>>>>>>
>>>>>> It seems the bridging occurs and dialplan processes, but no media
>>>>>> flows - until DTMF from the A-leg.
>>>>>> Call flow: PSTN (via carrier) to freeswitch A -> media and IVR ->
>>>>>> freeswitch B.
>>>>>>
>>>>>> Calls directly from carrier to Freeswitch B are fine.
>>>>>> Calls from a different carrier to Freeswitch A -> media and IVR ->
>>>>>> Freeswitch B are also fine.
>>>>>> So it sounds like a carrier/unique SIP/RTP issue, but since FS is in
>>>>>> the media path, it's an FS issue...
>>>>>>
>>>>>>
>>>>>> I actually mcguyvered this right now with a queue_dtmf before the
>>>>>> bridge, to force the audio stream to update.
>>>>>>
>>>>>> Here's the log on freeswitch B:
>>>>>>
>>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>>>>  log(DEBUG class chosen: 1234567)
>>>>>> 2021-10-07 09:16:24.343175 [DEBUG
>>>>>> ] mod_dptools.c:1879 class chosen: 1234567
>>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>>>>  javascript(conference/lookupAndJoinConference.js 1234567)
>>>>>> EXECUTE [depth=0] sofia/external/972581234567 at 172.123.123.123
>>>>>>  playback(class/hold-wait-teacher.wav)
>>>>>> 2021-10-07 09:16:24.363379 [DEBUG
>>>>>> ] sofia.c:7406 Channel sofia/external/972581234567 at 172.123.123.123
>>>>>>  entering state [completed][200]
>>>>>> 2021-10-07 09:16:24.363379 [DEBUG
>>>>>> ] sofia.c:7406 Channel sofia/external/972581234567 at 172.123.123.123
>>>>>>  entering state [ready][200]
>>>>>> 2021-10-07 09:16:24.363379 [DEBUG
>>>>>> ] switch_ivr_play_say.c:1486 Codec Activated L16 at 8000hz
>>>>>>  1 channels 20ms
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> 2021-10-07 09:16:34.903283 [DEBUG
>>>>>> ] switch_rtp.c:7793 Correct audio ip/port confirmed.
>>>>>> 2021-10-07 09:16:34.923190 [DEBUG
>>>>>> ] switch_rtp.c:8038 RTP RECV DTMF 3:2080
>>>>>> 2021-10-07 09:16:34.923190 [INFO
>>>>>> ] switch_channel.c:522 RECV DTMF 3:2080
>>>>>> 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3
>>>>>> 2021-10-07 09:16:37.143169 [DEBUG
>>>>>> ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav
>>>>>>
>>>>>>
>>>>>> You can see a 10 second gap between call ready 200 and correct
>>>>>> audio/ip and file done playing (it's a 2 second file), and this doesn't
>>>>>> happen automatically, only when I choose to press something.
>>>>>>
>>>>>>
>>>>>> Any ideas as to the root cause of this?
>>>>>>
>>>>>>
>>>>>> -Avi Marcus
>>>>>>
>>>>>> ---------- Forwarded message ---------
>>>>>> From: Avi Marcus <avi at avimarcus.net>
>>>>>> Date: Wed, Oct 6, 2021 at 3:32 PM
>>>>>> Subject: Bridge to other FS server has no audio ???
>>>>>> To: FreeSWITCH Users Help <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>
>>>>>>
>>>>>> Any ideas on why a call doesn't have media? It used to work, but I
>>>>>> think my upstream changed his SDP again.
>>>>>>
>>>>>> - FreeSWITCH Server A - call comes in and bypass_media bridges to FS
>>>>>> server B. Media works.
>>>>>> - FreeSWITCH Server A - call comes in and bridges to FS server B (not
>>>>>> on bypass). Media works.
>>>>>> - FreeSWITCH Server A - call comes in, gets answered, then bridges to
>>>>>> FS server B. Call looks OK, but no media is flowing (I don't hear anything,
>>>>>> PCAPs just have SIP, and there isn't 80kbps network traffic). All the same
>>>>>> codecs are set in the json cdrs (PCMU).
>>>>>>
>>>>>> FS server B is to join a conference if that matters.
>>>>>>
>>>>>> I was assuming it had to do with codecs, but setting
>>>>>> absolute_codec_string to PCMU doesn't make any difference in the logs  -
>>>>>> it's already always PCMU.
>>>>>>
>>>>>> I have NO clue what further could cause this other than codecs, which
>>>>>> seem to be fine. Any ideas please?
>>>>>>
>>>>>>
>>>>>> -Avi Marcus
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>
>>>>>> 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
>>>
>>> --
>>> 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>
> _________________________________________________________________________
>
> 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/20211007/8ae6421f/attachment-0001.html>

From davidswalkabout at gmail.com  Thu Oct  7 20:47:25 2021
From: davidswalkabout at gmail.com (David P)
Date: Fri, 8 Oct 2021 09:47:25 +1300
Subject: [Freeswitch-users] Avoiding DDoS with verto?
Message-ID: <CAE23trvy+mH9EPY4W0sND=azF7dibwtHEfY1Yvk8fLcYQjqOxQ@mail.gmail.com>

After reading about recent DDoS attacks on VOIP providers in
https://www.rtcsec.com/post/2021/09/massive-ddos-attacks-on-voip-providers-and-simulated-ddos-testing/
in which Freeswitch is mentioned, I wondered what current practices are for
services that must serve the public Internet.

For example, a service that is purely verto-based seems like it could
protect itself in this way:

1) Block requests on all ports (except the verto WSS login) unless the
request is from an address that's already part of signaling.

2) To protect signaling, put it behind AWS API Gateway, which provides
rate-limiting, and add an authorization check.

Does this seem like it would defend against DDoS? Can FS be configured to
do #1?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211008/91c1e539/attachment.html>

From bipin at xbipin.com  Thu Oct  7 12:04:20 2021
From: bipin at xbipin.com (Bipin Patel)
Date: Thu, 07 Oct 2021 16:04:20 +0400
Subject: [Freeswitch-users] mod_fifo onhook member uuid not expanding
Message-ID: <0a76fdaecf9e176ac14f24c7bee5cffd@xbipin.com>

hi,

im pretty much stuck in this situation where I have users dialing into a 
DID which FS answers and places callers in a fifo queue, the calls get 
bridged to on-hook members from the fifo list, they want the ability to 
press *2 at anytime to record the call, all this works but I need to 
store the recording filename with the uuid which doesnt seem to expand 
at all

my diaplan is as below:
     <extension name="support_fifo">
         <condition field="destination_number" expression="^0001$">
             <action application="answer"/>
             <action application="set" 
data="fifo_chime_list=$${base_dir}/sounds/en/us/callie/misc/8000/misc-freeswitch_is_state_of_the_art.wav"/>
             <action application="set" data="fifo_chime_freq=15"/>
             <action application="set" data="fifo_orbit_dialplan=XML"/>
             <action application="set" 
data="fifo_orbit_context=lanusersdialout"/>
             <action application="set" 
data="fifo_orbit_announce=$${base_dir}/sounds/en/us/callie/ivr/8000/ivr-connect_actual_human_being.wav"/>
             <action application="set" 
data="fifo_override_announce=$${base_dir}/sounds/en/us/callie/ivr/8000/ivr-one_moment_please.wav"/>
             <action application="fifo" data="support_fifo@$${domain} in 
undef $${base_dir}/sounds/music/8000/moh.wav"/>
         </condition>
     </extension>

my fifo config is as below:
         <fifo name="support_fifo@$${domain}" importance="0">
             <member timeout="15" simo="1" 
lag="5">{execute_on_answer='bind_meta_app 2 a s 
record_session::$${base_dir}/recordings/archive/${uuid}.mp3',Caller-Context=lanusersdialout,absolute_codec_string=PCMU,call_timeout=30,fifo_member_wait=nowait}user/88</member>
         </fifo>

anyways to pass any sort of uuid from the dialplan or get the uuid of 
the originating channel using any variables?

any help is very much appreciated


-- 
Regards,
Bipin



From adrian.worutowicz at esifrance.net  Mon Oct 11 12:49:53 2021
From: adrian.worutowicz at esifrance.net (Adrian Worutowicz)
Date: Mon, 11 Oct 2021 12:49:53 +0000
Subject: [Freeswitch-users] User-to-User RFC 7433
Message-ID: <PR0P264MB1625552013DBA7FAC550E27FF5B59@PR0P264MB1625.FRAP264.PROD.OUTLOOK.COM>

Hello,

I need to add the User-to-User field to the SIP header - for incoming calls.
I have not managed to do it (neither for incoming nor for outgoing calls).

In ./conf/dialplan/default.xml,  <extension> section, I added :

<action application="set" data="sip_h_User-to-User=123"/>
or
<action application="set"><![CDATA[sip_h_User-to-User=${sip_Call_ID};tenant=1101;DID=${sip_to_user}]]></action>

No result.


In sofia_presence.c, I've seen some code for "User-to-User", so it should be possible. But how to do it ?

Best regards,
Adrian.



-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211011/4a163842/attachment.html>

From brian at freeswitch.com  Mon Oct 11 13:18:29 2021
From: brian at freeswitch.com (Brian West)
Date: Mon, 11 Oct 2021 08:18:29 -0500
Subject: [Freeswitch-users] User-to-User RFC 7433
In-Reply-To: <PR0P264MB1625552013DBA7FAC550E27FF5B59@PR0P264MB1625.FRAP264.PROD.OUTLOOK.COM>
References: <PR0P264MB1625552013DBA7FAC550E27FF5B59@PR0P264MB1625.FRAP264.PROD.OUTLOOK.COM>
Message-ID: <CAEJMVkBXqsef9HCbUQyQsZFyGh+D9SC-1LYAt4XjiSrcG4GWjQ@mail.gmail.com>

Code will probably need to be written for that.

/b


On Mon, Oct 11, 2021 at 8:17 AM Adrian Worutowicz <
adrian.worutowicz at esifrance.net> wrote:

> Hello,
>
>
>
> I need to add the User-to-User field to the SIP header – for incoming
> calls.
>
> I have not managed to do it (neither for incoming nor for outgoing calls).
>
>
>
> In ./conf/dialplan/default.xml,  <extension> section, I added :
>
>
>
> <action application="set" data="sip_h_User-to-User=123"/>
>
> or
>
> <action
> application="set"><![CDATA[sip_h_User-to-User=${sip_Call_ID};tenant=1101;DID=${sip_to_user}]]></action>
>
>
>
> No result.
>
>
>
>
>
> In sofia_presence.c, I’ve seen some code for "User-to-User", so it should
> be possible. But how to do it ?
>
>
>
> Best regards,
>
> Adrian.
>
>
>
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

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/20211011/ad2bcb37/attachment.html>

From adrian.worutowicz at esifrance.net  Mon Oct 11 13:26:47 2021
From: adrian.worutowicz at esifrance.net (Adrian Worutowicz)
Date: Mon, 11 Oct 2021 13:26:47 +0000
Subject: [Freeswitch-users] User-to-User RFC 7433
In-Reply-To: <CAEJMVkBXqsef9HCbUQyQsZFyGh+D9SC-1LYAt4XjiSrcG4GWjQ@mail.gmail.com>
References: <PR0P264MB1625552013DBA7FAC550E27FF5B59@PR0P264MB1625.FRAP264.PROD.OUTLOOK.COM>
 <CAEJMVkBXqsef9HCbUQyQsZFyGh+D9SC-1LYAt4XjiSrcG4GWjQ@mail.gmail.com>
Message-ID: <PR0P264MB1625790DCDE0E98FBC20FC4AF5B59@PR0P264MB1625.FRAP264.PROD.OUTLOOK.COM>

Thanks Brian,

I’ve read that it is possible to insert any line to the SIP header.

Is it true ? If so, how to do it ?

Adrian.

De : Brian West <brian at freeswitch.com>
Envoyé : lundi 11 octobre 2021 15:18
À : Adrian Worutowicz <adrian.worutowicz at esifrance.net>
Cc : FreeSWITCH-users at lists.freeswitch.org
Objet : Re: [Freeswitch-users] User-to-User RFC 7433

Code will probably need to be written for that.

/b


On Mon, Oct 11, 2021 at 8:17 AM Adrian Worutowicz <adrian.worutowicz at esifrance.net<mailto:adrian.worutowicz at esifrance.net>> wrote:
Hello,

I need to add the User-to-User field to the SIP header – for incoming calls.
I have not managed to do it (neither for incoming nor for outgoing calls).

In ./conf/dialplan/default.xml,  <extension> section, I added :

<action application="set" data="sip_h_User-to-User=123"/>
or
<action application="set"><![CDATA[sip_h_User-to-User=${sip_Call_ID};tenant=1101;DID=${sip_to_user}]]></action>

No result.


In sofia_presence.c, I’ve seen some code for "User-to-User", so it should be possible. But how to do it ?

Best regards,
Adrian.



_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales at freeswitch.com<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/>

[cid:~WRD0000.jpg]<https://www.facebook.com/freeswitch>[cid:~WRD0000.jpg]<https://twitter.com/freeswitch>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211011/797fcaeb/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: ~WRD0000.jpg
Type: image/jpeg
Size: 823 bytes
Desc: ~WRD0000.jpg
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211011/797fcaeb/attachment-0001.jpg>

From babak.yakhchali at gmail.com  Tue Oct 12 10:10:43 2021
From: babak.yakhchali at gmail.com (Babak Yakhchali)
Date: Tue, 12 Oct 2021 13:40:43 +0330
Subject: [Freeswitch-users] Passing custom sip headers using re-invite
Message-ID: <CAD3oNZxacQhmrpHxWSZgBnQ1h9jGP5aRU-GY+ArUXQDodzEX7w@mail.gmail.com>

Hi
I need to pass some custom sip headers between 2 fs boxes after the call is
established. To do this I'm using "uuid_setvar sip_h_X-myvar=value" on the
channel and calling    uuid_media_reneg to create a re-invite, fs is
sending out the re-invite with new custom headers but the other fs box is
not adding these new custom sip headers to the channel. How can I make fs
to set new custom headers on the receiving side of the re-invite?
fs1 ----invite---> fs2
.... call is established
fs1 <---re-invite--- fs2 (re-invite with new custom sip X headers)
fs1 ... not adding new headers to channel

thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211012/2a0d4b84/attachment.html>

From dragos at freeswitch.org  Tue Oct 12 10:55:06 2021
From: dragos at freeswitch.org (Dragos Oancea)
Date: Tue, 12 Oct 2021 13:55:06 +0300
Subject: [Freeswitch-users] Passing custom sip headers using re-invite
In-Reply-To: <CAD3oNZxacQhmrpHxWSZgBnQ1h9jGP5aRU-GY+ArUXQDodzEX7w@mail.gmail.com>
References: <CAD3oNZxacQhmrpHxWSZgBnQ1h9jGP5aRU-GY+ArUXQDodzEX7w@mail.gmail.com>
Message-ID: <CAGvJV4qiS5swYrWTHYYSqLYr=T_UK7qJmkaUf693vqtxO0-=Mw@mail.gmail.com>

The custom header name, is it prefixed with "X-" ?

On Tue, Oct 12, 2021 at 1:11 PM Babak Yakhchali <babak.yakhchali at gmail.com>
wrote:

> Hi
> I need to pass some custom sip headers between 2 fs boxes after the call
> is established. To do this I'm using "uuid_setvar sip_h_X-myvar=value" on
> the channel and calling    uuid_media_reneg to create a re-invite, fs is
> sending out the re-invite with new custom headers but the other fs box is
> not adding these new custom sip headers to the channel. How can I make fs
> to set new custom headers on the receiving side of the re-invite?
> fs1 ----invite---> fs2
> .... call is established
> fs1 <---re-invite--- fs2 (re-invite with new custom sip X headers)
> fs1 ... not adding new headers to channel
>
> 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211012/8b31e2d3/attachment.html>

From babak.yakhchali at gmail.com  Tue Oct 12 12:50:30 2021
From: babak.yakhchali at gmail.com (Babak Yakhchali)
Date: Tue, 12 Oct 2021 16:20:30 +0330
Subject: [Freeswitch-users] Passing custom sip headers using re-invite
In-Reply-To: <CAGvJV4qiS5swYrWTHYYSqLYr=T_UK7qJmkaUf693vqtxO0-=Mw@mail.gmail.com>
References: <CAD3oNZxacQhmrpHxWSZgBnQ1h9jGP5aRU-GY+ArUXQDodzEX7w@mail.gmail.com>
 <CAGvJV4qiS5swYrWTHYYSqLYr=T_UK7qJmkaUf693vqtxO0-=Mw@mail.gmail.com>
Message-ID: <CAD3oNZzNfXqW3WMXdu=hsF9EWivX8EBy+9ghT92fo0KCC3gE7g@mail.gmail.com>

yes

On Tue, Oct 12, 2021 at 2:48 PM Dragos Oancea <dragos at freeswitch.org> wrote:

> The custom header name, is it prefixed with "X-" ?
>
> On Tue, Oct 12, 2021 at 1:11 PM Babak Yakhchali <babak.yakhchali at gmail.com>
> wrote:
>
>> Hi
>> I need to pass some custom sip headers between 2 fs boxes after the call
>> is established. To do this I'm using "uuid_setvar sip_h_X-myvar=value" on
>> the channel and calling    uuid_media_reneg to create a re-invite, fs is
>> sending out the re-invite with new custom headers but the other fs box is
>> not adding these new custom sip headers to the channel. How can I make fs
>> to set new custom headers on the receiving side of the re-invite?
>> fs1 ----invite---> fs2
>> .... call is established
>> fs1 <---re-invite--- fs2 (re-invite with new custom sip X headers)
>> fs1 ... not adding new headers to channel
>>
>> 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
>
> _________________________________________________________________________
>
> 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/20211012/1c2453c0/attachment.html>

From david.villasmil.work at gmail.com  Tue Oct 12 13:47:41 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 12 Oct 2021 14:47:41 +0100
Subject: [Freeswitch-users] Passing custom sip headers using re-invite
In-Reply-To: <CAD3oNZzNfXqW3WMXdu=hsF9EWivX8EBy+9ghT92fo0KCC3gE7g@mail.gmail.com>
References: <CAD3oNZxacQhmrpHxWSZgBnQ1h9jGP5aRU-GY+ArUXQDodzEX7w@mail.gmail.com>
 <CAGvJV4qiS5swYrWTHYYSqLYr=T_UK7qJmkaUf693vqtxO0-=Mw@mail.gmail.com>
 <CAD3oNZzNfXqW3WMXdu=hsF9EWivX8EBy+9ghT92fo0KCC3gE7g@mail.gmail.com>
Message-ID: <CAFGRPVojJ=dK+BuaV=RP+8efbrwMkuAGr8-b30CN6WJyqUdbTw@mail.gmail.com>

How are you trying to use the header on the other side?


On Tue, 12 Oct 2021 at 14:07, Babak Yakhchali <babak.yakhchali at gmail.com>
wrote:

> yes
>
> On Tue, Oct 12, 2021 at 2:48 PM Dragos Oancea <dragos at freeswitch.org>
> wrote:
>
>> The custom header name, is it prefixed with "X-" ?
>>
>> On Tue, Oct 12, 2021 at 1:11 PM Babak Yakhchali <
>> babak.yakhchali at gmail.com> wrote:
>>
>>> Hi
>>> I need to pass some custom sip headers between 2 fs boxes after the call
>>> is established. To do this I'm using "uuid_setvar sip_h_X-myvar=value" on
>>> the channel and calling    uuid_media_reneg to create a re-invite, fs is
>>> sending out the re-invite with new custom headers but the other fs box is
>>> not adding these new custom sip headers to the channel. How can I make fs
>>> to set new custom headers on the receiving side of the re-invite?
>>> fs1 ----invite---> fs2
>>> .... call is established
>>> fs1 <---re-invite--- fs2 (re-invite with new custom sip X headers)
>>> fs1 ... not adding new headers to channel
>>>
>>> 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
>>
>> _________________________________________________________________________
>>
>> 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/20211012/218938a5/attachment.html>

From babak.yakhchali at gmail.com  Tue Oct 12 16:25:11 2021
From: babak.yakhchali at gmail.com (Babak Yakhchali)
Date: Tue, 12 Oct 2021 19:55:11 +0330
Subject: [Freeswitch-users] Passing custom sip headers using re-invite
In-Reply-To: <CAFGRPVojJ=dK+BuaV=RP+8efbrwMkuAGr8-b30CN6WJyqUdbTw@mail.gmail.com>
References: <CAD3oNZxacQhmrpHxWSZgBnQ1h9jGP5aRU-GY+ArUXQDodzEX7w@mail.gmail.com>
 <CAGvJV4qiS5swYrWTHYYSqLYr=T_UK7qJmkaUf693vqtxO0-=Mw@mail.gmail.com>
 <CAD3oNZzNfXqW3WMXdu=hsF9EWivX8EBy+9ghT92fo0KCC3gE7g@mail.gmail.com>
 <CAFGRPVojJ=dK+BuaV=RP+8efbrwMkuAGr8-b30CN6WJyqUdbTw@mail.gmail.com>
Message-ID: <CAD3oNZxPJVqfVf7UAvbfa+ZzL3npmsFfSqGZ=rvQjeiNJjQRFg@mail.gmail.com>

I have a lua script as an event hook on sofia::reinvite, I want to get the
variable in the handler and do some logic on it.

On Tue, Oct 12, 2021 at 5:40 PM David Villasmil <
david.villasmil.work at gmail.com> wrote:

>
> How are you trying to use the header on the other side?
>
>
> On Tue, 12 Oct 2021 at 14:07, Babak Yakhchali <babak.yakhchali at gmail.com>
> wrote:
>
>> yes
>>
>> On Tue, Oct 12, 2021 at 2:48 PM Dragos Oancea <dragos at freeswitch.org>
>> wrote:
>>
>>> The custom header name, is it prefixed with "X-" ?
>>>
>>> On Tue, Oct 12, 2021 at 1:11 PM Babak Yakhchali <
>>> babak.yakhchali at gmail.com> wrote:
>>>
>>>> Hi
>>>> I need to pass some custom sip headers between 2 fs boxes after the
>>>> call is established. To do this I'm using "uuid_setvar sip_h_X-myvar=value"
>>>> on the channel and calling    uuid_media_reneg to create a re-invite, fs is
>>>> sending out the re-invite with new custom headers but the other fs box is
>>>> not adding these new custom sip headers to the channel. How can I make fs
>>>> to set new custom headers on the receiving side of the re-invite?
>>>> fs1 ----invite---> fs2
>>>> .... call is established
>>>> fs1 <---re-invite--- fs2 (re-invite with new custom sip X headers)
>>>> fs1 ... not adding new headers to channel
>>>>
>>>> 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
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211012/6b5f1aea/attachment-0001.html>

From david.villasmil.work at gmail.com  Tue Oct 12 22:27:54 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 12 Oct 2021 23:27:54 +0100
Subject: [Freeswitch-users] Passing custom sip headers using re-invite
In-Reply-To: <CAD3oNZxPJVqfVf7UAvbfa+ZzL3npmsFfSqGZ=rvQjeiNJjQRFg@mail.gmail.com>
References: <CAD3oNZxacQhmrpHxWSZgBnQ1h9jGP5aRU-GY+ArUXQDodzEX7w@mail.gmail.com>
 <CAGvJV4qiS5swYrWTHYYSqLYr=T_UK7qJmkaUf693vqtxO0-=Mw@mail.gmail.com>
 <CAD3oNZzNfXqW3WMXdu=hsF9EWivX8EBy+9ghT92fo0KCC3gE7g@mail.gmail.com>
 <CAFGRPVojJ=dK+BuaV=RP+8efbrwMkuAGr8-b30CN6WJyqUdbTw@mail.gmail.com>
 <CAD3oNZxPJVqfVf7UAvbfa+ZzL3npmsFfSqGZ=rvQjeiNJjQRFg@mail.gmail.com>
Message-ID: <CAFGRPVqO-qAS0EAmmkw1r=tyAFE4Hir=bJ0O4EFw+TWJ-AY_fg@mail.gmail.com>

again, exactly how are you trying to read the header?

Regards,

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


On Tue, Oct 12, 2021 at 5:45 PM Babak Yakhchali <babak.yakhchali at gmail.com>
wrote:

> I have a lua script as an event hook on sofia::reinvite, I want to get the
> variable in the handler and do some logic on it.
>
> On Tue, Oct 12, 2021 at 5:40 PM David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>>
>> How are you trying to use the header on the other side?
>>
>>
>> On Tue, 12 Oct 2021 at 14:07, Babak Yakhchali <babak.yakhchali at gmail.com>
>> wrote:
>>
>>> yes
>>>
>>> On Tue, Oct 12, 2021 at 2:48 PM Dragos Oancea <dragos at freeswitch.org>
>>> wrote:
>>>
>>>> The custom header name, is it prefixed with "X-" ?
>>>>
>>>> On Tue, Oct 12, 2021 at 1:11 PM Babak Yakhchali <
>>>> babak.yakhchali at gmail.com> wrote:
>>>>
>>>>> Hi
>>>>> I need to pass some custom sip headers between 2 fs boxes after the
>>>>> call is established. To do this I'm using "uuid_setvar sip_h_X-myvar=value"
>>>>> on the channel and calling    uuid_media_reneg to create a re-invite, fs is
>>>>> sending out the re-invite with new custom headers but the other fs box is
>>>>> not adding these new custom sip headers to the channel. How can I make fs
>>>>> to set new custom headers on the receiving side of the re-invite?
>>>>> fs1 ----invite---> fs2
>>>>> .... call is established
>>>>> fs1 <---re-invite--- fs2 (re-invite with new custom sip X headers)
>>>>> fs1 ... not adding new headers to channel
>>>>>
>>>>> 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
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> --
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> 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/20211012/12626b57/attachment.html>

From dujinfang at gmail.com  Wed Oct 13 01:26:50 2021
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 13 Oct 2021 09:26:50 +0800
Subject: [Freeswitch-users] Avoiding DDoS with verto?
In-Reply-To: <CAE23trvy+mH9EPY4W0sND=azF7dibwtHEfY1Yvk8fLcYQjqOxQ@mail.gmail.com>
References: <CAE23trvy+mH9EPY4W0sND=azF7dibwtHEfY1Yvk8fLcYQjqOxQ@mail.gmail.com>
Message-ID: <CAJs+-5pLJ-LfkoGobAq1q+-Qpa_VBZOm5SPG6rCkkMC53pXPKg@mail.gmail.com>

maybe tweak some code so it can log the src ip and fail2ban can catch it.

On Fri, Oct 8, 2021 at 5:11 AM David P <davidswalkabout at gmail.com> wrote:

> After reading about recent DDoS attacks on VOIP providers in
> https://www.rtcsec.com/post/2021/09/massive-ddos-attacks-on-voip-providers-and-simulated-ddos-testing/
> in which Freeswitch is mentioned, I wondered what current practices are for
> services that must serve the public Internet.
>
> For example, a service that is purely verto-based seems like it could
> protect itself in this way:
>
> 1) Block requests on all ports (except the verto WSS login) unless the
> request is from an address that's already part of signaling.
>
> 2) To protect signaling, put it behind AWS API Gateway, which provides
> rate-limiting, and add an authorization check.
>
> Does this seem like it would defend against DDoS? Can FS be configured to
> do #1?
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
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/20211013/d5e78c16/attachment-0001.html>

From babak.yakhchali at gmail.com  Wed Oct 13 12:21:38 2021
From: babak.yakhchali at gmail.com (Babak Yakhchali)
Date: Wed, 13 Oct 2021 15:51:38 +0330
Subject: [Freeswitch-users] Passing custom sip headers using re-invite
In-Reply-To: <CAFGRPVqO-qAS0EAmmkw1r=tyAFE4Hir=bJ0O4EFw+TWJ-AY_fg@mail.gmail.com>
References: <CAD3oNZxacQhmrpHxWSZgBnQ1h9jGP5aRU-GY+ArUXQDodzEX7w@mail.gmail.com>
 <CAGvJV4qiS5swYrWTHYYSqLYr=T_UK7qJmkaUf693vqtxO0-=Mw@mail.gmail.com>
 <CAD3oNZzNfXqW3WMXdu=hsF9EWivX8EBy+9ghT92fo0KCC3gE7g@mail.gmail.com>
 <CAFGRPVojJ=dK+BuaV=RP+8efbrwMkuAGr8-b30CN6WJyqUdbTw@mail.gmail.com>
 <CAD3oNZxPJVqfVf7UAvbfa+ZzL3npmsFfSqGZ=rvQjeiNJjQRFg@mail.gmail.com>
 <CAFGRPVqO-qAS0EAmmkw1r=tyAFE4Hir=bJ0O4EFw+TWJ-AY_fg@mail.gmail.com>
Message-ID: <CAD3oNZxkMxwni0s2t4OUauM4DWdg_zO_q62c8zy-w5WiTuFGkw@mail.gmail.com>

Hi again
sorry for the delay.I need to forward client ip address to an upstream
server,but the problem is client ip address is determined after the call
is  answered , for example when client is a call center agent, in this case
I need to somehow pass the header mid call after an agent  answers the call.
thanks

On Wed, Oct 13, 2021 at 2:13 AM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> again, exactly how are you trying to read the header?
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
>
> On Tue, Oct 12, 2021 at 5:45 PM Babak Yakhchali <babak.yakhchali at gmail.com>
> wrote:
>
>> I have a lua script as an event hook on sofia::reinvite, I want to get
>> the variable in the handler and do some logic on it.
>>
>> On Tue, Oct 12, 2021 at 5:40 PM David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>>
>>> How are you trying to use the header on the other side?
>>>
>>>
>>> On Tue, 12 Oct 2021 at 14:07, Babak Yakhchali <babak.yakhchali at gmail.com>
>>> wrote:
>>>
>>>> yes
>>>>
>>>> On Tue, Oct 12, 2021 at 2:48 PM Dragos Oancea <dragos at freeswitch.org>
>>>> wrote:
>>>>
>>>>> The custom header name, is it prefixed with "X-" ?
>>>>>
>>>>> On Tue, Oct 12, 2021 at 1:11 PM Babak Yakhchali <
>>>>> babak.yakhchali at gmail.com> wrote:
>>>>>
>>>>>> Hi
>>>>>> I need to pass some custom sip headers between 2 fs boxes after the
>>>>>> call is established. To do this I'm using "uuid_setvar sip_h_X-myvar=value"
>>>>>> on the channel and calling    uuid_media_reneg to create a re-invite, fs is
>>>>>> sending out the re-invite with new custom headers but the other fs box is
>>>>>> not adding these new custom sip headers to the channel. How can I make fs
>>>>>> to set new custom headers on the receiving side of the re-invite?
>>>>>> fs1 ----invite---> fs2
>>>>>> .... call is established
>>>>>> fs1 <---re-invite--- fs2 (re-invite with new custom sip X headers)
>>>>>> fs1 ... not adding new headers to channel
>>>>>>
>>>>>> 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
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> --
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.work at gmail.com
>>> phone: +34669448337
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> 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/20211013/dd053373/attachment.html>

From babak.yakhchali at gmail.com  Wed Oct 13 12:23:13 2021
From: babak.yakhchali at gmail.com (Babak Yakhchali)
Date: Wed, 13 Oct 2021 15:53:13 +0330
Subject: [Freeswitch-users] Passing custom sip headers using re-invite
In-Reply-To: <CAD3oNZxkMxwni0s2t4OUauM4DWdg_zO_q62c8zy-w5WiTuFGkw@mail.gmail.com>
References: <CAD3oNZxacQhmrpHxWSZgBnQ1h9jGP5aRU-GY+ArUXQDodzEX7w@mail.gmail.com>
 <CAGvJV4qiS5swYrWTHYYSqLYr=T_UK7qJmkaUf693vqtxO0-=Mw@mail.gmail.com>
 <CAD3oNZzNfXqW3WMXdu=hsF9EWivX8EBy+9ghT92fo0KCC3gE7g@mail.gmail.com>
 <CAFGRPVojJ=dK+BuaV=RP+8efbrwMkuAGr8-b30CN6WJyqUdbTw@mail.gmail.com>
 <CAD3oNZxPJVqfVf7UAvbfa+ZzL3npmsFfSqGZ=rvQjeiNJjQRFg@mail.gmail.com>
 <CAFGRPVqO-qAS0EAmmkw1r=tyAFE4Hir=bJ0O4EFw+TWJ-AY_fg@mail.gmail.com>
 <CAD3oNZxkMxwni0s2t4OUauM4DWdg_zO_q62c8zy-w5WiTuFGkw@mail.gmail.com>
Message-ID: <CAD3oNZzo4e+cM7YcftVKWN4MYzP48ritYYDNZ9581VmAwcvQ7Q@mail.gmail.com>

by answer I mean the call is really bridged to an endpoint, not answered by
IVR or by calling dialplan answer application.

On Wed, Oct 13, 2021 at 3:51 PM Babak Yakhchali <babak.yakhchali at gmail.com>
wrote:

> Hi again
> sorry for the delay.I need to forward client ip address to an upstream
> server,but the problem is client ip address is determined after the call
> is  answered , for example when client is a call center agent, in this case
> I need to somehow pass the header mid call after an agent  answers the call.
> thanks
>
> On Wed, Oct 13, 2021 at 2:13 AM David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> again, exactly how are you trying to read the header?
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>>
>>
>> On Tue, Oct 12, 2021 at 5:45 PM Babak Yakhchali <
>> babak.yakhchali at gmail.com> wrote:
>>
>>> I have a lua script as an event hook on sofia::reinvite, I want to get
>>> the variable in the handler and do some logic on it.
>>>
>>> On Tue, Oct 12, 2021 at 5:40 PM David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>>>
>>>> How are you trying to use the header on the other side?
>>>>
>>>>
>>>> On Tue, 12 Oct 2021 at 14:07, Babak Yakhchali <
>>>> babak.yakhchali at gmail.com> wrote:
>>>>
>>>>> yes
>>>>>
>>>>> On Tue, Oct 12, 2021 at 2:48 PM Dragos Oancea <dragos at freeswitch.org>
>>>>> wrote:
>>>>>
>>>>>> The custom header name, is it prefixed with "X-" ?
>>>>>>
>>>>>> On Tue, Oct 12, 2021 at 1:11 PM Babak Yakhchali <
>>>>>> babak.yakhchali at gmail.com> wrote:
>>>>>>
>>>>>>> Hi
>>>>>>> I need to pass some custom sip headers between 2 fs boxes after the
>>>>>>> call is established. To do this I'm using "uuid_setvar sip_h_X-myvar=value"
>>>>>>> on the channel and calling    uuid_media_reneg to create a re-invite, fs is
>>>>>>> sending out the re-invite with new custom headers but the other fs box is
>>>>>>> not adding these new custom sip headers to the channel. How can I make fs
>>>>>>> to set new custom headers on the receiving side of the re-invite?
>>>>>>> fs1 ----invite---> fs2
>>>>>>> .... call is established
>>>>>>> fs1 <---re-invite--- fs2 (re-invite with new custom sip X headers)
>>>>>>> fs1 ... not adding new headers to channel
>>>>>>>
>>>>>>> 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
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>
>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>> https://signalwire.com
>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>> services.
>>>>>> Build your next product on our scalable cloud platform.
>>>>>>
>>>>>> Join our online community to chat in real time
>>>>>> https://signalwire.community
>>>>>>
>>>>>> Professional FreeSWITCH Services
>>>>>> sales at freeswitch.com
>>>>>> https://freeswitch.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> https://freeswitch.com/oss
>>>>>> https://freeswitch.org/confluence
>>>>>> https://cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> https://freeswitch.com
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>> --
>>>> Regards,
>>>>
>>>> David Villasmil
>>>> email: david.villasmil.work at gmail.com
>>>> phone: +34669448337
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> 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/20211013/db608d6a/attachment-0001.html>

From Antony.Stone at freeswitch.open.source.it  Thu Oct 14 15:54:09 2021
From: Antony.Stone at freeswitch.open.source.it (Antony Stone)
Date: Thu, 14 Oct 2021 17:54:09 +0200
Subject: [Freeswitch-users] Dial with SIP credentials without defining
	gateway?
Message-ID: <202110141754.09432.Antony.Stone@freeswitch.open.source.it>

Hi.

I've recently started using FreeSwitch, after being familiar with Asterisk for 
many years.

I'm wondering whether FreeSwitch has an equivalent to Asterisk's facility for 
dialling out to a SIP server using credentials supplied as part of the dial 
command, instead of needing a pre-defined gateway?

In an Asterisk dialplan, you can say something like

	Dial(SIP/${Destination}:${SIPpass}::${SIPuser}@${SIPhost})

and that will dial to the Destination number on SIPhost using the SIP 
credentials SIPuser/SIPpass.  Nothing needs to be configured in advance.

Can FreeSwitch do something like that too?


Thanks,


Antony.

-- 
"The problem with television is that the people must sit and keep their eyes 
glued on a screen; the average American family hasn't time for it."

 - New York Times, following a demonstration at the 1939 World's Fair.

                                                   Please reply to the list;
                                                         please *don't* CC me.


