From nneul at mst.edu  Thu Feb  1 00:14:24 2018
From: nneul at mst.edu (Nathan Neulinger)
Date: Wed, 31 Jan 2018 18:14:24 -0600
Subject: [Freeswitch-users] The next version of FreeSWITCH will be...?
In-Reply-To: <CAEJMVkAfesifwd9PFQE89FectnL5XOUHz7V+winEsb3OG5XexA@mail.gmail.com>
References: <f31724f6-23c6-066f-3cbb-e7876badd934@hkcradio.com>
 <CAEJMVkAfesifwd9PFQE89FectnL5XOUHz7V+winEsb3OG5XexA@mail.gmail.com>
Message-ID: <2d5d02c9-726d-2086-81cf-7fe3890a8d6b@mst.edu>

Note that there is at least one merge/pull request that hasn't been processed adding a couple of additional allison 
sound files sitting out on Stash.

-- Nathan

On 1/31/18 4:59 PM, Brian West wrote:
> The Allison files are published here:
> 
> https://files.freeswitch.org/releases/sounds/
> 
> /b
> 
> On Wed, Jan 31, 2018 at 10:31 AM, Nick Giannak III <nick at hkcradio.com <mailto:nick at hkcradio.com>> wrote:
> 
>     Greetings.
> 
>          Last year, there was a lot of talk about FreeSWITCH 1.8, especially given that a new set of prompts from
>     Allison Smith was going to be included with that version. As an audio guy, I'm excited about this by comparison to
>     the prompts we already have, but I digress.
> 
>          However, a lot of that talk seems to have quietened down, and version 1.9 is shown as development on the
>     FreeSWITCH website. And yet, no 1.8 release materialized. So I'm asking for a status report...I'm confused about
>     what's coming next. Please alleviate my confusion?
> 
>     Thanks in advance,
> 
>     Nick
> 
> 
> 
> 
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com>
> 
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org <http://confluence.freeswitch.org>
>     http://www.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>
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> 
> 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/>
> 
> color-facebook-96.png <https://www.facebook.com/freeswitch/>color-twitter-96.png 
> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
> 

-- 
------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect


From loi.dangthanh at gmail.com  Thu Feb  1 03:03:42 2018
From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=)
Date: Thu, 01 Feb 2018 03:03:42 +0000
Subject: [Freeswitch-users] Memory Leaks in calling python script using API
Message-ID: <CAGsrq_tos0JJiL18zRTwhrp1XQ=ykM8C_5QvAxwCZCN=-7=mtg@mail.gmail.com>

Hi all, I reproduced my memory leaks to the simplest case
in xml dialplan:
<action application="python" data="fspy.memtesting"/>
<action application="hangup" data="CALL_REJECTED"/>
in memtesting.py:
api = freeswitch.API()
gwdown = api.executeString("sofia profile external gwlist down")

This is just to execute the api from the python script, and then the memory
usage from top keep growing and never drop down during my running of ~20000
calls.
No memory leaks in executing the api alone from xml dialplan:

<action application="eval" data="${sofia profile external gwlist down}}"/>

reproduced in both 1.6.17 (my running version) and 1.6.19
I'm using up to date CentOS Linux release 7.4.1708 (Core).
Any helps would be appreciated.

rgds,
Loi Dang
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From alex at freeswitch.com  Thu Feb  1 03:59:31 2018
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Thu, 1 Feb 2018 12:59:31 +0900
Subject: [Freeswitch-users] Memory Leaks in calling python script using
	API
In-Reply-To: <CAGsrq_tos0JJiL18zRTwhrp1XQ=ykM8C_5QvAxwCZCN=-7=mtg@mail.gmail.com>
References: <CAGsrq_tos0JJiL18zRTwhrp1XQ=ykM8C_5QvAxwCZCN=-7=mtg@mail.gmail.com>
Message-ID: <CAGJLwFjx4s=WvLMsrG67Wfno9Gs4RBCQ5asj4RG5cGQTOJpjLQ@mail.gmail.com>

Then you need to reproduce this problem on Debian 8 with latest master git
code. If it still exist then search through Jira on similar issues and
comment them. If no issues are present then open new issue after careful
and thoughtful reading of this
https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA

On Thu, Feb 1, 2018 at 12:03 PM, Lợi Đặng <loi.dangthanh at gmail.com> wrote:

> Hi all, I reproduced my memory leaks to the simplest case
> in xml dialplan:
> <action application="python" data="fspy.memtesting"/>
> <action application="hangup" data="CALL_REJECTED"/>
> in memtesting.py:
> api = freeswitch.API()
> gwdown = api.executeString("sofia profile external gwlist down")
>
> This is just to execute the api from the python script, and then the
> memory usage from top keep growing and never drop down during my running of
> ~20000 calls.
> No memory leaks in executing the api alone from xml dialplan:
>
> <action application="eval" data="${sofia profile external gwlist down}}"/>
>
> reproduced in both 1.6.17 (my running version) and 1.6.19
> I'm using up to date CentOS Linux release 7.4.1708 (Core).
> Any helps would be appreciated.
>
> rgds,
> Loi Dang
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

Alex Sibyakin | Support Engineer
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: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.
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From markus at ply.com  Thu Feb  1 15:42:27 2018
From: markus at ply.com (Markus Badberg -PLY)
Date: Thu, 01 Feb 2018 16:42:27 +0100
Subject: [Freeswitch-users] Trouble connecting Sangoma Vega 50 BRI to
 Freeswitch
In-Reply-To: <39A31E19-B763-48C1-BD8C-0A67FF924026@vallimamod.org>
References: <1516892941193-0.post@n2.nabble.com>
 <CAO_GGAptnpq6arU0VRsXQOSZk6-P-Zz4SzDkOiAwnFCtxREs6A@mail.gmail.com>
 <D690CA81.489E%markus@ply.com>
 <39A31E19-B763-48C1-BD8C-0A67FF924026@vallimamod.org>
Message-ID: <D698F315.4A94%markus@ply.com>


Thanks, i fixed many things. Now i have a non-registrar connection working.

But yet i have another problem with outbound routes.

I configured this regex to match all numbers more then 2 numbers routed to
gateway -> 

(^\d{3,15}$)


But it won¹t work.

Here the output from the logfile of the interesting section:

2018-02-01 16:25:58.140689 [INFO] mod_dialplan_xml.c:637 Processing Zentrale
-PLY <00>->01763426XXXX in context public freeswitch at greg-ph01-pbx> [DEBUG]
esl.c:1316 esl_recv_event() RECV HEADER [Content-Type] = [log/data] [DEBUG]
esl.c:1316 esl_recv_event() RECV HEADER [Content-Length] = [87] [DEBUG]
esl.c:1316 esl_recv_event() RECV HEADER [Log-Level] = [7] [DEBUG] esl.c:1316
esl_recv_event() RECV HEADER [Text-Channel] = [1] [DEBUG] esl.c:1316
esl_recv_event() RECV HEADER [Log-File] = [mod_dialplan_xml.c] [DEBUG]
esl.c:1316 esl_recv_event() RECV HEADER [Log-Func] = [dialplan_hunt] [DEBUG]
esl.c:1316 esl_recv_event() RECV HEADER [Log-Line] = [694] [DEBUG]
esl.c:1316 esl_recv_event() RECV HEADER [User-Data] =
[9896a97c-796d-4e45-84c1-baad7f48f4b0] [DEBUG] esl.c:1480 esl_recv_event()
RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: log/data Content-Length:
87 Log-Level: 7 Text-Channel: 1 Log-File: mod_dialplan_xml.c Log-Func:
dialplan_hunt Log-Line: 694 User-Data: 9896a97c-796d-4e45-84c1-baad7f48f4b0
Content-Length: 87 
Dialplan: sofia/internal/00 at 192.168.170.204 parsing [public->32595XXXX]
continue=false 
Dialplan: sofia/internal/00 at 192.168.170.204 parsing [public->32595XXXX]
continue=false freeswitch at greg-ph01-pbx> [DEBUG] esl.c:1316 esl_recv_event()
RECV HEADER [Content-Type] = [log/data] [DEBUG] esl.c:1316 esl_recv_event()
RECV HEADER [Content-Length] = [136] [DEBUG] esl.c:1316 esl_recv_event()
RECV HEADER [Log-Level] = [7] [DEBUG] esl.c:1316 esl_recv_event() RECV
HEADER [Text-Channel] = [1] [DEBUG] esl.c:1316 esl_recv_event() RECV HEADER
[Log-File] = [mod_dialplan_xml.c] [DEBUG] esl.c:1316 esl_recv_event() RECV
HEADER [Log-Func] = [parse_exten] [DEBUG] esl.c:1316 esl_recv_event() RECV
HEADER [Log-Line] = [424] [DEBUG] esl.c:1316 esl_recv_event() RECV HEADER
[User-Data] = [9896a97c-796d-4e45-84c1-baad7f48f4b0] [DEBUG] esl.c:1480
esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: log/data
Content-Length: 136 Log-Level: 7 Text-Channel: 1 Log-File:
mod_dialplan_xml.c Log-Func: parse_exten Log-Line: 424 User-Data:
9896a97c-796d-4e45-84c1-baad7f48f4b0 Content-Length: 136
Dialplan: sofia/internal/00 at 192.168.170.204 Regex (FAIL) [32595XXXX]
destination_number(01763426XXXX) =~ /^(32595XXXX)$/ break=on-false
Dialplan: sofia/internal/00 at 192.168.170.204 Regex (FAIL) [32595XXXX]
destination_number(01763426XXXX) =~ /^(32595XXXX)$/ break=on-false
freeswitch at greg-ph01-pbx> [DEBUG] esl.c:1316 esl_recv_event() RECV HEADER
[Content-Type] = [log/data] [DEBUG] esl.c:1316 esl_recv_event() RECV HEADER
[Content-Length] = [85] [DEBUG] esl.c:1316 esl_recv_event() RECV HEADER
[Log-Level] = [6] [DEBUG] esl.c:1316 esl_recv_event() RECV HEADER
[Text-Channel] = [3] [DEBUG] esl.c:1316 esl_recv_event() RECV HEADER
[Log-File] = [switch_core_state_machine.c] [DEBUG] esl.c:1316
esl_recv_event() RECV HEADER [Log-Func] = [switch_core_standard_on_routing]
[DEBUG] esl.c:1316 esl_recv_event() RECV HEADER [Log-Line] = [311] [DEBUG]
esl.c:1316 esl_recv_event() RECV HEADER [User-Data] =
[9896a97c-796d-4e45-84c1-baad7f48f4b0] [DEBUG] esl.c:1480 esl_recv_event()
RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: log/data Content-Length:
85 Log-Level: 6 Text-Channel: 3 Log-File: switch_core_state_machine.c
Log-Func: switch_core_standard_on_routing Log-Line: 311 User-Data:
9896a97c-796d-4e45-84c1-baad7f48f4b0 Content-Length: 85
2018-02-01 16:25:58.160687 [INFO] switch_core_state_machine.c:311 No Route,
Aborting 
2018-02-01 16:25:58.160687 [INFO] switch_core_state_machine.c:311 No Route,
Aborting freeswitch at greg-ph01-pbx> [DEBUG] esl.c:1316 esl_recv_event() RECV
HEADER [Content-Type] = [log/data] [DEBUG] esl.c:1316 esl_recv_event() RECV
HEADER [Content-Length] = [145] [DEBUG] esl.c:1316 esl_recv_event() RECV
HEADER [Log-Level] = [5] [DEBUG] esl.c:1316 esl_recv_event() RECV HEADER
[Text-Channel] = [0] [DEBUG] esl.c:1316 esl_recv_event() RECV HEADER
[Log-File] = [switch_core_state_machine.c] [DEBUG] esl.c:1316
esl_recv_event() RECV HEADER [Log-Func] = [switch_core_standard_on_routing]
[DEBUG] esl.c:1316 esl_recv_event() RECV HEADER [Log-Line] = [312] [DEBUG]
esl.c:1316 esl_recv_event() RECV HEADER [User-Data] =
[9896a97c-796d-4e45-84c1-baad7f48f4b0] [DEBUG] esl.c:1480 esl_recv_event()
RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: log/data Content-Length:
145 Log-Level: 5 Text-Channel: 0 Log-File: switch_core_state_machine.c
Log-Func: switch_core_standard_on_routing Log-Line: 312 User-Data:
9896a97c-796d-4e45-84c1-baad7f48f4b0 Content-Length: 145
2018-02-01 16:25:58.160687 [NOTICE] switch_core_state_machine.c:312 Hangup
sofia/internal/00 at 192.168.170.204 [CS_ROUTING] [NO_ROUTE_DESTINATION]
2018-02-01 16:25:58.160687 [NOTICE] switch_core_state_machine.c:312 Hangup
sofia/internal/00 at 192.168.170.204 [CS_ROUTING] [NO_ROUTE_DESTINATION]
freeswitch at greg-ph01-pbx> [DEBUG] esl.c:1316 esl_recv_event() RECV HEADER
[Content-Type] = [log/data] [DEBUG] esl.c:1316 esl_recv_event() RECV HEADER
[Content-Length] = [132] [DEBUG] esl.c:1316 esl_recv_event() RECV HEADER
[Log-Level] = [7] [DEBUG] esl.c:1316 esl_recv_event() RECV HEADER
[Text-Channel] = [3] [DEBUG] esl.c:1316 esl_recv_event() RECV HEADER
[Log-File] = [switch_core_state_machine.c] [DEBUG] esl.c:1316
esl_recv_event() RECV HEADER [Log-Func] = [switch_core_session_run] [DEBUG]
esl.c:1316 esl_recv_event() RECV HEADER [Log-Line] = [643] [DEBUG]
esl.c:1316 esl_recv_event() RECV HEADER [User-Data] =
[9896a97c-796d-4e45-84c1-baad7f48f4b0] [DEBUG] esl.c:1480 esl_recv_event()
RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: log/data Content-Length:
132 Log-Level: 7 Text-Channel: 3 Log-File: switch_core_state_machine.c
Log-Func: switch_core_session_run Log-Line: 643 User-Data:
9896a97c-796d-4e45-84c1-baad7f48f4b0 Content-Length: 132

