From svanherwaarden at precisionag.org  Mon Mar  1 10:45:16 2021
From: svanherwaarden at precisionag.org (Sam van Herwaarden)
Date: Mon, 1 Mar 2021 11:45:16 +0100
Subject: [Freeswitch-users] "Shadow calls" - duplicate calls executing
	in parallel
In-Reply-To: <CAE23truFiM4GGceRerdzcQt8Oq9EPMQWHN8Bmi1BNzcgiMTJ9Q@mail.gmail.com>
References: <mailman.1388.1614368359.1264.freeswitch-users@lists.freeswitch.org>
 <CAE23truFiM4GGceRerdzcQt8Oq9EPMQWHN8Bmi1BNzcgiMTJ9Q@mail.gmail.com>
Message-ID: <CAAtWTekYGSkCgKgFZiHYr4WnQkuqWmdT0a6qA3Q5DyvDewVJ1g@mail.gmail.com>

I see. I've seen this happening both on systems using SIP and systems using
PRI (no RTC involved), so it might be something else. Unfortunately I'm not
in a position to upgrade some of the systems (we don't run all of them) so
I guess I'll just have to live with this.

Best,
Sam

On Sat, Feb 27, 2021 at 3:36 AM David P <davidswalkabout at gmail.com> wrote:

> I think the last time we experienced that was with 10.4 (we're on 10.5
> now).
>
> It happened infrequently as you note. I assumed it was an error in the way
> our WebRTC app invoked verto. (I didn't have any browser console log to
> confirm that, though.)
>
> On Sat, 27 Feb 2021, 9:07 am , <
> freeswitch-users-request at lists.freeswitch.org> wrote:
>
>>
>>
>> ---------- Forwarded message ----------
>> From: Sam van Herwaarden <svanherwaarden at precisionag.org>
>>
>> Hi all,
>>
>> I was just wondering if others have experienced this: on a few of the
>> FreeSWITCH systems I've interacted with, I've observed situations where
>> incoming phone calls seem to be duplicated. So there seem to be two call
>> processes for the same phone number, also both receiving things like DTMF
>> events. The timestamps of events tend to be very close (sub-second) but not
>> necessarily identical.
>>
>> It's not very common, less than 1 per 1000 calls has this happening. It
>> does tend to throw off some of our tooling (both in call handling and
>> analysis). Curious to know if there could be an obvious cause/easy fix.
>>
>> Kind regards,
>> Sam
>>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From brian at freeswitch.com  Mon Mar  1 15:22:25 2021
From: brian at freeswitch.com (Brian West)
Date: Mon, 1 Mar 2021 09:22:25 -0600
Subject: [Freeswitch-users] B-leg ringing after A-leg hangs-up. LUA.
In-Reply-To: <CABGrKasrUzAMdUmP2yk_-5KBeCpnzXzdF84Sns6gVE6iH3M2Fw@mail.gmail.com>
References: <CABGrKasLSksOF_2zD5i-FqBYN2VsTJ101-ONiWEsqW3samakSQ@mail.gmail.com>
 <CAEJMVkDg8V7DLt0nJaqtSN+B7MB47Cp769wr2vgXoEXHZZ3e-w@mail.gmail.com>
 <CABGrKasrUzAMdUmP2yk_-5KBeCpnzXzdF84Sns6gVE6iH3M2Fw@mail.gmail.com>
Message-ID: <CAEJMVkDtmj6WCEQ80Tqr4d3kmSJBwgMFCc7yv16FMb-b4pynbg@mail.gmail.com>

That's because the second legs Lua is running in the first legs session
thread.  So it can't hang up yet.

On Fri, Feb 26, 2021 at 3:40 PM Тимур Ситдиков <sitdikovt at gmail.com> wrote:

> Hi Brian!
> Yes, B hangs up right after pick up. Also there's an error about non
> available channels in console.
>
>
>
> сб, 27 февр. 2021 г., 02:07 Brian West <brian at freeswitch.com>:
>
>> Tim,
>>
>> Based on your description, I would guess that the B-Leg hangs up when
>> they answer after the a-leg is gone??
>>
>> /b
>>
>>
>>
>>
>> On Fri, Feb 26, 2021 at 12:42 PM Тимур Ситдиков <sitdikovt at gmail.com>
>> wrote:
>>
>>> Hi all. Need help with a simple call script.
>>> I'm got this in dialplan
>>> <extension name="call_lua_home">
>>>       <condition field="destination_number" expression="^(1007)$">
>>>         <!-- <action application="set" data="session_in_hangup_hook=true"/> -->
>>> For hangup hook try
>>>         <action application="lua" data="fork.lua"/>
>>>       </condition>
>>> </extension>
>>> And this in fork.lua
>>> if session:ready() then
>>> api = freeswitch.API()
>>> contact = api:execute("sofia_contact", "*/1007 at compA.com");
>>> caller = session:getVariable("caller_id_number");
>>> session1 = freeswitch.Session("{origination_caller_id_name="..caller..
>>> "}[leg_timeout=30]"..contact.."");
>>> -- session:setHangupHook("HangupHook");
>>> if session1:ready() then
>>>     freeswitch.bridge(session, session1);
>>> end
>>> freeswitch.consoleLog("WARNING","=====SCRIPT END\n");
>>> session:hangup()
>>> end
>>>
>>> When Caller is hanging up before B-leg answers - Bleg is continuing to
>>> ring. There are still 2 channels in 'show channels' after A-leg hangs up.
>>> I want to drop B-leg when A-leg hangs up. Is it possible?
>>>
>>> UPD Tried to use hangup hook. It works with stramFile, but no luck with
>>> session.
>>> function HangupHook(s, status)
>>>     freeswitch.consoleLog("WARNING","Event fired breaking out\n");
>>>     return exit;
>>>     -- return die;
>>> end
>>>
>>> session:answer();
>>> session:setHangupHook("HangupHook");
>>> while (session:ready() == true) do
>>>
>>>     session:streamFile(
>>> "/usr/share/freeswitch/sounds/en/us/callie/ivr/16000/ivr-on_hold_indefinitely.wav"
>>> );
>>>
>>> -- session1 = freeswitch.Session("{origination_caller_id_name="..caller.."}[leg_timeout=30]"..contact.."");
>>> end
>>> freeswitch.consoleLog("WARNING","=====SCRIPT END\n");
>>> session:hangup()
>>> end
>>>
>>> Can anyone help me with this?
>>>
>>> Thanks! Regards,Tim
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>>
>> --
>>
>> Brian West | Co-founder and Developer
>>
>> Need Commercial support? email sales at freeswitch.com
>>
>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>
>> Email: brian at freeswitch.com
>>
>> Mobile: 918-424-9378
>>
>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>
>> [image: https://www.facebook.com/signalwireinc?src=email]
>> <https://www.facebook.com/freeswitch> [image:
>> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users 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>
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From sitdikovt at gmail.com  Tue Mar  2 04:50:25 2021
From: sitdikovt at gmail.com (=?UTF-8?B?0KLQuNC80YPRgCDQodC40YLQtNC40LrQvtCy?=)
Date: Tue, 2 Mar 2021 10:50:25 +0600
Subject: [Freeswitch-users] B-leg ringing after A-leg hangs-up. LUA.
In-Reply-To: <CAEJMVkDtmj6WCEQ80Tqr4d3kmSJBwgMFCc7yv16FMb-b4pynbg@mail.gmail.com>
References: <CABGrKasLSksOF_2zD5i-FqBYN2VsTJ101-ONiWEsqW3samakSQ@mail.gmail.com>
 <CAEJMVkDg8V7DLt0nJaqtSN+B7MB47Cp769wr2vgXoEXHZZ3e-w@mail.gmail.com>
 <CABGrKasrUzAMdUmP2yk_-5KBeCpnzXzdF84Sns6gVE6iH3M2Fw@mail.gmail.com>
 <CAEJMVkDtmj6WCEQ80Tqr4d3kmSJBwgMFCc7yv16FMb-b4pynbg@mail.gmail.com>
Message-ID: <CABGrKav6puQOWX5wu7=6s_f8_8HXQCZOcK+FBmqQXh1XMfDGPw@mail.gmail.com>

Ok, got it :-)
I'm new to lua, got this script based on the FS 1.8 book example script.
Bleg session is in Aleg thread entirely in that example.

Anyway, I've separated sessions and it works. Thank you a lot, Brian!

Regards, Tim

пн, 1 мар. 2021 г. в 22:04, Brian West <brian at freeswitch.com>:

> That's because the second legs Lua is running in the first legs session
> thread.  So it can't hang up yet.
>
> On Fri, Feb 26, 2021 at 3:40 PM Тимур Ситдиков <sitdikovt at gmail.com>
> wrote:
>
>> Hi Brian!
>> Yes, B hangs up right after pick up. Also there's an error about non
>> available channels in console.
>>
>>
>>
>> сб, 27 февр. 2021 г., 02:07 Brian West <brian at freeswitch.com>:
>>
>>> Tim,
>>>
>>> Based on your description, I would guess that the B-Leg hangs up when
>>> they answer after the a-leg is gone??
>>>
>>> /b
>>>
>>>
>>>
>>>
>>> On Fri, Feb 26, 2021 at 12:42 PM Тимур Ситдиков <sitdikovt at gmail.com>
>>> wrote:
>>>
>>>> Hi all. Need help with a simple call script.
>>>> I'm got this in dialplan
>>>> <extension name="call_lua_home">
>>>>       <condition field="destination_number" expression="^(1007)$">
>>>>         <!-- <action application="set" data="session_in_hangup_hook=true"/> -->
>>>> For hangup hook try
>>>>         <action application="lua" data="fork.lua"/>
>>>>       </condition>
>>>> </extension>
>>>> And this in fork.lua
>>>> if session:ready() then
>>>> api = freeswitch.API()
>>>> contact = api:execute("sofia_contact", "*/1007 at compA.com");
>>>> caller = session:getVariable("caller_id_number");
>>>> session1 = freeswitch.Session("{origination_caller_id_name="..caller..
>>>> "}[leg_timeout=30]"..contact.."");
>>>> -- session:setHangupHook("HangupHook");
>>>> if session1:ready() then
>>>>     freeswitch.bridge(session, session1);
>>>> end
>>>> freeswitch.consoleLog("WARNING","=====SCRIPT END\n");
>>>> session:hangup()
>>>> end
>>>>
>>>> When Caller is hanging up before B-leg answers - Bleg is continuing to
>>>> ring. There are still 2 channels in 'show channels' after A-leg hangs up.
>>>> I want to drop B-leg when A-leg hangs up. Is it possible?
>>>>
>>>> UPD Tried to use hangup hook. It works with stramFile, but no luck with
>>>> session.
>>>> function HangupHook(s, status)
>>>>     freeswitch.consoleLog("WARNING","Event fired breaking out\n");
>>>>     return exit;
>>>>     -- return die;
>>>> end
>>>>
>>>> session:answer();
>>>> session:setHangupHook("HangupHook");
>>>> while (session:ready() == true) do
>>>>
>>>>     session:streamFile(
>>>> "/usr/share/freeswitch/sounds/en/us/callie/ivr/16000/ivr-on_hold_indefinitely.wav"
>>>> );
>>>>
>>>> -- session1 = freeswitch.Session("{origination_caller_id_name="..caller.."}[leg_timeout=30]"..contact.."");
>>>> end
>>>> freeswitch.consoleLog("WARNING","=====SCRIPT END\n");
>>>> session:hangup()
>>>> end
>>>>
>>>> Can anyone help me with this?
>>>>
>>>> Thanks! Regards,Tim
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>>
>>>
>>> --
>>>
>>> Brian West | Co-founder and Developer
>>>
>>> Need Commercial support? email sales at freeswitch.com
>>>
>>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
>>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>>
>>> Email: brian at freeswitch.com
>>>
>>> Mobile: 918-424-9378
>>>
>>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>>
>>> [image: https://www.facebook.com/signalwireinc?src=email]
>>> <https://www.facebook.com/freeswitch> [image:
>>> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users 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
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From sitdikovt at gmail.com  Tue Mar  2 05:25:27 2021
From: sitdikovt at gmail.com (=?UTF-8?B?0KLQuNC80YPRgCDQodC40YLQtNC40LrQvtCy?=)
Date: Tue, 2 Mar 2021 11:25:27 +0600
Subject: [Freeswitch-users] B-leg ringing after A-leg hangs-up. LUA.
In-Reply-To: <CAEJMVkDtmj6WCEQ80Tqr4d3kmSJBwgMFCc7yv16FMb-b4pynbg@mail.gmail.com>
References: <CABGrKasLSksOF_2zD5i-FqBYN2VsTJ101-ONiWEsqW3samakSQ@mail.gmail.com>
 <CAEJMVkDg8V7DLt0nJaqtSN+B7MB47Cp769wr2vgXoEXHZZ3e-w@mail.gmail.com>
 <CABGrKasrUzAMdUmP2yk_-5KBeCpnzXzdF84Sns6gVE6iH3M2Fw@mail.gmail.com>
 <CAEJMVkDtmj6WCEQ80Tqr4d3kmSJBwgMFCc7yv16FMb-b4pynbg@mail.gmail.com>
Message-ID: <CABGrKaswhu8Ou+YAkfdyT5L6ef8r3cY0YaH-9kCWN4ZsDLhC_g@mail.gmail.com>

Sorry, I made a mistake in my dialplan when I sent the previous message.

My try to separate threads is not working :-(
Still ringing after Aleg hangs up.
Trying to do this
if session:ready() then
api = freeswitch.API()
contact = api:execute("sofia_contact", "*/1007 at compA.com");
caller = session:getVariable("caller_id_number");
session:execute("ring_ready")
session:setVariable("ringback", "%(2000,4000,440.0,480.0)");
end

session1 = freeswitch.Session("{origination_caller_id_name="..caller..
"}[leg_timeout=30]"..contact.."");
if session1:ready() then
    freeswitch.bridge(session, session1);
end

I'm totally noob in Lua. Can someone please point where to dig, or show an
example script?
Thank you!

Tim

пн, 1 мар. 2021 г. в 22:04, Brian West <brian at freeswitch.com>:

> That's because the second legs Lua is running in the first legs session
> thread.  So it can't hang up yet.
>
> On Fri, Feb 26, 2021 at 3:40 PM Тимур Ситдиков <sitdikovt at gmail.com>
> wrote:
>
>> Hi Brian!
>> Yes, B hangs up right after pick up. Also there's an error about non
>> available channels in console.
>>
>>
>>
>> сб, 27 февр. 2021 г., 02:07 Brian West <brian at freeswitch.com>:
>>
>>> Tim,
>>>
>>> Based on your description, I would guess that the B-Leg hangs up when
>>> they answer after the a-leg is gone??
>>>
>>> /b
>>>
>>>
>>>
>>>
>>> On Fri, Feb 26, 2021 at 12:42 PM Тимур Ситдиков <sitdikovt at gmail.com>
>>> wrote:
>>>
>>>> Hi all. Need help with a simple call script.
>>>> I'm got this in dialplan
>>>> <extension name="call_lua_home">
>>>>       <condition field="destination_number" expression="^(1007)$">
>>>>         <!-- <action application="set" data="session_in_hangup_hook=true"/> -->
>>>> For hangup hook try
>>>>         <action application="lua" data="fork.lua"/>
>>>>       </condition>
>>>> </extension>
>>>> And this in fork.lua
>>>> if session:ready() then
>>>> api = freeswitch.API()
>>>> contact = api:execute("sofia_contact", "*/1007 at compA.com");
>>>> caller = session:getVariable("caller_id_number");
>>>> session1 = freeswitch.Session("{origination_caller_id_name="..caller..
>>>> "}[leg_timeout=30]"..contact.."");
>>>> -- session:setHangupHook("HangupHook");
>>>> if session1:ready() then
>>>>     freeswitch.bridge(session, session1);
>>>> end
>>>> freeswitch.consoleLog("WARNING","=====SCRIPT END\n");
>>>> session:hangup()
>>>> end
>>>>
>>>> When Caller is hanging up before B-leg answers - Bleg is continuing to
>>>> ring. There are still 2 channels in 'show channels' after A-leg hangs up.
>>>> I want to drop B-leg when A-leg hangs up. Is it possible?
>>>>
>>>> UPD Tried to use hangup hook. It works with stramFile, but no luck with
>>>> session.
>>>> function HangupHook(s, status)
>>>>     freeswitch.consoleLog("WARNING","Event fired breaking out\n");
>>>>     return exit;
>>>>     -- return die;
>>>> end
>>>>
>>>> session:answer();
>>>> session:setHangupHook("HangupHook");
>>>> while (session:ready() == true) do
>>>>
>>>>     session:streamFile(
>>>> "/usr/share/freeswitch/sounds/en/us/callie/ivr/16000/ivr-on_hold_indefinitely.wav"
>>>> );
>>>>
>>>> -- session1 = freeswitch.Session("{origination_caller_id_name="..caller.."}[leg_timeout=30]"..contact.."");
>>>> end
>>>> freeswitch.consoleLog("WARNING","=====SCRIPT END\n");
>>>> session:hangup()
>>>> end
>>>>
>>>> Can anyone help me with this?
>>>>
>>>> Thanks! Regards,Tim
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>>
>>>
>>> --
>>>
>>> Brian West | Co-founder and Developer
>>>
>>> Need Commercial support? email sales at freeswitch.com
>>>
>>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
>>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>>
>>> Email: brian at freeswitch.com
>>>
>>> Mobile: 918-424-9378
>>>
>>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>>
>>> [image: https://www.facebook.com/signalwireinc?src=email]
>>> <https://www.facebook.com/freeswitch> [image:
>>> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users 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
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From brian at freeswitch.com  Tue Mar  2 14:21:52 2021
From: brian at freeswitch.com (Brian West)
Date: Tue, 2 Mar 2021 08:21:52 -0600
Subject: [Freeswitch-users] B-leg ringing after A-leg hangs-up. LUA.
In-Reply-To: <CABGrKaswhu8Ou+YAkfdyT5L6ef8r3cY0YaH-9kCWN4ZsDLhC_g@mail.gmail.com>
References: <CABGrKasLSksOF_2zD5i-FqBYN2VsTJ101-ONiWEsqW3samakSQ@mail.gmail.com>
 <CAEJMVkDg8V7DLt0nJaqtSN+B7MB47Cp769wr2vgXoEXHZZ3e-w@mail.gmail.com>
 <CABGrKasrUzAMdUmP2yk_-5KBeCpnzXzdF84Sns6gVE6iH3M2Fw@mail.gmail.com>
 <CAEJMVkDtmj6WCEQ80Tqr4d3kmSJBwgMFCc7yv16FMb-b4pynbg@mail.gmail.com>
 <CABGrKaswhu8Ou+YAkfdyT5L6ef8r3cY0YaH-9kCWN4ZsDLhC_g@mail.gmail.com>
Message-ID: <CAEJMVkCus++TUS51hRsqLJAJjfoVO-dQ3fHLTK9aZrcKDUKG9A@mail.gmail.com>

Try wrapping it in a while session:ready() then

On Mon, Mar 1, 2021 at 11:37 PM Тимур Ситдиков <sitdikovt at gmail.com> wrote:

> Sorry, I made a mistake in my dialplan when I sent the previous message.
>
> My try to separate threads is not working :-(
> Still ringing after Aleg hangs up.
> Trying to do this
> if session:ready() then
> api = freeswitch.API()
> contact = api:execute("sofia_contact", "*/1007 at compA.com");
> caller = session:getVariable("caller_id_number");
> session:execute("ring_ready")
> session:setVariable("ringback", "%(2000,4000,440.0,480.0)");
> end
>
> session1 = freeswitch.Session("{origination_caller_id_name="..caller..
> "}[leg_timeout=30]"..contact.."");
> if session1:ready() then
>     freeswitch.bridge(session, session1);
> end
>
> I'm totally noob in Lua. Can someone please point where to dig, or show an
> example script?
> Thank you!
>
> Tim
>
> пн, 1 мар. 2021 г. в 22:04, Brian West <brian at freeswitch.com>:
>
>> That's because the second legs Lua is running in the first legs session
>> thread.  So it can't hang up yet.
>>
>> On Fri, Feb 26, 2021 at 3:40 PM Тимур Ситдиков <sitdikovt at gmail.com>
>> wrote:
>>
>>> Hi Brian!
>>> Yes, B hangs up right after pick up. Also there's an error about non
>>> available channels in console.
>>>
>>>
>>>
>>> сб, 27 февр. 2021 г., 02:07 Brian West <brian at freeswitch.com>:
>>>
>>>> Tim,
>>>>
>>>> Based on your description, I would guess that the B-Leg hangs up when
>>>> they answer after the a-leg is gone??
>>>>
>>>> /b
>>>>
>>>>
>>>>
>>>>
>>>> On Fri, Feb 26, 2021 at 12:42 PM Тимур Ситдиков <sitdikovt at gmail.com>
>>>> wrote:
>>>>
>>>>> Hi all. Need help with a simple call script.
>>>>> I'm got this in dialplan
>>>>> <extension name="call_lua_home">
>>>>>       <condition field="destination_number" expression="^(1007)$">
>>>>>         <!-- <action application="set" data="session_in_hangup_hook=true"/> -->
>>>>> For hangup hook try
>>>>>         <action application="lua" data="fork.lua"/>
>>>>>       </condition>
>>>>> </extension>
>>>>> And this in fork.lua
>>>>> if session:ready() then
>>>>> api = freeswitch.API()
>>>>> contact = api:execute("sofia_contact", "*/1007 at compA.com");
>>>>> caller = session:getVariable("caller_id_number");
>>>>> session1 = freeswitch.Session("{origination_caller_id_name="..caller..
>>>>> "}[leg_timeout=30]"..contact.."");
>>>>> -- session:setHangupHook("HangupHook");
>>>>> if session1:ready() then
>>>>>     freeswitch.bridge(session, session1);
>>>>> end
>>>>> freeswitch.consoleLog("WARNING","=====SCRIPT END\n");
>>>>> session:hangup()
>>>>> end
>>>>>
>>>>> When Caller is hanging up before B-leg answers - Bleg is continuing to
>>>>> ring. There are still 2 channels in 'show channels' after A-leg hangs up.
>>>>> I want to drop B-leg when A-leg hangs up. Is it possible?
>>>>>
>>>>> UPD Tried to use hangup hook. It works with stramFile, but no luck
>>>>> with session.
>>>>> function HangupHook(s, status)
>>>>>     freeswitch.consoleLog("WARNING","Event fired breaking out\n");
>>>>>     return exit;
>>>>>     -- return die;
>>>>> end
>>>>>
>>>>> session:answer();
>>>>> session:setHangupHook("HangupHook");
>>>>> while (session:ready() == true) do
>>>>>
>>>>>     session:streamFile(
>>>>> "/usr/share/freeswitch/sounds/en/us/callie/ivr/16000/ivr-on_hold_indefinitely.wav"
>>>>> );
>>>>>
>>>>> -- session1 = freeswitch.Session("{origination_caller_id_name="..caller.."}[leg_timeout=30]"..contact.."");
>>>>> end
>>>>> freeswitch.consoleLog("WARNING","=====SCRIPT END\n");
>>>>> session:hangup()
>>>>> end
>>>>>
>>>>> Can anyone help me with this?
>>>>>
>>>>> Thanks! Regards,Tim
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> Brian West | Co-founder and Developer
>>>>
>>>> Need Commercial support? email sales at freeswitch.com
>>>>
>>>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
>>>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>>>
>>>> Email: brian at freeswitch.com
>>>>
>>>> Mobile: 918-424-9378
>>>>
>>>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>>>
>>>> [image: https://www.facebook.com/signalwireinc?src=email]
>>>> <https://www.facebook.com/freeswitch> [image:
>>>> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>>
>> --
>>
>> Brian West | Co-founder and Developer
>>
>> Need Commercial support? email sales at freeswitch.com
>>
>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>
>> Email: brian at freeswitch.com
>>
>> Mobile: 918-424-9378
>>
>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>
>> [image: https://www.facebook.com/signalwireinc?src=email]
>> <https://www.facebook.com/freeswitch> [image:
>> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

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>
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From david.villasmil.work at gmail.com  Tue Mar  2 14:25:00 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 2 Mar 2021 14:25:00 +0000
Subject: [Freeswitch-users] Freeswitch selecting the public ip
Message-ID: <CAFGRPVpYxrtK0-JzXN5muammoFgAvu5Rux6+EaC=mOFevfvCag@mail.gmail.com>

Hello Guys,

Whenever freeswitch triggers an even, on the variable FreeSWITCH-IPv4:
Freeswitch always selects the public IP.
I made an implementation some time ago which grabs that variable and uses
it for something.
I need to have that variable as the private ip, not the public one, is this
possible at all?

Thanks

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From david.villasmil.work at gmail.com  Tue Mar  2 14:34:39 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 2 Mar 2021 14:34:39 +0000
Subject: [Freeswitch-users] Freeswitch selecting the public ip
In-Reply-To: <CAFGRPVpYxrtK0-JzXN5muammoFgAvu5Rux6+EaC=mOFevfvCag@mail.gmail.com>
References: <CAFGRPVpYxrtK0-JzXN5muammoFgAvu5Rux6+EaC=mOFevfvCag@mail.gmail.com>
Message-ID: <CAFGRPVpJ=8Af0pqyc9vDKdOpzaCeJcxxtS094AFC=kxHt44mBQ@mail.gmail.com>

Maybe It's important to point out that this happens when a call comes in
and i send that call to a conference, and on the incoming profile there is
no public ip on SIGNALLING, only on RTP.

Regards,

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


On Tue, Mar 2, 2021 at 2:25 PM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello Guys,
>
> Whenever freeswitch triggers an even, on the variable FreeSWITCH-IPv4:
> Freeswitch always selects the public IP.
> I made an implementation some time ago which grabs that variable and uses
> it for something.
> I need to have that variable as the private ip, not the public one, is
> this possible at all?
>
> Thanks
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
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From sitdikovt at gmail.com  Tue Mar  2 17:48:37 2021
From: sitdikovt at gmail.com (=?UTF-8?B?0KLQuNC80YPRgCDQodC40YLQtNC40LrQvtCy?=)
Date: Tue, 2 Mar 2021 23:48:37 +0600
Subject: [Freeswitch-users] B-leg ringing after A-leg hangs-up. LUA.
In-Reply-To: <CAEJMVkCus++TUS51hRsqLJAJjfoVO-dQ3fHLTK9aZrcKDUKG9A@mail.gmail.com>
References: <CABGrKasLSksOF_2zD5i-FqBYN2VsTJ101-ONiWEsqW3samakSQ@mail.gmail.com>
 <CAEJMVkDg8V7DLt0nJaqtSN+B7MB47Cp769wr2vgXoEXHZZ3e-w@mail.gmail.com>
 <CABGrKasrUzAMdUmP2yk_-5KBeCpnzXzdF84Sns6gVE6iH3M2Fw@mail.gmail.com>
 <CAEJMVkDtmj6WCEQ80Tqr4d3kmSJBwgMFCc7yv16FMb-b4pynbg@mail.gmail.com>
 <CABGrKaswhu8Ou+YAkfdyT5L6ef8r3cY0YaH-9kCWN4ZsDLhC_g@mail.gmail.com>
 <CAEJMVkCus++TUS51hRsqLJAJjfoVO-dQ3fHLTK9aZrcKDUKG9A@mail.gmail.com>
Message-ID: <CABGrKauCTaXG16Db3TEBpdtWGhaAvad1J5PNNQJPe6EV=oCC0Q@mail.gmail.com>

Sorry, same thing. Tried this already.
while (session:ready() == true) do

    api = freeswitch.API();
    contact = api:execute("sofia_contact", "*/1007 at compA.com");
    caller = session:getVariable("caller_id_number");

    session:execute("ring_ready")
    session:setVariable("ringback", "%(2000,4000,440.0,480.0)");


    session_B = freeswitch.Session("{origination_caller_id_name="..caller..
"}[leg_timeout=30]"..contact.."");

        if session_B:ready() then
            freeswitch.bridge(session, session_B);
        end

    session:hangup();
end
Tried to nest if not (session:ready) in this loop, and then break  - no
luck.

вт, 2 мар. 2021 г. в 20:58, Brian West <brian at freeswitch.com>:

> Try wrapping it in a while session:ready() then
>
> On Mon, Mar 1, 2021 at 11:37 PM Тимур Ситдиков <sitdikovt at gmail.com>
> wrote:
>
>> Sorry, I made a mistake in my dialplan when I sent the previous message.
>>
>> My try to separate threads is not working :-(
>> Still ringing after Aleg hangs up.
>> Trying to do this
>> if session:ready() then
>> api = freeswitch.API()
>> contact = api:execute("sofia_contact", "*/1007 at compA.com");
>> caller = session:getVariable("caller_id_number");
>> session:execute("ring_ready")
>> session:setVariable("ringback", "%(2000,4000,440.0,480.0)");
>> end
>>
>> session1 = freeswitch.Session("{origination_caller_id_name="..caller..
>> "}[leg_timeout=30]"..contact.."");
>> if session1:ready() then
>>     freeswitch.bridge(session, session1);
>> end
>>
>> I'm totally noob in Lua. Can someone please point where to dig, or show
>> an example script?
>> Thank you!
>>
>> Tim
>>
>> пн, 1 мар. 2021 г. в 22:04, Brian West <brian at freeswitch.com>:
>>
>>> That's because the second legs Lua is running in the first legs session
>>> thread.  So it can't hang up yet.
>>>
>>> On Fri, Feb 26, 2021 at 3:40 PM Тимур Ситдиков <sitdikovt at gmail.com>
>>> wrote:
>>>
>>>> Hi Brian!
>>>> Yes, B hangs up right after pick up. Also there's an error about non
>>>> available channels in console.
>>>>
>>>>
>>>>
>>>> сб, 27 февр. 2021 г., 02:07 Brian West <brian at freeswitch.com>:
>>>>
>>>>> Tim,
>>>>>
>>>>> Based on your description, I would guess that the B-Leg hangs up when
>>>>> they answer after the a-leg is gone??
>>>>>
>>>>> /b
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Fri, Feb 26, 2021 at 12:42 PM Тимур Ситдиков <sitdikovt at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Hi all. Need help with a simple call script.
>>>>>> I'm got this in dialplan
>>>>>> <extension name="call_lua_home">
>>>>>>       <condition field="destination_number" expression="^(1007)$">
>>>>>>         <!-- <action application="set" data="session_in_hangup_hook=true"/> -->
>>>>>> For hangup hook try
>>>>>>         <action application="lua" data="fork.lua"/>
>>>>>>       </condition>
>>>>>> </extension>
>>>>>> And this in fork.lua
>>>>>> if session:ready() then
>>>>>> api = freeswitch.API()
>>>>>> contact = api:execute("sofia_contact", "*/1007 at compA.com");
>>>>>> caller = session:getVariable("caller_id_number");
>>>>>> session1 = freeswitch.Session("{origination_caller_id_name="..caller
>>>>>> .."}[leg_timeout=30]"..contact.."");
>>>>>> -- session:setHangupHook("HangupHook");
>>>>>> if session1:ready() then
>>>>>>     freeswitch.bridge(session, session1);
>>>>>> end
>>>>>> freeswitch.consoleLog("WARNING","=====SCRIPT END\n");
>>>>>> session:hangup()
>>>>>> end
>>>>>>
>>>>>> When Caller is hanging up before B-leg answers - Bleg is continuing
>>>>>> to ring. There are still 2 channels in 'show channels' after A-leg hangs up.
>>>>>> I want to drop B-leg when A-leg hangs up. Is it possible?
>>>>>>
>>>>>> UPD Tried to use hangup hook. It works with stramFile, but no luck
>>>>>> with session.
>>>>>> function HangupHook(s, status)
>>>>>>     freeswitch.consoleLog("WARNING","Event fired breaking out\n");
>>>>>>     return exit;
>>>>>>     -- return die;
>>>>>> end
>>>>>>
>>>>>> session:answer();
>>>>>> session:setHangupHook("HangupHook");
>>>>>> while (session:ready() == true) do
>>>>>>
>>>>>>     session:streamFile(
>>>>>> "/usr/share/freeswitch/sounds/en/us/callie/ivr/16000/ivr-on_hold_indefinitely.wav"
>>>>>> );
>>>>>>
>>>>>> -- session1 = freeswitch.Session("{origination_caller_id_name="..caller.."}[leg_timeout=30]"..contact.."");
>>>>>> end
>>>>>> freeswitch.consoleLog("WARNING","=====SCRIPT END\n");
>>>>>> session:hangup()
>>>>>> end
>>>>>>
>>>>>> Can anyone help me with this?
>>>>>>
>>>>>> Thanks! Regards,Tim
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>
>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>> https://signalwire.com
>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>> services.
>>>>>> Build your next product on our scalable cloud platform.
>>>>>>
>>>>>> Join our online community to chat in real time
>>>>>> https://signalwire.community
>>>>>>
>>>>>> Professional FreeSWITCH Services
>>>>>> sales at freeswitch.com
>>>>>> https://freeswitch.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> https://freeswitch.com/oss
>>>>>> https://freeswitch.org/confluence
>>>>>> https://cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> https://freeswitch.com
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>> Brian West | Co-founder and Developer
>>>>>
>>>>> Need Commercial support? email sales at freeswitch.com
>>>>>
>>>>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
>>>>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>>>>
>>>>> Email: brian at freeswitch.com
>>>>>
>>>>> Mobile: 918-424-9378
>>>>>
>>>>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>>>>
>>>>> [image: https://www.facebook.com/signalwireinc?src=email]
>>>>> <https://www.facebook.com/freeswitch> [image:
>>>>> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>>
>>>
>>> --
>>>
>>> Brian West | Co-founder and Developer
>>>
>>> Need Commercial support? email sales at freeswitch.com
>>>
>>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
>>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>>
>>> Email: brian at freeswitch.com
>>>
>>> Mobile: 918-424-9378
>>>
>>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>>
>>> [image: https://www.facebook.com/signalwireinc?src=email]
>>> <https://www.facebook.com/freeswitch> [image:
>>> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
>
> 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
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From christian.berger at foncloud.net  Tue Mar  2 16:33:52 2021
From: christian.berger at foncloud.net (Christian Berger)
Date: Tue, 2 Mar 2021 17:33:52 +0100
Subject: [Freeswitch-users] NOTIFYs seem unreliable under high number of
	SUBSCRIBEs
Message-ID: <275f41f7-f6dd-7ab8-0be0-81d6aa509d6a@foncloud.net>

Hello,

we are currently seeing some strange behavior under load conditions. 
Notifications for calls are sent to seemingly random subsets of 
subscribed users. This already happens with the default configuration.
To recreate this we have created 10000 SUBSCRIBEs with sipp for 1002.
sipp -sf sip.xml $ip_of_freeswitch -m 10000 -r 100

Meanwhile sngrep is running to record the dialogues, and we have 2 
telephones registered to 1001 and 1002. When 1002 calls 1001, NOTIFYs 
will be sent out to the subscriptions. However when there are 10000 
subscriptions only about 3000-4000 of them will get the NOTIFY. Trying 
it again will cause another subset of the SUBSCRIBEs to be answered with 
a NOTIFY.

This test was conducted locally on an otherwise idle server, so network 
loss can be ruled out rather well.

Has anybody else had a similar experience? I couldn't find any way 
sending out NOTIFYs could be limited and the number of NOTIFYs we get 
seems to be to random for such a limit.

Thank you very much for your time
    Christian Berger foncloud.net


Christian Berger


 

foncloud GmbH & Co KG
Hahlweg 2a
36093 Künzell

Tel: / 
Fax: +49 661 968990-99

 

Email: Christian.Berger at foncloud.net
 Web: www.foncloud.net





P.S.: Wussten Sie schon? 

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

 


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From lewis.bergman at gmail.com  Tue Mar  2 01:30:06 2021
From: lewis.bergman at gmail.com (Lewis Bergman)
Date: Mon, 1 Mar 2021 19:30:06 -0600
Subject: [Freeswitch-users] mod_sndfile error called from
 mod_tts_commandline - [WARNING] mod_sndfile.c:281 Error Opening File
Message-ID: <CAD2CnaQHdQ4pL1uXaGumGS+pM6MGyos9s+w2EUXJD52T0goB0w@mail.gmail.com>

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

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

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

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

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

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

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

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

-- 
Lewis Bergman


From mayamatakeshi at gmail.com  Tue Mar  2 23:12:09 2021
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Wed, 3 Mar 2021 08:12:09 +0900
Subject: [Freeswitch-users] mod_sndfile error called from
 mod_tts_commandline - [WARNING] mod_sndfile.c:281 Error Opening File
In-Reply-To: <CAD2CnaQHdQ4pL1uXaGumGS+pM6MGyos9s+w2EUXJD52T0goB0w@mail.gmail.com>
References: <CAD2CnaQHdQ4pL1uXaGumGS+pM6MGyos9s+w2EUXJD52T0goB0w@mail.gmail.com>
Message-ID: <CABaNFCabu6equvag0nkwvLm+eDZNf3VLgnop4SpZ_PMWCm4qmw@mail.gmail.com>

On Wed, Mar 3, 2021 at 4:12 AM Lewis Bergman <lewis.bergman at gmail.com>
wrote:

> Hello all. I am pretty new to Freeswitch and have an experimental container
> on Proxmox 6.3 Debian 10 to play with. The goal is to get mod_unimrcp
> working. To that end, I have mod_tts_commandline (installed from binary
> with FusionPBX as part of the default install) listed in modules.conf.xml
> /etc/freeswitch/autoload_configs/modules.conf.xml
>                 <!-- Speech Recognition / Text to Speech -->
>                 <load module="mod_unimrcp"/>
>                 <load module="mod_tts_commandline"/>
>                 <load module="mod_flite"/>
> I have tried it with and without mod_flite enabled.
> Mod_tts_commandline refuses to pull a sound file and the debug log seems to
> blame mod_sndfile. This is using the example from the mod_tts_commandline
> page
> <
> https://freeswitch.org/confluence/display/FREESWITCH/mod_tts_commandline#mod_tts_commandline-Configuringmod_tts_commandline
> >
> as dialplan shows following the debug log output below:
>
> 2021-02-26 15:19:24.713040 [NOTICE] mod_dptools.c:1406 Channel
> [sofia/internal/102 at 66.151.243.45] has been answered
> 2021-02-26 15:19:24.713040 [DEBUG] switch_channel.c:3865 (sofia/internal/
> 102 at 66.151.243.45) Callstate Change RINGING -> ACTIVE
> 2021-02-26 15:19:24.713040 [DEBUG] sofia.c:7326 Channel sofia/internal/
> 102 at 66.151.243.45 entering state [completed][200]
> EXECUTE [depth=0] sofia/internal/102 at 66.151.243.45
> speak(tts_commandline|pico|This is an example of using tts_commandline)
> 2021-02-26 15:19:24.713040 [DEBUG] switch_ivr_play_say.c:3023 OPEN TTS
> tts_commandline
> 2021-02-26 15:19:24.713040 [DEBUG] switch_ivr_play_say.c:3033 Raw Codec
> Activated
> 2021-02-26 15:19:24.713040 [DEBUG] mod_tts_commandline.c:160 Executing:
> echo 'This is an example of using tts_commandline' | text2wave -f 8000 >
> '/tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav'
> 2021-02-26 15:19:25.433013 [WARNING] mod_sndfile.c:281 Error Opening File
> [/tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav] [File contains data in
> an unknown format.]
> 2021-02-26 15:19:25.433013 [ERR] mod_tts_commandline.c:170 Failed to open
> file: /tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav


Since your goal is to use mod_unimrcp I think you should not spend your
time with mod_tts_commandline.
Anyway, what do you get when you run these in the bash shell:

which text2wave
echo 'This is an example of using tts_commandline' | text2wave -f 8000 >
test.wav
file test.wav
ls -l test.wav

Did you install festival? (text2wave comes with it)
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From william at williamcollsassoc.ca  Wed Mar  3 04:17:33 2021
From: william at williamcollsassoc.ca (William Colls)
Date: Tue, 2 Mar 2021 23:17:33 -0500
Subject: [Freeswitch-users] One Way Audio
Message-ID: <927623dd-f418-abe5-ef5e-f397f14eef8a@williamcollsassoc.ca>

I know I have seen this discussed before but can't find the information now.

Freeswitch 1.10.5 compiled from source.

When I try to make an outbound call the call connects, and the callee 
can hear me, but I can't hear him/her. Calls between phones registered 
on freeswitch work, Inbound calls work. The call appears to executed in 
the public context.

Any suggestions as to how to proceed greatly appreciated.

Thanks for you time.

William



From dragos at freeswitch.org  Wed Mar  3 07:25:29 2021
From: dragos at freeswitch.org (Dragos Oancea)
Date: Wed, 3 Mar 2021 09:25:29 +0200
Subject: [Freeswitch-users] mod_sndfile error called from
 mod_tts_commandline - [WARNING] mod_sndfile.c:281 Error Opening File
In-Reply-To: <CABaNFCabu6equvag0nkwvLm+eDZNf3VLgnop4SpZ_PMWCm4qmw@mail.gmail.com>
References: <CAD2CnaQHdQ4pL1uXaGumGS+pM6MGyos9s+w2EUXJD52T0goB0w@mail.gmail.com>
 <CABaNFCabu6equvag0nkwvLm+eDZNf3VLgnop4SpZ_PMWCm4qmw@mail.gmail.com>
Message-ID: <CAGvJV4rP5UwxYB3L8nPSKHUwUU_n8zpxrHwrq6yG-9j2VtwCFQ@mail.gmail.com>

If the wav file exists but is 0 size, mod_sndfile would say this:
mod_sndfile.c:281 Error Opening File [/tmp/foo.wav] [File contains data in
an unknown format.]
So yes, most likely text2wave is not installed.

On Wed, Mar 3, 2021 at 1:13 AM mayamatakeshi <mayamatakeshi at gmail.com>
wrote:

>
>
> On Wed, Mar 3, 2021 at 4:12 AM Lewis Bergman <lewis.bergman at gmail.com>
> wrote:
>
>> Hello all. I am pretty new to Freeswitch and have an experimental
>> container
>> on Proxmox 6.3 Debian 10 to play with. The goal is to get mod_unimrcp
>> working. To that end, I have mod_tts_commandline (installed from binary
>> with FusionPBX as part of the default install) listed in modules.conf.xml
>> /etc/freeswitch/autoload_configs/modules.conf.xml
>>                 <!-- Speech Recognition / Text to Speech -->
>>                 <load module="mod_unimrcp"/>
>>                 <load module="mod_tts_commandline"/>
>>                 <load module="mod_flite"/>
>> I have tried it with and without mod_flite enabled.
>> Mod_tts_commandline refuses to pull a sound file and the debug log seems
>> to
>> blame mod_sndfile. This is using the example from the mod_tts_commandline
>> page
>> <
>> https://freeswitch.org/confluence/display/FREESWITCH/mod_tts_commandline#mod_tts_commandline-Configuringmod_tts_commandline
>> >
>> as dialplan shows following the debug log output below:
>>
>> 2021-02-26 15:19:24.713040 [NOTICE] mod_dptools.c:1406 Channel
>> [sofia/internal/102 at 66.151.243.45] has been answered
>> 2021-02-26 15:19:24.713040 [DEBUG] switch_channel.c:3865 (sofia/internal/
>> 102 at 66.151.243.45) Callstate Change RINGING -> ACTIVE
>> 2021-02-26 15:19:24.713040 [DEBUG] sofia.c:7326 Channel sofia/internal/
>> 102 at 66.151.243.45 entering state [completed][200]
>> EXECUTE [depth=0] sofia/internal/102 at 66.151.243.45
>> speak(tts_commandline|pico|This is an example of using tts_commandline)
>> 2021-02-26 15:19:24.713040 [DEBUG] switch_ivr_play_say.c:3023 OPEN TTS
>> tts_commandline
>> 2021-02-26 15:19:24.713040 [DEBUG] switch_ivr_play_say.c:3033 Raw Codec
>> Activated
>> 2021-02-26 15:19:24.713040 [DEBUG] mod_tts_commandline.c:160 Executing:
>> echo 'This is an example of using tts_commandline' | text2wave -f 8000 >
>> '/tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav'
>> 2021-02-26 15:19:25.433013 [WARNING] mod_sndfile.c:281 Error Opening File
>> [/tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav] [File contains data in
>> an unknown format.]
>> 2021-02-26 15:19:25.433013 [ERR] mod_tts_commandline.c:170 Failed to open
>> file: /tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav
>
>
> Since your goal is to use mod_unimrcp I think you should not spend your
> time with mod_tts_commandline.
> Anyway, what do you get when you run these in the bash shell:
>
> which text2wave
> echo 'This is an example of using tts_commandline' | text2wave -f 8000 >
> test.wav
> file test.wav
> ls -l test.wav
>
> Did you install festival? (text2wave comes with it)
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From dragos at freeswitch.org  Wed Mar  3 07:32:08 2021
From: dragos at freeswitch.org (Dragos Oancea)
Date: Wed, 3 Mar 2021 09:32:08 +0200
Subject: [Freeswitch-users] One Way Audio
In-Reply-To: <927623dd-f418-abe5-ef5e-f397f14eef8a@williamcollsassoc.ca>
References: <927623dd-f418-abe5-ef5e-f397f14eef8a@williamcollsassoc.ca>
Message-ID: <CAGvJV4qsz=qjTmYONP0K5xkei2XFMa318visSJ4_1qsP2oLPeQ@mail.gmail.com>

Check iptables / firewall .