From kaduww at gmail.com  Fri Oct 15 01:23:03 2021
From: kaduww at gmail.com (Carlos Eduardo)
Date: Thu, 14 Oct 2021 22:23:03 -0300
Subject: [Freeswitch-users] Dial with SIP credentials without defining
	gateway?
In-Reply-To: <202110141754.09432.Antony.Stone@freeswitch.open.source.it>
References: <202110141754.09432.Antony.Stone@freeswitch.open.source.it>
Message-ID: <CAB=EcebdO4N_d=kJaEqTTtDDVUb_yC_MTyo8b-8j+B_68TbktQ@mail.gmail.com>

Hey,

It can, set the variables sip_auth_username and sip_auth_password

bridge
{sip_auth_username=<your_user_name>,sip_auth_password=<your_password>}sofia/external/${
destination_number}@12.34.56.78

Here you have the list of variables you can use
https://freeswitch.org/confluence/display/FREESWITCH/Variables+Master+List

Em qui., 14 de out. de 2021 às 21:32, Antony Stone <
Antony.Stone at freeswitch.open.source.it> escreveu:

> Hi.
>
> I've recently started using FreeSwitch, after being familiar with Asterisk
> for
> many years.
>
> I'm wondering whether FreeSwitch has an equivalent to Asterisk's facility
> for
> dialling out to a SIP server using credentials supplied as part of the
> dial
> command, instead of needing a pre-defined gateway?
>
> In an Asterisk dialplan, you can say something like
>
>         Dial(SIP/${Destination}:${SIPpass}::${SIPuser}@${SIPhost})
>
> and that will dial to the Destination number on SIPhost using the SIP
> credentials SIPuser/SIPpass.  Nothing needs to be configured in advance.
>
> Can FreeSwitch do something like that too?
>
>
> Thanks,
>
>
> Antony.
>
> --
> "The problem with television is that the people must sit and keep their
> eyes
> glued on a screen; the average American family hasn't time for it."
>
>  - New York Times, following a demonstration at the 1939 World's Fair.
>
>                                                    Please reply to the
> list;
>                                                          please *don't* CC
> me.
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
*Carlos E. Wagner*
*Tecnólogo em Telecomunicações, Opensips Certified Professional*

*Fone: +55 48 99981-0894*
*E-mail:* kaduww at gmail.com
*LinkedIn:* https://www.linkedin.com/in/carlos-eduardo-wagner-96bbb433/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211014/83793cd3/attachment.html>

From miguel.andrade at neoscopio.com  Fri Oct 15 15:43:24 2021
From: miguel.andrade at neoscopio.com (Miguel Andrade)
Date: Fri, 15 Oct 2021 16:43:24 +0100
Subject: [Freeswitch-users] Expert help needed debuging sip registration
	[logs included]
Message-ID: <CANyACTDy0q9-Bm-Ty_7GtqGMuqWAz-Md1GPJYQY1JriSbfTZ+A@mail.gmail.com>

Some how I'm able to register sip phones and softphones but not as a
freeswitch gateway. I've tried several combinations to no avail.


My provider instructions are:


· Endereço Registar (Host): sip.sapo.pt

· Endereço Proxy (Outbound Proxy): proxy.sip.sapo.pt

· Porta Proxy: 5070

· Username: +3513020xxxxx (*)

· Password: [de 6 a 8 carateres]

· Transporte Rede (Network Settings): UDP

· SIP URL: +3513020xxxxx (*)


I've entered this as username: +351xxx, password: xxx, proxy:
proxy.sip.sapo.pt:5070 , from domain: sip.sapo.pt

Doesn't work.

I went to advanced and set transport to udp and outbound proxy as
proxy.sip.sapo.pt:5070 didn't work wither.

Tried all combinations I could but also no cigar.


Can anyone help me debug this?


Here are some fs_cli logs:



2021-10-15 15:41:46.615028 [NOTICE] sofia_reg.c:454 Registering
ebf96e38-7592-43b7-82fa-4a9fd4c5facf

nua_stack.c:569 nua_stack_signal() nua(0x7f6730005880): recv signal
r_register

nua_dialog.c:336 nua_dialog_usage_add() nua(0x7f6730005880): adding
register usage

nta.c:2670 nta_tpn_by_url() nta: selecting scheme sip

nta.c:10831 outgoing_query_a() nta: for "proxy.sip.sapo.pt" query "
proxy.sip.sapo.pt" A (cached)

nta.c:10886 outgoing_answer_a() nta: proxy.sip.sapo.pt. IN A 194.65.2.241

tport.c:3274 tport_tsend() tport_tsend(0x7f6724195450) tpn = udp/
194.65.2.241:5070

tport.c:4697 tport_by_addrinfo() tport_by_addrinfo(0x7f6724195450): not
found by name udp/194.65.2.241:5070

tport.c:3609 tport_vsend() tport_vsend(0x7f6724195450): 642 bytes of 642 to
udp/194.65.2.241:5070

nua_stack.c:528 nua_signal() nua(0x7f6730005880): sent signal r_register

send 642 bytes to udp/[194.65.2.241]:5070 at 15:41:46.632220:

------------------------------------------------------------------------

REGISTER sip[image: image.gif]roxy.sip.sapo.pt:5070;transport=udp SIP/2.0

Via: SIP/2.0/UDP 92.222.246.205:5080;rport;branch=z9hG4bKc0QF2jmNeB8Qe

Max-Forwards: 70

From: <sip:+351302020304 at sip.sapo.pt>;tag=HjK0rD25pc27a

To: <sip:+351302020304 at sip.sapo.pt>

Call-ID: 8ab00b1b-ec4d-4564-b148-26fcb82622e1

CSeq: 42600661 REGISTER

Contact: <sip:gw+ebf96e38-7592-43b7-82fa-4a9fd4c5facf at 92.222.246.205:5080
;transport=udp;gw=ebf96e38-7592-43b7-82fa-4a9fd4c5facf>

Expires: 60

User-Agent: Grandstream HT802 1.0.3.2

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY

Supported: timer, path, replaces

Content-Length: 0


nta.c:8312 outgoing_send() nta: sent REGISTER (42600661) to udp/
194.65.2.241:5070

tport.c:4175 tport_pend() tport_pend(0x7f6724195450): pending
0x7f67241145e0 for udp/92.222.246.205:5080 (already 0)

tport.c:2751 tport_wakeup_pri() tport_wakeup_pri(0x7f6724195450): events IN

tport.c:2882 tport_recv_event() tport_recv_event(0x7f6724195450)

tport.c:3219 tport_recv_iovec() tport_recv_iovec(0x7f6724195450) msg
0x7f672416c140 from (udp/92.222.246.205:5080) has 602 bytes, veclen = 1

recv 602 bytes from udp/[194.65.2.241]:5070 at 15:41:46.753181:

------------------------------------------------------------------------

SIP/2.0 401 Unauthorized 11030230330

Via: SIP/2.0/UDP 92.222.246.205:5080
;received=92.222.246.205;branch=z9hG4bKc0QF2jmNeB8Qe;rport=5080

From: <sip:+351302020304 at sip.sapo.pt>;tag=HjK0rD25pc27a

To: <sip:+351302020304 at sip.sapo.pt>;tag=9848f21803d1705ce6531d6ca2f

Call-ID: 8ab00b1b-ec4d-4564-b148-26fcb82622e1

CSeq: 42600661 REGISTER

Content-Length: 0

WWW-Authenticate: Digest nonce="D734DDD02A93696100000000E09BC51B",realm="
ims.telecom.pt",algorithm=MD5,qop="auth"

P-Charging-Vector:
icid-value=3bnhe046p1ds2rk6ehcs2pg6hhoamfde9rpulo4623db2jmt;icid-generated-at=194.65.2.241;orig-ioi=internet


tport.c:3037 tport_deliver() tport_deliver(0x7f6724195450): msg
0x7f672416c140 (602 bytes) from udp/194.65.2.241:5080/sip next=(nil)

nta.c:3303 agent_recv_response() nta: received 401 Unauthorized 11030230330
for REGISTER (42600661)

nta.c:3370 agent_recv_response() nta: 401 Unauthorized 11030230330 is going
to a transaction

nta.c:9584 outgoing_estimate_delay() nta_outgoing: RTT is 121.007 ms

tport.c:4237 tport_release() tport_release(0x7f6724195450): 0x7f67241145e0
by 0x7f672454b980 with 0x7f672416c140

auth_digest.c:105 auth_digest_challenge_get() auth_digest_challenge_get():
got 6

nua_stack.c:271 nua_stack_event() nua(0x7f6730005880): event r_register 401
Unauthorized 11030230330

nua_stack.c:528 nua_signal() nua(0x7f6730005880): sent signal r_authenticate

nua_stack.c:569 nua_stack_signal() nua(0x7f6730005880): recv signal
r_authenticate

auth_digest.c:225 auth_digest_a1() auth_digest_a1() has A1 =
MD5(+351302020304:ims.telecom.pt:1q2w!Q) = 8038cbcec02479ac6c24e22c0d4d343a

auth_digest.c:317 auth_digest_response() A2 = MD5(REGISTER:sip[image:
image.gif]roxy.sip.sapo.pt:5070;transport=udp)

auth_digest.c:339 auth_digest_response() auth_response:
72688ed841494a95193f7d115c409667 = MD5(8038cbcec02479ac6c24e22c0d4d343a[image:
image.gif]
734DDD02A93696100000000E09BC51B:00000001:3UagCahoEjqJ2gIAACyybQ:auth:66f331

b6539f207dc0b673edce10bbbc) (qop=auth)

nta.c:2670 nta_tpn_by_url() nta: selecting scheme sip

nta.c:10831 outgoing_query_a() nta: for "proxy.sip.sapo.pt" query "
proxy.sip.sapo.pt" A (cached)

nta.c:10886 outgoing_answer_a() nta: proxy.sip.sapo.pt. IN A 194.65.2.241

tport.c:3274 tport_tsend() tport_tsend(0x7f6724195450) tpn = udp/
194.65.2.241:5070

tport.c:4697 tport_by_addrinfo() tport_by_addrinfo(0x7f6724195450): not
found by name udp/194.65.2.241:5070

tport.c:3609 tport_vsend() tport_vsend(0x7f6724195450): 920 bytes of 920 to
udp/194.65.2.241:5070

send 920 bytes to udp/[194.65.2.241]:5070 at 15:41:46.753392:

------------------------------------------------------------------------

REGISTER sip[image: image.gif]roxy.sip.sapo.pt:5070;transport=udp SIP/2.0

Via: SIP/2.0/UDP 92.222.246.205:5080;rport;branch=z9hG4bKD9g83D5rBmyaa

Max-Forwards: 70

From: <sip:+351302020304 at sip.sapo.pt>;tag=HjK0rD25pc27a

To: <sip:+351302020304 at sip.sapo.pt>

Call-ID: 8ab00b1b-ec4d-4564-b148-26fcb82622e1

CSeq: 42600662 REGISTER

Contact: <sip:gw+ebf96e38-7592-43b7-82fa-4a9fd4c5facf at 92.222.246.205:5080
;transport=udp;gw=ebf96e38-7592-43b7-82fa-4a9fd4c5facf>

Expires: 60

User-Agent: Grandstream HT802 1.0.3.2

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY

Supported: timer, path, replaces

Authorization: Digest username="+351302020304", realm="ims.telecom.pt",
nonce="D734DDD02A93696100000000E09BC51B", cnonce="3UagCahoEjqJ2gIAACyybQ",
algorithm=MD5, uri="sip[image: image.gif]roxy.sip.sapo.pt:5070;transport=

udp", response="72688ed841494a95193f7d115c409667", qop=auth, nc=00000001

Content-Length: 0


nta.c:8312 outgoing_send() nta: sent REGISTER (42600662) to udp/
194.65.2.241:5070

tport.c:4175 tport_pend() tport_pend(0x7f6724195450): pending
0x7f67242e5a90 for udp/92.222.246.205:5080 (already 0)

tport.c:2751 tport_wakeup_pri() tport_wakeup_pri(0x7f6724195450): events IN

tport.c:2882 tport_recv_event() tport_recv_event(0x7f6724195450)

tport.c:3219 tport_recv_iovec() tport_recv_iovec(0x7f6724195450) msg
0x7f67242d2000 from (udp/92.222.246.205:5080) has 602 bytes, veclen = 1

recv 602 bytes from udp/[194.65.2.241]:5070 at 15:41:46.871252:

------------------------------------------------------------------------

SIP/2.0 401 Unauthorized 1103023032F

Via: SIP/2.0/UDP 92.222.246.205:5080
;received=92.222.246.205;branch=z9hG4bKD9g83D5rBmyaa;rport=5080

From: <sip:+351302020304 at sip.sapo.pt>;tag=HjK0rD25pc27a

To: <sip:+351302020304 at sip.sapo.pt>;tag=9848f2180cea805ce6531d8948d

Call-ID: 8ab00b1b-ec4d-4564-b148-26fcb82622e1

CSeq: 42600662 REGISTER

Content-Length: 0

WWW-Authenticate: Digest nonce="47DFEAEB2A936961000000000782EF64",realm="
ims.telecom.pt",algorithm=MD5,qop="auth"

P-Charging-Vector:
icid-value=3bnhe046p1ds2rk6ehcs2pg6hhoamfde9rpulo462blb2jmt;icid-generated-at=194.65.2.241;orig-ioi=internet


tport.c:3037 tport_deliver() tport_deliver(0x7f6724195450): msg
0x7f67242d2000 (602 bytes) from udp/194.65.2.241:5080/sip next=(nil)

nta.c:3303 agent_recv_response() nta: received 401 Unauthorized 1103023032F
for REGISTER (42600662)

nta.c:3370 agent_recv_response() nta: 401 Unauthorized 1103023032F is going
to a transaction

nta.c:9584 outgoing_estimate_delay() nta_outgoing: RTT is 117.905 ms

tport.c:4237 tport_release() tport_release(0x7f6724195450): 0x7f67242e5a90
by 0x7f672410d3b0 with 0x7f67242d2000

auth_digest.c:105 auth_digest_challenge_get() auth_digest_challenge_get():
got 6

nua_client.c:1223 nua_base_client_check_restart() nua(0x7f6730005880): bad
credentials, clearing them

nua_stack.c:271 nua_stack_event() nua(0x7f6730005880): event r_register 401
Unauthorized 1103023032F

nua_stack.c:528 nua_signal() nua(0x7f6730005880): sent signal r_authenticate

nua_stack.c:569 nua_stack_signal() nua(0x7f6730005880): recv signal
r_authenticate

nua_stack.c:271 nua_stack_event() nua(0x7f6730005880): event r_register 904
Operation has no matching challenge

nua_dialog.c:395 nua_dialog_usage_remove_at() nua(0x7f6730005880): removing
register usage

2021-10-15 15:41:46.854983 [ERR] sofia_reg.c:2470
ebf96e38-7592-43b7-82fa-4a9fd4c5facf Failed Registration with status
Operation has no matching challenge  [904]. failure #1
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211015/4fc27e1d/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image.gif
Type: image/gif
Size: 42 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211015/4fc27e1d/attachment-0005.gif>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image.gif
Type: image/gif
Size: 42 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211015/4fc27e1d/attachment-0006.gif>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image.gif
Type: image/gif
Size: 42 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211015/4fc27e1d/attachment-0007.gif>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image.gif
Type: image/gif
Size: 42 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211015/4fc27e1d/attachment-0008.gif>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image.gif
Type: image/gif
Size: 42 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211015/4fc27e1d/attachment-0009.gif>

From bkk at ednt.de  Tue Oct 19 08:38:54 2021
From: bkk at ednt.de (Bernd Krueger-Knauber)
Date: Tue, 19 Oct 2021 10:38:54 +0200
Subject: [Freeswitch-users] playback results in silence
Message-ID: <933a3da5-c400-833f-44cc-b2d6ea087f5a@ednt.de>

Hi,

we use mod_nibblebill and want to play a wav when the low ballance 
amount is reached.
We noticed alreday that the example in our nibblebill.conf.xml 'play 
ding' is not working.
There is no application play and no file ding.
We use now 'playback misc/sorry.wav' for testing.

We can see in the log that the file is played and we can here a silence 
during the playback of the file,
but not the voice.

If we build a small dialplan entry with 'answer', 'playback 
misc/sorry.wav', 'hangup', we can here the voice.

Any ideas?

Best regards,
Bernd



From Antony.Stone at freeswitch.open.source.it  Wed Oct 20 09:14:35 2021
From: Antony.Stone at freeswitch.open.source.it (Antony Stone)
Date: Wed, 20 Oct 2021 11:14:35 +0200
Subject: [Freeswitch-users] Log dialplan actions?
Message-ID: <202110201114.36329.Antony.Stone@freeswitch.open.source.it>

Hi.

I'm just getting the hang of FreeSwitch (long-time user of Asterisk), and 
wondering:

 - is there a way to get FS to log dialplan actions as they are executed?

Basically, the equivalent of Asterisk's "messages", or even better, "verbose" 
log files.

I'm looking for a way to see how FS is processing calls as they come in, so I 
can be sure my dialplans are doing what I think they should be.

I know I can add "log" actions of my own, but is there something that will 
simply log each dialplan action automatically?


Antony.

-- 
BASIC is to computer languages what Roman numerals are to arithmetic.

                                                   Please reply to the list;
                                                         please *don't* CC me.


From freeswitch at femur.pl  Sun Oct 17 20:44:05 2021
From: freeswitch at femur.pl (=?UTF-8?Q?Pawe=c5=82_Femur_Wojtal?=)
Date: Sun, 17 Oct 2021 22:44:05 +0200
Subject: [Freeswitch-users] Hangup and wait for connection after dialed
	extension is busy.
Message-ID: <07fdf88a-23ae-5ae4-a96b-205ba9eb3c49@femur.pl>


Hello!

Imagine the situation.

I call internal extension. It is busy. So I press *0 and hangup. When 
leg B finishes its connection, freeswitch calls us both and bridges.

How do I get such functionality? I will be grateful for any suggestion.


-- 
fe


From support at telium.io  Mon Oct 18 22:00:51 2021
From: support at telium.io (Telium Technical Support)
Date: Mon, 18 Oct 2021 22:00:51 +0000
Subject: [Freeswitch-users] Recovery capabilities
Message-ID: <0100017c956bef7e-c066e025-cdd8-4e7c-8ee4-94d02bd8ddda-000000@email.amazonses.com>

I've been reading about the recovery capabilities of FreeSwitch.  It SOUNDS
a bit like a work in progress (possibly waiting for funding) - but maybe the
docs are out of date.

 

In terms of current state, can FS:

1.       resume calls in progress (on restart)

2.       place existing calls back into the dialplan where they where

3.       restore queue agents & callers (and their order)

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211018/4206011c/attachment.html>

From avi at avimarcus.net  Wed Oct 20 13:19:47 2021
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 20 Oct 2021 13:19:47 +0000
Subject: [Freeswitch-users] Log dialplan actions?
In-Reply-To: <202110201114.36329.Antony.Stone@freeswitch.open.source.it>
References: <202110201114.36329.Antony.Stone@freeswitch.open.source.it>
Message-ID: <0100017c9ddb9b63-18a1a4f7-cec4-49a3-923b-95b13f886fff-000000@email.amazonses.com>

I think what you want is:

Open fs_cli (with /log 7 if not already enabled)

Also you read afterards in /var/log/freeswitch/freeswitch.log (depending on
your distro/config)

I suggest modifying logfile.conf.xml to add: <param name="uuid"
value="true"/> so that you can do a grep to see just one call in the logs.
(and "reload mod_logfile" to make configuration take effect)

-Avi Marcus
BestFone


On Wed, Oct 20, 2021 at 4:10 PM Antony Stone <
Antony.Stone at freeswitch.open.source.it> wrote:

> Hi.
>
> I'm just getting the hang of FreeSwitch (long-time user of Asterisk), and
> wondering:
>
>  - is there a way to get FS to log dialplan actions as they are executed?
>
> Basically, the equivalent of Asterisk's "messages", or even better,
> "verbose"
> log files.
>
> I'm looking for a way to see how FS is processing calls as they come in,
> so I
> can be sure my dialplans are doing what I think they should be.
>
> I know I can add "log" actions of my own, but is there something that will
> simply log each dialplan action automatically?
>
>
> Antony.
>
> --
> BASIC is to computer languages what Roman numerals are to arithmetic.
>
>                                                    Please reply to the
> list;
>                                                          please *don't* CC
> me.
>
> _________________________________________________________________________
>
> 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/20211020/ddc4e6c7/attachment.html>

From david.villasmil.work at gmail.com  Wed Oct 20 18:41:22 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 20 Oct 2021 19:41:22 +0100
Subject: [Freeswitch-users] Log dialplan actions?
In-Reply-To: <0100017c9ddb9b63-18a1a4f7-cec4-49a3-923b-95b13f886fff-000000@email.amazonses.com>
References: <202110201114.36329.Antony.Stone@freeswitch.open.source.it>
 <0100017c9ddb9b63-18a1a4f7-cec4-49a3-923b-95b13f886fff-000000@email.amazonses.com>
Message-ID: <CAFGRPVp7pX9B=KVXNQ3GnziVQbSfN7TQ_ADj-4B0hxZZ0gsnsw@mail.gmail.com>

if you enable xml_cdr, you will get an "app_log" array containing
everything.
Regards,

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


On Wed, Oct 20, 2021 at 3:09 PM Avi Marcus <avi at avimarcus.net> wrote:

> I think what you want is:
>
> Open fs_cli (with /log 7 if not already enabled)
>
> Also you read afterards in /var/log/freeswitch/freeswitch.log (depending
> on your distro/config)
>
> I suggest modifying logfile.conf.xml to add: <param name="uuid"
> value="true"/> so that you can do a grep to see just one call in the logs.
> (and "reload mod_logfile" to make configuration take effect)
>
> -Avi Marcus
> BestFone
>
>
> On Wed, Oct 20, 2021 at 4:10 PM Antony Stone <
> Antony.Stone at freeswitch.open.source.it> wrote:
>
>> Hi.
>>
>> I'm just getting the hang of FreeSwitch (long-time user of Asterisk), and
>> wondering:
>>
>>  - is there a way to get FS to log dialplan actions as they are executed?
>>
>> Basically, the equivalent of Asterisk's "messages", or even better,
>> "verbose"
>> log files.
>>
>> I'm looking for a way to see how FS is processing calls as they come in,
>> so I
>> can be sure my dialplans are doing what I think they should be.
>>
>> I know I can add "log" actions of my own, but is there something that
>> will
>> simply log each dialplan action automatically?
>>
>>
>> Antony.
>>
>> --
>> BASIC is to computer languages what Roman numerals are to arithmetic.
>>
>>                                                    Please reply to the
>> list;
>>                                                          please *don't*
>> CC me.
>>
>> _________________________________________________________________________
>>
>> 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/20211020/441d46da/attachment-0001.html>

From avi at avimarcus.net  Wed Oct 20 18:51:50 2021
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 20 Oct 2021 18:51:50 +0000
Subject: [Freeswitch-users] Log dialplan actions?
In-Reply-To: <CAFGRPVp7pX9B=KVXNQ3GnziVQbSfN7TQ_ADj-4B0hxZZ0gsnsw@mail.gmail.com>
References: <202110201114.36329.Antony.Stone@freeswitch.open.source.it>
 <0100017c9ddb9b63-18a1a4f7-cec4-49a3-923b-95b13f886fff-000000@email.amazonses.com>
 <CAFGRPVp7pX9B=KVXNQ3GnziVQbSfN7TQ_ADj-4B0hxZZ0gsnsw@mail.gmail.com>
Message-ID: <0100017c9f0b9c7c-9c7f455f-7177-4329-a198-72c7b6744b7b-000000@email.amazonses.com>

Oh op didn't ask for this, but the full logs will show the dialplan
evaluation and matching, not just what ended up being executed.

Staring at those logs is how I experienced and understood how the dial plan
worked.

On Wed, Oct 20, 2021, 9:42 PM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> if you enable xml_cdr, you will get an "app_log" array containing
> everything.
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
>
> On Wed, Oct 20, 2021 at 3:09 PM Avi Marcus <avi at avimarcus.net> wrote:
>
>> I think what you want is:
>>
>> Open fs_cli (with /log 7 if not already enabled)
>>
>> Also you read afterards in /var/log/freeswitch/freeswitch.log (depending
>> on your distro/config)
>>
>> I suggest modifying logfile.conf.xml to add: <param name="uuid"
>> value="true"/> so that you can do a grep to see just one call in the logs.
>> (and "reload mod_logfile" to make configuration take effect)
>>
>> -Avi Marcus
>> BestFone
>>
>>
>> On Wed, Oct 20, 2021 at 4:10 PM Antony Stone <
>> Antony.Stone at freeswitch.open.source.it> wrote:
>>
>>> Hi.
>>>
>>> I'm just getting the hang of FreeSwitch (long-time user of Asterisk),
>>> and
>>> wondering:
>>>
>>>  - is there a way to get FS to log dialplan actions as they are executed?
>>>
>>> Basically, the equivalent of Asterisk's "messages", or even better,
>>> "verbose"
>>> log files.
>>>
>>> I'm looking for a way to see how FS is processing calls as they come in,
>>> so I
>>> can be sure my dialplans are doing what I think they should be.
>>>
>>> I know I can add "log" actions of my own, but is there something that
>>> will
>>> simply log each dialplan action automatically?
>>>
>>>
>>> Antony.
>>>
>>> --
>>> BASIC is to computer languages what Roman numerals are to arithmetic.
>>>
>>>                                                    Please reply to the
>>> list;
>>>                                                          please *don't*
>>> CC me.
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211020/cdc16ec3/attachment.html>

From david.villasmil.work at gmail.com  Wed Oct 20 20:10:26 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 20 Oct 2021 21:10:26 +0100
Subject: [Freeswitch-users] Log dialplan actions?
In-Reply-To: <0100017c9f0b9c7c-9c7f455f-7177-4329-a198-72c7b6744b7b-000000@email.amazonses.com>
References: <202110201114.36329.Antony.Stone@freeswitch.open.source.it>
 <0100017c9ddb9b63-18a1a4f7-cec4-49a3-923b-95b13f886fff-000000@email.amazonses.com>
 <CAFGRPVp7pX9B=KVXNQ3GnziVQbSfN7TQ_ADj-4B0hxZZ0gsnsw@mail.gmail.com>
 <0100017c9f0b9c7c-9c7f455f-7177-4329-a198-72c7b6744b7b-000000@email.amazonses.com>
Message-ID: <CAFGRPVp2cedxcATdZE+08Ss2yk3PKbbNt=LfEtmoQXMe-D82tA@mail.gmail.com>

👍 true
Regards,

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


On Wed, Oct 20, 2021 at 8:35 PM Avi Marcus <avi at avimarcus.net> wrote:

> Oh op didn't ask for this, but the full logs will show the dialplan
> evaluation and matching, not just what ended up being executed.
>
> Staring at those logs is how I experienced and understood how the dial
> plan worked.
>
> On Wed, Oct 20, 2021, 9:42 PM David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> if you enable xml_cdr, you will get an "app_log" array containing
>> everything.
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>>
>>
>> On Wed, Oct 20, 2021 at 3:09 PM Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> I think what you want is:
>>>
>>> Open fs_cli (with /log 7 if not already enabled)
>>>
>>> Also you read afterards in /var/log/freeswitch/freeswitch.log (depending
>>> on your distro/config)
>>>
>>> I suggest modifying logfile.conf.xml to add: <param name="uuid"
>>> value="true"/> so that you can do a grep to see just one call in the logs.
>>> (and "reload mod_logfile" to make configuration take effect)
>>>
>>> -Avi Marcus
>>> BestFone
>>>
>>>
>>> On Wed, Oct 20, 2021 at 4:10 PM Antony Stone <
>>> Antony.Stone at freeswitch.open.source.it> wrote:
>>>
>>>> Hi.
>>>>
>>>> I'm just getting the hang of FreeSwitch (long-time user of Asterisk),
>>>> and
>>>> wondering:
>>>>
>>>>  - is there a way to get FS to log dialplan actions as they are
>>>> executed?
>>>>
>>>> Basically, the equivalent of Asterisk's "messages", or even better,
>>>> "verbose"
>>>> log files.
>>>>
>>>> I'm looking for a way to see how FS is processing calls as they come
>>>> in, so I
>>>> can be sure my dialplans are doing what I think they should be.
>>>>
>>>> I know I can add "log" actions of my own, but is there something that
>>>> will
>>>> simply log each dialplan action automatically?
>>>>
>>>>
>>>> Antony.
>>>>
>>>> --
>>>> BASIC is to computer languages what Roman numerals are to arithmetic.
>>>>
>>>>                                                    Please reply to the
>>>> list;
>>>>                                                          please *don't*
>>>> CC me.
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211020/f3b15aaf/attachment-0001.html>

From mayamatakeshi at gmail.com  Thu Oct 21 01:03:52 2021
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 21 Oct 2021 10:03:52 +0900
Subject: [Freeswitch-users] Hangup and wait for connection after dialed
 extension is busy.
In-Reply-To: <07fdf88a-23ae-5ae4-a96b-205ba9eb3c49@femur.pl>
References: <07fdf88a-23ae-5ae4-a96b-205ba9eb3c49@femur.pl>
Message-ID: <CABaNFCY+T2ChQ58_mJHgGO8Nbx5wDJX5AFZcXU5+t7nhZqzZKA@mail.gmail.com>

On Wed, Oct 20, 2021 at 11:14 PM Paweł Femur Wojtal <freeswitch at femur.pl>
wrote:

>
> Hello!
>
> Imagine the situation.
>
> I call internal extension. It is busy. So I press *0 and hangup. When
> leg B finishes its connection, freeswitch calls us both and bridges.
>
> How do I get such functionality? I will be grateful for any suggestion.
>

Maybe someone knows an easier way but if I needed something like this, I
would try:
  - send the call to a lua script after the bridge fails due to BUSY
(channel variable last_bridge_hangup_cause==USER_BUSY)
  - ask the user to confirm if he/she wants to get a callback
  - if yes, find the UUID of the extension call. This could be obtained
from channel variables (probably originate_signal_bond or originated_legs)
if not, I would use Freeswitch.dbh (
https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference) to
search for the extension that is busy.
  - set a hangup hook to be execute when the extension call terminates (
https://freeswitch.org/confluence/display/FREESWITCH/api_hangup_hook)
  - the hook would be a lua script that would make the callback to you and
after answer try to call the extension.

obs:
  - if the hangup hook fails to be set, it could be because the call ended
while we were doing the processing so I would execute the callback right
away).
  - in the callback, before calling the extension I would ask for
confirmation again as by that time you might not want to call the extension
anymore
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211021/de844abd/attachment.html>

From mayamatakeshi at gmail.com  Thu Oct 21 10:12:26 2021
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 21 Oct 2021 19:12:26 +0900
Subject: [Freeswitch-users] Hangup and wait for connection after dialed
 extension is busy.
In-Reply-To: <CABaNFCY+T2ChQ58_mJHgGO8Nbx5wDJX5AFZcXU5+t7nhZqzZKA@mail.gmail.com>
References: <07fdf88a-23ae-5ae4-a96b-205ba9eb3c49@femur.pl>
 <CABaNFCY+T2ChQ58_mJHgGO8Nbx5wDJX5AFZcXU5+t7nhZqzZKA@mail.gmail.com>
Message-ID: <CABaNFCZQ3-LdFQAY82orY7SWWPJ70u4ZH293u2f730Vrm-XQdw@mail.gmail.com>

On Thu, Oct 21, 2021 at 10:03 AM mayamatakeshi <mayamatakeshi at gmail.com>
wrote:

>
>
> On Wed, Oct 20, 2021 at 11:14 PM Paweł Femur Wojtal <freeswitch at femur.pl>
> wrote:
>
>>
>> Hello!
>>
>> Imagine the situation.
>>
>> I call internal extension. It is busy. So I press *0 and hangup. When
>> leg B finishes its connection, freeswitch calls us both and bridges.
>>
>> How do I get such functionality? I will be grateful for any suggestion.
>>
>
> Maybe someone knows an easier way but if I needed something like this, I
> would try:
>   - send the call to a lua script after the bridge fails due to BUSY
> (channel variable last_bridge_hangup_cause==USER_BUSY)
>   - ask the user to confirm if he/she wants to get a callback
>   - if yes, find the UUID of the extension call. This could be obtained
> from channel variables (probably originate_signal_bond or originated_legs)
> if not, I would use Freeswitch.dbh (
> https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference)
> to search for the extension that is busy.
>

Sorry, actually, you cannot use the channel variables to find the UUID as
they would be for the Leg B you just attempted. You need to find the UUID
of the active call that is making the extension B busy.
So you need to use freeswitch.dbh.


>   - set a hangup hook to be execute when the extension call terminates (
> https://freeswitch.org/confluence/display/FREESWITCH/api_hangup_hook)
>   - the hook would be a lua script that would make the callback to you and
> after answer try to call the extension.
>
> obs:
>   - if the hangup hook fails to be set, it could be because the call ended
> while we were doing the processing so I would execute the callback right
> away).
>   - in the callback, before calling the extension I would ask for
> confirmation again as by that time you might not want to call the extension
> anymore
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211021/cae7fdeb/attachment.html>

From jhunter at voxboxcoms.co.uk  Thu Oct 21 13:10:05 2021
From: jhunter at voxboxcoms.co.uk (Jonathan Hunter)
Date: Thu, 21 Oct 2021 14:10:05 +0100
Subject: [Freeswitch-users] UDP/DTLS/SCTP webrtc-datachannel Freeswitch
	1.10.6 and sip.js
Message-ID: <CAPQovyQc218=DFjCGgyGrwSKqp0nHHikqyDAeRrW3rFO9uCF3w@mail.gmail.com>

Hi guys,

Just wondered if anyone had suggestions around this or had this issue
before, we have 2 web clients using the sip.js library and we are trying to
use the data channel to send messages to one another.

If I bypass media and send media peer to peer it works fine, however if I
try and send an invite with the data channel defined;
m=application 52004 UDP/DTLS/SCTP webrtc-datachannel

Freeswitch seems to consume this when bridging to the target end
point/user. Does freeswitch support SCTP/ webrtc-datachannel if I wanted to
present it from leg A to leg B so the media can flow via freeswitch?

This is using debian:buster container.

Thanks

Jon
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211021/0dfbcdd9/attachment.html>

From gmaruzz at gmail.com  Thu Oct 21 13:31:08 2021
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 21 Oct 2021 15:31:08 +0200
Subject: [Freeswitch-users] UDP/DTLS/SCTP webrtc-datachannel Freeswitch
 1.10.6 and sip.js
In-Reply-To: <CAPQovyQc218=DFjCGgyGrwSKqp0nHHikqyDAeRrW3rFO9uCF3w@mail.gmail.com>
References: <CAPQovyQc218=DFjCGgyGrwSKqp0nHHikqyDAeRrW3rFO9uCF3w@mail.gmail.com>
Message-ID: <CALXCt0qoLK3LHtXNCrRvECt5y528fm+PG8R7yKUiFymsfEFEPw@mail.gmail.com>

On Thu, Oct 21, 2021 at 3:10 PM Jonathan Hunter <jhunter at voxboxcoms.co.uk>
wrote:

> Hi guys,
>
> Just wondered if anyone had suggestions around this or had this issue
> before, we have 2 web clients using the sip.js library and we are trying to
> use the data channel to send messages to one another.
>
>
Why don't you use the MESSAGE sip method (supported by both freeswitch and
sip.js)?

You can even process the MESSAGE into freeswitch's chatplan, if needed.

-giovanni





> If I bypass media and send media peer to peer it works fine, however if I
> try and send an invite with the data channel defined;
> m=application 52004 UDP/DTLS/SCTP webrtc-datachannel
>
> Freeswitch seems to consume this when bridging to the target end
> point/user. Does freeswitch support SCTP/ webrtc-datachannel if I wanted to
> present it from leg A to leg B so the media can flow via freeswitch?
>
> This is using debian:buster container.
>
> Thanks
>
> Jon
>

-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211021/9d404037/attachment.html>

From chrisbware at yahoo.it  Thu Oct 21 13:35:59 2021
From: chrisbware at yahoo.it (Chris B. Ware)
Date: Thu, 21 Oct 2021 13:35:59 +0000 (UTC)
Subject: [Freeswitch-users] RE: Recovery capabilities
References: <2069632511.7841425.1634823359563.ref@mail.yahoo.com>
Message-ID: <2069632511.7841425.1634823359563@mail.yahoo.com>

Hi,
I'm playing with, so called, FS High Availability recently. I've had the same feeling of a work in progress features.It works on resuming calls (just few seconds of audio missing during the switchover) and probably keeps the status of the call, also respect the dialplan.
It is basically thought to handle two FS box with keepalived managing a shared IP, so if you need to shutdown one of the FS nodes ( I need it in my project)it doesn't work, since the dying FS will drop all the calls, cleaning Recovery table on DB.
In 2014 there was a project for a module called mod_ha_cluster, I think is a dead project.
Anyway this is a feature  that would deserve more attention, indeed.


KR,Chris


------------------------------
Message: 3Date: Mon, 18 Oct 2021 22:00:51 +0000From: "Telium Technical Support" <support at telium.io>To: <freeswitch-users at lists.freeswitch.org>Subject: [Freeswitch-users] Recovery capabilitiesMessage-ID:    <0100017c956bef7e-c066e025-cdd8-4e7c-8ee4-94d02bd8ddda-000000 at email.amazonses.com>   Content-Type: text/plain; charset="utf-8"
I've been reading about the recovery capabilities of FreeSwitch.  It SOUNDSa bit like a work in progress (possibly waiting for funding) - but maybe thedocs are out of date.


In terms of current state, can FS:
1.      resume calls in progress (on restart)
2.      place existing calls back into the dialplan where they where
3.      restore queue agents & callers (and their order)
-------------- next part --------------An HTML attachment was scrubbed...URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211018/4206011c/attachment-0001.html>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211021/4f84bee6/attachment.html>

From kaiduanx at yahoo.ca  Thu Oct 21 13:56:29 2021
From: kaiduanx at yahoo.ca (kaiduan xie)
Date: Thu, 21 Oct 2021 13:56:29 +0000 (UTC)
Subject: [Freeswitch-users] UDP/DTLS/SCTP webrtc-datachannel Freeswitch
 1.10.6 and sip.js
In-Reply-To: <CALXCt0qoLK3LHtXNCrRvECt5y528fm+PG8R7yKUiFymsfEFEPw@mail.gmail.com>
References: <CAPQovyQc218=DFjCGgyGrwSKqp0nHHikqyDAeRrW3rFO9uCF3w@mail.gmail.com>
 <CALXCt0qoLK3LHtXNCrRvECt5y528fm+PG8R7yKUiFymsfEFEPw@mail.gmail.com>
Message-ID: <1662564254.2821806.1634824589967@mail.yahoo.com>

 Freeswitch does not support webrtc SCTP data channel.
/Kaiduan
    On Thursday, October 21, 2021, 09:32:19 a.m. EDT, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:  
 
 On Thu, Oct 21, 2021 at 3:10 PM Jonathan Hunter <jhunter at voxboxcoms.co.uk> wrote:

Hi guys,
Just wondered if anyone had suggestions around this or had this issue before, we have 2 web clients using the sip.js library and we are trying to use the data channel to send messages to one another.