Von:  FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on
behalf of Vallimamod Abdullah <vma at vallimamod.org>
Antworten an:  FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Datum:  Freitag, 26. Januar 2018 um 18:29
An:  FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Betreff:  Re: [Freeswitch-users] Trouble connecting Sangoma Vega 50 BRI to
Freeswitch

Hi,

It looks that you have a problem with your config:

> nta.c:10803 outgoing_query_a() nta: for "9bde7ac0-4a8f-4937-ab2e-0c213e8c1d2f"
> query "9bde7ac0-4a8f-4937-ab2e-0c213e8c1d2f" A (cached)

Sofia is looking to resolve the "9bde7ac0-4a8f-4937-ab2e-0c213e8c1d2f"
domain, which obviously fails...

Btw, you don't have to worry about NAPTR or SRV records as if they are not
found, sofia falls back to A record.

Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sipsolutions.fr
.

> On 26 Jan 2018, at 12:11, Markus Badberg -PLY <markus at ply.com> wrote:
> 
> 
> It seems, that it has something to do with NAPTR Records.
> 
> I¹ve no clue about that. I¹m running a simple DNS-Resolver on pfsense. No full
> DNS Server. Maybe could be this the problem?
> How do i fix NAPTR Records?
> 
> Here is the logging output from freeswitch:
> tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7f529c0042c0): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7f529c0042c0)
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7f529c0042c0) msg
> 0x7f529c003230 from (udp/192.168.170.204:5060) has 4 bytes, veclen = 1
> tport.c:3023 tport_deliver() tport_deliver(0x7f529c0042c0): bad msg
> 0x7f529c003230 (4 bytes) from udp/192.168.170.10:5060/sip next=(nil)
> 2018-01-26 12:05:56.939301 [NOTICE] sofia_reg.c:448 Registering
> 9bde7ac0-4a8f-4937-ab2e-0c213e8c1d2f
> nua.c:622 nua_register() nua: nua_register: entering
> nua_stack.c:569 nua_stack_signal() nua(0x7f5294024b60): recv signal r_register
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:403 soa_set_params() soa_set_params(static::0x7f529c0360a0, ...) called
> nua_dialog.c:338 nua_dialog_usage_add() nua(0x7f5294024b60): adding register
> usage
> nua_stack.c:529 nua_signal() nua(0x7f5294024b60): sent signal r_register
> nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7f529c038a40)
> nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
> sres_cache.c:272 sres_cache_get() sres_cache_get(0x7f529c002c40, SRV,
> "_sip._udp.9bde7ac0-4a8f-4937-ab2e-0c213e8c1d2f.") called
> sres_cache.c:318 sres_cache_get() sres_cache_get(0x7f529c002c40, SRV,
> "_sip._udp.9bde7ac0-4a8f-4937-ab2e-0c213e8c1d2f.") returned 1 entries
> nta.c:10598 outgoing_query_srv() nta: for
> "9bde7ac0-4a8f-4937-ab2e-0c213e8c1d2f" query
> "_sip._udp.9bde7ac0-4a8f-4937-ab2e-0c213e8c1d2f" SRV (cached)
> sres_cache.c:272 sres_cache_get() sres_cache_get(0x7f529c002c40, A,
> "9bde7ac0-4a8f-4937-ab2e-0c213e8c1d2f.") called
> sres_cache.c:318 sres_cache_get() sres_cache_get(0x7f529c002c40, A,
> "9bde7ac0-4a8f-4937-ab2e-0c213e8c1d2f.") returned 1 entries
> nta.c:10803 outgoing_query_a() nta: for "9bde7ac0-4a8f-4937-ab2e-0c213e8c1d2f"
> query "9bde7ac0-4a8f-4937-ab2e-0c213e8c1d2f" A (cached)
> nta.c:1348 set_timeout() nta: timer shortened to 5000 ms
> nua_stack.c:271 nua_stack_event() nua(0x7f5294024b60): event r_register 503
> DNS Error
> nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f5294024b60): removing
> register usage
> nta.c:4470 nta_leg_destroy() nta_leg_destroy(0x7f529c038a40)
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2018-01-26 12:05:56.939301 [ERR] sofia_reg.c:2447
> 9bde7ac0-4a8f-4937-ab2e-0c213e8c1d2f Failed Registration with status DNS Error
> [503]. failure #99
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2018-01-26 12:05:57.939301 [WARNING] sofia_reg.c:505
> 9bde7ac0-4a8f-4937-ab2e-0c213e8c1d2f Failed Registration [503], setting retry
> to 30 seconds.
> nta.c:9101 outgoing_timer_dk() nta: timer K fired, terminate REGISTER
> (118146314)
> 
> 
> greetings
> Von:  FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on
> behalf of Volodymyr Fedorov <lexxua at gmail.com>
> Antworten an:  FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Datum:  Freitag, 26. Januar 2018 um 10:25
> An:  FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Betreff:  Re: [Freeswitch-users] Trouble connecting Sangoma Vega 50 BRI to
> Freeswitch
> 
> Hi, maybe pcap with register towards to Freeswitch will be useful.
> I had vega euro 50 with FXS ports and combination works without any issue.
> 
> On Thu, Jan 25, 2018 at 4:09 PM, badhills <markus at ply.com> wrote:
>> Hi,
>> 
>> i have trouble connecting my Sangoma vega 50 BRI to Freeswitch.
>> 
>> I setup the Vega in quick setup, with user and password.
>> 
>> Then i setup the gateway in freeswitch. But both sides are getting forbidden
>> and unauthorized messages.
>> In the freeswitch log is a dns failure, but dns seems to working great. I
>> can't find issues there.
>> 
>> Unfortunately, I haven't found anything suitable yet and I'm getting
>> desperate. It can't be that hard. If I configure extensions, it works fine
>> and I can make internal calls with 2 or more phones. The connection to the
>> gateway isn't very different, is it?
>> 
>> Under Asterisk with Freepbx I already got this to work, but I wanted to try
>> it with FreeSwitch.
>> 
>> Had anybody a working connection and could explain it a lil bit to me?
>> 
>> kind regards,
>> Markus
>> 
>> 
>> 
>> --
>> Sent from: http://freeswitch-users.2379917.n2.nabble.com/
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <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 <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Best regards,
> Volodymyr
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.orghttp://www.freeswitchsolutions.com
> <http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listi
> nfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp:/
> /www.freeswitch.org <http://www.freeswitch.org/>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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

_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
http://www.freeswitchsolutions.com Official FreeSWITCH Sites
http://www.freeswitch.org http://confluence.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 kinshuk1989 at gmail.com  Thu Feb  1 10:53:23 2018
From: kinshuk1989 at gmail.com (Kinshuk Bairagi)
Date: Thu, 1 Feb 2018 16:23:23 +0530
Subject: [Freeswitch-users] Inbound Event Socket Making Calls with UUID
	Tracking
Message-ID: <CACa8_yqNnA7SG=HAyW1r2T-A1TXQ-sXkJ-joK_73aeB0k_jp3g@mail.gmail.com>

Hi

I am trying to track the status of the calls while making call via
esl_inbound socket interface, but while making call, the call forks into
multiple legs and new uuid's are generated without any ability to track
them. Is there something wrong?