On Wed, Mar 3, 2021 at 6:18 AM William Colls <william at williamcollsassoc.ca>
wrote:

> I know I have seen this discussed before but can't find the information
> now.
>
> Freeswitch 1.10.5 compiled from source.
>
> When I try to make an outbound call the call connects, and the callee
> can hear me, but I can't hear him/her. Calls between phones registered
> on freeswitch work, Inbound calls work. The call appears to executed in
> the public context.
>
> Any suggestions as to how to proceed greatly appreciated.
>
> Thanks for you time.
>
> William
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From bilaln018 at gmail.com  Wed Mar  3 08:23:16 2021
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Wed, 3 Mar 2021 13:23:16 +0500
Subject: [Freeswitch-users] One Way Audio
In-Reply-To: <CAGvJV4qsz=qjTmYONP0K5xkei2XFMa318visSJ4_1qsP2oLPeQ@mail.gmail.com>
References: <927623dd-f418-abe5-ef5e-f397f14eef8a@williamcollsassoc.ca>
 <CAGvJV4qsz=qjTmYONP0K5xkei2XFMa318visSJ4_1qsP2oLPeQ@mail.gmail.com>
Message-ID: <CADSVerCM=0YnEX+4yxwzirKLnDidjM0YmEz+AsjriOwSY3840w@mail.gmail.com>

Hi,
Can you check sngrep logs, and see if the media/rtp IPs are advertised
properly.
After that see if you have firewall access properly setup.

Regards
Abbasi

On Wed, 3 Mar 2021 at 12:54 PM, Dragos Oancea <dragos at freeswitch.org> wrote:

> Check iptables / firewall .
>
> On Wed, Mar 3, 2021 at 6:18 AM William Colls <william at williamcollsassoc.ca>
> wrote:
>
>> I know I have seen this discussed before but can't find the information
>> now.
>>
>> Freeswitch 1.10.5 compiled from source.
>>
>> When I try to make an outbound call the call connects, and the callee
>> can hear me, but I can't hear him/her. Calls between phones registered
>> on freeswitch work, Inbound calls work. The call appears to executed in
>> the public context.
>>
>> Any suggestions as to how to proceed greatly appreciated.
>>
>> Thanks for you time.
>>
>> William
>>
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users 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
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From vbvbrj at gmail.com  Wed Mar  3 14:54:38 2021
From: vbvbrj at gmail.com (Mimiko)
Date: Wed, 3 Mar 2021 16:54:38 +0200
Subject: [Freeswitch-users] mod lcr cid conditional replace.
Message-ID: <5451082a-b479-a919-0e5d-4960ccc51a6f@gmail.com>

Hi.

I use mod_lcr for dialout operators. On the freeswitch pbx we have extensions 500 to 899. When calling outside number the internal extension is 
converted ok to for external cid to be accepted by each operator using lcr_cid regular pattern:
/(\d\d\d)/${operator_prefix}$1/

The problem lies with one operator where our extensions which start with 5 (500 to 599) must be converted to numbers which start with 9 (900 to 999) 
and then prefixed with operator's prefix. The rest of extensions does not have such a substitution of first number.

How can I do this using regular expression in lcr_cid?

Thank you.


From davidswalkabout at gmail.com  Thu Mar  4 02:46:11 2021
From: davidswalkabout at gmail.com (David P)
Date: Thu, 4 Mar 2021 15:46:11 +1300
Subject: [Freeswitch-users] rtp-timeout-sec VS media_timeout
In-Reply-To: <mailman.0.1610020801.56386.freeswitch-users@lists.freeswitch.org>
References: <mailman.0.1610020801.56386.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAE23trtppjOD6KnroPUKJ_nfysHzPOdO6KrAMHtD1KRihjXk+A@mail.gmail.com>

Hi Allan,

I don't know if the media_timeout=300 behavior you saw is a bug or not, but
I wanted to add my own observation of weirdness about hangup
cause MEDIA_TIMEOUT...

I just noticed a conference end abruptly after one leg spoke for 5 minutes.
The logs aren't clear why this happened, but it seems that <param
name="rtp-timeout-sec" value="300"/> in our sip_profiles/internal.xml is
the reason -- it seems that because the *other* leg of the conference
remained silent, the RTP timeout was reached.

I couldn't find any confluence pages about MEDIA_TIMEOUT by googling.

On Fri, Jan 8, 2021 at 1:08 AM <
freeswitch-users-request at lists.freeswitch.org> wrote:

>
> ---------- Forwarded message ----------
> From: Allan Kristensen <ak at hejdu.dk>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Wed, 6 Jan 2021 19:37:33 +0100
> Subject: [Freeswitch-users] rtp-timeout-sec VS media_timeout
> We had some problems with "hanging channels" for our webrtc clients (via
> kamailio).
> To solve the problem I tried to use "media_timeout" setting but it didn't
> really work. So I tried the deprecated "rtp-timeout-sec" and this actually
> works fine?
>
> Not working:
> <param name="media_timeout" value="300"/>
>
> Working:
> <param name="rtp-timeout-sec" value="300"/>
>
> How to reproduce: Make a call using webrtc and just close browser window,
> after some time freeswitch should close the channel because of missing rtp.
> Instead it just keeps writing: "switch_rtp.c:853 No audio stun for a long
> time!" forever...
>
> Anyway, it works now....just curious why...a typo or bug?
>
> /Allan
>
> Using:
> FreeSWITCH Version 1.10.5-release+git~20200818T185121Z~25569c1631~64bit
>
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From dragos at freeswitch.org  Thu Mar  4 09:04:26 2021
From: dragos at freeswitch.org (Dragos Oancea)
Date: Thu, 4 Mar 2021 11:04:26 +0200
Subject: [Freeswitch-users] rtp-timeout-sec VS media_timeout
In-Reply-To: <CAE23trtppjOD6KnroPUKJ_nfysHzPOdO6KrAMHtD1KRihjXk+A@mail.gmail.com>
References: <mailman.0.1610020801.56386.freeswitch-users@lists.freeswitch.org>
 <CAE23trtppjOD6KnroPUKJ_nfysHzPOdO6KrAMHtD1KRihjXk+A@mail.gmail.com>
Message-ID: <CAGvJV4qMqizn1ME9yW-tN0=TT+krvJjbCeC=yc1_Gq5asAhqGA@mail.gmail.com>

For media timeout there are the following chan vars:
media_timeout,
media_hold_timeout, media_timeout, media_hold_timeout_video,
media_hold_timeout_audio, media_timeout_audio
.

They are in milliseconds, not seconds like rtp-timeout-sec .

<param name="media_timeout" value="300000"/>

On Thu, Mar 4, 2021 at 4:47 AM David P <davidswalkabout at gmail.com> wrote:

> Hi Allan,
>
> I don't know if the media_timeout=300 behavior you saw is a bug or not,
> but I wanted to add my own observation of weirdness about hangup
> cause MEDIA_TIMEOUT...
>
> I just noticed a conference end abruptly after one leg spoke for 5
> minutes. The logs aren't clear why this happened, but it seems that <param
> name="rtp-timeout-sec" value="300"/> in our sip_profiles/internal.xml is
> the reason -- it seems that because the *other* leg of the conference
> remained silent, the RTP timeout was reached.
>
> I couldn't find any confluence pages about MEDIA_TIMEOUT by googling.
>
> On Fri, Jan 8, 2021 at 1:08 AM <
> freeswitch-users-request at lists.freeswitch.org> wrote:
>
>>
>> ---------- Forwarded message ----------
>> From: Allan Kristensen <ak at hejdu.dk>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Bcc:
>> Date: Wed, 6 Jan 2021 19:37:33 +0100
>> Subject: [Freeswitch-users] rtp-timeout-sec VS media_timeout
>> We had some problems with "hanging channels" for our webrtc clients (via
>> kamailio).
>> To solve the problem I tried to use "media_timeout" setting but it didn't
>> really work. So I tried the deprecated "rtp-timeout-sec" and this actually
>> works fine?
>>
>> Not working:
>> <param name="media_timeout" value="300"/>
>>
>> Working:
>> <param name="rtp-timeout-sec" value="300"/>
>>
>> How to reproduce: Make a call using webrtc and just close browser window,
>> after some time freeswitch should close the channel because of missing rtp.
>> Instead it just keeps writing: "switch_rtp.c:853 No audio stun for a long
>> time!" forever...
>>
>> Anyway, it works now....just curious why...a typo or bug?
>>
>> /Allan
>>
>> Using:
>> FreeSWITCH Version 1.10.5-release+git~20200818T185121Z~25569c1631~64bit
>>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From martin.paterson at technologywithin.com  Thu Mar  4 13:34:03 2021
From: martin.paterson at technologywithin.com (Martin Paterson)
Date: Thu, 4 Mar 2021 13:34:03 +0000
Subject: [Freeswitch-users] Don't deprecate this option - useful for buggy
	user agents
Message-ID: <VI1PR10MB2094B42A154665A1B8E79E5995979@VI1PR10MB2094.EURPRD10.PROD.OUTLOOK.COM>

I had a problem which I solved by turning on a FreeSWITCH option that logs a warning: "Enabling reg-deny-binding-fetch-and-no-lookup - this functionality is deprecated and will be removed - let FS devs know if you think it should stay". So here is me letting the FS devs know.

The problem is to do with buggy user agents and their handling of multiple Contact headers in the 200 OK response to their REGISTER.

My FreeSWITCH system uses mainly Snom phones and Zoiper soft phones. I allows multiple registrations and I have two sip profiles that the user agents connect to, one for Snoms and one for Zoiper. They share a registration database with the dbname option.

When multiple user agents register with the same credentials, the correct behaviour in the 200 OK to a REGISTER is to list all the user agents in multiple Contact headers with their individual registration expiry times. This is exactly what FreeSWITCH does. However both clients (Snom and Zoiper) fail in a similar way:

For their own reasons (https://www.zoiper.com/en/support/home/article/205/Zoiper%20Push%20Proxy) Zoiper clients register with a 1 month expiry time. If in the 200 OK to a Snom the Zoiper contact header is first, then the Snom assumes that the header is meant for it, reads the large expiry time and uses it (and complains in its log if it's over 14 days). The FreeSWITCH registration for the Snom then expires after 1 hour, which what the Snom asked for in the first place. The Snom then doesn't receive calls.

Annoyingly a similar thing happens the other way round - if the Snom Contact header is first in the 200 OK sent to Zoiper, then the Zoiper client just looks at the first header, and complains that the expiry time is too short with a message "You are connecting to a server with a low re-registration time. Battery life will not be optimal. Please contact your VoIP provider or PBX server administrator."

(An aside: it would be interesting to know why returning multiple Contact headers is the defined behaviour, and what a user agent is supposed to do with the others headers, and therefore what the consequences of not having them might be.)

We are contacting both Snom and Zoiper about this, but for the moment this is a useful FreeSWITCH option to cope with these user agents' behaviours.

Martin

Martin Paterson
Development Team
Phone: 0207 953 8840
Email: martin.paterson at technologywithin.com
Chevron Business Park, Limekiln Lane, Southampton, Hampshire, SO45 2QL
Registered Office: CP House, Otterspool Way, Watford, WD25 8JJ, U.K 
​Registered in England No: 5964349 | VAT Number: GB 902 5369 37 
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From botelist at gmail.com  Thu Mar  4 13:41:31 2021
From: botelist at gmail.com (Bote Man)
Date: Thu, 4 Mar 2021 08:41:31 -0500
Subject: [Freeswitch-users] rtp-timeout-sec VS media_timeout
In-Reply-To: <CAGvJV4qMqizn1ME9yW-tN0=TT+krvJjbCeC=yc1_Gq5asAhqGA@mail.gmail.com>
References: <mailman.0.1610020801.56386.freeswitch-users@lists.freeswitch.org>
 <CAE23trtppjOD6KnroPUKJ_nfysHzPOdO6KrAMHtD1KRihjXk+A@mail.gmail.com>
 <CAGvJV4qMqizn1ME9yW-tN0=TT+krvJjbCeC=yc1_Gq5asAhqGA@mail.gmail.com>
Message-ID: <005301d710fc$1622c760$42685620$@gmail.com>

I recall that the rtp_* variable names were changed to media_* some time ago. I updated the wiki to reflect this.

 

Now the question is: what version of code exhibited this behavior? 

 

If it was built before the change, then naturally the rtp_* series will be what it uses.

 

 

--- 

John Boteler 

BnC Group U.S.A. 

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Dragos Oancea
Sent: Thursday, 4 March, 2021 04:04
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] rtp-timeout-sec VS media_timeout

 

For media timeout there are the following chan vars:

media_timeout, media_hold_timeout, media_timeout, media_hold_timeout_video, media_hold_timeout_audio, media_timeout_audio .

 

They are in milliseconds, not seconds like rtp-timeout-sec . 

 

<param name="media_timeout" value="300000"/>

 

On Thu, Mar 4, 2021 at 4:47 AM David P <davidswalkabout at gmail.com <mailto:davidswalkabout at gmail.com> > wrote:

Hi Allan,

 

I don't know if the media_timeout=300 behavior you saw is a bug or not, but I wanted to add my own observation of weirdness about hangup cause MEDIA_TIMEOUT...

 

I just noticed a conference end abruptly after one leg spoke for 5 minutes. The logs aren't clear why this happened, but it seems that <param name="rtp-timeout-sec" value="300"/> in our sip_profiles/internal.xml is the reason -- it seems that because the *other* leg of the conference remained silent, the RTP timeout was reached.

 

I couldn't find any confluence pages about MEDIA_TIMEOUT by googling.

 

On Fri, Jan 8, 2021 at 1:08 AM <freeswitch-users-request at lists.freeswitch.org <mailto:freeswitch-users-request at lists.freeswitch.org> > wrote:


---------- Forwarded message ----------
From: Allan Kristensen <ak at hejdu.dk <mailto:ak at hejdu.dk> >
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Cc: 
Bcc: 
Date: Wed, 6 Jan 2021 19:37:33 +0100
Subject: [Freeswitch-users] rtp-timeout-sec VS media_timeout

We had some problems with "hanging channels" for our webrtc clients (via kamailio).

To solve the problem I tried to use "media_timeout" setting but it didn't really work. So I tried the deprecated "rtp-timeout-sec" and this actually works fine?

 

Not working:

<param name="media_timeout" value="300"/>

 

Working:

<param name="rtp-timeout-sec" value="300"/>

 

How to reproduce: Make a call using webrtc and just close browser window, after some time freeswitch should close the channel because of missing rtp.

Instead it just keeps writing: "switch_rtp.c:853 No audio stun for a long time!" forever...

 

Anyway, it works now....just curious why...a typo or bug?

 

/Allan

 

Using:

FreeSWITCH Version 1.10.5-release+git~20200818T185121Z~25569c1631~64bit

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product 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
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From vbvbrj at gmail.com  Thu Mar  4 14:19:48 2021
From: vbvbrj at gmail.com (Mimiko)
Date: Thu, 4 Mar 2021 16:19:48 +0200
Subject: [Freeswitch-users] Starting dtmf_generation with mod_lcr
Message-ID: <d6f0acc8-b890-dbea-80d6-e4c378728341@gmail.com>

Hello.

I use lcr for outbound connection to providers based on phone call price. On one provider I need to use start_dtmf_generate to force recoding sip info 
from local extensions to inband dtmf. Setting variable from lcr execute_on_answer=start_dtmf_generate does not work. Also gateway profile have

<variables>
	<variable direction="outbound" name="dtmf_type" value="none"/>
</variables>
<param name="dtmf-type" value="none"/>

this also does not help.

How can I accomplish per provider dtmf_generate for outbound calls?


From david.villaume at sewan.fr  Wed Mar  3 12:55:23 2021
From: david.villaume at sewan.fr (David VILLAUME)
Date: Wed, 3 Mar 2021 12:55:23 +0000
Subject: [Freeswitch-users] Freeswitch loacally acks 200 in bypass media
	scenarios
Message-ID: <PR1P264MB1166B5F172F31EC66BC2AC89FD989@PR1P264MB1166.FRAP264.PROD.OUTLOOK.COM>

Hello,

In bypass-media mode freeswitch locally acks the 200 ok (SDP), that leads to some Race condition when the party that answered the 200 sends a quick reinvite after Just after the Ack.

I tried on 1.10.1 and 1.10.5, with and without late-negociation, but couldn’t figure how to avoid this.

Has one of you experienced it or have any idea about this ?


          SBC                FREESWITCH                PBX
+---------          ----------+---------          ----------+
|        INVITE (SDP)         |                             |
| --------------------------> |                             |
|         100 Trying          |                             |
| <-------------------------- |                             |
|                             |        INVITE (SDP)         |
|                             | --------------------------> |
|                             |         100 Trying          |
|                             | <-------------------------- |
|                             |         180 Ringing         |
|                             | <-------------------------- |
|         180 Ringing         |                             |
| <-------------------------- |                             |
|                             |        200 OK (SDP)         |
|                             | <-------------------------- |
|                             |             ACK             |
|                             | --------------------------> |
|        200 OK (SDP)         |                             |
| <-------------------------- |                             |
|                             |        INVITE (SDP)         |
|                             | <-------------------------- |
|                             |             100             |
|                             | --------------------------> |
|        INVITE (SDP)         |                             |
| <-------------------------- |                             |
|             100             |                             |
| --------------------------> |                             |
|             491             |                             |
| --------------------------> |                             |
|                             |             491             |
|                             | --------------------------> |

Regards,
David
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From lewis.bergman at gmail.com  Thu Mar  4 13:02:29 2021
From: lewis.bergman at gmail.com (Lewis Bergman)
Date: Thu, 4 Mar 2021 07:02:29 -0600
Subject: [Freeswitch-users] mod_sndfile error called from
 mod_tts_commandline - [WARNING] mod_sndfile.c:281 Error Opening File
In-Reply-To: <CABaNFCabu6equvag0nkwvLm+eDZNf3VLgnop4SpZ_PMWCm4qmw@mail.gmail.com>
References: <CAD2CnaQHdQ4pL1uXaGumGS+pM6MGyos9s+w2EUXJD52T0goB0w@mail.gmail.com>
 <CABaNFCabu6equvag0nkwvLm+eDZNf3VLgnop4SpZ_PMWCm4qmw@mail.gmail.com>
Message-ID: <CAD2CnaTKNWq3FJ1yJujW=MXTwNMJ6Wn7QNf+aDL9CVaORVBBjA@mail.gmail.com>

That was exactly it for the record.
#which text2wave
#                         <-- before apt-get install festival
#which text2wave
/usr/bin/text2wave
# echo 'This is an example of using tts_commandline' | text2wave -f
8000 > test.wav
# file test.wav
test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16
bit, mono 8000 Hz
# ls -l test.wav
-rw-r--r-- 1 root root 60208 Mar  4 06:44 test.wav
And the freeswitch debug log that shows it works:
EXECUTE [depth=0] sofia/internal/102 at 66.151.243.45
speak(tts_commandline|text2wave|This is an example of using
tts_commandline)
2021-03-04 06:59:14.305216 [DEBUG] switch_ivr_play_say.c:3023 OPEN TTS
tts_commandline
2021-03-04 06:59:14.305216 [DEBUG] switch_ivr_play_say.c:3033 Raw
Codec Activated
2021-03-04 06:59:14.305216 [DEBUG] mod_tts_commandline.c:160
Executing: echo 'This is an example of using tts_commandline' |
text2wave -f 8000 >
'/tmp/05e71313-2a4d-48ce-a1e9-84600859112f.tmp.wav'
2021-03-04 06:59:15.545212 [DEBUG] switch_ivr_play_say.c:2741 Speaking
text: This is an example of using tts_commandline
2021-03-04 06:59:15.545212 [DEBUG] sofia.c:7326 Channel
sofia/internal/102 at 66.151.243.45 entering state [ready][200]
2021-03-04 06:59:15.565210 [DEBUG] switch_rtp.c:7759 Correct audio
ip/port confirmed.
2021-03-04 06:59:19.305225 [DEBUG] switch_ivr_play_say.c:2905 done speaking text

Thanks for the help.
The reason I tried to start with tts_commandine is I have been
struggling for a couple of weeks to get MRCP to produce something
useful so I thought I would try something simpler and I could swear
mod_tts_commandline was listed as a requirement. But it could have
been on something else as I have been pouring through every document I
can find to figure this out and I probably got confused.
Thanks again!

On Tue, Mar 2, 2021 at 6:36 PM mayamatakeshi <mayamatakeshi at gmail.com> wrote:
>
>
>
> On Wed, Mar 3, 2021 at 4:12 AM Lewis Bergman <lewis.bergman at gmail.com> wrote:
>>
>> Hello all. I am pretty new to Freeswitch and have an experimental container
>> on Proxmox 6.3 Debian 10 to play with. The goal is to get mod_unimrcp
>> working. To that end, I have mod_tts_commandline (installed from binary
>> with FusionPBX as part of the default install) listed in modules.conf.xml
>> /etc/freeswitch/autoload_configs/modules.conf.xml
>>                 <!-- Speech Recognition / Text to Speech -->
>>                 <load module="mod_unimrcp"/>
>>                 <load module="mod_tts_commandline"/>
>>                 <load module="mod_flite"/>
>> I have tried it with and without mod_flite enabled.
>> Mod_tts_commandline refuses to pull a sound file and the debug log seems to
>> blame mod_sndfile. This is using the example from the mod_tts_commandline
>> page
>> <https://freeswitch.org/confluence/display/FREESWITCH/mod_tts_commandline#mod_tts_commandline-Configuringmod_tts_commandline>
>> as dialplan shows following the debug log output below:
>>
>> 2021-02-26 15:19:24.713040 [NOTICE] mod_dptools.c:1406 Channel
>> [sofia/internal/102 at 66.151.243.45] has been answered
>> 2021-02-26 15:19:24.713040 [DEBUG] switch_channel.c:3865 (sofia/internal/
>> 102 at 66.151.243.45) Callstate Change RINGING -> ACTIVE
>> 2021-02-26 15:19:24.713040 [DEBUG] sofia.c:7326 Channel sofia/internal/
>> 102 at 66.151.243.45 entering state [completed][200]
>> EXECUTE [depth=0] sofia/internal/102 at 66.151.243.45
>> speak(tts_commandline|pico|This is an example of using tts_commandline)
>> 2021-02-26 15:19:24.713040 [DEBUG] switch_ivr_play_say.c:3023 OPEN TTS
>> tts_commandline
>> 2021-02-26 15:19:24.713040 [DEBUG] switch_ivr_play_say.c:3033 Raw Codec
>> Activated
>> 2021-02-26 15:19:24.713040 [DEBUG] mod_tts_commandline.c:160 Executing:
>> echo 'This is an example of using tts_commandline' | text2wave -f 8000 >
>> '/tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav'
>> 2021-02-26 15:19:25.433013 [WARNING] mod_sndfile.c:281 Error Opening File
>> [/tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav] [File contains data in
>> an unknown format.]
>> 2021-02-26 15:19:25.433013 [ERR] mod_tts_commandline.c:170 Failed to open
>> file: /tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav
>
>
> Since your goal is to use mod_unimrcp I think you should not spend your time with mod_tts_commandline.
> Anyway, what do you get when you run these in the bash shell:
>
> which text2wave
> echo 'This is an example of using tts_commandline' | text2wave -f 8000 > test.wav
> file test.wav
> ls -l test.wav
>
> Did you install festival? (text2wave comes with it)
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Lewis Bergman
325-439-0533 Cell


From brian at freeswitch.com  Thu Mar  4 19:38:40 2021
From: brian at freeswitch.com (Brian West)
Date: Thu, 4 Mar 2021 13:38:40 -0600
Subject: [Freeswitch-users] Freeswitch loacally acks 200 in bypass media
	scenarios
In-Reply-To: <PR1P264MB1166B5F172F31EC66BC2AC89FD989@PR1P264MB1166.FRAP264.PROD.OUTLOOK.COM>
References: <PR1P264MB1166B5F172F31EC66BC2AC89FD989@PR1P264MB1166.FRAP264.PROD.OUTLOOK.COM>
Message-ID: <CAEJMVkDvcTX2UnmqifLYO6xSAwfRJfbkA3WuMRX-tXuLwRXE9w@mail.gmail.com>

What do you mean 'locally acks'?

On Thu, Mar 4, 2021 at 1:25 PM David VILLAUME <david.villaume at sewan.fr>
wrote:

> Hello,
>
>
>
> In bypass-media mode freeswitch locally acks the 200 ok (SDP), that leads
> to some Race condition when the party that answered the 200 sends a quick
> reinvite after Just after the Ack.
>
>
>
> I tried on 1.10.1 and 1.10.5, with and without late-negociation, but
> couldn’t figure how to avoid this.
>
>
>
> Has one of you experienced it or have any idea about this ?
>
>
>
>
>
>           SBC                FREESWITCH                PBX
>
> +---------          ----------+---------          ----------+
>
> |        INVITE (SDP)         |                             |
>
> | --------------------------> |                             |
>
> |         100 Trying          |                             |
>
> | <-------------------------- |                             |
>
> |                             |        INVITE (SDP)         |
>
> |                             | --------------------------> |
>
> |                             |         100 Trying          |
>
> |                             | <-------------------------- |
>
> |                             |         180 Ringing         |
>
> |                             | <-------------------------- |
>
> |         180 Ringing         |                             |
>
> | <-------------------------- |                             |
>
> |                             |        200 OK (SDP)         |
>
> |                             | <-------------------------- |
>
> |                             |             ACK             |
>
> |                             | --------------------------> |
>
> |        200 OK (SDP)         |                             |
>
> | <-------------------------- |                             |
>
> |                             |        INVITE (SDP)         |
>
> |                             | <-------------------------- |
>
> |                             |             100             |
>
> |                             | --------------------------> |
>
> |        INVITE (SDP)         |                             |
>
> | <-------------------------- |                             |
>
> |             100             |                             |
>
> | --------------------------> |                             |
>
> |             491             |                             |
>
> | --------------------------> |                             |
>
> |                             |             491             |
>
> |                             | --------------------------> |
>
>
>
> Regards,
>
> David
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users 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>
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From brian at freeswitch.com  Thu Mar  4 19:51:53 2021
From: brian at freeswitch.com (Brian West)
Date: Thu, 4 Mar 2021 13:51:53 -0600
Subject: [Freeswitch-users] mod_sndfile error called from
 mod_tts_commandline - [WARNING] mod_sndfile.c:281 Error Opening File
In-Reply-To: <CAD2CnaTKNWq3FJ1yJujW=MXTwNMJ6Wn7QNf+aDL9CVaORVBBjA@mail.gmail.com>
References: <CAD2CnaQHdQ4pL1uXaGumGS+pM6MGyos9s+w2EUXJD52T0goB0w@mail.gmail.com>
 <CABaNFCabu6equvag0nkwvLm+eDZNf3VLgnop4SpZ_PMWCm4qmw@mail.gmail.com>
 <CAD2CnaTKNWq3FJ1yJujW=MXTwNMJ6Wn7QNf+aDL9CVaORVBBjA@mail.gmail.com>
Message-ID: <CAEJMVkBC3asdwwvWnK3C0+eNodEUddKXJOVhnUUeQawUrQuSng@mail.gmail.com>

$PATH might be a thing? :) Try the full path to the binary?

On Thu, Mar 4, 2021 at 1:36 PM Lewis Bergman <lewis.bergman at gmail.com>
wrote:

> That was exactly it for the record.
> #which text2wave
> #                         <-- before apt-get install festival
> #which text2wave
> /usr/bin/text2wave
> # echo 'This is an example of using tts_commandline' | text2wave -f
> 8000 > test.wav
> # file test.wav
> test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16
> bit, mono 8000 Hz
> # ls -l test.wav
> -rw-r--r-- 1 root root 60208 Mar  4 06:44 test.wav
> And the freeswitch debug log that shows it works:
> EXECUTE [depth=0] sofia/internal/102 at 66.151.243.45
> speak(tts_commandline|text2wave|This is an example of using
> tts_commandline)
> 2021-03-04 06:59:14.305216 [DEBUG] switch_ivr_play_say.c:3023 OPEN TTS
> tts_commandline
> 2021-03-04 06:59:14.305216 [DEBUG] switch_ivr_play_say.c:3033 Raw
> Codec Activated
> 2021-03-04 06:59:14.305216 [DEBUG] mod_tts_commandline.c:160
> Executing: echo 'This is an example of using tts_commandline' |
> text2wave -f 8000 >
> '/tmp/05e71313-2a4d-48ce-a1e9-84600859112f.tmp.wav'
> 2021-03-04 06:59:15.545212 [DEBUG] switch_ivr_play_say.c:2741 Speaking
> text: This is an example of using tts_commandline
> 2021-03-04 06:59:15.545212 [DEBUG] sofia.c:7326 Channel
> sofia/internal/102 at 66.151.243.45 entering state [ready][200]
> 2021-03-04 06:59:15.565210 [DEBUG] switch_rtp.c:7759 Correct audio
> ip/port confirmed.
> 2021-03-04 06:59:19.305225 [DEBUG] switch_ivr_play_say.c:2905 done
> speaking text
>
> Thanks for the help.
> The reason I tried to start with tts_commandine is I have been
> struggling for a couple of weeks to get MRCP to produce something
> useful so I thought I would try something simpler and I could swear
> mod_tts_commandline was listed as a requirement. But it could have
> been on something else as I have been pouring through every document I
> can find to figure this out and I probably got confused.
> Thanks again!
>
> On Tue, Mar 2, 2021 at 6:36 PM mayamatakeshi <mayamatakeshi at gmail.com>
> wrote:
> >
> >
> >
> > On Wed, Mar 3, 2021 at 4:12 AM Lewis Bergman <lewis.bergman at gmail.com>
> wrote:
> >>
> >> Hello all. I am pretty new to Freeswitch and have an experimental
> container
> >> on Proxmox 6.3 Debian 10 to play with. The goal is to get mod_unimrcp
> >> working. To that end, I have mod_tts_commandline (installed from binary
> >> with FusionPBX as part of the default install) listed in
> modules.conf.xml
> >> /etc/freeswitch/autoload_configs/modules.conf.xml
> >>                 <!-- Speech Recognition / Text to Speech -->
> >>                 <load module="mod_unimrcp"/>
> >>                 <load module="mod_tts_commandline"/>
> >>                 <load module="mod_flite"/>
> >> I have tried it with and without mod_flite enabled.
> >> Mod_tts_commandline refuses to pull a sound file and the debug log
> seems to
> >> blame mod_sndfile. This is using the example from the
> mod_tts_commandline
> >> page
> >> <
> https://freeswitch.org/confluence/display/FREESWITCH/mod_tts_commandline#mod_tts_commandline-Configuringmod_tts_commandline
> >
> >> as dialplan shows following the debug log output below:
> >>
> >> 2021-02-26 15:19:24.713040 [NOTICE] mod_dptools.c:1406 Channel
> >> [sofia/internal/102 at 66.151.243.45] has been answered
> >> 2021-02-26 15:19:24.713040 [DEBUG] switch_channel.c:3865
> (sofia/internal/
> >> 102 at 66.151.243.45) Callstate Change RINGING -> ACTIVE
> >> 2021-02-26 15:19:24.713040 [DEBUG] sofia.c:7326 Channel sofia/internal/
> >> 102 at 66.151.243.45 entering state [completed][200]
> >> EXECUTE [depth=0] sofia/internal/102 at 66.151.243.45
> >> speak(tts_commandline|pico|This is an example of using tts_commandline)
> >> 2021-02-26 15:19:24.713040 [DEBUG] switch_ivr_play_say.c:3023 OPEN TTS
> >> tts_commandline
> >> 2021-02-26 15:19:24.713040 [DEBUG] switch_ivr_play_say.c:3033 Raw Codec
> >> Activated
> >> 2021-02-26 15:19:24.713040 [DEBUG] mod_tts_commandline.c:160 Executing:
> >> echo 'This is an example of using tts_commandline' | text2wave -f 8000 >
> >> '/tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav'
> >> 2021-02-26 15:19:25.433013 [WARNING] mod_sndfile.c:281 Error Opening
> File
> >> [/tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav] [File contains data
> in
> >> an unknown format.]
> >> 2021-02-26 15:19:25.433013 [ERR] mod_tts_commandline.c:170 Failed to
> open
> >> file: /tmp/0665a30d-2bb8-40fe-b5a6-9625bd58306e.tmp.wav
> >
> >
> > Since your goal is to use mod_unimrcp I think you should not spend your
> time with mod_tts_commandline.
> > Anyway, what do you get when you run these in the bash shell:
> >
> > which text2wave
> > echo 'This is an example of using tts_commandline' | text2wave -f 8000 >
> test.wav
> > file test.wav
> > ls -l test.wav
> >
> > Did you install festival? (text2wave comes with it)
> >
> >
> >
> > _________________________________________________________________________
> >
> > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> > Build your next product on our scalable cloud platform.
> >
> > Join our online community to chat in real time
> https://signalwire.community
> >
> > Professional FreeSWITCH Services
> > sales at freeswitch.com
> > https://freeswitch.com
> >
> > Official FreeSWITCH Sites
> > https://freeswitch.com/oss
> > https://freeswitch.org/confluence
> > https://cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > https://freeswitch.com
>
>
>
> --
> Lewis Bergman
> 325-439-0533 Cell
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users 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>
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From mayamatakeshi at gmail.com  Thu Mar  4 23:05:23 2021
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Fri, 5 Mar 2021 08:05:23 +0900
Subject: [Freeswitch-users] mod_sndfile error called from
 mod_tts_commandline - [WARNING] mod_sndfile.c:281 Error Opening File
In-Reply-To: <CAD2CnaTKNWq3FJ1yJujW=MXTwNMJ6Wn7QNf+aDL9CVaORVBBjA@mail.gmail.com>
References: <CAD2CnaQHdQ4pL1uXaGumGS+pM6MGyos9s+w2EUXJD52T0goB0w@mail.gmail.com>
 <CABaNFCabu6equvag0nkwvLm+eDZNf3VLgnop4SpZ_PMWCm4qmw@mail.gmail.com>
 <CAD2CnaTKNWq3FJ1yJujW=MXTwNMJ6Wn7QNf+aDL9CVaORVBBjA@mail.gmail.com>
Message-ID: <CABaNFCZ+fPPyOHqzszEHRQ_1hbmacWLZoM0jPyC01K1240r7kQ@mail.gmail.com>

On Fri, Mar 5, 2021 at 4:41 AM Lewis Bergman <lewis.bergman at gmail.com>
wrote:

> That was exactly it for the record.
> #which text2wave
> #                         <-- before apt-get install festival
> #which text2wave
> /usr/bin/text2wave
> # echo 'This is an example of using tts_commandline' | text2wave -f
> 8000 > test.wav
> # file test.wav
> test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16
> bit, mono 8000 Hz
> # ls -l test.wav
> -rw-r--r-- 1 root root 60208 Mar  4 06:44 test.wav
> And the freeswitch debug log that shows it works:
> EXECUTE [depth=0] sofia/internal/102 at 66.151.243.45
> speak(tts_commandline|text2wave|This is an example of using
> tts_commandline)
> 2021-03-04 06:59:14.305216 [DEBUG] switch_ivr_play_say.c:3023 OPEN TTS
> tts_commandline
> 2021-03-04 06:59:14.305216 [DEBUG] switch_ivr_play_say.c:3033 Raw
> Codec Activated
> 2021-03-04 06:59:14.305216 [DEBUG] mod_tts_commandline.c:160
> Executing: echo 'This is an example of using tts_commandline' |
> text2wave -f 8000 >
> '/tmp/05e71313-2a4d-48ce-a1e9-84600859112f.tmp.wav'
> 2021-03-04 06:59:15.545212 [DEBUG] switch_ivr_play_say.c:2741 Speaking
> text: This is an example of using tts_commandline
> 2021-03-04 06:59:15.545212 [DEBUG] sofia.c:7326 Channel
> sofia/internal/102 at 66.151.243.45 entering state [ready][200]
> 2021-03-04 06:59:15.565210 [DEBUG] switch_rtp.c:7759 Correct audio
> ip/port confirmed.
> 2021-03-04 06:59:19.305225 [DEBUG] switch_ivr_play_say.c:2905 done
> speaking text
>
> Thanks for the help.
> The reason I tried to start with tts_commandine is I have been
> struggling for a couple of weeks to get MRCP to produce something
> useful so I thought I would try something simpler and I could swear
> mod_tts_commandline was listed as a requirement.


Working with mod_unimrcp should be simple.
But maybe this can be of help:
  https://github.com/MayamaTakeshi/mrcp_client
It is a node.js app that I wrote to test mrcp servers.
I don't know which MRCP server you are using (maybe unimrcp?).
But you could make requests directly to it and check if it's working as
expected without having to deal with FS.
Then once things are cleared up you can work on the FS configuration.

Also, if you want to get down checking SIP/MRCP messages, you could use
this sngrep fork:
  https://github.com/MayamaTakeshi/sngrep/tree/mrcp_support
(this is a patch that I wrote to support MRCP on sngrep, but this will not
be merged to upstream as sngrep main branch is only accepting bug fixes
right now: https://github.com/irontec/sngrep/pull/346) so you would need to
build it yourself.
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From davidswalkabout at gmail.com  Fri Mar  5 02:50:58 2021
From: davidswalkabout at gmail.com (David P)
Date: Fri, 5 Mar 2021 15:50:58 +1300
Subject: [Freeswitch-users] rtp-timeout-sec VS media_timeout
In-Reply-To: <mailman.1490.1614883434.1264.freeswitch-users@lists.freeswitch.org>
References: <mailman.1490.1614883434.1264.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAE23trtNCgKga7E6SFqG1isPQ+beQL3ACVs2dLt3DTNSLvLcVQ@mail.gmail.com>

We're using FS 10.5 on Debian Stretch installed via packages.

<param name="rtp-timeout-sec" value="300"/> in our
sip_profiles/internal.xml does effect calls. We have no media_timeout*
config.
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From brian at freeswitch.com  Fri Mar  5 17:22:46 2021
From: brian at freeswitch.com (Brian West)
Date: Fri, 5 Mar 2021 11:22:46 -0600
Subject: [Freeswitch-users] rtp-timeout-sec VS media_timeout
In-Reply-To: <CAE23trtNCgKga7E6SFqG1isPQ+beQL3ACVs2dLt3DTNSLvLcVQ@mail.gmail.com>
References: <mailman.1490.1614883434.1264.freeswitch-users@lists.freeswitch.org>
 <CAE23trtNCgKga7E6SFqG1isPQ+beQL3ACVs2dLt3DTNSLvLcVQ@mail.gmail.com>
Message-ID: <CAEJMVkASGE6Q0WePHc6SWPReSU7OR_HUw=JO6-Z92gctgK4UCw@mail.gmail.com>

David,

The profile level option is deprecated, These variables can be set per
session media_timeout plus media_timeout_audio, media_timeout_video, then
you have media_hold_timeout plus media_hold_timeout_audio
and media_hold_timeout_video

/b


On Thu, Mar 4, 2021 at 10:04 PM David P <davidswalkabout at gmail.com> wrote:

> We're using FS 10.5 on Debian Stretch installed via packages.
>
> <param name="rtp-timeout-sec" value="300"/> in our
> sip_profiles/internal.xml does effect calls. We have no media_timeout*
> config.
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users 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>
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From david.villaume at sewan.fr  Thu Mar  4 22:28:24 2021
From: david.villaume at sewan.fr (David VILLAUME)
Date: Thu, 4 Mar 2021 22:28:24 +0000
Subject: [Freeswitch-users] Freeswitch loacally acks 200 in bypass
	media	scenarios
In-Reply-To: <CAEJMVkDvcTX2UnmqifLYO6xSAwfRJfbkA3WuMRX-tXuLwRXE9w@mail.gmail.com>
References: <PR1P264MB1166B5F172F31EC66BC2AC89FD989@PR1P264MB1166.FRAP264.PROD.OUTLOOK.COM>
 <CAEJMVkDvcTX2UnmqifLYO6xSAwfRJfbkA3WuMRX-tXuLwRXE9w@mail.gmail.com>
Message-ID: <PR1P264MB116648A70B93B5E9892D37C2FD979@PR1P264MB1166.FRAP264.PROD.OUTLOOK.COM>

I mean that freeswitch sends the Ack on the Bleg,  before propagating to the 200 to the A leg.

I would imagine in a bypass media scenario the Ack would handle the 200 with SDP as a proxy, am I wrong about it ?

Regards,
David

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Brian West
Sent: Thursday, March 4, 2021 8:39 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Freeswitch loacally acks 200 in bypass media scenarios

What do you mean 'locally acks'?

On Thu, Mar 4, 2021 at 1:25 PM David VILLAUME <david.villaume at sewan.fr<mailto:david.villaume at sewan.fr>> wrote:
Hello,

In bypass-media mode freeswitch locally acks the 200 ok (SDP), that leads to some Race condition when the party that answered the 200 sends a quick reinvite after Just after the Ack.

I tried on 1.10.1 and 1.10.5, with and without late-negociation, but couldn’t figure how to avoid this.

Has one of you experienced it or have any idea about this ?


          SBC                FREESWITCH                PBX
+---------          ----------+---------          ----------+
|        INVITE (SDP)         |                             |
| --------------------------> |                             |
|         100 Trying          |                             |
| <-------------------------- |                             |
|                             |        INVITE (SDP)         |
|                             | --------------------------> |
|                             |         100 Trying          |
|                             | <-------------------------- |
|                             |         180 Ringing         |
|                             | <-------------------------- |
|         180 Ringing         |                             |
| <-------------------------- |                             |
|                             |        200 OK (SDP)         |
|                             | <-------------------------- |
|                             |             ACK             |
|                             | --------------------------> |
|        200 OK (SDP)         |                             |
| <-------------------------- |                             |
|                             |        INVITE (SDP)         |
|                             | <-------------------------- |
|                             |             100             |
|                             | --------------------------> |
|        INVITE (SDP)         |                             |
| <-------------------------- |                             |
|             100             |                             |
| --------------------------> |                             |
|             491             |                             |
| --------------------------> |                             |
|                             |             491             |
|                             | --------------------------> |

Regards,
David
_________________________________________________________________________

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

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

Professional FreeSWITCH Services
sales at freeswitch.com<mailto:sales at freeswitch.com>
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com


--


Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com<mailto:sales at freeswitch.com>

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

Email: brian at freeswitch.com<mailto:brian at freeswitch.com>

Mobile: 918-424-9378

Website: https://www.FreeSWITCH.com<https://www.freeswitch.com/>

[https://lh6.googleusercontent.com/AYfRoSNaDNtMPRMevPn_GqcVEMd5NDRFi0GlluGUWzV6I5TAY_3T2-Tt0IuIXeUtEdYsgNsM8DOYKRKhjmrG_-n2Ga-LCnoNk46sO8VyEma1sBFYdiGJcLRUvkrD1CYHN79qimeg]<https://www.facebook.com/freeswitch>[https://lh3.googleusercontent.com/W4SqXyybH2qdAozvtoKjcz736qOjk9LHDwldvs1ahc-WVU0putVMSsUH474KDrJ32jsqi6JDjyUWxqeEkN5I1xSlC5ShYrd1b8NIMUkDzDrtbWQfa6A_90UcygqesBtRLgeFirKa]<https://twitter.com/freeswitch>
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From lewis.bergman at gmail.com  Fri Mar  5 17:03:03 2021
From: lewis.bergman at gmail.com (Lewis Bergman)
Date: Fri, 5 Mar 2021 11:03:03 -0600
Subject: [Freeswitch-users] mod_sndfile error called from
 mod_tts_commandline - [WARNING] mod_sndfile.c:281 Error Opening File
In-Reply-To: <CABaNFCZ+fPPyOHqzszEHRQ_1hbmacWLZoM0jPyC01K1240r7kQ@mail.gmail.com>
References: <CAD2CnaQHdQ4pL1uXaGumGS+pM6MGyos9s+w2EUXJD52T0goB0w@mail.gmail.com>
 <CABaNFCabu6equvag0nkwvLm+eDZNf3VLgnop4SpZ_PMWCm4qmw@mail.gmail.com>
 <CAD2CnaTKNWq3FJ1yJujW=MXTwNMJ6Wn7QNf+aDL9CVaORVBBjA@mail.gmail.com>
 <CABaNFCZ+fPPyOHqzszEHRQ_1hbmacWLZoM0jPyC01K1240r7kQ@mail.gmail.com>
Message-ID: <CAD2CnaQqCVPVu1fR22M9zqtg8Umsa+jZCceunAHvq7wsJkHaqw@mail.gmail.com>

Thanks so much! That is very helpful info. Let me create a new thread
for this and explain what I have done so far.