Why don't you use the MESSAGE sip method (supported by both freeswitch and sip.js)?
You can even process the MESSAGE into freeswitch's chatplan, if needed.
-giovanni


 
If I bypass media and send media peer to peer it works fine, however if I try and send an invite with the data channel defined;m=application 52004 UDP/DTLS/SCTP webrtc-datachannel
Freeswitch seems to consume this when bridging to the target end point/user. Does freeswitch support SCTP/ webrtc-datachannel if I wanted to present it from leg A to leg B so the media can flow via freeswitch?
This is using debian:buster container.
Thanks
Jon


-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18

_________________________________________________________________________

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/20211021/0a316480/attachment-0001.html>

From jhunter at voxboxcoms.co.uk  Thu Oct 21 14:02:25 2021
From: jhunter at voxboxcoms.co.uk (Jonathan Hunter)
Date: Thu, 21 Oct 2021 15:02:25 +0100
Subject: [Freeswitch-users] UDP/DTLS/SCTP webrtc-datachannel Freeswitch
 1.10.6 and sip.js
In-Reply-To: <CALXCt0qoLK3LHtXNCrRvECt5y528fm+PG8R7yKUiFymsfEFEPw@mail.gmail.com>
References: <CAPQovyQc218=DFjCGgyGrwSKqp0nHHikqyDAeRrW3rFO9uCF3w@mail.gmail.com>
 <CALXCt0qoLK3LHtXNCrRvECt5y528fm+PG8R7yKUiFymsfEFEPw@mail.gmail.com>
Message-ID: <CAPQovyRA1HkXsZo1YU-s_W9yhxTbjYdg3eKz9zh1wUQGouMK7A@mail.gmail.com>

Hi Giovanni,

Thank you for the reply, yes good point I saw you implement that in the
cookbook some years ago :)

I was just interested in using the data channel but you are right I could
just use that for chat.

Thanks

Jon

On Thu, Oct 21, 2021 at 2:44 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> On Thu, Oct 21, 2021 at 3:10 PM Jonathan Hunter <jhunter at voxboxcoms.co.uk>
> wrote:
>
>> Hi guys,
>>
>> Just wondered if anyone had suggestions around this or had this issue
>> before, we have 2 web clients using the sip.js library and we are trying to
>> use the data channel to send messages to one another.
>>
>>
> Why don't you use the MESSAGE sip method (supported by both freeswitch and
> sip.js)?
>
> You can even process the MESSAGE into freeswitch's chatplan, if needed.
>
> -giovanni
>
>
>
>
>
>> If I bypass media and send media peer to peer it works fine, however if I
>> try and send an invite with the data channel defined;
>> m=application 52004 UDP/DTLS/SCTP webrtc-datachannel
>>
>> Freeswitch seems to consume this when bridging to the target end
>> point/user. Does freeswitch support SCTP/ webrtc-datachannel if I wanted to
>> present it from leg A to leg B so the media can flow via freeswitch?
>>
>> This is using debian:buster container.
>>
>> Thanks
>>
>> Jon
>>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
>
> 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



-- 
Jonathan Hunter
Principle Consultant
M:(+44) 7917 190 438
Email:jhunter at voxboxcoms.co.uk
www.voxboxcoms.co.uk/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211021/318a69fa/attachment.html>

From jhunter at voxboxcoms.co.uk  Thu Oct 21 14:07:54 2021
From: jhunter at voxboxcoms.co.uk (Jonathan Hunter)
Date: Thu, 21 Oct 2021 15:07:54 +0100
Subject: [Freeswitch-users] UDP/DTLS/SCTP webrtc-datachannel Freeswitch
 1.10.6 and sip.js
In-Reply-To: <CAPQovyRA1HkXsZo1YU-s_W9yhxTbjYdg3eKz9zh1wUQGouMK7A@mail.gmail.com>
References: <CAPQovyQc218=DFjCGgyGrwSKqp0nHHikqyDAeRrW3rFO9uCF3w@mail.gmail.com>
 <CALXCt0qoLK3LHtXNCrRvECt5y528fm+PG8R7yKUiFymsfEFEPw@mail.gmail.com>
 <CAPQovyRA1HkXsZo1YU-s_W9yhxTbjYdg3eKz9zh1wUQGouMK7A@mail.gmail.com>
Message-ID: <CAPQovySKjW7fX050C2-Fnp49cWV+NKAFD+9kwkmAQwDkLPe87g@mail.gmail.com>

For background, I was looking to send status events like isMuted via the
data channel for client status, I assume same can be achieved using MESSAGE
between client applications?

Jon

On Thu, Oct 21, 2021 at 3:02 PM Jonathan Hunter <jhunter at voxboxcoms.co.uk>
wrote:

> Hi Giovanni,
>
> Thank you for the reply, yes good point I saw you implement that in the
> cookbook some years ago :)
>
> I was just interested in using the data channel but you are right I could
> just use that for chat.
>
> Thanks
>
> Jon
>
> On Thu, Oct 21, 2021 at 2:44 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> On Thu, Oct 21, 2021 at 3:10 PM Jonathan Hunter <jhunter at voxboxcoms.co.uk>
>> wrote:
>>
>>> Hi guys,
>>>
>>> Just wondered if anyone had suggestions around this or had this issue
>>> before, we have 2 web clients using the sip.js library and we are trying to
>>> use the data channel to send messages to one another.
>>>
>>>
>> Why don't you use the MESSAGE sip method (supported by both freeswitch
>> and sip.js)?
>>
>> You can even process the MESSAGE into freeswitch's chatplan, if needed.
>>
>> -giovanni
>>
>>
>>
>>
>>
>>> If I bypass media and send media peer to peer it works fine, however if
>>> I try and send an invite with the data channel defined;
>>> m=application 52004 UDP/DTLS/SCTP webrtc-datachannel
>>>
>>> Freeswitch seems to consume this when bridging to the target end
>>> point/user. Does freeswitch support SCTP/ webrtc-datachannel if I wanted to
>>> present it from leg A to leg B so the media can flow via freeswitch?
>>>
>>> This is using debian:buster container.
>>>
>>> Thanks
>>>
>>> Jon
>>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
>> _________________________________________________________________________
>>
>> 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
>
>
>
> --
> Jonathan Hunter
> Principle Consultant
> M:(+44) 7917 190 438
> Email:jhunter at voxboxcoms.co.uk
> www.voxboxcoms.co.uk/
>
>

-- 
Jonathan Hunter
Principle Consultant
M:(+44) 7917 190 438
Email:jhunter at voxboxcoms.co.uk
www.voxboxcoms.co.uk/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211021/c964ad53/attachment.html>

From lists at telefaks.de  Thu Oct 21 15:20:20 2021
From: lists at telefaks.de (Peter Steinbach)
Date: Thu, 21 Oct 2021 17:20:20 +0200
Subject: [Freeswitch-users] Issue with Enterprise originate and 487
 "ORIGINATOR_CANCEL".
In-Reply-To: <CAPfjz3hJNH4rhbeMb3LkboN2NC3VQoW890cVr8i4UJbdRTjgkg@mail.gmail.com>
References: <CAPfjz3hJNH4rhbeMb3LkboN2NC3VQoW890cVr8i4UJbdRTjgkg@mail.gmail.com>
Message-ID: <26f9382e-1c6c-6da0-0b49-3db8633668c2@telefaks.de>

Hello,


this is a long lasting issue I firstly posted in 2017 and lastly saw in
2019 in this thread, and I am happy to announce, that we have found a
solution for this.

Here is the patch, which works pretty well in our environment

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

--- freeswitch_orig/src/switch_ivr_originate.c    2021-10-20
12:51:04.863328041 +0200
+++ freeswitch/src/switch_ivr_originate.c    2021-10-20
12:54:15.764616280 +0200
@@ -1789,7 +1789,13 @@
         if (cancel_cause && *cancel_cause > 0) {
             handles[i].cancel_cause = *cancel_cause;
         } else {
-            handles[i].cancel_cause = SWITCH_CAUSE_LOSE_RACE;
+            if ((hp!=NULL) && (hp->cause==SWITCH_CAUSE_SUCCESS)) {
//Was this call taken by another destination?
+                //Yes, the race was lost
+                handles[i].cancel_cause = SWITCH_CAUSE_LOSE_RACE;
+            } else {
+                //No, something else happened, probably Originator
Cancel   
+                handles[i].cancel_cause = SWITCH_CAUSE_ORIGINATOR_CANCEL;
+            }
         }
     }
===================================================================================================


Can someone retest this and check this in?


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de

Am 22.05.19 um 21:14 schrieb Jai Rangi:
> All,
> Running into a small issue with group calling, when the original
> caller hang up, phones dont show that call as missed call, cause of
> wrong reason code from freeswitch.
> Scenario:
>
>   * Caller -> FS (Groupcall) to 101 and 102
>   * Phones ring
>   * Caller hangup
>   * Phones should see missed calls,
>   * But free-switch send cancel with Reason: SIP;cause=200;text="Call
>     completed elsewhere", where is should send Reason:
>     SIP;cause=487;text="ORIGINATOR_CANCEL".
>
> Here is what my bridge data.
>
> bridge(<all_threads=true,absolute_codec_string=PCMU,PCMA,G729,group_confirm_file=playback
> /usr/local/freeswitch/sound/ivr/1443458342.wav,group_confirm_key=exec>{sip_cid_type=rpid,ignore_early_media=true,originate_timeout=30}[leg_timeout=20
> ]user/101 at 192.168.5.50:_:[ignore_early_media=true,leg_timeout=20
> ]user/102 at 192.168.5.50 <mailto:102 at 192.168.5.50>)
>
> If I send
> bridge(<all_threads=true,absolute_codec_string=PCMU,PCMA,G729,group_confirm_file=playback
> /usr/local/freeswitch/sound/ivr/1443458342.wav,group_confirm_key=exec>{sip_cid_type=rpid,ignore_early_media=true,originate_timeout=30}[leg_timeout=20
> ]user/101 at 192.168.5.50 <mailto:101 at 192.168.5.50>
>
> Everything works fine.
> I see some old posts but no resolution.
> Freeswitch version on Debian 8.9
> FreeSWITCH (Version 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit)
>
> Any idea or suggestion for workaround? 
>
> Jai
>
>
>
>
>
>
>
> _________________________________________________________________________
>
> 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/20211021/b72cb14d/attachment-0001.html>

From support at naud.io  Thu Oct 21 17:02:38 2021
From: support at naud.io (Support from NetworkedAudio LLC)
Date: Thu, 21 Oct 2021 17:02:38 +0000
Subject: [Freeswitch-users] Sending announcements to all callers
Message-ID: <SN4PR0701MB382296A8385542D2322A36A5A3BF9@SN4PR0701MB3822.namprd07.prod.outlook.com>

I have a system where a user dials in, get authenticated and placed into a standard conference. Other local resources get added to the conferences as needed, and that is working well.

For safety reasons, we need to play a single incoming PortAudio channel to all dialed in users.

If I do an auto call or originate I can add a deaf member to each conference which is a loopback to an eavesdrop. This is messy but works.

However, it uses a lot of resources and now my local people have the announcements twice, with delay.

What I want is just to send audio out to those dialing in.

I have tried sending eavesdropping just to the incoming call but it won’t let me then join a conference through the dialplan - processing stops at the eavesdrop.

I see broadcast but can only send tones or saved files.

Ideally the incoming external callers would hear the announcements and not the conferences themselves. I’d like to avoid loopback, too.

Does anyone have ideas, please?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211021/761ea351/attachment.html>

From Antony.Stone at freeswitch.open.source.it  Wed Oct 20 19:39:39 2021
From: Antony.Stone at freeswitch.open.source.it (Antony Stone)
Date: Wed, 20 Oct 2021 21:39:39 +0200
Subject: [Freeswitch-users] Log dialplan actions?
In-Reply-To: <CAFGRPVp7pX9B=KVXNQ3GnziVQbSfN7TQ_ADj-4B0hxZZ0gsnsw@mail.gmail.com>
References: <202110201114.36329.Antony.Stone@freeswitch.open.source.it>
 <0100017c9ddb9b63-18a1a4f7-cec4-49a3-923b-95b13f886fff-000000@email.amazonses.com>
 <CAFGRPVp7pX9B=KVXNQ3GnziVQbSfN7TQ_ADj-4B0hxZZ0gsnsw@mail.gmail.com>
Message-ID: <202110202139.39864.Antony.Stone@freeswitch.open.source.it>

On Wednesday 20 October 2021 at 20:41:22, David Villasmil wrote:

> if you enable xml_cdr, you will get an "app_log" array containing
> everything.

Really?  xml_cdr sounds (and looks from its config file) to me like it records 
CDRs, which are a summary of call legs, durations, billing, etc.

I'm looking for something which tells me, when a call comes in and gets 
processed, which parts of the dialplan are doing the processing, so I can 
correct my errors by spotting where it goes wrong.

Asterisk logs will show me each line of each context which a call gets 
processed through, so I can tell "ah!  that should not go there - it should go 
here instead".

What's the equivalent for FreeSwitch - or alternatively, how does anyone else 
debug their dialplans?

> On Wed, Oct 20, 2021 at 3:09 PM Avi Marcus <avi at avimarcus.net> wrote:
> > I think what you want is:
> > 
> > Open fs_cli (with /log 7 if not already enabled)
> > 
> > Also you read afterards in /var/log/freeswitch/freeswitch.log (depending
> > on your distro/config)

That seems to be *extraordinarily* verbose - it's telling me *everything* 
FreeSwitch is thinking about.

I'm just trying to follow the processing of a call through the dialplan.

> > I suggest modifying logfile.conf.xml to add: <param name="uuid"
> > value="true"/> so that you can do a grep to see just one call in the
> > logs. (and "reload mod_logfile" to make configuration take effect)
> > 
> > -Avi Marcus
> > BestFone
> > 
> > 
> > On Wed, Oct 20, 2021 at 4:10 PM Antony Stone wrote:
> >> Hi.
> >> 
> >> I'm just getting the hang of FreeSwitch (long-time user of Asterisk),
> >> and wondering:
> >>
> >>  - is there a way to get FS to log dialplan actions as they are
> >>  executed?
> >> 
> >> Basically, the equivalent of Asterisk's "messages", or even better,
> >> "verbose" log files.
> >> 
> >> I'm looking for a way to see how FS is processing calls as they come in,
> >> so I can be sure my dialplans are doing what I think they should be.
> >> 
> >> I know I can add "log" actions of my own, but is there something that
> >> will simply log each dialplan action automatically?
> >> 
> >> 
> >> Antony.

-- 
It is also possible that putting the birds in a laboratory setting 
inadvertently renders them relatively incompetent.

 - Daniel C Dennett

                                                   Please reply to the list;
                                                         please *don't* CC me.


From Antony.Stone at freeswitch.open.source.it  Thu Oct 21 13:12:02 2021
From: Antony.Stone at freeswitch.open.source.it (Antony Stone)
Date: Thu, 21 Oct 2021 15:12:02 +0200
Subject: [Freeswitch-users] Database docs?
Message-ID: <202110211512.02247.Antony.Stone@freeswitch.open.source.it>

Hi.

https://freeswitch.com/index.php/2019/08/20/freeswitch-v1-10-1-release/ tells 
me that there's a 2-year old module called mod_mariadb, but I cannot find 
anywhere on https://freeswitch.org/confluence/ the documentation about how to 
use it.

Can someone kindly point me in the right direction?

Also, just in case there's a simple answer, why isn't mod_odbc_query built 
into the Debian release packages available at 
https://files.freeswitch.org/repo/deb/debian-release/ ?

I'm running FS 1.10.6


Thanks,


Antony.

-- 
"Hi, I've found a fault with the English language and I need an entomologist."
"I think you mean an etymologist."
"No.  It's a bug, not a feature."

                                                   Please reply to the list;
                                                         please *don't* CC me.


From covici at ccs.covici.com  Fri Oct 22 04:27:41 2021
From: covici at ccs.covici.com (John Covici)
Date: Fri, 22 Oct 2021 00:27:41 -0400
Subject: [Freeswitch-users] Log dialplan actions?
In-Reply-To: <202110202139.39864.Antony.Stone@freeswitch.open.source.it>
References: <202110201114.36329.Antony.Stone@freeswitch.open.source.it>
 <0100017c9ddb9b63-18a1a4f7-cec4-49a3-923b-95b13f886fff-000000@email.amazonses.com>
 <CAFGRPVp7pX9B=KVXNQ3GnziVQbSfN7TQ_ADj-4B0hxZZ0gsnsw@mail.gmail.com>
 <202110202139.39864.Antony.Stone@freeswitch.open.source.it>
Message-ID: <m3pmrxr8ea.wl-covici@ccs.covici.com>

If you look at  your
/etc/freeswitch/conf/autoload_configs/logfile.conf.xml or wherever you
have it, and your log file is at the debug level or greater, then all
dialplan actions are logged, you get to see which condition its
parsing and then you get to see which dialplan actions are actually
executed for each condition.  The difference between asterisk and
freeswitch is that freeswitch does everything statically unless you
say inline on a particular action.  Also, turn on the parameter to
prefix each line with the uuid, so you search for the next line
containing that uuid, just like with asterisk you search for c000... .

I hope this helps.

On Wed, 20 Oct 2021 15:39:39 -0400,
Antony Stone wrote:
> 
> On Wednesday 20 October 2021 at 20:41:22, David Villasmil wrote:
> 
> > if you enable xml_cdr, you will get an "app_log" array containing
> > everything.
> 
> Really?  xml_cdr sounds (and looks from its config file) to me like it records 
> CDRs, which are a summary of call legs, durations, billing, etc.
> 
> I'm looking for something which tells me, when a call comes in and gets 
> processed, which parts of the dialplan are doing the processing, so I can 
> correct my errors by spotting where it goes wrong.
> 
> Asterisk logs will show me each line of each context which a call gets 
> processed through, so I can tell "ah!  that should not go there - it should go 
> here instead".
> 
> What's the equivalent for FreeSwitch - or alternatively, how does anyone else 
> debug their dialplans?
> 
> > On Wed, Oct 20, 2021 at 3:09 PM Avi Marcus <avi at avimarcus.net> wrote:
> > > I think what you want is:
> > > 
> > > Open fs_cli (with /log 7 if not already enabled)
> > > 
> > > Also you read afterards in /var/log/freeswitch/freeswitch.log (depending
> > > on your distro/config)
> 
> That seems to be *extraordinarily* verbose - it's telling me *everything* 
> FreeSwitch is thinking about.
> 
> I'm just trying to follow the processing of a call through the dialplan.
> 
> > > I suggest modifying logfile.conf.xml to add: <param name="uuid"
> > > value="true"/> so that you can do a grep to see just one call in the
> > > logs. (and "reload mod_logfile" to make configuration take effect)
> > > 
> > > -Avi Marcus
> > > BestFone
> > > 
> > > 
> > > On Wed, Oct 20, 2021 at 4:10 PM Antony Stone wrote:
> > >> Hi.
> > >> 
> > >> I'm just getting the hang of FreeSwitch (long-time user of Asterisk),
> > >> and wondering:
> > >>
> > >>  - is there a way to get FS to log dialplan actions as they are
> > >>  executed?
> > >> 
> > >> Basically, the equivalent of Asterisk's "messages", or even better,
> > >> "verbose" log files.
> > >> 
> > >> I'm looking for a way to see how FS is processing calls as they come in,
> > >> so I can be sure my dialplans are doing what I think they should be.
> > >> 
> > >> I know I can add "log" actions of my own, but is there something that
> > >> will simply log each dialplan action automatically?
> > >> 
> > >> 
> > >> Antony.
> 
> -- 
> It is also possible that putting the birds in a laboratory setting 
> inadvertently renders them relatively incompetent.
> 
>  - Daniel C Dennett
> 
>                                                    Please reply to the list;
>                                                          please *don't* CC me.
> 
> _________________________________________________________________________
> 
> 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 mayamatakeshi at gmail.com  Fri Oct 22 04:47:22 2021
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Fri, 22 Oct 2021 13:47:22 +0900
Subject: [Freeswitch-users] Log dialplan actions?
In-Reply-To: <202110202139.39864.Antony.Stone@freeswitch.open.source.it>
References: <202110201114.36329.Antony.Stone@freeswitch.open.source.it>
 <0100017c9ddb9b63-18a1a4f7-cec4-49a3-923b-95b13f886fff-000000@email.amazonses.com>
 <CAFGRPVp7pX9B=KVXNQ3GnziVQbSfN7TQ_ADj-4B0hxZZ0gsnsw@mail.gmail.com>
 <202110202139.39864.Antony.Stone@freeswitch.open.source.it>
Message-ID: <CABaNFCaO-kN8ARfrx+aEp+YiqR+pn9XLFYj1+_pBZy++KsGGVQ@mail.gmail.com>

On Fri, Oct 22, 2021 at 12:13 PM Antony Stone <
Antony.Stone at freeswitch.open.source.it> wrote:

> On Wednesday 20 October 2021 at 20:41:22, David Villasmil wrote:
>
> > if you enable xml_cdr, you will get an "app_log" array containing
> > everything.
>
> Really?  xml_cdr sounds (and looks from its config file) to me like it
> records
> CDRs, which are a summary of call legs, durations, billing, etc.
>
> I'm looking for something which tells me, when a call comes in and gets
> processed, which parts of the dialplan are doing the processing, so I can
> correct my errors by spotting where it goes wrong.
>
> Asterisk logs will show me each line of each context which a call gets
> processed through, so I can tell "ah!  that should not go there - it
> should go
> here instead".
>
> What's the equivalent for FreeSwitch - or alternatively, how does anyone
> else
> debug their dialplans?


The app_log helps.
It shows which applications were executed.
Here is an example:

<app_log>
<application app_name="set" app_data="hold_music=local_stream://moh"
app_stamp="1634291972221409"/>
<application app_name="set" app_data="transfer_ringback=local_stream://moh"
app_stamp="1634291972221589"/>
<application app_name="set" app_data="call_timeout=60"
app_stamp="1634291972222178"/>
<application app_name="set" app_data="hangup_after_bridge=true"
app_stamp="1634291972222358"/>
<application app_name="set" app_data="ignore_early_media=true"
app_stamp="1634291972222539"/>
<application app_name="set" app_data="effective_callee_id_name=fax"
app_stamp="1634291972222771"/>
<application app_name="set" app_data="sip_callee_id_number=fax"
app_stamp="1634291972222961"/>
<application app_name="set" app_data="continue_on_fail=true"
app_stamp="1634291972223152"/>
<application app_name="set" app_data="fax_enable_t38=true"
app_stamp="1634291972223727"/>
<application app_name="set" app_data="sip_execute_on_image=t38_gateway peer
nocng" app_stamp="1634291972223932"/>
<application app_name="bridge"
app_data="{sip_cid_type=none,sip_execute_on_image=t38_gateway self
nocng,media_mix_inbound_outbound_codecs=true,hold_music=local_stream://moh,transfer_ringback=local_stream://moh}[origination_caller_id_name=0311112222,origination_caller_id_number=0311112222]sofia/internal/
40DD809595F58EA0ACFC at 127.0.0.1" app_stamp="1634291972224122"/>
</app_log>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211022/11aa8407/attachment-0001.html>

From andywolk at gmail.com  Fri Oct 22 08:21:06 2021
From: andywolk at gmail.com (Andrey Volk)
Date: Fri, 22 Oct 2021 11:21:06 +0300
Subject: [Freeswitch-users] Database docs?
In-Reply-To: <202110211512.02247.Antony.Stone@freeswitch.open.source.it>
References: <202110211512.02247.Antony.Stone@freeswitch.open.source.it>
Message-ID: <CAD55zR38w8KragbUc55+sxpeUMgU8MNHYcNNdRfukOmu0mLxxw@mail.gmail.com>

https://freeswitch.org/jira/browse/FS-11727

Just incomment it in the config files:
https://github.com/signalwire/freeswitch/blob/master/conf/vanilla/autoload_configs/switch.conf.xml#L185
https://github.com/signalwire/freeswitch/blob/master/conf/vanilla/autoload_configs/pre_load_modules.conf.xml#L4

пт, 22 окт. 2021 г. в 04:43, Antony Stone <
Antony.Stone at freeswitch.open.source.it>:

> Hi.
>
> https://freeswitch.com/index.php/2019/08/20/freeswitch-v1-10-1-release/
> tells
> me that there's a 2-year old module called mod_mariadb, but I cannot find
> anywhere on https://freeswitch.org/confluence/ the documentation about
> how to
> use it.
>
> Can someone kindly point me in the right direction?
>
> Also, just in case there's a simple answer, why isn't mod_odbc_query built
> into the Debian release packages available at
> https://files.freeswitch.org/repo/deb/debian-release/ ?
>
> I'm running FS 1.10.6
>
>
> Thanks,
>
>
> Antony.
>
> --
> "Hi, I've found a fault with the English language and I need an
> entomologist."
> "I think you mean an etymologist."
> "No.  It's a bug, not a feature."
>
>                                                    Please reply to the
> list;
>                                                          please *don't* CC
> me.
>
> _________________________________________________________________________
>
> 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/20211022/0742c974/attachment.html>

From david.villasmil.work at gmail.com  Fri Oct 22 11:25:46 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 22 Oct 2021 12:25:46 +0100
Subject: [Freeswitch-users] Log dialplan actions?
In-Reply-To: <CABaNFCaO-kN8ARfrx+aEp+YiqR+pn9XLFYj1+_pBZy++KsGGVQ@mail.gmail.com>
References: <202110201114.36329.Antony.Stone@freeswitch.open.source.it>
 <0100017c9ddb9b63-18a1a4f7-cec4-49a3-923b-95b13f886fff-000000@email.amazonses.com>
 <CAFGRPVp7pX9B=KVXNQ3GnziVQbSfN7TQ_ADj-4B0hxZZ0gsnsw@mail.gmail.com>
 <202110202139.39864.Antony.Stone@freeswitch.open.source.it>
 <CABaNFCaO-kN8ARfrx+aEp+YiqR+pn9XLFYj1+_pBZy++KsGGVQ@mail.gmail.com>
Message-ID: <CAFGRPVqNveFrvnsGK2Yv3FjPeawb+C++f6CG+4_Ss8VUU36h4w@mail.gmail.com>

Go to switch.conf.xml and set the logging to debug and to the same on
console.Conf.xml.

You get a lot of stuff since it’s on debug, but you will see the dialplan
being executed step by step.

On Fri, 22 Oct 2021 at 06:17, mayamatakeshi <mayamatakeshi at gmail.com> wrote:

>
>
> On Fri, Oct 22, 2021 at 12:13 PM Antony Stone <
> Antony.Stone at freeswitch.open.source.it> wrote:
>
>> On Wednesday 20 October 2021 at 20:41:22, David Villasmil wrote:
>>
>> > if you enable xml_cdr, you will get an "app_log" array containing
>> > everything.
>>
>> Really?  xml_cdr sounds (and looks from its config file) to me like it
>> records
>> CDRs, which are a summary of call legs, durations, billing, etc.
>>
>> I'm looking for something which tells me, when a call comes in and gets
>> processed, which parts of the dialplan are doing the processing, so I can
>> correct my errors by spotting where it goes wrong.
>>
>> Asterisk logs will show me each line of each context which a call gets
>> processed through, so I can tell "ah!  that should not go there - it
>> should go
>> here instead".
>>
>> What's the equivalent for FreeSwitch - or alternatively, how does anyone
>> else
>> debug their dialplans?
>
>
> The app_log helps.
> It shows which applications were executed.
> Here is an example:
>
> <app_log>
> <application app_name="set" app_data="hold_music=local_stream://moh"
> app_stamp="1634291972221409"/>
> <application app_name="set"
> app_data="transfer_ringback=local_stream://moh"
> app_stamp="1634291972221589"/>
> <application app_name="set" app_data="call_timeout=60"
> app_stamp="1634291972222178"/>
> <application app_name="set" app_data="hangup_after_bridge=true"
> app_stamp="1634291972222358"/>
> <application app_name="set" app_data="ignore_early_media=true"
> app_stamp="1634291972222539"/>
> <application app_name="set" app_data="effective_callee_id_name=fax"
> app_stamp="1634291972222771"/>
> <application app_name="set" app_data="sip_callee_id_number=fax"
> app_stamp="1634291972222961"/>
> <application app_name="set" app_data="continue_on_fail=true"
> app_stamp="1634291972223152"/>
> <application app_name="set" app_data="fax_enable_t38=true"
> app_stamp="1634291972223727"/>
> <application app_name="set" app_data="sip_execute_on_image=t38_gateway
> peer nocng" app_stamp="1634291972223932"/>
> <application app_name="bridge"
> app_data="{sip_cid_type=none,sip_execute_on_image=t38_gateway self
> nocng,media_mix_inbound_outbound_codecs=true,hold_music=local_stream://moh,transfer_ringback=local_stream://moh}[origination_caller_id_name=0311112222,origination_caller_id_number=0311112222]sofia/internal/
> 40DD809595F58EA0ACFC at 127.0.0.1" app_stamp="1634291972224122"/>
> </app_log>
>
> _________________________________________________________________________
>
> 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/20211022/f0014a66/attachment.html>

From lists at telefaks.de  Fri Oct 22 13:55:15 2021
From: lists at telefaks.de (Peter Steinbach)
Date: Fri, 22 Oct 2021 15:55:15 +0200
Subject: [Freeswitch-users] Issue with Enterprise originate and 487
 "ORIGINATOR_CANCEL".
In-Reply-To: <26f9382e-1c6c-6da0-0b49-3db8633668c2@telefaks.de>
References: <CAPfjz3hJNH4rhbeMb3LkboN2NC3VQoW890cVr8i4UJbdRTjgkg@mail.gmail.com>
 <26f9382e-1c6c-6da0-0b49-3db8633668c2@telefaks.de>
Message-ID: <938b5330-fe6a-d155-37ce-3260c69a7bc1@telefaks.de>

Just to mention: The patch was developed by Christian Berger from our
client Foncloud. Thank you Christian!

Best regards
Peter


Am 21.10.21 um 17:20 schrieb Peter Steinbach:
>
> Hello,
>
>
> this is a long lasting issue I firstly posted in 2017 and lastly saw
> in 2019 in this thread, and I am happy to announce, that we have found
> a solution for this.
>
> Here is the patch, which works pretty well in our environment
>
> ===================================================================================================
>
> --- freeswitch_orig/src/switch_ivr_originate.c    2021-10-20
> 12:51:04.863328041 +0200
> +++ freeswitch/src/switch_ivr_originate.c    2021-10-20
> 12:54:15.764616280 +0200
> @@ -1789,7 +1789,13 @@
>          if (cancel_cause && *cancel_cause > 0) {
>              handles[i].cancel_cause = *cancel_cause;
>          } else {
> -            handles[i].cancel_cause = SWITCH_CAUSE_LOSE_RACE;
> +            if ((hp!=NULL) && (hp->cause==SWITCH_CAUSE_SUCCESS)) {
> //Was this call taken by another destination?
> +                //Yes, the race was lost
> +                handles[i].cancel_cause = SWITCH_CAUSE_LOSE_RACE;
> +            } else {
> +                //No, something else happened, probably Originator
> Cancel   
> +                handles[i].cancel_cause = SWITCH_CAUSE_ORIGINATOR_CANCEL;
> +            }
>          }
>      }
> ===================================================================================================
>
>
> Can someone retest this and check this in?
>
>
> -- 
> With kind regards
> Peter Steinbach 
>
> Telefaks Services GmbH
> mailto:lists (att) telefaks.de
> Internet: www.telefaks.de
>
> Am 22.05.19 um 21:14 schrieb Jai Rangi:
>> All,
>> Running into a small issue with group calling, when the original
>> caller hang up, phones dont show that call as missed call, cause of
>> wrong reason code from freeswitch.
>> Scenario:
>>
>>   * Caller -> FS (Groupcall) to 101 and 102
>>   * Phones ring
>>   * Caller hangup
>>   * Phones should see missed calls,
>>   * But free-switch send cancel with Reason: SIP;cause=200;text="Call
>>     completed elsewhere", where is should send Reason:
>>     SIP;cause=487;text="ORIGINATOR_CANCEL".
>>
>> Here is what my bridge data.
>>
>> bridge(<all_threads=true,absolute_codec_string=PCMU,PCMA,G729,group_confirm_file=playback
>> /usr/local/freeswitch/sound/ivr/1443458342.wav,group_confirm_key=exec>{sip_cid_type=rpid,ignore_early_media=true,originate_timeout=30}[leg_timeout=20
>> ]user/101 at 192.168.5.50:_:[ignore_early_media=true,leg_timeout=20
>> ]user/102 at 192.168.5.50 <mailto:102 at 192.168.5.50>)
>>
>> If I send
>> bridge(<all_threads=true,absolute_codec_string=PCMU,PCMA,G729,group_confirm_file=playback
>> /usr/local/freeswitch/sound/ivr/1443458342.wav,group_confirm_key=exec>{sip_cid_type=rpid,ignore_early_media=true,originate_timeout=30}[leg_timeout=20
>> ]user/101 at 192.168.5.50 <mailto:101 at 192.168.5.50>
>>
>> Everything works fine.
>> I see some old posts but no resolution.
>> Freeswitch version on Debian 8.9
>> FreeSWITCH (Version 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit)
>>
>> Any idea or suggestion for workaround? 
>>
>> Jai
>>
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>>
>> 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/20211022/bd51b16e/attachment.html>

From gregor at infomedia.si  Fri Oct 22 16:31:26 2021
From: gregor at infomedia.si (Gregor Nanger)
Date: Fri, 22 Oct 2021 18:31:26 +0200
Subject: [Freeswitch-users] Issue with Enterprise originate and 487
	"ORIGINATOR_CANCEL".
In-Reply-To: <26f9382e-1c6c-6da0-0b49-3db8633668c2@telefaks.de>
References: <CAPfjz3hJNH4rhbeMb3LkboN2NC3VQoW890cVr8i4UJbdRTjgkg@mail.gmail.com>
 <26f9382e-1c6c-6da0-0b49-3db8633668c2@telefaks.de>
Message-ID: <CAC7ttFGHhsk9bwMLZ8S1n5JBex6c_RafaXXM4zObAkdgYSt0_w@mail.gmail.com>

Is this an issue that we do not see missed call on the phone although call
wasn't taken? And yes, we are using enterprise originate?

Br, Gregor

On Thu, 21 Oct 2021, 17:21 Peter Steinbach, <lists at telefaks.de> wrote:

> Hello,
>
>
> this is a long lasting issue I firstly posted in 2017 and lastly saw in
> 2019 in this thread, and I am happy to announce, that we have found a
> solution for this.
>
> Here is the patch, which works pretty well in our environment
>
>
> ===================================================================================================
>
> --- freeswitch_orig/src/switch_ivr_originate.c    2021-10-20
> 12:51:04.863328041 +0200
> +++ freeswitch/src/switch_ivr_originate.c    2021-10-20 12:54:15.764616280
> +0200
> @@ -1789,7 +1789,13 @@
>          if (cancel_cause && *cancel_cause > 0) {
>              handles[i].cancel_cause = *cancel_cause;
>          } else {
> -            handles[i].cancel_cause = SWITCH_CAUSE_LOSE_RACE;
> +            if ((hp!=NULL) && (hp->cause==SWITCH_CAUSE_SUCCESS)) { //Was
> this call taken by another destination?
> +                //Yes, the race was lost
> +                handles[i].cancel_cause = SWITCH_CAUSE_LOSE_RACE;
> +            } else {
> +                //No, something else happened, probably Originator
> Cancel
> +                handles[i].cancel_cause = SWITCH_CAUSE_ORIGINATOR_CANCEL;
> +            }
>          }
>      }
>
> ===================================================================================================
>
>
> Can someone retest this and check this in?
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
> Internet: www.telefaks.de
>
> Am 22.05.19 um 21:14 schrieb Jai Rangi:
>
> All,
> Running into a small issue with group calling, when the original caller
> hang up, phones dont show that call as missed call, cause of wrong reason
> code from freeswitch.
> Scenario:
>
>    - Caller -> FS (Groupcall) to 101 and 102
>    - Phones ring
>    - Caller hangup
>    - Phones should see missed calls,
>    - But free-switch send cancel with Reason: SIP;cause=200;text="Call
>    completed elsewhere", where is should send Reason:
>    SIP;cause=487;text="ORIGINATOR_CANCEL".
>
> Here is what my bridge data.
>
> bridge(<all_threads=true,absolute_codec_string=PCMU,PCMA,G729,group_confirm_file=playback
> /usr/local/freeswitch/sound/ivr/1443458342.wav,group_confirm_key=exec>{sip_cid_type=rpid,ignore_early_media=true,originate_timeout=30}[leg_timeout=20
> ]user/101 at 192.168.5.50:_:[ignore_early_media=true,leg_timeout=20 ]user/
> 102 at 192.168.5.50)
>
> If I send
> bridge(<all_threads=true,absolute_codec_string=PCMU,PCMA,G729,group_confirm_file=playback
> /usr/local/freeswitch/sound/ivr/1443458342.wav,group_confirm_key=exec>{sip_cid_type=rpid,ignore_early_media=true,originate_timeout=30}[leg_timeout=20
> ]user/101 at 192.168.5.50
>
> Everything works fine.
> I see some old posts but no resolution.
> Freeswitch version on Debian 8.9
> FreeSWITCH (Version 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit)
>
> Any idea or suggestion for workaround?
>
> Jai
>
>
>
>
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time https://signalwire.community
>
> Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com
>
> Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211022/2ed569bd/attachment-0001.html>

From brian at freeswitch.com  Fri Oct 22 18:50:29 2021
From: brian at freeswitch.com (Brian West)
Date: Fri, 22 Oct 2021 13:50:29 -0500
Subject: [Freeswitch-users] Log dialplan actions?
In-Reply-To: <CAFGRPVqNveFrvnsGK2Yv3FjPeawb+C++f6CG+4_Ss8VUU36h4w@mail.gmail.com>
References: <202110201114.36329.Antony.Stone@freeswitch.open.source.it>
 <0100017c9ddb9b63-18a1a4f7-cec4-49a3-923b-95b13f886fff-000000@email.amazonses.com>
 <CAFGRPVp7pX9B=KVXNQ3GnziVQbSfN7TQ_ADj-4B0hxZZ0gsnsw@mail.gmail.com>
 <202110202139.39864.Antony.Stone@freeswitch.open.source.it>
 <CABaNFCaO-kN8ARfrx+aEp+YiqR+pn9XLFYj1+_pBZy++KsGGVQ@mail.gmail.com>
 <CAFGRPVqNveFrvnsGK2Yv3FjPeawb+C++f6CG+4_Ss8VUU36h4w@mail.gmail.com>
Message-ID: <CAEJMVkBwi3EESBSmigz+=b=FTV+r-xMv_dywqQNvYpBOrTQR4w@mail.gmail.com>

Also the dialplan in FS is evaluated before execution, not line by line
like Asterisk, See also inline=true from confluence to get a way to set a
variable as it is evaluated.