Command Issued :
originate
{origination_uuid=95622d76-9a98-48d7-a53b-6f55598d78e1,origination_caller_id_number=9876,origination_caller_id_number=0000,originate_timeout=30}sofia/internal/
1001 at 10.85.185.65 &park()

This causes multiple call-legs, with 3  uuids : Portal Screenshot
https://i.imgur.com/ahPdvFK.png

Log : https://gist.github.com/kingster/88d9e64368e3a2af4a6f603d81100457

How can I track co-relation between this call-legs? What is the reason that
3 call legs being created, any way to skip that?

Thanks
Kinshuk
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From speech at pobox.com  Thu Feb  1 14:21:44 2018
From: speech at pobox.com (M Yudkowsky)
Date: Thu, 1 Feb 2018 08:21:44 -0600
Subject: [Freeswitch-users] Skype-in no longer accepts Toll-Free PAI or even
	From
Message-ID: <544CB9F5-7E1A-4859-9EA4-35D1E5071DF1@pobox.com>

Comrades!

Over the past seven years I've dialed Skype-in numbers using a From of a toll DID, with a PAI of a toll-free TN along with a name. This sets the termination display to show the name and the toll-free TN, and informs them who is calling.

As of sometime on 2018-01-23, this seems to have stopped working. Details are here <https://stackoverflow.com/questions/48551204>. TL;DR -- if you send a toll-free TN in the From or PAI, the call fails with a 408.

Really, I'm trying to figure out if other people are having this problem, although I believe it's upstream of my SIP provider. And if someone knows of a workaround. I may send a toll DID in the meantime, but that's far less than optimal. 


-- 
  Moshe Yudkowsky
  Disaggregate Corporation
  2952 W Fargo
  Chicago, IL 60645 USA


   http://www.Disaggregate.com



From mike at jerris.com  Thu Feb  1 18:01:56 2018
From: mike at jerris.com (Michael Jerris)
Date: Thu, 1 Feb 2018 13:01:56 -0500
Subject: [Freeswitch-users] Inbound Event Socket Making Calls with UUID
 Tracking
In-Reply-To: <CACa8_yqNnA7SG=HAyW1r2T-A1TXQ-sXkJ-joK_73aeB0k_jp3g@mail.gmail.com>
References: <CACa8_yqNnA7SG=HAyW1r2T-A1TXQ-sXkJ-joK_73aeB0k_jp3g@mail.gmail.com>
Message-ID: <F94C0D76-F576-4D00-90ED-65689CAA2427@jerris.com>

you are making an outbound call to your own box, so the three calls are the outbound call, the inbound of the same call (which will of course have a different uuid), and i assume the 3rd is whatever happens with that inbound leg.  In this case, its tricky to track them, but also totally unnecessary to make a call that way.  Why are you looping back to yourself with a sip call instead of calling the correct intended destination to start with?

> On Feb 1, 2018, at 5:53 AM, Kinshuk Bairagi <kinshuk1989 at gmail.com> wrote:
> 
> Hi
> 
> I am trying to track the status of the calls while making call via esl_inbound socket interface, but while making call, the call forks into multiple legs and new uuid's are generated without any ability to track them. Is there something wrong?
> 
> Command Issued : 
> originate {origination_uuid=95622d76-9a98-48d7-a53b-6f55598d78e1,origination_caller_id_number=9876,origination_caller_id_number=0000,originate_timeout=30}sofia/internal/1001 at 10.85.185.65 <mailto:1001 at 10.85.185.65> &park()
> 
> This causes multiple call-legs, with 3  uuids : Portal Screenshot  https://i.imgur.com/ahPdvFK.png <https://i.imgur.com/ahPdvFK.png>
> 
> Log : https://gist.github.com/kingster/88d9e64368e3a2af4a6f603d81100457 <https://gist.github.com/kingster/88d9e64368e3a2af4a6f603d81100457>
> 
> How can I track co-relation between this call-legs? What is the reason that 3 call legs being created, any way to skip that?
> 
> Thanks
> Kinshuk

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From mike at jerris.com  Thu Feb  1 19:07:41 2018
From: mike at jerris.com (Michael Jerris)
Date: Thu, 1 Feb 2018 14:07:41 -0500
Subject: [Freeswitch-users] Skype-in no longer accepts Toll-Free PAI or
 even From
In-Reply-To: <544CB9F5-7E1A-4859-9EA4-35D1E5071DF1@pobox.com>
References: <544CB9F5-7E1A-4859-9EA4-35D1E5071DF1@pobox.com>
Message-ID: <82FA3536-B338-4580-BDDE-E66440543E31@jerris.com>

The workaround would be to send a toll number.  The fix would be to ask Skype or whatever carrier is rejecting the call to fix it.

> On Feb 1, 2018, at 9:21 AM, M Yudkowsky <speech at pobox.com> wrote:
> 
> Comrades!
> 
> Over the past seven years I've dialed Skype-in numbers using a From of a toll DID, with a PAI of a toll-free TN along with a name. This sets the termination display to show the name and the toll-free TN, and informs them who is calling.
> 
> As of sometime on 2018-01-23, this seems to have stopped working. Details are here <https://stackoverflow.com/questions/48551204>. TL;DR -- if you send a toll-free TN in the From or PAI, the call fails with a 408.
> 
> Really, I'm trying to figure out if other people are having this problem, although I believe it's upstream of my SIP provider. And if someone knows of a workaround. I may send a toll DID in the meantime, but that's far less than optimal. 



From speech at pobox.com  Thu Feb  1 19:18:12 2018
From: speech at pobox.com (M Yudkowsky)
Date: Thu, 1 Feb 2018 13:18:12 -0600
Subject: [Freeswitch-users] Skype-in no longer accepts Toll-Free PAI or
 even From
Message-ID: <C50968E1-525D-4128-B65E-B8E2C0547E00@pobox.com>

> 
> On Feb 1, 2018, at 13:07 , Michael Jerris <mike at jerris.com> wrote:
> 
> The workaround would be to send a toll number.  The fix would be to ask Skype or whatever carrier is rejecting the call to fix it.

Well, sending a toll number would be nice, and was my plan B or C, except that everyone receiving the call would be puzzled as to who's calling them -- bad for business.

Actually, I was just about to write: turns out that this is *not* a Skype-in number problem. The "obvious" thing to do, and I was too focused on tech-side to do it, was to try other non-toll numbers in the DID; e.g. 1 800 111 2222. That works, btw, since PAI's are never validated in my experience to date. As such I've determined that indeed, as Mr J states, this problem is specific to my TN.

Now I just have to find the right people at Skype to unblock my TN. I wonder how long it takes to get that done.
-- 
  Moshe Yudkowsky
  Disaggregate Corporation
  2952 W Fargo
  Chicago, IL 60645 USA

   http://www.Disaggregate.com
   http://www.PebbleAndAvalanche.com








From mike at jerris.com  Thu Feb  1 19:56:43 2018
From: mike at jerris.com (Michael Jerris)
Date: Thu, 1 Feb 2018 14:56:43 -0500
Subject: [Freeswitch-users] Skype-in no longer accepts Toll-Free PAI or
 even From
In-Reply-To: <C50968E1-525D-4128-B65E-B8E2C0547E00@pobox.com>
References: <C50968E1-525D-4128-B65E-B8E2C0547E00@pobox.com>
Message-ID: <834445F7-EB89-4B84-93FB-21F94599B984@jerris.com>

Were these calls that might be considered spammy calls or be reported as such?  Depending on what they think the calls are, they may not be willing to.

> On Feb 1, 2018, at 2:18 PM, M Yudkowsky <speech at pobox.com> wrote:
> 
>> 
>> On Feb 1, 2018, at 13:07 , Michael Jerris <mike at jerris.com> wrote:
>> 
>> The workaround would be to send a toll number.  The fix would be to ask Skype or whatever carrier is rejecting the call to fix it.
> 
> Well, sending a toll number would be nice, and was my plan B or C, except that everyone receiving the call would be puzzled as to who's calling them -- bad for business.
> 
> Actually, I was just about to write: turns out that this is *not* a Skype-in number problem. The "obvious" thing to do, and I was too focused on tech-side to do it, was to try other non-toll numbers in the DID; e.g. 1 800 111 2222. That works, btw, since PAI's are never validated in my experience to date. As such I've determined that indeed, as Mr J states, this problem is specific to my TN.
> 
> Now I just have to find the right people at Skype to unblock my TN. I wonder how long it takes to get that done.



From tculjaga at gmail.com  Thu Feb  1 20:14:51 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 1 Feb 2018 21:14:51 +0100
Subject: [Freeswitch-users] FreeSWITCH behind NAT
In-Reply-To: <BY2PR07MB613B067273E8F42C1F75344BEE40@BY2PR07MB613.namprd07.prod.outlook.com>
References: <e676ae53-fe64-6753-4d76-61670b678540@gmail.com>
 <c5e7635e-7827-bac7-b9a6-a9668d2db24f@gmail.com>
 <BY2PR07MB613B067273E8F42C1F75344BEE40@BY2PR07MB613.namprd07.prod.outlook.com>
Message-ID: <CAJoKKdX8N1tJ7YzxWpe8SZjx7GB6z4rJDdMq99sH4sJkP0iDKg@mail.gmail.com>

well, here is something you may find interesting:
https://www.youtube.com/watch?v=_WSx-T6TriI

have fun :=)

On 30 January 2018 at 19:03, Mundkowsky, Robert <rmundkowsky at ets.org> wrote:

> I am not an expert, you should get help from FS staff, but anyways, here
> is some info:
>
> > 1. Do ports need to be opened on the firewall under all circumstances?
> > UDP ports for RTP, for instance.
>
> Yes. If you block the ports then nothing can get thru.  For SIP/RTP, you
> need the SIP port open, and you need a range of ports open for RTP. See the
> configuration files for the port numbers.
>
> If you use WSS, then you need the WSS port open.
>
>
> > 2. Is this always a good idea to enable in sip_profiles/internal.xml?
> > <param name="nat-options-ping" value="true"/>
>
> Maybe. Some software does not support OPTIONS messages. If it yours does
> then yeah use it.  See https://freeswitch.org/
> confluence/display/FREESWITCH/NAT+Traversal
>
> > 3. Is it necessary/recommended to have STUN enabled in vars.xml AND
> > setup the nat-options-ping?
>
> I guess here, if you or your client are behind an asymmetric NAT then you
> need a STUN server. If a symmetric NAT then you need TURN server.  Keep in
> mind your clients might have all kinds of different situations.
>
>
> > 4. my sip_profile/internal.xml has this:
> > <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip"
> > value="auto-nat"/>
> >
> > Is this an improvement over what's  in confluence of:
> > <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>
> Not sure, read up on it https://freeswitch.org/
> confluence/display/FREESWITCH/Auto+Nat
>
>
> > 5. If the endpoints are configured to connect using TCP, does any of
> > this change what's above?
>
> Not sure, but my guess is no
>
> Robert
>
> -----Original Message-----
> From: FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] On Behalf Of jungle boogie
> Sent: Tuesday, January 30, 2018 12:15 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] FreeSWITCH behind NAT
>
> Hi All,
>
> Can anyone give me some advice? I'll update the docs, if it's needed.
>
> Thanks!
>
> Thus said Jungle Boogie on Sun, 28 Jan 2018 17:51:40 -0800
> > Hi All,
> >
> > I have some questions about this page and what folks do when
> > freeswitch is behind NAT:
> > https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ffrees
> > witch.org%2Fconfluence%2Fdisplay%2FFREESWITCH%2FNAT%2BTraversal&data=0
> > 2%7C01%7Crmundkowsky%40ets.org%7C505306e6c453490da26708d567a0f5d5%7C0b
> > a6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636528863387947052&sdata=L91K
> > SrefIPmwVZLdFSPioL6zcM7Be5MlgxPoyUOU70w%3D&reserved=0
> >
> > 1. Do ports need to be opened on the firewall under all circumstances?
> > UDP ports for RTP, for instance.
> >
> > 2. Is this always a good idea to enable in sip_profiles/internal.xml?
> > <param name="nat-options-ping" value="true"/>
> >
> >
> > 3. Is it necessary/recommended to have STUN enabled in vars.xml AND
> > setup the nat-options-ping?
> >
> >
> > 4. my sip_profile/internal.xml has this:
> > <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip"
> > value="auto-nat"/>
> >
> > Is this an improvement over what's  in confluence of:
> > <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> >
> > 5. If the endpoints are configured to connect using TCP, does any of
> > this change what's above?
> >
> > thanks!
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.
> freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%
> 7C505306e6c453490da26708d567a0f5d5%7C0ba6e9b760b34fae92f37e6ddd9e
> 9b65%7C0%7C0%7C636528863387947052&sdata=qvvmQaISaJ37%
> 2FHkzp8eNafIKcIfvWtYI9WYlMUb3HTs%3D&reserved=0
>
> Official FreeSWITCH Sites
> https://na01.safelinks.protection.outlook.com/?url=
> http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%
> 7C505306e6c453490da26708d567a0f5d5%7C0ba6e9b760b34fae92f37e6ddd9e
> 9b65%7C0%7C0%7C636528863387947052&sdata=2YGYIAS02v0lG%
> 2ByUtZZdCkzFJCgpYU4eUeGuWfcfnfY%3D&reserved=0
> https://na01.safelinks.protection.outlook.com/?url=
> http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%
> 7Crmundkowsky%40ets.org%7C505306e6c453490da26708d567a0f5d5%
> 7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636528863387947052&sdata=
> sWhGrHsJoqBxx9p%2BQ32vxrbyrclq0QCb4llrrfs3QRo%3D&reserved=0
> https://na01.safelinks.protection.outlook.com/?url=
> http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%
> 7C505306e6c453490da26708d567a0f5d5%7C0ba6e9b760b34fae92f37e6ddd9e
> 9b65%7C0%7C0%7C636528863387947052&sdata=wQP9XZziFFiMIkvWB5zr5DwDK5F5%
> 2BDbNJ3gtJCHhYt8%3D&reserved=0
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> https://na01.safelinks.protection.outlook.com/?url=
> http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%
> 2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%
> 7C505306e6c453490da26708d567a0f5d5%7C0ba6e9b760b34fae92f37e6ddd9e
> 9b65%7C0%7C0%7C636528863387947052&sdata=iKUQA2bDbZh4jgK4eN%
> 2F4sJCiHrEqRXTPyPHedufrJws%3D&reserved=0
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://na01.safelinks.protection.outlook.com/?url=
> http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%
> 7C505306e6c453490da26708d567a0f5d5%7C0ba6e9b760b34fae92f37e6ddd9e
> 9b65%7C0%7C0%7C636528863387947052&sdata=2YGYIAS02v0lG%
> 2ByUtZZdCkzFJCgpYU4eUeGuWfcfnfY%3D&reserved=0
>
> ________________________________
>
> This e-mail and any files transmitted with it may contain privileged or
> confidential information. It is solely for use by the individual for whom
> it is intended, even if addressed incorrectly. If you received this e-mail
> in error, please notify the sender; do not disclose, copy, distribute, or
> take any action in reliance on the contents of this information; and delete
> it from your system. Any other use of this e-mail is prohibited.
>
>
> Thank you for your compliance.
>
> ________________________________
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 speech at pobox.com  Thu Feb  1 20:18:54 2018
From: speech at pobox.com (M Yudkowsky)
Date: Thu, 1 Feb 2018 14:18:54 -0600
Subject: [Freeswitch-users] Skype-in no longer accepts Toll-Free PAI or
 even From
In-Reply-To: <834445F7-EB89-4B84-93FB-21F94599B984@jerris.com>
References: <C50968E1-525D-4128-B65E-B8E2C0547E00@pobox.com>
 <834445F7-EB89-4B84-93FB-21F94599B984@jerris.com>
Message-ID: <88E742D0-308C-4857-9C7F-91FAB4067D33@pobox.com>



> On Feb 1, 2018, at 13:56 , Michael Jerris <mike at jerris.com> wrote:
> 
> Were these calls that might be considered spammy calls or be reported as such?  Depending on what they think the calls are, they may not be willing to.

I'm amazed to report that Skype wrote, in response to my original PAI question, "This is an issue were working to fix. I have asked to get your user ID unblocked, absolute apologies for the disruption." 

My personal Skype-in TN is receives the PAI and displays it correctly. I'm trying to verify that it's fixed for everyone, not just me.



-- 
  Moshe Yudkowsky
  Disaggregate Corporation
  2952 W Fargo
  Chicago, IL 60645 USA


   http://www.Disaggregate.com
   http://www.PebbleAndAvalanche.com








From jungleboogie0 at gmail.com  Thu Feb  1 20:23:30 2018
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Thu, 1 Feb 2018 12:23:30 -0800
Subject: [Freeswitch-users] FreeSWITCH behind NAT
In-Reply-To: <CAJoKKdX8N1tJ7YzxWpe8SZjx7GB6z4rJDdMq99sH4sJkP0iDKg@mail.gmail.com>
References: <e676ae53-fe64-6753-4d76-61670b678540@gmail.com>
 <c5e7635e-7827-bac7-b9a6-a9668d2db24f@gmail.com>
 <BY2PR07MB613B067273E8F42C1F75344BEE40@BY2PR07MB613.namprd07.prod.outlook.com>
 <CAJoKKdX8N1tJ7YzxWpe8SZjx7GB6z4rJDdMq99sH4sJkP0iDKg@mail.gmail.com>
Message-ID: <CAKE2PDvrTNdpUmZoV_jqm212TEEGwSpZe3037=nDtMNy4c8GtA@mail.gmail.com>

Thanks everyone for the input! It seems the documentation is fairly
accurate at this point.


From asilva at wirelessmundi.com  Thu Feb  1 23:01:45 2018
From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=)
Date: Fri, 2 Feb 2018 00:01:45 +0100
Subject: [Freeswitch-users] mod_lua Too many open files
In-Reply-To: <d0ecb16e-9989-1eb0-ad38-19654a12aa5f@wirelessmundi.com>
References: <d0ecb16e-9989-1eb0-ad38-19654a12aa5f@wirelessmundi.com>
Message-ID: <3024211c-1652-924e-54d4-be2b4e3886d6@wirelessmundi.com>

this issue was fix between 05/2017 and now, putting a new version of FS 
git master solves my issue.

sorry for the noise.


On 01/22/2018 05:40 PM, António Silva wrote:
> hi,
>
> After a few weeks running fs without issues, i got the error "Too many 
> open files", i understand that i reach the limit of allowed open files 
> in the system. But it must be trigger for some "zombie fds" opened..
>
>
> Is there a way to get the "zombie" fds open by fs?
>
>
> I use lua scripts to check db mapping, so in every script i make sure 
> to call dbh:release() before returning.
> I also use esl sockets to check FS events.
>
> in the running process i have the following limits:
>
> cat /proc/10975/limits
>
> Limit                     Soft Limit           Hard Limit Units
> Max cpu time              unlimited            unlimited seconds
> Max file size             unlimited            unlimited bytes
> Max data size             unlimited            unlimited bytes
> Max stack size            245760               8388608 bytes
> Max core file size        unlimited            unlimited bytes
> Max resident set          unlimited            unlimited bytes
> Max processes             unlimited            unlimited processes
> Max open files            999999               999999 files
> Max locked memory         65536                65536 bytes
> Max address space         unlimited            unlimited bytes
> Max file locks            unlimited            unlimited locks
> Max pending signals       128123               128123 signals
> Max msgqueue size         819200               819200 bytes
> Max nice priority         0                    0
> Max realtime priority     unlimited            unlimited
> Max realtime timeout      unlimited            unlimited us
>
>
> currently i see the number of files change:
>
> while :; do echo -n "$(date +'%F %T') "; lsof -p $(cat 
> /var/run/freeswitch/freeswitch.pid) | wc -l; sleep 5; done;
> 2018-01-22 17:20:46 78938
> 2018-01-22 17:20:53 78920
> 2018-01-22 17:20:59 78902
> 2018-01-22 17:21:06 78911
> 2018-01-22 17:21:12 78883
> 2018-01-22 17:21:18 78883
> 2018-01-22 17:21:25 78899
> 2018-01-22 17:21:31 78913
> 2018-01-22 17:21:37 78925
> 2018-01-22 17:21:44 78911
> 2018-01-22 17:21:50 78913
> 2018-01-22 17:21:56 78935
>
>
> I see lot of entries:
> COMMAND    PID USER   FD   TYPE             DEVICE  SIZE/OFF NODE NAME
> freeswitc 2429 root *257u  0000               0,11         0 10734 
> anon_inode
> freeswitc 2429 root *258u  0000               0,11         0 10734 
> anon_inode
> freeswitc 2429 root *260u  0000               0,11         0 10734 
> anon_inode
> freeswitc 2429 root *261u  0000               0,11         0 10734 
> anon_inode
> freeswitc 2429 root *262u  0000               0,11         0 10734 
> anon_inode
> freeswitc 2429 root *264u  0000               0,11         0 10734 
> anon_inode
> freeswitc 2429 root *265u  0000               0,11         0 10734 
> anon_inode
> freeswitc 2429 root *266u  0000               0,11         0 10734 
> anon_inode
>
>
> grep "0000" openfiles | wc -l
> 78951
>
> but I don't understand the output for the TYPE= "0000"...
>
>
> thanks for the help.
>

-- 
Saludos / Regards / Cumprimentos
António Silva



From jaradmorgan at gmail.com  Fri Feb  2 13:55:29 2018
From: jaradmorgan at gmail.com (Jarad Morgan)
Date: Fri, 02 Feb 2018 13:55:29 +0000
Subject: [Freeswitch-users] auto create presence_data_cols natively?
Message-ID: <CANYMxhhMTqA0WJnK8u-tQdZ+1YSfuF6r_Uck6aauRjKQ9_PLdw@mail.gmail.com>

Hello! When FreeSWITCH starts up and creates the schema and/or database if
not present.. is there any way natively to have it create the fields
defined in presence_data_cols as well? Or is the only option to manually
alter schema outside of FreeSWITCH?