On Thu, Mar 4, 2021 at 5:48 PM mayamatakeshi <mayamatakeshi at gmail.com> wrote:
>
>
>
> On Fri, Mar 5, 2021 at 4:41 AM Lewis Bergman <lewis.bergman at gmail.com> wrote:
>>
>> That was exactly it for the record.
>> #which text2wave
>> #                         <-- before apt-get install festival
>> #which text2wave
>> /usr/bin/text2wave
>> # echo 'This is an example of using tts_commandline' | text2wave -f
>> 8000 > test.wav
>> # file test.wav
>> test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16
>> bit, mono 8000 Hz
>> # ls -l test.wav
>> -rw-r--r-- 1 root root 60208 Mar  4 06:44 test.wav
>> And the freeswitch debug log that shows it works:
>> EXECUTE [depth=0] sofia/internal/102 at 66.151.243.45
>> speak(tts_commandline|text2wave|This is an example of using
>> tts_commandline)
>> 2021-03-04 06:59:14.305216 [DEBUG] switch_ivr_play_say.c:3023 OPEN TTS
>> tts_commandline
>> 2021-03-04 06:59:14.305216 [DEBUG] switch_ivr_play_say.c:3033 Raw
>> Codec Activated
>> 2021-03-04 06:59:14.305216 [DEBUG] mod_tts_commandline.c:160
>> Executing: echo 'This is an example of using tts_commandline' |
>> text2wave -f 8000 >
>> '/tmp/05e71313-2a4d-48ce-a1e9-84600859112f.tmp.wav'
>> 2021-03-04 06:59:15.545212 [DEBUG] switch_ivr_play_say.c:2741 Speaking
>> text: This is an example of using tts_commandline
>> 2021-03-04 06:59:15.545212 [DEBUG] sofia.c:7326 Channel
>> sofia/internal/102 at 66.151.243.45 entering state [ready][200]
>> 2021-03-04 06:59:15.565210 [DEBUG] switch_rtp.c:7759 Correct audio
>> ip/port confirmed.
>> 2021-03-04 06:59:19.305225 [DEBUG] switch_ivr_play_say.c:2905 done speaking text
>>
>> Thanks for the help.
>> The reason I tried to start with tts_commandine is I have been
>> struggling for a couple of weeks to get MRCP to produce something
>> useful so I thought I would try something simpler and I could swear
>> mod_tts_commandline was listed as a requirement.
>
>
> Working with mod_unimrcp should be simple.
> But maybe this can be of help:
>   https://github.com/MayamaTakeshi/mrcp_client
> It is a node.js app that I wrote to test mrcp servers.
> I don't know which MRCP server you are using (maybe unimrcp?).
> But you could make requests directly to it and check if it's working as expected without having to deal with FS.
> Then once things are cleared up you can work on the FS configuration.
>
> Also, if you want to get down checking SIP/MRCP messages, you could use this sngrep fork:
>   https://github.com/MayamaTakeshi/sngrep/tree/mrcp_support
> (this is a patch that I wrote to support MRCP on sngrep, but this will not be merged to upstream as sngrep main branch is only accepting bug fixes right now: https://github.com/irontec/sngrep/pull/346) so you would need to build it yourself.
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Lewis Bergman
325-439-0533 Cell


From brian at freeswitch.com  Fri Mar  5 20:50:40 2021
From: brian at freeswitch.com (Brian West)
Date: Fri, 5 Mar 2021 14:50:40 -0600
Subject: [Freeswitch-users] Freeswitch loacally acks 200 in bypass media
	scenarios
In-Reply-To: <PR1P264MB116648A70B93B5E9892D37C2FD979@PR1P264MB1166.FRAP264.PROD.OUTLOOK.COM>
References: <PR1P264MB1166B5F172F31EC66BC2AC89FD989@PR1P264MB1166.FRAP264.PROD.OUTLOOK.COM>
 <CAEJMVkDvcTX2UnmqifLYO6xSAwfRJfbkA3WuMRX-tXuLwRXE9w@mail.gmail.com>
 <PR1P264MB116648A70B93B5E9892D37C2FD979@PR1P264MB1166.FRAP264.PROD.OUTLOOK.COM>
Message-ID: <CAEJMVkBLW56t0u0GEU8Ds4BmDmrMLbfHT0xWe8KHE+AXMrim8Q@mail.gmail.com>

That's because we're not a proxy.

On Fri, Mar 5, 2021 at 1:26 PM David VILLAUME <david.villaume at sewan.fr>
wrote:

> I mean that freeswitch sends the Ack on the Bleg,  before propagating to
> the 200 to the A leg.
>
>
>
> I would imagine in a bypass media scenario the Ack would handle the 200
> with SDP as a proxy, am I wrong about it ?
>
>
>
> Regards,
>
> David
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Brian West
> *Sent:* Thursday, March 4, 2021 8:39 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Freeswitch loacally acks 200 in bypass
> media scenarios
>
>
>
> What do you mean 'locally acks'?
>
>
>
> On Thu, Mar 4, 2021 at 1:25 PM David VILLAUME <david.villaume at sewan.fr>
> wrote:
>
> Hello,
>
>
>
> In bypass-media mode freeswitch locally acks the 200 ok (SDP), that leads
> to some Race condition when the party that answered the 200 sends a quick
> reinvite after Just after the Ack.
>
>
>
> I tried on 1.10.1 and 1.10.5, with and without late-negociation, but
> couldn’t figure how to avoid this.
>
>
>
> Has one of you experienced it or have any idea about this ?
>
>
>
>
>
>
>           SBC                FREESWITCH                PBX
>
> +---------          ----------+---------          ----------+
>
> |        INVITE (SDP)         |                             |
>
> | --------------------------> |                             |
>
> |         100 Trying          |                             |
>
> | <-------------------------- |                             |
>
> |                             |        INVITE (SDP)         |
>
> |                             | --------------------------> |
>
> |                             |         100 Trying          |
>
> |                             | <-------------------------- |
>
> |                             |         180 Ringing         |
>
> |                             | <-------------------------- |
>
> |         180 Ringing         |                             |
>
> | <-------------------------- |                             |
>
> |                             |        200 OK (SDP)         |
>
> |                             | <-------------------------- |
>
> |                             |             ACK             |
>
> |                             | --------------------------> |
>
> |        200 OK (SDP)         |                             |
>
> | <-------------------------- |                             |
>
> |                             |        INVITE (SDP)         |
>
> |                             | <-------------------------- |
>
> |                             |             100             |
>
> |                             | --------------------------> |
>
> |        INVITE (SDP)         |                             |
>
> | <-------------------------- |                             |
>
> |             100             |                             |
>
> | --------------------------> |                             |
>
> |             491             |                             |
>
> | --------------------------> |                             |
>
> |                             |             491             |
>
> |                             | --------------------------> |
>
>
>
> Regards,
>
> David
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users 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/>
>
> <https://www.facebook.com/freeswitch> <https://twitter.com/freeswitch>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

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>
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From rasheed.kalapurackal at gmail.com  Sat Mar  6 18:19:28 2021
From: rasheed.kalapurackal at gmail.com (Rasheed Kalapurackal)
Date: Sat, 6 Mar 2021 23:49:28 +0530
Subject: [Freeswitch-users] Installing Modules
Message-ID: <CAKU4o50sSOxAhW1=kWcqZ0N66Ru=g9Uqf0ti8qTpYYzKs+3vxA@mail.gmail.com>

Hello ,

            I completed the installation of Freeswitch on Debian 10 . Now i
need to install an additional module in to the Freeswitch .  Is it possible
to install Modules in to an already deployed Freeswitch instance ?  Any
quick help on this would be greatly appreciated.

Thanks

Abdul Rasheed
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From botelist at gmail.com  Mon Mar  8 19:31:52 2021
From: botelist at gmail.com (Bote Man)
Date: Mon, 8 Mar 2021 14:31:52 -0500
Subject: [Freeswitch-users] Installing Modules
In-Reply-To: <CAKU4o50sSOxAhW1=kWcqZ0N66Ru=g9Uqf0ti8qTpYYzKs+3vxA@mail.gmail.com>
References: <CAKU4o50sSOxAhW1=kWcqZ0N66Ru=g9Uqf0ti8qTpYYzKs+3vxA@mail.gmail.com>
Message-ID: <018501d71451$b1765df0$146319d0$@gmail.com>

I am not certain about other dependencies, but you simply uncomment the line in 

${conf_dir}/autoload_configs/modules.conf.xml

to tell FreeSWITCH to load that module.

 

Hope this helps.

 

 

--- 

John Boteler 

BnC Group U.S.A. 

 

 

 

From: Rasheed Kalapurackal
Sent: Saturday, 6 March, 2021 13:19
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Installing Modules

 

Hello , 

 

            I completed the installation of Freeswitch on Debian 10 . Now i need to install an additional module in to the Freeswitch .  Is it possible to install Modules in to an already deployed Freeswitch instance ?  Any quick help on this would be greatly appreciated. 

 

Thanks 

 

Abdul Rasheed

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From david.villasmil.work at gmail.com  Mon Mar  8 19:49:04 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 8 Mar 2021 19:49:04 +0000
Subject: [Freeswitch-users] Installing Modules
In-Reply-To: <018501d71451$b1765df0$146319d0$@gmail.com>
References: <CAKU4o50sSOxAhW1=kWcqZ0N66Ru=g9Uqf0ti8qTpYYzKs+3vxA@mail.gmail.com>
 <018501d71451$b1765df0$146319d0$@gmail.com>
Message-ID: <CAFGRPVqXYo1j7j_8hqZTE0eO-MS30jF-n-dB8AWeYa4iYdDQKA@mail.gmail.com>

How did you install? Is the modules already in the libs path? if so, just
edit modules.conf and uncommet the module.

Regards,

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


On Mon, Mar 8, 2021 at 7:32 PM Bote Man <botelist at gmail.com> wrote:

> I am not certain about other dependencies, but you simply uncomment the
> line in
>
> ${conf_dir}/autoload_configs/modules.conf.xml
>
> to tell FreeSWITCH to load that module.
>
>
>
> Hope this helps.
>
>
>
>
>
> ---
>
> John Boteler
>
> BnC Group U.S.A.
>
>
>
>
>
>
>
> *From:* Rasheed Kalapurackal
> *Sent:* Saturday, 6 March, 2021 13:19
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Installing Modules
>
>
>
> Hello ,
>
>
>
>             I completed the installation of Freeswitch on Debian 10 . Now
> i need to install an additional module in to the Freeswitch .  Is it
> possible to install Modules in to an already deployed Freeswitch instance
> ?  Any quick help on this would be greatly appreciated.
>
>
>
> Thanks
>
>
>
> Abdul Rasheed
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From lewis.bergman at gmail.com  Mon Mar  8 19:27:21 2021
From: lewis.bergman at gmail.com (Lewis Bergman)
Date: Mon, 8 Mar 2021 13:27:21 -0600
Subject: [Freeswitch-users] Installing Modules
In-Reply-To: <CAKU4o50sSOxAhW1=kWcqZ0N66Ru=g9Uqf0ti8qTpYYzKs+3vxA@mail.gmail.com>
References: <CAKU4o50sSOxAhW1=kWcqZ0N66Ru=g9Uqf0ti8qTpYYzKs+3vxA@mail.gmail.com>
Message-ID: <CAD2CnaTD5+RpVo32fosYgXkgySfGW7sgW1M1b2K+FD+im0yG7A@mail.gmail.com>

apt-get install freeswitch-mod-whatever should work.
-- 
Lewis Bergman
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From josefu at gmail.com  Tue Mar  9 10:39:48 2021
From: josefu at gmail.com (=?UTF-8?Q?Jose_Fco=2E_Irles_Dur=C3=A1?=)
Date: Tue, 9 Mar 2021 11:39:48 +0100
Subject: [Freeswitch-users] Problem with answered call and multiple
	re-invites
Message-ID: <CAJOPU6Umk6bV9H8_o-deX491neA9pSdP5XFRGLw-e6Bzx2O9og@mail.gmail.com>

Hi,

I have an audio problem with some user agents (Alcatel OXO) that sends
multiple re-invites.

Example scenario:

FreeSWITCH with public IP (example: 1.1.1.1)
Alcatel behind NAT (private IP 192.168.20.200, public IP 2.2.2.2)

1. I send the call to Alcatel and the sdp negotiation works well. The
remote port for RTP is 32014.
In the capture I can see that FreeSWITCH sends first RTP packets to
the IP 192.168.20.200 and port 32014. Later, the audio stream from
Alcatel arrives from IP 2.2.2.2 and port 32014 and FreeSWITCH changes
and sends RTP to 2.2.2.2:32014 (rtp auto adjust), so I can listen the
audio (the Alcatel has an IVR menu).

2. In the IVR menu I press one option and the Alcatel sends me an
INVITE changing media. It sends a new port (32016), FreeSWITCH accepts
the re-invite without changing its source port.
In the capture I can see that FreeSWITCH sends first RTP packets to
the IP 192.168.20.200 and port 32016, but when the first packets from
the real IP:PORT (2.2.2.2:32016) arrives, FreeSWITCH don't change its
RTP stream to the new port, so I can't hear the audio from the
Alcatel.

3. Later, FreeSWITCH receives another re-invite changing remote port to 32020.
First, FreeSWITCH sends audio to the private IP:PORT
(192.168.20.200:32020), but when it receives the real stream from
2.2.2.2:32020, it doesn't change to the new IP:PORT (2.2.2.2:32020).

>From point 2, I don't hear anything.

This problem happens with the last stable version (1.10.5).

I think that it's a bug, but I don't know if there are some config
that change that behaviour.

I have uploaded the capture (png):
https://i.imgur.com/emvnaCt.png

Best regards

--
Jose Fco. Irles Durá


From rasheed.kalapurackal at gmail.com  Tue Mar  9 16:21:13 2021
From: rasheed.kalapurackal at gmail.com (Rasheed Kalapurackal)
Date: Tue, 9 Mar 2021 21:51:13 +0530
Subject: [Freeswitch-users] Installing Modules
In-Reply-To: <018501d71451$b1765df0$146319d0$@gmail.com>
References: <CAKU4o50sSOxAhW1=kWcqZ0N66Ru=g9Uqf0ti8qTpYYzKs+3vxA@mail.gmail.com>
 <018501d71451$b1765df0$146319d0$@gmail.com>
Message-ID: <CAKU4o51uqn+8=tMf_h-nYt47noNHS52xiNK1fLcAuLS7yEW1gA@mail.gmail.com>

Hello John ,

I uncommented it from the module.conf.xml , but it seems that the module
was not there in the lib directory. .  Since it is a custom module , i
could not install it  using apt-get install freeswitch-mod-whatever .  But
i have the .so file which i can copy it to lib directory , i will try if
that will resolve the issue.

Thanks
Rasheed

On Tue, Mar 9, 2021 at 1:17 AM Bote Man <botelist at gmail.com> wrote:

> I am not certain about other dependencies, but you simply uncomment the
> line in
>
> ${conf_dir}/autoload_configs/modules.conf.xml
>
> to tell FreeSWITCH to load that module.
>
>
>
> Hope this helps.
>
>
>
>
>
> ---
>
> John Boteler
>
> BnC Group U.S.A.
>
>
>
>
>
>
>
> *From:* Rasheed Kalapurackal
> *Sent:* Saturday, 6 March, 2021 13:19
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Installing Modules
>
>
>
> Hello ,
>
>
>
>             I completed the installation of Freeswitch on Debian 10 . Now
> i need to install an additional module in to the Freeswitch .  Is it
> possible to install Modules in to an already deployed Freeswitch instance
> ?  Any quick help on this would be greatly appreciated.
>
>
>
> Thanks
>
>
>
> Abdul Rasheed
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From brian at freeswitch.com  Tue Mar  9 16:42:11 2021
From: brian at freeswitch.com (Brian West)
Date: Tue, 9 Mar 2021 10:42:11 -0600
Subject: [Freeswitch-users] Installing Modules
In-Reply-To: <CAKU4o51uqn+8=tMf_h-nYt47noNHS52xiNK1fLcAuLS7yEW1gA@mail.gmail.com>
References: <CAKU4o50sSOxAhW1=kWcqZ0N66Ru=g9Uqf0ti8qTpYYzKs+3vxA@mail.gmail.com>
 <018501d71451$b1765df0$146319d0$@gmail.com>
 <CAKU4o51uqn+8=tMf_h-nYt47noNHS52xiNK1fLcAuLS7yEW1gA@mail.gmail.com>
Message-ID: <CAEJMVkCDBvBf5X-UwhQFohuUz+OibATS04hOavXAVf7NbnWDPQ@mail.gmail.com>

Sounds like you have your own custom module that you're trying to install
or build, what type of module is it?

Thanks,
Brina


On Tue, Mar 9, 2021 at 10:36 AM Rasheed Kalapurackal <
rasheed.kalapurackal at gmail.com> wrote:

> Hello John ,
>
> I uncommented it from the module.conf.xml , but it seems that the module
> was not there in the lib directory. .  Since it is a custom module , i
> could not install it  using apt-get install freeswitch-mod-whatever .  But
> i have the .so file which i can copy it to lib directory , i will try if
> that will resolve the issue.
>
> Thanks
> Rasheed
>
> On Tue, Mar 9, 2021 at 1:17 AM Bote Man <botelist at gmail.com> wrote:
>
>> I am not certain about other dependencies, but you simply uncomment the
>> line in
>>
>> ${conf_dir}/autoload_configs/modules.conf.xml
>>
>> to tell FreeSWITCH to load that module.
>>
>>
>>
>> Hope this helps.
>>
>>
>>
>>
>>
>> ---
>>
>> John Boteler
>>
>> BnC Group U.S.A.
>>
>>
>>
>>
>>
>>
>>
>> *From:* Rasheed Kalapurackal
>> *Sent:* Saturday, 6 March, 2021 13:19
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* [Freeswitch-users] Installing Modules
>>
>>
>>
>> Hello ,
>>
>>
>>
>>             I completed the installation of Freeswitch on Debian 10 . Now
>> i need to install an additional module in to the Freeswitch .  Is it
>> possible to install Modules in to an already deployed Freeswitch instance
>> ?  Any quick help on this would be greatly appreciated.
>>
>>
>>
>> Thanks
>>
>>
>>
>> Abdul Rasheed
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

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>
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From marcb at voicemeup.com  Mon Mar  8 20:48:40 2021
From: marcb at voicemeup.com (Marc Bernard)
Date: Mon, 8 Mar 2021 15:48:40 -0500
Subject: [Freeswitch-users] Odd RTP skew behavior
In-Reply-To: <CAEJMVkCmCK5jYemysSUmmNH+u=5kU92QVi8aW+W5LqDYtDj17A@mail.gmail.com>
References: <14b301d6f650$ffa9c120$fefd4360$@voicemeup.com>
 <CAEJMVkCmCK5jYemysSUmmNH+u=5kU92QVi8aW+W5LqDYtDj17A@mail.gmail.com>
Message-ID: <07d301d7145c$6bc86180$43592480$@voicemeup.com>

Sorry for late reply.

I think this is mostly happen with both 1.6.20 and 1.10.5 in the path


--


From: Brian West <brian at freeswitch.com> 
Sent: Monday, February 1, 2021 11:51
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Odd RTP skew behavior

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




From arslan.saeed at overthewire.com.au  Tue Mar  9 05:27:51 2021
From: arslan.saeed at overthewire.com.au (Arslan Saeed)
Date: Tue, 9 Mar 2021 16:27:51 +1100
Subject: [Freeswitch-users] Freeswitch DTMF issue - wrong timestamp base
Message-ID: <CA+QohWiLc3d2SYiO6LOHdynaEqN65uxatjcA3ZxQrzO+ULUJRA@mail.gmail.com>

Hi All,

We have come across an issue where freeswitch ((Version 1.10.5
-release-17-25569c1631 64bit)
seems to be doing something wrong with regards to RFC4733 DTMF event
packets generation.

What we see is that freeswitch would start using a new timestamp base for
the RTP event packets (sometimes even for a few RTP packets preceeding the
DTMF packetS) and then at the end of the DTMF event revert to older
timestamp base. Although the marker bit is set on when new timestamp base
is used but as far as I know this is not RFC compliant and causes issues
for some of the PBX devices.

RFC 4733, section 2.5.1.2 says:
DTMF digits and other named telephone events are carried as part of
the audio stream, and they MUST use the same sequence number and
timestamp base as the regular audio channel to simplify the
generation of audio waveforms at a gateway.```
https://tools.ietf.org/html/rfc4733#section-2.5.1

--

I think if freeswitch were to change the timestamp for DTMF event packets
then it must also change the RTP  SSRC at same time (which does not happen)
Although I don't understand the need to change timestamp base at all for
the start of DTMF event  packets or even with the few proceeding RTP
packets.

I have tried to use following parameter but that does not help
<param name="auto-rtp-bugs" data="CHANGE_SSRC_ON_MARKER"/>

 I have also tried to test with these additional parameters (in order to
infleunce freeswitch to not use a new timestamp base for the start of dtmf
event packets)  but no change in freeswitch behaviour.

  <param name="pass-rfc2833" value="true"/>
<param name="rtp-rewrite-timestamps"value="true"/>
<param name="auto-rtp-bugs"value="CISCO_SKIP_MARK_BIT_2833"/>
<param name="auto-rtp-bugs"value="clear"/>

In addition I have tried to use some jitter related sofia profile
parameters but no affect.

Another DTMF related issue that I have seen with freeswitch is that
it sends DTMF event packets in a short burst and not evenly spaced out per
the codec sample rate.
For example if RTP is being sent at 20ms, freeswitch would send dtmf event
packets at different interval like it may send 8ms apart or 13 ms apat and
at times it sends multiple event packets just a few ms apart.

To fix that behaviour, when we explicity set the RTP-timer-name to "soft"
using

<param name="rtp-timer-name" value="soft"/>


 that fixes it and ensures evenly spaced DTMF packets say 20 ms apart (that
fixes interop with some of the devices)

Any help is highly appreciated.

Thanks
Arslan
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From mircea.huh at gmail.com  Tue Mar  9 13:56:14 2021
From: mircea.huh at gmail.com (Mircea Botoca-Huh)
Date: Tue, 9 Mar 2021 15:56:14 +0200
Subject: [Freeswitch-users] No audio on bridged call
Message-ID: <2BB61A75-937C-4D36-A913-E82C38D65701@hxcore.ol>

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From lewis.bergman at gmail.com  Wed Mar 10 01:44:05 2021
From: lewis.bergman at gmail.com (Lewis Bergman)
Date: Tue, 9 Mar 2021 19:44:05 -0600
Subject: [Freeswitch-users] using mod_unimrcp to transcribe voicemails
Message-ID: <CAD2CnaQ4L_XF8axypdd+j=6KBqKkg50fcFpcx5vuQuJuP_vAGg@mail.gmail.com>

Does anyone have guidance or code to get this to work? I have looked at the
transcription documentation in Freeswitch and it all seems to be related to
command line type implementations using Google, Wartson, Azure, etc, with
curl. using a FusionPBX Lua script (
/var/www/fusionpbx/app/scripts/resources/scripts/app/voicemail/resources/functions/record_message.lua
)
If someone has made voicemail transcription work with mod_unimrcp I would
be grateful in hearing how it was done. It seems like some Lua programming
might be required at the minimum. If making it work is described an idea of
where to start looking for documentation would be great.
-- 
Lewis Bergman
325-439-0533 Cell
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From max at nysolutions.com  Wed Mar 10 02:25:05 2021
From: max at nysolutions.com (Moishe Grunstein)
Date: Wed, 10 Mar 2021 02:25:05 +0000
Subject: [Freeswitch-users] using mod_unimrcp to transcribe voicemails
In-Reply-To: <CAD2CnaQ4L_XF8axypdd+j=6KBqKkg50fcFpcx5vuQuJuP_vAGg@mail.gmail.com>
References: <CAD2CnaQ4L_XF8axypdd+j=6KBqKkg50fcFpcx5vuQuJuP_vAGg@mail.gmail.com>
Message-ID: <a3a24a932c994635bd27cb03d29dda38@nysolutions.com>

See https://docs.fusionpbx.com/en/latest/applications/voicemail_transcription.html?highlight=transcribe
And https://docs.fusionpbx.com/en/latest/applications/voicemail.html

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com<mailto:support at nysolutions.com>
[cid:image001.jpg at 01D7152B.393CEC70]<http://www.nysolutions.com/>
Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Lewis Bergman
Sent: Tuesday, March 9, 2021 8:44 PM
To: FreeSWITCH Users Help <FreeSWITCH-users at lists.freeswitch.org>
Subject: [Freeswitch-users] using mod_unimrcp to transcribe voicemails

Does anyone have guidance or code to get this to work? I have looked at the transcription documentation in Freeswitch and it all seems to be related to command line type implementations using Google, Wartson, Azure, etc, with curl. using a FusionPBX Lua script (/var/www/fusionpbx/app/scripts/resources/scripts/app/voicemail/resources/functions/record_message.lua)
If someone has made voicemail transcription work with mod_unimrcp I would be grateful in hearing how it was done. It seems like some Lua programming might be required at the minimum. If making it work is described an idea of where to start looking for documentation would be great.
--
Lewis Bergman
325-439-0533 Cell
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From lewis.bergman at gmail.com  Wed Mar 10 04:07:06 2021
From: lewis.bergman at gmail.com (Lewis Bergman)
Date: Tue, 9 Mar 2021 22:07:06 -0600
Subject: [Freeswitch-users] using mod_unimrcp to transcribe voicemails
In-Reply-To: <a3a24a932c994635bd27cb03d29dda38@nysolutions.com>
References: <CAD2CnaQ4L_XF8axypdd+j=6KBqKkg50fcFpcx5vuQuJuP_vAGg@mail.gmail.com>
 <a3a24a932c994635bd27cb03d29dda38@nysolutions.com>
Message-ID: <CAD2CnaSrqRq6ESmYEMVfgbD5xTZkdo4LU4nrsRC7C_biEj69NA@mail.gmail.com>

Thanks for the response but I have looked at those.

See
> https://docs.fusionpbx.com/en/latest/applications/voicemail_transcription.html?highlight=transcribe
>
The file that these setting use is
${baseidr}/fusionpbx/app/scripts/resources/scripts/app/voicemail/resources/functions/record_message.lua
None of the options there (presumably custom) fit, or I don't believe they
will work, for mod_unimrcp.


> And https://docs.fusionpbx.com/en/latest/applications/voicemail.html
>
>
>
I have looked at this one as well but I did find that on further review I
don't have the file mentioned at the very end on my system -
applications/voicemail_transcription.rst . That would seem to be important.
Maybe that is my missing link. Have you been able to get mod_unimrcp
working for vm transcription? I would think it is a fairly common thing to
do with unimrcp and freeswitch.
Thanks for your help,
-- 
Lewis Bergman
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From lewis.bergman at gmail.com  Wed Mar 10 04:11:22 2021
From: lewis.bergman at gmail.com (Lewis Bergman)
Date: Tue, 9 Mar 2021 22:11:22 -0600
Subject: [Freeswitch-users] using mod_unimrcp to transcribe voicemails
In-Reply-To: <CAD2CnaSrqRq6ESmYEMVfgbD5xTZkdo4LU4nrsRC7C_biEj69NA@mail.gmail.com>
References: <CAD2CnaQ4L_XF8axypdd+j=6KBqKkg50fcFpcx5vuQuJuP_vAGg@mail.gmail.com>
 <a3a24a932c994635bd27cb03d29dda38@nysolutions.com>
 <CAD2CnaSrqRq6ESmYEMVfgbD5xTZkdo4LU4nrsRC7C_biEj69NA@mail.gmail.com>
Message-ID: <CAD2CnaQzVSksVPXAgOMpDhyx49uk5Gr-=Dv=gAHDRaaVEQhsxw@mail.gmail.com>

Ahhh. That rst is a circular reference back to the first doc. Well that
doesn't help.
--
Lewis Bergman
325-439-0533 Cell
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From davidswalkabout at gmail.com  Wed Mar 10 05:06:16 2021
From: davidswalkabout at gmail.com (David P)
Date: Wed, 10 Mar 2021 18:06:16 +1300
Subject: [Freeswitch-users] using mod_unimrcp to transcribe voicemails
Message-ID: <CAE23trtEEiXZ-MurMTNVdLVuhZBR=_7nybTcawOcRL9haS16Ng@mail.gmail.com>

One of the ClueCon 2020 talks was by someone who uses mod_unimrcp and has a
github project showing its FS integration, iirc.
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From dragos at freeswitch.org  Wed Mar 10 13:04:19 2021
From: dragos at freeswitch.org (Dragos Oancea)
Date: Wed, 10 Mar 2021 15:04:19 +0200
Subject: [Freeswitch-users] No audio on bridged call
In-Reply-To: <2BB61A75-937C-4D36-A913-E82C38D65701@hxcore.ol>
References: <2BB61A75-937C-4D36-A913-E82C38D65701@hxcore.ol>
Message-ID: <CAGvJV4p-w-oqsKxhUkSMt0VsDHC9Mtkec6KzUtpgo87TU6HAjg@mail.gmail.com>

Check param ext-rtp-ip in the external SIP profile.


On Wed, Mar 10, 2021 at 12:35 AM Mircea Botoca-Huh <mircea.huh at gmail.com>
wrote:

> Hello,
>
>
>
> I have freeswitch version 1.10.5 -release-17-25569c1631 64bit installed.
>
>
>
> I have one SIP provider, no nat.
>
>
>
> I use a simple dialplan which bridge the received call from SIP provider
> to a mobile number using the same provider.
>
>
>
> The call goes through but I have no ringback tone and if I answer there is
> no audio.
>
>
>
> If I send the call to one internal extension, it is working fine.
>
>
>
> The only difference I can see in the logs is that, on the call with no
> audio, I do not have in logs „Correct ip/port confirmed”.
>
>
>
> Any suggestion is greatly appreciated.
>
>
>
> Thank you.
>
>
>
> Best regards,
>
> Mircea.
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From dragos at freeswitch.org  Wed Mar 10 13:15:24 2021
From: dragos at freeswitch.org (Dragos Oancea)
Date: Wed, 10 Mar 2021 15:15:24 +0200
Subject: [Freeswitch-users] Freeswitch DTMF issue - wrong timestamp base
In-Reply-To: <CA+QohWiLc3d2SYiO6LOHdynaEqN65uxatjcA3ZxQrzO+ULUJRA@mail.gmail.com>
References: <CA+QohWiLc3d2SYiO6LOHdynaEqN65uxatjcA3ZxQrzO+ULUJRA@mail.gmail.com>
Message-ID: <CAGvJV4pwg_CFYqotwa7k=USVqjbY_4CnYedYYES6auQLcwEg0Q@mail.gmail.com>

you could open a github issue:
https://github.com/signalwire/freeswitch/issues
Attach FS debug logs and pcap file.


On Wed, Mar 10, 2021 at 12:35 AM Arslan Saeed <
arslan.saeed at overthewire.com.au> wrote:

> Hi All,
>
> We have come across an issue where freeswitch ((Version 1.10.5
> -release-17-25569c1631 64bit)
> seems to be doing something wrong with regards to RFC4733 DTMF event
> packets generation.
>
> What we see is that freeswitch would start using a new timestamp base for
> the RTP event packets (sometimes even for a few RTP packets preceeding the
> DTMF packetS) and then at the end of the DTMF event revert to older
> timestamp base. Although the marker bit is set on when new timestamp base
> is used but as far as I know this is not RFC compliant and causes issues
> for some of the PBX devices.
>
> RFC 4733, section 2.5.1.2 says:
> DTMF digits and other named telephone events are carried as part of
> the audio stream, and they MUST use the same sequence number and
> timestamp base as the regular audio channel to simplify the
> generation of audio waveforms at a gateway.```
> https://tools.ietf.org/html/rfc4733#section-2.5.1
>
> --
>
> I think if freeswitch were to change the timestamp for DTMF event packets
> then it must also change the RTP  SSRC at same time (which does not happen)
> Although I don't understand the need to change timestamp base at all for
> the start of DTMF event  packets or even with the few proceeding RTP
> packets.
>
> I have tried to use following parameter but that does not help
> <param name="auto-rtp-bugs" data="CHANGE_SSRC_ON_MARKER"/>
>
>  I have also tried to test with these additional parameters (in order to
> infleunce freeswitch to not use a new timestamp base for the start of dtmf
> event packets)  but no change in freeswitch behaviour.
>
>   <param name="pass-rfc2833" value="true"/>
> <param name="rtp-rewrite-timestamps"value="true"/>
> <param name="auto-rtp-bugs"value="CISCO_SKIP_MARK_BIT_2833"/>
> <param name="auto-rtp-bugs"value="clear"/>
>
> In addition I have tried to use some jitter related sofia profile
> parameters but no affect.
>
> Another DTMF related issue that I have seen with freeswitch is that
> it sends DTMF event packets in a short burst and not evenly spaced out per
> the codec sample rate.
> For example if RTP is being sent at 20ms, freeswitch would send dtmf event
> packets at different interval like it may send 8ms apart or 13 ms apat and
> at times it sends multiple event packets just a few ms apart.
>
> To fix that behaviour, when we explicity set the RTP-timer-name to "soft"
> using
>
> <param name="rtp-timer-name" value="soft"/>
>
>
>  that fixes it and ensures evenly spaced DTMF packets say 20 ms apart
> (that fixes interop with some of the devices)
>
> Any help is highly appreciated.
>
> Thanks
> Arslan
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From vishalmpai at gmail.com  Wed Mar 10 13:21:38 2021
From: vishalmpai at gmail.com (Vishal Pai)
Date: Wed, 10 Mar 2021 18:51:38 +0530
Subject: [Freeswitch-users] NOTIFYs Event
Message-ID: <CACOmXtAx4O0ymzQVJcypxwy8VuZLrgprRyhNe7rfJeqKWRZqOQ@mail.gmail.com>

Hello Everyone

Few days back we had a issue with Notify request from park+*5903 at abc.com
the freeswitch was sending the notify request but the endpoint side was
responding late on checking it out it was
router on which sip alg was enabled so we disable the that and all started
working to normal.

My question is that is there any way to identify such issues and block the
IP or unregistered that EXT automatically via fail2ban or any other way is
possible.

Thanks
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From lists at telefaks.de  Wed Mar 10 15:54:53 2021
From: lists at telefaks.de (Peter Steinbach)
Date: Wed, 10 Mar 2021 16:54:53 +0100
Subject: [Freeswitch-users] BLF under high load
Message-ID: <6bb719f3-a6cf-37c8-fd82-7c5350ee58e5@telefaks.de>

Hello


we have a Freeswitch installation for a multi tenant environment. We
have extensively tested our Freeswitch for all call scenarios and all
worked fine. This is the good news.

However, under higher load (>250 Channels and  >6000 Subscriptions),
presence does not seem to work reliably. I've found some ealier posts on
this issue, which seem to lead to the following 2 solutions

  * use PostgreSQL (we use Mariadb via ODBC, also tested mod_mariadb,
    which made things worse)
  * Move presence handling to Kamailio

As your environment is based on MySQL, we do not want to move to
PostgreSQL without knowing, whether this will improve the situation.

So far we are catching events via ESL or even grepping the network and
correct the status of presence informatons sent to the phone. However we
discover, that even if we send corrected PRESENCE_IN via event socket,
our Freeswitch keeps on sending the wrong BLF information afterwards to
the phone. Means BLFs are blinking or in state red event if the watched
phone is not in a call. So at present we are grepping all presence
informations on the network, crosscheck this with the channels table and
resend the corrected BLF status via event_socket if needed. But still
this is not sufficiant.

So I am now asking the community: What is your approach to overcome this?

  * Do newer Freeswitch versions perform better for this case?(Ours is
    "FreeSWITCH Version 1.10.5-release+git~20200818T185121Z")
  * which work have you done to overcome this?
  * does anybody have an idea how to query all internal presence status
    from Freewitch or maybe how to correct it?

Thank you in advance. Any help is appreciated.

Peter

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From grcamauer at gmail.com  Wed Mar 10 17:07:53 2021
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Wed, 10 Mar 2021 14:07:53 -0300
Subject: [Freeswitch-users] BLF under high load
In-Reply-To: <6bb719f3-a6cf-37c8-fd82-7c5350ee58e5@telefaks.de>
References: <6bb719f3-a6cf-37c8-fd82-7c5350ee58e5@telefaks.de>
Message-ID: <CAPNTnnrkQhLQGXftzooAYNVhYR1XMrf-_xnzZeqW=vZO3dJ2AQ@mail.gmail.com>

>From your description, it sounds like the DB is not keeping up with the
updates FreeSwitch is making.  Have you tried tuning the db?  More memory,
faster/more disks?

Guillermo

On Wed, Mar 10, 2021 at 1:16 PM Peter Steinbach <lists at telefaks.de> wrote:

> Hello
>
>
> we have a Freeswitch installation for a multi tenant environment. We have
> extensively tested our Freeswitch for all call scenarios and all worked
> fine. This is the good news.
>
> However, under higher load (>250 Channels and  >6000 Subscriptions),
> presence does not seem to work reliably. I've found some ealier posts on
> this issue, which seem to lead to the following 2 solutions
>
>    - use PostgreSQL (we use Mariadb via ODBC, also tested mod_mariadb,
>    which made things worse)
>    - Move presence handling to Kamailio
>
> As your environment is based on MySQL, we do not want to move to
> PostgreSQL without knowing, whether this will improve the situation.
>
> So far we are catching events via ESL or even grepping the network and
> correct the status of presence informatons sent to the phone. However we
> discover, that even if we send corrected PRESENCE_IN via event socket, our
> Freeswitch keeps on sending the wrong BLF information afterwards to the
> phone. Means BLFs are blinking or in state red event if the watched phone
> is not in a call. So at present we are grepping all presence informations
> on the network, crosscheck this with the channels table and resend the
> corrected BLF status via event_socket if needed. But still this is not
> sufficiant.
>
> So I am now asking the community: What is your approach to overcome this?
>
>    - Do newer Freeswitch versions perform better for this case?(Ours is
>    "FreeSWITCH Version 1.10.5-release+git~20200818T185121Z")
>    - which work have you done to overcome this?
>    - does anybody have an idea how to query all internal presence status
>    from Freewitch or maybe how to correct it?
>
> Thank you in advance. Any help is appreciated.
>
> Peter
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users 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
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From mircea.huh at gmail.com  Wed Mar 10 17:16:57 2021
From: mircea.huh at gmail.com (Mircea Botoca-Huh)
Date: Wed, 10 Mar 2021 19:16:57 +0200
Subject: [Freeswitch-users] No audio on bridged call
In-Reply-To: <CAGvJV4p-w-oqsKxhUkSMt0VsDHC9Mtkec6KzUtpgo87TU6HAjg@mail.gmail.com>
References: <2BB61A75-937C-4D36-A913-E82C38D65701@hxcore.ol>
 <CAGvJV4p-w-oqsKxhUkSMt0VsDHC9Mtkec6KzUtpgo87TU6HAjg@mail.gmail.com>
Message-ID: <CAL0pLpbYxomvdbveJsxNWVC0zSTR+BL_07Wr3WmQ4LL0SiL9ww@mail.gmail.com>

Thank you for your answer.

I have ext-rtp-ip set to correct IP address.

Only the forwarded call is not working. The incoming to extension has
audio.

BR,

Mircea


mie., 10 mar. 2021, 15:05 Dragos Oancea <dragos at freeswitch.org> a scris:

> Check param ext-rtp-ip in the external SIP profile.
>
>
> On Wed, Mar 10, 2021 at 12:35 AM Mircea Botoca-Huh <mircea.huh at gmail.com>
> wrote:
>
>> Hello,
>>
>>
>>
>> I have freeswitch version 1.10.5 -release-17-25569c1631 64bit installed.
>>
>>
>>
>> I have one SIP provider, no nat.
>>
>>
>>
>> I use a simple dialplan which bridge the received call from SIP provider
>> to a mobile number using the same provider.
>>
>>
>>
>> The call goes through but I have no ringback tone and if I answer there
>> is no audio.
>>
>>
>>
>> If I send the call to one internal extension, it is working fine.
>>
>>
>>
>> The only difference I can see in the logs is that, on the call with no
>> audio, I do not have in logs „Correct ip/port confirmed”.
>>
>>
>>
>> Any suggestion is greatly appreciated.
>>
>>
>>
>> Thank you.
>>
>>
>>
>> Best regards,
>>
>> Mircea.
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users 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
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From david.villasmil.work at gmail.com  Wed Mar 10 17:54:55 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 10 Mar 2021 17:54:55 +0000
Subject: [Freeswitch-users] No audio on bridged call
In-Reply-To: <CAL0pLpbYxomvdbveJsxNWVC0zSTR+BL_07Wr3WmQ4LL0SiL9ww@mail.gmail.com>
References: <2BB61A75-937C-4D36-A913-E82C38D65701@hxcore.ol>
 <CAGvJV4p-w-oqsKxhUkSMt0VsDHC9Mtkec6KzUtpgo87TU6HAjg@mail.gmail.com>
 <CAL0pLpbYxomvdbveJsxNWVC0zSTR+BL_07Wr3WmQ4LL0SiL9ww@mail.gmail.com>
Message-ID: <CAFGRPVqYMf8oAef+Ycqp1H9szkG+yZboG2uyat2bHWBq==WmKw@mail.gmail.com>

you should send the sip trace
Regards,

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


On Wed, Mar 10, 2021 at 5:42 PM Mircea Botoca-Huh <mircea.huh at gmail.com>
wrote:

> Thank you for your answer.
>
> I have ext-rtp-ip set to correct IP address.
>
> Only the forwarded call is not working. The incoming to extension has
> audio.
>
> BR,
>
> Mircea
>
>
> mie., 10 mar. 2021, 15:05 Dragos Oancea <dragos at freeswitch.org> a scris:
>
>> Check param ext-rtp-ip in the external SIP profile.
>>
>>
>> On Wed, Mar 10, 2021 at 12:35 AM Mircea Botoca-Huh <mircea.huh at gmail.com>
>> wrote:
>>
>>> Hello,
>>>
>>>
>>>
>>> I have freeswitch version 1.10.5 -release-17-25569c1631 64bit installed.
>>>
>>>
>>>
>>> I have one SIP provider, no nat.
>>>
>>>
>>>
>>> I use a simple dialplan which bridge the received call from SIP provider
>>> to a mobile number using the same provider.
>>>
>>>
>>>
>>> The call goes through but I have no ringback tone and if I answer there
>>> is no audio.
>>>
>>>
>>>
>>> If I send the call to one internal extension, it is working fine.
>>>
>>>
>>>
>>> The only difference I can see in the logs is that, on the call with no
>>> audio, I do not have in logs „Correct ip/port confirmed”.
>>>
>>>
>>>
>>> Any suggestion is greatly appreciated.
>>>
>>>
>>>
>>> Thank you.
>>>
>>>
>>>
>>> Best regards,
>>>
>>> Mircea.
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users 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
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From rasheed.kalapurackal at gmail.com  Wed Mar 10 18:15:57 2021
From: rasheed.kalapurackal at gmail.com (Rasheed Kalapurackal)
Date: Wed, 10 Mar 2021 23:45:57 +0530
Subject: [Freeswitch-users] Installing Modules
In-Reply-To: <CAEJMVkCDBvBf5X-UwhQFohuUz+OibATS04hOavXAVf7NbnWDPQ@mail.gmail.com>
References: <CAKU4o50sSOxAhW1=kWcqZ0N66Ru=g9Uqf0ti8qTpYYzKs+3vxA@mail.gmail.com>
 <018501d71451$b1765df0$146319d0$@gmail.com>
 <CAKU4o51uqn+8=tMf_h-nYt47noNHS52xiNK1fLcAuLS7yEW1gA@mail.gmail.com>
 <CAEJMVkCDBvBf5X-UwhQFohuUz+OibATS04hOavXAVf7NbnWDPQ@mail.gmail.com>
Message-ID: <CAKU4o50eyqrwRXudedXp36ExxPV2P51UyhtT=Z9+zfnw3Z33cA@mail.gmail.com>

Hi Brian ,

               I am using the module called audio_fork from Drachtio
freeswitch modules. i was able to install a special build for Drachtio
media resource framework which includes the freeswitch with these custom
modules in it and i use this freeswitch build for my use. That resolved my
problem atleast for now.

Thanks
Rasheed

On Tue, Mar 9, 2021 at 10:41 PM Brian West <brian at freeswitch.com> wrote:

> Sounds like you have your own custom module that you're trying to install
> or build, what type of module is it?
>
> Thanks,
> Brina
>
>
> On Tue, Mar 9, 2021 at 10:36 AM Rasheed Kalapurackal <
> rasheed.kalapurackal at gmail.com> wrote:
>
>> Hello John ,
>>
>> I uncommented it from the module.conf.xml , but it seems that the module
>> was not there in the lib directory. .  Since it is a custom module , i
>> could not install it  using apt-get install freeswitch-mod-whatever .  But
>> i have the .so file which i can copy it to lib directory , i will try if
>> that will resolve the issue.
>>
>> Thanks
>> Rasheed
>>
>> On Tue, Mar 9, 2021 at 1:17 AM Bote Man <botelist at gmail.com> wrote:
>>
>>> I am not certain about other dependencies, but you simply uncomment the
>>> line in
>>>
>>> ${conf_dir}/autoload_configs/modules.conf.xml
>>>
>>> to tell FreeSWITCH to load that module.
>>>
>>>
>>>
>>> Hope this helps.
>>>
>>>
>>>
>>>
>>>
>>> ---
>>>
>>> John Boteler
>>>
>>> BnC Group U.S.A.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> *From:* Rasheed Kalapurackal
>>> *Sent:* Saturday, 6 March, 2021 13:19
>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> *Subject:* [Freeswitch-users] Installing Modules
>>>
>>>
>>>
>>> Hello ,
>>>
>>>
>>>
>>>             I completed the installation of Freeswitch on Debian 10 .
>>> Now i need to install an additional module in to the Freeswitch .  Is it
>>> possible to install Modules in to an already deployed Freeswitch instance
>>> ?  Any quick help on this would be greatly appreciated.
>>>
>>>
>>>
>>> Thanks
>>>
>>>
>>>
>>> Abdul Rasheed
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
>
> 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
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From brian at freeswitch.com  Wed Mar 10 18:24:54 2021
From: brian at freeswitch.com (Brian West)
Date: Wed, 10 Mar 2021 12:24:54 -0600
Subject: [Freeswitch-users] No audio on bridged call
In-Reply-To: <CAL0pLpbYxomvdbveJsxNWVC0zSTR+BL_07Wr3WmQ4LL0SiL9ww@mail.gmail.com>
References: <2BB61A75-937C-4D36-A913-E82C38D65701@hxcore.ol>
 <CAGvJV4p-w-oqsKxhUkSMt0VsDHC9Mtkec6KzUtpgo87TU6HAjg@mail.gmail.com>
 <CAL0pLpbYxomvdbveJsxNWVC0zSTR+BL_07Wr3WmQ4LL0SiL9ww@mail.gmail.com>
Message-ID: <CAEJMVkCcev4v++H-pGmHWqb_=1YuGSR2SZX7iLwGBgSCvgZp9g@mail.gmail.com>

Are you saying a hair pin call thru your FreeSWITCH has no audio (i.e. call
comes in and goes right back out)?

for giggles try this at fs_cli:

global_setvar execute_on_answer01='playback silence_stream://500'

What you describe sounds like both ends are in a standoff waiting on the
other to send media.