/b


On Fri, Oct 22, 2021 at 6:47 AM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Go to switch.conf.xml and set the logging to debug and to the same on
> console.Conf.xml.
>
> You get a lot of stuff since it’s on debug, but you will see the dialplan
> being executed step by step.
>
> On Fri, 22 Oct 2021 at 06:17, mayamatakeshi <mayamatakeshi at gmail.com>
> wrote:
>
>>
>>
>> On Fri, Oct 22, 2021 at 12:13 PM Antony Stone <
>> Antony.Stone at freeswitch.open.source.it> wrote:
>>
>>> On Wednesday 20 October 2021 at 20:41:22, David Villasmil wrote:
>>>
>>> > if you enable xml_cdr, you will get an "app_log" array containing
>>> > everything.
>>>
>>> Really?  xml_cdr sounds (and looks from its config file) to me like it
>>> records
>>> CDRs, which are a summary of call legs, durations, billing, etc.
>>>
>>> I'm looking for something which tells me, when a call comes in and gets
>>> processed, which parts of the dialplan are doing the processing, so I
>>> can
>>> correct my errors by spotting where it goes wrong.
>>>
>>> Asterisk logs will show me each line of each context which a call gets
>>> processed through, so I can tell "ah!  that should not go there - it
>>> should go
>>> here instead".
>>>
>>> What's the equivalent for FreeSwitch - or alternatively, how does anyone
>>> else
>>> debug their dialplans?
>>
>>
>> The app_log helps.
>> It shows which applications were executed.
>> Here is an example:
>>
>> <app_log>
>> <application app_name="set" app_data="hold_music=local_stream://moh"
>> app_stamp="1634291972221409"/>
>> <application app_name="set"
>> app_data="transfer_ringback=local_stream://moh"
>> app_stamp="1634291972221589"/>
>> <application app_name="set" app_data="call_timeout=60"
>> app_stamp="1634291972222178"/>
>> <application app_name="set" app_data="hangup_after_bridge=true"
>> app_stamp="1634291972222358"/>
>> <application app_name="set" app_data="ignore_early_media=true"
>> app_stamp="1634291972222539"/>
>> <application app_name="set" app_data="effective_callee_id_name=fax"
>> app_stamp="1634291972222771"/>
>> <application app_name="set" app_data="sip_callee_id_number=fax"
>> app_stamp="1634291972222961"/>
>> <application app_name="set" app_data="continue_on_fail=true"
>> app_stamp="1634291972223152"/>
>> <application app_name="set" app_data="fax_enable_t38=true"
>> app_stamp="1634291972223727"/>
>> <application app_name="set" app_data="sip_execute_on_image=t38_gateway
>> peer nocng" app_stamp="1634291972223932"/>
>> <application app_name="bridge"
>> app_data="{sip_cid_type=none,sip_execute_on_image=t38_gateway self
>> nocng,media_mix_inbound_outbound_codecs=true,hold_music=local_stream://moh,transfer_ringback=local_stream://moh}[origination_caller_id_name=0311112222,origination_caller_id_number=0311112222]sofia/internal/
>> 40DD809595F58EA0ACFC at 127.0.0.1" app_stamp="1634291972224122"/>
>> </app_log>
>>
>> _________________________________________________________________________
>>
>> 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



-- 

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/20211022/baa60600/attachment.html>

From Prometheus001 at gmx.net  Fri Oct 22 21:13:50 2021
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 22 Oct 2021 23:13:50 +0200
Subject: [Freeswitch-users] Issue with Enterprise originate and 487
 "ORIGINATOR_CANCEL".
In-Reply-To: <CAC7ttFGHhsk9bwMLZ8S1n5JBex6c_RafaXXM4zObAkdgYSt0_w@mail.gmail.com>
References: <CAPfjz3hJNH4rhbeMb3LkboN2NC3VQoW890cVr8i4UJbdRTjgkg@mail.gmail.com>
 <26f9382e-1c6c-6da0-0b49-3db8633668c2@telefaks.de>
 <CAC7ttFGHhsk9bwMLZ8S1n5JBex6c_RafaXXM4zObAkdgYSt0_w@mail.gmail.com>
Message-ID: <054c544e-a65c-ff55-dc72-b3e0c3e34f3a@gmx.net>

Yes, that's the point


Am 22.10.21 um 18:31 schrieb Gregor Nanger:
> Is this an issue that we do not see missed call on the phone although
> call wasn't taken? And yes, we are using enterprise originate?
>
> Br, Gregor
>
> On Thu, 21 Oct 2021, 17:21 Peter Steinbach, <lists at telefaks.de
> <mailto:lists at telefaks.de>> wrote:
>
>     Hello,
>
>
>     this is a long lasting issue I firstly posted in 2017 and lastly
>     saw in 2019 in this thread, and I am happy to announce, that we
>     have found a solution for this.
>
>     Here is the patch, which works pretty well in our environment
>
>     ===================================================================================================
>
>     --- freeswitch_orig/src/switch_ivr_originate.c    2021-10-20
>     12:51:04.863328041 +0200
>     +++ freeswitch/src/switch_ivr_originate.c    2021-10-20
>     12:54:15.764616280 +0200
>     @@ -1789,7 +1789,13 @@
>              if (cancel_cause && *cancel_cause > 0) {
>                  handles[i].cancel_cause = *cancel_cause;
>              } else {
>     -            handles[i].cancel_cause = SWITCH_CAUSE_LOSE_RACE;
>     +            if ((hp!=NULL) && (hp->cause==SWITCH_CAUSE_SUCCESS))
>     { //Was this call taken by another destination?
>     +                //Yes, the race was lost
>     +                handles[i].cancel_cause = SWITCH_CAUSE_LOSE_RACE;
>     +            } else {
>     +                //No, something else happened, probably
>     Originator Cancel   
>     +                handles[i].cancel_cause =
>     SWITCH_CAUSE_ORIGINATOR_CANCEL;
>     +            }
>              }
>          }
>     ===================================================================================================
>
>
>     Can someone retest this and check this in?
>
>
>     --
>     With kind regards
>     Peter Steinbach
>
>     Telefaks Services GmbH
>     mailto:lists (att) telefaks.de <http://telefaks.de>
>     Internet: www.telefaks.de <http://www.telefaks.de>
>
>     Am 22.05.19 um 21:14 schrieb Jai Rangi:
>>     All,
>>     Running into a small issue with group calling, when the original
>>     caller hang up, phones dont show that call as missed call, cause
>>     of wrong reason code from freeswitch.
>>     Scenario:
>>
>>       * Caller -> FS (Groupcall) to 101 and 102
>>       * Phones ring
>>       * Caller hangup
>>       * Phones should see missed calls,
>>       * But free-switch send cancel with Reason:
>>         SIP;cause=200;text="Call completed elsewhere", where is
>>         should send Reason: SIP;cause=487;text="ORIGINATOR_CANCEL".
>>
>>     Here is what my bridge data.
>>
>>     bridge(<all_threads=true,absolute_codec_string=PCMU,PCMA,G729,group_confirm_file=playback
>>     /usr/local/freeswitch/sound/ivr/1443458342.wav,group_confirm_key=exec>{sip_cid_type=rpid,ignore_early_media=true,originate_timeout=30}[leg_timeout=20
>>     ]user/101 at 192.168.5.50:_:[ignore_early_media=true,leg_timeout=20
>>     ]user/102 at 192.168.5.50 <mailto:102 at 192.168.5.50>)
>>
>>     If I send
>>     bridge(<all_threads=true,absolute_codec_string=PCMU,PCMA,G729,group_confirm_file=playback
>>     /usr/local/freeswitch/sound/ivr/1443458342.wav,group_confirm_key=exec>{sip_cid_type=rpid,ignore_early_media=true,originate_timeout=30}[leg_timeout=20
>>     ]user/101 at 192.168.5.50 <mailto:101 at 192.168.5.50>
>>
>>     Everything works fine.
>>     I see some old posts but no resolution.
>>     Freeswitch version on Debian 8.9
>>     FreeSWITCH (Version 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit)
>>
>>     Any idea or suggestion for workaround? 
>>
>>     Jai
>>
>>
>>
>>
>>
>>
>>
>>     _________________________________________________________________________
>>
>>     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
>
>
> _________________________________________________________________________
>
> 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/20211022/8561ae40/attachment-0001.html>

From lists at telefaks.de  Fri Oct 22 21:16:35 2021
From: lists at telefaks.de (Peter Steinbach)
Date: Fri, 22 Oct 2021 23:16:35 +0200
Subject: [Freeswitch-users] Issue with Enterprise originate and 487
 "ORIGINATOR_CANCEL".
In-Reply-To: <CAC7ttFGHhsk9bwMLZ8S1n5JBex6c_RafaXXM4zObAkdgYSt0_w@mail.gmail.com>
References: <CAPfjz3hJNH4rhbeMb3LkboN2NC3VQoW890cVr8i4UJbdRTjgkg@mail.gmail.com>
 <26f9382e-1c6c-6da0-0b49-3db8633668c2@telefaks.de>
 <CAC7ttFGHhsk9bwMLZ8S1n5JBex6c_RafaXXM4zObAkdgYSt0_w@mail.gmail.com>
Message-ID: <c2edad45-4942-f32a-0d56-9dfe0bb26ada@telefaks.de>

Yes, that's the point


Am 22.10.21 um 18:31 schrieb Gregor Nanger:
> Is this an issue that we do not see missed call on the phone although
> call wasn't taken? And yes, we are using enterprise originate?
>
> Br, Gregor
>
> On Thu, 21 Oct 2021, 17:21 Peter Steinbach, <lists at telefaks.de
> <mailto:lists at telefaks.de>> wrote:
>
>     Hello,
>
>
>     this is a long lasting issue I firstly posted in 2017 and lastly
>     saw in 2019 in this thread, and I am happy to announce, that we
>     have found a solution for this.
>
>     Here is the patch, which works pretty well in our environment
>
>     ===================================================================================================
>
>     --- freeswitch_orig/src/switch_ivr_originate.c    2021-10-20
>     12:51:04.863328041 +0200
>     +++ freeswitch/src/switch_ivr_originate.c    2021-10-20
>     12:54:15.764616280 +0200
>     @@ -1789,7 +1789,13 @@
>              if (cancel_cause && *cancel_cause > 0) {
>                  handles[i].cancel_cause = *cancel_cause;
>              } else {
>     -            handles[i].cancel_cause = SWITCH_CAUSE_LOSE_RACE;
>     +            if ((hp!=NULL) && (hp->cause==SWITCH_CAUSE_SUCCESS))
>     { //Was this call taken by another destination?
>     +                //Yes, the race was lost
>     +                handles[i].cancel_cause = SWITCH_CAUSE_LOSE_RACE;
>     +            } else {
>     +                //No, something else happened, probably
>     Originator Cancel   
>     +                handles[i].cancel_cause =
>     SWITCH_CAUSE_ORIGINATOR_CANCEL;
>     +            }
>              }
>          }
>     ===================================================================================================
>
>
>     Can someone retest this and check this in?
>
>
>     -- 
>     With kind regards
>     Peter Steinbach 
>
>     Telefaks Services GmbH
>     mailto:lists (att) telefaks.de <http://telefaks.de>
>     Internet: www.telefaks.de <http://www.telefaks.de>
>
>     Am 22.05.19 um 21:14 schrieb Jai Rangi:
>>     All,
>>     Running into a small issue with group calling, when the original
>>     caller hang up, phones dont show that call as missed call, cause
>>     of wrong reason code from freeswitch.
>>     Scenario:
>>
>>       * Caller -> FS (Groupcall) to 101 and 102
>>       * Phones ring
>>       * Caller hangup
>>       * Phones should see missed calls,
>>       * But free-switch send cancel with Reason:
>>         SIP;cause=200;text="Call completed elsewhere", where is
>>         should send Reason: SIP;cause=487;text="ORIGINATOR_CANCEL".
>>
>>     Here is what my bridge data.
>>
>>     bridge(<all_threads=true,absolute_codec_string=PCMU,PCMA,G729,group_confirm_file=playback
>>     /usr/local/freeswitch/sound/ivr/1443458342.wav,group_confirm_key=exec>{sip_cid_type=rpid,ignore_early_media=true,originate_timeout=30}[leg_timeout=20
>>     ]user/101 at 192.168.5.50:_:[ignore_early_media=true,leg_timeout=20
>>     ]user/102 at 192.168.5.50 <mailto:102 at 192.168.5.50>)
>>
>>     If I send
>>     bridge(<all_threads=true,absolute_codec_string=PCMU,PCMA,G729,group_confirm_file=playback
>>     /usr/local/freeswitch/sound/ivr/1443458342.wav,group_confirm_key=exec>{sip_cid_type=rpid,ignore_early_media=true,originate_timeout=30}[leg_timeout=20
>>     ]user/101 at 192.168.5.50 <mailto:101 at 192.168.5.50>
>>
>>     Everything works fine.
>>     I see some old posts but no resolution.
>>     Freeswitch version on Debian 8.9
>>     FreeSWITCH (Version 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit)
>>
>>     Any idea or suggestion for workaround? 
>>
>>     Jai
>>
>>
>>
>>
>>
>>
>>
>>     _________________________________________________________________________
>>
>>     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
>
>
> _________________________________________________________________________
>
> 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/20211022/1d8b3011/attachment.html>

From lists at telefaks.de  Fri Oct 22 21:29:00 2021
From: lists at telefaks.de (Peter Steinbach)
Date: Fri, 22 Oct 2021 23:29:00 +0200
Subject: [Freeswitch-users] Internal Interface suddenly freezes
In-Reply-To: <79ef33e2-c520-c3a6-7ae2-fa425f19ac09@telefaks.de>
References: <40B08719-BA62-4E3F-984B-556AB40A5A91@wnt.at>
 <CAFGRPVrj-ewOpY8h9HUdHz1T=A8KCg=Nsqp-dxGoC4NR23i_ZA@mail.gmail.com>
 <VI1P190MB0494FA8017F747D0C79C8F8F867E9@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <001b01d724a1$ba791130$2f6b3390$@gmail.com>
 <VI1P190MB049439288D7951FC99D99C75867E9@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <CAFGRPVqGyM7vuYqHH-j_w1KUpQZjh1c7Jj8KFS5aYo8TH0=ZkA@mail.gmail.com>
 <E3DD2F0E-0153-4DA0-BEA2-A55ADBF4BB84@wnt.at>
 <CAL0pLpZD57B9safu17hd=Va0O0YRPkWZLdvSCV88ejcfYDAgYw@mail.gmail.com>
 <001801d7259d$db351020$919f3060$@gmail.com>
 <9B9FC99A-90CD-4BC5-8B52-FE407CF759DE@wnt.at>
 <E9072A99-2776-4124-9D22-D8AD99D0B6B5@wnt.at>
 <79ef33e2-c520-c3a6-7ae2-fa425f19ac09@telefaks.de>
Message-ID: <8d07ba9d-b306-65c1-e9f6-ff34745cadc9@telefaks.de>

Hello,


for the stuck sofia internal profile, I received a patch from Brian we
have in place since June,11 under high load (several thousand
registrations with lot's of presence involved (sometimes some hundred
Notifys per second)). Since then our problems nearly went away. Feel
free to test it. I would love to hear from your experiences.

Here it is (it's not formatted, indented yet)

root at fs /usr/src/freeswitch/src/mod/endpoints/mod_sofia # diff
sofia_presence.c.org <http://sofia_presence.c.org> sofia_presence.c
724,733d723
<               sql = switch_mprintf("select sip_registrations.sip_user, "
<                                                               
"sip_registrations.sub_host, "
<                                                               
"sip_registrations.status, "
<                                                               
"sip_registrations.rpid, "
<                                                                "'', "
<                                                               
"sip_dialogs.uuid, "
<                                                               
"sip_dialogs.state, "
<                                                               
"sip_dialogs.direction, "
<                                                               
"sip_dialogs.sip_to_user, "
<                                                               
"sip_dialogs.sip_to_host, "
735,740c725,757
<                                                               
"sip_presence.status,"
<                                                               
"sip_presence.rpid,"
<                                                               
"sip_dialogs.presence_id, "
<                                                               
"sip_presence.open_closed,"
<                                                               
"'%q','%q','%q' "
<                                                                "from
sip_registrations "
---
> sql = switch_mprintf("select sip_registrations.sip_user, "
> "sip_registrations.sub_host, "
> "sip_registrations.status, "
> "sip_registrations.rpid, "
> "'', "
> "sip_dialogs.uuid, "
> "sip_dialogs.state, "
> "sip_dialogs.direction, "
> "sip_dialogs.sip_to_user, "
> "sip_dialogs.sip_to_host, "
> "sip_presence.status,"
> "sip_presence.rpid,"
> "sip_dialogs.presence_id, "
> "sip_presence.open_closed,"
> "'%q','%q','%q' "
> "from sip_registrations "
> "left join sip_dialogs on "
> "sip_dialogs.hostname = sip_registrations.hostname and
sip_dialogs.profile_name = sip_registrations.profile_name and "
> "sip_dialogs.call_info_state != 'seized' and "
> "(sip_dialogs.presence_id = sip_registrations.sip_user %q '@' %q
sip_registrations.sub_host "
> "or (sip_dialogs.sip_from_user = sip_registrations.sip_user "
> "and sip_dialogs.sip_from_host = sip_registrations.sip_host)) "
> "left join sip_presence on "
> "sip_presence.hostname=sip_registrations.hostname and "
> "sip_registrations.sip_user=sip_presence.sip_user and
sip_registrations.orig_server_host=sip_presence.sip_host and "
> "sip_registrations.profile_name=sip_presence.profile_name "
> "where sip_registrations.hostname='%q' and
sip_registrations.profile_name='%q' "
> "and sip_registrations.sip_user='%q' and "
> "(sip_registrations.orig_server_host='%q' or
sip_registrations.sub_host='%q' "
> ")",
> dh.status, dh.rpid, switch_str_nil(sub_call_id),
> switch_sql_concat(), switch_sql_concat(),
> mod_sofia_globals.hostname, profile->name, probe_euser, probe_host, 
probe_euser, probe_host, probe_host);
742,758d758
<                                                                "left
join sip_dialogs on "
<                                                               
"sip_dialogs.hostname = sip_registrations.hostname and
sip_dialogs.profile_name = sip_registrations.profile_name and ("
<                                                               
"sip_dialogs.presence_id = sip_registrations.sip_user %q '@' %q
sip_registrations.sub_host "
<                                                                "or
(sip_dialogs.sip_from_user = sip_registrations.sip_user "
<                                                                "and
sip_dialogs.sip_from_host = sip_registrations.sip_host)) "
<
<                                                                "left
join sip_presence on "
<                                                               
"sip_presence.hostname=sip_registrations.hostname and "
<                                                               
"(sip_registrations.sip_user=sip_presence.sip_user and
sip_registrations.orig_server_host=sip_presence.sip_host and "
<                                                               
"sip_registrations.profile_name=sip_presence.profile_name) "
<                                                                "where
sip_registrations.hostname='%q' and sip_registrations.profile_name='%q'
and sip_dialogs.call_info_state != 'seized' "
<                                                                "and
sip_dialogs.presence_id='%q@%q' or (sip_registrations.sip_user='%q' and "
<                                                               
"(sip_registrations.orig_server_host='%q' or
sip_registrations.sub_host='%q' "
<                                                                "))",
<                                                               
dh.status, dh.rpid, switch_str_nil(sub_call_id),
<                                                               
switch_sql_concat(), switch_sql_concat(),
<                                                               
mod_sofia_globals.hostname, profile->name, probe_euser, probe_host, 
probe_euser, probe_host, probe_host);


Am 01.04.21 um 18:40 schrieb Peter Steinbach:
>
> I think, I have a similar problem with registers (and presence) via
> XML-Curl. Freswitch is not sending a Curl request to the server then
> (as we grepped on the network interface).
>
> This happens under high load with a lot of presence involved. Suddenly
> Freeswitch internal profile stops responding and after a while (some
> minutes) it responds again. During that time, no register requests are
> answered at all. Only a few XML-Curl requests are sent to the server,
> but they do not contain the user info (which are needed for
> registrations) and are related to mod_voicemail.c etc.
>
> My debug log looks like this
>
> tport.c:3286 tport_tsend() tport_tsend(0x7f9c68005470) tpn =
> */xx.xxx.xxx.x:39672
> tport.c:4075 tport_resolve() tport_resolve addrinfo = xx.xxx.xxx.x:39672
> tport.c:4709 tport_by_addrinfo() tport_by_addrinfo(0x7f9c68005470):
> not found by name */xx.xxx.xxx.x:39672
> tport.c:3623 tport_vsend() tport_vsend(0x7f9c68005470): 1065 bytes of
> 1065 to udp/xx.xxx.xxx.x:39672
> tport.c:3521 tport_send_msg() tport_vsend returned 1065
> nta.c:8348 outgoing_send() nta: sent NOTIFY (304240670) to
> */xx.xxx.xxx.x:39672
> tport.c:4189 tport_pend() tport_pend(0x7f9c68005470): pending
> 0x7f9997dcf880 for udp/xxx.xx.x.xxx:5060 (already 124658)
> nua_stack.c:569 nua_stack_signal() nua(0x7f9a0bd45ae0): recv signal
> r_notify
> nua_params.c:484 nua_stack_set_params() nua: nua_stack_set_params:
> entering
> soa.c:280 soa_clone() soa_clone(static::0x7f9c68001b90,
> 0x7f9c68001390, 0x7f9a0bd45ae0) called
> soa.c:403 soa_set_params() soa_set_params(static::0x7f9997dd1c90, ...)
> called
> soa.c:403 soa_set_params() soa_set_params(static::0x7f9997dd1c90, ...)
> called
> nta.c:4446 nta_leg_tcreate() nta_leg_tcreate(0x7f9997dcd870)
> nta.c:2694 nta_tpn_by_url() nta: selecting scheme sip
> tport.c:3286 tport_tsend() tport_tsend(0x7f9c68005470) tpn =
> udp/xx.xxx.xxx.x:58694
> tport.c:4075 tport_resolve() tport_resolve addrinfo = xx.xxx.xxx.x:58694
> tport.c:4709 tport_by_addrinfo() tport_by_addrinfo(0x7f9c68005470):
> not found by name udp/xx.xxx.xxx.x:58694
> tport.c:3623 tport_vsend() tport_vsend(0x7f9c68005470): 1114 bytes of
> 1114 to udp/xx.xxx.xxx.x:58694
> tport.c:3521 tport_send_msg() tport_vsend returned 1114
> nta.c:8348 outgoing_send() nta: sent NOTIFY (304240671) to
> udp/xx.xxx.xxx.x:58694
> tport.c:4189 tport_pend() tport_pend(0x7f9c68005470): pending
> 0x7f9997dd2650 for udp/xxx.xx.x.xxx:5060 (already 124659)
>
> Looks like some event queue is stuck?
>
> /Peter
>
>
> Am 01.04.21 um 00:10 schrieb Stefan Kainz:
>> Hi guys, 
>>
>> Me again. 
>> It just happened again :)
>> ( With extra debugging lines in my register.lua and a not found xml
>> like you suggested if the result is empty )
>>
>> This time I turned up the logging to 9 while it was happening.
>> This is what I found:
>>
>> While its happening, a Register request causes the following to be
>> logged: 
>>
>> nta.c:2880 agent_recv_request() nta: received REGISTER sip:xxxxx
>> <sip:xxxxx> SIP/2.0 (CSeq 73038)
>> nta.c:3012 agent_recv_request() nta: REGISTER (73038) going to
>> existing REGISTER transaction
>> tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7fab1c0042b0):
>> events IN
>> tport.c:2864 tport_recv_event() tport_recv_event(0x7fab1c0042b0)
>> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fab1c0042b0) msg
>> 0x7fab1d095280 from (udp/xxx.xxx.xxx.xxx:5060) has 809 bytes, veclen = 1
>> tport.c:3023 tport_deliver() tport_deliver(0x7fab1c0042b0): msg
>> 0x7fab1d095280 (809 bytes) from udp/xxx.xxx.xxx.xxx:5060/sip next=(nil)
>>
>> And that’s it. it doesn’t even execute my register.lua anymore, I
>> just tries to update an existing Register transaction. 
>> When I start an ngrep on port 5060, all I see is the register, no
>> unauthorised or ok response.
>>
>> While a Register Request after I restarted freeswitch and everything
>> worked again looks like this: 
>>
>> nta.c:2880 agent_recv_request() nta: received REGISTER sip:xxxxx
>> <sip:xxxxx> SIP/2.0 (CSeq 37884)
>> nta.c:3085 agent_recv_request() nta: REGISTER (37884) going to a
>> default leg
>> nua_server.c:102 nua_stack_process_request() nua:
>> nua_stack_process_request: entering
>> nua_stack.c:899 nh_create() nua: nh_create: entering
>> nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>> entering
>> soa.c:280 soa_clone() soa_clone(static::0x7f8514001930,
>> 0x7f8514001130, 0x7f85142225b0) called
>> soa.c:403 soa_set_params() soa_set_params(static::0x7f85141f6f70,
>> ...) called
>> nua_stack.c:271 nua_stack_event() nua(0x7f85142225b0): event
>> i_register 100 Trying
>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>> entering
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> 2021-03-31 23:47:11.485911 [INFO] switch_cpp.cpp:1328 register.lua:
>> Connect to database
>> ...
>>
>> The last entry “Connect to database” is already a consoleLog from my
>> register.lua. 
>> From here on out everything works great again … 
>>
>> So, do you think it could still be true that one request to my
>> register.lua holds up all subsequent requests?
>>
>> Thank you very much!
>>
>> Regards, 
>>
>>
>>
>>
>>> On 31.03.2021, at 09:16, Stefan Kainz <s.kainz at wnt.at
>>> <mailto:s.kainz at wnt.at>> wrote:
>>>
>>> Ok thanks!
>>>
>>> I will try to return a not found and add debug lines to the
>>> register.lua. 
>>> I will let you know If that helped. 
>>>
>>> Regards, 
>>> Stefan
>>>
>>>> On 30.03.2021, at 21:49, Bote Man <botelist at gmail.com
>>>> <mailto:botelist at gmail.com>> wrote:
>>>>
>>>> And add some logging lines to the Lua script so that it can tell
>>>> you what is happening at each stage. Perhaps it is doing something
>>>> that you don’t want it to do, or not doing something that you do
>>>> want it to do?
>>>>  
>>>> Bote
>>>>  
>>>>  
>>>> *From:* FreeSWITCH-users
>>>> <freeswitch-users-bounces at lists.freeswitch.org
>>>> <mailto:freeswitch-users-bounces at lists.freeswitch.org>> *On Behalf
>>>> Of *Mircea Botoca-Huh
>>>> *Sent:* Tuesday, 30 March, 2021 15:21
>>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
>>>> <mailto:freeswitch-users at lists.freeswitch.org>>
>>>> *Subject:* Re: [Freeswitch-users] Internal Interface suddenly freezes
>>>>  
>>>> Hi, 
>>>>  
>>>> If you don't wish to serve an action from your Lua script, you must
>>>> return not found. 
>>>>  
>>>> This is from confluence doc:
>>>>  
>>>>
>>>> If your LUA application receives a request and you don't wish to
>>>> serve dialplan or like to fallback to plain XML dialplan, then you
>>>> should return the following "not found" result.
>>>>
>>>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>>>> <document type="freeswitch/xml">
>>>>   <section name="result">
>>>>     <result status="not found" />
>>>>   </section>
>>>> </document>
>>>>  
>>>>  
>>>> Best regards, 
>>>> Mircea 
>>>>  
>>>> mar., 30 mar. 2021, 22:05 Stefan Kainz <s.kainz at wnt.at
>>>> <mailto:s.kainz at wnt.at>> a scris:
>>>>> Hi,  
>>>>>  
>>>>> The lua script is called on a directory request. 
>>>>> Your know, autoload_configs, lua.conf.xml: 
>>>>>  
>>>>> <param name="xml-handler-script" value="register.lua"/>
>>>>> <param name="xml-handler-bindings" value="directory,dialplan”/>
>>>>>  
>>>>> Maybe the dialplan in the xml-handler-bindings is doing something
>>>>> I don’t want … 
>>>>>  
>>>>> In the register.lua I do basic sanity checks, and of course a
>>>>> database query that returns the user-data.
>>>>> After that I generate the xml that gets sent to freeswitch. 
>>>>>  
>>>>> Yesterday I added cacheable=“120" to the <user> tag.
>>>>> Thats no solution of course, but it looks like when the data is in
>>>>> the cache the lua script doesn’t get called. 
>>>>> So maybe the error won’t occur as often as before … 
>>>>>  
>>>>> But, I have an idea, maybe you can confirm this. 
>>>>> With dialplan enabled in xml-handler-bindings, the lua script is
>>>>> also called on calls, not just on registrations. 
>>>>> ( I think that’s necessary to authenticate the users on an
>>>>> outgoing call, please correct me if im wrong )
>>>>> In the register.lua I only handle directory-requests and
>>>>> action==user_call.
>>>>> All other requests to my register.lua are unhandled and therefore
>>>>> I don’t return any xml at all. 
>>>>>  
>>>>> Maybe I get a request to my register.lua that is not a
>>>>> directory-request and not a user_call and since I don’t return a
>>>>> xml, Sofia hangs.
>>>>>  
>>>>> Does that sound plausible?
>>>>>  
>>>>> Thank you in advance!
>>>>>  
>>>>> Regards, 
>>>>> Stefan
>>>>>  
>>>>>  
>>>>>
>>>>>
>>>>>> On 29.03.2021, at 23:57, David Villasmil
>>>>>> <david.villasmil.work at gmail.com
>>>>>> <mailto:david.villasmil.work at gmail.com>> wrote:
>>>>>>  
>>>>>> That was going to be my next question: what’s lua doing and at
>>>>>> what point?
>>>>>>  
>>>>>> On Mon, 29 Mar 2021 at 16:09, Stefan Kainz <s.kainz at wnt.at
>>>>>> <mailto:s.kainz at wnt.at>> wrote:
>>>>>>> I just found an issue on jira, where it seems someone had the
>>>>>>> same problem I have. 
>>>>>>> https://freeswitch.org/jira/browse/FS-3328
>>>>>>> I don’t have mod_xml_curl enabled though. 
>>>>>>>  
>>>>>>> But knowing that sofia can handle only one register at a time
>>>>>>> and then blocking all subsequent Registers is a good starting
>>>>>>> point … 
>>>>>>>  
>>>>>>> Regards, 
>>>>>>>  
>>>>>>> *Von:* FreeSWITCH-users
>>>>>>> <freeswitch-users-bounces at lists.freeswitch.org
>>>>>>> <mailto:freeswitch-users-bounces at lists.freeswitch.org>> *Im
>>>>>>> Auftrag von *Bote Man
>>>>>>> *Gesendet:* Montag, 29. März 2021 15:45
>>>>>>> *An:* 'FreeSWITCH Users Help'
>>>>>>> <freeswitch-users at lists.freeswitch.org
>>>>>>> <mailto:freeswitch-users at lists.freeswitch.org>>
>>>>>>> *Betreff:* Re: [Freeswitch-users] Internal Interface suddenly
>>>>>>> freezes
>>>>>>>  
>>>>>>> The one common element is your Lua script.
>>>>>>>  
>>>>>>> I am certainly no expert on script writing, but I have seen a
>>>>>>> number of problems on the mailing list over the years with
>>>>>>> scripts doing “too much” work during critical sections of the
>>>>>>> dialplan. Perhaps there is a race condition?
>>>>>>>  
>>>>>>> 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 *Stefan Kainz
>>>>>>> *Sent:* Monday, 29 March, 2021 07:49
>>>>>>> *To:* FreeSWITCH Users Help
>>>>>>> <freeswitch-users at lists.freeswitch.org
>>>>>>> <mailto:freeswitch-users at lists.freeswitch.org>>
>>>>>>> *Subject:* Re: [Freeswitch-users] Internal Interface suddenly
>>>>>>> freezes
>>>>>>>  
>>>>>>> Thank you for your answer!
>>>>>>>  
>>>>>>> Hmm, I also tested it on two completely different servers ( no
>>>>>>> virtualization ) and the problem exists on both. 
>>>>>>> Im also going to try it on a third server, also completely
>>>>>>> different, but I cant really image that this is a hardware-thing … 
>>>>>>>  
>>>>>>> We also have many freeswitch servers in production ( Exactly the
>>>>>>> same hardware as the server with the problem ). 
>>>>>>> The only difference is that one of those servers handles
>>>>>>> registrations, and one doesn’t.
>>>>>>> The one handling the registrations has the problem, the other
>>>>>>> one doesn’t. 
>>>>>>>  
>>>>>>> Its really strange … 
>>>>>>>  
>>>>>>> Regards,
>>>>>>>  
>>>>>>> *Von:* FreeSWITCH-users
>>>>>>> <freeswitch-users-bounces at lists.freeswitch.org
>>>>>>> <mailto:freeswitch-users-bounces at lists.freeswitch.org>> *Im
>>>>>>> Auftrag von *David Villasmil
>>>>>>> *Gesendet:* Montag, 29. März 2021 13:12
>>>>>>> *An:* FreeSWITCH Users Help
>>>>>>> <freeswitch-users at lists.freeswitch.org
>>>>>>> <mailto:freeswitch-users at lists.freeswitch.org>>
>>>>>>> *Betreff:* Re: [Freeswitch-users] Internal Interface suddenly
>>>>>>> freezes
>>>>>>>  
>>>>>>> That looks to be more on the hardware side than software. It’d
>>>>>>> be an extremely coincidence those versions and all those OS have
>>>>>>> some issue somewhere.
>>>>>>> Change hardware. 
>>>>>>>  
>>>>>>> On Sun, 28 Mar 2021 at 09:12, Stefan Kainz <s.kainz at wnt.at
>>>>>>> <mailto:s.kainz at wnt.at>> wrote:
>>>>>>>> Hi everbody, 
>>>>>>>>
>>>>>>>> I have a little bit of a problem. 
>>>>>>>> Im using Version 1.10.3. ( but this problem also occurs on
>>>>>>>> version 1.4.18 )
>>>>>>>>
>>>>>>>> Sometimes the internal Sofia interface just stops responding to
>>>>>>>> SIP Requests. 
>>>>>>>> It sometimes happens once every day, and sometimes once a week. 
>>>>>>>> It happens at completely random times, like one day in the
>>>>>>>> morning, and the next day in the middle of the night. 
>>>>>>>> The freeswitch.log gives me nothing, its like the Sofia
>>>>>>>> interface was stopped. 
>>>>>>>>
>>>>>>>> When I try to restart the interface with "sofia profile
>>>>>>>> internal restart” nothing happens. The fs_cli just remains
>>>>>>>> stuck with that command.
>>>>>>>>
>>>>>>>> The solution is to restart the freeswitch service.
>>>>>>>>
>>>>>>>> Sometimes when I recognise it too late, for example in the
>>>>>>>> middle of the night, it seems like the problem solves itself
>>>>>>>> after about 2 hours. 
>>>>>>>> The profile just starts working again, without somebody doing
>>>>>>>> anything.
>>>>>>>>
>>>>>>>> I have checked a variety of things, including the firewall &
>>>>>>>> fail2ban, network connection, made sure watchdog is disabled,
>>>>>>>> and also tested it on different Debian-versions and freeswitch
>>>>>>>> versions. 
>>>>>>>> It seems this problem occurs on every freeswitch version i have
>>>>>>>> tested. 
>>>>>>>>
>>>>>>>> The external-profile on the other hand, keeps working like
>>>>>>>> nothing happened. 
>>>>>>>>
>>>>>>>> Both Interfaces listen on the same network-device with a public
>>>>>>>> ip. 
>>>>>>>> The only difference is, the internal profile uses a Lua file to
>>>>>>>> handle registrations.  
>>>>>>>>
>>>>>>>> Has anybody come across anything similar? 
>>>>>>>>
>>>>>>>> Any help is much appreciated!
>>>>>>>>
>>>>>>>> Regards, 
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>>
>>>>>>>> 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.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
>>>>>>>> UNSUBSCRIBE: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 <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.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
>>>>>>> UNSUBSCRIBE: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 <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.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
>>>>>> UNSUBSCRIBE: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.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
>>>>> UNSUBSCRIBE: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.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
>>>> UNSUBSCRIBE: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
>>>
>>> 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/20211022/957ccaf0/attachment-0001.html>

From brian at freeswitch.com  Fri Oct 22 22:21:48 2021
From: brian at freeswitch.com (Brian West)
Date: Fri, 22 Oct 2021 17:21:48 -0500
Subject: [Freeswitch-users] Internal Interface suddenly freezes
In-Reply-To: <8d07ba9d-b306-65c1-e9f6-ff34745cadc9@telefaks.de>
References: <40B08719-BA62-4E3F-984B-556AB40A5A91@wnt.at>
 <CAFGRPVrj-ewOpY8h9HUdHz1T=A8KCg=Nsqp-dxGoC4NR23i_ZA@mail.gmail.com>
 <VI1P190MB0494FA8017F747D0C79C8F8F867E9@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <001b01d724a1$ba791130$2f6b3390$@gmail.com>
 <VI1P190MB049439288D7951FC99D99C75867E9@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <CAFGRPVqGyM7vuYqHH-j_w1KUpQZjh1c7Jj8KFS5aYo8TH0=ZkA@mail.gmail.com>
 <E3DD2F0E-0153-4DA0-BEA2-A55ADBF4BB84@wnt.at>
 <CAL0pLpZD57B9safu17hd=Va0O0YRPkWZLdvSCV88ejcfYDAgYw@mail.gmail.com>
 <001801d7259d$db351020$919f3060$@gmail.com>
 <9B9FC99A-90CD-4BC5-8B52-FE407CF759DE@wnt.at>
 <E9072A99-2776-4124-9D22-D8AD99D0B6B5@wnt.at>
 <79ef33e2-c520-c3a6-7ae2-fa425f19ac09@telefaks.de>
 <8d07ba9d-b306-65c1-e9f6-ff34745cadc9@telefaks.de>
Message-ID: <CAEJMVkAB-B4_t_KF5Ce1nQa7=-yDDX6Hq2faONoWAQMESHYFXg@mail.gmail.com>

Biggest blocker to getting this merged is a test set to prove it in a
controlled environment.