Thanks!!
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From mike at jerris.com  Fri Feb  2 16:38:46 2018
From: mike at jerris.com (Michael Jerris)
Date: Fri, 2 Feb 2018 11:38:46 -0500
Subject: [Freeswitch-users] auto create presence_data_cols natively?
In-Reply-To: <CANYMxhhMTqA0WJnK8u-tQdZ+1YSfuF6r_Uck6aauRjKQ9_PLdw@mail.gmail.com>
References: <CANYMxhhMTqA0WJnK8u-tQdZ+1YSfuF6r_Uck6aauRjKQ9_PLdw@mail.gmail.com>
Message-ID: <29A9CB11-B4EF-41C1-922B-91BAA7D8CFD1@jerris.com>

when using custom schema you must maintain that manually.


> On Feb 2, 2018, at 8:55 AM, Jarad Morgan <jaradmorgan at gmail.com> wrote:
> 
> Hello! When FreeSWITCH starts up and creates the schema and/or database if not present.. is there any way natively to have it create the fields defined in presence_data_cols as well? Or is the only option to manually alter schema outside of FreeSWITCH?



From william at williamcollsassoc.ca  Sat Feb  3 22:40:09 2018
From: william at williamcollsassoc.ca (William Colls)
Date: Sat, 3 Feb 2018 17:40:09 -0500
Subject: [Freeswitch-users] Debian 9 and Freeswitch 1.8
Message-ID: <5ed9762c-0735-0135-682b-d1458daf891e@williamcollsassoc.ca>


Is there an expected date when Freeswitch 1.8/Debian 9 will become the 
preferred production configuration?



From ssinyagin at gmail.com  Sun Feb  4 02:36:47 2018
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sun, 4 Feb 2018 03:36:47 +0100
Subject: [Freeswitch-users] Debian 9 and Freeswitch 1.8
In-Reply-To: <5ed9762c-0735-0135-682b-d1458daf891e@williamcollsassoc.ca>
References: <5ed9762c-0735-0135-682b-d1458daf891e@williamcollsassoc.ca>
Message-ID: <CACOYK=hHj15WuNu7AYhqdT8NGS5i+fAx=MUK-6Rw6tbDqhW_Ww@mail.gmail.com>

BTW, I run a Debian 8 in a privileged LXC container on a Debian 9
host, and FreeSWITCH is running fine. The container has its own
physical NIC, and is used as a voice quality testing probe. Because
the container has full privileges (the default in Debian), FreeSWITCH
daemon gets the necessary realtime priority in the host kernel. I was
also monitoring the quality of outbound RTP stream, and the best
results are achieved with "timerfd" RTP timer in the SIP profile. The
default "soft" timer resulted in sporadic RTP delays in some calls.



On Sat, Feb 3, 2018 at 11:40 PM, William Colls
<william at williamcollsassoc.ca> wrote:
>
> Is there an expected date when Freeswitch 1.8/Debian 9 will become the
> preferred production configuration?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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


From mike at ascendency.net  Sun Feb  4 16:47:18 2018
From: mike at ascendency.net (Mike Loiterman)
Date: Sun, 4 Feb 2018 10:47:18 -0600
Subject: [Freeswitch-users] mod_soundtouch
Message-ID: <C188AE2A-ED39-47B5-85DD-68DE82783834@ascendency.net>

I’m trying to use mod_sound touch, but the audio doesn’t appear to be processing correctly.  The module is loaded, but something isn’t right because all I get on the B Leg is an almost imperceptible bit of static when the A Leg is speaking.

Dialplan:
    <extension name="Soundtouch">
      <condition field="destination_number" expression="^(\d+)$">
        <action application="soundtouch" data="hook_dtmf -0.3o 0.8r"/>
        <action application="bridge" data="sofia/gateway/gateway1/$1"/>
      </condition>
    </extension>

Versions:
Linux 3.16.0-5-amd64 #1 SMP Debian 3.16.51-3+deb8u1 (2018-01-08) x86_64 GNU/Linux
FreeSWITCH Version 1.6.20-37-987c9b9~64bit (-37-987c9b9 64bit)
freeswitch-mod-soundtouch 1.6.20~37~987c9b9-1~jessie+1 amd64

I’ve emailed the developer at max at evolux.net.br, but the email was just bounced back saying the address doesn’t exist.  

Seems like this module is dead?  Appreciate any input.


------------------------------
Mike Loiterman
Cell: 630-302-4944
Email: mike at ascendency.net



From michael at mailworks.org  Mon Feb  5 06:25:59 2018
From: michael at mailworks.org (Michael Avers)
Date: Sun, 04 Feb 2018 23:25:59 -0700
Subject: [Freeswitch-users] freeswitch-mod-ilbc
In-Reply-To: <C188AE2A-ED39-47B5-85DD-68DE82783834@ascendency.net>
References: <C188AE2A-ED39-47B5-85DD-68DE82783834@ascendency.net>
Message-ID: <1517811959.181284.1259552336.08D63091@webmail.messagingengine.com>

Hello,

Is there a reason why freeswitch-mod-ilbc is no longer included in the Debian packaging? It looks like after 1.6.13 it no longer got packaged. I just did an apt-update and all my packages updated to 1.6.20 and in the process it removed the older mod-ilbc package.

Any ideas?

Mike


From michaelt at mdevt.com  Mon Feb  5 09:43:53 2018
From: michaelt at mdevt.com (Michael Toop)
Date: Mon, 05 Feb 2018 09:43:53 +0000
Subject: [Freeswitch-users] freeswitch.org Site Certificate Has Expired
Message-ID: <CAF34jFESMxqSQqGUFfrrEvWwc-C69dQL75eyif4hxob3dA8e1A@mail.gmail.com>

Hi, FYI the wildcard domain: "*.freeswitch.org" certificate has expired.

Not sure who maintains it, but it isn't me :).
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From brian at freeswitch.com  Mon Feb  5 14:09:27 2018
From: brian at freeswitch.com (Brian West)
Date: Mon, 5 Feb 2018 08:09:27 -0600
Subject: [Freeswitch-users] freeswitch.org Site Certificate Has Expired
In-Reply-To: <CAF34jFESMxqSQqGUFfrrEvWwc-C69dQL75eyif4hxob3dA8e1A@mail.gmail.com>
References: <CAF34jFESMxqSQqGUFfrrEvWwc-C69dQL75eyif4hxob3dA8e1A@mail.gmail.com>
Message-ID: <CAEJMVkD8wUM4JDKzHWt7aB3e6+zpce-Jo-WpUuYTgChmhM5WpQ@mail.gmail.com>

Yes, I replaced it last Friday, I missed a couple of systems, So where were
you seeing this expired at?

On Mon, Feb 5, 2018 at 3:43 AM, Michael Toop <michaelt at mdevt.com> wrote:

> Hi, FYI the wildcard domain: "*.freeswitch.org" certificate has expired.
>
> Not sure who maintains it, but it isn't me :).
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From mike at jerris.com  Mon Feb  5 16:13:00 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 5 Feb 2018 11:13:00 -0500
Subject: [Freeswitch-users] Debian 9 and Freeswitch 1.8
In-Reply-To: <5ed9762c-0735-0135-682b-d1458daf891e@williamcollsassoc.ca>
References: <5ed9762c-0735-0135-682b-d1458daf891e@williamcollsassoc.ca>
Message-ID: <4B263F26-BC14-43AD-8869-DA0F8EE6E022@jerris.com>

There are some known issues with Debian 9 and how its handled the openssl upgrade that we are still working through to confirm what exactly needs to get done to fix it.  The problem is, a bunch of system libs were not ready for the new openssl when deb9 was locked down, so they took a half way there approach and its caused issues where multiple versions of openssl end up in the same process, and we expect that this may cause some issues.  I expect this to take a little while to verify completely.


> On Feb 3, 2018, at 5:40 PM, William Colls <william at williamcollsassoc.ca> wrote:
> 
> 
> Is there an expected date when Freeswitch 1.8/Debian 9 will become the preferred production configuration?



From scott at tgifriday.com  Sun Feb  4 19:37:08 2018
From: scott at tgifriday.com (Scott Howell)
Date: Sun, 04 Feb 2018 19:37:08 +0000
Subject: [Freeswitch-users] Stream inbound Freeswitch (RTMP?)
Message-ID: <CAMALkh2pQ9u3ZWzWB+PPCMQh-p0mgMcRzXG4VJgiRBYm58ba3A@mail.gmail.com>

How would I stream audio into Freeswitch for callers to dial in and
listen.  I suspect RTMP, but I'm uncertain of the configuration,
particularly in My FusionBox Setup, although I can configure the underlying
freeswitch if necessary.

I have an audio/video stream that I already set up, I would like to use the
audio portion as a "conference" that users can dial into and listen live.