/b



On Wed, Mar 10, 2021 at 11:50 AM Mircea Botoca-Huh <mircea.huh at gmail.com>
wrote:

> Thank you for your answer.
>
> I have ext-rtp-ip set to correct IP address.
>
> Only the forwarded call is not working. The incoming to extension has
> audio.
>
> BR,
>
> Mircea
>
>
> mie., 10 mar. 2021, 15:05 Dragos Oancea <dragos at freeswitch.org> a scris:
>
>> Check param ext-rtp-ip in the external SIP profile.
>>
>>
>> On Wed, Mar 10, 2021 at 12:35 AM Mircea Botoca-Huh <mircea.huh at gmail.com>
>> wrote:
>>
>>> Hello,
>>>
>>>
>>>
>>> I have freeswitch version 1.10.5 -release-17-25569c1631 64bit installed.
>>>
>>>
>>>
>>> I have one SIP provider, no nat.
>>>
>>>
>>>
>>> I use a simple dialplan which bridge the received call from SIP provider
>>> to a mobile number using the same provider.
>>>
>>>
>>>
>>> The call goes through but I have no ringback tone and if I answer there
>>> is no audio.
>>>
>>>
>>>
>>> If I send the call to one internal extension, it is working fine.
>>>
>>>
>>>
>>> The only difference I can see in the logs is that, on the call with no
>>> audio, I do not have in logs „Correct ip/port confirmed”.
>>>
>>>
>>>
>>> Any suggestion is greatly appreciated.
>>>
>>>
>>>
>>> Thank you.
>>>
>>>
>>>
>>> Best regards,
>>>
>>> Mircea.
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users 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



-- 

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>
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From mircea.huh at gmail.com  Wed Mar 10 21:22:04 2021
From: mircea.huh at gmail.com (Mircea Botoca-Huh)
Date: Wed, 10 Mar 2021 23:22:04 +0200
Subject: [Freeswitch-users] No audio on bridged call
In-Reply-To: <CAEJMVkCcev4v++H-pGmHWqb_=1YuGSR2SZX7iLwGBgSCvgZp9g@mail.gmail.com>
References: <2BB61A75-937C-4D36-A913-E82C38D65701@hxcore.ol>
 <CAGvJV4p-w-oqsKxhUkSMt0VsDHC9Mtkec6KzUtpgo87TU6HAjg@mail.gmail.com>
 <CAL0pLpbYxomvdbveJsxNWVC0zSTR+BL_07Wr3WmQ4LL0SiL9ww@mail.gmail.com>,
 <CAEJMVkCcev4v++H-pGmHWqb_=1YuGSR2SZX7iLwGBgSCvgZp9g@mail.gmail.com>
Message-ID: <75A3C23C-3DF5-47C7-A8E8-C0DC511E7301@hxcore.ol>

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From gregor at infomedia.si  Wed Mar 10 21:28:14 2021
From: gregor at infomedia.si (Gregor Nanger)
Date: Wed, 10 Mar 2021 22:28:14 +0100
Subject: [Freeswitch-users] No audio on bridged call
In-Reply-To: <75A3C23C-3DF5-47C7-A8E8-C0DC511E7301@hxcore.ol>
References: <2BB61A75-937C-4D36-A913-E82C38D65701@hxcore.ol>
 <CAGvJV4p-w-oqsKxhUkSMt0VsDHC9Mtkec6KzUtpgo87TU6HAjg@mail.gmail.com>
 <CAL0pLpbYxomvdbveJsxNWVC0zSTR+BL_07Wr3WmQ4LL0SiL9ww@mail.gmail.com>
 <CAEJMVkCcev4v++H-pGmHWqb_=1YuGSR2SZX7iLwGBgSCvgZp9g@mail.gmail.com>
 <75A3C23C-3DF5-47C7-A8E8-C0DC511E7301@hxcore.ol>
Message-ID: <CAC7ttFH6VhvgwS0uTS_NxRhhjucn2XgEoyAQWirPhWNespx28A@mail.gmail.com>

Good giggles, Brian 👍

On Wed, 10 Mar 2021 at 22:22, Mircea Botoca-Huh <mircea.huh at gmail.com>
wrote:

> Hello Brian,
>
>
>
> You were absolutely right.
>
>
>
> I followed your advise and indeed with this variable set the call has
> audio.
>
>
>
> Thank you all very much.
>
>
>
> Best regards,
>
> Mircea.
>
>
>
> *From: *Brian West <brian at freeswitch.com>
> *Sent: *miercuri, 10 martie 2021 20:25
> *To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject: *Re: [Freeswitch-users] No audio on bridged call
>
>
>
> Are you saying a hair pin call thru your FreeSWITCH has no audio (i.e.
> call comes in and goes right back out)?
>
>
>
> for giggles try this at fs_cli:
>
>
>
> global_setvar execute_on_answer01='playback silence_stream://500'
>
>
>
> What you describe sounds like both ends are in a standoff waiting on the
> other to send media.
>
>
>
> /b
>
>
>
>
>
>
>
> On Wed, Mar 10, 2021 at 11:50 AM Mircea Botoca-Huh <mircea.huh at gmail.com>
> wrote:
>
> Thank you for your answer.
>
>
>
> I have ext-rtp-ip set to correct IP address.
>
>
>
> Only the forwarded call is not working. The incoming to extension has
> audio.
>
>
>
> BR,
>
>
>
> Mircea
>
>
>
> mie., 10 mar. 2021, 15:05 Dragos Oancea <dragos at freeswitch.org> a scris:
>
> Check param ext-rtp-ip in the external SIP profile.
>
>
>
>
>
> On Wed, Mar 10, 2021 at 12:35 AM Mircea Botoca-Huh <mircea.huh at gmail.com>
> wrote:
>
> Hello,
>
>
>
> I have freeswitch version 1.10.5 -release-17-25569c1631 64bit installed.
>
>
>
> I have one SIP provider, no nat.
>
>
>
> I use a simple dialplan which bridge the received call from SIP provider
> to a mobile number using the same provider.
>
>
>
> The call goes through but I have no ringback tone and if I answer there is
> no audio.
>
>
>
> If I send the call to one internal extension, it is working fine.
>
>
>
> The only difference I can see in the logs is that, on the call with no
> audio, I do not have in logs „Correct ip/port confirmed”.
>
>
>
> Any suggestion is greatly appreciated.
>
>
>
> Thank you.
>
>
>
> Best regards,
>
> Mircea.
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users 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
>
>
>
>
> --
>
>
>
> 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/>
>
> <https://www.facebook.com/freeswitch> <https://twitter.com/freeswitch>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From brian at freeswitch.com  Wed Mar 10 21:44:26 2021
From: brian at freeswitch.com (Brian West)
Date: Wed, 10 Mar 2021 15:44:26 -0600
Subject: [Freeswitch-users] Odd RTP skew behavior
In-Reply-To: <07d301d7145c$6bc86180$43592480$@voicemeup.com>
References: <14b301d6f650$ffa9c120$fefd4360$@voicemeup.com>
 <CAEJMVkCmCK5jYemysSUmmNH+u=5kU92QVi8aW+W5LqDYtDj17A@mail.gmail.com>
 <07d301d7145c$6bc86180$43592480$@voicemeup.com>
Message-ID: <CAEJMVkAPpP3tMa3nDsoomUN-mgVogvW7fAn_-C9ui=qH8djWtA@mail.gmail.com>

Well nobody should be on 1.6.x or 1.8.x, there is no telling what could be
going on then.

On Tue, Mar 9, 2021 at 5:02 PM Marc Bernard <marcb at voicemeup.com> wrote:

> Sorry for late reply.
>
> I think this is mostly happen with both 1.6.20 and 1.10.5 in the path
>
>
> --
>
>
> From: Brian West <brian at freeswitch.com>
> Sent: Monday, February 1, 2021 11:51
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Odd RTP skew behavior
>
> You didn't mention the revision of freeswitch you're using?
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users 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>
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From lists at telefaks.de  Thu Mar 11 12:08:11 2021
From: lists at telefaks.de (Peter Steinbach)
Date: Thu, 11 Mar 2021 13:08:11 +0100
Subject: [Freeswitch-users] BLF under high load
In-Reply-To: <CAPNTnnrkQhLQGXftzooAYNVhYR1XMrf-_xnzZeqW=vZO3dJ2AQ@mail.gmail.com>
References: <6bb719f3-a6cf-37c8-fd82-7c5350ee58e5@telefaks.de>
 <CAPNTnnrkQhLQGXftzooAYNVhYR1XMrf-_xnzZeqW=vZO3dJ2AQ@mail.gmail.com>
Message-ID: <5a7ab417-672c-26c7-0545-3e866f0aff50@telefaks.de>

We made some investigations on our Production database. We set mysql
slow log to 0.02 sec and we did not discover any long running qeuries
concerning presence.

The command

    select * from INFORMATION_SCHEMA.PROCESSLIST  where command <> 'Sleep';

shows, that presence queries run between 0.1 and 1 milliseconds. So I do
not really feel, that the Mysql database is the bottleneck here. MySQL
show about 10-15%CPU on an 8 core machine. Freeswitch runs at about 50%
CPU on a different 8 core machine.

Nevertheless we have severe problems with presence. I have e.g. a Snom
phone and a Yealink phone on my desk with a BLF for a 3rd phone, which
is not in a call at present. BLF on Snom is "on" and BLF on Yealink is
"blinking". Both should be "off". This is very strange to me. How can
this happen? BTW when this system was under no load, before we migrated
our customers, everything worked fine.

So any help is very much appreciated here.

Another question is still:

  * does anybody have an idea how to query all internal presence status
    from Freewitch or maybe how to correct it?
      o e.g. "sofia_presence_data list number at domain" returns e.g.

status|rpid|user_agent|network_ip|network_port
unknown|unknown|N510 IP PRO/42.250.00.000.000|xx.xxx.xxx.xx|21549
unknown|unknown|Yealink SIP-T41S 66.83.0.35 805EC025FDA9|xx.xx.xxx.xxx|20901


      * and "sofia_presence_data status number at domain" always returns

unknownunknown

      * Database table "sip_subscriptions" however is fine


Best regards

Peter


Am 10.03.21 um 18:07 schrieb Guillermo Ruiz Camauer:
> From your description, it sounds like the DB is not keeping up with
> the updates FreeSwitch is making.  Have you tried tuning the db?  More
> memory, faster/more disks?
>
> Guillermo
>
> On Wed, Mar 10, 2021 at 1:16 PM Peter Steinbach <lists at telefaks.de
> <mailto:lists at telefaks.de>> wrote:
>
>     Hello
>
>
>     we have a Freeswitch installation for a multi tenant environment.
>     We have extensively tested our Freeswitch for all call scenarios
>     and all worked fine. This is the good news.
>
>     However, under higher load (>250 Channels and  >6000
>     Subscriptions), presence does not seem to work reliably. I've
>     found some ealier posts on this issue, which seem to lead to the
>     following 2 solutions
>
>       * use PostgreSQL (we use Mariadb via ODBC, also tested
>         mod_mariadb, which made things worse)
>       * Move presence handling to Kamailio
>
>     As your environment is based on MySQL, we do not want to move to
>     PostgreSQL without knowing, whether this will improve the situation.
>
>     So far we are catching events via ESL or even grepping the network
>     and correct the status of presence informatons sent to the phone.
>     However we discover, that even if we send corrected PRESENCE_IN
>     via event socket, our Freeswitch keeps on sending the wrong BLF
>     information afterwards to the phone. Means BLFs are blinking or in
>     state red event if the watched phone is not in a call. So at
>     present we are grepping all presence informations on the network,
>     crosscheck this with the channels table and resend the corrected
>     BLF status via event_socket if needed. But still this is not
>     sufficiant.
>
>     So I am now asking the community: What is your approach to
>     overcome this?
>
>       * Do newer Freeswitch versions perform better for this
>         case?(Ours is "FreeSWITCH Version
>         1.10.5-release+git~20200818T185121Z")
>       * which work have you done to overcome this?
>       * does anybody have an idea how to query all internal presence
>         status from Freewitch or maybe how to correct it?
>
>     Thank you in advance. Any help is appreciated.
>
>     Peter
>
>     _________________________________________________________________________
>
>     The FreeSWITCH project is sponsored by SignalWire
>     https://signalwire.com
>     Enhance your FreeSWITCH install with disruptive priced SMS and
>     PSTN services.
>     Build your next product 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
>
>
>
> -- 
> Guillermo Ruiz Camauer
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From lists at telefaks.de  Thu Mar 11 13:09:34 2021
From: lists at telefaks.de (Peter Steinbach)
Date: Thu, 11 Mar 2021 14:09:34 +0100
Subject: [Freeswitch-users] BLF under high load
In-Reply-To: <5a7ab417-672c-26c7-0545-3e866f0aff50@telefaks.de>
References: <6bb719f3-a6cf-37c8-fd82-7c5350ee58e5@telefaks.de>
 <CAPNTnnrkQhLQGXftzooAYNVhYR1XMrf-_xnzZeqW=vZO3dJ2AQ@mail.gmail.com>
 <5a7ab417-672c-26c7-0545-3e866f0aff50@telefaks.de>
Message-ID: <e7449327-41f5-409d-6657-5df4609eab4f@telefaks.de>

While debugging presence, I found a lot of the following lines on the
freeswitch console

2021-03-11 13:26:34.496493 [CRIT] sofia_presence.c:1377 CHECK SQL:
9886 at customer.mydomain.net [select state,status,rpid,presence_id,uuid
from sip_dialogs where call_info_state != 'seized' and
hostname='fs02.mydomain.net' and profile_name='internal' and
((sip_from_user='9886' and sip_from_host='customer.mydomain.net') or
presence_id='9886 at customer.mydomain.net') order by rcd desc]
2021-03-11 13:26:34.496493 [ERR] sofia_presence.c:1435 PRES SQL update
sip_subscriptions set version=version+1 where
hostname='fs02.mydomain.net' and profile_name='internal' and
sip_subscriptions.event != 'line-seize' and
sip_subscriptions.proto='sip' and (event='presence' or event='dialog')
and sub_to_user='9886' and (sub_to_host='customer.mydomain.net' or
sub_to_host='xxx.xx.x.xxx' or sub_to_host='N/A' or presence_hosts like
'%customer.mydomain.net%') and (sip_subscriptions.profile_name =
'internal' or presence_hosts like '%customer.mydomain.net%')

  * The first query returns 0 entries, when I run it manually, line is
    marked as [CRIT]
      o my impression is that this query is always empty because _all_
        values of sip_dialogs.call_info_state are empty, allthough we
        have about 100 entries in sip_dialogs table. Running 100k
        continuous queries against this table, I could not find any
        other value than '' for sip_dialogs.call_info_state
  * The second query updates 7 entries, when I run it manually, but line
    is marked as [ERR]


Maybe this info creates some new ideas.

Best regards
Peter


Am 11.03.21 um 13:08 schrieb Peter Steinbach:
>
> We made some investigations on our Production database. We set mysql
> slow log to 0.02 sec and we did not discover any long running qeuries
> concerning presence.
>
> The command
>
>     select * from INFORMATION_SCHEMA.PROCESSLIST  where command <>
> 'Sleep';
>
> shows, that presence queries run between 0.1 and 1 milliseconds. So I
> do not really feel, that the Mysql database is the bottleneck here.
> MySQL show about 10-15%CPU on an 8 core machine. Freeswitch runs at
> about 50% CPU on a different 8 core machine.
>
> Nevertheless we have severe problems with presence. I have e.g. a Snom
> phone and a Yealink phone on my desk with a BLF for a 3rd phone, which
> is not in a call at present. BLF on Snom is "on" and BLF on Yealink is
> "blinking". Both should be "off". This is very strange to me. How can
> this happen? BTW when this system was under no load, before we
> migrated our customers, everything worked fine.
>
> So any help is very much appreciated here.
>
> Another question is still:
>
>   * does anybody have an idea how to query all internal presence
>     status from Freewitch or maybe how to correct it?
>       o e.g. "sofia_presence_data list number at domain" returns e.g.
>
> status|rpid|user_agent|network_ip|network_port
> unknown|unknown|N510 IP PRO/42.250.00.000.000|xx.xxx.xxx.xx|21549
> unknown|unknown|Yealink SIP-T41S 66.83.0.35
> 805EC025FDA9|xx.xx.xxx.xxx|20901
>
>
>       * and "sofia_presence_data status number at domain" always returns
>
> unknownunknown
>
>       * Database table "sip_subscriptions" however is fine
>
>
> Best regards
>
> Peter
>
>
> Am 10.03.21 um 18:07 schrieb Guillermo Ruiz Camauer:
>> From your description, it sounds like the DB is not keeping up with
>> the updates FreeSwitch is making.  Have you tried tuning the db? 
>> More memory, faster/more disks?
>>
>> Guillermo
>>
>> On Wed, Mar 10, 2021 at 1:16 PM Peter Steinbach <lists at telefaks.de
>> <mailto:lists at telefaks.de>> wrote:
>>
>>     Hello
>>
>>
>>     we have a Freeswitch installation for a multi tenant environment.
>>     We have extensively tested our Freeswitch for all call scenarios
>>     and all worked fine. This is the good news.
>>
>>     However, under higher load (>250 Channels and  >6000
>>     Subscriptions), presence does not seem to work reliably. I've
>>     found some ealier posts on this issue, which seem to lead to the
>>     following 2 solutions
>>
>>       * use PostgreSQL (we use Mariadb via ODBC, also tested
>>         mod_mariadb, which made things worse)
>>       * Move presence handling to Kamailio
>>
>>     As your environment is based on MySQL, we do not want to move to
>>     PostgreSQL without knowing, whether this will improve the situation.
>>
>>     So far we are catching events via ESL or even grepping the
>>     network and correct the status of presence informatons sent to
>>     the phone. However we discover, that even if we send corrected
>>     PRESENCE_IN via event socket, our Freeswitch keeps on sending the
>>     wrong BLF information afterwards to the phone. Means BLFs are
>>     blinking or in state red event if the watched phone is not in a
>>     call. So at present we are grepping all presence informations on
>>     the network, crosscheck this with the channels table and resend
>>     the corrected BLF status via event_socket if needed. But still
>>     this is not sufficiant.
>>
>>     So I am now asking the community: What is your approach to
>>     overcome this?
>>
>>       * Do newer Freeswitch versions perform better for this
>>         case?(Ours is "FreeSWITCH Version
>>         1.10.5-release+git~20200818T185121Z")
>>       * which work have you done to overcome this?
>>       * does anybody have an idea how to query all internal presence
>>         status from Freewitch or maybe how to correct it?
>>
>>     Thank you in advance. Any help is appreciated.
>>
>>     Peter
>>
>>     _________________________________________________________________________
>>
>>     The FreeSWITCH project is sponsored by SignalWire
>>     https://signalwire.com
>>     Enhance your FreeSWITCH install with disruptive priced SMS and
>>     PSTN services.
>>     Build your next product 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
>>
>>
>>
>> -- 
>> Guillermo Ruiz Camauer
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com

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From brian at freeswitch.com  Thu Mar 11 14:09:14 2021
From: brian at freeswitch.com (Brian West)
Date: Thu, 11 Mar 2021 08:09:14 -0600
Subject: [Freeswitch-users] No audio on bridged call
In-Reply-To: <75A3C23C-3DF5-47C7-A8E8-C0DC511E7301@hxcore.ol>
References: <2BB61A75-937C-4D36-A913-E82C38D65701@hxcore.ol>
 <CAGvJV4p-w-oqsKxhUkSMt0VsDHC9Mtkec6KzUtpgo87TU6HAjg@mail.gmail.com>
 <CAL0pLpbYxomvdbveJsxNWVC0zSTR+BL_07Wr3WmQ4LL0SiL9ww@mail.gmail.com>
 <CAEJMVkCcev4v++H-pGmHWqb_=1YuGSR2SZX7iLwGBgSCvgZp9g@mail.gmail.com>
 <75A3C23C-3DF5-47C7-A8E8-C0DC511E7301@hxcore.ol>
Message-ID: <CAEJMVkBQ4XgxfG2e0LOL88M2FvNBc8RD2vsjHTtbHz=10fcoZw@mail.gmail.com>

I think the feature you're hitting in the upstream switch is called 'RTP
Clamping'.  They won't start sending media till you do, it should only be
doing this on inbound or outbound calls NOT BOTH.  What I gave you was a
trick to make FreeSWITCH inject 500ms of silence, you can lower this
number, you can also just set it inside the {} on the outbound originate
too.

/b


On Wed, Mar 10, 2021 at 3:49 PM Mircea Botoca-Huh <mircea.huh at gmail.com>
wrote:

> Hello Brian,
>
>
>
> You were absolutely right.
>
>
>
> I followed your advise and indeed with this variable set the call has
> audio.
>
>
>
> Thank you all very much.
>
>
>
> Best regards,
>
> Mircea.
>
>
>
> *From: *Brian West <brian at freeswitch.com>
> *Sent: *miercuri, 10 martie 2021 20:25
> *To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject: *Re: [Freeswitch-users] No audio on bridged call
>
>
>
> Are you saying a hair pin call thru your FreeSWITCH has no audio (i.e.
> call comes in and goes right back out)?
>
>
>
> for giggles try this at fs_cli:
>
>
>
> global_setvar execute_on_answer01='playback silence_stream://500'
>
>
>
> What you describe sounds like both ends are in a standoff waiting on the
> other to send media.
>
>
>
> /b
>
>
>
>
>
>
>
> On Wed, Mar 10, 2021 at 11:50 AM Mircea Botoca-Huh <mircea.huh at gmail.com>
> wrote:
>
> Thank you for your answer.
>
>
>
> I have ext-rtp-ip set to correct IP address.
>
>
>
> Only the forwarded call is not working. The incoming to extension has
> audio.
>
>
>
> BR,
>
>
>
> Mircea
>
>
>
> mie., 10 mar. 2021, 15:05 Dragos Oancea <dragos at freeswitch.org> a scris:
>
> Check param ext-rtp-ip in the external SIP profile.
>
>
>
>
>
> On Wed, Mar 10, 2021 at 12:35 AM Mircea Botoca-Huh <mircea.huh at gmail.com>
> wrote:
>
> Hello,
>
>
>
> I have freeswitch version 1.10.5 -release-17-25569c1631 64bit installed.
>
>
>
> I have one SIP provider, no nat.
>
>
>
> I use a simple dialplan which bridge the received call from SIP provider
> to a mobile number using the same provider.
>
>
>
> The call goes through but I have no ringback tone and if I answer there is
> no audio.
>
>
>
> If I send the call to one internal extension, it is working fine.
>
>
>
> The only difference I can see in the logs is that, on the call with no
> audio, I do not have in logs „Correct ip/port confirmed”.
>
>
>
> Any suggestion is greatly appreciated.
>
>
>
> Thank you.
>
>
>
> Best regards,
>
> Mircea.
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users 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
>
>
>
>
> --
>
>
>
> 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/>
>
> <https://www.facebook.com/freeswitch> <https://twitter.com/freeswitch>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

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>
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From lexxua at gmail.com  Thu Mar 11 19:16:28 2021
From: lexxua at gmail.com (Volodymyr Fedorov)
Date: Thu, 11 Mar 2021 20:16:28 +0100
Subject: [Freeswitch-users] Pass sendonly from b-leg to a-leg
Message-ID: <CAO_GGAqWjLbjKmx1Yc-=RAi0u43O8pet7WtmR=9abBb6841iNQ@mail.gmail.com>

Hello,
I have following scenario : A (GW) -> Freeswitch -> B(Phone)
When B-puts call on-hold and send to Freeswtich sendonly attribute,
Freeswitch starts to play moh to A-party without re-invite with
sendonly.
Is there a way to send to A (GW) sendonly attribute using dialplan variable?
Thanks for advice!

-- 
Best regards,
Volodymyr


From brian at freeswitch.com  Thu Mar 11 19:56:54 2021
From: brian at freeswitch.com (Brian West)
Date: Thu, 11 Mar 2021 13:56:54 -0600
Subject: [Freeswitch-users] Pass sendonly from b-leg to a-leg
In-Reply-To: <CAO_GGAqWjLbjKmx1Yc-=RAi0u43O8pet7WtmR=9abBb6841iNQ@mail.gmail.com>
References: <CAO_GGAqWjLbjKmx1Yc-=RAi0u43O8pet7WtmR=9abBb6841iNQ@mail.gmail.com>
Message-ID: <CAEJMVkD6inr1Wg08PNfUB3gVHssJGFj2x=GcAd5HpUgEp=C=dw@mail.gmail.com>

FreeSWITCH IS NOT A PROXY, it's a B2BUA

/b


On Thu, Mar 11, 2021 at 1:44 PM Volodymyr Fedorov <lexxua at gmail.com> wrote:

> Hello,
> I have following scenario : A (GW) -> Freeswitch -> B(Phone)
> When B-puts call on-hold and send to Freeswtich sendonly attribute,
> Freeswitch starts to play moh to A-party without re-invite with
> sendonly.
> Is there a way to send to A (GW) sendonly attribute using dialplan
> variable?
> Thanks for advice!
>
> --
> Best regards,
> Volodymyr
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users 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>
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From joe at expert.net  Fri Mar 12 04:14:19 2021
From: joe at expert.net (Joseph Barrero)
Date: Thu, 11 Mar 2021 22:14:19 -0600
Subject: [Freeswitch-users] Un-Subscribing on 481
Message-ID: <CAGAa33svGkW6=-p07vQSRuzxbqHi+a0fJ1bK_XrrvZ5gFaYB4g@mail.gmail.com>

Hi, everyone.

I recently noticed that Freeswitch continues sending NOTIFY messages even
after receiving a 481 from my proxy after not being able to deliver the
messages to the UA.  This seems to happen when the UA disconnects from the
Internet briefly and registers again before the previous subscription
expires.

Below is the code I found in sofia.c that seems to handle the 481 response.

if (status == 481 && sip && !sip->sip_retry_after && sip->sip_call_id &&
(!sofia_private || !sofia_private->is_call)) {
        char *sql;

        sql = switch_mprintf("delete from sip_subscriptions where
call_id='%q'", sip->sip_call_id->i_id);
        switch_assert(sql != NULL);
        sofia_glue_execute_sql(profile, &sql, SWITCH_TRUE);
        nua_handle_destroy(nh);
}


Assuming that the variable sofia_private->is_call evaluates to true when
NOTIFY messages are sent during a call, it seems to curtail the removal of
the SUBSCRIPTION even after receiving a 481 from the proxy.

Is there a drawback to removing a subscription on a 481 response during a
call?  Would it make sense to simply remove the check if there is a call in
place? Is checking ofia_private->is_call there for another purpose?

Thanks,
Joe Barrero
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From lexxua at gmail.com  Fri Mar 12 06:34:01 2021
From: lexxua at gmail.com (Volodymyr Fedorov)
Date: Fri, 12 Mar 2021 07:34:01 +0100
Subject: [Freeswitch-users] Pass sendonly from b-leg to a-leg
In-Reply-To: <CAEJMVkD6inr1Wg08PNfUB3gVHssJGFj2x=GcAd5HpUgEp=C=dw@mail.gmail.com>
References: <CAO_GGAqWjLbjKmx1Yc-=RAi0u43O8pet7WtmR=9abBb6841iNQ@mail.gmail.com>
 <CAEJMVkD6inr1Wg08PNfUB3gVHssJGFj2x=GcAd5HpUgEp=C=dw@mail.gmail.com>
Message-ID: <CAO_GGApzs6m6cCeQwdQ8Fqoj7jrZqHF41R4UhwyEJQWQJ7W0+w@mail.gmail.com>

Hi Brian,
That is clear, but in proxy media mode freeswitch passes sendonly from b to
a leg.
So I was wondering about  other options.
Best regards,
Vova

чт, 11 мар. 2021 г., 21:19 Brian West <brian at freeswitch.com>:

> FreeSWITCH IS NOT A PROXY, it's a B2BUA
>
> /b
>
>
> On Thu, Mar 11, 2021 at 1:44 PM Volodymyr Fedorov <lexxua at gmail.com>
> wrote:
>
>> Hello,
>> I have following scenario : A (GW) -> Freeswitch -> B(Phone)
>> When B-puts call on-hold and send to Freeswtich sendonly attribute,
>> Freeswitch starts to play moh to A-party without re-invite with
>> sendonly.
>> Is there a way to send to A (GW) sendonly attribute using dialplan
>> variable?
>> Thanks for advice!
>>
>> --
>> Best regards,
>> Volodymyr
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users 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
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From mircea.huh at gmail.com  Fri Mar 12 12:06:27 2021
From: mircea.huh at gmail.com (Mircea Botoca-Huh)
Date: Fri, 12 Mar 2021 14:06:27 +0200
Subject: [Freeswitch-users] No audio on bridged call
In-Reply-To: <CAEJMVkBQ4XgxfG2e0LOL88M2FvNBc8RD2vsjHTtbHz=10fcoZw@mail.gmail.com>
References: <2BB61A75-937C-4D36-A913-E82C38D65701@hxcore.ol>
 <CAGvJV4p-w-oqsKxhUkSMt0VsDHC9Mtkec6KzUtpgo87TU6HAjg@mail.gmail.com>
 <CAL0pLpbYxomvdbveJsxNWVC0zSTR+BL_07Wr3WmQ4LL0SiL9ww@mail.gmail.com>
 <CAEJMVkCcev4v++H-pGmHWqb_=1YuGSR2SZX7iLwGBgSCvgZp9g@mail.gmail.com>
 <75A3C23C-3DF5-47C7-A8E8-C0DC511E7301@hxcore.ol>
 <CAEJMVkBQ4XgxfG2e0LOL88M2FvNBc8RD2vsjHTtbHz=10fcoZw@mail.gmail.com>
Message-ID: <CAL0pLpb_ECu3F2BJVNWwBgF0YJuajL+Oj08uYnRf01m=t2Rk1w@mail.gmail.com>

Indeed Brian, this is what I did. I set it on outbound originate and it is
working perfectly.

Thank you for the tip. You are the best.

BR,
Mircea.

joi, 11 mar. 2021, 16:10 Brian West <brian at freeswitch.com> a scris:

> I think the feature you're hitting in the upstream switch is called 'RTP
> Clamping'.  They won't start sending media till you do, it should only be
> doing this on inbound or outbound calls NOT BOTH.  What I gave you was a
> trick to make FreeSWITCH inject 500ms of silence, you can lower this
> number, you can also just set it inside the {} on the outbound originate
> too.
>
> /b
>
>
> On Wed, Mar 10, 2021 at 3:49 PM Mircea Botoca-Huh <mircea.huh at gmail.com>
> wrote:
>
>> Hello Brian,
>>
>>
>>
>> You were absolutely right.
>>
>>
>>
>> I followed your advise and indeed with this variable set the call has
>> audio.
>>
>>
>>
>> Thank you all very much.
>>
>>
>>
>> Best regards,
>>
>> Mircea.
>>
>>
>>
>> *From: *Brian West <brian at freeswitch.com>
>> *Sent: *miercuri, 10 martie 2021 20:25
>> *To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject: *Re: [Freeswitch-users] No audio on bridged call
>>
>>
>>
>> Are you saying a hair pin call thru your FreeSWITCH has no audio (i.e.
>> call comes in and goes right back out)?
>>
>>
>>
>> for giggles try this at fs_cli:
>>
>>
>>
>> global_setvar execute_on_answer01='playback silence_stream://500'
>>
>>
>>
>> What you describe sounds like both ends are in a standoff waiting on the
>> other to send media.
>>
>>
>>
>> /b
>>
>>
>>
>>
>>
>>
>>
>> On Wed, Mar 10, 2021 at 11:50 AM Mircea Botoca-Huh <mircea.huh at gmail.com>
>> wrote:
>>
>> Thank you for your answer.
>>
>>
>>
>> I have ext-rtp-ip set to correct IP address.
>>
>>
>>
>> Only the forwarded call is not working. The incoming to extension has
>> audio.
>>
>>
>>
>> BR,
>>
>>
>>
>> Mircea
>>
>>
>>
>> mie., 10 mar. 2021, 15:05 Dragos Oancea <dragos at freeswitch.org> a scris:
>>
>> Check param ext-rtp-ip in the external SIP profile.
>>
>>
>>
>>
>>
>> On Wed, Mar 10, 2021 at 12:35 AM Mircea Botoca-Huh <mircea.huh at gmail.com>
>> wrote:
>>
>> Hello,
>>
>>
>>
>> I have freeswitch version 1.10.5 -release-17-25569c1631 64bit installed.
>>
>>
>>
>> I have one SIP provider, no nat.
>>
>>
>>
>> I use a simple dialplan which bridge the received call from SIP provider
>> to a mobile number using the same provider.
>>
>>
>>
>> The call goes through but I have no ringback tone and if I answer there
>> is no audio.
>>
>>
>>
>> If I send the call to one internal extension, it is working fine.
>>
>>
>>
>> The only difference I can see in the logs is that, on the call with no
>> audio, I do not have in logs „Correct ip/port confirmed”.
>>
>>
>>
>> Any suggestion is greatly appreciated.
>>
>>
>>
>> Thank you.
>>
>>
>>
>> Best regards,
>>
>> Mircea.
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users 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
>>
>>
>>
>>
>> --
>>
>>
>>
>> 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/>
>>
>> <https://www.facebook.com/freeswitch> <https://twitter.com/freeswitch>
>>
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
>
> 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
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From lewis.bergman at gmail.com  Fri Mar 12 12:14:23 2021
From: lewis.bergman at gmail.com (Lewis Bergman)
Date: Fri, 12 Mar 2021 06:14:23 -0600
Subject: [Freeswitch-users] using mod_unimrcp to transcribe voicemails
In-Reply-To: <CAE23trtEEiXZ-MurMTNVdLVuhZBR=_7nybTcawOcRL9haS16Ng@mail.gmail.com>
References: <CAE23trtEEiXZ-MurMTNVdLVuhZBR=_7nybTcawOcRL9haS16Ng@mail.gmail.com>
Message-ID: <CAD2CnaS6fDpQM2=aa1c7P=gxB_q_ieDeC67EPxLAEGNwSoyL4A@mail.gmail.com>

I couldn't find anything. Is there an archive of presentations I am blind
to?

On Tue, Mar 9, 2021 at 11:53 PM David P <davidswalkabout at gmail.com> wrote:

> One of the ClueCon 2020 talks was by someone who uses mod_unimrcp and has
> a github project showing its FS integration, iirc.
>
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From kaiduanx at yahoo.ca  Fri Mar 12 14:29:49 2021
From: kaiduanx at yahoo.ca (kaiduan xie)
Date: Fri, 12 Mar 2021 14:29:49 +0000 (UTC)
Subject: [Freeswitch-users] Odd RTP skew behavior
In-Reply-To: <CAEJMVkAPpP3tMa3nDsoomUN-mgVogvW7fAn_-C9ui=qH8djWtA@mail.gmail.com>
References: <14b301d6f650$ffa9c120$fefd4360$@voicemeup.com>
 <CAEJMVkCmCK5jYemysSUmmNH+u=5kU92QVi8aW+W5LqDYtDj17A@mail.gmail.com>
 <07d301d7145c$6bc86180$43592480$@voicemeup.com>
 <CAEJMVkAPpP3tMa3nDsoomUN-mgVogvW7fAn_-C9ui=qH8djWtA@mail.gmail.com>
Message-ID: <58639871.115304.1615559389514@mail.yahoo.com>

 Brian,
A lot of FS in production are 1.6x or 1.8x, are there any particular reasons why you say above?
Thanks,
/Kaiduan
    On Wednesday, March 10, 2021, 04:45:12 p.m. EST, Brian West <brian at freeswitch.com> wrote:  
 
 Well nobody should be on 1.6.x or 1.8.x, there is no telling what could be going on then.
On Tue, Mar 9, 2021 at 5:02 PM Marc Bernard <marcb at voicemeup.com> wrote:

Sorry for late reply.

I think this is mostly happen with both 1.6.20 and 1.10.5 in the path


--


From: Brian West <brian at freeswitch.com> 
Sent: Monday, February 1, 2021 11:51
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Odd RTP skew behavior

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



_________________________________________________________________________

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

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

Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com


-- 


Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com 

FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045

Email: brian at freeswitch.com

Mobile: 918-424-9378

Website: https://www.FreeSWITCH.com

 

_________________________________________________________________________

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

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

Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com  
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From rawat.anshuman at gmail.com  Fri Mar 12 14:54:29 2021
From: rawat.anshuman at gmail.com (Anshuman Rawat)
Date: Fri, 12 Mar 2021 09:54:29 -0500
Subject: [Freeswitch-users] using mod_unimrcp to transcribe voicemails
In-Reply-To: <CAD2CnaS6fDpQM2=aa1c7P=gxB_q_ieDeC67EPxLAEGNwSoyL4A@mail.gmail.com>
References: <CAE23trtEEiXZ-MurMTNVdLVuhZBR=_7nybTcawOcRL9haS16Ng@mail.gmail.com>
 <CAD2CnaS6fDpQM2=aa1c7P=gxB_q_ieDeC67EPxLAEGNwSoyL4A@mail.gmail.com>
Message-ID: <CACrNUca+F6AC1ipEunNpixdLuiXXC66e60QwycUvNbUTRTKWCQ@mail.gmail.com>

I don't know the solution to your problem, but is there a reason you want
to use mod_unimrcp to transcribe voicemails? Assuming you have the VM
recording, alternatively you could use APIs provided by google/AWS to
upload the recording & get the transcript back.


On Fri, Mar 12, 2021 at 7:38 AM Lewis Bergman <lewis.bergman at gmail.com>
wrote:

> I couldn't find anything. Is there an archive of presentations I am blind
> to?
>
> On Tue, Mar 9, 2021 at 11:53 PM David P <davidswalkabout at gmail.com> wrote:
>
>> One of the ClueCon 2020 talks was by someone who uses mod_unimrcp and has
>> a github project showing its FS integration, iirc.
>>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From lewis.bergman at gmail.com  Fri Mar 12 15:26:18 2021
From: lewis.bergman at gmail.com (Lewis Bergman)
Date: Fri, 12 Mar 2021 09:26:18 -0600
Subject: [Freeswitch-users] using mod_unimrcp to transcribe voicemails
In-Reply-To: <CACrNUca+F6AC1ipEunNpixdLuiXXC66e60QwycUvNbUTRTKWCQ@mail.gmail.com>
References: <CAE23trtEEiXZ-MurMTNVdLVuhZBR=_7nybTcawOcRL9haS16Ng@mail.gmail.com>
 <CAD2CnaS6fDpQM2=aa1c7P=gxB_q_ieDeC67EPxLAEGNwSoyL4A@mail.gmail.com>
 <CACrNUca+F6AC1ipEunNpixdLuiXXC66e60QwycUvNbUTRTKWCQ@mail.gmail.com>
Message-ID: <CAD2CnaSyKqoda0gaFLQN1aSvcZWPSt_JGE5iKJ440bH7hzOEpw@mail.gmail.com>

I would like to standardize on mrcp so that I can switch to another
provider, or use many of them at once to minimize cost and compare which is
better at what type of SR. If I have to code something I only have to do it
once with mod_unimrcp. I can change the provider by using a different
profile as an argument.

On Fri, Mar 12, 2021 at 9:16 AM Anshuman Rawat <rawat.anshuman at gmail.com>
wrote:

> I don't know the solution to your problem, but is there a reason you want
> to use mod_unimrcp to transcribe voicemails? Assuming you have the VM
> recording, alternatively you could use APIs provided by google/AWS to
> upload the recording & get the transcript back.
>
>
> On Fri, Mar 12, 2021 at 7:38 AM Lewis Bergman <lewis.bergman at gmail.com>
> wrote:
>
>> I couldn't find anything. Is there an archive of presentations I am blind
>> to?
>>
>> On Tue, Mar 9, 2021 at 11:53 PM David P <davidswalkabout at gmail.com>
>> wrote:
>>
>>> One of the ClueCon 2020 talks was by someone who uses mod_unimrcp and
>>> has a github project showing its FS integration, iirc.
>>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users 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



-- 
Lewis Bergman
325-439-0533 Cell
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From david.villasmil.work at gmail.com  Fri Mar 12 16:13:42 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 12 Mar 2021 16:13:42 +0000
Subject: [Freeswitch-users] No audio on bridged call
In-Reply-To: <CAL0pLpb_ECu3F2BJVNWwBgF0YJuajL+Oj08uYnRf01m=t2Rk1w@mail.gmail.com>
References: <2BB61A75-937C-4D36-A913-E82C38D65701@hxcore.ol>
 <CAGvJV4p-w-oqsKxhUkSMt0VsDHC9Mtkec6KzUtpgo87TU6HAjg@mail.gmail.com>
 <CAL0pLpbYxomvdbveJsxNWVC0zSTR+BL_07Wr3WmQ4LL0SiL9ww@mail.gmail.com>
 <CAEJMVkCcev4v++H-pGmHWqb_=1YuGSR2SZX7iLwGBgSCvgZp9g@mail.gmail.com>
 <75A3C23C-3DF5-47C7-A8E8-C0DC511E7301@hxcore.ol>
 <CAEJMVkBQ4XgxfG2e0LOL88M2FvNBc8RD2vsjHTtbHz=10fcoZw@mail.gmail.com>
 <CAL0pLpb_ECu3F2BJVNWwBgF0YJuajL+Oj08uYnRf01m=t2Rk1w@mail.gmail.com>
Message-ID: <CAFGRPVqwB62d4gbK8RCFRFEOXx0uVJwE9g7F9nooZP-Wgm4Xgg@mail.gmail.com>

This may be a dumb suggestion, but wouldn’t doing that by default with a
gateway or mod_sofia parameter parameter solve this issue without having to
go to the mailing list?



On Fri, 12 Mar 2021 at 12:52, Mircea Botoca-Huh <mircea.huh at gmail.com>
wrote:

> Indeed Brian, this is what I did. I set it on outbound originate and it is
> working perfectly.
>
> Thank you for the tip. You are the best.
>
> BR,
> Mircea.
>
> joi, 11 mar. 2021, 16:10 Brian West <brian at freeswitch.com> a scris:
>
>> I think the feature you're hitting in the upstream switch is called 'RTP
>> Clamping'.  They won't start sending media till you do, it should only be
>> doing this on inbound or outbound calls NOT BOTH.  What I gave you was a
>> trick to make FreeSWITCH inject 500ms of silence, you can lower this
>> number, you can also just set it inside the {} on the outbound originate
>> too.
>>
>> /b
>>
>>
>> On Wed, Mar 10, 2021 at 3:49 PM Mircea Botoca-Huh <mircea.huh at gmail.com>
>> wrote:
>>
>>> Hello Brian,
>>>
>>>
>>>
>>> You were absolutely right.
>>>
>>>
>>>
>>> I followed your advise and indeed with this variable set the call has
>>> audio.
>>>
>>>
>>>
>>> Thank you all very much.
>>>
>>>
>>>
>>> Best regards,
>>>
>>> Mircea.
>>>
>>>
>>>
>>> *From: *Brian West <brian at freeswitch.com>
>>> *Sent: *miercuri, 10 martie 2021 20:25
>>> *To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> *Subject: *Re: [Freeswitch-users] No audio on bridged call
>>>
>>>
>>>
>>> Are you saying a hair pin call thru your FreeSWITCH has no audio (i.e.
>>> call comes in and goes right back out)?
>>>
>>>
>>>
>>> for giggles try this at fs_cli:
>>>
>>>
>>>
>>> global_setvar execute_on_answer01='playback silence_stream://500'
>>>
>>>
>>>
>>> What you describe sounds like both ends are in a standoff waiting on the
>>> other to send media.
>>>
>>>
>>>
>>> /b
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Mar 10, 2021 at 11:50 AM Mircea Botoca-Huh <mircea.huh at gmail.com>
>>> wrote:
>>>
>>> Thank you for your answer.
>>>
>>>
>>>
>>> I have ext-rtp-ip set to correct IP address.
>>>
>>>
>>>
>>> Only the forwarded call is not working. The incoming to extension has
>>> audio.
>>>
>>>
>>>
>>> BR,
>>>
>>>
>>>
>>> Mircea
>>>
>>>
>>>
>>> mie., 10 mar. 2021, 15:05 Dragos Oancea <dragos at freeswitch.org> a scris:
>>>
>>> Check param ext-rtp-ip in the external SIP profile.
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Mar 10, 2021 at 12:35 AM Mircea Botoca-Huh <mircea.huh at gmail.com>
>>> wrote:
>>>
>>> Hello,
>>>
>>>
>>>
>>> I have freeswitch version 1.10.5 -release-17-25569c1631 64bit installed.
>>>
>>>
>>>
>>> I have one SIP provider, no nat.
>>>
>>>
>>>
>>> I use a simple dialplan which bridge the received call from SIP provider
>>> to a mobile number using the same provider.
>>>
>>>
>>>
>>> The call goes through but I have no ringback tone and if I answer there
>>> is no audio.
>>>
>>>
>>>
>>> If I send the call to one internal extension, it is working fine.
>>>
>>>
>>>
>>> The only difference I can see in the logs is that, on the call with no
>>> audio, I do not have in logs „Correct ip/port confirmed”.
>>>
>>>
>>>
>>> Any suggestion is greatly appreciated.
>>>
>>>
>>>
>>> Thank you.
>>>
>>>
>>>
>>> Best regards,
>>>
>>> Mircea.
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users 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
>>>
>>>
>>>
>>>
>>> --
>>>
>>>
>>>
>>> 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/>
>>>
>>> <https://www.facebook.com/freeswitch> <https://twitter.com/freeswitch>
>>>
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>>
>> --
>>
>> 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
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From craigwilson51 at hotmail.com  Thu Mar 11 16:50:13 2021
From: craigwilson51 at hotmail.com (CRAIG WILSON)
Date: Thu, 11 Mar 2021 16:50:13 +0000
Subject: [Freeswitch-users] User_Not_Registered
Message-ID: <DBBPR08MB6026CF68A8F9A5100CA64711B5909@DBBPR08MB6026.eurprd08.prod.outlook.com>

Hi,

I am using ASTPP on top of FS. I am receiving an inbound DID. However, my SIP Trace shows
error "User_Not_Registered" see (sip trace log below).

Running command list_user reveals nothing and sofia_contact reveals "User_Not_Registered".

Show Registrations - reveals:
9856937967,myrevbill.com,8vSSYnN7QcIRw5u3nfg5pA..,sofia/sip-ip/sip:9856937967 at 92.7.187.203:39706;transport=UDP;rinstance=ec30c6121794d08f,1615480452,92.7.187.203,39706,udp,astpp.myrevbill.com,

1 total.