/b


On Fri, Oct 22, 2021 at 4:52 PM Peter Steinbach <lists at telefaks.de> wrote:

> Hello,
>
>
> for the stuck sofia internal profile, I received a patch from Brian we
> have in place since June,11 under high load (several thousand registrations
> with lot's of presence involved (sometimes some hundred Notifys per
> second)). Since then our problems nearly went away. Feel free to test it. I
> would love to hear from your experiences.
>
> Here it is (it's not formatted, indented yet)
> root at fs /usr/src/freeswitch/src/mod/endpoints/mod_sofia # diff
> sofia_presence.c.org sofia_presence.c
> 724,733d723
> <               sql = switch_mprintf("select sip_registrations.sip_user, "
> <
> "sip_registrations.sub_host, "
> <
> "sip_registrations.status, "
> <
> "sip_registrations.rpid, "
> <                                                                "'', "
> <
> "sip_dialogs.uuid, "
> <
> "sip_dialogs.state, "
> <
> "sip_dialogs.direction, "
> <
> "sip_dialogs.sip_to_user, "
> <
> "sip_dialogs.sip_to_host, "
> 735,740c725,757
> <
> "sip_presence.status,"
> <
> "sip_presence.rpid,"
> <
> "sip_dialogs.presence_id, "
> <
> "sip_presence.open_closed,"
> <
> "'%q','%q','%q' "
> <                                                                "from
> sip_registrations "
> ---
> > sql = switch_mprintf("select sip_registrations.sip_user, "
> > "sip_registrations.sub_host, "
> > "sip_registrations.status, "
> > "sip_registrations.rpid, "
> > "'', "
> > "sip_dialogs.uuid, "
> > "sip_dialogs.state, "
> > "sip_dialogs.direction, "
> > "sip_dialogs.sip_to_user, "
> > "sip_dialogs.sip_to_host, "
> > "sip_presence.status,"
> > "sip_presence.rpid,"
> > "sip_dialogs.presence_id, "
> > "sip_presence.open_closed,"
> > "'%q','%q','%q' "
> > "from sip_registrations "
> > "left join sip_dialogs on "
> > "sip_dialogs.hostname = sip_registrations.hostname and
> sip_dialogs.profile_name = sip_registrations.profile_name and "
> > "sip_dialogs.call_info_state != 'seized' and "
> > "(sip_dialogs.presence_id = sip_registrations.sip_user %q '@' %q
> sip_registrations.sub_host "
> > "or (sip_dialogs.sip_from_user = sip_registrations.sip_user "
> > "and sip_dialogs.sip_from_host = sip_registrations.sip_host)) "
> > "left join sip_presence on "
> > "sip_presence.hostname=sip_registrations.hostname and "
> > "sip_registrations.sip_user=sip_presence.sip_user and
> sip_registrations.orig_server_host=sip_presence.sip_host and "
> > "sip_registrations.profile_name=sip_presence.profile_name "
> > "where sip_registrations.hostname='%q' and
> sip_registrations.profile_name='%q' "
> > "and sip_registrations.sip_user='%q' and "
> > "(sip_registrations.orig_server_host='%q' or
> sip_registrations.sub_host='%q' "
> > ")",
> > dh.status, dh.rpid, switch_str_nil(sub_call_id),
> > switch_sql_concat(), switch_sql_concat(),
> > mod_sofia_globals.hostname, profile->name, probe_euser, probe_host,
> probe_euser, probe_host, probe_host);
> 742,758d758
> <                                                                "left
> join sip_dialogs on "
> <
> "sip_dialogs.hostname = sip_registrations.hostname and
> sip_dialogs.profile_name = sip_registrations.profile_name and ("
> <
> "sip_dialogs.presence_id = sip_registrations.sip_user %q '@' %q
> sip_registrations.sub_host "
> <                                                                "or
> (sip_dialogs.sip_from_user = sip_registrations.sip_user "
> <                                                                "and
> sip_dialogs.sip_from_host = sip_registrations.sip_host)) "
> <
> <                                                                "left
> join sip_presence on "
> <
> "sip_presence.hostname=sip_registrations.hostname and "
> <
> "(sip_registrations.sip_user=sip_presence.sip_user and
> sip_registrations.orig_server_host=sip_presence.sip_host and "
> <
> "sip_registrations.profile_name=sip_presence.profile_name) "
> <                                                                "where
> sip_registrations.hostname='%q' and sip_registrations.profile_name='%q' and
> sip_dialogs.call_info_state != 'seized' "
> <                                                                "and
> sip_dialogs.presence_id='%q@%q' or (sip_registrations.sip_user='%q' and "
> <
> "(sip_registrations.orig_server_host='%q' or
> sip_registrations.sub_host='%q' "
> <                                                                "))",
> <
> dh.status, dh.rpid, switch_str_nil(sub_call_id),
> <
> switch_sql_concat(), switch_sql_concat(),
> <
> mod_sofia_globals.hostname, profile->name, probe_euser, probe_host,
> probe_euser, probe_host, probe_host);
>
>
> Am 01.04.21 um 18:40 schrieb Peter Steinbach:
>
> I think, I have a similar problem with registers (and presence) via
> XML-Curl. Freswitch is not sending a Curl request to the server then (as we
> grepped on the network interface).
>
> This happens under high load with a lot of presence involved. Suddenly
> Freeswitch internal profile stops responding and after a while (some
> minutes) it responds again. During that time, no register requests are
> answered at all. Only a few XML-Curl requests are sent to the server, but
> they do not contain the user info (which are needed for registrations) and
> are related to mod_voicemail.c etc.
>
> My debug log looks like this
>
> tport.c:3286 tport_tsend() tport_tsend(0x7f9c68005470) tpn =
> */xx.xxx.xxx.x:39672
> tport.c:4075 tport_resolve() tport_resolve addrinfo = xx.xxx.xxx.x:39672
> tport.c:4709 tport_by_addrinfo() tport_by_addrinfo(0x7f9c68005470): not
> found by name */xx.xxx.xxx.x:39672
> tport.c:3623 tport_vsend() tport_vsend(0x7f9c68005470): 1065 bytes of 1065
> to udp/xx.xxx.xxx.x:39672
> tport.c:3521 tport_send_msg() tport_vsend returned 1065
> nta.c:8348 outgoing_send() nta: sent NOTIFY (304240670) to
> */xx.xxx.xxx.x:39672
> tport.c:4189 tport_pend() tport_pend(0x7f9c68005470): pending
> 0x7f9997dcf880 for udp/xxx.xx.x.xxx:5060 (already 124658)
> nua_stack.c:569 nua_stack_signal() nua(0x7f9a0bd45ae0): recv signal
> r_notify
> nua_params.c:484 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:280 soa_clone() soa_clone(static::0x7f9c68001b90, 0x7f9c68001390,
> 0x7f9a0bd45ae0) called
> soa.c:403 soa_set_params() soa_set_params(static::0x7f9997dd1c90, ...)
> called
> soa.c:403 soa_set_params() soa_set_params(static::0x7f9997dd1c90, ...)
> called
> nta.c:4446 nta_leg_tcreate() nta_leg_tcreate(0x7f9997dcd870)
> nta.c:2694 nta_tpn_by_url() nta: selecting scheme sip
> tport.c:3286 tport_tsend() tport_tsend(0x7f9c68005470) tpn =
> udp/xx.xxx.xxx.x:58694
> tport.c:4075 tport_resolve() tport_resolve addrinfo = xx.xxx.xxx.x:58694
> tport.c:4709 tport_by_addrinfo() tport_by_addrinfo(0x7f9c68005470): not
> found by name udp/xx.xxx.xxx.x:58694
> tport.c:3623 tport_vsend() tport_vsend(0x7f9c68005470): 1114 bytes of 1114
> to udp/xx.xxx.xxx.x:58694
> tport.c:3521 tport_send_msg() tport_vsend returned 1114
> nta.c:8348 outgoing_send() nta: sent NOTIFY (304240671) to
> udp/xx.xxx.xxx.x:58694
> tport.c:4189 tport_pend() tport_pend(0x7f9c68005470): pending
> 0x7f9997dd2650 for udp/xxx.xx.x.xxx:5060 (already 124659)
> Looks like some event queue is stuck?
>
> /Peter
>
>
> Am 01.04.21 um 00:10 schrieb Stefan Kainz:
>
> Hi guys,
>
> Me again.
> It just happened again :)
> ( With extra debugging lines in my register.lua and a not found xml like
> you suggested if the result is empty )
>
> This time I turned up the logging to 9 while it was happening.
> This is what I found:
>
> While its happening, a Register request causes the following to be logged:
>
> nta.c:2880 agent_recv_request() nta: received REGISTER sip:xxxxx SIP/2.0
> (CSeq 73038)
> nta.c:3012 agent_recv_request() nta: REGISTER (73038) going to existing
> REGISTER transaction
> tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7fab1c0042b0): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7fab1c0042b0)
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fab1c0042b0) msg
> 0x7fab1d095280 from (udp/xxx.xxx.xxx.xxx:5060) has 809 bytes, veclen = 1
> tport.c:3023 tport_deliver() tport_deliver(0x7fab1c0042b0): msg
> 0x7fab1d095280 (809 bytes) from udp/xxx.xxx.xxx.xxx:5060/sip next=(nil)
>
> And that’s it. it doesn’t even execute my register.lua anymore, I just
> tries to update an existing Register transaction.
> When I start an ngrep on port 5060, all I see is the register, no
> unauthorised or ok response.
>
> While a Register Request after I restarted freeswitch and everything
> worked again looks like this:
>
> nta.c:2880 agent_recv_request() nta: received REGISTER sip:xxxxx SIP/2.0
> (CSeq 37884)
> nta.c:3085 agent_recv_request() nta: REGISTER (37884) going to a default
> leg
> nua_server.c:102 nua_stack_process_request() nua:
> nua_stack_process_request: entering
> nua_stack.c:899 nh_create() nua: nh_create: entering
> nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:280 soa_clone() soa_clone(static::0x7f8514001930, 0x7f8514001130,
> 0x7f85142225b0) called
> soa.c:403 soa_set_params() soa_set_params(static::0x7f85141f6f70, ...)
> called
> nua_stack.c:271 nua_stack_event() nua(0x7f85142225b0): event i_register
> 100 Trying
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2021-03-31 23:47:11.485911 [INFO] switch_cpp.cpp:1328 register.lua:
> Connect to database
> ...
>
> The last entry “Connect to database” is already a consoleLog from my
> register.lua.
> From here on out everything works great again …
>
> So, do you think it could still be true that one request to my
> register.lua holds up all subsequent requests?
>
> Thank you very much!
>
> Regards,
>
>
>
>
> On 31.03.2021, at 09:16, Stefan Kainz <s.kainz at wnt.at> wrote:
>
> Ok thanks!
>
> I will try to return a not found and add debug lines to the register.lua.
> I will let you know If that helped.
>
> Regards,
> Stefan
>
> On 30.03.2021, at 21:49, Bote Man <botelist at gmail.com> wrote:
>
> And add some logging lines to the Lua script so that it can tell you what
> is happening at each stage. Perhaps it is doing something that you don’t
> want it to do, or not doing something that you do want it to do?
>
> Bote
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Mircea Botoca-Huh
> *Sent:* Tuesday, 30 March, 2021 15:21
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Internal Interface suddenly freezes
>
> Hi,
>
> If you don't wish to serve an action from your Lua script, you must return
> not found.
>
> This is from confluence doc:
>
>
> If your LUA application receives a request and you don't wish to serve
> dialplan or like to fallback to plain XML dialplan, then you should return
> the following "not found" result.
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>
> <document type="freeswitch/xml">
>
>   <section name="result">
>
>     <result status="not found" />
>
>   </section>
>
> </document>
>
>
>
> Best regards,
> Mircea
>
> mar., 30 mar. 2021, 22:05 Stefan Kainz <s.kainz at wnt.at> a scris:
>
> Hi,
>
> The lua script is called on a directory request.
> Your know, autoload_configs, lua.conf.xml:
>
> <param name="xml-handler-script" value="register.lua"/>
> <param name="xml-handler-bindings" value="directory,dialplan”/>
>
> Maybe the dialplan in the xml-handler-bindings is doing something I don’t
> want …
>
> In the register.lua I do basic sanity checks, and of course a database
> query that returns the user-data.
> After that I generate the xml that gets sent to freeswitch.
>
> Yesterday I added cacheable=“120" to the <user> tag.
> Thats no solution of course, but it looks like when the data is in the
> cache the lua script doesn’t get called.
> So maybe the error won’t occur as often as before …
>
> But, I have an idea, maybe you can confirm this.
> With dialplan enabled in xml-handler-bindings, the lua script is also
> called on calls, not just on registrations.
> ( I think that’s necessary to authenticate the users on an outgoing call,
> please correct me if im wrong )
> In the register.lua I only handle directory-requests and action==user_call.
> All other requests to my register.lua are unhandled and therefore I don’t
> return any xml at all.
>
> Maybe I get a request to my register.lua that is not a directory-request
> and not a user_call and since I don’t return a xml, Sofia hangs.
>
> Does that sound plausible?
>
> Thank you in advance!
>
> Regards,
> Stefan
>
>
>
>
> On 29.03.2021, at 23:57, David Villasmil <david.villasmil.work at gmail.com>
> wrote:
>
> That was going to be my next question: what’s lua doing and at what point?
>
> On Mon, 29 Mar 2021 at 16:09, Stefan Kainz <s.kainz at wnt.at> wrote:
>
> I just found an issue on jira, where it seems someone had the same problem
> I have.
> https://freeswitch.org/jira/browse/FS-3328
> I don’t have mod_xml_curl enabled though.
>
> But knowing that sofia can handle only one register at a time and then
> blocking all subsequent Registers is a good starting point …
>
> Regards,
>
> *Von:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *Im
> Auftrag von *Bote Man
> *Gesendet:* Montag, 29. März 2021 15:45
> *An:* 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
> *Betreff:* Re: [Freeswitch-users] Internal Interface suddenly freezes
>
> The one common element is your Lua script.
>
> I am certainly no expert on script writing, but I have seen a number of
> problems on the mailing list over the years with scripts doing “too much”
> work during critical sections of the dialplan. Perhaps there is a race
> condition?
>
> Hope this helps.
>
>
> ---
> John Boteler
> BnC Group U.S.A.
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Stefan Kainz
> *Sent:* Monday, 29 March, 2021 07:49
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Internal Interface suddenly freezes
>
> Thank you for your answer!
>
> Hmm, I also tested it on two completely different servers ( no
> virtualization ) and the problem exists on both.
> Im also going to try it on a third server, also completely different, but
> I cant really image that this is a hardware-thing …
>
> We also have many freeswitch servers in production ( Exactly the same
> hardware as the server with the problem ).
> The only difference is that one of those servers handles registrations,
> and one doesn’t.
> The one handling the registrations has the problem, the other one doesn’t.
>
>
> Its really strange …
>
> Regards,
>
> *Von:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *Im
> Auftrag von *David Villasmil
> *Gesendet:* Montag, 29. März 2021 13:12
> *An:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Betreff:* Re: [Freeswitch-users] Internal Interface suddenly freezes
>
> That looks to be more on the hardware side than software. It’d be an
> extremely coincidence those versions and all those OS have some issue
> somewhere.
> Change hardware.
>
> On Sun, 28 Mar 2021 at 09:12, Stefan Kainz <s.kainz at wnt.at> wrote:
>
> Hi everbody,
>
> I have a little bit of a problem.
> Im using Version 1.10.3. ( but this problem also occurs on version 1.4.18 )
>
> Sometimes the internal Sofia interface just stops responding to SIP
> Requests.
> It sometimes happens once every day, and sometimes once a week.
> It happens at completely random times, like one day in the morning, and
> the next day in the middle of the night.
> The freeswitch.log gives me nothing, its like the Sofia interface was
> stopped.
>
> When I try to restart the interface with "sofia profile internal restart”
> nothing happens. The fs_cli just remains stuck with that command.
>
> The solution is to restart the freeswitch service.
>
> Sometimes when I recognise it too late, for example in the middle of the
> night, it seems like the problem solves itself after about 2 hours.
> The profile just starts working again, without somebody doing anything.
>
> I have checked a variety of things, including the firewall & fail2ban,
> network connection, made sure watchdog is disabled, and also tested it on
> different Debian-versions and freeswitch versions.
> It seems this problem occurs on every freeswitch version i have tested.
>
> The external-profile on the other hand, keeps working like nothing
> happened.
>
> Both Interfaces listen on the same network-device with a public ip.
> The only difference is, the internal profile uses a Lua file to handle
> registrations.
>
> Has anybody come across anything similar?
>
> Any help is much appreciated!
>
> 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
>
> --
> 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
>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time https://signalwire.community
>
> Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com
>
> Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<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/20211022/74dad8c7/attachment-0001.html>

From support at naud.io  Sat Oct 23 01:46:22 2021
From: support at naud.io (Support from NetworkedAudio LLC)
Date: Sat, 23 Oct 2021 01:46:22 +0000
Subject: [Freeswitch-users] Sending announcements to all callers
In-Reply-To: <SN4PR0701MB382296A8385542D2322A36A5A3BF9@SN4PR0701MB3822.namprd07.prod.outlook.com>
References: <SN4PR0701MB382296A8385542D2322A36A5A3BF9@SN4PR0701MB3822.namprd07.prod.outlook.com>
Message-ID: <SN4PR0701MB3822B9B40440C40EEDD17787A3819@SN4PR0701MB3822.namprd07.prod.outlook.com>

Can't seem to do this at the call level, so have set up an autocall to the eavesdrop when the conference is made.

Then when the local resources access the conference, they put the eavesdrop call on hold and then store the UUID of that call in a lookup with the conference name.
The hangup hook on those local resources checks the conference xml_list and sees how many local members are left, and if none takes the eavesdrop off hold.

No loopback, resources seem OK. I'd prefer to do it pre-conference, but will see how this goes.

Any advice still welcome.

________________________________
From: Support from NetworkedAudio LLC
Sent: Thursday, October 21, 2021 12:02 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Sending announcements to all callers

I have a system where a user dials in, get authenticated and placed into a standard conference. Other local resources get added to the conferences as needed, and that is working well.

For safety reasons, we need to play a single incoming PortAudio channel to all dialed in users.

If I do an auto call or originate I can add a deaf member to each conference which is a loopback to an eavesdrop. This is messy but works.

However, it uses a lot of resources and now my local people have the announcements twice, with delay.

What I want is just to send audio out to those dialing in.

I have tried sending eavesdropping just to the incoming call but it won’t let me then join a conference through the dialplan - processing stops at the eavesdrop.

I see broadcast but can only send tones or saved files.

Ideally the incoming external callers would hear the announcements and not the conferences themselves. I’d like to avoid loopback, too.

Does anyone have ideas, please?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211023/9f1a7863/attachment.html>

From clive at lansink.co.nz  Sun Oct 24 01:09:41 2021
From: clive at lansink.co.nz (Clive Lansink)
Date: Sun, 24 Oct 2021 01:09:41 +0000
Subject: [Freeswitch-users] PortAudio and codecs
Message-ID: <0100017cafd89be1-6f731606-c9d9-433f-ba2b-1ef59e274de6-000000@email.amazonses.com>

An embedded and charset-unspecified text was scrubbed...
Name: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211024/837b8c39/attachment.ksh>

From clive at lansink.co.nz  Sun Oct 24 19:31:55 2021
From: clive at lansink.co.nz (Clive Lansink)
Date: Sun, 24 Oct 2021 19:31:55 +0000
Subject: [Freeswitch-users] PortAudio and codecs
Message-ID: <0100017cb3c9bd78-83042e8c-3dbf-4ed1-985c-e104873df8a8-000000@email.amazonses.com>

An embedded and charset-unspecified text was scrubbed...
Name: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211024/0d763fdb/attachment.ksh>

From sdevoy at bizfocused.com  Mon Oct 25 16:13:56 2021
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Mon, 25 Oct 2021 16:13:56 +0000
Subject: Internal extension and external CELL phone - revisited
Message-ID: <BL0PR0102MB3425C0D6B09E06E9016DBB35BB839@BL0PR0102MB3425.prod.exchangelabs.com>

Hi,

With Avi's help I have the dial string corrected to:
Dial an internal extension. Delay 10 seconds, then dial external number (which happens to be a cell phone)
The goal is to set the call timeout to keep the voicemail in FreeSwitch. (For this discussion let's ignore the variability of time to make a connection to a cell phone.)

After 10 second delay, FreeSwitch dials the external number just fine.  Then the external leg gets a
switch_core_media.c:8980 sofia/external/14108089999 Set 2833 dtmf receive payload to 101
It immediately hangs up the internal extension with a
switch_ivr_originate.c:3716 Hangup sofia/internal/220 at 68.143.96.162:1458<mailto:sofia/internal/220 at 68.143.96.162:1458> [CS_CONSUME_MEDIA] [LOSE_RACE]

Note, the cell phone never gets answered by the recipient. Sadly, the call does go to the cell voicemail.

Here are my questions:
1 - Is that because the Cell Carrier has actually "answered" the call and is now routing it to the phone?
2 - Can I even go to voicemail on FS after that event or is the call "answered"?

Regards,
Sean
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211025/cff24b18/attachment.html>

From brian at freeswitch.com  Mon Oct 25 16:58:03 2021
From: brian at freeswitch.com (Brian West)
Date: Mon, 25 Oct 2021 11:58:03 -0500
Subject: [Freeswitch-users] Internal extension and external CELL phone -
	revisited
In-Reply-To: <mailman.4183.1635178471.1264.freeswitch-users@lists.freeswitch.org>
References: <mailman.4183.1635178471.1264.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAEJMVkA+sGN1tgevxQWP_c+j1YuuKq+QowvGqPLXOQ=fOZthDA@mail.gmail.com>

ignore_early_media=true will be required here.

On Mon, Oct 25, 2021 at 11:49 AM Sean Devoy via FreeSWITCH-users <
freeswitch-users at lists.freeswitch.org> wrote:

>
>
>
> ---------- Forwarded message ----------
> From: Sean Devoy <sdevoy at bizfocused.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Mon, 25 Oct 2021 16:13:56 +0000
> Subject: Internal extension and external CELL phone - revisited
>
> Hi,
>
>
>
> With Avi’s help I have the dial string corrected to:
>
> Dial an internal extension. Delay 10 seconds, then dial external number
> (which happens to be a cell phone)
>
> The goal is to set the call timeout to keep the voicemail in FreeSwitch.
> (For this discussion let’s ignore the variability of time to make a
> connection to a cell phone.)
>
>
>
> After 10 second delay, FreeSwitch dials the external number just fine.
> Then the external leg gets a
>
> switch_core_media.c:8980 sofia/external/14108089999 Set 2833 dtmf receive
> payload to 101
>
> It immediately hangs up the internal extension with a
>
> switch_ivr_originate.c:3716 Hangup sofia/internal/220 at 68.143.96.162:1458
> [CS_CONSUME_MEDIA] [LOSE_RACE]
>
>
>
> Note, the cell phone never gets answered by the recipient. Sadly, the call
> does go to the cell voicemail.
>
>
>
> Here are my questions:
>
> 1 – Is that because the Cell Carrier has actually “answered” the call and
> is now routing it to the phone?
>
> 2 – Can I even go to voicemail on FS after that event or is the call
> “answered”?
>
>
>
> Regards,
>
> Sean
>
>
>
> ---------- Forwarded message ----------
> From: Sean Devoy via FreeSWITCH-users <
> freeswitch-users at lists.freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Mon, 25 Oct 2021 09:49:37 -0700 (PDT)
> Subject: [Freeswitch-users] Internal extension and external CELL phone -
> revisited
> _________________________________________________________________________
>
> 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/20211025/ceede72e/attachment-0001.html>

From sdevoy at bizfocused.com  Mon Oct 25 19:19:44 2021
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Mon, 25 Oct 2021 19:19:44 +0000
Subject: [Freeswitch-users] Internal extension and external CELL phone -
 revisited
In-Reply-To: <CAEJMVkA+sGN1tgevxQWP_c+j1YuuKq+QowvGqPLXOQ=fOZthDA@mail.gmail.com>
References: <mailman.4183.1635178471.1264.freeswitch-users@lists.freeswitch.org>
 <CAEJMVkA+sGN1tgevxQWP_c+j1YuuKq+QowvGqPLXOQ=fOZthDA@mail.gmail.com>
Message-ID: <BL0PR0102MB34254F5D8D57A0F7AE95A853BB839@BL0PR0102MB3425.prod.exchangelabs.com>

Thank you for the quick response without the obvious RTFM. Unfortunately if I don't know what to search for finding the answer is hard.

I really like FS a LOT, but I love this forum.  
YOU GENTLEMEN and LADIES ROCK.

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Brian West
Sent: Monday, October 25, 2021 12:58 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Internal extension and external CELL phone - revisited

ignore_early_media=true will be required here.

On Mon, Oct 25, 2021 at 11:49 AM Sean Devoy via FreeSWITCH-users <mailto:freeswitch-users at lists.freeswitch.org> wrote:



---------- Forwarded message ----------
From: Sean Devoy <mailto:sdevoy at bizfocused.com>
To: FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org>
Cc: 
Bcc: 
Date: Mon, 25 Oct 2021 16:13:56 +0000
Subject: Internal extension and external CELL phone - revisited
Hi,
 
With Avi’s help I have the dial string corrected to:
Dial an internal extension. Delay 10 seconds, then dial external number (which happens to be a cell phone)
The goal is to set the call timeout to keep the voicemail in FreeSwitch. (For this discussion let’s ignore the variability of time to make a connection to a cell phone.)
 
After 10 second delay, FreeSwitch dials the external number just fine.  Then the external leg gets a 
        switch_core_media.c:8980 sofia/external/14108089999 Set 2833 dtmf receive payload to 101
It immediately hangs up the internal extension with a 
        switch_ivr_originate.c:3716 Hangup mailto:sofia/internal/220 at 68.143.96.162:1458 [CS_CONSUME_MEDIA] [LOSE_RACE]
 
Note, the cell phone never gets answered by the recipient. Sadly, the call does go to the cell voicemail.
 
Here are my questions:
1 – Is that because the Cell Carrier has actually “answered” the call and is now routing it to the phone?
2 – Can I even go to voicemail on FS after that event or is the call “answered”?
 
Regards,
Sean



---------- Forwarded message ----------
From: Sean Devoy via FreeSWITCH-users <mailto:freeswitch-users at lists.freeswitch.org>
To: FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org>
Cc: 
Bcc: 
Date: Mon, 25 Oct 2021 09:49:37 -0700 (PDT)
Subject: [Freeswitch-users] Internal extension and external CELL phone - revisited
_________________________________________________________________________

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
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
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 mailto:sales at freeswitch.com 
FreeSWITCH Solutions | https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g
Email: mailto:brian at freeswitch.com
Mobile: 918-424-9378
Website: https://www.freeswitch.com/


From vishalmpai at gmail.com  Tue Oct 26 18:02:54 2021
From: vishalmpai at gmail.com (Vishal Pai)
Date: Tue, 26 Oct 2021 23:32:54 +0530
Subject: [Freeswitch-users] AUDIO RTP REPORTS ERROR: [Remote Address Error!]
 -- [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
Message-ID: <CACOmXtBBR_o4o4BYggHNwERuC7OfcZm08U8Yppi2M=R2N8Xnhw@mail.gmail.com>

Hello Everyone

I am using webrtc client sip.js + freeswitch 1.10.2 installed from src.

Whenever the first call comes to sip.js it gets disconnected with error
AUDIO RTP REPORTS ERROR: [Remote Address Error!] --
[INCOMPATIBLE_DESTINATION]

Here is the log file attached.

Is there any bug in the freeswitch version ? Or am I using the wrong
configuration?

Note there is no NAT on the server, it DO droplet which is directly
connected to public IP.

Any help will be highly appreciated.

Thanks.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211026/045ace18/attachment.html>
-------------- next part --------------
2021-10-26 12:52:29.050821 [NOTICE] switch_channel.c:1118 New Channel sofia/internal/2l3s73da at ppcf1t3o2201.invalid [d0dc30cc-9ae7-4c04-b1d6-4917c2d279ec]
2021-10-26 12:52:29.050821 [DEBUG] mod_sofia.c:5089 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) State Change CS_NEW -> CS_INIT
2021-10-26 12:52:29.050821 [DEBUG] switch_core_state_machine.c:585 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) Running State Change CS_INIT (Cur 2 Tot 33)
2021-10-26 12:52:29.050821 [DEBUG] switch_core_state_machine.c:628 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) State INIT
2021-10-26 12:52:29.050821 [DEBUG] mod_sofia.c:93 sofia/internal/2l3s73da at ppcf1t3o2201.invalid SOFIA INIT
2021-10-26 12:52:29.050821 [DEBUG] sofia_glue.c:1587 sip:2l3s73da at 49.36.66.155:54529;transport=wss Setting proxy route to sofia/internal/2l3s73da at ppcf1t3o2201.invalid
2021-10-26 12:52:29.050821 [DEBUG] sofia_glue.c:1618 sofia/internal/2l3s73da at ppcf1t3o2201.invalid sending invite version: 1.10.2 -release 64bit
Local SDP:
v=0
o=FreeSWITCH 1635243287 1635243288 IN IP4 159.6X.1XX.1X
s=FreeSWITCH
c=IN IP4 159.6X.1XX.1X
t=0 0
a=msid-semantic: WMS SUEn7PzWk1vj1UlaPB66hNYKnESNmJO5
m=audio 27462 RTP/SAVPF 9 0 8 3 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fingerprint:sha-256 05:2D:ED:8F:CD:13:CE:A4:E2:7E:39:E3:56:52:D4:57:FF:4C:D6:6D:A6:34:B6:C0:30:AC:F5:FE:86:0A:1A:57
a=setup:actpass
a=rtcp-mux
a=rtcp:27462 IN IP4 159.6X.1XX.1X
a=ssrc:1166113573 cname:ujjsYJe8BAhsltt0
a=ssrc:1166113573 msid:SUEn7PzWk1vj1UlaPB66hNYKnESNmJO5 a0
a=ssrc:1166113573 mslabel:SUEn7PzWk1vj1UlaPB66hNYKnESNmJO5
a=ssrc:1166113573 label:SUEn7PzWk1vj1UlaPB66hNYKnESNmJO5a0
a=ice-ufrag:UpW23LDgflTsVyog
a=ice-pwd:Hyzv7Euz4CRfVb6GqSyju1pA
a=candidate:2385770488 1 udp 659136 159.6X.1XX.1X 27462 typ host generation 0
a=candidate:2385770488 2 udp 659136 159.6X.1XX.1X 27462 typ host generation 0
a=ptime:20
a=sendrecv

2021-10-26 12:52:29.050821 [DEBUG] switch_core_state_machine.c:40 sofia/internal/2l3s73da at ppcf1t3o2201.invalid Standard INIT
2021-10-26 12:52:29.050821 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) State Change CS_INIT -> CS_ROUTING
2021-10-26 12:52:29.050821 [DEBUG] switch_core_state_machine.c:628 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) State INIT going to sleep
2021-10-26 12:52:29.050821 [DEBUG] switch_core_state_machine.c:585 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) Running State Change CS_ROUTING (Cur 2 Tot 33)
2021-10-26 12:52:29.050821 [DEBUG] sofia.c:7301 Channel sofia/internal/2l3s73da at ppcf1t3o2201.invalid entering state [calling][0]
2021-10-26 12:52:29.050821 [DEBUG] switch_core_state_machine.c:644 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) State ROUTING
2021-10-26 12:52:29.050821 [DEBUG] mod_sofia.c:154 sofia/internal/2l3s73da at ppcf1t3o2201.invalid SOFIA ROUTING
2021-10-26 12:52:29.050821 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2021-10-26 12:52:29.050821 [DEBUG] switch_core_state_machine.c:644 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) State ROUTING going to sleep
2021-10-26 12:52:29.050821 [DEBUG] switch_core_state_machine.c:585 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 33)
2021-10-26 12:52:29.050821 [DEBUG] switch_core_state_machine.c:663 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) State CONSUME_MEDIA
2021-10-26 12:52:29.050821 [DEBUG] switch_core_state_machine.c:663 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) State CONSUME_MEDIA going to sleep
2021-10-26 12:52:29.270824 [DEBUG] sofia.c:7301 Channel sofia/internal/2l3s73da at ppcf1t3o2201.invalid entering state [proceeding][180]
2021-10-26 12:52:29.270824 [NOTICE] sofia.c:7412 Ring-Ready sofia/internal/2l3s73da at ppcf1t3o2201.invalid!
2021-10-26 12:52:29.270824 [DEBUG] switch_channel.c:3437 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) Callstate Change DOWN -> RINGING
2021-10-26 12:52:38.810821 [DEBUG] sofia.c:7301 Channel sofia/internal/2l3s73da at ppcf1t3o2201.invalid entering state [completing][200]
2021-10-26 12:52:38.810821 [DEBUG] sofia.c:7311 Remote SDP:
v=0
o=- 5763818288033074925 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS 7z7Ljk2XXKuNEXsvU3Esz2fbmVvl5URTXwlP
m=audio 50131 UDP/TLS/RTP/SAVPF 9 0 8 101
c=IN IP4 49.36.66.155
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=rtcp:9 IN IP4 192.0.2.63
a=candidate:840328171 1 udp 2113937151 d26fe240-6e5f-4d34-8488-39f093b59d67.local 50131 typ host generation 0 network-cost 999
a=candidate:2430225745 1 udp 2113939711 e31f962c-3822-4856-aa21-25478692ef23.local 50132 typ host generation 0 network-cost 999
a=candidate:842163049 1 udp 1677729535 49.36.66.155 50131 typ srflx raddr 192.0.2.63 rport 0 generation 0 network-cost 999
a=ice-ufrag:eyg7
a=ice-pwd:2S4J8PnNelLFAW6kaef2ZKEh
a=ice-options:trickle
a=fingerprint:sha-256 1C:3D:B2:DF:8D:EB:C3:1A:57:1F:1B:FF:41:EF:56:F6:2D:D4:A3:A6:69:C8:B0:49:B4:7B:39:82:22:BA:57:3E
a=setup:active
a=mid:0
a=rtcp-mux
a=ssrc:3396866038 cname:yLBRvE6bQhZN8vDr
a=ssrc:3396866038 msid:7z7Ljk2XXKuNEXsvU3Esz2fbmVvl5URTXwlP 64ccd886-3848-44a7-bd97-e474846a0969
a=ssrc:3396866038 mslabel:7z7Ljk2XXKuNEXsvU3Esz2fbmVvl5URTXwlP
a=ssrc:3396866038 label:64ccd886-3848-44a7-bd97-e474846a0969

2021-10-26 12:52:38.810821 [DEBUG] sofia.c:7301 Channel sofia/internal/2l3s73da at ppcf1t3o2201.invalid entering state [ready][200]
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:5508 Set telephone-event payload to 101 at 8000
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:3837 Set Codec sofia/internal/2l3s73da at ppcf1t3o2201.invalid G722/8000 20 ms 160 samples 64000 bits 1 channels
2021-10-26 12:52:38.810821 [DEBUG] switch_core_codec.c:111 sofia/internal/2l3s73da at ppcf1t3o2201.invalid Original read codec set to G722:9
2021-10-26 12:52:38.810821 [WARNING] switch_core_media.c:4252 NO candidate ACL defined, Defaulting to wan.auto
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:4282 Save audio Candidate cid: 1 proto: udp type: host addr: d26fe240-6e5f-4d34-8488-39f093b59d67.local:50131
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:4282 Save audio Candidate cid: 1 proto: udp type: host addr: e31f962c-3822-4856-aa21-25478692ef23.local:50132
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:4282 Save audio Candidate cid: 1 proto: udp type: srflx addr: 49.36.66.155:50131
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:4327 Searching for rtp candidate.
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:4336 Choose rtp candidate, index 0, d26fe240-6e5f-4d34-8488-39f093b59d67.local:50131
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:4102 sofia/internal/2l3s73da at ppcf1t3o2201.invalid choosing family v4
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:4347 Choose same candidate, index 0, for rtcp based on rtcp-mux attribute d26fe240-6e5f-4d34-8488-39f093b59d67.local:50131
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:4399 setting remote audio ice addr to index 0 d26fe240-6e5f-4d34-8488-39f093b59d67.local:50131 based on candidate
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:4434 Setting remote rtcp audio addr to d26fe240-6e5f-4d34-8488-39f093b59d67.local:50131 based on candidate
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:5851 Set telephone-event payload to 101 at 8000
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:5909 sofia/internal/2l3s73da at ppcf1t3o2201.invalid Set 2833 dtmf send payload to 101 recv payload to 101
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:8659 AUDIO RTP [sofia/internal/2l3s73da at ppcf1t3o2201.invalid] 159.6X.1XX.1X port 27462 -> d26fe240-6e5f-4d34-8488-39f093b59d67.local port 50131 codec: 9 ms: 20
2021-10-26 12:52:38.810821 [DEBUG] switch_rtp.c:4408 Starting timer [soft] 160 bytes per 20ms
2021-10-26 12:52:38.810821 [ERR] switch_core_media.c:9679 AUDIO RTP REPORTS ERROR: [Remote Address Error!]
2021-10-26 12:52:38.810821 [NOTICE] switch_core_media.c:9680 Hangup sofia/internal/2l3s73da at ppcf1t3o2201.invalid [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
2021-10-26 12:52:38.810821 [ERR] sofia.c:8481 RTP Error!
2021-10-26 12:52:38.810821 [DEBUG] sofia.c:1548 Channel is already hungup.
2021-10-26 12:52:38.810821 [DEBUG] switch_core_state_machine.c:585 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) Running State Change CS_HANGUP (Cur 2 Tot 33)
2021-10-26 12:52:38.810821 [DEBUG] switch_core_state_machine.c:848 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) Callstate Change RINGING -> HANGUP
2021-10-26 12:52:38.810821 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) State HANGUP
2021-10-26 12:52:38.810821 [DEBUG] mod_sofia.c:453 Channel sofia/internal/2l3s73da at ppcf1t3o2201.invalid hanging up, cause: INCOMPATIBLE_DESTINATION
2021-10-26 12:52:38.810821 [DEBUG] mod_sofia.c:507 Sending BYE to sofia/internal/2l3s73da at ppcf1t3o2201.invalid
2021-10-26 12:52:38.810821 [DEBUG] switch_core_state_machine.c:60 sofia/internal/2l3s73da at ppcf1t3o2201.invalid Standard HANGUP, cause: INCOMPATIBLE_DESTINATION
2021-10-26 12:52:38.810821 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) State HANGUP going to sleep
2021-10-26 12:52:38.810821 [DEBUG] switch_core_state_machine.c:620 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) State Change CS_HANGUP -> CS_REPORTING
2021-10-26 12:52:38.810821 [DEBUG] switch_core_state_machine.c:585 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) Running State Change CS_REPORTING (Cur 2 Tot 33)
2021-10-26 12:52:38.810821 [DEBUG] switch_core_state_machine.c:936 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) State REPORTING
2021-10-26 12:52:38.810821 [DEBUG] switch_ivr_originate.c:3952 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
2021-10-26 12:52:38.810821 [NOTICE] switch_ivr_originate.c:2955 Cannot create outgoing channel of type [user] cause: [INCOMPATIBLE_DESTINATION]
2021-10-26 12:52:38.810821 [DEBUG] switch_ivr_originate.c:3952 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
2021-10-26 12:52:38.830888 [DEBUG] switch_core_state_machine.c:174 sofia/internal/2l3s73da at ppcf1t3o2201.invalid Standard REPORTING, cause: INCOMPATIBLE_DESTINATION
2021-10-26 12:52:38.830888 [DEBUG] switch_core_state_machine.c:936 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) State REPORTING going to sleep
2021-10-26 12:52:38.830888 [DEBUG] switch_core_state_machine.c:611 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) State Change CS_REPORTING -> CS_DESTROY
2021-10-26 12:52:38.830888 [DEBUG] switch_core_session.c:1726 Session 33 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) Locked, Waiting on external entities
2021-10-26 12:52:38.830888 [NOTICE] switch_core_session.c:1744 Session 33 (sofia/internal/2l3s73da at ppcf1t3o2201.invalid) Ended
2021-10-26 12:52:38.830888 [NOTICE] switch_core_session.c:1748 Close Channel sofia/internal/2l3s73da at ppcf1t3o2201.invalid [CS_DESTROY]

From grcamauer at gmail.com  Tue Oct 26 20:10:02 2021
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Tue, 26 Oct 2021 17:10:02 -0300
Subject: [Freeswitch-users] AUDIO RTP REPORTS ERROR: [Remote Address
 Error!] -- [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
In-Reply-To: <CACOmXtBBR_o4o4BYggHNwERuC7OfcZm08U8Yppi2M=R2N8Xnhw@mail.gmail.com>
References: <CACOmXtBBR_o4o4BYggHNwERuC7OfcZm08U8Yppi2M=R2N8Xnhw@mail.gmail.com>
Message-ID: <CAPNTnnr=VL=om=QGCjwRwaUmcD0NctJn58-oDyU5esCsHd6rQQ@mail.gmail.com>

Vishal,

You should probably be using version 1.10.6.  In fact, version 1.10.7 was
announced recently.