Thank you,
-Scott
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From mike at jerris.com  Mon Feb  5 16:24:19 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 5 Feb 2018 11:24:19 -0500
Subject: [Freeswitch-users] freeswitch-mod-ilbc
In-Reply-To: <1517811959.181284.1259552336.08D63091@webmail.messagingengine.com>
References: <C188AE2A-ED39-47B5-85DD-68DE82783834@ascendency.net>
 <1517811959.181284.1259552336.08D63091@webmail.messagingengine.com>
Message-ID: <D3232709-87EA-4887-AF67-F817327BF8F2@jerris.com>

The commit that changed this was:

commit 0a50536aa837cd18a5744962e8ca49a22f71bc63
Author: Travis Cross <tc at traviscross.com>
Date:   Tue May 28 21:58:36 2013 +0000

    Add generic mechanism for building non-DFSG packages

> On Feb 5, 2018, at 1:25 AM, Michael Avers <michael at mailworks.org> wrote:
> 
> Hello,
> 
> Is there a reason why freeswitch-mod-ilbc is no longer included in the Debian packaging? It looks like after 1.6.13 it no longer got packaged. I just did an apt-update and all my packages updated to 1.6.20 and in the process it removed the older mod-ilbc package.
> 
> Any ideas?
> 
> Mike
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 kathleen at freeswitch.com  Mon Feb  5 17:46:48 2018
From: kathleen at freeswitch.com (Kathleen King)
Date: Mon, 5 Feb 2018 09:46:48 -0800
Subject: [Freeswitch-users] ClueCon Weekly: Women in VoIP
Message-ID: <CAOStUPYQ9B4DHuzS3aoo-otnyCdJO9pU4o0=EQ2RrQaTRe+spw@mail.gmail.com>

Hi FreeSWITCHers,

This week we are having another wonderful edition of Women in VoIP on
ClueCon Weekly! The conference is at 12:00pm central time every Wednesday.
You can join the call live by dialing 888 at
https://conference.freeswitch.org/vc/ or watch it live on our Youtube
channel: https://youtu.be/YCIoey-bW6k

This week we have gathered together women from various companies in the
communication industry to talk technology!



Kathleen King | Public Relations / Administrative Assistant

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

Email: Kathleen at freeswitch.com

Mobile: 703-859-3757

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

[image: color-facebook-96.png] <https://www.facebook.com/freeswitch/> [image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From david.villasmil.work at gmail.com  Mon Feb  5 19:30:25 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 5 Feb 2018 20:30:25 +0100
Subject: [Freeswitch-users] freeswitch odbc in the core and RDS
Message-ID: <CAFGRPVrsHCoUSGfqietuJjm3G_PZRuEUSt_pX1PrRcwnNuRVMg@mail.gmail.com>

Hello guys,

trying to use RDS as a backend for freeswitch core, but when creating the
tables, RDS comes back with:

[STATE: HY000 CODE 1118 ERROR: [MySQL][ODBC 5.1
Driver][mysqld-5.6.37-log]Row size too large. The maximum row size for the
used table type, not counting BLOBs, is 65535. This includes storage
overhead, check the manual. You have to change some columns to TEXT or BLOBs

for some tables, thus messing up everything

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
ᐧ
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From michael at mailworks.org  Mon Feb  5 19:43:54 2018
From: michael at mailworks.org (Michael Avers)
Date: Mon, 05 Feb 2018 12:43:54 -0700
Subject: [Freeswitch-users] freeswitch-mod-ilbc
In-Reply-To: <D3232709-87EA-4887-AF67-F817327BF8F2@jerris.com>
References: <C188AE2A-ED39-47B5-85DD-68DE82783834@ascendency.net>
 <1517811959.181284.1259552336.08D63091@webmail.messagingengine.com>
 <D3232709-87EA-4887-AF67-F817327BF8F2@jerris.com>
Message-ID: <1517859834.2137258.1260378608.30C929B1@webmail.messagingengine.com>

What would be the right way to bring back ILBC codec support to a
freeswitch 1.6.20 installed from the official Debian packages? Is there
even a way?
Mike


On Mon, Feb 5, 2018, at 9:24 AM, Michael Jerris wrote:
> The commit that changed this was:
> 
> commit 0a50536aa837cd18a5744962e8ca49a22f71bc63
> Author: Travis Cross <tc at traviscross.com>
> Date:   Tue May 28 21:58:36 2013 +0000
> 
>     Add generic mechanism for building non-DFSG packages
> 
>> On Feb 5, 2018, at 1:25 AM, Michael Avers
>> <michael at mailworks.org> wrote:>> 
>> Hello,
>> 
>> Is there a reason why freeswitch-mod-ilbc is no longer included in
>> the Debian packaging? It looks like after 1.6.13 it no longer got
>> packaged. I just did an apt-update and all my packages updated to
>> 1.6.20 and in the process it removed the older mod-ilbc package.>> 
>> Any ideas?
>> 
>> Mike
>> 
>> __________________________________________________________________-
>> _______>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
> ___________________________________________________________________-
> ________> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 mike at jerris.com  Mon Feb  5 22:14:09 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 5 Feb 2018 17:14:09 -0500
Subject: [Freeswitch-users] freeswitch-mod-ilbc
In-Reply-To: <1517859834.2137258.1260378608.30C929B1@webmail.messagingengine.com>
References: <C188AE2A-ED39-47B5-85DD-68DE82783834@ascendency.net>
 <1517811959.181284.1259552336.08D63091@webmail.messagingengine.com>
 <D3232709-87EA-4887-AF67-F817327BF8F2@jerris.com>
 <1517859834.2137258.1260378608.30C929B1@webmail.messagingengine.com>
Message-ID: <820679C8-E189-4C46-9EF6-07FDF28E5E1F@jerris.com>

I honestly didn’t realize we were not building it by default.  File a jira on the issue and we can try to take a look at it.  Its not a codec that is used a lot so probably just no one has noticed.


> On Feb 5, 2018, at 2:43 PM, Michael Avers <michael at mailworks.org> wrote:
> 
> What would be the right way to bring back ILBC codec support to a freeswitch 1.6.20 installed from the official Debian packages? Is there even a way?
> 
> Mike
> 
> 
> On Mon, Feb 5, 2018, at 9:24 AM, Michael Jerris wrote:
>> The commit that changed this was:
>> 
>> commit 0a50536aa837cd18a5744962e8ca49a22f71bc63
>> Author: Travis Cross <tc at traviscross.com <mailto:tc at traviscross.com>>
>> Date:   Tue May 28 21:58:36 2013 +0000
>> 
>>     Add generic mechanism for building non-DFSG packages
>> 
>>> On Feb 5, 2018, at 1:25 AM, Michael Avers <michael at mailworks.org <mailto:michael at mailworks.org>> wrote:
>>> 
>>> Hello,
>>> 
>>> Is there a reason why freeswitch-mod-ilbc is no longer included in the Debian packaging? It looks like after 1.6.13 it no longer got packaged. I just did an apt-update and all my packages updated to 1.6.20 and in the process it removed the older mod-ilbc package.
>>> 
>>> Any ideas?
>>> 
>>> Mike
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>  
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.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>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.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>
> http://www.freeswitch.org <http://www.freeswitch.org/>
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From michael at mailworks.org  Mon Feb  5 22:27:02 2018
From: michael at mailworks.org (Michael Avers)
Date: Mon, 05 Feb 2018 15:27:02 -0700
Subject: [Freeswitch-users] freeswitch-mod-ilbc
In-Reply-To: <820679C8-E189-4C46-9EF6-07FDF28E5E1F@jerris.com>
References: <C188AE2A-ED39-47B5-85DD-68DE82783834@ascendency.net>
 <1517811959.181284.1259552336.08D63091@webmail.messagingengine.com>
 <D3232709-87EA-4887-AF67-F817327BF8F2@jerris.com>
 <1517859834.2137258.1260378608.30C929B1@webmail.messagingengine.com>
 <820679C8-E189-4C46-9EF6-07FDF28E5E1F@jerris.com>
Message-ID: <1517869622.3011619.1260571864.2B862226@webmail.messagingengine.com>

There is this one:

https://freeswitch.org/jira/browse/FS-5415

Is it possible to reopen it or is it required to open a new one? Its the
same issue.
Thanks
Mike


On Mon, Feb 5, 2018, at 3:14 PM, Michael Jerris wrote:
> I honestly didn’t realize we were not building it by default.  File a
> jira on the issue and we can try to take a look at it.  Its not a
> codec that is used a lot so probably just no one has noticed.> 
> 
>> On Feb 5, 2018, at 2:43 PM, Michael Avers
>> <michael at mailworks.org> wrote:>> 
>> What would be the right way to bring back ILBC codec support to a
>> freeswitch 1.6.20 installed from the official Debian packages? Is
>> there even a way?>> 
>> Mike
>> 
>> 
>> On Mon, Feb 5, 2018, at 9:24 AM, Michael Jerris wrote:
>>> The commit that changed this was:
>>> 
>>> commit 0a50536aa837cd18a5744962e8ca49a22f71bc63
>>> Author: Travis Cross <tc at traviscross.com>
>>> Date:   Tue May 28 21:58:36 2013 +0000
>>> 
>>>     Add generic mechanism for building non-DFSG packages
>>> 
>>>> On Feb 5, 2018, at 1:25 AM, Michael Avers <michael at mailworks.org>
>>>> wrote:>>>> 
>>>> Hello,
>>>> 
>>>> Is there a reason why freeswitch-mod-ilbc is no longer included in
>>>> the Debian packaging? It looks like after 1.6.13 it no longer got
>>>> packaged. I just did an apt-update and all my packages updated to
>>>> 1.6.20 and in the process it removed the older mod-ilbc package.>>>> 
>>>> Any ideas?
>>>> 
>>>> Mike
>>>> 
>>>> __________________________________________________________________-
>>>> _______>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.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
>>> ___________________________________________________________________-
>>> ________>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com[1]
>>>  
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org[2]
>>> http://confluence.freeswitch.org[3]
>>> http://www.cluecon.com[4]
>>>  
>>> FreeSWITCH-users mailing 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[5]
>> 
>> __________________________________________________________________-
>> _______>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com[6]
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org[7]
>> http://confluence.freeswitch.org[8]
>> http://www.cluecon.com[9]
>> 
>> FreeSWITCH-users mailing 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[10]
> ___________________________________________________________________-
> ________> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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


Links:

   1. http://www.freeswitchsolutions.com/
   2. http://www.freeswitch.org/
   3. http://confluence.freeswitch.org/
   4. http://www.cluecon.com/
   5. http://www.freeswitch.org/
   6. http://www.freeswitchsolutions.com/
   7. http://www.freeswitch.org/
   8. http://confluence.freeswitch.org/
   9. http://www.cluecon.com/
  10. http://www.freeswitch.org/
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From brian at freeswitch.com  Mon Feb  5 22:36:01 2018
From: brian at freeswitch.com (Brian West)
Date: Mon, 5 Feb 2018 16:36:01 -0600
Subject: [Freeswitch-users] freeswitch-mod-ilbc
In-Reply-To: <1517869622.3011619.1260571864.2B862226@webmail.messagingengine.com>
References: <C188AE2A-ED39-47B5-85DD-68DE82783834@ascendency.net>
 <1517811959.181284.1259552336.08D63091@webmail.messagingengine.com>
 <D3232709-87EA-4887-AF67-F817327BF8F2@jerris.com>
 <1517859834.2137258.1260378608.30C929B1@webmail.messagingengine.com>
 <820679C8-E189-4C46-9EF6-07FDF28E5E1F@jerris.com>
 <1517869622.3011619.1260571864.2B862226@webmail.messagingengine.com>
Message-ID: <CAEJMVkCQw6+qG6o+WyHUu5H1kxQ0C8WZA4i1jk=rh5jcputcog@mail.gmail.com>

https://freeswitch.org/jira/browse/FS-10386

On Mon, Feb 5, 2018 at 4:27 PM, Michael Avers <michael at mailworks.org> wrote:

> There is this one:
>
> https://freeswitch.org/jira/browse/FS-5415
>
> Is it possible to reopen it or is it required to open a new one? Its the
> same issue.
>
> Thanks
> Mike
>
>
> On Mon, Feb 5, 2018, at 3:14 PM, Michael Jerris wrote:
>
> I honestly didn’t realize we were not building it by default.  File a jira
> on the issue and we can try to take a look at it.  Its not a codec that is
> used a lot so probably just no one has noticed.
>
>
> On Feb 5, 2018, at 2:43 PM, Michael Avers <michael at mailworks.org> wrote:
>
> What would be the right way to bring back ILBC codec support to a
> freeswitch 1.6.20 installed from the official Debian packages? Is there
> even a way?
>
> Mike
>
>
> On Mon, Feb 5, 2018, at 9:24 AM, Michael Jerris wrote:
>
> The commit that changed this was:
>
> commit 0a50536aa837cd18a5744962e8ca49a22f71bc63
> Author: Travis Cross <tc at traviscross.com>
> Date:   Tue May 28 21:58:36 2013 +0000
>
>     Add generic mechanism for building non-DFSG packages
>
> On Feb 5, 2018, at 1:25 AM, Michael Avers <michael at mailworks.org> wrote:
>
> Hello,
>
> Is there a reason why freeswitch-mod-ilbc is no longer included in the
> Debian packaging? It looks like after 1.6.13 it no longer got packaged. I
> just did an apt-update and all my packages updated to 1.6.20 and in the
> process it removed the older mod-ilbc package.
>
> Any ideas?
>
> Mike
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
> *_________________________________________________________________________*
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
> *_________________________________________________________________________*
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From colin.morelli at gmail.com  Tue Feb  6 00:56:56 2018
From: colin.morelli at gmail.com (Colin Morelli)
Date: Mon, 5 Feb 2018 19:56:56 -0500
Subject: [Freeswitch-users] rtp-timer-name / timer issues
Message-ID: <CAPtU-UqfL-kQNYzPzts84UppsEjhvVbv=wJOpRnxZTCm-d184Q@mail.gmail.com>

Hey list,

I'm running FS on EC2 (I know, I know). Having some issues with random
packet loss, which I believe almost certainly I have narrowed down to timer
issues and/or network latency/jitter (seems surprising since I'm using
c5.xlarge instances).

Behavior is that, during a call, brief pauses or notable audio loss will
occur. This is on high bandwidth links that are otherwise stable.
Freeswitch logs with max debug spew out "Hot Hit 1" through "Hot Hit 10"
and eventually "auto-flush catching up 1 packet(s)" in rapid succession
(usually going through the cycle 4-5 times) before things settle again.
Obviously that means a minimum of 4-5 audio packets were dropped within the
span of a second which results in considerable audio artifacting.

Changing rtp-timer-name to none, which I understand to perform synchronous
reads of RTP audio (as opposed to timer-based async reads) makes the audio
notably smoother. That said, I'm having a hard time uncovering the
consequences of doing this. Obviously I understand that reads will block
the RTP thread, but I can't seem to understand the potential ramifications
of this. Could anyone help clarify?

My other question is: assuming "timer while hot" indicates what I believe
it does (that when the timer hit there was >1 packet in the queue to be
read), couldn't this issue also just be caused by network jitter, and not
necessarily just timer inconsistencies?

Thanks in advance.

Best,
Colin
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From gmina at connectfirst.com  Tue Feb  6 01:22:36 2018
From: gmina at connectfirst.com (Geoff Mina)
Date: Mon, 5 Feb 2018 18:22:36 -0700
Subject: [Freeswitch-users] rtp-timer-name / timer issues
In-Reply-To: <CAPtU-UqfL-kQNYzPzts84UppsEjhvVbv=wJOpRnxZTCm-d184Q@mail.gmail.com>
References: <CAPtU-UqfL-kQNYzPzts84UppsEjhvVbv=wJOpRnxZTCm-d184Q@mail.gmail.com>
Message-ID: <AED238AC-BEC6-4A1F-A28E-B7E2709ACFDA@connectfirst.com>

What OS are you using? We run FS at over 600 concurrent calls on a smaller EC2 (c4.xlarge) size and have not had any issues in either US-East or US-West. 

Running FS Installed from YUM. CentOS7. 

> On Feb 5, 2018, at 5:56 PM, Colin Morelli <colin.morelli at gmail.com> wrote:
> 
> Hey list,
> 
> I'm running FS on EC2 (I know, I know). Having some issues with random packet loss, which I believe almost certainly I have narrowed down to timer issues and/or network latency/jitter (seems surprising since I'm using c5.xlarge instances).
> 
> Behavior is that, during a call, brief pauses or notable audio loss will occur. This is on high bandwidth links that are otherwise stable. Freeswitch logs with max debug spew out "Hot Hit 1" through "Hot Hit 10" and eventually "auto-flush catching up 1 packet(s)" in rapid succession (usually going through the cycle 4-5 times) before things settle again. Obviously that means a minimum of 4-5 audio packets were dropped within the span of a second which results in considerable audio artifacting.
> 
> Changing rtp-timer-name to none, which I understand to perform synchronous reads of RTP audio (as opposed to timer-based async reads) makes the audio notably smoother. That said, I'm having a hard time uncovering the consequences of doing this. Obviously I understand that reads will block the RTP thread, but I can't seem to understand the potential ramifications of this. Could anyone help clarify?
> 
> My other question is: assuming "timer while hot" indicates what I believe it does (that when the timer hit there was >1 packet in the queue to be read), couldn't this issue also just be caused by network jitter, and not necessarily just timer inconsistencies?
> 
> Thanks in advance.
> 
> Best,
> Colin
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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


From colin.morelli at gmail.com  Tue Feb  6 02:07:03 2018
From: colin.morelli at gmail.com (Colin Morelli)
Date: Tue, 06 Feb 2018 02:07:03 +0000
Subject: [Freeswitch-users] rtp-timer-name / timer issues
In-Reply-To: <AED238AC-BEC6-4A1F-A28E-B7E2709ACFDA@connectfirst.com>
References: <CAPtU-UqfL-kQNYzPzts84UppsEjhvVbv=wJOpRnxZTCm-d184Q@mail.gmail.com>
 <AED238AC-BEC6-4A1F-A28E-B7E2709ACFDA@connectfirst.com>
Message-ID: <CAPtU-UqzGnKbv650Vv3fnMWJqnskh+GvvPPGu96UQh95_vKaAA@mail.gmail.com>

Ubuntu 16.04 LTS. Do you have any custom kernel settings or just a stock
CentOS instance?
On Mon, Feb 5, 2018 at 8:23 PM Geoff Mina <gmina at connectfirst.com> wrote:

> What OS are you using? We run FS at over 600 concurrent calls on a smaller
> EC2 (c4.xlarge) size and have not had any issues in either US-East or
> US-West.
>
> Running FS Installed from YUM. CentOS7.
>
> > On Feb 5, 2018, at 5:56 PM, Colin Morelli <colin.morelli at gmail.com>
> wrote:
> >
> > Hey list,
> >
> > I'm running FS on EC2 (I know, I know). Having some issues with random
> packet loss, which I believe almost certainly I have narrowed down to timer
> issues and/or network latency/jitter (seems surprising since I'm using
> c5.xlarge instances).
> >
> > Behavior is that, during a call, brief pauses or notable audio loss will
> occur. This is on high bandwidth links that are otherwise stable.
> Freeswitch logs with max debug spew out "Hot Hit 1" through "Hot Hit 10"
> and eventually "auto-flush catching up 1 packet(s)" in rapid succession
> (usually going through the cycle 4-5 times) before things settle again.
> Obviously that means a minimum of 4-5 audio packets were dropped within the
> span of a second which results in considerable audio artifacting.
> >
> > Changing rtp-timer-name to none, which I understand to perform
> synchronous reads of RTP audio (as opposed to timer-based async reads)
> makes the audio notably smoother. That said, I'm having a hard time
> uncovering the consequences of doing this. Obviously I understand that
> reads will block the RTP thread, but I can't seem to understand the
> potential ramifications of this. Could anyone help clarify?
> >
> > My other question is: assuming "timer while hot" indicates what I
> believe it does (that when the timer hit there was >1 packet in the queue
> to be read), couldn't this issue also just be caused by network jitter, and
> not necessarily just timer inconsistencies?
> >
> > Thanks in advance.
> >
> > Best,
> > Colin
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 gmina at connectfirst.com  Tue Feb  6 02:28:15 2018
From: gmina at connectfirst.com (Geoff Mina)
Date: Mon, 5 Feb 2018 19:28:15 -0700
Subject: [Freeswitch-users] rtp-timer-name / timer issues
In-Reply-To: <CAPtU-UqzGnKbv650Vv3fnMWJqnskh+GvvPPGu96UQh95_vKaAA@mail.gmail.com>
References: <CAPtU-UqfL-kQNYzPzts84UppsEjhvVbv=wJOpRnxZTCm-d184Q@mail.gmail.com>
 <AED238AC-BEC6-4A1F-A28E-B7E2709ACFDA@connectfirst.com>
 <CAPtU-UqzGnKbv650Vv3fnMWJqnskh+GvvPPGu96UQh95_vKaAA@mail.gmail.com>
Message-ID: <CAKzROs1GWmcpdL5R33RN5d6Y46UjjRXqHehAWBO3dSZz5h0wLA@mail.gmail.com>

Stock CentOS w/ Kernel 3.10.0-693.el7.x86_64.  Freeswitch 1.6.19~64bit

We have TLS Websocket /DTLS-SRTP connections on one side and standard UDP
SIP on the other.  Both sides using G.711, so no transcoding.