Then I ran command below:
sofia status profile sip-ip  reg

Registrations:
=================================================================================================
Call-ID:        S3gXf-fjxpA9ws47L8bh9w..
User:           9856937967 at myrevbill.com
Contact:        "" <sip:9856937967 at 92.7.187.203:62990;transport=UDP;rinstance=df94511808fbcf07>
Agent:          Z 5.4.9 rv2.10.11.7
Status:         Registered(UDP)(unknown) EXP(2021-03-11 11:03:29) EXPSECS(102)
Ping-Status:    Reachable
Ping-Time:      0.00
Host:           astpp.myrevbill.com
IP:             92.7.187.203
Port:           62990
Auth-User:      9856937967
Auth-Realm:     myrevbill.com
MWI-Account:    9856937967 at myrevbill.com

Total items returned: 1



Sip Trace Log:

2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:584 (sofia/sip-ip/01226971834 at 10.0.1.29) Running State Change CS_NEW (Cur 1 Tot 1)
2021-03-10 12:20:17.534757 [DEBUG] sofia.c:9873 sofia/sip-ip/01226971834 at 10.0.1.29 receiving invite from 10.0.1.29:5060 version: 1.6.20  64bit
2021-03-10 12:20:17.534757 [DEBUG] sofia.c:9989 IP 10.0.1.29 Approved by acl "default[]". Access Granted.
2021-03-10 12:20:17.534757 [DEBUG] sofia.c:7084 Channel sofia/sip-ip/01226971834 at 10.0.1.29 entering state [received][100]
2021-03-10 12:20:17.534757 [DEBUG] sofia.c:7094 Remote SDP:
v=0
o=- 283383691 283383691 IN IP4 10.0.1.29
s=Asterisk
c=IN IP4 10.0.1.29
t=0 0
m=audio 11144 RTP/AVP 0 8 9 4 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150

2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G729:18:8000:20:8000:1]
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G729:18:8000:20:8000:1]
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G722:9:8000:20:64000:1]/[G729:18:8000:20:8000:1]
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G723:4:8000:20:6300:1]/[PCMU:0:8000:20:64000:1]
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G723:4:8000:20:6300:1]/[PCMA:8:8000:20:64000:1]
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G723:4:8000:20:6300:1]/[G729:18:8000:20:8000:1]
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMU:0:8000:20:64000:1]
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMA:8:8000:20:64000:1]
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [GSM:3:8000:20:13200:1]/[G729:18:8000:20:8000:1]
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4365 Set telephone-event payload to 101 at 8000
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:3061 Set Codec sofia/sip-ip/01226971834 at 10.0.1.29 PCMU/8000 20 ms 160 samples 64000 bits 1 channels
2021-03-10 12:20:17.534757 [DEBUG] switch_core_codec.c:111 sofia/sip-ip/01226971834 at 10.0.1.29 Original read codec set to PCMU:0
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4708 Set telephone-event payload to 101 at 8000
2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4767 sofia/sip-ip/01226971834 at 10.0.1.29 Set 2833 dtmf send payload to 101 recv payload to 101
2021-03-10 12:20:17.534757 [DEBUG] sofia.c:7507 (sofia/sip-ip/01226971834 at 10.0.1.29) State Change CS_NEW -> CS_INIT
2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:603 (sofia/sip-ip/01226971834 at 10.0.1.29) State NEW
2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:584 (sofia/sip-ip/01226971834 at 10.0.1.29) Running State Change CS_INIT (Cur 1 Tot 1)
2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:627 (sofia/sip-ip/01226971834 at 10.0.1.29) State INIT
2021-03-10 12:20:17.534757 [DEBUG] mod_sofia.c:90 sofia/sip-ip/01226971834 at 10.0.1.29 SOFIA INIT
2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:40 sofia/sip-ip/01226971834 at 10.0.1.29 Standard INIT
2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:48 (sofia/sip-ip/01226971834 at 10.0.1.29) State Change CS_INIT -> CS_ROUTING
2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:627 (sofia/sip-ip/01226971834 at 10.0.1.29) State INIT going to sleep
2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:584 (sofia/sip-ip/01226971834 at 10.0.1.29) Running State Change CS_ROUTING (Cur 1 Tot 1)
2021-03-10 12:20:17.534757 [DEBUG] switch_channel.c:2249 (sofia/sip-ip/01226971834 at 10.0.1.29) Callstate Change DOWN -> RINGING
2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:643 (sofia/sip-ip/01226971834 at 10.0.1.29) State ROUTING
2021-03-10 12:20:17.534757 [DEBUG] mod_sofia.c:143 sofia/sip-ip/01226971834 at 10.0.1.29 SOFIA ROUTING
2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:236 sofia/sip-ip/01226971834 at 10.0.1.29 Standard ROUTING
2021-03-10 12:20:17.534757 [INFO] mod_dialplan_xml.c:637 Processing 01226971834 <01226971834>->03302290443 in context default
2021-03-10 12:20:17.596174 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f3ebc1cae70 Connected.
2021-03-10 12:20:17.596174 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [LOAD_CONF] Query :SELECT name,value FROM `system` WHERE group_title IN ('global','opensips','callingcard','calls','InternationalPrefixes')
2021-03-10 12:20:17.596174 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [LOAD_ADDON_CONF] Query :SELECT package_name FROM addons
2021-03-10 12:20:17.596174 [INFO] switch_cpp.cpp:1365 [ASTPP] [Dialplan] Dialed number : 03302290443
2021-03-10 12:20:17.596174 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [DOAUTHORIZATION] Query :SELECT access_number FROM accessnumber WHERE access_number = '03302290443' AND status=0 limit 1
2021-03-10 12:20:17.596174 [INFO] switch_cpp.cpp:1365 [ASTPP] [Dialplan] Caller Id name / number  : 01226971834 / 01226971834
2021-03-10 12:20:17.596174 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [CHECK_DID] Query :SELECT A.id as id,A.number as did_number,B.id as accountid,B.number as account_code,A.number as did_number,A.connectcost,A.includedseconds,A.cost,A.inc,A.extensions,A.maxchannels,A.call_type,A.city,A.province,A.init_inc,A.leg_timeout,A.status,A.country_id,A.call_type_vm_flag FROM dids AS A,accounts AS B WHERE B.status=0 AND B.deleted=0 AND B.id=A.accountid AND A.number ="03302290443" LIMIT 1
2021-03-10 12:20:17.596174 [INFO] switch_cpp.cpp:1365 [ASTPP] [Dialplan] Call direction : inbound
2021-03-10 12:20:17.596174 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [IPAUTHENTICATION] Query :SELECT ip_map.*, (SELECT number FROM accounts where id=accountid AND status=0 AND deleted=0) AS account_code FROM ip_map WHERE INET_ATON("10.0.1.29") BETWEEN(INET_ATON(SUBSTRING_INDEX(`ip`, '/', 1)) & 0xffffffff ^((0x1 <<(32 -  SUBSTRING_INDEX(`ip`, '/', -1))) -1 )) AND(INET_ATON(SUBSTRING_INDEX(`ip`, '/', 1)) |((0x100000000 >> SUBSTRING_INDEX(`ip`,'/', -1)) -1))  AND "03302290443"  LIKE CONCAT(prefix,'%') ORDER BY LENGTH(prefix) DESC LIMIT 1
2021-03-10 12:20:17.634761 [INFO] switch_cpp.cpp:1365 [ASTPP] [Accountcode : 9856937967]
2021-03-10 12:20:17.634761 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [DOAUTHORIZATION] Query :SELECT * FROM accounts WHERE number = "9856937967" AND deleted = 0 limit 1
2021-03-10 12:20:17.634761 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [CHECK_SPEEDDIAL] Query :SELECT A.number FROM speed_dial as A,accounts as B WHERE B.status=0 AND B.deleted=0 AND B.id=A.accountid AND A.speed_num ="03302290443" AND A.accountid = '13' limit 1
2021-03-10 12:20:17.634761 [INFO] switch_cpp.cpp:1365 [ASTPP] [Dialplan] SPEED DIAL NUMBER : 03302290443
2021-03-10 12:20:17.634761 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [CHECK_DID] Query :SELECT A.id as id,A.number as did_number,B.id as accountid,B.number as account_code,A.number as did_number,A.connectcost,A.includedseconds,A.cost,A.inc,A.extensions,A.maxchannels,A.call_type,A.city,A.province,A.init_inc,A.leg_timeout,A.status,A.country_id,A.call_type_vm_flag FROM dids AS A,accounts AS B WHERE B.status=0 AND B.deleted=0 AND B.id=A.accountid AND A.number ="03302290443" LIMIT 1
2021-03-10 12:20:17.634761 [INFO] switch_cpp.cpp:1365 [ASTPP] [Dialplan] New Call direction : inbound
2021-03-10 12:20:17.634761 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [IS_CHECK_DID] Query :SELECT * FROM dids WHERE number ="03302290443" AND (accountid = 0 OR status = 1) LIMIT 1
2021-03-10 12:20:17.634761 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [GET_PACKAGE_INFO] call_direction :inbound
2021-03-10 12:20:17.634761 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [GET_PACKAGE_INFO] DID_ACCOUNTID :2
2021-03-10 12:20:17.634761 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [GET_PACKAGE_INFO] Query :SELECT *,P.id as package_id,P.product_id as product_id FROM packages_view as P inner join package_patterns as PKGPTR on P.product_id = PKGPTR.product_id WHERE (patterns = '^03302290443.*' OR patterns = '^0330229044.*' OR patterns = '^033022904.*' OR patterns = '^03302290.*' OR patterns = '^0330229.*' OR patterns = '^033022.*' OR patterns = '^03302.*' OR patterns = '^0330.*' OR patterns = '^033.*' OR patterns = '^03.*' OR patterns = '^0.*' OR patterns ='--') AND accountid = 2 ORDER BY LENGTH(PKGPTR.patterns) DESC
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] =============== Account Information ===================
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] User id : 13
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Account code : 9856937967
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Balance : 10000
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Type : 0 [0:prepaid,1:postpaid]
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Ratecard id : 1
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] ========================================================
2021-03-10 12:20:17.655679 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [GET_PRICELIST_INFO] Query :select * from pricelists WHERE id = 1 AND status = 0
2021-03-10 12:20:17.655679 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [GET_RATES] call_direction :inbound
2021-03-10 12:20:17.655679 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [CHECK_DID] Query :SELECT A.id as id,A.number as did_number,B.id as accountid,B.number as account_code,A.number as did_number,A.connectcost,A.includedseconds,A.cost,A.inc,A.extensions,A.maxchannels,A.call_type,A.city,A.province,A.init_inc,A.leg_timeout,A.status,A.country_id,A.call_type_vm_flag FROM dids AS A,accounts AS B WHERE B.status=0 AND B.deleted=0 AND B.id=A.accountid AND A.number ="03302290443" LIMIT 1
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] call_direction:::::: inbound
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] =============== Rates Information ===================
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] ID : 1
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Connectcost : 0.00000
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Includedseconds : 0
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Cost : 1.30378
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] comment : 03302290443
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Country Id : 200
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Accid : 13
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] ================================================================
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] [FIND_MAXLENGTH] Your10000 balance Accountid 13 !!!
2021-03-10 12:20:17.655679 [NOTICE] switch_cpp.cpp:1365 [ASTPP] [FIND_MAXLENGTH] Limiting call to config max length 100 mins!
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Call Max length duration : 100 minutes
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] [userinfo] INB_FREE:TRUE
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] [userinfo] free_inbound:1
2021-03-10 12:20:17.655679 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [GET_OVERRIDE_CALLERID] Query :SELECT callerid_name as cid_name,callerid_number as cid_number,accountid FROM accounts_callerid WHERE accountid = 13 AND status=0 LIMIT 1
2021-03-10 12:20:17.655679 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [DOAUTHORIZATION] Query :SELECT * FROM accounts WHERE id = "2" AND deleted = 0 limit 1
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] [userinfo] Actual CustomerInfo XML:13
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] [userinfo] Userinfo XML:13
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] [userinfo] Actual CustomerInfo XML : 13
2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] maxlength::::::::: 100
2021-03-10 12:20:17.655679 [DEBUG] switch_cpp.cpp:1365 [ASTPP] custom_function_name:::::::::::::::::::::::::custom_inbound_0
2021-03-10 12:20:17.655679 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [Dialplan] Generated XML:<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="dialplan" description="ASTPP Dialplan">
<context name="default">
<extension name="03302290443">
<condition field="destination_number" expression="03302290443">
<action application="set" data="effective_destination_number=03302290443"/>
<action application="set" data="bridge_pre_execute_bleg_app=sched_hangup"/>
<action application="set" data="bridge_pre_execute_bleg_data=+6000 normal_clearing"/>
<action application="set" data="callstart=2021-03-10 12:20:17"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="continue_on_fail=TRUE"/>
<action application="set" data="account_id=13"/>
<action application="set" data="parent_id=0"/>
<action application="set" data="entity_id=3"/>
<action application="set" data="call_processed=internal"/>
<action application="set" data="call_direction=inbound"/>
<action application="set" data="accountname=FreePBX"/>
<action application="set" data="origination_rates_did=ID:1|CODE:^03302290443.*|DESTINATION:03302290443|CONNECTIONCOST:0.00000|INCLUDEDSECONDS:0|CT:0|COST:1.30378|INC:0|INITIALBLOCK:0|RATEGROUP:0|MARKUP:0|CI:200|ACCID:2"/>
<action application="set" data="origination_rates=0"/>
<action application="set" data="original_caller_id_name=01226971834"/>
<action application="set" data="original_caller_id_number=01226971834"/>
<action application="set" data="effective_caller_id_name=01226971834"/>
<action application="set" data="effective_caller_id_number=01226971834"/>
<action application="set" data="receiver_accid=2"/>
<action application="export" data="presence_data=x|||Dial9(9856937967)|||^03302290443.* // 03302290443 // 1.30378||||||DID"/>
<action application="export" data="call_type=0"/>
<action application="set" data="calltype=DID-LOCAL"/>
<action application="bridge" data="[leg_timeout=0]user/9856937967@${domain_name}"/>
<condition field="${cond(${user_data 9856937967@${domain_name} param vm-enabled} == true ? YES : NO)}" expression="^YES$">
<action application="answer"/>
<action application="export" data="voicemail_alternate_greet_id=03302290443"/>
<action application="voicemail" data="default $${domain_name} 9856937967"/>
<anti-action application="hangup" data="${originate_disposition}"/>
</condition>
</condition>
</extension>
</context>
</section>
</document>
2021-03-10 12:20:17.655679 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f3ebc1cae70 released.
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 parsing [default->03302290443] continue=false
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Regex (PASS) [03302290443] destination_number(03302290443) =~ /03302290443/ break=on-false
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(effective_destination_number=03302290443)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(bridge_pre_execute_bleg_app=sched_hangup)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(bridge_pre_execute_bleg_data=+6000 normal_clearing)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(callstart=2021-03-10 12:20:17)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(hangup_after_bridge=true)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(continue_on_fail=TRUE)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(account_id=13)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(parent_id=0)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(entity_id=3)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(call_processed=internal)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(call_direction=inbound)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(accountname=FreePBX)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(origination_rates_did=ID:1|CODE:^03302290443.*|DESTINATION:03302290443|CONNECTIONCOST:0.00000|INCLUDEDSECONDS:0|CT:0|COST:1.30378|INC:0|INITIALBLOCK:0|RATEGROUP:0|MARKUP:0|CI:200|ACCID:2)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(origination_rates=0)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(original_caller_id_name=01226971834)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(original_caller_id_number=01226971834)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(effective_caller_id_name=01226971834)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(effective_caller_id_number=01226971834)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(receiver_accid=2)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action export(presence_data=x|||Dial9(9856937967)|||^03302290443.* // 03302290443 // 1.30378||||||DID)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action export(call_type=0)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(calltype=DID-LOCAL)
Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action bridge([leg_timeout=0]user/9856937967@${domain_name})
|--- Dialplan: Processing recursive conditions level:1 [03302290443_recur_1] require-nested=TRUE
|--- Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Regex (PASS) [03302290443_recur_1] ${cond(${user_data 9856937967@${domain_name} param vm-enabled} == true ? YES : NO)}(YES) =~ /^YES$/ break=on-false
|--- Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action answer()
|--- Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action export(voicemail_alternate_greet_id=03302290443)
|--- Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action voicemail(default $${domain_name} 9856937967)
2021-03-10 12:20:17.682238 [DEBUG] switch_core_state_machine.c:286 (sofia/sip-ip/01226971834 at 10.0.1.29) State Change CS_ROUTING -> CS_EXECUTE
2021-03-10 12:20:17.682238 [DEBUG] switch_core_state_machine.c:643 (sofia/sip-ip/01226971834 at 10.0.1.29) State ROUTING going to sleep
2021-03-10 12:20:17.682238 [DEBUG] switch_core_state_machine.c:584 (sofia/sip-ip/01226971834 at 10.0.1.29) Running State Change CS_EXECUTE (Cur 1 Tot 1)
2021-03-10 12:20:17.682238 [DEBUG] switch_core_state_machine.c:650 (sofia/sip-ip/01226971834 at 10.0.1.29) State EXECUTE
2021-03-10 12:20:17.682238 [DEBUG] mod_sofia.c:198 sofia/sip-ip/01226971834 at 10.0.1.29 SOFIA EXECUTE
2021-03-10 12:20:17.682238 [DEBUG] switch_core_state_machine.c:328 sofia/sip-ip/01226971834 at 10.0.1.29 Standard EXECUTE
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(effective_destination_number=03302290443)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [effective_destination_number]=[03302290443]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(bridge_pre_execute_bleg_app=sched_hangup)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [bridge_pre_execute_bleg_app]=[sched_hangup]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(bridge_pre_execute_bleg_data=+6000 normal_clearing)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [bridge_pre_execute_bleg_data]=[+6000 normal_clearing]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(callstart=2021-03-10 12:20:17)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [callstart]=[2021-03-10 12:20:17]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(hangup_after_bridge=true)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [hangup_after_bridge]=[true]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(continue_on_fail=TRUE)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [continue_on_fail]=[TRUE]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(account_id=13)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [account_id]=[13]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(parent_id=0)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [parent_id]=[0]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(entity_id=3)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [entity_id]=[3]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(call_processed=internal)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [call_processed]=[internal]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(call_direction=inbound)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [call_direction]=[inbound]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(accountname=FreePBX)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [accountname]=[FreePBX]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(origination_rates_did=ID:1|CODE:^03302290443.*|DESTINATION:03302290443|CONNECTIONCOST:0.00000|INCLUDEDSECONDS:0|CT:0|COST:1.30378|INC:0|INITIALBLOCK:0|RATEGROUP:0|MARKUP:0|CI:200|ACCID:2)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [origination_rates_did]=[ID:1|CODE:^03302290443.*|DESTINATION:03302290443|CONNECTIONCOST:0.00000|INCLUDEDSECONDS:0|CT:0|COST:1.30378|INC:0|INITIALBLOCK:0|RATEGROUP:0|MARKUP:0|CI:200|ACCID:2]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(origination_rates=0)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [origination_rates]=[0]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(original_caller_id_name=01226971834)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [original_caller_id_name]=[01226971834]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(original_caller_id_number=01226971834)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [original_caller_id_number]=[01226971834]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(effective_caller_id_name=01226971834)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [effective_caller_id_name]=[01226971834]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(effective_caller_id_number=01226971834)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [effective_caller_id_number]=[01226971834]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(receiver_accid=2)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [receiver_accid]=[2]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 export(presence_data=x|||Dial9(9856937967)|||^03302290443.* // 03302290443 // 1.30378||||||DID)
2021-03-10 12:20:17.682238 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [presence_data]=[x|||Dial9(9856937967)|||^03302290443.* // 03302290443 // 1.30378||||||DID]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 export(call_type=0)
2021-03-10 12:20:17.682238 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [call_type]=[0]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(calltype=DID-LOCAL)
2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET sofia/sip-ip/01226971834 at 10.0.1.29 [calltype]=[DID-LOCAL]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 bridge([leg_timeout=0]user/9856937967 at 10.0.1.212)
2021-03-10 12:20:17.682238 [DEBUG] switch_channel.c:1250 sofia/sip-ip/01226971834 at 10.0.1.29 EXPORTING[export_vars] [presence_data]=[x|||Dial9(9856937967)|||^03302290443.* // 03302290443 // 1.30378||||||DID] to event
2021-03-10 12:20:17.682238 [DEBUG] switch_channel.c:1250 sofia/sip-ip/01226971834 at 10.0.1.29 EXPORTING[export_vars] [call_type]=[0] to event
2021-03-10 12:20:17.682238 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables
2021-03-10 12:20:17.682238 [DEBUG] switch_ivr_originate.c:2669 Parsing session specific variables
2021-03-10 12:20:17.682238 [DEBUG] switch_channel.c:1250 sofia/sip-ip/01226971834 at 10.0.1.29 EXPORTING[export_vars] [presence_data]=[x|||Dial9(9856937967)|||^03302290443.* // 03302290443 // 1.30378||||||DID] to event
2021-03-10 12:20:17.682238 [DEBUG] switch_channel.c:1250 sofia/sip-ip/01226971834 at 10.0.1.29 EXPORTING[export_vars] [call_type]=[0] to event
2021-03-10 12:20:17.682238 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables
2021-03-10 12:20:17.682238 [NOTICE] switch_ivr_originate.c:2851 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2021-03-10 12:20:17.682238 [DEBUG] switch_ivr_originate.c:3848 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2021-03-10 12:20:17.682238 [NOTICE] switch_ivr_originate.c:2851 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
2021-03-10 12:20:17.682238 [DEBUG] switch_ivr_originate.c:3848 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2021-03-10 12:20:17.682238 [INFO] mod_dptools.c:3436 Originate Failed.  Cause: USER_NOT_REGISTERED
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 answer()
2021-03-10 12:20:17.682238 [DEBUG] switch_core_media.c:6878 AUDIO RTP [sofia/sip-ip/01226971834 at 10.0.1.29] 10.0.1.212 port 16840 -> 10.0.1.29 port 11144 codec: 0 ms: 20
2021-03-10 12:20:17.682238 [DEBUG] switch_rtp.c:4159 Not using a timer
2021-03-10 12:20:17.682238 [DEBUG] switch_core_media.c:7180 sofia/sip-ip/01226971834 at 10.0.1.29 Set 2833 dtmf send payload to 101
2021-03-10 12:20:17.682238 [DEBUG] switch_core_media.c:7187 sofia/sip-ip/01226971834 at 10.0.1.29 Set 2833 dtmf receive payload to 101
2021-03-10 12:20:17.682238 [DEBUG] switch_core_media.c:7210 sofia/sip-ip/01226971834 at 10.0.1.29 Set rtp dtmf delay to 40
2021-03-10 12:20:17.682238 [DEBUG] mod_sofia.c:850 Local SDP sofia/sip-ip/01226971834 at 10.0.1.29:
v=0
o=FreeSWITCH 1615361977 1615361978 IN IP4 10.0.1.212
s=FreeSWITCH
c=IN IP4 10.0.1.212
t=0 0
m=audio 16840 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2021-03-10 12:20:17.682238 [NOTICE] mod_dptools.c:1312 Channel [sofia/sip-ip/01226971834 at 10.0.1.29] has been answered
2021-03-10 12:20:17.682238 [DEBUG] switch_channel.c:3773 (sofia/sip-ip/01226971834 at 10.0.1.29) Callstate Change RINGING -> ACTIVE
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 export(voicemail_alternate_greet_id=03302290443)
2021-03-10 12:20:17.682238 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [voicemail_alternate_greet_id]=[03302290443]
EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 voicemail(default 10.0.1.212 9856937967)
2021-03-10 12:20:17.715099 [DEBUG] sofia.c:7084 Channel sofia/sip-ip/01226971834 at 10.0.1.29 entering state [completed][200]
2021-03-10 12:20:17.715099 [DEBUG] sofia.c:7084 Channel sofia/sip-ip/01226971834 at 10.0.1.29 entering state [ready][200]
2021-03-10 12:20:49.734765 [NOTICE] sofia.c:1012 Hangup sofia/sip-ip/01226971834 at 10.0.1.29 [CS_EXECUTE] [NORMAL_CLEARING]
2021-03-10 12:20:49.734765 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en]
2021-03-10 12:20:49.734765 [DEBUG] switch_core_session.c:2815 sofia/sip-ip/01226971834 at 10.0.1.29 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2021-03-10 12:20:49.734765 [DEBUG] switch_core_state_machine.c:650 (sofia/sip-ip/01226971834 at 10.0.1.29) State EXECUTE going to sleep
2021-03-10 12:20:49.734765 [DEBUG] switch_core_state_machine.c:584 (sofia/sip-ip/01226971834 at 10.0.1.29) Running State Change CS_HANGUP (Cur 1 Tot 1)
2021-03-10 12:20:49.734765 [DEBUG] switch_core_state_machine.c:850 (sofia/sip-ip/01226971834 at 10.0.1.29) Callstate Change ACTIVE -> HANGUP
2021-03-10 12:20:49.734765 [DEBUG] switch_core_state_machine.c:852 (sofia/sip-ip/01226971834 at 10.0.1.29) State HANGUP
2021-03-10 12:20:49.734765 [DEBUG] mod_sofia.c:438 Channel sofia/sip-ip/01226971834 at 10.0.1.29 hanging up, cause: NORMAL_CLEARING
2021-03-10 12:20:49.734765 [DEBUG] switch_core_state_machine.c:60 sofia/sip-ip/01226971834 at 10.0.1.29 Standard HANGUP, cause: NORMAL_CLEARING
2021-03-10 12:20:49.734765 [DEBUG] switch_core_state_machine.c:852 (sofia/sip-ip/01226971834 at 10.0.1.29) State HANGUP going to sleep
2021-03-10 12:20:49.734765 [DEBUG] switch_core_state_machine.c:619 (sofia/sip-ip/01226971834 at 10.0.1.29) State Change CS_HANGUP -> CS_REPORTING
2021-03-10 12:20:49.734765 [DEBUG] switch_core_state_machine.c:584 (sofia/sip-ip/01226971834 at 10.0.1.29) Running State Change CS_REPORTING (Cur 1 Tot 1)
2021-03-10 12:20:49.734765 [DEBUG] switch_core_state_machine.c:938 (sofia/sip-ip/01226971834 at 10.0.1.29) State REPORTING
2021-03-10 12:20:49.734765 [INFO] mod_json_cdr.c:271 Process [f9793c2e-819a-11eb-b65f-8914a0e01ab2.cdr.json]
2021-03-10 12:20:49.794754 [DEBUG] switch_core_state_machine.c:174 sofia/sip-ip/01226971834 at 10.0.1.29 Standard REPORTING, cause: NORMAL_CLEARING
2021-03-10 12:20:49.794754 [DEBUG] switch_core_state_machine.c:938 (sofia/sip-ip/01226971834 at 10.0.1.29) State REPORTING going to sleep
2021-03-10 12:20:49.794754 [DEBUG] switch_core_state_machine.c:610 (sofia/sip-ip/01226971834 at 10.0.1.29) State Change CS_REPORTING -> CS_DESTROY
2021-03-10 12:20:49.794754 [DEBUG] switch_core_session.c:1665 Session 1 (sofia/sip-ip/01226971834 at 10.0.1.29) Locked, Waiting on external entities
2021-03-10 12:20:49.794754 [NOTICE] switch_core_session.c:1683 Session 1 (sofia/sip-ip/01226971834 at 10.0.1.29) Ended
2021-03-10 12:20:49.794754 [NOTICE] switch_core_session.c:1687 Close Channel sofia/sip-ip/01226971834 at 10.0.1.29 [CS_DESTROY]
2021-03-10 12:20:49.794754 [DEBUG] switch_core_state_machine.c:741 (sofia/sip-ip/01226971834 at 10.0.1.29) Running State Change CS_DESTROY (Cur 0 Tot 1)
2021-03-10 12:20:49.794754 [DEBUG] switch_core_state_machine.c:751 (sofia/sip-ip/01226971834 at 10.0.1.29) State DESTROY
2021-03-10 12:20:49.794754 [DEBUG] mod_sofia.c:343 sofia/sip-ip/01226971834 at 10.0.1.29 SOFIA DESTROY
2021-03-10 12:20:49.794754 [DEBUG] switch_core_state_machine.c:181 sofia/sip-ip/01226971834 at 10.0.1.29 Standard DESTROY
2021-03-10 12:20:49.794754 [DEBUG] switch_core_state_machine.c:751 (sofia/sip-ip/01226971834 at 10.0.1.29) State DESTROY going to sleep
freeswitch at astpp.myrevbill.com>

Thanks!!


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From infinite3219 at gmail.com  Fri Mar 12 03:24:37 2021
From: infinite3219 at gmail.com (Terry C)
Date: Thu, 11 Mar 2021 21:24:37 -0600
Subject: [Freeswitch-users] Opus packet loss
Message-ID: <CA+pHSymbmz8q0TyzCDMa1=TLYD08W5AXUu+pOmjuUebUKJ5V-g@mail.gmail.com>

Hi,

We use FreeSwitch 1.10.5 with Opus audio for clients. We target voice
quality to be wideband. The config settings are below. We also set a jitter
buffer in the dialplan.

Is there a downside to increasing the packet loss percent to 40-50% other
than using more bandwidth? Is there a voice quality impact if you set it
too high (what would be considered too high)?

Thanks.

<param name="use-vbr" value="1"/>
<param name="use-dtx" value="1"/>
<param name="complexity" value="10"/>
<param name="packet-loss-percent" value="30"/>
<param name="keep-fec-enabled" value="1"/>
<param name="use-jb-lookahead" value="1"/>
<param name="maxaveragebitrate" value="40000"/>
<param name="maxplaybackrate" value="16000"/>
<param name="sprop-maxcapturerate" value="16000"/>
<param name="adjust-bitrate" value="1"/>
</settings>
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From mitachundkrach at gmail.com  Fri Mar 12 16:41:07 2021
From: mitachundkrach at gmail.com (R G)
Date: Fri, 12 Mar 2021 17:41:07 +0100
Subject: [Freeswitch-users] Record Frequency Problems
Message-ID: <3bbabfa0-fe4a-6434-7939-10ea5fd92b15@gmail.com>

Hi all,

i've got a weird bug with the record_session function.

I added he record_session function into my dialplan and set the 
record_sample_rate to 16000.

When i use Google Chrome the record is fine and the wave file has a 
16khz Sample Rate.

When i use Mozilla Firefox the Sample Rate is at 32khz.

my additions to the dialplan:

       <action  application="set"  data="RECORD_READ_ONLY=true"/>
       <action  application="set"  data="record_sample_rate=8000"/>
       <action  application="record_session"  data="/var/freeswitch/meetings/${strftime(%Y-%m-%d-%H-%M-%S)}_${call_uuid}.wav"/>


Kind regards,

Robert

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From brian at freeswitch.com  Fri Mar 12 17:42:32 2021
From: brian at freeswitch.com (Brian West)
Date: Fri, 12 Mar 2021 11:42:32 -0600
Subject: [Freeswitch-users] Un-Subscribing on 481
In-Reply-To: <CAGAa33svGkW6=-p07vQSRuzxbqHi+a0fJ1bK_XrrvZ5gFaYB4g@mail.gmail.com>
References: <CAGAa33svGkW6=-p07vQSRuzxbqHi+a0fJ1bK_XrrvZ5gFaYB4g@mail.gmail.com>
Message-ID: <CAEJMVkC0FCPCU7=t8rUcPpxZ7d2dGXJoFDp-phxRwa=BAcZ+hw@mail.gmail.com>

Please file issues here https://github.com/signalwire/freeswitch/issues

/b

On Thu, Mar 11, 2021 at 10:46 PM Joseph Barrero <joe at expert.net> wrote:

> Hi, everyone.
>
> I recently noticed that Freeswitch continues sending NOTIFY messages even
> after receiving a 481 from my proxy after not being able to deliver the
> messages to the UA.  This seems to happen when the UA disconnects from the
> Internet briefly and registers again before the previous subscription
> expires.
>
> Below is the code I found in sofia.c that seems to handle the 481 response.
>
> if (status == 481 && sip && !sip->sip_retry_after && sip->sip_call_id &&
> (!sofia_private || !sofia_private->is_call)) {
>         char *sql;
>
>         sql = switch_mprintf("delete from sip_subscriptions where
> call_id='%q'", sip->sip_call_id->i_id);
>         switch_assert(sql != NULL);
>         sofia_glue_execute_sql(profile, &sql, SWITCH_TRUE);
>         nua_handle_destroy(nh);
> }
>
>
> Assuming that the variable sofia_private->is_call evaluates to true when
> NOTIFY messages are sent during a call, it seems to curtail the removal of
> the SUBSCRIPTION even after receiving a 481 from the proxy.
>
> Is there a drawback to removing a subscription on a 481 response during a
> call?  Would it make sense to simply remove the check if there is a call in
> place? Is checking ofia_private->is_call there for another purpose?
>
> Thanks,
> Joe Barrero
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users 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>
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From brian at freeswitch.com  Fri Mar 12 19:04:27 2021
From: brian at freeswitch.com (Brian West)
Date: Fri, 12 Mar 2021 13:04:27 -0600
Subject: [Freeswitch-users] Opus packet loss
In-Reply-To: <CA+pHSymbmz8q0TyzCDMa1=TLYD08W5AXUu+pOmjuUebUKJ5V-g@mail.gmail.com>
References: <CA+pHSymbmz8q0TyzCDMa1=TLYD08W5AXUu+pOmjuUebUKJ5V-g@mail.gmail.com>
Message-ID: <CAEJMVkBA0F8=et3EE5PRZ3KPc9oNnRWf4_S4fz0CmaAG2n=kWQ@mail.gmail.com>

CPU and bandwidth.

/b


On Fri, Mar 12, 2021 at 12:40 PM Terry C <infinite3219 at gmail.com> wrote:

> Hi,
>
> We use FreeSwitch 1.10.5 with Opus audio for clients. We target voice
> quality to be wideband. The config settings are below. We also set a jitter
> buffer in the dialplan.
>
> Is there a downside to increasing the packet loss percent to 40-50% other
> than using more bandwidth? Is there a voice quality impact if you set it
> too high (what would be considered too high)?
>
> Thanks.
>
> <param name="use-vbr" value="1"/>
> <param name="use-dtx" value="1"/>
> <param name="complexity" value="10"/>
> <param name="packet-loss-percent" value="30"/>
> <param name="keep-fec-enabled" value="1"/>
> <param name="use-jb-lookahead" value="1"/>
> <param name="maxaveragebitrate" value="40000"/>
> <param name="maxplaybackrate" value="16000"/>
> <param name="sprop-maxcapturerate" value="16000"/>
> <param name="adjust-bitrate" value="1"/>
> </settings>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users 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>
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From davidswalkabout at gmail.com  Sat Mar 13 03:15:55 2021
From: davidswalkabout at gmail.com (David P)
Date: Sat, 13 Mar 2021 16:15:55 +1300
Subject: [Freeswitch-users] rtp-timeout-sec VS media_timeout
Message-ID: <CAE23truYg_jDprWbiNCc2kH6bWUXNPCkF+eLB-UyDQMojZL+eQ@mail.gmail.com>

Using FS10.5 installed from packages on Debian Stretch...

I wanted to replicate the behavior we had with <param name="rtp-timeout-sec"
value="300"/> so that we'll hangup calls in which leg B of a conference has
sent no media for 5 mins.

I edited our dialplan like this...

<action application="conference_set_auto_outcall"
 data="['conference_member_flags=endconf,jitterbuffer_msec=5p:100p,media_timeout=300000']sofia/gateway/...deleted..."/>

and restarted like this...

fs_cli -x 'fsctl shutdown elegant restart'

But after 8 mins in such a call there was no hangup.

---------- Forwarded message ----------
From: Dragos Oancea <dragos at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Cc:
Bcc:
Date: Thu, 4 Mar 2021 11:04:26 +0200
Subject: Re: [Freeswitch-users] rtp-timeout-sec VS media_timeout
For media timeout there are the following chan vars:
media_timeout, media_hold_timeout, media_timeout,
media_hold_timeout_video, media_hold_timeout_audio,
media_timeout_audio
.

They are in milliseconds, not seconds like rtp-timeout-sec .

<param name="media_timeout" value="300000"/>
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From dragos at freeswitch.org  Sat Mar 13 12:26:28 2021
From: dragos at freeswitch.org (Dragos Oancea)
Date: Sat, 13 Mar 2021 14:26:28 +0200
Subject: [Freeswitch-users] Opus packet loss
In-Reply-To: <CAEJMVkBA0F8=et3EE5PRZ3KPc9oNnRWf4_S4fz0CmaAG2n=kWQ@mail.gmail.com>
References: <CA+pHSymbmz8q0TyzCDMa1=TLYD08W5AXUu+pOmjuUebUKJ5V-g@mail.gmail.com>
 <CAEJMVkBA0F8=et3EE5PRZ3KPc9oNnRWf4_S4fz0CmaAG2n=kWQ@mail.gmail.com>
Message-ID: <CAGvJV4qPNFOM-Z-DWJe4O3F756-Kt3yCOSWK-pD=ktbRcX2vZw@mail.gmail.com>

When trying to understand how it works make sure you are not mixing
decoding side settings with the encoding side settings.
packet-loss-percent is encoding side and is the initial value for loss if
you use adjust-bitrate (callback with SCC_AUDIO_PACKET_LOSS from the core)
.
The callback comes only if there is incoming RTCP - which for audio is
typically at 5 seconds interval.So the adjust-bitrate feature will update
the loss too, not only the bitrate.

Otherwise, without adjust-birate, it's the default value
for OPUS_SET_PACKET_LOSS_PERC() which will stay like this along the call
and depending on the preset bitrate it will generate a certain amount of
FEC on top of the payload of some of the packets (encoder decisions) .  So
no, you should not set that very high, since you can't know the real
network conditions before the call is made. 15 or 20 are good values.


On Fri, Mar 12, 2021 at 9:05 PM Brian West <brian at freeswitch.com> wrote:

> CPU and bandwidth.
>
> /b
>
>
> On Fri, Mar 12, 2021 at 12:40 PM Terry C <infinite3219 at gmail.com> wrote:
>
>> Hi,
>>
>> We use FreeSwitch 1.10.5 with Opus audio for clients. We target voice
>> quality to be wideband. The config settings are below. We also set a jitter
>> buffer in the dialplan.
>>
>> Is there a downside to increasing the packet loss percent to 40-50% other
>> than using more bandwidth? Is there a voice quality impact if you set it
>> too high (what would be considered too high)?
>>
>> Thanks.
>>
>> <param name="use-vbr" value="1"/>
>> <param name="use-dtx" value="1"/>
>> <param name="complexity" value="10"/>
>> <param name="packet-loss-percent" value="30"/>
>> <param name="keep-fec-enabled" value="1"/>
>> <param name="use-jb-lookahead" value="1"/>
>> <param name="maxaveragebitrate" value="40000"/>
>> <param name="maxplaybackrate" value="16000"/>
>> <param name="sprop-maxcapturerate" value="16000"/>
>> <param name="adjust-bitrate" value="1"/>
>> </settings>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users 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
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From mario_fs at mgtech.com  Mon Mar 15 17:28:10 2021
From: mario_fs at mgtech.com (mario_fs)
Date: Mon, 15 Mar 2021 10:28:10 -0700
Subject: [Freeswitch-users] Opus packet loss
In-Reply-To: <CAGvJV4qPNFOM-Z-DWJe4O3F756-Kt3yCOSWK-pD=ktbRcX2vZw@mail.gmail.com>
References: <CA+pHSymbmz8q0TyzCDMa1=TLYD08W5AXUu+pOmjuUebUKJ5V-g@mail.gmail.com>
 <CAEJMVkBA0F8=et3EE5PRZ3KPc9oNnRWf4_S4fz0CmaAG2n=kWQ@mail.gmail.com>
 <CAGvJV4qPNFOM-Z-DWJe4O3F756-Kt3yCOSWK-pD=ktbRcX2vZw@mail.gmail.com>
Message-ID: <D5E2B512-B588-4E07-97C1-90153B95638D@mgtech.com>

Could it be related to this which is outstanding?, I can duplicate it every time:
https://github.com/signalwire/freeswitch/issues/963 <https://github.com/signalwire/freeswitch/issues/963>
Opus warbly, drops sections or unintelligible audio due to gaps in RTP timestamps

> On Mar 13, 2021, at 4:26 AM, Dragos Oancea <dragos at freeswitch.org> wrote:
> 
> When trying to understand how it works make sure you are not mixing decoding side settings with the encoding side settings.
> packet-loss-percent is encoding side and is the initial value for loss if you use adjust-bitrate (callback with SCC_AUDIO_PACKET_LOSS from the core) . 
> The callback comes only if there is incoming RTCP - which for audio is typically at 5 seconds interval.So the adjust-bitrate feature will update the loss too, not only the bitrate. 
>  
> Otherwise, without adjust-birate, it's the default value for OPUS_SET_PACKET_LOSS_PERC() which will stay like this along the call and depending on the preset bitrate it will generate a certain amount of FEC on top of the payload of some of the packets (encoder decisions) .  So no, you should not set that very high, since you can't know the real network conditions before the call is made. 15 or 20 are good values.
> 
> 
> On Fri, Mar 12, 2021 at 9:05 PM Brian West <brian at freeswitch.com <mailto:brian at freeswitch.com>> wrote:
> CPU and bandwidth.
> 
> /b
> 
> 
> On Fri, Mar 12, 2021 at 12:40 PM Terry C <infinite3219 at gmail.com <mailto:infinite3219 at gmail.com>> wrote:
> Hi,
> 
> We use FreeSwitch 1.10.5 with Opus audio for clients. We target voice quality to be wideband. The config settings are below. We also set a jitter buffer in the dialplan.
> 
> Is there a downside to increasing the packet loss percent to 40-50% other than using more bandwidth? Is there a voice quality impact if you set it too high (what would be considered too high)?
> 
> Thanks.
> 
> <param name="use-vbr" value="1"/>
> <param name="use-dtx" value="1"/>
> <param name="complexity" value="10"/>
> <param name="packet-loss-percent" value="30"/>
> <param name="keep-fec-enabled" value="1"/>
> <param name="use-jb-lookahead" value="1"/>
> <param name="maxaveragebitrate" value="40000"/>
> <param name="maxplaybackrate" value="16000"/>
> <param name="sprop-maxcapturerate" value="16000"/>
> <param name="adjust-bitrate" value="1"/>
> </settings>
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>
> 
> -- 
> 
> Brian West | Co-founder and Developer
> Need Commercial support? email sales at freeswitch.com <mailto:sales at freeswitch.com> 
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
> Email: brian at freeswitch.com <mailto:brian at freeswitch.com>
> Mobile: 918-424-9378
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>  <https://www.facebook.com/freeswitch>  <https://twitter.com/freeswitch>
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>_________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From davidswalkabout at gmail.com  Mon Mar 15 22:26:34 2021
From: davidswalkabout at gmail.com (David P)
Date: Tue, 16 Mar 2021 11:26:34 +1300
Subject: [Freeswitch-users] rtp-timeout-sec VS media_timeout
In-Reply-To: <mailman.0.1615636802.20708.freeswitch-users@lists.freeswitch.org>
References: <mailman.0.1615636802.20708.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAE23trtV1mnd2Y5NNj+vAqcSASsRTwMb5tn=xH4CQQ=5fYeOTg@mail.gmail.com>

Although I didn't see that adding...

  media_timeout=300000

...to our conference settings have any effect after restarting FS10.5 this
way...

  fs_cli -x 'fsctl shutdown elegant restart'

...I do see it having an effect now. I suspect there's been a reboot since
then. FYI

On Sun, Mar 14, 2021 at 1:00 AM <
freeswitch-users-request at lists.freeswitch.org> wrote:

>
> ---------- Forwarded message ----------
> From: David P <davidswalkabout at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Sat, 13 Mar 2021 16:15:55 +1300
> Subject: Re: [Freeswitch-users] rtp-timeout-sec VS media_timeout
> Using FS10.5 installed from packages on Debian Stretch...
>
> I wanted to replicate the behavior we had with <param name="rtp-timeout-sec"
> value="300"/> so that we'll hangup calls in which leg B of a conference has
> sent no media for 5 mins.
>
> I edited our dialplan like this...
>
> <action application="conference_set_auto_outcall"
>
>  data="['conference_member_flags=endconf,jitterbuffer_msec=5p:100p,media_timeout=300000']sofia/gateway/...deleted..."/>
>
> and restarted like this...
>
> fs_cli -x 'fsctl shutdown elegant restart'
>
> But after 8 mins in such a call there was no hangup.
>
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From schoch+freeswitch.org at xwin32.com  Tue Mar 16 01:59:40 2021
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Mon, 15 Mar 2021 18:59:40 -0700
Subject: [Freeswitch-users] Faxing with FreeSWITCH
In-Reply-To: <em2dc11070-bd98-43f9-8764-593b49f319cf@asftul-022812>
References: <CA+q4kR+4xqF-Sta+3VaypQmsUe6NhfyHQv79unRnrfYB2q6ciQ@mail.gmail.com>
 <em2dc11070-bd98-43f9-8764-593b49f319cf@asftul-022812>
Message-ID: <CAOkuKdczgFN5-TQt_hRb8UZMbVpDLVGE05ytrYcUE4S90TCcMg@mail.gmail.com>

Hi, Bruce!
Yes, it's been 7 years, but I'm setting up a FreeSWITCH system for a local
non-profit, and part of this is FAX.
I have been using your fsfax script at my company for a long time, and they
just keep working, so I never made any changes.
When I was on your GitHub repository (https://github.com/bwmarrin/fsfax) I
noticed the code hasn't been updated for 6 years. Is this because:

1. It's already perfect?
2. You updated it, but forgot to push?
3. You found something better?

Let me know which, so I can decide which FAX solution to use.