Incompatible destination usually means that a compatible CODEC could not be
negotiated by both sides.  Do you have the correct codecs loaded?

Guillermo

On Tue, Oct 26, 2021 at 3:12 PM Vishal Pai <vishalmpai at gmail.com> wrote:

> Hello Everyone
>
> I am using webrtc client sip.js + freeswitch 1.10.2 installed from src.
>
> Whenever the first call comes to sip.js it gets disconnected with error
> AUDIO RTP REPORTS ERROR: [Remote Address Error!] --
> [INCOMPATIBLE_DESTINATION]
>
> Here is the log file attached.
>
> Is there any bug in the freeswitch version ? Or am I using the wrong
> configuration?
>
> Note there is no NAT on the server, it DO droplet which is directly
> connected to public IP.
>
> Any help will be highly appreciated.
>
> 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



-- 
Guillermo Ruiz Camauer
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211026/5a35b89b/attachment-0001.html>

From gregor at infomedia.si  Tue Oct 26 21:16:17 2021
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 26 Oct 2021 23:16:17 +0200
Subject: [Freeswitch-users] Buld freeswitch - Visual Studio
Message-ID: <CAC7ttFEbLSWF-MRvnPaYBkn8uGWpN5gv087EhzYDwQO=p6cDZg@mail.gmail.com>

Hi!

I wanted to build the latest version of FS in Visual Studio 2017.

I get following errors:
Error MSB4044 The "SwitchVersionTask" task was not given a value for the
required parameter "revision". FreeSwitchCoreLib
C:\freeswitch-1.10.7\w32\switch_version.props 128

Error The "SignFile" task was not given a value for the required parameter
"SigningTarget". Setup.2017
C:\freeswitch-1.10.7\w32\Setup\Setup.2017.wixproj 1025

Since FreeSwitchCoreLib is not built, I got a lot of other errors.

Can someone help me with this?

BR, Gregor
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211026/6196c72c/attachment.html>

From covici at ccs.covici.com  Wed Oct 27 14:47:52 2021
From: covici at ccs.covici.com (John Covici)
Date: Wed, 27 Oct 2021 10:47:52 -0400
Subject: [Freeswitch-users] problems compiling 1.10.7
Message-ID: <m3ee86r0br.wl-covici@ccs.covici.com>

Hi.  I am compiling freeswitch under gentoo and getting the following
error:

/bin/sh /usr/src/freeswitch/libs/apr/libtool --silent --mode=compile
gcc -g -O2 -pthread  -g -O2  -DHAVE_CONFIG_H -D_REENTRANT
-D_GNU_SOURCE   -I./include
-I/usr/src/freeswitch/libs/apr/include/arch/unix -I./include/arch/unix
-I/usr/src/fr\eeswitch/libs/apr/include  -o threadproc/unix/signals.lo
-c threadproc/unix/signals.c && touch threadproc/unix/signals.lo
In file included from ./include/apr_pools.h:39,
                 from
			  /usr/src/freeswitch/libs/apr/include/arch/unix/../apr_private_common.h:24,
			                   from
						    /usr/src/freeswitch/libs/apr/include/arch/unix/apr_private.h:881,
						                     from
										 /usr/src/freeswitch/libs/apr/include/arch/unix/apr_arch_threadproc.h:18,
										                  from
													   threadproc/unix/signals.c:18:
													   ./include/apr_general.h:33:
													   warning:
													   "NSIG"
													   redefined
													      33
														 |
														 #define
                                                                        NSIG
														  __DARWIN_NSIG
														        |
															   In
															   file
															   included
															   from
															   /usr/include/sys/wait.h:36,
															                    from
															   ./include/apr.h:131,
															                    from
															   /usr/src/freeswitch/libs/apr/include/arch/unix/apr_arch_threadproc.h:17,
															                    from
															   threadproc/unix/signals.c:18:
															   /usr/include/signal.h:184:
															   note:
															   this
															   is
															   the
															   location
															   of
															   the
															   previous
															   definition
															     184
															   |
															   # define
                                                                                NSIG
																_NSIG
																      |
																	 In
																	 file
																	 included
																	 from
																	 ./include/apr_pools.h:39,
																	                  from
																	 /usr/src/freeswitch/libs/apr/include/arch/unix/../apr_private_common.h:24,
																	                  from
																	 /usr/src/freeswitch/libs/apr/include/arch/unix/apr_private.h:881,
																	                  from
																	 /usr/src/freeswitch/libs/apr/include/arch/unix/apr_arch_threadproc.h:18,
																	                  from
																	 threadproc/unix/signals.c:18:
																	 ./include/apr_general.h:33:14:
																	 error:
																	 `__DARWIN_NSIG'
																	 undeclared
																	 here
																	 (not
																	 in
																	 a
																	 function)
																	    33
																	 |
																	 #define
                                                                                       NSIG
																	  __DARWIN_NSIG
																	        |
																		   ^~~~~~~~~~~~~
																		   threadproc/unix/signals.c:127:20:
																		   note:
																		   in
																		   expansion
																		   of
																		   macro
																		   `NSIG'
																		     127
																		   |
																		   #define
                                                                                              APR_NUMSIG
																		    NSIG
																		          |
																				^~~~
																				threadproc/unix/signals.c:136:39:
																				note:
																				in
																				expansion
																				of
																				macro
																				`APR_NUMSIG'
																				  136
																				|
																				static
																				const
																				char
																				*signal_description[APR_NUMSIG];
																				      |
																				^~~~~~~~~~
																				make[2]:
																				***
																				[/usr/src/freeswitch/libs/apr/build/apr_rules.mk:201:
																				threadproc/unix/signals.lo]
																				Error
																				1

Anyone have a clue?

Thanks in advance for any suggestions.

-- 
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 abdirahman.osm at gmail.com  Wed Oct 27 15:32:48 2021
From: abdirahman.osm at gmail.com (Abdirahman Osman)
Date: Wed, 27 Oct 2021 11:32:48 -0400
Subject: [Freeswitch-users] Postgress BDR insert failed for "complete" table
Message-ID: <CAEjtZA2t+wLpGN5O1z7ba5jQktYqO7=8K8RFRoBNG+phtQvy-w@mail.gmail.com>

Hello,

I am using postgres  with BDR bidirectional replication which requires
primary keys for the freeswitch tables.

 So far so good , except freeswitch is complaining about one table here is
the error

2021-10-26 23:40:47.772796 95.47% [DEBUG] mod_pgsql.c:784 Query (insert
into complete values (0,'hash_remote', 'list', '', '', '', '', '', '', '',
'',  'dagobah-api-gw-1-2')) returned PGRES_FATAL_ERROR
2021-10-26 23:40:47.772796 95.47% [ERR] mod_pgsql.c:551 Error executing
query:
ERROR:  column "complete_uuid" is of type uuid but expression is of type
integer
LINE 1: insert into complete values (0,'hash_remote', 'list', '', ''...

What I noticed is that freeswitch is doing an INSERT INTO command with no
column names specified. I checked the source code and found this to be the
only case for the "complete" table whereas the other tables when INSERT
INTO is used column names are specified.

Here is the structure of my  complete table with the new column
complete_uuid as primary key.

CREATE TABLE :myschema.complete (
complete_uuid uuid DEFAULT public.gen_random_uuid() PRIMARY KEY,
sticky  INTEGER,
a1  VARCHAR(128),
a2  VARCHAR(128),
a3  VARCHAR(128),
a4  VARCHAR(128),
a5  VARCHAR(128),
a6  VARCHAR(128),
a7  VARCHAR(128),
a8  VARCHAR(128),
a9  VARCHAR(128),
a10 VARCHAR(128),
hostname VARCHAR(256)
        );

Any suggestions?

Thanks

Abdirahman A. Osman
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211027/9b93c5cf/attachment.html>

From jprangi at gmail.com  Fri Oct 22 17:17:55 2021
From: jprangi at gmail.com (Jai Rangi)
Date: Fri, 22 Oct 2021 10:17:55 -0700
Subject: [Freeswitch-users] Issue with Enterprise originate and 487
	"ORIGINATOR_CANCEL".
In-Reply-To: <938b5330-fe6a-d155-37ce-3260c69a7bc1@telefaks.de>
References: <CAPfjz3hJNH4rhbeMb3LkboN2NC3VQoW890cVr8i4UJbdRTjgkg@mail.gmail.com>
 <26f9382e-1c6c-6da0-0b49-3db8633668c2@telefaks.de>
 <938b5330-fe6a-d155-37ce-3260c69a7bc1@telefaks.de>
Message-ID: <CAPfjz3jLeub7A0aA1Y8FNSoPBOTch9MzFD6iuNMX=RkCs9eJ6Q@mail.gmail.com>

Thank you Peter and Christian.

On Fri, Oct 22, 2021 at 7:19 AM Peter Steinbach <lists at telefaks.de> wrote:

> Just to mention: The patch was developed by Christian Berger from our
> client Foncloud. Thank you Christian!
>
> Best regards
> Peter
>
>
> Am 21.10.21 um 17:20 schrieb Peter Steinbach:
>
> Hello,
>
>
> this is a long lasting issue I firstly posted in 2017 and lastly saw in
> 2019 in this thread, and I am happy to announce, that we have found a
> solution for this.
>
> Here is the patch, which works pretty well in our environment
>
>
> ===================================================================================================
>
> --- freeswitch_orig/src/switch_ivr_originate.c    2021-10-20
> 12:51:04.863328041 +0200
> +++ freeswitch/src/switch_ivr_originate.c    2021-10-20 12:54:15.764616280
> +0200
> @@ -1789,7 +1789,13 @@
>          if (cancel_cause && *cancel_cause > 0) {
>              handles[i].cancel_cause = *cancel_cause;
>          } else {
> -            handles[i].cancel_cause = SWITCH_CAUSE_LOSE_RACE;
> +            if ((hp!=NULL) && (hp->cause==SWITCH_CAUSE_SUCCESS)) { //Was
> this call taken by another destination?
> +                //Yes, the race was lost
> +                handles[i].cancel_cause = SWITCH_CAUSE_LOSE_RACE;
> +            } else {
> +                //No, something else happened, probably Originator
> Cancel
> +                handles[i].cancel_cause = SWITCH_CAUSE_ORIGINATOR_CANCEL;
> +            }
>          }
>      }
>
> ===================================================================================================
>
>
> Can someone retest this and check this in?
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
> Internet: www.telefaks.de
>
> Am 22.05.19 um 21:14 schrieb Jai Rangi:
>
> All,
> Running into a small issue with group calling, when the original caller
> hang up, phones dont show that call as missed call, cause of wrong reason
> code from freeswitch.
> Scenario:
>
>    - Caller -> FS (Groupcall) to 101 and 102
>    - Phones ring
>    - Caller hangup
>    - Phones should see missed calls,
>    - But free-switch send cancel with Reason: SIP;cause=200;text="Call
>    completed elsewhere", where is should send Reason:
>    SIP;cause=487;text="ORIGINATOR_CANCEL".
>
> Here is what my bridge data.
>
> bridge(<all_threads=true,absolute_codec_string=PCMU,PCMA,G729,group_confirm_file=playback
> /usr/local/freeswitch/sound/ivr/1443458342.wav,group_confirm_key=exec>{sip_cid_type=rpid,ignore_early_media=true,originate_timeout=30}[leg_timeout=20
> ]user/101 at 192.168.5.50:_:[ignore_early_media=true,leg_timeout=20 ]user/
> 102 at 192.168.5.50)
>
> If I send
> bridge(<all_threads=true,absolute_codec_string=PCMU,PCMA,G729,group_confirm_file=playback
> /usr/local/freeswitch/sound/ivr/1443458342.wav,group_confirm_key=exec>{sip_cid_type=rpid,ignore_early_media=true,originate_timeout=30}[leg_timeout=20
> ]user/101 at 192.168.5.50
>
> Everything works fine.
> I see some old posts but no resolution.
> Freeswitch version on Debian 8.9
> FreeSWITCH (Version 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit)
>
> Any idea or suggestion for workaround?
>
> Jai
>
>
>
>
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time https://signalwire.community
>
> Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com
>
> Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211022/13707839/attachment-0001.html>

From mrsuperduper415 at gmail.com  Tue Oct 26 03:12:49 2021
From: mrsuperduper415 at gmail.com (Jimmy Simpson)
Date: Mon, 25 Oct 2021 20:12:49 -0700
Subject: [Freeswitch-users] disable video from being automatically added to
	sdp
Message-ID: <CAGzVJPZrbfhn-XRZViBEZFJhVYWD5hEK+z3wAh1kMkv+sgDtiQ@mail.gmail.com>

I can not seem to disable and stop Freeswitch from adding the m=video to
the SDP:

m=video 17076 RTP/AVP 104 105

Is there a channel variable I can set to disable all Video.

The extra m=video keeps getting added.

I only need audio only. This takes up an extra 3 ports that I do not want
to take up.

--------- in INVITE - From: udp:1.1.1.1:5060 To: udp:2.2.2.2:5060 ------
where 1.1.1.1 is the freeswitch, and 2.2.2.2 is the proxy

INVITE sip:107 at testsystem.example-domain.com SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKX2j1QgS2cepaK
Route: <sip:testsystem.example-domain.com:5060>
Max-Forwards: 67
From: "test10first test10last" <sip:106 at testsystem.example-domain.com
>;tag=yDttpta215tXc
To: <sip:107 at testsystem.example-domain.com>
Call-ID: 6bb53aa6-b0aa-123a-8c82-0050568c1d26
CSeq: 43054544 INVITE
Contact: <sip:mod_sofia at 1.1.1.1:5060>
User-Agent: Freeswitch
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 991
X-Type: audio
X-FS-Support: update_display,send_info
Remote-Party-ID: "test10first test10last" <
sip:106 at testsystem.example-domain.com>;party=calling;screen=yes;privacy=off

v=0
o=Cisco9 1635189532 1635189533 IN IP4 1.1.1.1
s=Cisco9
c=IN IP4 1.1.1.1
t=0 0
m=audio 27140 RTP/AVP 102 18 103 9 0 8 106 101
a=rtpmap:102 opus/48000/2
a=fmtp:102 useinbandfec=1; maxaveragebitrate=14000; maxplaybackrate=8000;
sprop-maxcapturerate=8000; ptime=20; minptime=10; maxptime=40
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:103 opus/48000/2
a=fmtp:103 useinbandfec=1; maxaveragebitrate=14000; maxplaybackrate=8000;
sprop-maxcapturerate=8000; ptime=20; minptime=10; maxptime=120
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 telephone-event/48000
a=fmtp:106 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
m=video 17076 RTP/AVP 104 105
b=AS:3072
a=rtpmap:104 VP9/90000
a=rtpmap:105 VP8/90000
a=inactive
a=rtcp-fb:104 ccm fir
a=rtcp-fb:104 ccm tmmbr
a=rtcp-fb:104 nack
a=rtcp-fb:104 nack pli
a=rtcp-fb:105 ccm fir
a=rtcp-fb:105 ccm tmmbr
a=rtcp-fb:105 nack
a=rtcp-fb:105 nack pli
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211025/a55602b2/attachment-0001.html>

From uppuluriaditya at gmail.com  Tue Oct 26 10:01:33 2021
From: uppuluriaditya at gmail.com (Aditya Uppuluri)
Date: Tue, 26 Oct 2021 15:31:33 +0530
Subject: [Freeswitch-users] Bridging a channel via Push Notifcation
Message-ID: <CAKWpYbm+2FPrdXZ9C1oq8cZFDCNWca9OOyvvxnFTLs6RmED0cQ@mail.gmail.com>

Hello All,
Thank you so much for the amazing software. I am new to freeswitch and
trying understand how to do the following use case (mobiles):

1. Call is received from LinPhone to User-B
2. Freeswitch (esl code) checks if User-B is a registered user or not
3. If a registered user, forward the call
4. If not a registered user, then send a push notification to the user-B's
mobile.
5. Freeswitch parks the call from LinPhone user
6. User-B's mobile gets push notification and get's a call screen
7. User-B accepts the call (Actually, he will send an INVITE with the data
received in the payload)
8. Freeswitch receives the invite and should bridge the Linphone user with
User-B

I am able to perform all the operations except the last one (Bridging the
call). If I try to use `uuid_bridge`, it says at least one call must be
answered. (*Neither channel is answered, cannot bridge them*.
)

May I know how to bridge these two legs in freeswitch?

FYI: I have the following park-only dialplan and controlling everything
from esl

<extension name="fspark">
<condition field="destination_number" expression="^(.*)$">
<action application="set" data="bypass_media=true"/>
<action application="set" data="bypass_media_after_bridge=true"/>
<action application="park"/>
</condition>
</extension>

Thanks in advance

Regards,
Aditya
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211026/8dd0ed5f/attachment-0001.html>

From uppuluriaditya at gmail.com  Wed Oct 27 16:19:05 2021
From: uppuluriaditya at gmail.com (Aditya Uppuluri)
Date: Wed, 27 Oct 2021 21:49:05 +0530
Subject: [Freeswitch-users] Bridging a channel via Push Notifcation
In-Reply-To: <CAKWpYbm+2FPrdXZ9C1oq8cZFDCNWca9OOyvvxnFTLs6RmED0cQ@mail.gmail.com>
References: <CAKWpYbm+2FPrdXZ9C1oq8cZFDCNWca9OOyvvxnFTLs6RmED0cQ@mail.gmail.com>
Message-ID: <CAKWpYbmc0OvV0h1YU_fixfXjUMb7bpo4EXxkRQVCYhaczUoaZw@mail.gmail.com>

Hello Team,
Does anyone know how to do this?

Thank you in advance


Regards,
Aditya


On Tue, Oct 26, 2021 at 3:31 PM Aditya Uppuluri <uppuluriaditya at gmail.com>
wrote:

> Hello All,
> Thank you so much for the amazing software. I am new to freeswitch and
> trying understand how to do the following use case (mobiles):
>
> 1. Call is received from LinPhone to User-B
> 2. Freeswitch (esl code) checks if User-B is a registered user or not
> 3. If a registered user, forward the call
> 4. If not a registered user, then send a push notification to the user-B's
> mobile.
> 5. Freeswitch parks the call from LinPhone user
> 6. User-B's mobile gets push notification and get's a call screen
> 7. User-B accepts the call (Actually, he will send an INVITE with the data
> received in the payload)
> 8. Freeswitch receives the invite and should bridge the Linphone user with
> User-B
>
> I am able to perform all the operations except the last one (Bridging the
> call). If I try to use `uuid_bridge`, it says at least one call must be
> answered. (*Neither channel is answered, cannot bridge them*.
> )
>
> May I know how to bridge these two legs in freeswitch?
>
> FYI: I have the following park-only dialplan and controlling everything
> from esl
>
> <extension name="fspark">
> <condition field="destination_number" expression="^(.*)$">
> <action application="set" data="bypass_media=true"/>
> <action application="set" data="bypass_media_after_bridge=true"/>
> <action application="park"/>
> </condition>
> </extension>
>
> Thanks in advance
>
> Regards,
> Aditya
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211027/1ae3717c/attachment-0001.html>

From rahman.duran at erzurum.edu.tr  Mon Oct 25 06:52:59 2021
From: rahman.duran at erzurum.edu.tr (Rahman Duran)
Date: Mon, 25 Oct 2021 09:52:59 +0300
Subject: [Freeswitch-users] Codec problem,
	Freeswitch does not regard absolute_codec_string?
In-Reply-To: <CAEMJzSxTL3sgQR6830P8LA+ZARSf8SBNDN1mWg+G+c102ajx7w@mail.gmail.com>
References: <CAEMJzSxTL3sgQR6830P8LA+ZARSf8SBNDN1mWg+G+c102ajx7w@mail.gmail.com>
Message-ID: <CAEMJzSz9ujt0iK7vG5MMKJh8xmsanN92agrq3pVwuPRv+qwxZA@mail.gmail.com>

Hi,

I am struggling tuning Freeswitch for late codec negotiation. Here is local
profile settings:

<param name="inbound-late-negotiation" value="true"/>
<param name="inbound-codec-prefs" value="G722,PCMA"/>
<param name="outbound-codec-prefs" value="G722,PCMA"/>
<param name="inbound-codec-negotiation" value="greedy"/>

As you see I enabled late-negotiation and set codecs prefs to allow G722
and PCMA. My goal is to use G722 if both sides support it and fall back to
PCMA if G722 is not an option for A leg or B leg.

Here is the local dial plan config where I bridge calls:

    <extension name="local_accounts_freeswitch">
      <condition regex="all">
        <regex field="destination_number" expression="^[0-9]{4}$"/>
        <regex field="${local_user_exists}"
expression="^true$|^TRUE$|^True$"/>
        <regex field="${santral_kontrol}" expression="^freeswitch$"/>
        <action application="set" inline="true"
data="call_pickup_group=${user_data(${destination_number}@${domain_name}
var call_pickup_group)}"/>
        <action application="hash"
data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/>
        <action application="hash"
data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/>
        <action application="hash"
data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/>
        <action application="log" data="ep_codec_string --
${ep_codec_string}" />
        <action application="set" data="inherit_codec=true"/>
        <action application="export"
data="nolocal:absolute_codec_string='${ep_codec_string}'"/>
        <action application="set_profile_var"
data="callee_id_name=${user_data(${destination_number}@${domain_name} var
effective_caller_id_name)}"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="call_timeout=30"/>
        <action application="set" data="execute_on_answer=sched_hangup
+21600 alloted_timeout" />
        <action application="bridge"
data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name}
var effective_caller_id_name)}}user/${destination_number}@${domain_name}"/>
        <action application="execute_extension"
data="hata-${originate_disposition} XML hata_anonslari"/>
      </condition>
    </extension>

So I export absolute_codec_string to B leg using ep_codec_string. When I
test a call from 2730->2110, the call fails with
"incompetible_destination".  Call logs here:
https://pastebin.freeswitch.org/view/b2978e79 The 2730 is a Cisco phone
with "G722,PCMU,PCMA,G729,L16,iLBC" codec support. The 2110 is a Samsung
phone that only supports "PCMA,PCMU,G729" codecs. As you see cisco invites
with all codecs it supports and freeswitch sets absolute_codec_string with
"G722,PCMA" as expected from the config. But as you see in the call logs
Freeswitch only sends "G722" in SDP to Samsung phone so Samsung rejects the
call.

Why does Freeswitch only send G722? How can I fix this? What am I doing
wrong?

P.S. using debian packages: FreeSWITCH Version
1.10.6-release-18-1ff9d0a60e~64bit (-release-18-1ff9d0a60e 64bit)

Regards,

Rahman Duran
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211025/7785fb23/attachment.html>

From Antony.Stone at freeswitch.open.source.it  Fri Oct 22 13:41:33 2021
From: Antony.Stone at freeswitch.open.source.it (Antony Stone)
Date: Fri, 22 Oct 2021 15:41:33 +0200
Subject: [Freeswitch-users] Log dialplan actions?
In-Reply-To: <CAFGRPVqNveFrvnsGK2Yv3FjPeawb+C++f6CG+4_Ss8VUU36h4w@mail.gmail.com>
References: <202110201114.36329.Antony.Stone@freeswitch.open.source.it>
 <CABaNFCaO-kN8ARfrx+aEp+YiqR+pn9XLFYj1+_pBZy++KsGGVQ@mail.gmail.com>
 <CAFGRPVqNveFrvnsGK2Yv3FjPeawb+C++f6CG+4_Ss8VUU36h4w@mail.gmail.com>
Message-ID: <202110221541.33766.Antony.Stone@freeswitch.open.source.it>

On Friday 22 October 2021 at 13:25:46, David Villasmil wrote:

> Go to switch.conf.xml and set the logging to debug and to the same on
> console.Conf.xml.
> 
> You get a lot of stuff since it’s on debug, but you will see the dialplan
> being executed step by step.

Thanks; I think I'm getting the hang of this :)

Antony.

-- 
I don't know, maybe if we all waited then cosmic rays would write all our 
software for us. Of course it might take a while.

 - Ron Minnich, Los Alamos National Laboratory

                                                   Please reply to the list;
                                                         please *don't* CC me.


From rahman.duran at erzurum.edu.tr  Tue Oct 26 06:14:00 2021
From: rahman.duran at erzurum.edu.tr (Rahman Duran)
Date: Tue, 26 Oct 2021 09:14:00 +0300
Subject: [Freeswitch-users] Overriding hangup cause in CDR logs
Message-ID: <CAEMJzSy3t5gw0ccwUmG9XyA7no9g-WbiVpeya1G5a0DfW=3LRg@mail.gmail.com>

Hi,

I am using announcements for fail hungup causes like busy, no_answer etc.
Here is my dial plan

<extension name="local_accounts_freeswitch">
      <condition regex="all">
        <regex field="destination_number" expression="^[0-9]{4}$"/>
        <regex field="${local_user_exists}"
expression="^true$|^TRUE$|^True$"/>
        <regex field="${santral_kontrol}" expression="^freeswitch$"/>
        <action application="set" inline="true"
data="call_pickup_group=${user_data(${destination_number}@${domain_name}
var call_pickup_group)}"/>
        <action application="hash"
data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/>
        <action application="hash"
data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/>
        <action application="hash"
data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/>
        <action application="log" data="ep_codec_string --
${ep_codec_string}" />
        <action application="set" data="inherit_codec=true"/>
        <action application="export"
data="nolocal:absolute_codec_string=${ep_codec_string}"/>
        <action application="set_profile_var"
data="callee_id_name=${user_data(${destination_number}@${domain_name} var
effective_caller_id_name)}"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="call_timeout=30"/>
        <action application="set" data="execute_on_answer=sched_hangup
+21600 alloted_timeout" />
        <action application="bridge"
data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name}
var effective_caller_id_name)}}user/${destination_number}@${domain_name}"/>
        <action application="execute_extension"
data="hata-${originate_disposition} XML hata_anonslari"/>
      </condition>
    </extension>

And here is announcements context that  I handle hangup causes:

<include>
  <!-- Burada iç çağrılarda alınan hata durumlarının anonsları işlenecek.
    -->
  <context name="hata_anonslari">
    <!-- yerel dahili  -->
    <extension name="error-causes-user_busy">
      <condition field="destination_number" expression="^hata-USER_BUSY$">
        <action application="export" data="cdr_hata_anonsu=USER_BUSY"/>
        <action application="pre_answer"/>
        <action application="playback"
data="$${anons_dosya_yolu}/user_busy.wav"/>
        <action application="sleep" data="1000"/>
        <action application="hangup" data="${originate_disposition}"/>
      </condition>
    </extension>
    <extension name="error-causes-no_answer">
      <condition field="destination_number" expression="^hata-NO_ANSWER$">
        <action application="export" data="cdr_hata_anonsu=NO_ANSWER"/>
        <action application="pre_answer"/>
        <action application="playback"
data="$${anons_dosya_yolu}/no_answer.wav"/>
        <action application="sleep" data="1000"/>
        <action application="hangup" data="${originate_disposition}"/>
      </condition>
    </extension>
    <extension name="error-causes-user_not_registered">
      <condition field="destination_number"
expression="^hata-USER_NOT_REGISTERED$">
        <action application="export"
data="cdr_hata_anonsu=USER_NOT_REGISTERED"/>
        <action application="pre_answer"/>
        <action application="playback"
data="$${anons_dosya_yolu}/user_not_registered.wav"/>
        <action application="sleep" data="1000"/>
        <action application="hangup" data="${originate_disposition}"/>
      </condition>
    </extension>
    <extension name="error-causes-any">
      <condition field="destination_number" expression="^hata-(.*)$">
        <action application="export"
data="cdr_hata_anonsu=${originate_disposition}"/>
        <action application="hangup" data="${originate_disposition}"/>
      </condition>
    </extension>
  <!-- context hata_anonslari sonu -->
  </context>
</include>


The problem is if the calling leg does not listen the announcement to the
end and hangup, CDR logs shows "Originator Cancel" as hangup cause. As I
already know the real hangup cause, how can I override the CDR hangup cause
with the real one? For now I am setting another variable (cdr_hata_anonsu)
and added it to CDR logs, but if possible I want to fix this with a real
solution.

Regards,

Rahman Duran
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211026/095c27ed/attachment.html>

From n.borisenkov at zonatelecom.ru  Wed Oct 27 07:29:09 2021
From: n.borisenkov at zonatelecom.ru (=?UTF-8?B?0J3QuNC60LjRgtCwINCR0L7RgNC40YHQtdC90LrQvtCy?=)
Date: Wed, 27 Oct 2021 10:29:09 +0300
Subject: [Freeswitch-users] AUDIO RTP REPORTS ERROR: [Remote Address
 Error!] -- [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
In-Reply-To: <CACOmXtBBR_o4o4BYggHNwERuC7OfcZm08U8Yppi2M=R2N8Xnhw@mail.gmail.com>
References: <CACOmXtBBR_o4o4BYggHNwERuC7OfcZm08U8Yppi2M=R2N8Xnhw@mail.gmail.com>
Message-ID: <04fae375-4b15-f714-e379-e32cc28e1256@zonatelecom.ru>

2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:4336 Choose rtp 
candidate, index 0, d26fe240-6e5f-4d34-8488-39f093b59d67.local:50131
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:4102 
sofia/internal/2l3s73da at ppcf1t3o2201.invalid choosing family v4
2021-10-26 12:52:38.810821 [DEBUG] switch_core_media.c:8659 AUDIO RTP 
[sofia/internal/2l3s73da at ppcf1t3o2201.invalid] 159.6X.1XX.1X port 27462 
-> d26fe240-6e5f-4d34-8488-39f093b59d67.local port 50131 codec: 9 ms: 20
2021-10-26 12:52:38.810821 [ERR] switch_core_media.c:9679 AUDIO RTP 
REPORTS ERROR: [Remote Address Error!]

Try to install the latest freeswitch. 
d26fe240-6e5f-4d34-8488-39f093b59d67.local - This value is the Multicast 
DNS (mDNS ) name ( 
https://rtcweb-wg.github.io/mdns-ice-candidates/draft-ietf-rtcweb-mdns-ice-candidates.html 
) . The server must translate this name into an IP address when 
receiving an mDNS address. Most likely the resolution of mdns addresses 
does not work on the local machine. Is avahi daemon or similar software 
running?



26.10.2021 21:02, Vishal Pai пишет:
> Hello Everyone
>
> I am using webrtc client sip.js + freeswitch 1.10.2 installed from src.
>
> Whenever the first call comes to sip.js it gets disconnected with 
> error AUDIO RTP REPORTS ERROR: [Remote Address Error!] -- 
> [INCOMPATIBLE_DESTINATION]
>
> Here is the log file attached.
>
> Is there any bug in the freeswitch version ? Or am I using the wrong 
> configuration?
>
> Note there is no NAT on the server, it DO droplet which is directly 
> connected to public IP.
>
> Any help will be highly appreciated.
>
> 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211027/071193fe/attachment-0001.html>

From gmaruzz at gmail.com  Wed Oct 27 17:20:36 2021
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 27 Oct 2021 19:20:36 +0200
Subject: [Freeswitch-users] Postgress BDR insert failed for "complete"
	table
In-Reply-To: <CAEjtZA2t+wLpGN5O1z7ba5jQktYqO7=8K8RFRoBNG+phtQvy-w@mail.gmail.com>
References: <CAEjtZA2t+wLpGN5O1z7ba5jQktYqO7=8K8RFRoBNG+phtQvy-w@mail.gmail.com>
Message-ID: <CALXCt0pAX3q9tkTXPCBLPxXVju0f+AJgiru5gyL=f2edRhZRPQ@mail.gmail.com>

On Wed, Oct 27, 2021 at 5:44 PM Abdirahman Osman <abdirahman.osm at gmail.com>
wrote:

> Hello,
>
> I am using postgres  with BDR bidirectional replication which requires
> primary keys for the freeswitch tables.
>
>  So far so good , except freeswitch is complaining about one table here is
> the error
>
> 2021-10-26 23:40:47.772796 95.47% [DEBUG] mod_pgsql.c:784 Query (insert
> into complete values (0,'hash_remote', 'list', '', '', '', '', '', '', '',
> '',  'dagobah-api-gw-1-2')) returned PGRES_FATAL_ERROR
> 2021-10-26 23:40:47.772796 95.47% [ERR] mod_pgsql.c:551 Error executing
> query:
> ERROR:  column "complete_uuid" is of type uuid but expression is of type
> integer
> LINE 1: insert into complete values (0,'hash_remote', 'list', '', ''...
>
> What I noticed is that freeswitch is doing an INSERT INTO command with no
> column names specified. I checked the source code and found this to be the
> only case for the "complete" table whereas the other tables when INSERT
> INTO is used column names are specified.
>
> Here is the structure of my  complete table with the new column
> complete_uuid as primary key.
>
> CREATE TABLE :myschema.complete (
> complete_uuid uuid DEFAULT public.gen_random_uuid() PRIMARY KEY,
> sticky  INTEGER,
> a1  VARCHAR(128),
> a2  VARCHAR(128),
> a3  VARCHAR(128),
> a4  VARCHAR(128),
> a5  VARCHAR(128),
> a6  VARCHAR(128),
> a7  VARCHAR(128),
> a8  VARCHAR(128),
> a9  VARCHAR(128),
> a10 VARCHAR(128),
> hostname VARCHAR(256)
>         );
>
> Any suggestions?
>
>
What is the table "complete" used for? by which module?

You can't do anything, only modify source in a way to explicit use fields,
and give a default to generate pk, or fill the pk explicitly.

Anyway, which module uses this table?

-giovanni


--
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211027/2171588e/attachment.html>

From jurijs.ivolga at gmail.com  Wed Oct 27 18:40:22 2021
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Wed, 27 Oct 2021 21:40:22 +0300
Subject: [Freeswitch-users] Bridging a channel via Push Notifcation
In-Reply-To: <CAKWpYbmc0OvV0h1YU_fixfXjUMb7bpo4EXxkRQVCYhaczUoaZw@mail.gmail.com>
References: <CAKWpYbm+2FPrdXZ9C1oq8cZFDCNWca9OOyvvxnFTLs6RmED0cQ@mail.gmail.com>
 <CAKWpYbmc0OvV0h1YU_fixfXjUMb7bpo4EXxkRQVCYhaczUoaZw@mail.gmail.com>
Message-ID: <CAOrA2Ub94kzcZgou+iydTAX2tvXxHq4o4BQkF4_2N_UprE5jkQ@mail.gmail.com>

Hi,

Before bridging, maybe try to answer a call from User-B?

So point 8 will be to answer a call from User-B and on point 9 please try
to bridge calls.

Jurijs


On Wed, Oct 27, 2021 at 9:25 PM Aditya Uppuluri <uppuluriaditya at gmail.com>
wrote:

> Hello Team,
> Does anyone know how to do this?
>
> Thank you in advance
>
>
> Regards,
> Aditya
>
>
> On Tue, Oct 26, 2021 at 3:31 PM Aditya Uppuluri <uppuluriaditya at gmail.com>
> wrote:
>
>> Hello All,
>> Thank you so much for the amazing software. I am new to freeswitch and
>> trying understand how to do the following use case (mobiles):
>>
>> 1. Call is received from LinPhone to User-B
>> 2. Freeswitch (esl code) checks if User-B is a registered user or not
>> 3. If a registered user, forward the call
>> 4. If not a registered user, then send a push notification to the
>> user-B's mobile.
>> 5. Freeswitch parks the call from LinPhone user
>> 6. User-B's mobile gets push notification and get's a call screen
>> 7. User-B accepts the call (Actually, he will send an INVITE with the
>> data received in the payload)
>> 8. Freeswitch receives the invite and should bridge the Linphone user
>> with User-B
>>
>> I am able to perform all the operations except the last one (Bridging the
>> call). If I try to use `uuid_bridge`, it says at least one call must be
>> answered. (*Neither channel is answered, cannot bridge them*.
>> )
>>
>> May I know how to bridge these two legs in freeswitch?
>>
>> FYI: I have the following park-only dialplan and controlling everything
>> from esl
>>
>> <extension name="fspark">
>> <condition field="destination_number" expression="^(.*)$">
>> <action application="set" data="bypass_media=true"/>
>> <action application="set" data="bypass_media_after_bridge=true"/>
>> <action application="park"/>
>> </condition>
>> </extension>
>>
>> Thanks in advance
>>
>> Regards,
>> Aditya
>>
> _________________________________________________________________________
>
> 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/20211027/4e90184b/attachment.html>

From jelledejong at powercraft.nl  Thu Oct 28 10:42:02 2021
From: jelledejong at powercraft.nl (Jelle de Jong)
Date: Thu, 28 Oct 2021 12:42:02 +0200
Subject: [Freeswitch-users] log format change breaking fail2ban? percentage
 added to logs? how to revert?
Message-ID: <0bf33652-0fab-9224-7f66-760bb75557a4@powercraft.nl>

Hello everybody,

The log format seem to have changed, how can I get it back to what it 
used to without the percentage thing? A search on Google failed me to 
provide an answer...

# fs_cli -x version
FreeSWITCH Version 1.10.7-release-19-883d2cb662~64bit 
(-release-19-883d2cb662 64bit)

# grep pref_line  /etc/fail2ban/filter.d/freeswitch.conf
_pref_line = ^%(__prefix_line)s(?:\d+-\d+-\d+ \d+:\d+:\d+\.\d+)?
failregex = %(_pref_line)s \[WARNING\] sofia_reg\.c:\d+ SIP auth 
(failure|challenge) \((REGISTER|INVITE)\) on sofia profile \'[^']+\' for 
\[[^\]]*\] from ip <HOST>$
             %(_pref_line)s \[WARNING\] sofia_reg\.c:\d+ Can't find user 
\[[^@]+@[^\]]+\] from <HOST>$

2021-10-28 12:28:55.736515 99.20% [WARNING] sofia_reg.c:1806 SIP auth 
failure (REGISTER) on sofia profile 'internal' for [30640 at 82.94.245.xxx] 
from ip 217.138.220.158

2021-05-20 16:44:54.159814 [WARNING] sofia_reg.c:1740 SIP auth failure 
(REGISTER) on sofia profile 'internal' for [123456 at 82.94.245.xxx] from 
ip 193.107.216.182

# fgrep percentage /etc/freeswitch/ -R

What is the use case of this new percentage in the log files?

Kind regards,

Jelle de Jong


From rahman.duran at erzurum.edu.tr  Thu Oct 28 08:00:30 2021
From: rahman.duran at erzurum.edu.tr (Rahman Duran)
Date: Thu, 28 Oct 2021 11:00:30 +0300
Subject: [Freeswitch-users] Updating caller id with attended transfer while
	phone still ringing
Message-ID: <CAEMJzSzU-mTNHrXmV0LxC0jREOrqSgY+K8ega_AGp8s2W3EoTg@mail.gmail.com>

Hi,

When doing attended transfer (A -> B -> attended transfer -> C) C only
updates its caller id display when it answers the call. Is there anyway to
update caller id on C, when B transfer the call while its still ringing?

The problem is we have some Cisco and Samsung SIP phones that does not have
a real "Blind Transfer" support. If you want to transfer without talking to
"C", you just press the transfer button quickly, while C starts to ring. So
I want to send the updated caller id to "C" after "B" transferred but
before "C" answers.

I have Yeastar PBX (asterisk based) works just like that but I can't
figure out how to do it with freeswitch.

Regards,

Rahman Duran
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211028/661722ef/attachment.html>

From Antony.Stone at freeswitch.open.source.it  Thu Oct 28 12:50:38 2021
From: Antony.Stone at freeswitch.open.source.it (Antony Stone)
Date: Thu, 28 Oct 2021 14:50:38 +0200
Subject: [Freeswitch-users] Trying to get short hostname
Message-ID: <202110281450.39060.Antony.Stone@freeswitch.open.source.it>

Hi.

I'm using FreeSwitch 1.10.7 as a client registering to other PBXs as well as a 
SIP server itself.

I am setting the user-agent string in the SIP profile, such as:

<param name="user-agent-string" value="FS $${CFGver} $${hostname}"/>

The variable $${CFGver}} is a version string I'm definiing in vars.xml

This is working, but it gives me the FQDN of the machine as $${hostname}} and 
I really only want the machine name without the domain.

I've found 
https://freeswitch.org/confluence/display/FREESWITCH/Regular+Expression and 
https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+regex and 
I've tried to use them to cut the domain off the value of $${hostname} but with 
no success:

In vars.xml:
<X-PRE-PROCESS cmd="set" data="shorthostname=${regex($${hostname}|([^\.]*).*|
%1)}"/>

This results in the variable shorthostname being assigned the value:
 	%{regex(abcd.efgh.tld|([^\.]*).*|%1)}

where abcd.efgh.tld is the server's FQDN.

Instead of doing that in vars.xml and creating the variable shorthostname, I 
tried putting the regex into the SIP profile:

<param name="user-agent-string" value="FS $${CFGver} ${regex($${hostname}|
([^\.]*).*|%1)}"/>

This ends up with precisely the same result.  The ${regex(....)} function 
simply isn't being executed.


Given that the machine's hostname is also available from the system command 
"hostname -s" I decided to try that instead:

In vars.xml:
<X-PRE-PROCESS cmd="set" data="shorthostname=${system(hostname -s)}"/>

Or in the SIP profile definition:
<param name="user-agent-string" value="FS $${CFGver} ${system(hostname -s)}"/>

These both produce the output:
	FS 0.65 ${system(hostname -s)}

In other words, the ${system(....)} command isn't working any more than 
${regex(....)} did.


What am I doing wrong please?


How can I get the machine's short hostname into a user-agent string for 
registration to a server?


Thanks,


Antony.

-- 
Perfection in design is achieved not when there is nothing left to add, but 
rather when there is nothing left to take away.

 - Antoine de Saint-Exupery

                                                   Please reply to the list;
                                                         please *don't* CC me.


From uppuluriaditya at gmail.com  Thu Oct 28 14:38:28 2021
From: uppuluriaditya at gmail.com (Aditya Uppuluri)
Date: Thu, 28 Oct 2021 20:08:28 +0530
Subject: [Freeswitch-users] Bridging a channel via Push Notifcation
In-Reply-To: <CAKWpYbmc0OvV0h1YU_fixfXjUMb7bpo4EXxkRQVCYhaczUoaZw@mail.gmail.com>
References: <CAKWpYbm+2FPrdXZ9C1oq8cZFDCNWca9OOyvvxnFTLs6RmED0cQ@mail.gmail.com>
 <CAKWpYbmc0OvV0h1YU_fixfXjUMb7bpo4EXxkRQVCYhaczUoaZw@mail.gmail.com>
Message-ID: <CAKWpYbke-STzW9dh7kYZsOQ5++xpqVSUTQdfnJ_xSAo4k6rS_w@mail.gmail.com>

Hello Team,
I could possibly answer the channel but I don't want the freeswitch to be
in the media path. Answering a channel from freeswitch, makes the
freeswitch in the middle of the media path doesn't it?

My Setup looks like this: kamailio(sbc) === Freeswitch (bypass media) ===
Customer's VOIP (freeswitch/asterisk).

So, Customer's VOIP network will anchor the media but not us.

Is there a way I could make the uuid_bridge or simple bridge work without
media for a non-registered user?

Thanks in advance



Regards,
Aditya Uppuluri


On Wed, Oct 27, 2021 at 9:49 PM Aditya Uppuluri <uppuluriaditya at gmail.com>
wrote:

> Hello Team,
> Does anyone know how to do this?
>
> Thank you in advance
>
>
> Regards,
> Aditya
>
>
> On Tue, Oct 26, 2021 at 3:31 PM Aditya Uppuluri <uppuluriaditya at gmail.com>
> wrote:
>
>> Hello All,
>> Thank you so much for the amazing software. I am new to freeswitch and
>> trying understand how to do the following use case (mobiles):
>>
>> 1. Call is received from LinPhone to User-B
>> 2. Freeswitch (esl code) checks if User-B is a registered user or not
>> 3. If a registered user, forward the call
>> 4. If not a registered user, then send a push notification to the
>> user-B's mobile.
>> 5. Freeswitch parks the call from LinPhone user
>> 6. User-B's mobile gets push notification and get's a call screen
>> 7. User-B accepts the call (Actually, he will send an INVITE with the
>> data received in the payload)
>> 8. Freeswitch receives the invite and should bridge the Linphone user
>> with User-B
>>
>> I am able to perform all the operations except the last one (Bridging the
>> call). If I try to use `uuid_bridge`, it says at least one call must be
>> answered. (*Neither channel is answered, cannot bridge them*.
>> )
>>
>> May I know how to bridge these two legs in freeswitch?
>>
>> FYI: I have the following park-only dialplan and controlling everything
>> from esl
>>
>> <extension name="fspark">
>> <condition field="destination_number" expression="^(.*)$">
>> <action application="set" data="bypass_media=true"/>
>> <action application="set" data="bypass_media_after_bridge=true"/>
>> <action application="park"/>
>> </condition>
>> </extension>
>>
>> Thanks in advance
>>
>> Regards,
>> Aditya
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211028/c64ae3cf/attachment.html>

From koldo.aingeru at sarenet.es  Thu Oct 28 12:30:39 2021
From: koldo.aingeru at sarenet.es (Koldo Aingeru)
Date: Thu, 28 Oct 2021 14:30:39 +0200
Subject: [Freeswitch-users] Issue with mod_event_multicast in 1.10.7
Message-ID: <A6F7EBE5-FAFA-409B-965A-DDDF692C1B14@sarenet.es>

Hi,

I just updated a test server from FS 1.10.6 to 1.10.7 on Debian 10 (apt update/upgrade from the official repo) and without changing anything on my config I got a buffer overflow when running "/usr/bin/freeswitch -u freeswitch -g freeswitch -c -nonat ":

*** buffer overflow detected ***: /usr/bin/freeswitch terminated
Aborted

I tried the original modules.xml that comes with the package and it starts up without problems. 

Doing a bit of trial & error with my modules.conf.xml I noticed that the problema is with the  mod_event_multicast module. Loading it from modules.conf.xml or by doing “load mod_event_multicast” result in a crash.

This is the current module:
# sha1sum /usr/lib/freeswitch/mod/mod_event_multicast.so
4daee6fa8ae8ed59c5ebc32393cb7357fead8198  /usr/lib/freeswitch/mod/mod_event_multicast.so

Overwriting the module with the 1.10.6 one seems to work:


freeswitch at pbx.mydomain <mailto:freeswitch at pbx.mydomain>.com> load mod_event_multicast
+OK Reloading XML
+OK


2021-10-28 14:21:23.038592 96.17% [CONSOLE] switch_loadable_module.c:1768 Successfully Loaded [mod_event_multicast]
2021-10-28 14:21:23.038592 96.17% [NOTICE] switch_loadable_module.c:389 Adding API Function 'multicast_peers'


but even with the module loaded the events aren't being sent to my other server, my event_multicast_conf.xml is this:


<configuration name="event_multicast.conf" description="Multicast Event">
  <settings>
    <param name="address" value="225.1.1.1"/>
    <param name="port" value="4242"/>
    <param name="bindings" value="CUSTOM sofia::register"/>
    <param name="ttl" value="1"/>
    <!-- <param name="loopback" value="no"/>-->
    <!-- Uncomment this to enable pre-shared key encryption on the packets. -->
    <!-- For this option to work, you'll need to have the openssl development -->
    <!-- headers installed when you ran ./configure -->
    <param name="psk" value=“pbx.mydomain.com"/>
  </settings>
</configuration>

I don’t see any multicast traffic being sent from this server nor the " Propagating registration for….” message.

Any idea what can be happening? 

Thanks!

========================================
Koldo Aingeru Marcos Fdez.  
Ingeniería de Sistemas
Sarenet S.A.U.
944209470
Parque Tecnológico, Edificio 103
48170 Zamudio, Bizkaia
www.sarenet.es
========================================







-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211028/55dea18f/attachment-0001.html>

From abdirahman.osm at gmail.com  Thu Oct 28 15:21:58 2021
From: abdirahman.osm at gmail.com (Abdirahman Osman)
Date: Thu, 28 Oct 2021 11:21:58 -0400
Subject: [Freeswitch-users] Postgress BDR insert failed for "complete"
	table
In-Reply-To: <CALXCt0pAX3q9tkTXPCBLPxXVju0f+AJgiru5gyL=f2edRhZRPQ@mail.gmail.com>
References: <CAEjtZA2t+wLpGN5O1z7ba5jQktYqO7=8K8RFRoBNG+phtQvy-w@mail.gmail.com>
 <CALXCt0pAX3q9tkTXPCBLPxXVju0f+AJgiru5gyL=f2edRhZRPQ@mail.gmail.com>
Message-ID: <CAEjtZA2AatCjDGAGorC-wpGsGE_HHeMDaweuSL6bieQJQ+NPFA@mail.gmail.com>

Hi Giovanni,

Thanks for your comment.

I think the table is used by fs_cli to list available commands.
Another possible solution I came across  is to use Replica Identity " ALTER
TABLE mytable REPLICA IDENTITY FULL", will give it a try.

Abdirahman A. Osman



On Wed, 27 Oct 2021 at 13:53, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> On Wed, Oct 27, 2021 at 5:44 PM Abdirahman Osman <abdirahman.osm at gmail.com>
> wrote:
>
>> Hello,
>>
>> I am using postgres  with BDR bidirectional replication which requires
>> primary keys for the freeswitch tables.
>>
>>  So far so good , except freeswitch is complaining about one table here
>> is the error
>>
>> 2021-10-26 23:40:47.772796 95.47% [DEBUG] mod_pgsql.c:784 Query (insert
>> into complete values (0,'hash_remote', 'list', '', '', '', '', '', '', '',
>> '',  'dagobah-api-gw-1-2')) returned PGRES_FATAL_ERROR
>> 2021-10-26 23:40:47.772796 95.47% [ERR] mod_pgsql.c:551 Error executing
>> query:
>> ERROR:  column "complete_uuid" is of type uuid but expression is of type
>> integer
>> LINE 1: insert into complete values (0,'hash_remote', 'list', '', ''...
>>
>> What I noticed is that freeswitch is doing an INSERT INTO command with no
>> column names specified. I checked the source code and found this to be the
>> only case for the "complete" table whereas the other tables when INSERT
>> INTO is used column names are specified.
>>
>> Here is the structure of my  complete table with the new column
>> complete_uuid as primary key.
>>
>> CREATE TABLE :myschema.complete (
>> complete_uuid uuid DEFAULT public.gen_random_uuid() PRIMARY KEY,
>> sticky  INTEGER,
>> a1  VARCHAR(128),
>> a2  VARCHAR(128),
>> a3  VARCHAR(128),
>> a4  VARCHAR(128),
>> a5  VARCHAR(128),
>> a6  VARCHAR(128),
>> a7  VARCHAR(128),
>> a8  VARCHAR(128),
>> a9  VARCHAR(128),
>> a10 VARCHAR(128),
>> hostname VARCHAR(256)
>>         );
>>
>> Any suggestions?
>>
>>
> What is the table "complete" used for? by which module?
>
> You can't do anything, only modify source in a way to explicit use fields,
> and give a default to generate pk, or fill the pk explicitly.
>
> Anyway, which module uses this table?
>
> -giovanni
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
>
> 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/20211028/1058e276/attachment.html>

From avi at avimarcus.net  Thu Oct 28 15:29:22 2021
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 28 Oct 2021 15:29:22 +0000
Subject: [Freeswitch-users] Trying to get short hostname
In-Reply-To: <202110281450.39060.Antony.Stone@freeswitch.open.source.it>
References: <202110281450.39060.Antony.Stone@freeswitch.open.source.it>
Message-ID: <0100017cc7851cbe-539c6003-da03-4c4b-a6d2-046f0031a3e7-000000@email.amazonses.com>

System doesn't listen to stdout, it's only for running the command.

On Thu, Oct 28, 2021, 6:22 PM Antony Stone <
Antony.Stone at freeswitch.open.source.it> wrote:

> Hi.
>
> I'm using FreeSwitch 1.10.7 as a client registering to other PBXs as well
> as a
> SIP server itself.
>
> I am setting the user-agent string in the SIP profile, such as:
>
> <param name="user-agent-string" value="FS $${CFGver} $${hostname}"/>
>
> The variable $${CFGver}} is a version string I'm definiing in vars.xml
>
> This is working, but it gives me the FQDN of the machine as $${hostname}}
> and
> I really only want the machine name without the domain.
>
> I've found
> https://freeswitch.org/confluence/display/FREESWITCH/Regular+Expression
> and
> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+regex
> and
> I've tried to use them to cut the domain off the value of $${hostname} but
> with
> no success:
>
> In vars.xml:
> <X-PRE-PROCESS cmd="set"
> data="shorthostname=${regex($${hostname}|([^\.]*).*|
> %1)}"/>
>
> This results in the variable shorthostname being assigned the value:
>         %{regex(abcd.efgh.tld|([^\.]*).*|%1)}
>
> where abcd.efgh.tld is the server's FQDN.
>
> Instead of doing that in vars.xml and creating the variable shorthostname,
> I
> tried putting the regex into the SIP profile:
>
> <param name="user-agent-string" value="FS $${CFGver} ${regex($${hostname}|
> ([^\.]*).*|%1)}"/>
>
> This ends up with precisely the same result.  The ${regex(....)} function
> simply isn't being executed.
>
>
> Given that the machine's hostname is also available from the system
> command
> "hostname -s" I decided to try that instead:
>
> In vars.xml:
> <X-PRE-PROCESS cmd="set" data="shorthostname=${system(hostname -s)}"/>
>
> Or in the SIP profile definition:
> <param name="user-agent-string" value="FS $${CFGver} ${system(hostname
> -s)}"/>
>
> These both produce the output:
>         FS 0.65 ${system(hostname -s)}
>
> In other words, the ${system(....)} command isn't working any more than
> ${regex(....)} did.
>
>
> What am I doing wrong please?
>
>
> How can I get the machine's short hostname into a user-agent string for
> registration to a server?
>
>
> Thanks,
>
>
> Antony.
>
> --
> Perfection in design is achieved not when there is nothing left to add,
> but
> rather when there is nothing left to take away.
>
>  - Antoine de Saint-Exupery
>
>                                                    Please reply to the
> list;
>                                                          please *don't* CC
> me.
>
> _________________________________________________________________________
>
> 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/20211028/eb31c802/attachment.html>

From gmaruzz at gmail.com  Thu Oct 28 17:20:09 2021
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 28 Oct 2021 19:20:09 +0200
Subject: [Freeswitch-users] Postgress BDR insert failed for "complete"
	table
In-Reply-To: <CAEjtZA2AatCjDGAGorC-wpGsGE_HHeMDaweuSL6bieQJQ+NPFA@mail.gmail.com>
References: <CAEjtZA2t+wLpGN5O1z7ba5jQktYqO7=8K8RFRoBNG+phtQvy-w@mail.gmail.com>
 <CALXCt0pAX3q9tkTXPCBLPxXVju0f+AJgiru5gyL=f2edRhZRPQ@mail.gmail.com>
 <CAEjtZA2AatCjDGAGorC-wpGsGE_HHeMDaweuSL6bieQJQ+NPFA@mail.gmail.com>
Message-ID: <CALXCt0oXnDxbNmEpYh7OUVpA-zs1OVJN+huEDAVmOmkVy7=3_Q@mail.gmail.com>

On Thu, Oct 28, 2021 at 5:23 PM Abdirahman Osman <abdirahman.osm at gmail.com>
wrote:

>
> I think the table is used by fs_cli to list available commands.
> Another possible solution I came across  is to use Replica Identity "
> ALTER TABLE mytable REPLICA IDENTITY FULL", will give it a try.
>
>
You can add an autogenerated uuid (eg: defaults to some uuid generation
function) pk to that table, and it will work with BDR

-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211028/e00c6174/attachment.html>

From davidswalkabout at gmail.com  Fri Oct 29 00:26:15 2021
From: davidswalkabout at gmail.com (David P)
Date: Fri, 29 Oct 2021 13:26:15 +1300
Subject: [Freeswitch-users] Changing device on one leg improves the audio
	recorded from another leg?
Message-ID: <CAE23trumahY8d3=HQm=9bh2deJgTk_01kO1cn_oXcmkKCZW9hg@mail.gmail.com>

We had this bizarre experience today: A user on this device...

Dell Latitude 5580
i7-7600U CPU
Windows 10 Education OS 19042.685

...made verto calls in which he heard clicks and pops every few 100ms from
our virtual agent on another leg, and those audio defects also appear in
the mp4 recording. When he then switched to this device *on the same desk*
using the same network...

Dell Latitude 5590
i5-8350U CPU
Windows 10 Education OS 17763.1821

...there were no audio defects. Playing audio from other sources on the
problematic device revealed no problems with its speaker. At least one of
the failed calls used srflx candidates, not any relaying.

How could changing the device on leg A affect the audio recording that
apparently comes from leg B?

If it's relevant:

   - We use  "jitterbuffer_msec=5p:100p"
   - xml_cdr logs for both legs show seemingly high values for some aspects
   like these...
   - <skip_packet_count>187</skip_packet_count>
   <jitter_packet_count>1620</jitter_packet_count>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211029/3dd76efb/attachment-0001.html>

From dragos at freeswitch.org  Fri Oct 29 10:20:36 2021
From: dragos at freeswitch.org (Dragos Oancea)
Date: Fri, 29 Oct 2021 13:20:36 +0300
Subject: [Freeswitch-users] log format change breaking fail2ban?
 percentage added to logs? how to revert?
In-Reply-To: <0bf33652-0fab-9224-7f66-760bb75557a4@powercraft.nl>
References: <0bf33652-0fab-9224-7f66-760bb75557a4@powercraft.nl>
Message-ID: <CAGvJV4rY+rTxgSX2GmZ6EUckCcyM=ekt+nz5t_-aQeqW7_oDJQ@mail.gmail.com>

It adds idle cpu to log lines.
check commits 03d5730ffcdd0dc69247693c58239757623b9319
and 241d397e39fad336c13b0f43fecfa697182a4a3b if you wish to revert.


On Thu, Oct 28, 2021 at 1:42 PM Jelle de Jong <jelledejong at powercraft.nl>
wrote:

> Hello everybody,
>
> The log format seem to have changed, how can I get it back to what it
> used to without the percentage thing? A search on Google failed me to
> provide an answer...
>
> # fs_cli -x version
> FreeSWITCH Version 1.10.7-release-19-883d2cb662~64bit
> (-release-19-883d2cb662 64bit)
>
> # grep pref_line  /etc/fail2ban/filter.d/freeswitch.conf
> _pref_line = ^%(__prefix_line)s(?:\d+-\d+-\d+ \d+:\d+:\d+\.\d+)?
> failregex = %(_pref_line)s \[WARNING\] sofia_reg\.c:\d+ SIP auth
> (failure|challenge) \((REGISTER|INVITE)\) on sofia profile \'[^']+\' for
> \[[^\]]*\] from ip <HOST>$
>              %(_pref_line)s \[WARNING\] sofia_reg\.c:\d+ Can't find user
> \[[^@]+@[^\]]+\] from <HOST>$
>
> 2021-10-28 12:28:55.736515 99.20% [WARNING] sofia_reg.c:1806 SIP auth
> failure (REGISTER) on sofia profile 'internal' for [30640 at 82.94.245.xxx]
> from ip 217.138.220.158
>
> 2021-05-20 16:44:54.159814 [WARNING] sofia_reg.c:1740 SIP auth failure
> (REGISTER) on sofia profile 'internal' for [123456 at 82.94.245.xxx] from
> ip 193.107.216.182
>
> # fgrep percentage /etc/freeswitch/ -R
>
> What is the use case of this new percentage in the log files?
>
> Kind regards,
>
> Jelle de Jong
>
> _________________________________________________________________________
>
> 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/20211029/eeb4a32e/attachment.html>

From dragos at freeswitch.org  Fri Oct 29 10:37:47 2021
From: dragos at freeswitch.org (Dragos Oancea)
Date: Fri, 29 Oct 2021 13:37:47 +0300
Subject: [Freeswitch-users] Expert help needed debuging sip registration
 [logs included]
In-Reply-To: <CANyACTDy0q9-Bm-Ty_7GtqGMuqWAz-Md1GPJYQY1JriSbfTZ+A@mail.gmail.com>
References: <CANyACTDy0q9-Bm-Ty_7GtqGMuqWAz-Md1GPJYQY1JriSbfTZ+A@mail.gmail.com>
Message-ID: <CAGvJV4r7SujP=onaYUDCzwa-jBUAC=K3f4b_-AogeCEgkCeLmw@mail.gmail.com>

Maybe it's the way a1 hash is calculated. Take a trace while registering a
sip phone  and compare with what FS sends in the Authorization header
(response).

On Fri, Oct 15, 2021 at 10:24 PM Miguel Andrade <
miguel.andrade at neoscopio.com> wrote:

> Some how I'm able to register sip phones and softphones but not as a
> freeswitch gateway. I've tried several combinations to no avail.
>
>
> My provider instructions are:
>
>
> · Endereço Registar (Host): sip.sapo.pt
>
> · Endereço Proxy (Outbound Proxy): proxy.sip.sapo.pt
>
> · Porta Proxy: 5070
>
> · Username: +3513020xxxxx (*)
>
> · Password: [de 6 a 8 carateres]
>
> · Transporte Rede (Network Settings): UDP
>
> · SIP URL: +3513020xxxxx (*)
>
>
> I've entered this as username: +351xxx, password: xxx, proxy:
> proxy.sip.sapo.pt:5070 , from domain: sip.sapo.pt
>
> Doesn't work.
>
> I went to advanced and set transport to udp and outbound proxy as
> proxy.sip.sapo.pt:5070 didn't work wither.
>
> Tried all combinations I could but also no cigar.
>
>
> Can anyone help me debug this?
>
>
> Here are some fs_cli logs:
>
>
>
> 2021-10-15 15:41:46.615028 [NOTICE] sofia_reg.c:454 Registering
> ebf96e38-7592-43b7-82fa-4a9fd4c5facf
>
> nua_stack.c:569 nua_stack_signal() nua(0x7f6730005880): recv signal
> r_register
>
> nua_dialog.c:336 nua_dialog_usage_add() nua(0x7f6730005880): adding
> register usage
>
> nta.c:2670 nta_tpn_by_url() nta: selecting scheme sip
>
> nta.c:10831 outgoing_query_a() nta: for "proxy.sip.sapo.pt" query "
> proxy.sip.sapo.pt" A (cached)
>
> nta.c:10886 outgoing_answer_a() nta: proxy.sip.sapo.pt. IN A 194.65.2.241
>
> tport.c:3274 tport_tsend() tport_tsend(0x7f6724195450) tpn = udp/
> 194.65.2.241:5070
>
> tport.c:4697 tport_by_addrinfo() tport_by_addrinfo(0x7f6724195450): not
> found by name udp/194.65.2.241:5070
>
> tport.c:3609 tport_vsend() tport_vsend(0x7f6724195450): 642 bytes of 642
> to udp/194.65.2.241:5070
>
> nua_stack.c:528 nua_signal() nua(0x7f6730005880): sent signal r_register
>
> send 642 bytes to udp/[194.65.2.241]:5070 at 15:41:46.632220:
>
> ------------------------------------------------------------------------
>
> REGISTER sip[image: image.gif]roxy.sip.sapo.pt:5070;transport=udp SIP/2.0
>
> Via: SIP/2.0/UDP 92.222.246.205:5080;rport;branch=z9hG4bKc0QF2jmNeB8Qe
>
> Max-Forwards: 70
>
> From: <sip:+351302020304 at sip.sapo.pt>;tag=HjK0rD25pc27a
>
> To: <sip:+351302020304 at sip.sapo.pt>
>
> Call-ID: 8ab00b1b-ec4d-4564-b148-26fcb82622e1
>
> CSeq: 42600661 REGISTER
>
> Contact: <sip:gw+ebf96e38-7592-43b7-82fa-4a9fd4c5facf at 92.222.246.205:5080
> ;transport=udp;gw=ebf96e38-7592-43b7-82fa-4a9fd4c5facf>
>
> Expires: 60
>
> User-Agent: Grandstream HT802 1.0.3.2
>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY
>
> Supported: timer, path, replaces
>
> Content-Length: 0
>
>
> nta.c:8312 outgoing_send() nta: sent REGISTER (42600661) to udp/
> 194.65.2.241:5070
>
> tport.c:4175 tport_pend() tport_pend(0x7f6724195450): pending
> 0x7f67241145e0 for udp/92.222.246.205:5080 (already 0)
>
> tport.c:2751 tport_wakeup_pri() tport_wakeup_pri(0x7f6724195450): events IN
>
> tport.c:2882 tport_recv_event() tport_recv_event(0x7f6724195450)
>
> tport.c:3219 tport_recv_iovec() tport_recv_iovec(0x7f6724195450) msg
> 0x7f672416c140 from (udp/92.222.246.205:5080) has 602 bytes, veclen = 1
>
> recv 602 bytes from udp/[194.65.2.241]:5070 at 15:41:46.753181:
>
> ------------------------------------------------------------------------
>
> SIP/2.0 401 Unauthorized 11030230330
>
> Via: SIP/2.0/UDP 92.222.246.205:5080
> ;received=92.222.246.205;branch=z9hG4bKc0QF2jmNeB8Qe;rport=5080
>
> From: <sip:+351302020304 at sip.sapo.pt>;tag=HjK0rD25pc27a
>
> To: <sip:+351302020304 at sip.sapo.pt>;tag=9848f21803d1705ce6531d6ca2f
>
> Call-ID: 8ab00b1b-ec4d-4564-b148-26fcb82622e1
>
> CSeq: 42600661 REGISTER
>
> Content-Length: 0
>
> WWW-Authenticate: Digest nonce="D734DDD02A93696100000000E09BC51B",realm="
> ims.telecom.pt",algorithm=MD5,qop="auth"
>
> P-Charging-Vector:
> icid-value=3bnhe046p1ds2rk6ehcs2pg6hhoamfde9rpulo4623db2jmt;icid-generated-at=194.65.2.241;orig-ioi=internet
>
>
> tport.c:3037 tport_deliver() tport_deliver(0x7f6724195450): msg
> 0x7f672416c140 (602 bytes) from udp/194.65.2.241:5080/sip next=(nil)
>
> nta.c:3303 agent_recv_response() nta: received 401 Unauthorized
> 11030230330 for REGISTER (42600661)
>
> nta.c:3370 agent_recv_response() nta: 401 Unauthorized 11030230330 is
> going to a transaction
>
> nta.c:9584 outgoing_estimate_delay() nta_outgoing: RTT is 121.007 ms
>
> tport.c:4237 tport_release() tport_release(0x7f6724195450): 0x7f67241145e0
> by 0x7f672454b980 with 0x7f672416c140
>
> auth_digest.c:105 auth_digest_challenge_get() auth_digest_challenge_get():
> got 6
>
> nua_stack.c:271 nua_stack_event() nua(0x7f6730005880): event r_register
> 401 Unauthorized 11030230330
>
> nua_stack.c:528 nua_signal() nua(0x7f6730005880): sent signal
> r_authenticate
>
> nua_stack.c:569 nua_stack_signal() nua(0x7f6730005880): recv signal
> r_authenticate
>
> auth_digest.c:225 auth_digest_a1() auth_digest_a1() has A1 =
> MD5(+351302020304:ims.telecom.pt:1q2w!Q) =
> 8038cbcec02479ac6c24e22c0d4d343a
>
> auth_digest.c:317 auth_digest_response() A2 = MD5(REGISTER:sip[image:
> image.gif]roxy.sip.sapo.pt:5070;transport=udp)
>
> auth_digest.c:339 auth_digest_response() auth_response:
> 72688ed841494a95193f7d115c409667 = MD5(8038cbcec02479ac6c24e22c0d4d343a[image:
> image.gif]
> 734DDD02A93696100000000E09BC51B:00000001:3UagCahoEjqJ2gIAACyybQ:auth:66f331
>
> b6539f207dc0b673edce10bbbc) (qop=auth)
>
> nta.c:2670 nta_tpn_by_url() nta: selecting scheme sip
>
> nta.c:10831 outgoing_query_a() nta: for "proxy.sip.sapo.pt" query "
> proxy.sip.sapo.pt" A (cached)
>
> nta.c:10886 outgoing_answer_a() nta: proxy.sip.sapo.pt. IN A 194.65.2.241
>
> tport.c:3274 tport_tsend() tport_tsend(0x7f6724195450) tpn = udp/
> 194.65.2.241:5070
>
> tport.c:4697 tport_by_addrinfo() tport_by_addrinfo(0x7f6724195450): not
> found by name udp/194.65.2.241:5070
>
> tport.c:3609 tport_vsend() tport_vsend(0x7f6724195450): 920 bytes of 920
> to udp/194.65.2.241:5070
>
> send 920 bytes to udp/[194.65.2.241]:5070 at 15:41:46.753392:
>
> ------------------------------------------------------------------------
>
> REGISTER sip[image: image.gif]roxy.sip.sapo.pt:5070;transport=udp SIP/2.0
>
> Via: SIP/2.0/UDP 92.222.246.205:5080;rport;branch=z9hG4bKD9g83D5rBmyaa
>
> Max-Forwards: 70
>
> From: <sip:+351302020304 at sip.sapo.pt>;tag=HjK0rD25pc27a
>
> To: <sip:+351302020304 at sip.sapo.pt>
>
> Call-ID: 8ab00b1b-ec4d-4564-b148-26fcb82622e1
>
> CSeq: 42600662 REGISTER
>
> Contact: <sip:gw+ebf96e38-7592-43b7-82fa-4a9fd4c5facf at 92.222.246.205:5080
> ;transport=udp;gw=ebf96e38-7592-43b7-82fa-4a9fd4c5facf>
>
> Expires: 60
>
> User-Agent: Grandstream HT802 1.0.3.2
>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY
>
> Supported: timer, path, replaces
>
> Authorization: Digest username="+351302020304", realm="ims.telecom.pt",
> nonce="D734DDD02A93696100000000E09BC51B", cnonce="3UagCahoEjqJ2gIAACyybQ",
> algorithm=MD5, uri="sip[image: image.gif]roxy.sip.sapo.pt:5070;transport=
>
> udp", response="72688ed841494a95193f7d115c409667", qop=auth, nc=00000001
>
> Content-Length: 0
>
>
> nta.c:8312 outgoing_send() nta: sent REGISTER (42600662) to udp/
> 194.65.2.241:5070
>
> tport.c:4175 tport_pend() tport_pend(0x7f6724195450): pending
> 0x7f67242e5a90 for udp/92.222.246.205:5080 (already 0)
>
> tport.c:2751 tport_wakeup_pri() tport_wakeup_pri(0x7f6724195450): events IN
>
> tport.c:2882 tport_recv_event() tport_recv_event(0x7f6724195450)
>
> tport.c:3219 tport_recv_iovec() tport_recv_iovec(0x7f6724195450) msg
> 0x7f67242d2000 from (udp/92.222.246.205:5080) has 602 bytes, veclen = 1
>
> recv 602 bytes from udp/[194.65.2.241]:5070 at 15:41:46.871252:
>
> ------------------------------------------------------------------------
>
> SIP/2.0 401 Unauthorized 1103023032F
>
> Via: SIP/2.0/UDP 92.222.246.205:5080
> ;received=92.222.246.205;branch=z9hG4bKD9g83D5rBmyaa;rport=5080
>
> From: <sip:+351302020304 at sip.sapo.pt>;tag=HjK0rD25pc27a
>
> To: <sip:+351302020304 at sip.sapo.pt>;tag=9848f2180cea805ce6531d8948d
>
> Call-ID: 8ab00b1b-ec4d-4564-b148-26fcb82622e1
>
> CSeq: 42600662 REGISTER
>
> Content-Length: 0
>
> WWW-Authenticate: Digest nonce="47DFEAEB2A936961000000000782EF64",realm="
> ims.telecom.pt",algorithm=MD5,qop="auth"
>
> P-Charging-Vector:
> icid-value=3bnhe046p1ds2rk6ehcs2pg6hhoamfde9rpulo462blb2jmt;icid-generated-at=194.65.2.241;orig-ioi=internet
>
>
> tport.c:3037 tport_deliver() tport_deliver(0x7f6724195450): msg
> 0x7f67242d2000 (602 bytes) from udp/194.65.2.241:5080/sip next=(nil)
>
> nta.c:3303 agent_recv_response() nta: received 401 Unauthorized
> 1103023032F for REGISTER (42600662)
>
> nta.c:3370 agent_recv_response() nta: 401 Unauthorized 1103023032F is
> going to a transaction
>
> nta.c:9584 outgoing_estimate_delay() nta_outgoing: RTT is 117.905 ms
>
> tport.c:4237 tport_release() tport_release(0x7f6724195450): 0x7f67242e5a90
> by 0x7f672410d3b0 with 0x7f67242d2000
>
> auth_digest.c:105 auth_digest_challenge_get() auth_digest_challenge_get():
> got 6
>
> nua_client.c:1223 nua_base_client_check_restart() nua(0x7f6730005880): bad
> credentials, clearing them
>
> nua_stack.c:271 nua_stack_event() nua(0x7f6730005880): event r_register
> 401 Unauthorized 1103023032F
>
> nua_stack.c:528 nua_signal() nua(0x7f6730005880): sent signal
> r_authenticate
>
> nua_stack.c:569 nua_stack_signal() nua(0x7f6730005880): recv signal
> r_authenticate
>
> nua_stack.c:271 nua_stack_event() nua(0x7f6730005880): event r_register
> 904 Operation has no matching challenge
>
> nua_dialog.c:395 nua_dialog_usage_remove_at() nua(0x7f6730005880):
> removing register usage
>
> 2021-10-15 15:41:46.854983 [ERR] sofia_reg.c:2470
> ebf96e38-7592-43b7-82fa-4a9fd4c5facf Failed Registration with status
> Operation has no matching challenge  [904]. failure #1
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211029/3e4039fe/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image.gif
Type: image/gif
Size: 42 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211029/3e4039fe/attachment-0005.gif>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image.gif
Type: image/gif
Size: 42 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211029/3e4039fe/attachment-0006.gif>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image.gif
Type: image/gif
Size: 42 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211029/3e4039fe/attachment-0007.gif>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image.gif
Type: image/gif
Size: 42 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211029/3e4039fe/attachment-0008.gif>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image.gif
Type: image/gif
Size: 42 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211029/3e4039fe/attachment-0009.gif>

From dragos at freeswitch.org  Fri Oct 29 11:30:34 2021
From: dragos at freeswitch.org (Dragos Oancea)
Date: Fri, 29 Oct 2021 14:30:34 +0300
Subject: [Freeswitch-users] Codec problem,
	Freeswitch does not regard absolute_codec_string?
In-Reply-To: <CAEMJzSz9ujt0iK7vG5MMKJh8xmsanN92agrq3pVwuPRv+qwxZA@mail.gmail.com>
References: <CAEMJzSxTL3sgQR6830P8LA+ZARSf8SBNDN1mWg+G+c102ajx7w@mail.gmail.com>
 <CAEMJzSz9ujt0iK7vG5MMKJh8xmsanN92agrq3pVwuPRv+qwxZA@mail.gmail.com>
Message-ID: <CAGvJV4rSZOJeh6LyQq4uP2KeGeFbxoHJ_xnCbG7fOW890++vYg@mail.gmail.com>

maybe the comments on this issue will help:
https://github.com/signalwire/freeswitch/issues/1340 .