-- 
Steve

On Tue, Mar 25, 2014 at 8:42 AM <bruce at sqls.net> wrote:

>
>
> ------ Original Message ------
> From: "Ali Pey" <alipey at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: 3/25/2014 9:52:41 AM
> Subject: Re: [Freeswitch-users] Faxing with FreeSWITCH
>
>
> Have you done any performance testing? How many concurrent faxes can you
> do?
>
>
> Yes, in Lua you can do transfer to FAX_DETECT extension. It's fairly
> simple. If you choose to go that route, I can help you to set it up.
>
> Try api_hangup_hook for post call processing. It gives you whole lots of
> information. You can also save additional parameters in channel variables
> and access them there in hangup hook.
>
>
> No, I haven't done any performance tests.  I'm not sure how I would do
> that either :).  I'm pretty new to FreeSWITCH and I've never used LUA
> before I started this project so all and all.. I'm mostly clueless.
> There's probably a number of best-practices I just haven't read or learned
> about yet.
>
> Right now I'm just trying to get it to log everything and be as reliable
> as possible :)   I am curious about how to do the extension transfer stuff
> in LUA.  Everything I learn at this point is a big bonus to making this
> work best.
>
> Right now all the post call processing is just happening in the same LUA
> script.  I do call "hangup" after the fax is received though.  Is there a
> big advantage to splitting the script up into two scripts? One for the call
> and another for post call stuff (logging, emailing, etc). ?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From schoch+freeswitch.org at xwin32.com  Tue Mar 16 23:08:46 2021
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Tue, 16 Mar 2021 16:08:46 -0700
Subject: [Freeswitch-users] Will fail2ban work for this?
Message-ID: <CAOkuKdf9dKVjiGSpkP+4_DDh9h61XHFPc2a+e8yX3a-p0nyNSA@mail.gmail.com>

I just set up a new FreeSWITCH system on my home network, and set a forward
for port 5080 to connect to Flowroute. While I'm debugging some call
routing stuff, my logs are getting overrun with stuff like this:

2021-03-16 15:52:02.267501 [NOTICE] switch_channel.c:1118 New Channel
sofia/external/7750@<my IP> [2de89b87-cd07-4c0f-b9fb-3da8e5a68d37]

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585
(sofia/external/7750@<my IP>) Running State Change CS_NEW (Cur 1 Tot 7822)

2021-03-16 15:52:02.267501 [DEBUG] sofia.c:10280 sofia/external/7750@<my
IP> receiving invite from 80.94.93.12:62635 version: 1.10.5
-release-17-25569c1631 64bit

2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7326 Channel sofia/external/7750@<my
IP> entering state [received][100]

2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7336 Remote SDP:

v=0

o=- 81921704 81921704 IN IP4 0.0.0.0

s=pplsip

c=IN IP4 0.0.0.0

t=0 0

m=audio 7628 RTP/AVP 100 6 0 8 3 18 5 101

a=rtpmap:100 speex/16000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-11

a=alt:1 1 : DF50DC48 0000001F 0.0.0.0 7628


2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7739 (sofia/external/7750@<my
IP>) State Change CS_NEW -> CS_INIT

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:604
(sofia/external/7750@<my IP>) State NEW

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585
(sofia/external/7750@<my IP>) Running State Change CS_INIT (Cur 1 Tot 7822)

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:628
(sofia/external/7750@<my IP>) State INIT

2021-03-16 15:52:02.267501 [DEBUG] mod_sofia.c:93 sofia/external/7750@<my
IP> SOFIA INIT

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:40
sofia/external/7750@<my IP> Standard INIT

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:48
(sofia/external/7750@<my IP>) State Change CS_INIT -> CS_ROUTING

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:628
(sofia/external/7750@<my IP>) State INIT going to sleep

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585
(sofia/external/7750@<my IP>) Running State Change CS_ROUTING (Cur 1 Tot
7822)

2021-03-16 15:52:02.267501 [DEBUG] switch_channel.c:2332
(sofia/external/7750@<my IP>) Callstate Change DOWN -> RINGING

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:644
(sofia/external/7750@<my IP>) State ROUTING

2021-03-16 15:52:02.267501 [DEBUG] mod_sofia.c:154 sofia/external/7750@<my
IP> SOFIA ROUTING

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:236
sofia/external/7750@<my IP> Standard ROUTING

2021-03-16 15:52:02.267501 [INFO] mod_dialplan_xml.c:637 Processing 7750
<7750>->900442037697855 in context public


I thought fail2ban was designed for stuff like this, but I don't see any
auth attempts here (I set "log-auth-failures" to "true"). These are coming
in a bit faster than 1 per second. It appears they are dialing random
extensions. How can I make them stop?

-- 
Steve
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From david.villasmil.work at gmail.com  Wed Mar 17 01:29:39 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 17 Mar 2021 01:29:39 +0000
Subject: [Freeswitch-users] Will fail2ban work for this?
In-Reply-To: <CAOkuKdf9dKVjiGSpkP+4_DDh9h61XHFPc2a+e8yX3a-p0nyNSA@mail.gmail.com>
References: <CAOkuKdf9dKVjiGSpkP+4_DDh9h61XHFPc2a+e8yX3a-p0nyNSA@mail.gmail.com>
Message-ID: <CAFGRPVo2x=oRVsKgCKBhHr2B+Wn7+mP5L6bh+2VyLyBQTEzjrQ@mail.gmail.com>

It works, sure. But needs to be configured.

https://freeswitch.org/confluence/display/FREESWITCH/Fail2Ban should help
you, especially the configuration part.

For fail2ban to work, it needs to see a line in the logfile with the
originating IP address, for that to work on failed call attempts you need
to add a specific failure log. Something like adding a catch-all extension
at the very end of the dialplan and log the originating IP. Then grab that
with fail2ban.

something like:

<extension name="catch-all">
  <condition field="destination_number" expression="^.*$">
<condition field="${sip_authorized}" expression="^false$">
     <action application="log" data="CRIT caught
${variable_sip_received_ip} trying to call."/>
     <action application="hangup" data=""/>
</condition>
  </condition>
</extension>

Then a regexp on *filter.d/freeswitch.local*

[Definition]
failregex = ^.* caught <HOST> trying to call$

NOTE: I didn't test any of this, you'll need to test yourself, but it
should be a starting point.


Another option, which i like on top of the already mentioned, is to _not_
use a default port 5080, use something like 9909 (security by obscurity)



Regards,

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


On Tue, Mar 16, 2021 at 11:40 PM Steven Schoch <
schoch+freeswitch.org at xwin32.com> wrote:

> I just set up a new FreeSWITCH system on my home network, and set a
> forward for port 5080 to connect to Flowroute. While I'm debugging some
> call routing stuff, my logs are getting overrun with stuff like this:
>
> 2021-03-16 15:52:02.267501 [NOTICE] switch_channel.c:1118 New Channel
> sofia/external/7750@<my IP> [2de89b87-cd07-4c0f-b9fb-3da8e5a68d37]
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585
> (sofia/external/7750@<my IP>) Running State Change CS_NEW (Cur 1 Tot 7822)
>
> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:10280 sofia/external/7750@<my
> IP> receiving invite from 80.94.93.12:62635 version: 1.10.5
> -release-17-25569c1631 64bit
>
> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7326 Channel
> sofia/external/7750@<my IP> entering state [received][100]
>
> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7336 Remote SDP:
>
> v=0
>
> o=- 81921704 81921704 IN IP4 0.0.0.0
>
> s=pplsip
>
> c=IN IP4 0.0.0.0
>
> t=0 0
>
> m=audio 7628 RTP/AVP 100 6 0 8 3 18 5 101
>
> a=rtpmap:100 speex/16000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-11
>
> a=alt:1 1 : DF50DC48 0000001F 0.0.0.0 7628
>
>
> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7739 (sofia/external/7750@<my
> IP>) State Change CS_NEW -> CS_INIT
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:604
> (sofia/external/7750@<my IP>) State NEW
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585
> (sofia/external/7750@<my IP>) Running State Change CS_INIT (Cur 1 Tot
> 7822)
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:628
> (sofia/external/7750@<my IP>) State INIT
>
> 2021-03-16 15:52:02.267501 [DEBUG] mod_sofia.c:93 sofia/external/7750@<my
> IP> SOFIA INIT
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:40
> sofia/external/7750@<my IP> Standard INIT
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:48
> (sofia/external/7750@<my IP>) State Change CS_INIT -> CS_ROUTING
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:628
> (sofia/external/7750@<my IP>) State INIT going to sleep
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585
> (sofia/external/7750@<my IP>) Running State Change CS_ROUTING (Cur 1 Tot
> 7822)
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_channel.c:2332
> (sofia/external/7750@<my IP>) Callstate Change DOWN -> RINGING
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:644
> (sofia/external/7750@<my IP>) State ROUTING
>
> 2021-03-16 15:52:02.267501 [DEBUG] mod_sofia.c:154 sofia/external/7750@<my
> IP> SOFIA ROUTING
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:236
> sofia/external/7750@<my IP> Standard ROUTING
>
> 2021-03-16 15:52:02.267501 [INFO] mod_dialplan_xml.c:637 Processing 7750
> <7750>->900442037697855 in context public
>
>
> I thought fail2ban was designed for stuff like this, but I don't see any
> auth attempts here (I set "log-auth-failures" to "true"). These are coming
> in a bit faster than 1 per second. It appears they are dialing random
> extensions. How can I make them stop?
>
> --
> Steve
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From schoch+freeswitch.org at xwin32.com  Wed Mar 17 01:59:31 2021
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Tue, 16 Mar 2021 18:59:31 -0700
Subject: [Freeswitch-users] Will fail2ban work for this?
In-Reply-To: <CAFGRPVo2x=oRVsKgCKBhHr2B+Wn7+mP5L6bh+2VyLyBQTEzjrQ@mail.gmail.com>
References: <CAOkuKdf9dKVjiGSpkP+4_DDh9h61XHFPc2a+e8yX3a-p0nyNSA@mail.gmail.com>
 <CAFGRPVo2x=oRVsKgCKBhHr2B+Wn7+mP5L6bh+2VyLyBQTEzjrQ@mail.gmail.com>
Message-ID: <CAOkuKdej=JOv=qtjep5u1k_xNZJDgKTwPe7y5sYhsWVy8LomMg@mail.gmail.com>

I like your 2nd option. I always assumed 5080 was safe because it isn't the
SIP port. It is listed as the "OnScreen Data Collection Service" in the
official port number database (
https://www.iana.org/assignments/service-names-port-numbers/service-names-port-numbers.xhtml?&page=89),
but I guess the hackers know the SIP people like to use it. I'll try
switching to another port.

-- 
Steve

On Tue, Mar 16, 2021 at 6:30 PM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> It works, sure. But needs to be configured.
>
> https://freeswitch.org/confluence/display/FREESWITCH/Fail2Ban should help
> you, especially the configuration part.
>
> For fail2ban to work, it needs to see a line in the logfile with the
> originating IP address, for that to work on failed call attempts you need
> to add a specific failure log. Something like adding a catch-all extension
> at the very end of the dialplan and log the originating IP. Then grab that
> with fail2ban.
>
> something like:
>
> <extension name="catch-all">
>   <condition field="destination_number" expression="^.*$">
> <condition field="${sip_authorized}" expression="^false$">
>      <action application="log" data="CRIT caught
> ${variable_sip_received_ip} trying to call."/>
>      <action application="hangup" data=""/>
> </condition>
>   </condition>
> </extension>
>
> Then a regexp on *filter.d/freeswitch.local*
>
> [Definition]
> failregex = ^.* caught <HOST> trying to call$
>
> NOTE: I didn't test any of this, you'll need to test yourself, but it
> should be a starting point.
>
>
> Another option, which i like on top of the already mentioned, is to _not_
> use a default port 5080, use something like 9909 (security by obscurity)
>
>
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
>
> On Tue, Mar 16, 2021 at 11:40 PM Steven Schoch <
> schoch+freeswitch.org at xwin32.com> wrote:
>
>> I just set up a new FreeSWITCH system on my home network, and set a
>> forward for port 5080 to connect to Flowroute. While I'm debugging some
>> call routing stuff, my logs are getting overrun with stuff like this:
>>
>> 2021-03-16 15:52:02.267501 [NOTICE] switch_channel.c:1118 New Channel
>> sofia/external/7750@<my IP> [2de89b87-cd07-4c0f-b9fb-3da8e5a68d37]
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585
>> (sofia/external/7750@<my IP>) Running State Change CS_NEW (Cur 1 Tot
>> 7822)
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:10280 sofia/external/7750@<my
>> IP> receiving invite from 80.94.93.12:62635 version: 1.10.5
>> -release-17-25569c1631 64bit
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7326 Channel
>> sofia/external/7750@<my IP> entering state [received][100]
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7336 Remote SDP:
>>
>> v=0
>>
>> o=- 81921704 81921704 IN IP4 0.0.0.0
>>
>> s=pplsip
>>
>> c=IN IP4 0.0.0.0
>>
>> t=0 0
>>
>> m=audio 7628 RTP/AVP 100 6 0 8 3 18 5 101
>>
>> a=rtpmap:100 speex/16000
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-11
>>
>> a=alt:1 1 : DF50DC48 0000001F 0.0.0.0 7628
>>
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7739 (sofia/external/7750@<my
>> IP>) State Change CS_NEW -> CS_INIT
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:604
>> (sofia/external/7750@<my IP>) State NEW
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585
>> (sofia/external/7750@<my IP>) Running State Change CS_INIT (Cur 1 Tot
>> 7822)
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:628
>> (sofia/external/7750@<my IP>) State INIT
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] mod_sofia.c:93 sofia/external/7750@<my
>> IP> SOFIA INIT
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:40
>> sofia/external/7750@<my IP> Standard INIT
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:48
>> (sofia/external/7750@<my IP>) State Change CS_INIT -> CS_ROUTING
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:628
>> (sofia/external/7750@<my IP>) State INIT going to sleep
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585
>> (sofia/external/7750@<my IP>) Running State Change CS_ROUTING (Cur 1 Tot
>> 7822)
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_channel.c:2332
>> (sofia/external/7750@<my IP>) Callstate Change DOWN -> RINGING
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:644
>> (sofia/external/7750@<my IP>) State ROUTING
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] mod_sofia.c:154 sofia/external/7750@<my
>> IP> SOFIA ROUTING
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:236
>> sofia/external/7750@<my IP> Standard ROUTING
>>
>> 2021-03-16 15:52:02.267501 [INFO] mod_dialplan_xml.c:637 Processing 7750
>> <7750>->900442037697855 in context public
>>
>>
>> I thought fail2ban was designed for stuff like this, but I don't see any
>> auth attempts here (I set "log-auth-failures" to "true"). These are coming
>> in a bit faster than 1 per second. It appears they are dialing random
>> extensions. How can I make them stop?
>>
>> --
>> Steve
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From mayamatakeshi at gmail.com  Wed Mar 17 02:10:05 2021
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Wed, 17 Mar 2021 11:10:05 +0900
Subject: [Freeswitch-users] Will fail2ban work for this?
In-Reply-To: <CAOkuKdf9dKVjiGSpkP+4_DDh9h61XHFPc2a+e8yX3a-p0nyNSA@mail.gmail.com>
References: <CAOkuKdf9dKVjiGSpkP+4_DDh9h61XHFPc2a+e8yX3a-p0nyNSA@mail.gmail.com>
Message-ID: <CABaNFCZQK9bvLBykwd4-f1fBYipoxqWKFpHom8UUH4Ps5opNDQ@mail.gmail.com>

On Wed, Mar 17, 2021 at 8:37 AM Steven Schoch <
schoch+freeswitch.org at xwin32.com> wrote:

> I just set up a new FreeSWITCH system on my home network, and set a
> forward for port 5080 to connect to Flowroute. While I'm debugging some
> call routing stuff, my logs are getting overrun with stuff like this:
>
> 2021-03-16 15:52:02.267501 [NOTICE] switch_channel.c:1118 New Channel
> sofia/external/7750@<my IP> [2de89b87-cd07-4c0f-b9fb-3da8e5a68d37]
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585
> (sofia/external/7750@<my IP>) Running State Change CS_NEW (Cur 1 Tot 7822)
>
> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:10280 sofia/external/7750@<my
> IP> receiving invite from 80.94.93.12:62635 version: 1.10.5
> -release-17-25569c1631 64bit
>
> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7326 Channel
> sofia/external/7750@<my IP> entering state [received][100]
>
> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7336 Remote SDP:
>
> v=0
>
> o=- 81921704 81921704 IN IP4 0.0.0.0
>
> s=pplsip
>
> c=IN IP4 0.0.0.0
>
> t=0 0
>
> m=audio 7628 RTP/AVP 100 6 0 8 3 18 5 101
>
> a=rtpmap:100 speex/16000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-11
>
> a=alt:1 1 : DF50DC48 0000001F 0.0.0.0 7628
>
>
> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7739 (sofia/external/7750@<my
> IP>) State Change CS_NEW -> CS_INIT
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:604
> (sofia/external/7750@<my IP>) State NEW
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585
> (sofia/external/7750@<my IP>) Running State Change CS_INIT (Cur 1 Tot
> 7822)
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:628
> (sofia/external/7750@<my IP>) State INIT
>
> 2021-03-16 15:52:02.267501 [DEBUG] mod_sofia.c:93 sofia/external/7750@<my
> IP> SOFIA INIT
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:40
> sofia/external/7750@<my IP> Standard INIT
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:48
> (sofia/external/7750@<my IP>) State Change CS_INIT -> CS_ROUTING
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:628
> (sofia/external/7750@<my IP>) State INIT going to sleep
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585
> (sofia/external/7750@<my IP>) Running State Change CS_ROUTING (Cur 1 Tot
> 7822)
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_channel.c:2332
> (sofia/external/7750@<my IP>) Callstate Change DOWN -> RINGING
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:644
> (sofia/external/7750@<my IP>) State ROUTING
>
> 2021-03-16 15:52:02.267501 [DEBUG] mod_sofia.c:154 sofia/external/7750@<my
> IP> SOFIA ROUTING
>
> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:236
> sofia/external/7750@<my IP> Standard ROUTING
>
> 2021-03-16 15:52:02.267501 [INFO] mod_dialplan_xml.c:637 Processing 7750
> <7750>->900442037697855 in context public
>
>
> I thought fail2ban was designed for stuff like this, but I don't see any
> auth attempts here (I set "log-auth-failures" to "true"). These are coming
> in a bit faster than 1 per second. It appears they are dialing random
> extensions. How can I make them stop?
>

I suppose:
  "in context public"
in the above log indicates the call entered your FS without need for
authentication.
So you should switch to a context/profile that requires authentication.,
then log-auth-failures should work.
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From s.safarov at gmail.com  Wed Mar 17 05:30:01 2021
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 17 Mar 2021 08:30:01 +0300
Subject: [Freeswitch-users] Will fail2ban work for this?
In-Reply-To: <CABaNFCZQK9bvLBykwd4-f1fBYipoxqWKFpHom8UUH4Ps5opNDQ@mail.gmail.com>
References: <CAOkuKdf9dKVjiGSpkP+4_DDh9h61XHFPc2a+e8yX3a-p0nyNSA@mail.gmail.com>
 <CABaNFCZQK9bvLBykwd4-f1fBYipoxqWKFpHom8UUH4Ps5opNDQ@mail.gmail.com>
Message-ID: <CAHtxdDfVMNF-otrK19pkggN9nmk0cTPuQO__Di6fzQ_2aaLuow@mail.gmail.com>

mod_failban designed to log auth failure.
So not need to parse all FreeSwitch logs by failban daemon.

Sergey

On Wed, Mar 17, 2021 at 5:48 AM mayamatakeshi <mayamatakeshi at gmail.com>
wrote:

>
>
> On Wed, Mar 17, 2021 at 8:37 AM Steven Schoch <
> schoch+freeswitch.org at xwin32.com> wrote:
>
>> I just set up a new FreeSWITCH system on my home network, and set a
>> forward for port 5080 to connect to Flowroute. While I'm debugging some
>> call routing stuff, my logs are getting overrun with stuff like this:
>>
>> 2021-03-16 15:52:02.267501 [NOTICE] switch_channel.c:1118 New Channel
>> sofia/external/7750@<my IP> [2de89b87-cd07-4c0f-b9fb-3da8e5a68d37]
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585
>> (sofia/external/7750@<my IP>) Running State Change CS_NEW (Cur 1 Tot
>> 7822)
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:10280 sofia/external/7750@<my
>> IP> receiving invite from 80.94.93.12:62635 version: 1.10.5
>> -release-17-25569c1631 64bit
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7326 Channel
>> sofia/external/7750@<my IP> entering state [received][100]
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7336 Remote SDP:
>>
>> v=0
>>
>> o=- 81921704 81921704 IN IP4 0.0.0.0
>>
>> s=pplsip
>>
>> c=IN IP4 0.0.0.0
>>
>> t=0 0
>>
>> m=audio 7628 RTP/AVP 100 6 0 8 3 18 5 101
>>
>> a=rtpmap:100 speex/16000
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-11
>>
>> a=alt:1 1 : DF50DC48 0000001F 0.0.0.0 7628
>>
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7739 (sofia/external/7750@<my
>> IP>) State Change CS_NEW -> CS_INIT
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:604
>> (sofia/external/7750@<my IP>) State NEW
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585
>> (sofia/external/7750@<my IP>) Running State Change CS_INIT (Cur 1 Tot
>> 7822)
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:628
>> (sofia/external/7750@<my IP>) State INIT
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] mod_sofia.c:93 sofia/external/7750@<my
>> IP> SOFIA INIT
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:40
>> sofia/external/7750@<my IP> Standard INIT
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:48
>> (sofia/external/7750@<my IP>) State Change CS_INIT -> CS_ROUTING
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:628
>> (sofia/external/7750@<my IP>) State INIT going to sleep
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585
>> (sofia/external/7750@<my IP>) Running State Change CS_ROUTING (Cur 1 Tot
>> 7822)
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_channel.c:2332
>> (sofia/external/7750@<my IP>) Callstate Change DOWN -> RINGING
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:644
>> (sofia/external/7750@<my IP>) State ROUTING
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] mod_sofia.c:154 sofia/external/7750@<my
>> IP> SOFIA ROUTING
>>
>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:236
>> sofia/external/7750@<my IP> Standard ROUTING
>>
>> 2021-03-16 15:52:02.267501 [INFO] mod_dialplan_xml.c:637 Processing 7750
>> <7750>->900442037697855 in context public
>>
>>
>> I thought fail2ban was designed for stuff like this, but I don't see any
>> auth attempts here (I set "log-auth-failures" to "true"). These are coming
>> in a bit faster than 1 per second. It appears they are dialing random
>> extensions. How can I make them stop?
>>
>
> I suppose:
>   "in context public"
> in the above log indicates the call entered your FS without need for
> authentication.
> So you should switch to a context/profile that requires authentication.,
> then log-auth-failures should work.
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From rbetancor at gmail.com  Wed Mar 17 06:58:32 2021
From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=)
Date: Wed, 17 Mar 2021 06:58:32 +0000
Subject: [Freeswitch-users] Will fail2ban work for this?
In-Reply-To: <CAOkuKdej=JOv=qtjep5u1k_xNZJDgKTwPe7y5sYhsWVy8LomMg@mail.gmail.com>
References: <CAOkuKdf9dKVjiGSpkP+4_DDh9h61XHFPc2a+e8yX3a-p0nyNSA@mail.gmail.com>
 <CAFGRPVo2x=oRVsKgCKBhHr2B+Wn7+mP5L6bh+2VyLyBQTEzjrQ@mail.gmail.com>
 <CAOkuKdej=JOv=qtjep5u1k_xNZJDgKTwPe7y5sYhsWVy8LomMg@mail.gmail.com>
Message-ID: <CAKsfQBjk9YgAaZXq=52vZoD0kBWzctikt3qhXYDKFMGODuv0cQ@mail.gmail.com>

Switching SIP port, is not the solution, sooner than later, they will find
you.

The best approach is to use a combination of solutions, like a blacklist of
know hackers IPs as voipbl.org, correctly setup fail2ban, put your FS
behind a Kamailio with the pike module and other security measures, etc.

On Wed, Mar 17, 2021 at 2:19 AM Steven Schoch <
schoch+freeswitch.org at xwin32.com> wrote:

> I like your 2nd option. I always assumed 5080 was safe because it isn't
> the SIP port. It is listed as the "OnScreen Data Collection Service" in the
> official port number database (
> https://www.iana.org/assignments/service-names-port-numbers/service-names-port-numbers.xhtml?&page=89),
> but I guess the hackers know the SIP people like to use it. I'll try
> switching to another port.
>
> --
> Steve
>
> On Tue, Mar 16, 2021 at 6:30 PM David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> It works, sure. But needs to be configured.
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/Fail2Ban should
>> help you, especially the configuration part.
>>
>> For fail2ban to work, it needs to see a line in the logfile with the
>> originating IP address, for that to work on failed call attempts you need
>> to add a specific failure log. Something like adding a catch-all extension
>> at the very end of the dialplan and log the originating IP. Then grab that
>> with fail2ban.
>>
>> something like:
>>
>> <extension name="catch-all">
>>   <condition field="destination_number" expression="^.*$">
>> <condition field="${sip_authorized}" expression="^false$">
>>      <action application="log" data="CRIT caught
>> ${variable_sip_received_ip} trying to call."/>
>>      <action application="hangup" data=""/>
>> </condition>
>>   </condition>
>> </extension>
>>
>> Then a regexp on *filter.d/freeswitch.local*
>>
>> [Definition]
>> failregex = ^.* caught <HOST> trying to call$
>>
>> NOTE: I didn't test any of this, you'll need to test yourself, but it
>> should be a starting point.
>>
>>
>> Another option, which i like on top of the already mentioned, is to _not_
>> use a default port 5080, use something like 9909 (security by obscurity)
>>
>>
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>>
>>
>> On Tue, Mar 16, 2021 at 11:40 PM Steven Schoch <
>> schoch+freeswitch.org at xwin32.com> wrote:
>>
>>> I just set up a new FreeSWITCH system on my home network, and set a
>>> forward for port 5080 to connect to Flowroute. While I'm debugging some
>>> call routing stuff, my logs are getting overrun with stuff like this:
>>>
>>> 2021-03-16 15:52:02.267501 [NOTICE] switch_channel.c:1118 New Channel
>>> sofia/external/7750@<my IP> [2de89b87-cd07-4c0f-b9fb-3da8e5a68d37]
>>>
>>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585
>>> (sofia/external/7750@<my IP>) Running State Change CS_NEW (Cur 1 Tot
>>> 7822)
>>>
>>> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:10280 sofia/external/7750@<my
>>> IP> receiving invite from 80.94.93.12:62635 version: 1.10.5
>>> -release-17-25569c1631 64bit
>>>
>>> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7326 Channel
>>> sofia/external/7750@<my IP> entering state [received][100]
>>>
>>> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7336 Remote SDP:
>>>
>>> v=0
>>>
>>> o=- 81921704 81921704 IN IP4 0.0.0.0
>>>
>>> s=pplsip
>>>
>>> c=IN IP4 0.0.0.0
>>>
>>> t=0 0
>>>
>>> m=audio 7628 RTP/AVP 100 6 0 8 3 18 5 101
>>>
>>> a=rtpmap:100 speex/16000
>>>
>>> a=rtpmap:101 telephone-event/8000
>>>
>>> a=fmtp:101 0-11
>>>
>>> a=alt:1 1 : DF50DC48 0000001F 0.0.0.0 7628
>>>
>>>
>>> 2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7739 (sofia/external/7750@<my
>>> IP>) State Change CS_NEW -> CS_INIT
>>>
>>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:604
>>> (sofia/external/7750@<my IP>) State NEW
>>>
>>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585
>>> (sofia/external/7750@<my IP>) Running State Change CS_INIT (Cur 1 Tot
>>> 7822)
>>>
>>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:628
>>> (sofia/external/7750@<my IP>) State INIT
>>>
>>> 2021-03-16 15:52:02.267501 [DEBUG] mod_sofia.c:93 sofia/external/7750@<my
>>> IP> SOFIA INIT
>>>
>>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:40
>>> sofia/external/7750@<my IP> Standard INIT
>>>
>>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:48
>>> (sofia/external/7750@<my IP>) State Change CS_INIT -> CS_ROUTING
>>>
>>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:628
>>> (sofia/external/7750@<my IP>) State INIT going to sleep
>>>
>>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585
>>> (sofia/external/7750@<my IP>) Running State Change CS_ROUTING (Cur 1
>>> Tot 7822)
>>>
>>> 2021-03-16 15:52:02.267501 [DEBUG] switch_channel.c:2332
>>> (sofia/external/7750@<my IP>) Callstate Change DOWN -> RINGING
>>>
>>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:644
>>> (sofia/external/7750@<my IP>) State ROUTING
>>>
>>> 2021-03-16 15:52:02.267501 [DEBUG] mod_sofia.c:154 sofia/external/7750@<my
>>> IP> SOFIA ROUTING
>>>
>>> 2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:236
>>> sofia/external/7750@<my IP> Standard ROUTING
>>>
>>> 2021-03-16 15:52:02.267501 [INFO] mod_dialplan_xml.c:637 Processing 7750
>>> <7750>->900442037697855 in context public
>>>
>>>
>>> I thought fail2ban was designed for stuff like this, but I don't see any
>>> auth attempts here (I set "log-auth-failures" to "true"). These are coming
>>> in a bit faster than 1 per second. It appears they are dialing random
>>> extensions. How can I make them stop?
>>>
>>> --
>>> Steve
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From mayamatakeshi at gmail.com  Wed Mar 17 09:52:39 2021
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Wed, 17 Mar 2021 18:52:39 +0900
Subject: [Freeswitch-users] Stopping TTS when play_and_detect_speech gets
	MRCP START-OF-INPUT
Message-ID: <CABaNFCZy9mtj_95aX=u+iMqZR_vvXQz9T9Vm_mFMnoZZoUPm9Q@mail.gmail.com>

Hi,
I'm trying play_and_detect_speech with an ESL app but START-OF-INPUT
doesn't interrupt the TTS being played.
Is this expected?
(
https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+play_and_detect_speech
is not clear about it)
Or maybe I need to set some channel variable for this to work.
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From martin.paterson at technologywithin.com  Wed Mar 17 15:42:06 2021
From: martin.paterson at technologywithin.com (Martin Paterson)
Date: Wed, 17 Mar 2021 15:42:06 +0000
Subject: [Freeswitch-users] Will fail2ban work for this?
In-Reply-To: <CAKsfQBjk9YgAaZXq=52vZoD0kBWzctikt3qhXYDKFMGODuv0cQ@mail.gmail.com>
References: <CAOkuKdf9dKVjiGSpkP+4_DDh9h61XHFPc2a+e8yX3a-p0nyNSA@mail.gmail.com>
 <CAFGRPVo2x=oRVsKgCKBhHr2B+Wn7+mP5L6bh+2VyLyBQTEzjrQ@mail.gmail.com>
 <CAOkuKdej=JOv=qtjep5u1k_xNZJDgKTwPe7y5sYhsWVy8LomMg@mail.gmail.com>
 <CAKsfQBjk9YgAaZXq=52vZoD0kBWzctikt3qhXYDKFMGODuv0cQ@mail.gmail.com>
Message-ID: <VI1PR10MB20944C0CBAD4C4843ECBB915956A9@VI1PR10MB2094.EURPRD10.PROD.OUTLOOK.COM>

APIBAN is also good for this (https://www.apiban.org/doc.html). It basically sends you a list of known bad IP addresses and modifies your firewall to block them, it’s really easy to install and get running. I found out about it at a ClueCon talk (this one: https://youtu.be/JvUGU3YtgzE?t=3132). The rest of Fred’s talk is also interesting and touches on security.

Martin.


Martin Paterson
Development Team
Phone: 0207 953 8840
Email: martin.paterson at technologywithin.com
Chevron Business Park, Limekiln Lane, Southampton, Hampshire, SO45 2QL
Registered Office: CP House, Otterspool Way, Watford, WD25 8JJ, U.K 
​Registered in England No: 5964349 | VAT Number: GB 902 5369 37 
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Raúl Alexis Betancor Santana
Sent: 17 March 2021 06:59
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Will fail2ban work for this?

Switching SIP port, is not the solution, sooner than later, they will find you.

The best approach is to use a combination of solutions, like a blacklist of know hackers IPs as voipbl.org<http://voipbl.org>, correctly setup fail2ban, put your FS behind a Kamailio with the pike module and other security measures, etc.

On Wed, Mar 17, 2021 at 2:19 AM Steven Schoch <schoch+freeswitch.org at xwin32.com<mailto:schoch%2Bfreeswitch.org at xwin32.com>> wrote:
I like your 2nd option. I always assumed 5080 was safe because it isn't the SIP port. It is listed as the "OnScreen Data Collection Service" in the official port number database (https://www.iana.org/assignments/service-names-port-numbers/service-names-port-numbers.xhtml?&page=89), but I guess the hackers know the SIP people like to use it. I'll try switching to another port.

--
Steve

On Tue, Mar 16, 2021 at 6:30 PM David Villasmil <david.villasmil.work at gmail.com<mailto:david.villasmil.work at gmail.com>> wrote:
It works, sure. But needs to be configured.

https://freeswitch.org/confluence/display/FREESWITCH/Fail2Ban should help you, especially the configuration part.

For fail2ban to work, it needs to see a line in the logfile with the originating IP address, for that to work on failed call attempts you need to add a specific failure log. Something like adding a catch-all extension at the very end of the dialplan and log the originating IP. Then grab that with fail2ban.

something like:

<extension name="catch-all">
  <condition field="destination_number" expression="^.*$">
<condition field="${sip_authorized}" expression="^false$">
     <action application="log" data="CRIT caught ${variable_sip_received_ip} trying to call."/>
     <action application="hangup" data=""/>
</condition>
  </condition>
</extension>

Then a regexp on filter.d/freeswitch.local

[Definition]
failregex = ^.* caught <HOST> trying to call$

NOTE: I didn't test any of this, you'll need to test yourself, but it should be a starting point.


Another option, which i like on top of the already mentioned, is to _not_ use a default port 5080, use something like 9909 (security by obscurity)



Regards,

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


On Tue, Mar 16, 2021 at 11:40 PM Steven Schoch <schoch+freeswitch.org at xwin32.com<mailto:schoch%2Bfreeswitch.org at xwin32.com>> wrote:
I just set up a new FreeSWITCH system on my home network, and set a forward for port 5080 to connect to Flowroute. While I'm debugging some call routing stuff, my logs are getting overrun with stuff like this:


2021-03-16 15:52:02.267501 [NOTICE] switch_channel.c:1118 New Channel sofia/external/7750@<my IP> [2de89b87-cd07-4c0f-b9fb-3da8e5a68d37]

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585 (sofia/external/7750@<my IP>) Running State Change CS_NEW (Cur 1 Tot 7822)

2021-03-16 15:52:02.267501 [DEBUG] sofia.c:10280 sofia/external/7750@<my IP> receiving invite from 80.94.93.12:62635<http://80.94.93.12:62635> version: 1.10.5 -release-17-25569c1631 64bit

2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7326 Channel sofia/external/7750@<my IP> entering state [received][100]

2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7336 Remote SDP:

v=0

o=- 81921704 81921704 IN IP4 0.0.0.0

s=pplsip

c=IN IP4 0.0.0.0

t=0 0

m=audio 7628 RTP/AVP 100 6 0 8 3 18 5 101

a=rtpmap:100 speex/16000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-11

a=alt:1 1 : DF50DC48 0000001F 0.0.0.0 7628



2021-03-16 15:52:02.267501 [DEBUG] sofia.c:7739 (sofia/external/7750@<my IP>) State Change CS_NEW -> CS_INIT

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:604 (sofia/external/7750@<my IP>) State NEW

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585 (sofia/external/7750@<my IP>) Running State Change CS_INIT (Cur 1 Tot 7822)

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:628 (sofia/external/7750@<my IP>) State INIT

2021-03-16 15:52:02.267501 [DEBUG] mod_sofia.c:93 sofia/external/7750@<my IP> SOFIA INIT

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:40 sofia/external/7750@<my IP> Standard INIT

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:48 (sofia/external/7750@<my IP>) State Change CS_INIT -> CS_ROUTING

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:628 (sofia/external/7750@<my IP>) State INIT going to sleep

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:585 (sofia/external/7750@<my IP>) Running State Change CS_ROUTING (Cur 1 Tot 7822)

2021-03-16 15:52:02.267501 [DEBUG] switch_channel.c:2332 (sofia/external/7750@<my IP>) Callstate Change DOWN -> RINGING

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:644 (sofia/external/7750@<my IP>) State ROUTING

2021-03-16 15:52:02.267501 [DEBUG] mod_sofia.c:154 sofia/external/7750@<my IP> SOFIA ROUTING

2021-03-16 15:52:02.267501 [DEBUG] switch_core_state_machine.c:236 sofia/external/7750@<my IP> Standard ROUTING

2021-03-16 15:52:02.267501 [INFO] mod_dialplan_xml.c:637 Processing 7750 <7750>->900442037697855 in context public


I thought fail2ban was designed for stuff like this, but I don't see any auth attempts here (I set "log-auth-failures" to "true"). These are coming in a bit faster than 1 per second. It appears they are dialing random extensions. How can I make them stop?

--
Steve
_________________________________________________________________________

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

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

Professional FreeSWITCH Services
sales at freeswitch.com<mailto:sales at freeswitch.com>
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
_________________________________________________________________________

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

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

Professional FreeSWITCH Services
sales at freeswitch.com<mailto:sales at freeswitch.com>
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product 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
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From mayamatakeshi at gmail.com  Thu Mar 18 03:23:19 2021
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 18 Mar 2021 12:23:19 +0900
Subject: [Freeswitch-users] Stopping TTS when play_and_detect_speech
	gets MRCP START-OF-INPUT
In-Reply-To: <CABaNFCZy9mtj_95aX=u+iMqZR_vvXQz9T9Vm_mFMnoZZoUPm9Q@mail.gmail.com>
References: <CABaNFCZy9mtj_95aX=u+iMqZR_vvXQz9T9Vm_mFMnoZZoUPm9Q@mail.gmail.com>
Message-ID: <CABaNFCYxUj1M6FtyaCRLFR9hJM87O6pfQF8v03nPPBKtyQzxDg@mail.gmail.com>

On Wed, Mar 17, 2021 at 6:52 PM mayamatakeshi <mayamatakeshi at gmail.com>
wrote:

> Hi,
> I'm trying play_and_detect_speech with an ESL app but START-OF-INPUT
> doesn't interrupt the TTS being played.
> Is this expected?
> (
> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+play_and_detect_speech
> is not clear about it)
> Or maybe I need to set some channel variable for this to work.
>

Hi,
My app is a fork of https://github.com/plivo/plivoframework
I debugged FS code and found the cause of the problem:
I verified that the DETECTED_SPEECH events (including the one
with Speech-Type begin-speaking) were not being fired.
This was because my app was not setting this variable:
  https://freeswitch.org/confluence/display/FREESWITCH/fire_asr_events
Then I set fire_asr_events=true
but still the prompt didn't get interrupted by speech.
Then I checked how FS code was fetching (dequeuing) events and I realized
it checks for divert_events.
Then I changed my app to send
  divert_events off
in the ESL socket
and after that it worked and FS stopped the prompt upon speech start.
However, it is not stopping the prompt when a DTMF digit is received (it
was already this way before I did the changes).
So things are better but still there is something amiss.
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From brian at freeswitch.com  Thu Mar 18 03:46:54 2021
From: brian at freeswitch.com (Brian West)
Date: Wed, 17 Mar 2021 22:46:54 -0500
Subject: [Freeswitch-users] rtp-timeout-sec VS media_timeout
In-Reply-To: <CAE23trtV1mnd2Y5NNj+vAqcSASsRTwMb5tn=xH4CQQ=5fYeOTg@mail.gmail.com>
References: <mailman.0.1615636802.20708.freeswitch-users@lists.freeswitch.org>
 <CAE23trtV1mnd2Y5NNj+vAqcSASsRTwMb5tn=xH4CQQ=5fYeOTg@mail.gmail.com>
Message-ID: <CAEJMVkC0-ftpATgVNHSKfDAs=2tRyORQinocRz=c35Pfz-9DPg@mail.gmail.com>

Are you trying to hang the call up?

On Mon, Mar 15, 2021 at 5:45 PM David P <davidswalkabout at gmail.com> wrote:

> Although I didn't see that adding...
>
>   media_timeout=300000
>
> ...to our conference settings have any effect after restarting FS10.5 this
> way...
>
>   fs_cli -x 'fsctl shutdown elegant restart'
>
> ...I do see it having an effect now. I suspect there's been a reboot since
> then. FYI
>
> On Sun, Mar 14, 2021 at 1:00 AM <
> freeswitch-users-request at lists.freeswitch.org> wrote:
>
>>
>> ---------- Forwarded message ----------
>> From: David P <davidswalkabout at gmail.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Bcc:
>> Date: Sat, 13 Mar 2021 16:15:55 +1300
>> Subject: Re: [Freeswitch-users] rtp-timeout-sec VS media_timeout
>> Using FS10.5 installed from packages on Debian Stretch...
>>
>> I wanted to replicate the behavior we had with <param name="rtp-timeout-sec"
>> value="300"/> so that we'll hangup calls in which leg B of a conference has
>> sent no media for 5 mins.
>>
>> I edited our dialplan like this...
>>
>> <action application="conference_set_auto_outcall"
>>
>>  data="['conference_member_flags=endconf,jitterbuffer_msec=5p:100p,media_timeout=300000']sofia/gateway/...deleted..."/>
>>
>> and restarted like this...
>>
>> fs_cli -x 'fsctl shutdown elegant restart'
>>
>> But after 8 mins in such a call there was no hangup.
>>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users 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>
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From mayamatakeshi at gmail.com  Thu Mar 18 06:59:59 2021
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 18 Mar 2021 15:59:59 +0900
Subject: [Freeswitch-users] Stopping TTS when play_and_detect_speech
	gets MRCP START-OF-INPUT
In-Reply-To: <CABaNFCYxUj1M6FtyaCRLFR9hJM87O6pfQF8v03nPPBKtyQzxDg@mail.gmail.com>
References: <CABaNFCZy9mtj_95aX=u+iMqZR_vvXQz9T9Vm_mFMnoZZoUPm9Q@mail.gmail.com>
 <CABaNFCYxUj1M6FtyaCRLFR9hJM87O6pfQF8v03nPPBKtyQzxDg@mail.gmail.com>
Message-ID: <CABaNFCb9durk0tjXN1AWg7imrCHzig5QmrFNTpr8x2PaW4M6hw@mail.gmail.com>

On Thu, Mar 18, 2021 at 12:23 PM mayamatakeshi <mayamatakeshi at gmail.com>
wrote:

>
>
> On Wed, Mar 17, 2021 at 6:52 PM mayamatakeshi <mayamatakeshi at gmail.com>
> wrote:
>
>> Hi,
>> I'm trying play_and_detect_speech with an ESL app but START-OF-INPUT
>> doesn't interrupt the TTS being played.
>> Is this expected?
>> (
>> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+play_and_detect_speech
>> is not clear about it)
>> Or maybe I need to set some channel variable for this to work.
>>
>
> Hi,
> My app is a fork of https://github.com/plivo/plivoframework
> I debugged FS code and found the cause of the problem:
> I verified that the DETECTED_SPEECH events (including the one
> with Speech-Type begin-speaking) were not being fired.
> This was because my app was not setting this variable:
>   https://freeswitch.org/confluence/display/FREESWITCH/fire_asr_events
> Then I set fire_asr_events=true
> but still the prompt didn't get interrupted by speech.
> Then I checked how FS code was fetching (dequeuing) events and I realized
> it checks for divert_events.
> Then I changed my app to send
>   divert_events off
> in the ESL socket
> and after that it worked and FS stopped the prompt upon speech start.
> However, it is not stopping the prompt when a DTMF digit is received (it
> was already this way before I did the changes).
> So things are better but still there is something amiss.
>

OK. I found the reason:
I was using 'playback_terminators=any'
which reading the FS code shows will result in:
  terminators = "1234567890*#"
But I was sending DTMF 'a' so it was not causing the prompt to be
terminated.
Then I changed my test script to send '1' instead and then this caused the
prompt to terminate.
However, this also terminated the speech detection operation as FS sent
STOP to the MRCP server.
And the reason is because differently from begin-speaking, a terminator
will set the operation as done (switch_ivr_async.c):

                      } else if (!strcasecmp(speech_type,
"begin-speaking")) {

switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO,
"(%s) START OF SPEECH\n", switch_channel_get_name(channel));
                          return SWITCH_STATUS_BREAK;
                      }

                      if (terminators && strchr(terminators, dtmf->digit)) {

switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
"(%s) ACCEPT TERMINATOR %c\n", switch_channel_get_name(channel),
dtmf->digit);
                          switch_channel_set_variable_printf(channel,
SWITCH_PLAYBACK_TERMINATOR_USED, "%c", dtmf->digit);
                          state->result =
switch_core_session_sprintf(session, "DIGIT: %c", dtmf->digit);
                          state->done = PLAY_AND_DETECT_DONE;
                          return SWITCH_STATUS_BREAK;
                      }

This might be OK for someone but in my case  there is another issue that
the termination of the speech detection is not notified to my app via ESL.

I am hoping to solve this by leaving the DTMF collection to be done by the
MRCP server but currently the MRCP server I'm using (UniMRCP) doesn't
report START-OF-INPUT for DTMF (I only get the RECOGNITION-COMPLETE after
all digits are input). (my use case is to allow the user to speak some
number sequence or dial it).

So I will switch to use detect_speech instead of play_and_detect_speech.
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From mayamatakeshi at gmail.com  Thu Mar 18 07:46:41 2021
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 18 Mar 2021 16:46:41 +0900
Subject: [Freeswitch-users] Stopping TTS when play_and_detect_speech
	gets MRCP START-OF-INPUT
In-Reply-To: <CABaNFCb9durk0tjXN1AWg7imrCHzig5QmrFNTpr8x2PaW4M6hw@mail.gmail.com>
References: <CABaNFCZy9mtj_95aX=u+iMqZR_vvXQz9T9Vm_mFMnoZZoUPm9Q@mail.gmail.com>
 <CABaNFCYxUj1M6FtyaCRLFR9hJM87O6pfQF8v03nPPBKtyQzxDg@mail.gmail.com>
 <CABaNFCb9durk0tjXN1AWg7imrCHzig5QmrFNTpr8x2PaW4M6hw@mail.gmail.com>
Message-ID: <CABaNFCaAd7b3123a_ce0Zf337cSvRK6tBqww6E+5c9SRFGjaNA@mail.gmail.com>

On Thu, Mar 18, 2021 at 3:59 PM mayamatakeshi <mayamatakeshi at gmail.com>
wrote:

>
>
> On Thu, Mar 18, 2021 at 12:23 PM mayamatakeshi <mayamatakeshi at gmail.com>
> wrote:
>
>>
>>
>> On Wed, Mar 17, 2021 at 6:52 PM mayamatakeshi <mayamatakeshi at gmail.com>
>> wrote:
>>
>>> Hi,
>>> I'm trying play_and_detect_speech with an ESL app but START-OF-INPUT
>>> doesn't interrupt the TTS being played.
>>> Is this expected?
>>> (
>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+play_and_detect_speech
>>> is not clear about it)
>>> Or maybe I need to set some channel variable for this to work.
>>>
>>
>> Hi,
>> My app is a fork of https://github.com/plivo/plivoframework
>> I debugged FS code and found the cause of the problem:
>> I verified that the DETECTED_SPEECH events (including the one
>> with Speech-Type begin-speaking) were not being fired.
>> This was because my app was not setting this variable:
>>   https://freeswitch.org/confluence/display/FREESWITCH/fire_asr_events
>> Then I set fire_asr_events=true
>> but still the prompt didn't get interrupted by speech.
>> Then I checked how FS code was fetching (dequeuing) events and I realized
>> it checks for divert_events.
>> Then I changed my app to send
>>   divert_events off
>> in the ESL socket
>> and after that it worked and FS stopped the prompt upon speech start.
>> However, it is not stopping the prompt when a DTMF digit is received (it
>> was already this way before I did the changes).
>> So things are better but still there is something amiss.
>>
>
> OK. I found the reason:
> I was using 'playback_terminators=any'
> which reading the FS code shows will result in:
>   terminators = "1234567890*#"
> But I was sending DTMF 'a' so it was not causing the prompt to be
> terminated.
> Then I changed my test script to send '1' instead and then this caused the
> prompt to terminate.
> However, this also terminated the speech detection operation as FS sent
> STOP to the MRCP server.
> And the reason is because differently from begin-speaking, a terminator
> will set the operation as done (switch_ivr_async.c):
>
>                       } else if (!strcasecmp(speech_type,
> "begin-speaking")) {
>
> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO,
> "(%s) START OF SPEECH\n", switch_channel_get_name(channel));
>                           return SWITCH_STATUS_BREAK;
>                       }
>
>                       if (terminators && strchr(terminators, dtmf->digit))
> {
>
> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
> "(%s) ACCEPT TERMINATOR %c\n", switch_channel_get_name(channel),
> dtmf->digit);
>                           switch_channel_set_variable_printf(channel,
> SWITCH_PLAYBACK_TERMINATOR_USED, "%c", dtmf->digit);
>                           state->result =
> switch_core_session_sprintf(session, "DIGIT: %c", dtmf->digit);
>                           state->done = PLAY_AND_DETECT_DONE;
>                           return SWITCH_STATUS_BREAK;
>                       }
>
> This might be OK for someone but in my case  there is another issue that
> the termination of the speech detection is not notified to my app via ESL.
>
> I am hoping to solve this by leaving the DTMF collection to be done by the
> MRCP server but currently the MRCP server I'm using (UniMRCP) doesn't
> report START-OF-INPUT for DTMF (I only get the RECOGNITION-COMPLETE after
> all digits are input). (my use case is to allow the user to speak some
> number sequence or dial it).
>

Sorry,
checking again, there is some specific condition that causes UniMRCP to not
send the START-OF-INPUT.
I'm not sure what it is yet but it may be by design or following something
in the protocol that I am not aware of.


>
> So I will switch to use detect_speech instead of play_and_detect_speech.
>
>
>
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From lewis.bergman at gmail.com  Thu Mar 18 14:03:08 2021
From: lewis.bergman at gmail.com (Lewis Bergman)
Date: Thu, 18 Mar 2021 09:03:08 -0500
Subject: [Freeswitch-users] Stopping TTS when play_and_detect_speech
 gets MRCP START-OF-INPUT
In-Reply-To: <CABaNFCaAd7b3123a_ce0Zf337cSvRK6tBqww6E+5c9SRFGjaNA@mail.gmail.com>
References: <CABaNFCZy9mtj_95aX=u+iMqZR_vvXQz9T9Vm_mFMnoZZoUPm9Q@mail.gmail.com>
 <CABaNFCYxUj1M6FtyaCRLFR9hJM87O6pfQF8v03nPPBKtyQzxDg@mail.gmail.com>
 <CABaNFCb9durk0tjXN1AWg7imrCHzig5QmrFNTpr8x2PaW4M6hw@mail.gmail.com>
 <CABaNFCaAd7b3123a_ce0Zf337cSvRK6tBqww6E+5c9SRFGjaNA@mail.gmail.com>
Message-ID: <CAD2CnaQhUGYXyTEs_2o+EuDpmj8sfsRoRw7Zo_h_dFWMtoQZog@mail.gmail.com>

Arsen is very good about responding on the unimrcp list.
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From davidswalkabout at gmail.com  Fri Mar 19 06:54:02 2021
From: davidswalkabout at gmail.com (David P)
Date: Fri, 19 Mar 2021 19:54:02 +1300
Subject: [Freeswitch-users] rtp-timeout-sec VS media_timeout
In-Reply-To: <mailman.1763.1616050841.1264.freeswitch-users@lists.freeswitch.org>
References: <mailman.1763.1616050841.1264.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAE23trsCYcf_eDfAPods92DaaXJdDHPBGoa4Q1JMR0Nq2-sd8A@mail.gmail.com>

>
> Yes, we want to hangup if media_timeout expires on leg b of the
conference. This seems to work but seemingly only after reboot, not the
'elegant' restart I tried initially.

Btw, still eager for any news on 'unified-plan' support.

---------- Forwarded message ----------
>
From: Brian West <brian at freeswitch.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Wed, 17 Mar 2021 22:46:54 -0500
> Subject: Re: [Freeswitch-users] rtp-timeout-sec VS media_timeout
> Are you trying to hang the call up?
>
> On Mon, Mar 15, 2021 at 5:45 PM David P <davidswalkabout at gmail.com> wrote:
>
>> Although I didn't see that adding...
>>
>>   media_timeout=300000
>>
>> ...to our conference settings have any effect after restarting FS10.5
>> this way...
>>
>>   fs_cli -x 'fsctl shutdown elegant restart'
>>
>> ...I do see it having an effect now. I suspect there's been a reboot
>> since then. FYI
>>
>> On Sun, Mar 14, 2021 at 1:00 AM <
>> freeswitch-users-request at lists.freeswitch.org> wrote:
>>
>>>
>>> ---------- Forwarded message ----------
>>> From: David P <davidswalkabout at gmail.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Cc:
>>> Bcc:
>>> Date: Sat, 13 Mar 2021 16:15:55 +1300
>>> Subject: Re: [Freeswitch-users] rtp-timeout-sec VS media_timeout
>>> Using FS10.5 installed from packages on Debian Stretch...
>>>
>>> I wanted to replicate the behavior we had with <param name="rtp-timeout-sec"
>>> value="300"/> so that we'll hangup calls in which leg B of a conference has
>>> sent no media for 5 mins.
>>>
>>> I edited our dialplan like this...
>>>
>>> <action application="conference_set_auto_outcall"
>>>
>>>  data="['conference_member_flags=endconf,jitterbuffer_msec=5p:100p,media_timeout=300000']sofia/gateway/...deleted..."/>
>>>
>>> and restarted like this...
>>>
>>> fs_cli -x 'fsctl shutdown elegant restart'
>>>
>>> But after 8 mins in such a call there was no hangup.
>>>
>>
>>
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From mayamatakeshi at gmail.com  Sun Mar 21 06:31:07 2021
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Sun, 21 Mar 2021 15:31:07 +0900
Subject: [Freeswitch-users] Stopping TTS when play_and_detect_speech
	gets MRCP START-OF-INPUT
In-Reply-To: <CABaNFCaAd7b3123a_ce0Zf337cSvRK6tBqww6E+5c9SRFGjaNA@mail.gmail.com>
References: <CABaNFCZy9mtj_95aX=u+iMqZR_vvXQz9T9Vm_mFMnoZZoUPm9Q@mail.gmail.com>
 <CABaNFCYxUj1M6FtyaCRLFR9hJM87O6pfQF8v03nPPBKtyQzxDg@mail.gmail.com>
 <CABaNFCb9durk0tjXN1AWg7imrCHzig5QmrFNTpr8x2PaW4M6hw@mail.gmail.com>
 <CABaNFCaAd7b3123a_ce0Zf337cSvRK6tBqww6E+5c9SRFGjaNA@mail.gmail.com>
Message-ID: <CABaNFCYcsZ91v8DfZh8pMXAJEn=Q32i_aAVsS8RZGp5-0tfjfw@mail.gmail.com>

On Thu, Mar 18, 2021 at 4:46 PM mayamatakeshi <mayamatakeshi at gmail.com>
wrote:

>
>
> On Thu, Mar 18, 2021 at 3:59 PM mayamatakeshi <mayamatakeshi at gmail.com>
> wrote:
>
>>
>>
>> On Thu, Mar 18, 2021 at 12:23 PM mayamatakeshi <mayamatakeshi at gmail.com>
>> wrote:
>>
>>>
>>>
>>> On Wed, Mar 17, 2021 at 6:52 PM mayamatakeshi <mayamatakeshi at gmail.com>
>>> wrote:
>>>
>>>> Hi,
>>>> I'm trying play_and_detect_speech with an ESL app but START-OF-INPUT
>>>> doesn't interrupt the TTS being played.
>>>> Is this expected?
>>>> (
>>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+play_and_detect_speech
>>>> is not clear about it)
>>>> Or maybe I need to set some channel variable for this to work.
>>>>
>>>
>>> Hi,
>>> My app is a fork of https://github.com/plivo/plivoframework
>>> I debugged FS code and found the cause of the problem:
>>> I verified that the DETECTED_SPEECH events (including the one
>>> with Speech-Type begin-speaking) were not being fired.
>>> This was because my app was not setting this variable:
>>>   https://freeswitch.org/confluence/display/FREESWITCH/fire_asr_events
>>> Then I set fire_asr_events=true
>>> but still the prompt didn't get interrupted by speech.
>>> Then I checked how FS code was fetching (dequeuing) events and I
>>> realized it checks for divert_events.
>>> Then I changed my app to send
>>>   divert_events off
>>> in the ESL socket
>>> and after that it worked and FS stopped the prompt upon speech start.
>>> However, it is not stopping the prompt when a DTMF digit is received (it
>>> was already this way before I did the changes).
>>> So things are better but still there is something amiss.
>>>
>>
>> OK. I found the reason:
>> I was using 'playback_terminators=any'
>> which reading the FS code shows will result in:
>>   terminators = "1234567890*#"
>> But I was sending DTMF 'a' so it was not causing the prompt to be
>> terminated.
>> Then I changed my test script to send '1' instead and then this caused
>> the prompt to terminate.
>> However, this also terminated the speech detection operation as FS sent
>> STOP to the MRCP server.
>> And the reason is because differently from begin-speaking, a terminator
>> will set the operation as done (switch_ivr_async.c):
>>
>>                       } else if (!strcasecmp(speech_type,
>> "begin-speaking")) {
>>
>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO,
>> "(%s) START OF SPEECH\n", switch_channel_get_name(channel));
>>                           return SWITCH_STATUS_BREAK;
>>                       }
>>
>>                       if (terminators && strchr(terminators,
>> dtmf->digit)) {
>>
>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
>> "(%s) ACCEPT TERMINATOR %c\n", switch_channel_get_name(channel),
>> dtmf->digit);
>>                           switch_channel_set_variable_printf(channel,
>> SWITCH_PLAYBACK_TERMINATOR_USED, "%c", dtmf->digit);
>>                           state->result =
>> switch_core_session_sprintf(session, "DIGIT: %c", dtmf->digit);
>>                           state->done = PLAY_AND_DETECT_DONE;
>>                           return SWITCH_STATUS_BREAK;
>>                       }
>>
>> This might be OK for someone but in my case  there is another issue that
>> the termination of the speech detection is not notified to my app via ESL.
>>
>> I am hoping to solve this by leaving the DTMF collection to be done by
>> the MRCP server but currently the MRCP server I'm using (UniMRCP) doesn't
>> report START-OF-INPUT for DTMF (I only get the RECOGNITION-COMPLETE after
>> all digits are input). (my use case is to allow the user to speak some
>> number sequence or dial it).
>>
>
> Sorry,
> checking again, there is some specific condition that causes UniMRCP to
> not send the START-OF-INPUT.
> I'm not sure what it is yet but it may be by design or following something
> in the protocol that I am not aware of.
>

Just closing this issue and leaving some final details in case someone gets
into a similar problem:
  - I was testing with unimrcp server 1.6.0 and under some conditions this
version would not send START-OF-INPUT upon DTMF detection
  - after updating to unimrcp server 1.7.0, the problem went away.

And in case someone needs to follow SIP/MRCP/DTMF flows, you can try my
sngrep fork. Details here:
  https://github.com/irontec/sngrep/issues/78

(it also supports UTF-8 which is particularly relevant to see the text in
MRCP SPEAK, DEFINE-GRAMMAR and RECOGNITION-COMPLETE messages)

Here is a snapshot:
[image: sngrep.dtmf_and_mrcp.png]


>
>>
>> So I will switch to use detect_speech instead of play_and_detect_speech.
>>
>>
So, using play_and_detect_speech is viable for my needs as I can leave DTMF
handling to unimrcp server.
I might eventually need to implement a solution using detect_speech like in
case of max-number-of-digits which unimrcp server doesn't support at the
moment.
But for now, play_and_detect_speech should suffice.
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From lucas at toptive.co  Fri Mar 19 19:57:03 2021
From: lucas at toptive.co (Lucas Ducculi)
Date: Fri, 19 Mar 2021 16:57:03 -0300
Subject: [Freeswitch-users] Check if an audio file exists to play an
	alternative one if not
Message-ID: <CALWHNi1g1BnR1ra9BKzDEi4A1ZoK30XzbdY9SH1PogYGeBP+zw@mail.gmail.com>

Different macros are defined in the phrases.xml file, I want to know if it
is possible to implement a condition in the macro to check if the audio
file exists and if not, play an alternative one.
Thanks.

<macro name="greeting">
    <input pattern="(.*)">
        <match>
            <action function="play-file"
data="/home/admin/freeswitch/conf/lang/en/ivr/wav/greeting.wav"/>
        </match>
    </input>
</macro>

-- 
Lucas Eduardo Dúcculi
Software Developer
 lucaseduc at hotmail.com
 +54 0358 4853837
 lucas at toptive.co
------------------------------
TOP TIER DEVELOPERS
Av. Italia 1684 - Plata Alta, Río Cuarto, Cba. Argentina
www.toptive.co
[image: LinkedIn icon] <https://www.linkedin.com/company/toptive> [image:
Facebook icon] <https://www.facebook.com/toptive/> [image: Facebook icon]
<https://www.instagram.com/toptierdevelopers/> [image: Facebook icon]
<https://medium.com/@toptive>

This email is confidential and for the exclusive use of the person(s) to
whom it is addressed. If you have received this email by mistake we ask you
to immediately notify the person who sent it.
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From asilva at wirelessmundi.com  Tue Mar 23 15:49:51 2021
From: asilva at wirelessmundi.com (=?utf-8?Q?Ant=C3=B3nio_Silva?=)
Date: Tue, 23 Mar 2021 15:49:51 +0000
Subject: Mod_lua: dbh open to many connections to database 
Message-ID: <71B0A039-98D2-425E-88B0-91F884D1CAC9@wirelessmundi.com>

Hi,

After migrate an old server from debian jessie to debian buster, hit a problem of cpu load increasing and block the server at 100% of cpu usage. I notice that was the number of postgresql open process.

Doing a simple sipp of 40cps to stress the server I notice that the problem is in a lua script, the script check if the caller is a table (database postgres) and allows or not the call.
in debian jessie the db handles open (using db_cache status) are  less than 20, but now in debian buster it scales to more that 100, and I start to see error messages that postgress cannot open more connections (that is true I’ve limited to 100 connections).

Anyone experience this problem?

I also try to set the max-db-handles open to 50, in switch.xml, but it ignores this parameter - I think that is only use in internal modules and not for lua scripts… 
Can I limit the number of open connections when using FS dbh?
 

The function I get to fetch the result:
filename = "pgsql://host=/var/run/postgresql/ user=postgres dbname=list"

callback_query_fetch = function (row)
	if (single_row == true) then
		result = row
		return 1 -- break the loop
	end
	if (result == false) then result = {} end
	table.insert(result, row)
	return 0
end

function db.fetch_single(filename, sql)
	local dbh = freeswitch.Dbh(filename)
	result = false
	single_row = true
	if (dbh:query(sql, callback_query_fetch) == false) then
		dbh:release()
		return false
	end
	dbh:release()
	return result
end



--
Saludos / Regards / Cumprimentos
António Silva




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From thilo at ginkel.com  Wed Mar 24 10:40:30 2021
From: thilo at ginkel.com (Thilo-Alexander Ginkel)
Date: Wed, 24 Mar 2021 11:40:30 +0100
Subject: [Freeswitch-users] Using TLS certificate with intermediate CA
Message-ID: <CANvSZQ8QGMcsLeSHKk3dsz8XtgZtDOf+=D399A3mZL723Ce2gA@mail.gmail.com>

Hello everyone,

I am currently struggling to get FreeSWITCH
(1.10.5-release-17-25569c1631~64bit) to send the intermediate CA
certificate for a Let's Encrypt X.509 certificate to be used for
protecting SIPS traffic.

I included the certificate chain in agent.pem:

-- 8< --
-----BEGIN EC PARAMETERS-----
*REDACTED*
-----END EC PARAMETERS-----
-----BEGIN EC PRIVATE KEY-----
*REDACTED*
-----END EC PRIVATE KEY-----
-----BEGIN CERTIFICATE-----
*SERVER CERT*
-----END CERTIFICATE-----

-----BEGIN CERTIFICATE-----
*INTERMEDIATE CERT*
-----END CERTIFICATE-----
-- 8< --

Still, clients are complaining about an invalid CA and openssl s_client
hints at only the server cert being sent in the server hello.

What did I miss?

Thanks,
Thilo
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From hamid2kviii at hotmail.com  Thu Mar 25 15:04:17 2021
From: hamid2kviii at hotmail.com (Hamid Hashmi)
Date: Thu, 25 Mar 2021 15:04:17 +0000
Subject: [Freeswitch-users] No 200 Okay on Re-Invite (Hold)
Message-ID: <PU1PR03MB29083E6C4D4A5CB3A0EE628EE0629@PU1PR03MB2908.apcprd03.prod.outlook.com>

  1.  A called B through FS server with bypass-media=true and proxy-hold=true.
  2.  A holds Call after Call is answered.
  3.  FS relayed INVITE with SDP Attribute Sendonly to B.
  4.  B responded with 200 Okay with SDP Attribute recvonly.
  5.  FS doesn't relay 200 okay back to A.

What I am doing wrong ? Is there any variable I need to set with proxy-hold ?


Regards

Hamid R. Hashmi
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From schoch+freeswitch.org at xwin32.com  Sun Mar 28 04:17:50 2021
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Sat, 27 Mar 2021 21:17:50 -0700
Subject: [Freeswitch-users] Configuring mod_portaudio for Debian built-in
	(JACK?) audio
Message-ID: <CAOkuKddJ5th0avYvTSayyeBKwuhrwY9yyQJm7zmCBy9T9-+EUA@mail.gmail.com>

I have an HP Prodesk tower with Debian and FreeSWITCH installed. It has an
audio jack that is plugged into a speaker. I configured the system so that
the devices in /dev/snd are group "freeswitch". Now I want to transfer a
call to the speaker with <action application="bridge"
data="portaudio/auto_answer"/>.

However, I am getting errors when I load mod_portaudio that look like this:
2021-03-27 21:15:30.747447 [ERR] mod_portaudio.c:1617 Invalid outstream
specified for endpoint 'default'

It looks like a config file issue, but I have no clue where to start
because I couldn't find a good example. Any suggestions?

-- 
Steve
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From s.kainz at wnt.at  Sun Mar 28 07:56:00 2021
From: s.kainz at wnt.at (Stefan Kainz)
Date: Sun, 28 Mar 2021 07:56:00 +0000
Subject: [Freeswitch-users] Internal Interface suddenly freezes
Message-ID: <40B08719-BA62-4E3F-984B-556AB40A5A91@wnt.at>

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, 


From botelist at gmail.com  Sun Mar 28 20:40:38 2021
From: botelist at gmail.com (Bote Man)
Date: Sun, 28 Mar 2021 16:40:38 -0400
Subject: [Freeswitch-users] Configuring mod_portaudio for Debian
	built-in	(JACK?) audio
In-Reply-To: <CAOkuKddJ5th0avYvTSayyeBKwuhrwY9yyQJm7zmCBy9T9-+EUA@mail.gmail.com>
References: <CAOkuKddJ5th0avYvTSayyeBKwuhrwY9yyQJm7zmCBy9T9-+EUA@mail.gmail.com>
Message-ID: <009e01d72412$9d1f1470$d75d3d50$@gmail.com>

I have never tried any of the linux audio drivers, but I have never heard a good word about any of them. YMMV.

 

In ${conf_dir}/autoload_configs/portaudio.conf.xml I see:

 

  <settings>

    <!-- indev, outdev, ringdev:

   partial case sensitive string match on something in the name

   or the device number prefixed with # eg "#1" (or blank for default) -->

 

    <!-- device to use for input -->

    <param name="indev" value=""/>

    <!-- device to use for output -->

    <param name="outdev" value=""/>

 

The wiki has some good ideas on how to list the available devices from fs_cli in:

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

 

One suggestion to aid you in testing would be to use the “playback” dialplan app to generate test tones with tone_stream or play a .wav file.

 

 

---

Bote Man

Bote Communications

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Steven Schoch
Sent: Sunday, 28 March, 2021 00:18
To: freeswitch-users <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Configuring mod_portaudio for Debian built-in (JACK?) audio

 

I have an HP Prodesk tower with Debian and FreeSWITCH installed. It has an audio jack that is plugged into a speaker. I configured the system so that the devices in /dev/snd are group "freeswitch". Now I want to transfer a call to the speaker with <action application="bridge" data="portaudio/auto_answer"/>.

 

However, I am getting errors when I load mod_portaudio that look like this:

2021-03-27 21:15:30.747447 [ERR] mod_portaudio.c:1617 Invalid outstream specified for endpoint 'default'

 

It looks like a config file issue, but I have no clue where to start because I couldn't find a good example. Any suggestions?

 

-- 

Steve

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From asilva at wirelessmundi.com  Mon Mar 29 09:32:53 2021
From: asilva at wirelessmundi.com (=?utf-8?Q?Ant=C3=B3nio_Silva?=)
Date: Mon, 29 Mar 2021 10:32:53 +0100
Subject: Mod_lua: dbh open to many connections to database 
In-Reply-To: <71B0A039-98D2-425E-88B0-91F884D1CAC9@wirelessmundi.com>
References: <71B0A039-98D2-425E-88B0-91F884D1CAC9@wirelessmundi.com>
Message-ID: <CE18C0C5-2BF7-4870-9C45-39D624369A0C@wirelessmundi.com>

Still investigating the issue, using the debug_sql, I notice that the connection is correctly open and release - my script is doing what is expected:

2021-03-29 10:49:34.623831 [ALERT] freeswitch_lua.cpp:371 Reuse Unused Cached DB handle db="host=/var/run/postgresql/ user=postgres dbname=cmcore",type="database_interface" [Database interface prefix: pgsql]
2021-03-29 10:49:34.623831 [ALERT] freeswitch_lua.cpp:372 DBH handle 0x55aa3c24e110 Connected.
2021-03-29 10:49:34.623831 [ALERT] freeswitch_lua.cpp:401 DBH handle 0x55aa3c24e110 released.


Which explain why the max_db_handle setting is not affecting the number of connections opened in postgres, 
Condition: 
	while(runtime.max_db_handles && sql_manager.total_handles >= runtime.max_db_handles && sql_manager.total_used_handles >= sql_manager.total_handles) {

But on release:
	sql_manager.total_used_handles--;

So for FS the connection is closed but is not coherent to Postgres. 


During the start of sipp it works as expected I don't see more than 3 open connections to postgresql but during the increase of new stablished calls, it starts to open more connections to postgresl and don’t re-use the open handlers. I can’t find out why... 

Any hint to see what is triggering this behaviour?

Thank you 




The output on postgresql open process:
2021-03-29 11:17:45
root     16611  0.0  0.0   8604   892 pts/1    S+   11:17   0:00  |       \_ grep cmcore
postgres  8970  0.2  0.0 211444 13108 ?        Ss   11:16   0:00  \_ postgres: postgres cmcore [local] idle
postgres  8971  1.7  0.0 211444 13136 ?        Ss   11:16   0:01  \_ postgres: postgres cmcore [local] idle
postgres 11885  4.5  0.0 211444 13756 ?        Rs   11:16   0:02  \_ postgres: postgres cmcore [local] idle
2021-03-29 11:17:46
root     16620  0.0  0.0   8604   884 pts/1    S+   11:17   0:00  |       \_ grep cmcore
postgres  8970  0.2  0.0 211444 13108 ?        Ss   11:16   0:00  \_ postgres: postgres cmcore [local] idle
postgres  8971  1.7  0.0 211444 13136 ?        Ss   11:16   0:01  \_ postgres: postgres cmcore [local] idle
postgres 11885  4.4  0.0 211444 13756 ?        Ss   11:16   0:02  \_ postgres: postgres cmcore [local] idle
2021-03-29 11:17:47
root     16645  0.0  0.0   8604   888 pts/1    S+   11:17   0:00  |       \_ grep cmcore
postgres  8970  0.2  0.0 211444 13108 ?        Ss   11:16   0:00  \_ postgres: postgres cmcore [local] idle
postgres  8971  1.7  0.0 211444 13136 ?        Ss   11:16   0:01  \_ postgres: postgres cmcore [local] idle
postgres 11885  4.4  0.0 211444 13756 ?        Rs   11:16   0:02  \_ postgres: postgres cmcore [local] SELECT
2021-03-29 11:17:48
root     16669  0.0  0.0   8604   828 pts/1    S+   11:17   0:00  |       \_ grep cmcore
postgres  8970  0.2  0.0 211444 13108 ?        Ss   11:16   0:00  \_ postgres: postgres cmcore [local] idle
postgres  8971  1.7  0.0 211444 13136 ?        Ss   11:16   0:01  \_ postgres: postgres cmcore [local] idle
postgres 11885  4.4  0.0 211444 13756 ?        Ss   11:16   0:02  \_ postgres: postgres cmcore [local] idle
postgres 16655  0.0  0.0 211444 13160 ?        Ss   11:17   0:00  \_ postgres: postgres cmcore [local] idle
postgres 16656  0.0  0.0 211444 13160 ?        Ss   11:17   0:00  \_ postgres: postgres cmcore [local] idle
postgres 16658  0.0  0.0 211444 13128 ?        Ss   11:17   0:00  \_ postgres: postgres cmcore [local] idle
postgres 16659  0.0  0.0 211444 13152 ?        Ss   11:17   0:00  \_ postgres: postgres cmcore [local] idle
postgres 16660  0.0  0.0 211444 13132 ?        Ss   11:17   0:00  \_ postgres: postgres cmcore [local] idle
postgres 16661  0.0  0.0 211444 13164 ?        Ss   11:17   0:00  \_ postgres: postgres cmcore [local] idle
postgres 16663  0.0  0.0 211444 13164 ?        Ss   11:17   0:00  \_ postgres: postgres cmcore [local] idle


And freeswitch db_cache status: (for sofia I use a different connection than the used for lua script)

**************
2021-03-29 11:17:45
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 0
	Total used: 14296
	Flags: Unlocked, Detached(0)
	Creator: mod_callcenter.c:609
	Last User: src/switch_console.c:253
db="host=/var/run/postgresql/ user=postgres dbname=cmcore",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 0
	Total used: 13359
	Flags: Unlocked, Detached(0)
	Creator: freeswitch_lua.cpp:371
	Last User: freeswitch_lua.cpp:371
db="host=/var/run/postgresql/ user=postgres dbname=cmcore",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 0
	Total used: 7333
	Flags: Unlocked, Detached(0)
	Creator: freeswitch_lua.cpp:371
	Last User: freeswitch_lua.cpp:371
db="host=/var/run/postgresql/ user=postgres dbname=cmcore",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 0
	Total used: 1020
	Flags: Unlocked, Detached(0)
	Creator: freeswitch_lua.cpp:371
	Last User: freeswitch_lua.cpp:371
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 0
	Total used: 42779
	Flags: Unlocked, Detached(0)
	Creator: mod_callcenter.c:609
	Last User: sofia_glue.c:2833
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 0
	Total used: 13398
	Flags: Unlocked, Detached(0)
	Creator: sofia_glue.c:2833
	Last User: mod_callcenter.c:609
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3469
	Total used: 5
	Flags: Locked, Attached(1)
	Creator: sofia_glue.c:2833
	Last User: src/switch_core_sqldb.c:2276
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3471
	Total used: 2
	Flags: Locked, Attached(1)
	Creator: sofia_glue.c:2602
	Last User: src/switch_core_sqldb.c:2276
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3471
	Total used: 2
	Flags: Locked, Attached(1)
	Creator: sofia_glue.c:2602
	Last User: src/switch_core_sqldb.c:2276
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3471
	Total used: 2
	Flags: Locked, Attached(1)
	Creator: sofia_glue.c:2602
	Last User: src/switch_core_sqldb.c:2276
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3471
	Total used: 2
	Flags: Locked, Attached(1)
	Creator: sofia_glue.c:2602
	Last User: src/switch_core_sqldb.c:2276
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3471
	Total used: 1
	Flags: Locked, Attached(1)
	Creator: src/switch_core_sqldb.c:2276
	Last User: 
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3471
	Total used: 3
	Flags: Locked, Attached(1)
	Creator: src/switch_core_sqldb.c:3636
	Last User: src/switch_core_sqldb.c:2276
13 total. 7 in use.

**************
2021-03-29 11:17:46
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 0
	Total used: 14725
	Flags: Unlocked, Detached(0)
	Creator: mod_callcenter.c:609
	Last User: src/switch_console.c:253
db="host=/var/run/postgresql/ user=postgres dbname=cmcore",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 0
	Total used: 13580
	Flags: Unlocked, Detached(0)
	Creator: freeswitch_lua.cpp:371
	Last User: freeswitch_lua.cpp:371
db="host=/var/run/postgresql/ user=postgres dbname=cmcore",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 0
	Total used: 7402
	Flags: Unlocked, Detached(0)
	Creator: freeswitch_lua.cpp:371
	Last User: freeswitch_lua.cpp:371
db="host=/var/run/postgresql/ user=postgres dbname=cmcore",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 1
	Total used: 1027
	Flags: Unlocked, Detached(0)
	Creator: freeswitch_lua.cpp:371
	Last User: freeswitch_lua.cpp:371
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 0
	Total used: 42823
	Flags: Unlocked, Detached(0)
	Creator: mod_callcenter.c:609
	Last User: sofia_glue.c:2833
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 0
	Total used: 13408
	Flags: Unlocked, Detached(0)
	Creator: sofia_glue.c:2833
	Last User: mod_callcenter.c:609
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3470
	Total used: 5
	Flags: Locked, Attached(1)
	Creator: sofia_glue.c:2833
	Last User: src/switch_core_sqldb.c:2276
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3472
	Total used: 2
	Flags: Locked, Attached(1)
	Creator: sofia_glue.c:2602
	Last User: src/switch_core_sqldb.c:2276
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3472
	Total used: 2
	Flags: Locked, Attached(1)
	Creator: sofia_glue.c:2602
	Last User: src/switch_core_sqldb.c:2276
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3472
	Total used: 2
	Flags: Locked, Attached(1)
	Creator: sofia_glue.c:2602
	Last User: src/switch_core_sqldb.c:2276
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3472
	Total used: 2
	Flags: Locked, Attached(1)
	Creator: sofia_glue.c:2602
	Last User: src/switch_core_sqldb.c:2276
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3472
	Total used: 1
	Flags: Locked, Attached(1)
	Creator: src/switch_core_sqldb.c:2276
	Last User: 
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3472
	Total used: 3
	Flags: Locked, Attached(1)
	Creator: src/switch_core_sqldb.c:3636
	Last User: src/switch_core_sqldb.c:2276
13 total. 7 in use.

**************
2021-03-29 11:17:47
db="host=/var/run/postgresql/ user=postgres dbname=cmcore",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 0
	Total used: 8
	Flags: Unlocked, Detached(0)
	Creator: freeswitch_lua.cpp:371
	Last User: freeswitch_lua.cpp:371
db="host=/var/run/postgresql/ user=postgres dbname=cmcore",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 0
	Total used: 16
	Flags: Unlocked, Detached(0)
	Creator: freeswitch_lua.cpp:371
	Last User: freeswitch_lua.cpp:371
db="host=/var/run/postgresql/ user=postgres dbname=cmcore",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 0
	Total used: 18
	Flags: Unlocked, Detached(0)
	Creator: freeswitch_lua.cpp:371
	Last User: freeswitch_lua.cpp:371
db="host=/var/run/postgresql/ user=postgres dbname=cmcore",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 0
	Total used: 18
	Flags: Unlocked, Detached(0)
	Creator: freeswitch_lua.cpp:371
	Last User: freeswitch_lua.cpp:371
db="host=/var/run/postgresql/ user=postgres dbname=cmcore",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 1
	Total used: 13
	Flags: Unlocked, Detached(0)
	Creator: freeswitch_lua.cpp:371
	Last User: freeswitch_lua.cpp:371
db="host=/var/run/postgresql/ user=postgres dbname=cmcore",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 1
	Total used: 8
	Flags: Unlocked, Detached(0)
	Creator: freeswitch_lua.cpp:371
	Last User: freeswitch_lua.cpp:371
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 0
	Total used: 15211
	Flags: Unlocked, Detached(0)
	Creator: mod_callcenter.c:609
	Last User: src/switch_console.c:253
db="host=/var/run/postgresql/ user=postgres dbname=cmcore",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 1
	Total used: 13747
	Flags: Unlocked, Detached(0)
	Creator: freeswitch_lua.cpp:371
	Last User: freeswitch_lua.cpp:371
db="host=/var/run/postgresql/ user=postgres dbname=cmcore",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 1
	Total used: 7485
	Flags: Unlocked, Detached(0)
	Creator: freeswitch_lua.cpp:371
	Last User: freeswitch_lua.cpp:371
db="host=/var/run/postgresql/ user=postgres dbname=cmcore",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 1
	Total used: 1035
	Flags: Unlocked, Detached(0)
	Creator: freeswitch_lua.cpp:371
	Last User: freeswitch_lua.cpp:371
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 1
	Total used: 42864
	Flags: Unlocked, Detached(0)
	Creator: mod_callcenter.c:609
	Last User: sofia_glue.c:2833
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 1
	Total used: 13426
	Flags: Unlocked, Detached(0)
	Creator: sofia_glue.c:2833
	Last User: sofia_glue.c:2833
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3472
	Total used: 5
	Flags: Locked, Attached(1)
	Creator: sofia_glue.c:2833
	Last User: src/switch_core_sqldb.c:2276
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3474
	Total used: 2
	Flags: Locked, Attached(1)
	Creator: sofia_glue.c:2602
	Last User: src/switch_core_sqldb.c:2276
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3474
	Total used: 2
	Flags: Locked, Attached(1)
	Creator: sofia_glue.c:2602
	Last User: src/switch_core_sqldb.c:2276
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3474
	Total used: 2
	Flags: Locked, Attached(1)
	Creator: sofia_glue.c:2602
	Last User: src/switch_core_sqldb.c:2276
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3474
	Total used: 2
	Flags: Locked, Attached(1)
	Creator: sofia_glue.c:2602
	Last User: src/switch_core_sqldb.c:2276
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3474
	Total used: 1
	Flags: Locked, Attached(1)
	Creator: src/switch_core_sqldb.c:2276
	Last User: 
db="host=/var/run/pgsql-fs user=freeswitch password=",type="database_interface"
	Type: DATABASE_INTERFACE
	Last used: 3474
	Total used: 3
	Flags: Locked, Attached(1)
	Creator: src/switch_core_sqldb.c:3636
	Last User: src/switch_core_sqldb.c:2276
19 total. 7 in use.




--
Saludos / Regards / Cumprimentos
António Silva




> On 23 Mar 2021, at 15:49, António Silva <asilva at wirelessmundi.com> wrote:
> 
> Hi,
> 
> After migrate an old server from debian jessie to debian buster, hit a problem of cpu load increasing and block the server at 100% of cpu usage. I notice that was the number of postgresql open process.
> 
> Doing a simple sipp of 40cps to stress the server I notice that the problem is in a lua script, the script check if the caller is a table (database postgres) and allows or not the call.
> in debian jessie the db handles open (using db_cache status) are  less than 20, but now in debian buster it scales to more that 100, and I start to see error messages that postgress cannot open more connections (that is true I’ve limited to 100 connections).
> 
> Anyone experience this problem?
> 
> I also try to set the max-db-handles open to 50, in switch.xml, but it ignores this parameter - I think that is only use in internal modules and not for lua scripts… 
> Can I limit the number of open connections when using FS dbh?
>  
> 
> The function I get to fetch the result:
> filename = "pgsql://host=/var/run/postgresql/ <pgsql://host=/var/run/postgresql/> user=postgres dbname=list"
> 
> callback_query_fetch = function (row)
> 	if (single_row == true) then
> 		result = row
> 		return 1 -- break the loop
> 	end
> 	if (result == false) then result = {} end
> 	table.insert(result, row)
> 	return 0
> end
> 
> function db.fetch_single(filename, sql)
> 	local dbh = freeswitch.Dbh(filename)
> 	result = false
> 	single_row = true
> 	if (dbh:query(sql, callback_query_fetch) == false) then
> 		dbh:release()
> 		return false
> 	end
> 	dbh:release()
> 	return result
> end
> 
> 
> 
> --
> Saludos / Regards / Cumprimentos
> António Silva
> 
> 
> 
> 

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From david.villasmil.work at gmail.com  Mon Mar 29 11:12:22 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 29 Mar 2021 12:12:22 +0100
Subject: [Freeswitch-users] Internal Interface suddenly freezes
In-Reply-To: <40B08719-BA62-4E3F-984B-556AB40A5A91@wnt.at>
References: <40B08719-BA62-4E3F-984B-556AB40A5A91@wnt.at>
Message-ID: <CAFGRPVrj-ewOpY8h9HUdHz1T=A8KCg=Nsqp-dxGoC4NR23i_ZA@mail.gmail.com>

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
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From s.kainz at wnt.at  Mon Mar 29 11:49:24 2021
From: s.kainz at wnt.at (Stefan Kainz)
Date: Mon, 29 Mar 2021 11:49:24 +0000
Subject: [Freeswitch-users] Internal Interface suddenly freezes
In-Reply-To: <CAFGRPVrj-ewOpY8h9HUdHz1T=A8KCg=Nsqp-dxGoC4NR23i_ZA@mail.gmail.com>
References: <40B08719-BA62-4E3F-984B-556AB40A5A91@wnt.at>
 <CAFGRPVrj-ewOpY8h9HUdHz1T=A8KCg=Nsqp-dxGoC4NR23i_ZA@mail.gmail.com>
Message-ID: <VI1P190MB0494FA8017F747D0C79C8F8F867E9@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>

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<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
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

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

Professional FreeSWITCH Services
sales at freeswitch.com<mailto:sales at freeswitch.com>
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
--
Regards,

David Villasmil
email: david.villasmil.work at gmail.com<mailto:david.villasmil.work at gmail.com>
phone: +34669448337
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From botelist at gmail.com  Mon Mar 29 13:45:05 2021
From: botelist at gmail.com (Bote Man)
Date: Mon, 29 Mar 2021 09:45:05 -0400
Subject: [Freeswitch-users] Internal Interface suddenly freezes
In-Reply-To: <VI1P190MB0494FA8017F747D0C79C8F8F867E9@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
References: <40B08719-BA62-4E3F-984B-556AB40A5A91@wnt.at>
 <CAFGRPVrj-ewOpY8h9HUdHz1T=A8KCg=Nsqp-dxGoC4NR23i_ZA@mail.gmail.com>
 <VI1P190MB0494FA8017F747D0C79C8F8F867E9@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
Message-ID: <001b01d724a1$ba791130$2f6b3390$@gmail.com>

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 <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
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

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

Professional FreeSWITCH Services
sales at freeswitch.com <mailto:sales at freeswitch.com> 
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org> 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com

-- 

Regards,

 

David Villasmil

email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com> 

phone: +34669448337

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From s.kainz at wnt.at  Mon Mar 29 14:15:57 2021
From: s.kainz at wnt.at (Stefan Kainz)
Date: Mon, 29 Mar 2021 14:15:57 +0000
Subject: [Freeswitch-users] Internal Interface suddenly freezes
In-Reply-To: <001b01d724a1$ba791130$2f6b3390$@gmail.com>
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>
Message-ID: <VI1P190MB049417F108E2F45F3AEF94EC867E9@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>

Hmm, thats an angle I didn’t think of before …
I’ll check that, thank you!

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<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
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

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

Professional FreeSWITCH Services
sales at freeswitch.com<mailto:sales at freeswitch.com>
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
--
Regards,

David Villasmil
email: david.villasmil.work at gmail.com<mailto:david.villasmil.work at gmail.com>
phone: +34669448337
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From s.kainz at wnt.at  Mon Mar 29 14:26:04 2021
From: s.kainz at wnt.at (Stefan Kainz)
Date: Mon, 29 Mar 2021 14:26:04 +0000
Subject: [Freeswitch-users] Internal Interface suddenly freezes
In-Reply-To: <001b01d724a1$ba791130$2f6b3390$@gmail.com>
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>
Message-ID: <VI1P190MB049439288D7951FC99D99C75867E9@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>

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<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
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

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

Professional FreeSWITCH Services
sales at freeswitch.com<mailto:sales at freeswitch.com>
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
--
Regards,

David Villasmil
email: david.villasmil.work at gmail.com<mailto:david.villasmil.work at gmail.com>
phone: +34669448337
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From martin at pattersong.co.uk  Thu Mar 25 09:50:25 2021
From: martin at pattersong.co.uk (Martin Paterson)
Date: Thu, 25 Mar 2021 09:50:25 +0000
Subject: [Freeswitch-users] FW: User_Not_Registered
In-Reply-To: <VI1PR10MB209492DECDA639055A82503895629@VI1PR10MB2094.EURPRD10.PROD.OUTLOOK.COM>
References: <DBBPR08MB6026CF68A8F9A5100CA64711B5909@DBBPR08MB6026.eurprd08.prod.outlook.com>
 <VI1PR10MB209492DECDA639055A82503895629@VI1PR10MB2094.EURPRD10.PROD.OUTLOOK.COM>
Message-ID: <CAM8=8aDuno_vexJb2fWuT0yX3HekzmkzELZrhY4KOLKzxMH0KQ@mail.gmail.com>

Your registration is to <number>@myrevbill.com, but the bridge is going to
<number>@10.0.1.29.

Best wishes,

Martin.