On Wed, Oct 27, 2021 at 8:13 PM Rahman Duran <rahman.duran at erzurum.edu.tr>
wrote:

> Hi,
>
> I am struggling tuning Freeswitch for late codec negotiation. Here is
> local profile settings:
>
> <param name="inbound-late-negotiation" value="true"/>
> <param name="inbound-codec-prefs" value="G722,PCMA"/>
> <param name="outbound-codec-prefs" value="G722,PCMA"/>
> <param name="inbound-codec-negotiation" value="greedy"/>
>
> As you see I enabled late-negotiation and set codecs prefs to allow G722
> and PCMA. My goal is to use G722 if both sides support it and fall back to
> PCMA if G722 is not an option for A leg or B leg.
>
> Here is the local dial plan config where I bridge calls:
>
>     <extension name="local_accounts_freeswitch">
>       <condition regex="all">
>         <regex field="destination_number" expression="^[0-9]{4}$"/>
>         <regex field="${local_user_exists}"
> expression="^true$|^TRUE$|^True$"/>
>         <regex field="${santral_kontrol}" expression="^freeswitch$"/>
>         <action application="set" inline="true"
> data="call_pickup_group=${user_data(${destination_number}@${domain_name}
> var call_pickup_group)}"/>
>         <action application="hash"
> data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/>
>         <action application="hash"
> data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/>
>         <action application="hash"
> data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/>
>         <action application="log" data="ep_codec_string --
> ${ep_codec_string}" />
>         <action application="set" data="inherit_codec=true"/>
>         <action application="export"
> data="nolocal:absolute_codec_string='${ep_codec_string}'"/>
>         <action application="set_profile_var"
> data="callee_id_name=${user_data(${destination_number}@${domain_name} var
> effective_caller_id_name)}"/>
>         <action application="set" data="hangup_after_bridge=true"/>
>         <action application="set" data="continue_on_fail=true"/>
>         <action application="set" data="call_timeout=30"/>
>         <action application="set" data="execute_on_answer=sched_hangup
> +21600 alloted_timeout" />
>         <action application="bridge"
> data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name}
> var effective_caller_id_name)}}user/${destination_number}@
> ${domain_name}"/>
>         <action application="execute_extension"
> data="hata-${originate_disposition} XML hata_anonslari"/>
>       </condition>
>     </extension>
>
> So I export absolute_codec_string to B leg using ep_codec_string. When I
> test a call from 2730->2110, the call fails with
> "incompetible_destination".  Call logs here:
> https://pastebin.freeswitch.org/view/b2978e79 The 2730 is a Cisco phone
> with "G722,PCMU,PCMA,G729,L16,iLBC" codec support. The 2110 is a Samsung
> phone that only supports "PCMA,PCMU,G729" codecs. As you see cisco invites
> with all codecs it supports and freeswitch sets absolute_codec_string with
> "G722,PCMA" as expected from the config. But as you see in the call logs
> Freeswitch only sends "G722" in SDP to Samsung phone so Samsung rejects the
> call.
>
> Why does Freeswitch only send G722? How can I fix this? What am I doing
> wrong?
>
> P.S. using debian packages: FreeSWITCH Version
> 1.10.6-release-18-1ff9d0a60e~64bit (-release-18-1ff9d0a60e 64bit)
>
> Regards,
>
> Rahman Duran
> _________________________________________________________________________
>
> 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/20211029/2493dfda/attachment.html>

From jurijs.ivolga at gmail.com  Fri Oct 29 11:45:48 2021
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Fri, 29 Oct 2021 14:45:48 +0300
Subject: [Freeswitch-users] Bridging a channel via Push Notifcation
In-Reply-To: <CAKWpYbke-STzW9dh7kYZsOQ5++xpqVSUTQdfnJ_xSAo4k6rS_w@mail.gmail.com>
References: <CAKWpYbm+2FPrdXZ9C1oq8cZFDCNWca9OOyvvxnFTLs6RmED0cQ@mail.gmail.com>
 <CAKWpYbmc0OvV0h1YU_fixfXjUMb7bpo4EXxkRQVCYhaczUoaZw@mail.gmail.com>
 <CAKWpYbke-STzW9dh7kYZsOQ5++xpqVSUTQdfnJ_xSAo4k6rS_w@mail.gmail.com>
Message-ID: <CAOrA2Ubs=HFGR-+d3-q5ahi-iMOTFYpkaPGAyFGK3W040yKmxA@mail.gmail.com>

Hi,

Then why do you need freeswitch on your setup, if you don't want to handle
media? Just use a single Kamailio and that is it. Check out the tsilo
module in Kamailio, I think it will help you with your flow.

Additionally, I don't really get why you send out invite from Device-B. In
most cases you will need to wait for registration from device-B and then
send invite to it.

Jurijs


On Thu, Oct 28, 2021 at 6:45 PM Aditya Uppuluri <uppuluriaditya at gmail.com>
wrote:

> Hello Team,
> I could possibly answer the channel but I don't want the freeswitch to be
> in the media path. Answering a channel from freeswitch, makes the
> freeswitch in the middle of the media path doesn't it?
>
> My Setup looks like this: kamailio(sbc) === Freeswitch (bypass media) ===
> Customer's VOIP (freeswitch/asterisk).
>
> So, Customer's VOIP network will anchor the media but not us.
>
> Is there a way I could make the uuid_bridge or simple bridge work without
> media for a non-registered user?
>
> Thanks in advance
>
>
>
> Regards,
> Aditya Uppuluri
>
>
> On Wed, Oct 27, 2021 at 9:49 PM Aditya Uppuluri <uppuluriaditya at gmail.com>
> wrote:
>
>> Hello Team,
>> Does anyone know how to do this?
>>
>> Thank you in advance
>>
>>
>> Regards,
>> Aditya
>>
>>
>> On Tue, Oct 26, 2021 at 3:31 PM Aditya Uppuluri <uppuluriaditya at gmail.com>
>> wrote:
>>
>>> Hello All,
>>> Thank you so much for the amazing software. I am new to freeswitch and
>>> trying understand how to do the following use case (mobiles):
>>>
>>> 1. Call is received from LinPhone to User-B
>>> 2. Freeswitch (esl code) checks if User-B is a registered user or not
>>> 3. If a registered user, forward the call
>>> 4. If not a registered user, then send a push notification to the
>>> user-B's mobile.
>>> 5. Freeswitch parks the call from LinPhone user
>>> 6. User-B's mobile gets push notification and get's a call screen
>>> 7. User-B accepts the call (Actually, he will send an INVITE with the
>>> data received in the payload)
>>> 8. Freeswitch receives the invite and should bridge the Linphone user
>>> with User-B
>>>
>>> I am able to perform all the operations except the last one (Bridging
>>> the call). If I try to use `uuid_bridge`, it says at least one call must be
>>> answered. (*Neither channel is answered, cannot bridge them*.
>>> )
>>>
>>> May I know how to bridge these two legs in freeswitch?
>>>
>>> FYI: I have the following park-only dialplan and controlling everything
>>> from esl
>>>
>>> <extension name="fspark">
>>> <condition field="destination_number" expression="^(.*)$">
>>> <action application="set" data="bypass_media=true"/>
>>> <action application="set" data="bypass_media_after_bridge=true"/>
>>> <action application="park"/>
>>> </condition>
>>> </extension>
>>>
>>> Thanks in advance
>>>
>>> Regards,
>>> Aditya
>>>
>> _________________________________________________________________________
>
> 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/20211029/ac8eecaf/attachment-0001.html>

From support at naud.io  Fri Oct 29 18:57:52 2021
From: support at naud.io (Support from NetworkedAudio LLC)
Date: Fri, 29 Oct 2021 18:57:52 +0000
Subject: [Freeswitch-users] Sending announcements to all callers
In-Reply-To: <SN4PR0701MB3822B9B40440C40EEDD17787A3819@SN4PR0701MB3822.namprd07.prod.outlook.com>
References: <SN4PR0701MB382296A8385542D2322A36A5A3BF9@SN4PR0701MB3822.namprd07.prod.outlook.com>
 <SN4PR0701MB3822B9B40440C40EEDD17787A3819@SN4PR0701MB3822.namprd07.prod.outlook.com>
Message-ID: <SN4PR0701MB3822B6356CB7297A61C5A5B5A3879@SN4PR0701MB3822.namprd07.prod.outlook.com>

Spoke too soon. Seems to be a memory leak doing it this way - will simplify and add a JIRA.

Switched to an Announce conference - portaudio is deaf, every autocall is mute. Still had to use loopback - @conference(Announce) doesn’t work.

Emailing for the sake of future searches.
________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Support from NetworkedAudio LLC <support at naud.io>
Sent: Friday, October 22, 2021 8:46 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Sending announcements to all callers

Can't seem to do this at the call level, so have set up an autocall to the eavesdrop when the conference is made.

Then when the local resources access the conference, they put the eavesdrop call on hold and then store the UUID of that call in a lookup with the conference name.
The hangup hook on those local resources checks the conference xml_list and sees how many local members are left, and if none takes the eavesdrop off hold.

No loopback, resources seem OK. I'd prefer to do it pre-conference, but will see how this goes.

Any advice still welcome.

________________________________
From: Support from NetworkedAudio LLC
Sent: Thursday, October 21, 2021 12:02 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Sending announcements to all callers

I have a system where a user dials in, get authenticated and placed into a standard conference. Other local resources get added to the conferences as needed, and that is working well.

For safety reasons, we need to play a single incoming PortAudio channel to all dialed in users.

If I do an auto call or originate I can add a deaf member to each conference which is a loopback to an eavesdrop. This is messy but works.

However, it uses a lot of resources and now my local people have the announcements twice, with delay.

What I want is just to send audio out to those dialing in.

I have tried sending eavesdropping just to the incoming call but it won’t let me then join a conference through the dialplan - processing stops at the eavesdrop.

I see broadcast but can only send tones or saved files.

Ideally the incoming external callers would hear the announcements and not the conferences themselves. I’d like to avoid loopback, too.

Does anyone have ideas, please?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211029/37815c9c/attachment.html>

From nico at vthadden.de  Fri Oct 29 20:12:57 2021
From: nico at vthadden.de (Nicola von Thadden)
Date: Fri, 29 Oct 2021 22:12:57 +0200
Subject: [Freeswitch-users] Installation failed Debian : SETSCHEDULER
In-Reply-To: <etPan.564bfbfa.d7a88a1.30d1@MacbookAir.local>
References: <etPan.564bfbfa.d7a88a1.30d1@MacbookAir.local>
Message-ID: <c2bb9272-abc8-7ed9-26ac-9e2a771aa1ca@vthadden.de>

Hi all,

just because this issue hit my just in a new container:
the container running freeswitch has to be privileged. Running an 
unprivileged container (in my case, it's with proxmox) won't have the 
necessary permissions to set the required CPU scheduler.
Making the container privileged solved the problem and freeswitch starts 
as expected.

Starting freeswitch manually worked before, because the scheduler is set 
by systemd (defined in the unit file).
Hope that helps if anyone else finds that message via your favorite 
search engine ;)

Regards
Nico


On 11/17/15 19:32, pgoergler at gmail.com wrote:
>
> # tail /var/log/syslog
>
> Nov 17 16:50:15 freeswitch systemd[20020]: Failed at step SETSCHEDULER 
> spawning /usr/bin/freeswitch: Operation not permitted
>
> Nov 17 16:50:15 freeswitch systemd[1]: freeswitch.service: control 
> process exited, code=exited status=214
>
> Nov 17 16:50:15 freeswitch systemd[1]: Failed to start freeswitch.
>
> Nov 17 16:50:15 freeswitch systemd[1]: Unit freeswitch.service entered 
> failed state.
>
> Nov 17 16:50:15 freeswitch systemd[1]: freeswitch.service holdoff time 
> over, scheduling restart.
>
> Nov 17 16:50:15 freeswitch systemd[1]: Stopping freeswitch...
>
> Nov 17 16:50:15 freeswitch systemd[1]: Starting freeswitch...
>
> Nov 17 16:50:15 freeswitch systemd[1]: freeswitch.service start 
> request repeated too quickly, refusing to start.
>
> Nov 17 16:50:15 freeswitch systemd[1]: Failed to start freeswitch.
>
> Nov 17 16:50:15 freeswitch systemd[1]: Unit freeswitch.service entered 
> failed state.
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211029/565c70c5/attachment.html>

From jelledejong at powercraft.nl  Fri Oct 29 20:21:56 2021
From: jelledejong at powercraft.nl (Jelle de Jong)
Date: Fri, 29 Oct 2021 22:21:56 +0200
Subject: [Freeswitch-users] log format change breaking fail2ban?
 percentage added to logs? how to revert?
In-Reply-To: <CAGvJV4rY+rTxgSX2GmZ6EUckCcyM=ekt+nz5t_-aQeqW7_oDJQ@mail.gmail.com>
References: <0bf33652-0fab-9224-7f66-760bb75557a4@powercraft.nl>
 <CAGvJV4rY+rTxgSX2GmZ6EUckCcyM=ekt+nz5t_-aQeqW7_oDJQ@mail.gmail.com>
Message-ID: <963bec80-c0d1-b4a2-2a60-45748e0d0959@powercraft.nl>

Thank you Dragos, looking at the commits there is no configure option to 
change the new behaviour.

I tried creating a new regex line based on the added percentage

_pref_line = ^%(__prefix_line)s(?:\d+-\d+-\d+ \d+:\d+:\d+\.\d+ \d+\.\d+%%)?

I can not get it to work successfully any-more, if someone has a updated 
fail2ban filter for the new log format please share.

Kind regards,

Jelle de Jong

On 10/29/21 12:20 PM, Dragos Oancea wrote:
> It adds idle cpu to log lines.
> check commits 03d5730ffcdd0dc69247693c58239757623b9319 
> and 241d397e39fad336c13b0f43fecfa697182a4a3b if you wish to revert.
> 
> 
> On Thu, Oct 28, 2021 at 1:42 PM Jelle de Jong <jelledejong at powercraft.nl 
> <mailto:jelledejong at powercraft.nl>> wrote:
> 
>     Hello everybody,
> 
>     The log format seem to have changed, how can I get it back to what it
>     used to without the percentage thing? A search on Google failed me to
>     provide an answer...
> 
>     # fs_cli -x version
>     FreeSWITCH Version 1.10.7-release-19-883d2cb662~64bit
>     (-release-19-883d2cb662 64bit)
> 
>     # grep pref_line  /etc/fail2ban/filter.d/freeswitch.conf
>     _pref_line = ^%(__prefix_line)s(?:\d+-\d+-\d+ \d+:\d+:\d+\.\d+)?
>     failregex = %(_pref_line)s \[WARNING\] sofia_reg\.c:\d+ SIP auth
>     (failure|challenge) \((REGISTER|INVITE)\) on sofia profile \'[^']+\'
>     for
>     \[[^\]]*\] from ip <HOST>$
>                   %(_pref_line)s \[WARNING\] sofia_reg\.c:\d+ Can't find
>     user
>     \[[^@]+@[^\]]+\] from <HOST>$
> 
>     2021-10-28 12:28:55.736515 99.20% [WARNING] sofia_reg.c:1806 SIP auth
>     failure (REGISTER) on sofia profile 'internal' for
>     [30640 at 82.94.245.xxx]
>     from ip 217.138.220.158
> 
>     2021-05-20 16:44:54.159814 [WARNING] sofia_reg.c:1740 SIP auth failure
>     (REGISTER) on sofia profile 'internal' for [123456 at 82.94.245.xxx] from
>     ip 193.107.216.182
> 
>     # fgrep percentage /etc/freeswitch/ -R
> 
>     What is the use case of this new percentage in the log files?
> 
>     Kind regards,
> 
>     Jelle de Jong
> 
>     _________________________________________________________________________
> 
>     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
> 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 support at telium.io  Sat Oct 30 21:30:57 2021
From: support at telium.io (Telium Technical Support)
Date: Sat, 30 Oct 2021 21:30:57 +0000
Subject: [Freeswitch-users] SIP over DTLS support
Message-ID: <0100017cd31ce035-9e146ede-4cae-4f12-a0a0-7a1c071e1a69-000000@email.amazonses.com>

Does FreeSwitch support SIP of DTLS?  Not clear from the docs (though
mentions of webrtc possibly using this)

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211030/624575e9/attachment.html>

From rahman.duran at erzurum.edu.tr  Tue Oct 26 06:14:00 2021
From: rahman.duran at erzurum.edu.tr (Rahman Duran)
Date: Tue, 26 Oct 2021 09:14:00 +0300
Subject: [Freeswitch-users] Overriding hangup cause in CDR logs
Message-ID: <CAEMJzSy3t5gw0ccwUmG9XyA7no9g-WbiVpeya1G5a0DfW=3LRg@mail.gmail.com>

Hi,

I am using announcements for fail hungup causes like busy, no_answer etc.
Here is my dial plan

<extension name="local_accounts_freeswitch">
      <condition regex="all">
        <regex field="destination_number" expression="^[0-9]{4}$"/>
        <regex field="${local_user_exists}"
expression="^true$|^TRUE$|^True$"/>
        <regex field="${santral_kontrol}" expression="^freeswitch$"/>
        <action application="set" inline="true"
data="call_pickup_group=${user_data(${destination_number}@${domain_name}
var call_pickup_group)}"/>
        <action application="hash"
data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/>
        <action application="hash"
data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/>
        <action application="hash"
data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/>
        <action application="log" data="ep_codec_string --
${ep_codec_string}" />
        <action application="set" data="inherit_codec=true"/>
        <action application="export"
data="nolocal:absolute_codec_string=${ep_codec_string}"/>
        <action application="set_profile_var"
data="callee_id_name=${user_data(${destination_number}@${domain_name} var
effective_caller_id_name)}"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="call_timeout=30"/>
        <action application="set" data="execute_on_answer=sched_hangup
+21600 alloted_timeout" />
        <action application="bridge"
data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name}
var effective_caller_id_name)}}user/${destination_number}@${domain_name}"/>
        <action application="execute_extension"
data="hata-${originate_disposition} XML hata_anonslari"/>
      </condition>
    </extension>

And here is announcements context that  I handle hangup causes:

<include>
  <!-- Burada iç çağrılarda alınan hata durumlarının anonsları işlenecek.
    -->
  <context name="hata_anonslari">
    <!-- yerel dahili  -->
    <extension name="error-causes-user_busy">
      <condition field="destination_number" expression="^hata-USER_BUSY$">
        <action application="export" data="cdr_hata_anonsu=USER_BUSY"/>
        <action application="pre_answer"/>
        <action application="playback"
data="$${anons_dosya_yolu}/user_busy.wav"/>
        <action application="sleep" data="1000"/>
        <action application="hangup" data="${originate_disposition}"/>
      </condition>
    </extension>
    <extension name="error-causes-no_answer">
      <condition field="destination_number" expression="^hata-NO_ANSWER$">
        <action application="export" data="cdr_hata_anonsu=NO_ANSWER"/>
        <action application="pre_answer"/>
        <action application="playback"
data="$${anons_dosya_yolu}/no_answer.wav"/>
        <action application="sleep" data="1000"/>
        <action application="hangup" data="${originate_disposition}"/>
      </condition>
    </extension>
    <extension name="error-causes-user_not_registered">
      <condition field="destination_number"
expression="^hata-USER_NOT_REGISTERED$">
        <action application="export"
data="cdr_hata_anonsu=USER_NOT_REGISTERED"/>
        <action application="pre_answer"/>
        <action application="playback"
data="$${anons_dosya_yolu}/user_not_registered.wav"/>
        <action application="sleep" data="1000"/>
        <action application="hangup" data="${originate_disposition}"/>
      </condition>
    </extension>
    <extension name="error-causes-any">
      <condition field="destination_number" expression="^hata-(.*)$">
        <action application="export"
data="cdr_hata_anonsu=${originate_disposition}"/>
        <action application="hangup" data="${originate_disposition}"/>
      </condition>
    </extension>
  <!-- context hata_anonslari sonu -->
  </context>
</include>


The problem is if the calling leg does not listen the announcement to the
end and hangup, CDR logs shows "Originator Cancel" as hangup cause. As I
already know the real hangup cause, how can I override the CDR hangup cause
with the real one? For now I am setting another variable (cdr_hata_anonsu)
and added it to CDR logs, but if possible I want to fix this with a real
solution.

Regards,

Rahman Duran
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211026/095c27ed/attachment-0003.html>

From rahman.duran at erzurum.edu.tr  Thu Oct 28 08:00:30 2021
From: rahman.duran at erzurum.edu.tr (Rahman Duran)
Date: Thu, 28 Oct 2021 11:00:30 +0300
Subject: [Freeswitch-users] Updating caller id with attended transfer while
	phone still ringing
Message-ID: <CAEMJzSzU-mTNHrXmV0LxC0jREOrqSgY+K8ega_AGp8s2W3EoTg@mail.gmail.com>

Hi,

When doing attended transfer (A -> B -> attended transfer -> C) C only
updates its caller id display when it answers the call. Is there anyway to
update caller id on C, when B transfer the call while its still ringing?

The problem is we have some Cisco and Samsung SIP phones that does not have
a real "Blind Transfer" support. If you want to transfer without talking to
"C", you just press the transfer button quickly, while C starts to ring. So
I want to send the updated caller id to "C" after "B" transferred but
before "C" answers.

I have Yeastar PBX (asterisk based) works just like that but I can't
figure out how to do it with freeswitch.

Regards,

Rahman Duran
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211028/661722ef/attachment-0003.html>

From rahman.duran at erzurum.edu.tr  Mon Oct 25 06:52:59 2021
From: rahman.duran at erzurum.edu.tr (Rahman Duran)
Date: Mon, 25 Oct 2021 09:52:59 +0300
Subject: [Freeswitch-users] Codec problem,
	Freeswitch does not regard absolute_codec_string?
In-Reply-To: <CAEMJzSxTL3sgQR6830P8LA+ZARSf8SBNDN1mWg+G+c102ajx7w@mail.gmail.com>
References: <CAEMJzSxTL3sgQR6830P8LA+ZARSf8SBNDN1mWg+G+c102ajx7w@mail.gmail.com>
Message-ID: <CAEMJzSz9ujt0iK7vG5MMKJh8xmsanN92agrq3pVwuPRv+qwxZA@mail.gmail.com>

Hi,

I am struggling tuning Freeswitch for late codec negotiation. Here is local
profile settings:

<param name="inbound-late-negotiation" value="true"/>
<param name="inbound-codec-prefs" value="G722,PCMA"/>
<param name="outbound-codec-prefs" value="G722,PCMA"/>
<param name="inbound-codec-negotiation" value="greedy"/>

As you see I enabled late-negotiation and set codecs prefs to allow G722
and PCMA. My goal is to use G722 if both sides support it and fall back to
PCMA if G722 is not an option for A leg or B leg.

Here is the local dial plan config where I bridge calls:

    <extension name="local_accounts_freeswitch">
      <condition regex="all">
        <regex field="destination_number" expression="^[0-9]{4}$"/>
        <regex field="${local_user_exists}"
expression="^true$|^TRUE$|^True$"/>
        <regex field="${santral_kontrol}" expression="^freeswitch$"/>
        <action application="set" inline="true"
data="call_pickup_group=${user_data(${destination_number}@${domain_name}
var call_pickup_group)}"/>
        <action application="hash"
data="insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}"/>
        <action application="hash"
data="insert/${domain_name}-group_pickup_last_uuid/${call_pickup_group}/${uuid}"/>
        <action application="hash"
data="insert/${domain_name}-pickup_last_uuid/${destination_number}/${uuid}"/>
        <action application="log" data="ep_codec_string --
${ep_codec_string}" />
        <action application="set" data="inherit_codec=true"/>
        <action application="export"
data="nolocal:absolute_codec_string='${ep_codec_string}'"/>
        <action application="set_profile_var"
data="callee_id_name=${user_data(${destination_number}@${domain_name} var
effective_caller_id_name)}"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="call_timeout=30"/>
        <action application="set" data="execute_on_answer=sched_hangup
+21600 alloted_timeout" />
        <action application="bridge"
data="{origination_callee_id_name=${user_data(${destination_number}@${domain_name}
var effective_caller_id_name)}}user/${destination_number}@${domain_name}"/>
        <action application="execute_extension"
data="hata-${originate_disposition} XML hata_anonslari"/>
      </condition>
    </extension>

So I export absolute_codec_string to B leg using ep_codec_string. When I
test a call from 2730->2110, the call fails with
"incompetible_destination".  Call logs here:
https://pastebin.freeswitch.org/view/b2978e79 The 2730 is a Cisco phone
with "G722,PCMU,PCMA,G729,L16,iLBC" codec support. The 2110 is a Samsung
phone that only supports "PCMA,PCMU,G729" codecs. As you see cisco invites
with all codecs it supports and freeswitch sets absolute_codec_string with
"G722,PCMA" as expected from the config. But as you see in the call logs
Freeswitch only sends "G722" in SDP to Samsung phone so Samsung rejects the
call.

Why does Freeswitch only send G722? How can I fix this? What am I doing
wrong?

P.S. using debian packages: FreeSWITCH Version
1.10.6-release-18-1ff9d0a60e~64bit (-release-18-1ff9d0a60e 64bit)

Regards,

Rahman Duran
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211025/7785fb23/attachment-0003.html>

From freeswitch13 at mailbox.org  Sat Oct 30 06:52:38 2021
From: freeswitch13 at mailbox.org (Gregor Maier)
Date: Sat, 30 Oct 2021 08:52:38 +0200
Subject: REGISTER with Telekom SIP Trunk
In-Reply-To: <8844a4633ca043d38ae46564467b9e55@secunet.com>
References: <8844a4633ca043d38ae46564467b9e55@secunet.com>
Message-ID: <54ee785e-c49b-c3c1-a6b0-e5e8991dce22@mailbox.org>

Hello Johannes, hello list!

I just found your post because I'm trying to achieve the same thing waht's already 
implemented for asterisk long time ago.

A few things to clarify the goal of the desired change a bit more:

- The standard registration with Telekom AllIP works just fine (regarding SIP over
   TLS e.g.)
- But if you additionally want to use encrypted RTP, you have to go through
   mediasec enabled registration. That's what you described partly below.


I want to go a bit more into deep how mediasec works at Deutsche Telekom. It's an 
extension of the regular registration process. It works like this (RFC3329) (only 
the additional headers of the regular register added):


(1) ----> (Request sent by FS)
REGISTER sip:tel.t-online.de SIP/2.0
CSeq: 12735 REGISTER
...
Security-Client: sdes-srtp;mediasec
Proxy-Require: mediasec
Require: mediasec

(2) <---- (Response sent to FS)
SIP/2.0 494 Security Agreement Required
CSeq: 12735 REGISTER
...
Security-Server: msrp-tls;mediasec
Security-Server: sdes-srtp;mediasec
Security-Server: dtls-srtp;mediase

(3) ---->
REGISTER sip:tel.t-online.de SIP/2.0
CSeq: 12736 REGISTER
...
Security-Verify: msrp-tls;mediasec
Security-Verify: sdes-srtp;mediasec
Security-Verify: dtls-srtp;mediasec

(4) <-----
SIP/2.0 401 Unauthorized 11030030345
CSeq: 12736 REGISTER
...
Security-Server: msrp-tls;mediasec
Security-Server: sdes-srtp;mediasec
Security-Server: dtls-srtp;mediasec

(5) ---->
REGISTER sip:tel.t-online.de SIP/2.0
CSeq: 12737 REGISTER
...
Security-Verify: msrp-tls;mediasec
Security-Verify: sdes-srtp;mediasec
Security-Verify: dtls-srtp;mediasec

(6) <----
SIP/2.0 200 OK
...


To summarize it:
mediasec adds one additional request / response process at the beginning of the 
known registration process and a few additional headers:
Requesting mediasec (it's the known register start plus additional headers) and 
the response is 494 Security Agreement Required. From now on registration works as 
usual except for those three additional headers.

I searched and found lots of documentation. Therefore I found a pretty easy way to 
add those additional headers (in the C code) and I'm able to start the mediasec 
process. Unfortunately, I'm not able to handle / route this additional 
request/response process right at the beginning, because sofia doesn't provide any 
means to modify the registration process based on freeswitch (id est: without 
changing sofia). Is this correct? Or is there nevertheless a way to integrate this 
additional request / response because I overlooked something?

I am able to get the 494 in freeswitch, but I don't know how to exactly proceed 
because sofia doesn't know anything about this process and sofia has its own 
definition on how to do registration. Maybe some of you do have any idea?

Basically it's pretty easy: after the first preceeding request / response, the 
"real" registration has to be (re)started. That's how it's possible in asterisk.

The key point is, that the "restarted" registration process must know about the 
fact that the mediasec process has already been done, because after the 494, you 
have to set three other headers in the register as before (compare (1) and (3) above).

The registration process is started with nua_register(). It probably would work if 
it would be possible to just silently (= do not send anything to the SIP provider 
at all) end the complete registration process after the 494 and immediately 
restart it again right from the beginning (plus three additional headers) - at 
this moment, the registration process itself works normally as sofia expects it.


If registration proceeds, the next and last step is to add the
a=3ge2ae:requested
to the outgoing invite's SDP. At the moment, I don't know how to do it - but the 
reason is pretty simple: I first have to solve registration - afterwards let's do 
the next step.



Thanks for any help!
Gregor

On 17.08.21 at 09:00 Schäffner, Johannes wrote:
> Hello,
> 
> 
> 
> I am trying to register with a Deutsche Telekom SIP Trunk.
> 
> The Deutsche Telekom technical documentation states that the client must set the following SIP header fields:
> 
> 
> 
> For an initial REGISTER without Authentication Challenge the SIP Header fields
> 
> - Security-Client: sdes-srtp;mediasec
> 
> - Proxy Request: mediasec
> 
> - Requirement: mediasec
> 
> shall be included. The platform replies with SIP 401 Unauthorised and contains the SIP header fields
> 
> - Security server: msrp-tls;mediasec
> 
> - Security server: sdes-srtp;mediasec
> 
> - Security server: dtls-srtp;mediasec
> 
> with indication of the possible encryption methods. For the following REGISTER with Authentication Challenge,
> 
> in addition to the originally included SIP header fields Security-Client, Proxy-Require, Require the SIP header fields
> 
> - Security Check: msrp-tls;mediasec
> 
> - Security check: sdes-srtp;mediasec
> 
> - Security check: dtls-srtp;mediasec
> 
> shall also be included. According to RFC3261, the Security-Verify header fields can also be combined as a single header field.
> 
> 
> 
> I have seen, that the Sofia-Stack has implemented these Fields.
> 
> But I have not found a way to set these header fields in the REGISTER message.
> 
> (For the INVITE messages I'm using the set action sip_h_*; not sure if this is correct.
> 
> There I have the Problem, that I'm not able to set the SDP Attribute
> 
> a=3ge2ae:requested)
> 
> 
> 
> 
> 
> Regards, Johannes


From Antony.Stone at freeswitch.open.source.it  Fri Oct 29 15:21:46 2021
From: Antony.Stone at freeswitch.open.source.it (Antony Stone)
Date: Fri, 29 Oct 2021 17:21:46 +0200
Subject: [Freeswitch-users] Developers / packagers here?
Message-ID: <202110291721.46635.Antony.Stone@freeswitch.open.source.it>

Hi.

Can anyone tell me why 
https://freeswitch.org/confluence/display/FREESWITCH/mod_odbc_query is not 
built as part of the standard Debian packages available from 
https://files.freeswitch.org/repo/deb/ ?

Thanks,


Antony.

(PS: This message sent at 15:21 UTC 2021-10-29, just wondering when it will 
appear on the list.)

-- 
90% of networking problems are routing problems.
9 of the remaining 10% are routing problems in the other direction.
The remaining 1% might be something else, but check the routing anyway.

                                                   Please reply to the list;
                                                         please *don't* CC me.


From Antony.Stone at freeswitch.open.source.it  Fri Oct 29 09:04:24 2021
From: Antony.Stone at freeswitch.open.source.it (Antony Stone)
Date: Fri, 29 Oct 2021 11:04:24 +0200
Subject: [Freeswitch-users] Changing device on one leg improves the
	audio recorded from another leg?
In-Reply-To: <CAE23trumahY8d3=HQm=9bh2deJgTk_01kO1cn_oXcmkKCZW9hg@mail.gmail.com>
References: <CAE23trumahY8d3=HQm=9bh2deJgTk_01kO1cn_oXcmkKCZW9hg@mail.gmail.com>
Message-ID: <202110291104.24468.Antony.Stone@freeswitch.open.source.it>

On Friday 29 October 2021 at 02:26:15, David P wrote:

> How could changing the device on leg A affect the audio recording that
> apparently comes from leg B?

Any chance the devices performed a codec re-negotiation when swapped over?

I've seen (or heard) somethng very similar (although using Asterisk) when:

 - answering a call (audio is perfect)
 - placing the caller on hold (hold music is fine)
 - resuming the call (audio is essentially incomprehensible)

note: no change in devices at all.

It turned out that the "resume" after hold was re-negotiating the codec in the 
SDP.


Antony.

-- 
Python is executable pseudocode.
Perl is executable line noise.

                                                   Please reply to the list;
                                                         please *don't* CC me.


From Antony.Stone at freeswitch.open.source.it  Thu Oct 28 19:08:10 2021
From: Antony.Stone at freeswitch.open.source.it (Antony Stone)
Date: Thu, 28 Oct 2021 21:08:10 +0200
Subject: [Freeswitch-users] Trying to get short hostname
In-Reply-To: <0100017cc7851cbe-539c6003-da03-4c4b-a6d2-046f0031a3e7-000000@email.amazonses.com>
References: <202110281450.39060.Antony.Stone@freeswitch.open.source.it>
 <0100017cc7851cbe-539c6003-da03-4c4b-a6d2-046f0031a3e7-000000@email.amazonses.com>
Message-ID: <202110282108.11163.Antony.Stone@freeswitch.open.source.it>

On Thursday 28 October 2021 at 17:29:22, Avi Marcus wrote:

> System doesn't listen to stdout, it's only for running the command.

Er, sorry?

https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+system

And anyway, what's wrong with my regex?


Antony.

> On Thu, Oct 28, 2021, 6:22 PM Antony Stone wrote:
> > Hi.
> > 
> > I'm using FreeSwitch 1.10.7 as a client registering to other PBXs as well
> > as a SIP server itself.
> > 
> > I am setting the user-agent string in the SIP profile, such as:
> > 
> > <param name="user-agent-string" value="FS $${CFGver} $${hostname}"/>
> > 
> > The variable $${CFGver}} is a version string I'm definiing in vars.xml
> > 
> > This is working, but it gives me the FQDN of the machine as $${hostname}}
> > and I really only want the machine name without the domain.
> > 
> > I've found
> > https://freeswitch.org/confluence/display/FREESWITCH/Regular+Expression
> > and
> > https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+regex
> > and I've tried to use them to cut the domain off the value of $${hostname}
> > but with no success:
> > 
> > In vars.xml:
> > <X-PRE-PROCESS cmd="set"
> > data="shorthostname=${regex($${hostname}|([^\.]*).*|
> > %1)}"/>
> > 
> > This results in the variable shorthostname being assigned the value:
> >         %{regex(abcd.efgh.tld|([^\.]*).*|%1)}
> > 
> > where abcd.efgh.tld is the server's FQDN.
> > 
> > Instead of doing that in vars.xml and creating the variable
> > shorthostname, I tried putting the regex into the SIP profile:
> > 
> > <param name="user-agent-string" value="FS $${CFGver}
> > ${regex($${hostname}| ([^\.]*).*|%1)}"/>
> > 
> > This ends up with precisely the same result.  The ${regex(....)} function
> > simply isn't being executed.
> > 
> > 
> > Given that the machine's hostname is also available from the system
> > command "hostname -s" I decided to try that instead:
> > 
> > In vars.xml:
> > <X-PRE-PROCESS cmd="set" data="shorthostname=${system(hostname -s)}"/>
> > 
> > Or in the SIP profile definition:
> > <param name="user-agent-string" value="FS $${CFGver} ${system(hostname
> > -s)}"/>
> > 
> > These both produce the output:
> >         FS 0.65 ${system(hostname -s)}
> > 
> > In other words, the ${system(....)} command isn't working any more than
> > ${regex(....)} did.
> > 
> > 
> > What am I doing wrong please?
> > 
> > 
> > How can I get the machine's short hostname into a user-agent string for
> > registration to a server?
> > 
> > 
> > Thanks,
> > 
> > 
> > Antony.

-- 
My life is going completely according to plan.

I do sometimes wish it had been *my* plan, though.

                                                   Please reply to the list;
                                                         please *don't* CC me.