Martin Paterson, Pattersong Music
Reduced orchestrations of G&S

*From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *CRAIG WILSON
> *Sent:* 11 March 2021 16:50
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] User_Not_Registered
>
>
>
> Hi,
>
>
>
> I am using ASTPP on top of FS. I am receiving an inbound DID. However, my
> SIP Trace shows
>
> error "User_Not_Registered" see (sip trace log below).
>
>
>
> Running command list_user reveals nothing and sofia_contact reveals
> "User_Not_Registered".
>
>
>
> Show Registrations - reveals:
>
> 9856937967,myrevbill.com
> ,8vSSYnN7QcIRw5u3nfg5pA..,sofia/sip-ip/sip:9856937967 at 92.7.187.203:39706
> ;transport=UDP;rinstance=ec30c6121794d08f,1615480452,92.7.187.203,39706,udp,
> astpp.myrevbill.com,
>
>
>
> 1 total.
>
>
>
> Then I ran command below:
>
> sofia status profile sip-ip  reg
>
>
>
> Registrations:
>
>
> =================================================================================================
>
> Call-ID:        S3gXf-fjxpA9ws47L8bh9w..
>
> User:           9856937967 at myrevbill.com
>
> Contact:        "" <
> sip:9856937967 at 92.7.187.203:62990;transport=UDP;rinstance=df94511808fbcf07
> >
>
> Agent:          Z 5.4.9 rv2.10.11.7
>
> Status:         Registered(UDP)(unknown) EXP(2021-03-11 11:03:29)
> EXPSECS(102)
>
> Ping-Status:    Reachable
>
> Ping-Time:      0.00
>
> Host:           astpp.myrevbill.com
>
> IP:             92.7.187.203
>
> Port:           62990
>
> Auth-User:      9856937967
>
> Auth-Realm:     myrevbill.com
>
> MWI-Account:    9856937967 at myrevbill.com
>
>
>
> Total items returned: 1
>
>
>
>
>
>
>
> Sip Trace Log:
>
>
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:584 (
> sofia/sip-ip/01226971834 at 10.0.1.29) Running State Change CS_NEW (Cur 1
> Tot 1)
>
> 2021-03-10 12:20:17.534757 [DEBUG] sofia.c:9873
> sofia/sip-ip/01226971834 at 10.0.1.29 receiving invite from 10.0.1.29:5060
> version: 1.6.20  64bit
>
> 2021-03-10 12:20:17.534757 [DEBUG] sofia.c:9989 IP 10.0.1.29 Approved by
> acl "default[]". Access Granted.
>
> 2021-03-10 12:20:17.534757 [DEBUG] sofia.c:7084 Channel
> sofia/sip-ip/01226971834 at 10.0.1.29 entering state [received][100]
>
> 2021-03-10 12:20:17.534757 [DEBUG] sofia.c:7094 Remote SDP:
>
> v=0
>
> o=- 283383691 283383691 IN IP4 10.0.1.29
>
> s=Asterisk
>
> c=IN IP4 10.0.1.29
>
> t=0 0
>
> m=audio 11144 RTP/AVP 0 8 9 4 3 101
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:9 G722/8000
>
> a=rtpmap:4 G723/8000
>
> a=rtpmap:3 GSM/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=ptime:20
>
> a=maxptime:150
>
>
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4504 Audio Codec
> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[G729:18:8000:20:8000:1]
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4504 Audio Codec
> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[G729:18:8000:20:8000:1]
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [G722:9:8000:20:64000:1]/[G729:18:8000:20:8000:1]
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [G723:4:8000:20:6300:1]/[PCMU:0:8000:20:64000:1]
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [G723:4:8000:20:6300:1]/[PCMA:8:8000:20:64000:1]
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [G723:4:8000:20:6300:1]/[G729:18:8000:20:8000:1]
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [GSM:3:8000:20:13200:1]/[PCMU:0:8000:20:64000:1]
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [GSM:3:8000:20:13200:1]/[PCMA:8:8000:20:64000:1]
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4449 Audio Codec
> Compare [GSM:3:8000:20:13200:1]/[G729:18:8000:20:8000:1]
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4365 Set
> telephone-event payload to 101 at 8000
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:3061 Set Codec
> sofia/sip-ip/01226971834 at 10.0.1.29 PCMU/8000 20 ms 160 samples 64000 bits
> 1 channels
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_codec.c:111
> sofia/sip-ip/01226971834 at 10.0.1.29 Original read codec set to PCMU:0
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4708 Set
> telephone-event payload to 101 at 8000
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_media.c:4767
> sofia/sip-ip/01226971834 at 10.0.1.29 Set 2833 dtmf send payload to 101 recv
> payload to 101
>
> 2021-03-10 12:20:17.534757 [DEBUG] sofia.c:7507 (
> sofia/sip-ip/01226971834 at 10.0.1.29) State Change CS_NEW -> CS_INIT
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:603 (
> sofia/sip-ip/01226971834 at 10.0.1.29) State NEW
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:584 (
> sofia/sip-ip/01226971834 at 10.0.1.29) Running State Change CS_INIT (Cur 1
> Tot 1)
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:627 (
> sofia/sip-ip/01226971834 at 10.0.1.29) State INIT
>
> 2021-03-10 12:20:17.534757 [DEBUG] mod_sofia.c:90
> sofia/sip-ip/01226971834 at 10.0.1.29 SOFIA INIT
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:40
> sofia/sip-ip/01226971834 at 10.0.1.29 Standard INIT
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:48 (
> sofia/sip-ip/01226971834 at 10.0.1.29) State Change CS_INIT -> CS_ROUTING
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:627 (
> sofia/sip-ip/01226971834 at 10.0.1.29) State INIT going to sleep
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:584 (
> sofia/sip-ip/01226971834 at 10.0.1.29) Running State Change CS_ROUTING (Cur
> 1 Tot 1)
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_channel.c:2249 (
> sofia/sip-ip/01226971834 at 10.0.1.29) Callstate Change DOWN -> RINGING
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:643 (
> sofia/sip-ip/01226971834 at 10.0.1.29) State ROUTING
>
> 2021-03-10 12:20:17.534757 [DEBUG] mod_sofia.c:143
> sofia/sip-ip/01226971834 at 10.0.1.29 SOFIA ROUTING
>
> 2021-03-10 12:20:17.534757 [DEBUG] switch_core_state_machine.c:236
> sofia/sip-ip/01226971834 at 10.0.1.29 Standard ROUTING
>
> 2021-03-10 12:20:17.534757 [INFO] mod_dialplan_xml.c:637 Processing
> 01226971834 <01226971834>->03302290443 in context default
>
> 2021-03-10 12:20:17.596174 [DEBUG] freeswitch_lua.cpp:365 DBH handle
> 0x7f3ebc1cae70 Connected.
>
> 2021-03-10 12:20:17.596174 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [LOAD_CONF]
> Query :SELECT name,value FROM `system` WHERE group_title IN
> ('global','opensips','callingcard','calls','InternationalPrefixes')
>
> 2021-03-10 12:20:17.596174 [DEBUG] switch_cpp.cpp:1365 [ASTPP]
> [LOAD_ADDON_CONF] Query :SELECT package_name FROM addons
>
> 2021-03-10 12:20:17.596174 [INFO] switch_cpp.cpp:1365 [ASTPP] [Dialplan]
> Dialed number : 03302290443
>
> 2021-03-10 12:20:17.596174 [DEBUG] switch_cpp.cpp:1365 [ASTPP]
> [DOAUTHORIZATION] Query :SELECT access_number FROM accessnumber WHERE
> access_number = '03302290443' AND status=0 limit 1
>
> 2021-03-10 12:20:17.596174 [INFO] switch_cpp.cpp:1365 [ASTPP] [Dialplan]
> Caller Id name / number  : 01226971834 / 01226971834
>
> 2021-03-10 12:20:17.596174 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [CHECK_DID]
> Query :SELECT A.id as id,A.number as did_number,B.id as accountid,B.number
> as account_code,A.number as
> did_number,A.connectcost,A.includedseconds,A.cost,A.inc,A.extensions,A.maxchannels,A.call_type,A.city,A.province,A.init_inc,A.leg_timeout,A.status,A.country_id,A.call_type_vm_flag
> FROM dids AS A,accounts AS B WHERE B.status=0 AND B.deleted=0 AND
> B.id=A.accountid AND A.number ="03302290443" LIMIT 1
>
> 2021-03-10 12:20:17.596174 [INFO] switch_cpp.cpp:1365 [ASTPP] [Dialplan]
> Call direction : inbound
>
> 2021-03-10 12:20:17.596174 [DEBUG] switch_cpp.cpp:1365 [ASTPP]
> [IPAUTHENTICATION] Query :SELECT ip_map.*, (SELECT number FROM accounts
> where id=accountid AND status=0 AND deleted=0) AS account_code FROM ip_map
> WHERE INET_ATON("10.0.1.29") BETWEEN(INET_ATON(SUBSTRING_INDEX(`ip`, '/',
> 1)) & 0xffffffff ^((0x1 <<(32 -  SUBSTRING_INDEX(`ip`, '/', -1))) -1 ))
> AND(INET_ATON(SUBSTRING_INDEX(`ip`, '/', 1)) |((0x100000000 >>
> SUBSTRING_INDEX(`ip`,'/', -1)) -1))  AND "03302290443"  LIKE
> CONCAT(prefix,'%') ORDER BY LENGTH(prefix) DESC LIMIT 1
>
> 2021-03-10 12:20:17.634761 [INFO] switch_cpp.cpp:1365 [ASTPP] [Accountcode
> : 9856937967]
>
> 2021-03-10 12:20:17.634761 [DEBUG] switch_cpp.cpp:1365 [ASTPP]
> [DOAUTHORIZATION] Query :SELECT * FROM accounts WHERE number = "9856937967"
> AND deleted = 0 limit 1
>
> 2021-03-10 12:20:17.634761 [DEBUG] switch_cpp.cpp:1365 [ASTPP]
> [CHECK_SPEEDDIAL] Query :SELECT A.number FROM speed_dial as A,accounts as B
> WHERE B.status=0 AND B.deleted=0 AND B.id=A.accountid AND A.speed_num
> ="03302290443" AND A.accountid = '13' limit 1
>
> 2021-03-10 12:20:17.634761 [INFO] switch_cpp.cpp:1365 [ASTPP] [Dialplan]
> SPEED DIAL NUMBER : 03302290443
>
> 2021-03-10 12:20:17.634761 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [CHECK_DID]
> Query :SELECT A.id as id,A.number as did_number,B.id as accountid,B.number
> as account_code,A.number as
> did_number,A.connectcost,A.includedseconds,A.cost,A.inc,A.extensions,A.maxchannels,A.call_type,A.city,A.province,A.init_inc,A.leg_timeout,A.status,A.country_id,A.call_type_vm_flag
> FROM dids AS A,accounts AS B WHERE B.status=0 AND B.deleted=0 AND
> B.id=A.accountid AND A.number ="03302290443" LIMIT 1
>
> 2021-03-10 12:20:17.634761 [INFO] switch_cpp.cpp:1365 [ASTPP] [Dialplan]
> New Call direction : inbound
>
> 2021-03-10 12:20:17.634761 [DEBUG] switch_cpp.cpp:1365 [ASTPP]
> [IS_CHECK_DID] Query :SELECT * FROM dids WHERE number ="03302290443" AND
> (accountid = 0 OR status = 1) LIMIT 1
>
> 2021-03-10 12:20:17.634761 [DEBUG] switch_cpp.cpp:1365 [ASTPP]
> [GET_PACKAGE_INFO] call_direction :inbound
>
> 2021-03-10 12:20:17.634761 [DEBUG] switch_cpp.cpp:1365 [ASTPP]
> [GET_PACKAGE_INFO] DID_ACCOUNTID :2
>
> 2021-03-10 12:20:17.634761 [DEBUG] switch_cpp.cpp:1365 [ASTPP]
> [GET_PACKAGE_INFO] Query :SELECT *,P.id as package_id,P.product_id as
> product_id FROM packages_view as P inner join package_patterns as PKGPTR on
> P.product_id = PKGPTR.product_id WHERE (patterns = '^03302290443.*' OR
> patterns = '^0330229044.*' OR patterns = '^033022904.*' OR patterns =
> '^03302290.*' OR patterns = '^0330229.*' OR patterns = '^033022.*' OR
> patterns = '^03302.*' OR patterns = '^0330.*' OR patterns = '^033.*' OR
> patterns = '^03.*' OR patterns = '^0.*' OR patterns ='--') AND accountid =
> 2 ORDER BY LENGTH(PKGPTR.patterns) DESC
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP]
> =============== Account Information ===================
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] User id : 13
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Account code
> : 9856937967
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Balance :
> 10000
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Type : 0
> [0:prepaid,1:postpaid]
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Ratecard id
> : 1
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP]
> ========================================================
>
> 2021-03-10 12:20:17.655679 [DEBUG] switch_cpp.cpp:1365 [ASTPP]
> [GET_PRICELIST_INFO] Query :select * from pricelists WHERE id = 1 AND
> status = 0
>
> 2021-03-10 12:20:17.655679 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [GET_RATES]
> call_direction :inbound
>
> 2021-03-10 12:20:17.655679 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [CHECK_DID]
> Query :SELECT A.id as id,A.number as did_number,B.id as accountid,B.number
> as account_code,A.number as
> did_number,A.connectcost,A.includedseconds,A.cost,A.inc,A.extensions,A.maxchannels,A.call_type,A.city,A.province,A.init_inc,A.leg_timeout,A.status,A.country_id,A.call_type_vm_flag
> FROM dids AS A,accounts AS B WHERE B.status=0 AND B.deleted=0 AND
> B.id=A.accountid AND A.number ="03302290443" LIMIT 1
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP]
> call_direction:::::: inbound
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP]
> =============== Rates Information ===================
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] ID : 1
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Connectcost
> : 0.00000
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP]
> Includedseconds : 0
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Cost :
> 1.30378
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] comment :
> 03302290443
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Country Id :
> 200
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Accid : 13
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP]
> ================================================================
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP]
> [FIND_MAXLENGTH] Your10000 balance Accountid 13 !!!
>
> 2021-03-10 12:20:17.655679 [NOTICE] switch_cpp.cpp:1365 [ASTPP]
> [FIND_MAXLENGTH] Limiting call to config max length 100 mins!
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] Call Max
> length duration : 100 minutes
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] [userinfo]
> INB_FREE:TRUE
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] [userinfo]
> free_inbound:1
>
> 2021-03-10 12:20:17.655679 [DEBUG] switch_cpp.cpp:1365 [ASTPP]
> [GET_OVERRIDE_CALLERID] Query :SELECT callerid_name as
> cid_name,callerid_number as cid_number,accountid FROM accounts_callerid
> WHERE accountid = 13 AND status=0 LIMIT 1
>
> 2021-03-10 12:20:17.655679 [DEBUG] switch_cpp.cpp:1365 [ASTPP]
> [DOAUTHORIZATION] Query :SELECT * FROM accounts WHERE id = "2" AND deleted
> = 0 limit 1
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] [userinfo]
> Actual CustomerInfo XML:13
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] [userinfo]
> Userinfo XML:13
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP] [userinfo]
> Actual CustomerInfo XML : 13
>
> 2021-03-10 12:20:17.655679 [INFO] switch_cpp.cpp:1365 [ASTPP]
> maxlength::::::::: 100
>
> 2021-03-10 12:20:17.655679 [DEBUG] switch_cpp.cpp:1365 [ASTPP]
> custom_function_name:::::::::::::::::::::::::custom_inbound_0
>
> 2021-03-10 12:20:17.655679 [DEBUG] switch_cpp.cpp:1365 [ASTPP] [Dialplan]
> Generated XML:<?xml version="1.0" encoding="UTF-8" standalone="no"?>
>
> <document type="freeswitch/xml">
>
> <section name="dialplan" description="ASTPP Dialplan">
>
> <context name="default">
>
> <extension name="03302290443">
>
> <condition field="destination_number" expression="03302290443">
>
> <action application="set" data="effective_destination_number=03302290443"/>
>
> <action application="set" data="bridge_pre_execute_bleg_app=sched_hangup"/>
>
> <action application="set" data="bridge_pre_execute_bleg_data=+6000
> normal_clearing"/>
>
> <action application="set" data="callstart=2021-03-10 12:20:17"/>
>
> <action application="set" data="hangup_after_bridge=true"/>
>
> <action application="set" data="continue_on_fail=TRUE"/>
>
> <action application="set" data="account_id=13"/>
>
> <action application="set" data="parent_id=0"/>
>
> <action application="set" data="entity_id=3"/>
>
> <action application="set" data="call_processed=internal"/>
>
> <action application="set" data="call_direction=inbound"/>
>
> <action application="set" data="accountname=FreePBX"/>
>
> <action application="set"
> data="origination_rates_did=ID:1|CODE:^03302290443.*|DESTINATION:03302290443|CONNECTIONCOST:0.00000|INCLUDEDSECONDS:0|CT:0|COST:1.30378|INC:0|INITIALBLOCK:0|RATEGROUP:0|MARKUP:0|CI:200|ACCID:2"/>
>
> <action application="set" data="origination_rates=0"/>
>
> <action application="set" data="original_caller_id_name=01226971834"/>
>
> <action application="set" data="original_caller_id_number=01226971834"/>
>
> <action application="set" data="effective_caller_id_name=01226971834"/>
>
> <action application="set" data="effective_caller_id_number=01226971834"/>
>
> <action application="set" data="receiver_accid=2"/>
>
> <action application="export"
> data="presence_data=x|||Dial9(9856937967)|||^03302290443.* // 03302290443
> // 1.30378||||||DID"/>
>
> <action application="export" data="call_type=0"/>
>
> <action application="set" data="calltype=DID-LOCAL"/>
>
> <action application="bridge" data="[leg_timeout=0]user/9856937967@
> ${domain_name}"/>
>
> <condition field="${cond(${user_data 9856937967@${domain_name} param
> vm-enabled} == true ? YES : NO)}" expression="^YES$">
>
> <action application="answer"/>
>
> <action application="export"
> data="voicemail_alternate_greet_id=03302290443"/>
>
> <action application="voicemail" data="default $${domain_name} 9856937967"/>
>
> <anti-action application="hangup" data="${originate_disposition}"/>
>
> </condition>
>
> </condition>
>
> </extension>
>
> </context>
>
> </section>
>
> </document>
>
> 2021-03-10 12:20:17.655679 [DEBUG] freeswitch_lua.cpp:382 DBH handle
> 0x7f3ebc1cae70 released.
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 parsing
> [default->03302290443] continue=false
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Regex (PASS) [03302290443]
> destination_number(03302290443) =~ /03302290443/ break=on-false
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> set(effective_destination_number=03302290443)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> set(bridge_pre_execute_bleg_app=sched_hangup)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> set(bridge_pre_execute_bleg_data=+6000 normal_clearing)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> set(callstart=2021-03-10 12:20:17)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> set(hangup_after_bridge=true)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> set(continue_on_fail=TRUE)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(account_id=13)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(parent_id=0)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(entity_id=3)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> set(call_processed=internal)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> set(call_direction=inbound)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> set(accountname=FreePBX)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> set(origination_rates_did=ID:1|CODE:^03302290443.*|DESTINATION:03302290443|CONNECTIONCOST:0.00000|INCLUDEDSECONDS:0|CT:0|COST:1.30378|INC:0|INITIALBLOCK:0|RATEGROUP:0|MARKUP:0|CI:200|ACCID:2)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> set(origination_rates=0)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> set(original_caller_id_name=01226971834)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> set(original_caller_id_number=01226971834)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> set(effective_caller_id_name=01226971834)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> set(effective_caller_id_number=01226971834)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action set(receiver_accid=2)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> export(presence_data=x|||Dial9(9856937967)|||^03302290443.* // 03302290443
> // 1.30378||||||DID)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action export(call_type=0)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> set(calltype=DID-LOCAL)
>
> Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> bridge([leg_timeout=0]user/9856937967@${domain_name})
>
> |--- Dialplan: Processing recursive conditions level:1
> [03302290443_recur_1] require-nested=TRUE
>
> |--- Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Regex (PASS)
> [03302290443_recur_1] ${cond(${user_data 9856937967@${domain_name} param
> vm-enabled} == true ? YES : NO)}(YES) =~ /^YES$/ break=on-false
>
> |--- Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action answer()
>
> |--- Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> export(voicemail_alternate_greet_id=03302290443)
>
> |--- Dialplan: sofia/sip-ip/01226971834 at 10.0.1.29 Action
> voicemail(default $${domain_name} 9856937967)
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_core_state_machine.c:286 (
> sofia/sip-ip/01226971834 at 10.0.1.29) State Change CS_ROUTING -> CS_EXECUTE
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_core_state_machine.c:643 (
> sofia/sip-ip/01226971834 at 10.0.1.29) State ROUTING going to sleep
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_core_state_machine.c:584 (
> sofia/sip-ip/01226971834 at 10.0.1.29) Running State Change CS_EXECUTE (Cur
> 1 Tot 1)
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_core_state_machine.c:650 (
> sofia/sip-ip/01226971834 at 10.0.1.29) State EXECUTE
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_sofia.c:198
> sofia/sip-ip/01226971834 at 10.0.1.29 SOFIA EXECUTE
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_core_state_machine.c:328
> sofia/sip-ip/01226971834 at 10.0.1.29 Standard EXECUTE
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29
> set(effective_destination_number=03302290443)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29
> [effective_destination_number]=[03302290443]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29
> set(bridge_pre_execute_bleg_app=sched_hangup)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29
> [bridge_pre_execute_bleg_app]=[sched_hangup]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29
> set(bridge_pre_execute_bleg_data=+6000 normal_clearing)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29 [bridge_pre_execute_bleg_data]=[+6000
> normal_clearing]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(callstart=2021-03-10
> 12:20:17)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29 [callstart]=[2021-03-10 12:20:17]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(hangup_after_bridge=true)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29 [hangup_after_bridge]=[true]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(continue_on_fail=TRUE)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29 [continue_on_fail]=[TRUE]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(account_id=13)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29 [account_id]=[13]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(parent_id=0)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29 [parent_id]=[0]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(entity_id=3)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29 [entity_id]=[3]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(call_processed=internal)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29 [call_processed]=[internal]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(call_direction=inbound)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29 [call_direction]=[inbound]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(accountname=FreePBX)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29 [accountname]=[FreePBX]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29
> set(origination_rates_did=ID:1|CODE:^03302290443.*|DESTINATION:03302290443|CONNECTIONCOST:0.00000|INCLUDEDSECONDS:0|CT:0|COST:1.30378|INC:0|INITIALBLOCK:0|RATEGROUP:0|MARKUP:0|CI:200|ACCID:2)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29
> [origination_rates_did]=[ID:1|CODE:^03302290443.*|DESTINATION:03302290443|CONNECTIONCOST:0.00000|INCLUDEDSECONDS:0|CT:0|COST:1.30378|INC:0|INITIALBLOCK:0|RATEGROUP:0|MARKUP:0|CI:200|ACCID:2]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(origination_rates=0)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29 [origination_rates]=[0]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29
> set(original_caller_id_name=01226971834)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29 [original_caller_id_name]=[01226971834]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29
> set(original_caller_id_number=01226971834)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29
> [original_caller_id_number]=[01226971834]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29
> set(effective_caller_id_name=01226971834)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29
> [effective_caller_id_name]=[01226971834]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29
> set(effective_caller_id_number=01226971834)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29
> [effective_caller_id_number]=[01226971834]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(receiver_accid=2)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29 [receiver_accid]=[2]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29
> export(presence_data=x|||Dial9(9856937967)|||^03302290443.* // 03302290443
> // 1.30378||||||DID)
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_channel.c:1296 EXPORT
> (export_vars) [presence_data]=[x|||Dial9(9856937967)|||^03302290443.* //
> 03302290443 // 1.30378||||||DID]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 export(call_type=0)
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_channel.c:1296 EXPORT
> (export_vars) [call_type]=[0]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 set(calltype=DID-LOCAL)
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_dptools.c:1548 SET
> sofia/sip-ip/01226971834 at 10.0.1.29 [calltype]=[DID-LOCAL]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 bridge([leg_timeout=0]
> user/9856937967 at 10.0.1.212)
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_channel.c:1250
> sofia/sip-ip/01226971834 at 10.0.1.29 EXPORTING[export_vars]
> [presence_data]=[x|||Dial9(9856937967)|||^03302290443.* // 03302290443 //
> 1.30378||||||DID] to event
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_channel.c:1250
> sofia/sip-ip/01226971834 at 10.0.1.29 EXPORTING[export_vars] [call_type]=[0]
> to event
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_ivr_originate.c:2142 Parsing
> global variables
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_ivr_originate.c:2669 Parsing
> session specific variables
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_channel.c:1250
> sofia/sip-ip/01226971834 at 10.0.1.29 EXPORTING[export_vars]
> [presence_data]=[x|||Dial9(9856937967)|||^03302290443.* // 03302290443 //
> 1.30378||||||DID] to event
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_channel.c:1250
> sofia/sip-ip/01226971834 at 10.0.1.29 EXPORTING[export_vars] [call_type]=[0]
> to event
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_ivr_originate.c:2142 Parsing
> global variables
>
> 2021-03-10 12:20:17.682238 [NOTICE] switch_ivr_originate.c:2851 Cannot
> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_ivr_originate.c:3848 Originate
> Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
>
> 2021-03-10 12:20:17.682238 [NOTICE] switch_ivr_originate.c:2851 Cannot
> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_ivr_originate.c:3848 Originate
> Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
>
> 2021-03-10 12:20:17.682238 [INFO] mod_dptools.c:3436 Originate Failed.
> Cause: USER_NOT_REGISTERED
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 answer()
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_core_media.c:6878 AUDIO RTP
> [sofia/sip-ip/01226971834 at 10.0.1.29] 10.0.1.212 port 16840 -> 10.0.1.29
> port 11144 codec: 0 ms: 20
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_rtp.c:4159 Not using a timer
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_core_media.c:7180
> sofia/sip-ip/01226971834 at 10.0.1.29 Set 2833 dtmf send payload to 101
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_core_media.c:7187
> sofia/sip-ip/01226971834 at 10.0.1.29 Set 2833 dtmf receive payload to 101
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_core_media.c:7210
> sofia/sip-ip/01226971834 at 10.0.1.29 Set rtp dtmf delay to 40
>
> 2021-03-10 12:20:17.682238 [DEBUG] mod_sofia.c:850 Local SDP
> sofia/sip-ip/01226971834 at 10.0.1.29:
>
> v=0
>
> o=FreeSWITCH 1615361977 1615361978 IN IP4 10.0.1.212
>
> s=FreeSWITCH
>
> c=IN IP4 10.0.1.212
>
> t=0 0
>
> m=audio 16840 RTP/AVP 0 101
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=ptime:20
>
> a=sendrecv
>
>
>
> 2021-03-10 12:20:17.682238 [NOTICE] mod_dptools.c:1312 Channel
> [sofia/sip-ip/01226971834 at 10.0.1.29] has been answered
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_channel.c:3773 (
> sofia/sip-ip/01226971834 at 10.0.1.29) Callstate Change RINGING -> ACTIVE
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29
> export(voicemail_alternate_greet_id=03302290443)
>
> 2021-03-10 12:20:17.682238 [DEBUG] switch_channel.c:1296 EXPORT
> (export_vars) [voicemail_alternate_greet_id]=[03302290443]
>
> EXECUTE sofia/sip-ip/01226971834 at 10.0.1.29 voicemail(default 10.0.1.212
> 9856937967)
>
> 2021-03-10 12:20:17.715099 [DEBUG] sofia.c:7084 Channel
> sofia/sip-ip/01226971834 at 10.0.1.29 entering state [completed][200]
>
> 2021-03-10 12:20:17.715099 [DEBUG] sofia.c:7084 Channel
> sofia/sip-ip/01226971834 at 10.0.1.29 entering state [ready][200]
>
> 2021-03-10 12:20:49.734765 [NOTICE] sofia.c:1012 Hangup
> sofia/sip-ip/01226971834 at 10.0.1.29 [CS_EXECUTE] [NORMAL_CLEARING]
>
> 2021-03-10 12:20:49.734765 [DEBUG] switch_ivr_play_say.c:70 No language
> specified - Using [en]
>
> 2021-03-10 12:20:49.734765 [DEBUG] switch_core_session.c:2815
> sofia/sip-ip/01226971834 at 10.0.1.29 skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>
> 2021-03-10 12:20:49.734765 [DEBUG] switch_core_state_machine.c:650 (
> sofia/sip-ip/01226971834 at 10.0.1.29) State EXECUTE going to sleep
>
> 2021-03-10 12:20:49.734765 [DEBUG] switch_core_state_machine.c:584 (
> sofia/sip-ip/01226971834 at 10.0.1.29) Running State Change CS_HANGUP (Cur 1
> Tot 1)
>
> 2021-03-10 12:20:49.734765 [DEBUG] switch_core_state_machine.c:850 (
> sofia/sip-ip/01226971834 at 10.0.1.29) Callstate Change ACTIVE -> HANGUP
>
> 2021-03-10 12:20:49.734765 [DEBUG] switch_core_state_machine.c:852 (
> sofia/sip-ip/01226971834 at 10.0.1.29) State HANGUP
>
> 2021-03-10 12:20:49.734765 [DEBUG] mod_sofia.c:438 Channel
> sofia/sip-ip/01226971834 at 10.0.1.29 hanging up, cause: NORMAL_CLEARING
>
> 2021-03-10 12:20:49.734765 [DEBUG] switch_core_state_machine.c:60
> sofia/sip-ip/01226971834 at 10.0.1.29 Standard HANGUP, cause: NORMAL_CLEARING
>
> 2021-03-10 12:20:49.734765 [DEBUG] switch_core_state_machine.c:852 (
> sofia/sip-ip/01226971834 at 10.0.1.29) State HANGUP going to sleep
>
> 2021-03-10 12:20:49.734765 [DEBUG] switch_core_state_machine.c:619 (
> sofia/sip-ip/01226971834 at 10.0.1.29) State Change CS_HANGUP -> CS_REPORTING
>
> 2021-03-10 12:20:49.734765 [DEBUG] switch_core_state_machine.c:584 (
> sofia/sip-ip/01226971834 at 10.0.1.29) Running State Change CS_REPORTING
> (Cur 1 Tot 1)
>
> 2021-03-10 12:20:49.734765 [DEBUG] switch_core_state_machine.c:938 (
> sofia/sip-ip/01226971834 at 10.0.1.29) State REPORTING
>
> 2021-03-10 12:20:49.734765 [INFO] mod_json_cdr.c:271 Process
> [f9793c2e-819a-11eb-b65f-8914a0e01ab2.cdr.json]
>
> 2021-03-10 12:20:49.794754 [DEBUG] switch_core_state_machine.c:174
> sofia/sip-ip/01226971834 at 10.0.1.29 Standard REPORTING, cause:
> NORMAL_CLEARING
>
> 2021-03-10 12:20:49.794754 [DEBUG] switch_core_state_machine.c:938 (
> sofia/sip-ip/01226971834 at 10.0.1.29) State REPORTING going to sleep
>
> 2021-03-10 12:20:49.794754 [DEBUG] switch_core_state_machine.c:610 (
> sofia/sip-ip/01226971834 at 10.0.1.29) State Change CS_REPORTING ->
> CS_DESTROY
>
> 2021-03-10 12:20:49.794754 [DEBUG] switch_core_session.c:1665 Session 1 (
> sofia/sip-ip/01226971834 at 10.0.1.29) Locked, Waiting on external entities
>
> 2021-03-10 12:20:49.794754 [NOTICE] switch_core_session.c:1683 Session 1 (
> sofia/sip-ip/01226971834 at 10.0.1.29) Ended
>
> 2021-03-10 12:20:49.794754 [NOTICE] switch_core_session.c:1687 Close
> Channel sofia/sip-ip/01226971834 at 10.0.1.29 [CS_DESTROY]
>
> 2021-03-10 12:20:49.794754 [DEBUG] switch_core_state_machine.c:741 (
> sofia/sip-ip/01226971834 at 10.0.1.29) Running State Change CS_DESTROY (Cur
> 0 Tot 1)
>
> 2021-03-10 12:20:49.794754 [DEBUG] switch_core_state_machine.c:751 (
> sofia/sip-ip/01226971834 at 10.0.1.29) State DESTROY
>
> 2021-03-10 12:20:49.794754 [DEBUG] mod_sofia.c:343
> sofia/sip-ip/01226971834 at 10.0.1.29 SOFIA DESTROY
>
> 2021-03-10 12:20:49.794754 [DEBUG] switch_core_state_machine.c:181
> sofia/sip-ip/01226971834 at 10.0.1.29 Standard DESTROY
>
> 2021-03-10 12:20:49.794754 [DEBUG] switch_core_state_machine.c:751 (
> sofia/sip-ip/01226971834 at 10.0.1.29) State DESTROY going to sleep
>
> freeswitch at astpp.myrevbill.com>
>
>
>
> Thanks!!
>
>
>
>
>
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From gidoramothra at gmail.com  Wed Mar 24 16:29:30 2021
From: gidoramothra at gmail.com (Stefan)
Date: Wed, 24 Mar 2021 17:29:30 +0100
Subject: [Freeswitch-users] Using TLS certificate with intermediate CA
In-Reply-To: <CANvSZQ8QGMcsLeSHKk3dsz8XtgZtDOf+=D399A3mZL723Ce2gA@mail.gmail.com>
References: <CANvSZQ8QGMcsLeSHKk3dsz8XtgZtDOf+=D399A3mZL723Ce2gA@mail.gmail.com>
Message-ID: <YFto6oBGrB/QIGvl@localhost.localdomain>

Hello, I also got certs from let's encrypt, and use a little script to make
freeswitch and the clients (polyphone, linphone, verto communicator)
happy. Just copy the contents of /etc/letsencrypt/live/your.host.name
to /etc/freeswitch/tls (or wherever your installation stores the certs)
and then do the following:

cat fullchain.pem privkey.pem > all.pem
ln -s all.pem tls.pem
ln -s all.pem agent.pem
ln -s all.pem wss.pem
ln -s all.pem dtls-srtp.pem

For me it works without even providing the real root ca cert, but if you
want that too, download it from letsencrypt like so:

wget -O ca.pem https://letsencrypt.org/certs/trustid-x3-root.pem.txt
cat chain.pem ca.pem > cafile.pem

Hope that works for You too. Polycoms need at least ucs v4.0.15 to
accept the letsencrypt certs (as far as I have tested it).

__
s.


On Wed, Mar 24, 2021 at 11:40:30AM +0100, Thilo-Alexander Ginkel wrote:
> Hello everyone,
> 
> I am currently struggling to get FreeSWITCH
> (1.10.5-release-17-25569c1631~64bit) to send the intermediate CA
> certificate for a Let's Encrypt X.509 certificate to be used for
> protecting SIPS traffic.
> 
> I included the certificate chain in agent.pem:
> 
> -- 8< --
> -----BEGIN EC PARAMETERS-----
> *REDACTED*
> -----END EC PARAMETERS-----
> -----BEGIN EC PRIVATE KEY-----
> *REDACTED*
> -----END EC PRIVATE KEY-----
> -----BEGIN CERTIFICATE-----
> *SERVER CERT*
> -----END CERTIFICATE-----
> 
> -----BEGIN CERTIFICATE-----
> *INTERMEDIATE CERT*
> -----END CERTIFICATE-----
> -- 8< --
> 
> Still, clients are complaining about an invalid CA and openssl s_client
> hints at only the server cert being sent in the server hello.
> 
> What did I miss?
> 
> Thanks,
> Thilo

> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users 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 nalmeida at gocontact.pt  Mon Mar 29 19:58:06 2021
From: nalmeida at gocontact.pt (nalmeida at gocontact.pt)
Date: Mon, 29 Mar 2021 20:58:06 +0100
Subject: [Freeswitch-users] BACKGROUND_JOB event received before action
	execution
In-Reply-To: <006401d724c5$4084a0f0$c18de2d0$@gocontact.pt>
References: <006401d724c5$4084a0f0$c18de2d0$@gocontact.pt>
Message-ID: <007101d724d5$d6939870$83bac950$@gocontact.pt>

Hi all.

 

I'm using ESL to send "bgapi" commands to Freeswitch and I was expecting to
receive the BACKGROUND_JOB event with the command's response after the
action was effectively executed, which is not the case.

 

I noticed that this happens with commands such as uuid_bridge, uuid_park,
uuid_transfer, etc. but it does not happen when executing an "originate",
where the BACKGROUND_JOB event is fired only after the call establishment or
rejection.

 

This leads to the execution of subsequent commands assuming that the call is
in a given state that might not be true.

 

Does someone faced a similar issue?

What should be the correct implementation here to make sure that the
subsequent logic is only executed when the current command was actually
done?

 

Thanks in advance! Regards, Nuno

 

 

Version: FreeSWITCH Version 1.10.2-release~64bit ( 64bit)

 

Example for uuid_bridge:

{"Event-Name":"API","Event-Date-Local":"2021-03-25
15:19:32","Event-Date-GMT":"Thu, 25 Mar 2021 15:19:32
GMT","Event-Date-Timestamp":"1616685572613681","Event-Calling-File":"switch_
loadable_module.c","Event-Calling-Function":"switch_api_execute","Event-Call
ing-Line-Number":"2992","Event-Sequence":"1548099","API-Command":"uuid_bridg
e","API-Command-Argument":"8112abfd-6d77-47ba-bedb-4748452abd78
0f693a5f-8e47-4aa0-b21a-5d5c95195e68"}

 

{"Event-Name":"BACKGROUND_JOB","Event-Date-Local":"2021-03-25
15:19:32","Event-Date-GMT":"Thu, 25 Mar 2021 15:19:32
GMT","Event-Date-Timestamp":"1616685572613681","Event-Calling-File":"mod_eve
nt_socket.c","Event-Calling-Function":"api_exec","Event-Calling-Line-Number"
:"1562","Event-Sequence":"1548100","Job-UUID":"bdcff921-7df3-4209-a6f6-6667f
39ac8a6","Job-Command":"uuid_bridge","Job-Command-Arg":"8112abfd-6d77-47ba-b
edb-4748452abd78
0f693a5f-8e47-4aa0-b21a-5d5c95195e68","Content-Length":"41","_body":"+OK
0f693a5f-8e47-4aa0-b21a-5d5c95195e68\n"}

 

{"Event-Name":"CHANNEL_EXECUTE","Event-Date-Local":"2021-03-25
15:19:33","Event-Date-GMT":"Thu, 25 Mar 2021 15:19:33
GMT","Event-Date-Timestamp":"1616685573913991","Event-Calling-File":"switch_
ivr_bridge.c","Event-Calling-Function":"uuid_bridge_on_soft_execute","Event-
Calling-Line-Number":"1241","Event-Sequence":"1548251","Application":"uuid_b
ridge","Application-Data":"0f693a5f-8e47-4aa0-b21a-5d5c95195e68",...}

 

{"Event-Name":"CHANNEL_EXECUTE","Event-Date-Local":"2021-03-25
15:19:33","Event-Date-GMT":"Thu, 25 Mar 2021 15:19:33
GMT","Event-Date-Timestamp":"1616685573913991","Event-Calling-File":"switch_
ivr_bridge.c","Event-Calling-Function":"uuid_bridge_on_soft_execute","Event-
Calling-Line-Number":"1248","Event-Sequence":"1548252","Application":"uuid_b
ridge","Application-Data":"8112abfd-6d77-47ba-bedb-4748452abd78"...}

 

{"Event-Name":"CHANNEL_BRIDGE","Event-Date-Local":"2021-03-25
15:19:33","Event-Date-GMT":"Thu, 25 Mar 2021 15:19:33
GMT","Event-Date-Timestamp":"1616685573913991","Event-Calling-File":"switch_
ivr_bridge.c","Event-Calling-Function":"switch_ivr_multi_threaded_bridge","E
vent-Calling-Line-Number":"1694","Event-Sequence":"1548253",...}

 

{"Event-Name":"CHANNEL_STATE","Event-Date-Local":"2021-03-25
15:19:33","Event-Date-GMT":"Thu, 25 Mar 2021 15:19:33
GMT","Event-Date-Timestamp":"1616685573913991","Event-Calling-File":"switch_
channel.c","Event-Calling-Function":"switch_channel_perform_set_running_stat
e","Event-Calling-Line-Number":"2341","Event-Sequence":"1548254",...}

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From david.villasmil.work at gmail.com  Mon Mar 29 21:57:32 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 29 Mar 2021 22:57:32 +0100
Subject: [Freeswitch-users] Internal Interface suddenly freezes
In-Reply-To: <VI1P190MB049439288D7951FC99D99C75867E9@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
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>
Message-ID: <CAFGRPVqGyM7vuYqHH-j_w1KUpQZjh1c7Jj8KFS5aYo8TH0=ZkA@mail.gmail.com>

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
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From s.kainz at wnt.at  Tue Mar 30 19:04:48 2021
From: s.kainz at wnt.at (Stefan Kainz)
Date: Tue, 30 Mar 2021 19:04:48 +0000
Subject: [Freeswitch-users] Internal Interface suddenly freezes
In-Reply-To: <CAFGRPVqGyM7vuYqHH-j_w1KUpQZjh1c7Jj8KFS5aYo8TH0=ZkA@mail.gmail.com>
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>
Message-ID: <E3DD2F0E-0153-4DA0-BEA2-A55ADBF4BB84@wnt.at>

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
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product 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
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com

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From mircea.huh at gmail.com  Tue Mar 30 19:20:36 2021
From: mircea.huh at gmail.com (Mircea Botoca-Huh)
Date: Tue, 30 Mar 2021 22:20:36 +0300
Subject: [Freeswitch-users] Internal Interface suddenly freezes
In-Reply-To: <E3DD2F0E-0153-4DA0-BEA2-A55ADBF4BB84@wnt.at>
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>
Message-ID: <CAL0pLpZD57B9safu17hd=Va0O0YRPkWZLdvSCV88ejcfYDAgYw@mail.gmail.com>

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
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From botelist at gmail.com  Tue Mar 30 19:49:54 2021
From: botelist at gmail.com (Bote Man)
Date: Tue, 30 Mar 2021 15:49:54 -0400
Subject: [Freeswitch-users] Internal Interface suddenly freezes
In-Reply-To: <CAL0pLpZD57B9safu17hd=Va0O0YRPkWZLdvSCV88ejcfYDAgYw@mail.gmail.com>
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>
Message-ID: <001801d7259d$db351020$919f3060$@gmail.com>

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 <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
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product 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

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From vishalmpai at gmail.com  Wed Mar 31 07:07:53 2021
From: vishalmpai at gmail.com (Vishal Pai)
Date: Wed, 31 Mar 2021 12:37:53 +0530
Subject: [Freeswitch-users] Ringless Voicemail
Message-ID: <CACOmXtAhFW5M7Oq3-=O5fW7k1PZXQBsfF6UGBXeXBHR4j0JnKA@mail.gmail.com>

Hello Everyone

These days lots of companies use/provide solutions for Ringless voicemail.
Also I found one url stating the RVM using freeswitch.

https://www.edoceo.com/dev/call-to-voicemail

Can we use freeswitch for it. or it is provided by the carrier/gateway to
drop messages directly in the VM bypass ring.

In the above URL if we refer then 2 calls are being made to make 1 line
busy and second to drop vim.


Last question: Is it being legal nowadays?

Thanks
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From s.kainz at wnt.at  Wed Mar 31 07:16:25 2021
From: s.kainz at wnt.at (Stefan Kainz)
Date: Wed, 31 Mar 2021 07:16:25 +0000
Subject: [Freeswitch-users] Internal Interface suddenly freezes
In-Reply-To: <001801d7259d$db351020$919f3060$@gmail.com>
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>
Message-ID: <9B9FC99A-90CD-4BC5-8B52-FE407CF759DE@wnt.at>

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
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product 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
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From grcamauer at gmail.com  Wed Mar 31 14:13:19 2021
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Wed, 31 Mar 2021 11:13:19 -0300
Subject: [Freeswitch-users] Ringless Voicemail
In-Reply-To: <CACOmXtAhFW5M7Oq3-=O5fW7k1PZXQBsfF6UGBXeXBHR4j0JnKA@mail.gmail.com>
References: <CACOmXtAhFW5M7Oq3-=O5fW7k1PZXQBsfF6UGBXeXBHR4j0JnKA@mail.gmail.com>
Message-ID: <CAPNTnnp920=UUJ=35qgxn4RPtd2_+G+hXVxHsyrhTbmhfMF80A@mail.gmail.com>

Vishal,

If you use one line to make the line busy you are taking the "Ringless" out
of it...  I doubt the end user will appreciate that first call.
I believe that true Ringless voicemail involves APIs with the carriers that
allow the messages to be inserted directly into the voicemail systems.
Those APIs are accessed only by "partner"companies (companies with which
they have signed agreements).

Regards,

Guillermo



On Wed, Mar 31, 2021 at 4:43 AM Vishal Pai <vishalmpai at gmail.com> wrote:

> Hello Everyone
>
> These days lots of companies use/provide solutions for Ringless voicemail.
> Also I found one url stating the RVM using freeswitch.
>
> https://www.edoceo.com/dev/call-to-voicemail
>
> Can we use freeswitch for it. or it is provided by the carrier/gateway to
> drop messages directly in the VM bypass ring.
>
> In the above URL if we refer then 2 calls are being made to make 1 line
> busy and second to drop vim.
>
>
> Last question: Is it being legal nowadays?
>
> 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
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From mike at freeswitch.org  Wed Mar 31 15:39:53 2021
From: mike at freeswitch.org (Mike Jerris)
Date: Wed, 31 Mar 2021 09:39:53 -0600
Subject: [Freeswitch-users] Ringless Voicemail
In-Reply-To: <CAPNTnnp920=UUJ=35qgxn4RPtd2_+G+hXVxHsyrhTbmhfMF80A@mail.gmail.com>
References: <CACOmXtAhFW5M7Oq3-=O5fW7k1PZXQBsfF6UGBXeXBHR4j0JnKA@mail.gmail.com>
 <CAPNTnnp920=UUJ=35qgxn4RPtd2_+G+hXVxHsyrhTbmhfMF80A@mail.gmail.com>
Message-ID: <D43896F4-CDE8-4467-A9D9-8DA786EEC996@freeswitch.org>

This is all correct. In practice what you will “usually” see is an almost immediate missed call indication on the phone (and in my testing my phone rarely made a ring sound at all) as that first call is hung up very quickly… usually… except when it doesn’t happen like that… which you have no control over...


> On Mar 31, 2021, at 8:13 AM, Guillermo Ruiz Camauer <grcamauer at gmail.com> wrote:
> 
> Vishal,
> 
> If you use one line to make the line busy you are taking the "Ringless" out of it...  I doubt the end user will appreciate that first call.
> I believe that true Ringless voicemail involves APIs with the carriers that allow the messages to be inserted directly into the voicemail systems.
> Those APIs are accessed only by "partner"companies (companies with which they have signed agreements).  
> 
> Regards,
> 
> Guillermo
> 
> 
> 
> On Wed, Mar 31, 2021 at 4:43 AM Vishal Pai <vishalmpai at gmail.com <mailto:vishalmpai at gmail.com>> wrote:
> Hello Everyone
> 
> These days lots of companies use/provide solutions for Ringless voicemail. Also I found one url stating the RVM using freeswitch.
> 
> https://www.edoceo.com/dev/call-to-voicemail <https://www.edoceo.com/dev/call-to-voicemail>
> 
> Can we use freeswitch for it. or it is provided by the carrier/gateway to drop messages directly in the VM bypass ring.
> 
> In the above URL if we refer then 2 calls are being made to make 1 line busy and second to drop vim.
> 
> 
> Last question: Is it being legal nowadays?
> 
> Thanks

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From s.kainz at wnt.at  Wed Mar 31 22:10:54 2021
From: s.kainz at wnt.at (Stefan Kainz)
Date: Wed, 31 Mar 2021 22:10:54 +0000
Subject: [Freeswitch-users] Internal Interface suddenly freezes
In-Reply-To: <9B9FC99A-90CD-4BC5-8B52-FE407CF759DE@wnt.at>
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>
Message-ID: <E9072A99-2776-4124-9D22-D8AD99D0B6B5@wnt.at>

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<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
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product 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
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com

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