On Mon, Feb 5, 2018 at 7:07 PM, Colin Morelli <colin.morelli at gmail.com>
wrote:

> Ubuntu 16.04 LTS. Do you have any custom kernel settings or just a stock
> CentOS instance?
>
> On Mon, Feb 5, 2018 at 8:23 PM Geoff Mina <gmina at connectfirst.com> wrote:
>
>> What OS are you using? We run FS at over 600 concurrent calls on a
>> smaller EC2 (c4.xlarge) size and have not had any issues in either US-East
>> or US-West.
>>
>> Running FS Installed from YUM. CentOS7.
>>
>> > On Feb 5, 2018, at 5:56 PM, Colin Morelli <colin.morelli at gmail.com>
>> wrote:
>> >
>> > Hey list,
>> >
>> > I'm running FS on EC2 (I know, I know). Having some issues with random
>> packet loss, which I believe almost certainly I have narrowed down to timer
>> issues and/or network latency/jitter (seems surprising since I'm using
>> c5.xlarge instances).
>> >
>> > Behavior is that, during a call, brief pauses or notable audio loss
>> will occur. This is on high bandwidth links that are otherwise stable.
>> Freeswitch logs with max debug spew out "Hot Hit 1" through "Hot Hit 10"
>> and eventually "auto-flush catching up 1 packet(s)" in rapid succession
>> (usually going through the cycle 4-5 times) before things settle again.
>> Obviously that means a minimum of 4-5 audio packets were dropped within the
>> span of a second which results in considerable audio artifacting.
>> >
>> > Changing rtp-timer-name to none, which I understand to perform
>> synchronous reads of RTP audio (as opposed to timer-based async reads)
>> makes the audio notably smoother. That said, I'm having a hard time
>> uncovering the consequences of doing this. Obviously I understand that
>> reads will block the RTP thread, but I can't seem to understand the
>> potential ramifications of this. Could anyone help clarify?
>> >
>> > My other question is: assuming "timer while hot" indicates what I
>> believe it does (that when the timer hit there was >1 packet in the queue
>> to be read), couldn't this issue also just be caused by network jitter, and
>> not necessarily just timer inconsistencies?
>> >
>> > Thanks in advance.
>> >
>> > Best,
>> > Colin
>> >
>> >
>> > ____________________________________________________________
>> _____________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 colin.morelli at gmail.com  Tue Feb  6 02:38:56 2018
From: colin.morelli at gmail.com (Colin Morelli)
Date: Mon, 5 Feb 2018 21:38:56 -0500
Subject: [Freeswitch-users] rtp-timer-name / timer issues
In-Reply-To: <CAKzROs1GWmcpdL5R33RN5d6Y46UjjRXqHehAWBO3dSZz5h0wLA@mail.gmail.com>
References: <CAPtU-UqfL-kQNYzPzts84UppsEjhvVbv=wJOpRnxZTCm-d184Q@mail.gmail.com>
 <AED238AC-BEC6-4A1F-A28E-B7E2709ACFDA@connectfirst.com>
 <CAPtU-UqzGnKbv650Vv3fnMWJqnskh+GvvPPGu96UQh95_vKaAA@mail.gmail.com>
 <CAKzROs1GWmcpdL5R33RN5d6Y46UjjRXqHehAWBO3dSZz5h0wLA@mail.gmail.com>
Message-ID: <CAPtU-Up5y_OjbV5Sn1EkaC8oMsh9uS_q0E9y8C4KKZO3MhDKsw@mail.gmail.com>

Appreciate the info. We're running on Kernel 4.4.0, with freeswitch 1.9.0
(compiled from git). Large differences on versions there but this seems so
fundamental and core to FS that I'd be surprised if later versions made
things worse. That said, I'm not sure when things like timerfd became
standard in freeswitch, and/or if that (or related concerns) could cause
the issue. We're also running in Docker, but we run it without any limits
(and can replicate the issue running outside of a conctainer).

Similar situation, TLS Websocket w/ DTLS-SRTP on one side and standard
TLS+SRTP on the other. All of our issues appear to occur on the WebRTC
legs, however, which would leads me to lean towards network issues being
the primary culprit. However, iperf shows 400Mbps easily (and consistently)
between the endpoints. Issue occurs whether we do G.711 on both ends, or
opus on the public internet side (although opus PLC can help mask some of
the issues)

Just to be clear, the audio is still intelligible, there's just noticeable
dropped packets every 5-10 seconds on average.

Do you know what timer your instance is using? And your settings for
rtp-timer-name?

Thanks again,
Colin

On Mon, Feb 5, 2018 at 9:28 PM, Geoff Mina <gmina at connectfirst.com> wrote:

> Stock CentOS w/ Kernel 3.10.0-693.el7.x86_64.  Freeswitch 1.6.19~64bit
>
> We have TLS Websocket /DTLS-SRTP connections on one side and standard UDP
> SIP on the other.  Both sides using G.711, so no transcoding.
>
>
>
>
> On Mon, Feb 5, 2018 at 7:07 PM, Colin Morelli <colin.morelli at gmail.com>
> wrote:
>
>> Ubuntu 16.04 LTS. Do you have any custom kernel settings or just a stock
>> CentOS instance?
>>
>> On Mon, Feb 5, 2018 at 8:23 PM Geoff Mina <gmina at connectfirst.com> wrote:
>>
>>> What OS are you using? We run FS at over 600 concurrent calls on a
>>> smaller EC2 (c4.xlarge) size and have not had any issues in either US-East
>>> or US-West.
>>>
>>> Running FS Installed from YUM. CentOS7.
>>>
>>> > On Feb 5, 2018, at 5:56 PM, Colin Morelli <colin.morelli at gmail.com>
>>> wrote:
>>> >
>>> > Hey list,
>>> >
>>> > I'm running FS on EC2 (I know, I know). Having some issues with random
>>> packet loss, which I believe almost certainly I have narrowed down to timer
>>> issues and/or network latency/jitter (seems surprising since I'm using
>>> c5.xlarge instances).
>>> >
>>> > Behavior is that, during a call, brief pauses or notable audio loss
>>> will occur. This is on high bandwidth links that are otherwise stable.
>>> Freeswitch logs with max debug spew out "Hot Hit 1" through "Hot Hit 10"
>>> and eventually "auto-flush catching up 1 packet(s)" in rapid succession
>>> (usually going through the cycle 4-5 times) before things settle again.
>>> Obviously that means a minimum of 4-5 audio packets were dropped within the
>>> span of a second which results in considerable audio artifacting.
>>> >
>>> > Changing rtp-timer-name to none, which I understand to perform
>>> synchronous reads of RTP audio (as opposed to timer-based async reads)
>>> makes the audio notably smoother. That said, I'm having a hard time
>>> uncovering the consequences of doing this. Obviously I understand that
>>> reads will block the RTP thread, but I can't seem to understand the
>>> potential ramifications of this. Could anyone help clarify?
>>> >
>>> > My other question is: assuming "timer while hot" indicates what I
>>> believe it does (that when the timer hit there was >1 packet in the queue
>>> to be read), couldn't this issue also just be caused by network jitter, and
>>> not necessarily just timer inconsistencies?
>>> >
>>> > Thanks in advance.
>>> >
>>> > Best,
>>> > Colin
>>> >
>>> >
>>> > ____________________________________________________________
>>> _____________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://confluence.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
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 colin.morelli at gmail.com  Tue Feb  6 03:54:53 2018
From: colin.morelli at gmail.com (Colin Morelli)
Date: Mon, 5 Feb 2018 22:54:53 -0500
Subject: [Freeswitch-users] rtp-timer-name / timer issues
In-Reply-To: <CAPtU-Up5y_OjbV5Sn1EkaC8oMsh9uS_q0E9y8C4KKZO3MhDKsw@mail.gmail.com>
References: <CAPtU-UqfL-kQNYzPzts84UppsEjhvVbv=wJOpRnxZTCm-d184Q@mail.gmail.com>
 <AED238AC-BEC6-4A1F-A28E-B7E2709ACFDA@connectfirst.com>
 <CAPtU-UqzGnKbv650Vv3fnMWJqnskh+GvvPPGu96UQh95_vKaAA@mail.gmail.com>
 <CAKzROs1GWmcpdL5R33RN5d6Y46UjjRXqHehAWBO3dSZz5h0wLA@mail.gmail.com>
 <CAPtU-Up5y_OjbV5Sn1EkaC8oMsh9uS_q0E9y8C4KKZO3MhDKsw@mail.gmail.com>
Message-ID: <CAPtU-Uqn_q+mg7YNiQrK5uDA4ON75CEkQZGNVURJ9zOQGykYDw@mail.gmail.com>

Alright, so I've captured a recording on the PSTN leg and WebRTC leg
simultaneously. On the PSTN leg, the audio is perfectly smooth with no
dropped packets. The recording on the WebRTC leg does have noticeable
dropped packets. I believe this would all but eliminate network concerns,
unless I'm misunderstanding something.

Additionally, running iperf between a machine that experiences the issue,
and the media server, easily pushes !00Mbps with < 0.5ms jitter. This
happens on multiple client internet connections ranging from 50Mbps to
1Gbps, both residential and commercial.

Any guidance on diving deeper into timer issues? Or other settings to
change?

Thanks,
Colin

On Mon, Feb 5, 2018 at 9:38 PM, Colin Morelli <colin.morelli at gmail.com>
wrote:

> Appreciate the info. We're running on Kernel 4.4.0, with freeswitch 1.9.0
> (compiled from git). Large differences on versions there but this seems so
> fundamental and core to FS that I'd be surprised if later versions made
> things worse. That said, I'm not sure when things like timerfd became
> standard in freeswitch, and/or if that (or related concerns) could cause
> the issue. We're also running in Docker, but we run it without any limits
> (and can replicate the issue running outside of a conctainer).
>
> Similar situation, TLS Websocket w/ DTLS-SRTP on one side and standard
> TLS+SRTP on the other. All of our issues appear to occur on the WebRTC
> legs, however, which would leads me to lean towards network issues being
> the primary culprit. However, iperf shows 400Mbps easily (and consistently)
> between the endpoints. Issue occurs whether we do G.711 on both ends, or
> opus on the public internet side (although opus PLC can help mask some of
> the issues)
>
> Just to be clear, the audio is still intelligible, there's just noticeable
> dropped packets every 5-10 seconds on average.
>
> Do you know what timer your instance is using? And your settings for
> rtp-timer-name?
>
> Thanks again,
> Colin
>
> On Mon, Feb 5, 2018 at 9:28 PM, Geoff Mina <gmina at connectfirst.com> wrote:
>
>> Stock CentOS w/ Kernel 3.10.0-693.el7.x86_64.  Freeswitch 1.6.19~64bit
>>
>> We have TLS Websocket /DTLS-SRTP connections on one side and standard UDP
>> SIP on the other.  Both sides using G.711, so no transcoding.
>>
>>
>>
>>
>> On Mon, Feb 5, 2018 at 7:07 PM, Colin Morelli <colin.morelli at gmail.com>
>> wrote:
>>
>>> Ubuntu 16.04 LTS. Do you have any custom kernel settings or just a stock
>>> CentOS instance?
>>>
>>> On Mon, Feb 5, 2018 at 8:23 PM Geoff Mina <gmina at connectfirst.com>
>>> wrote:
>>>
>>>> What OS are you using? We run FS at over 600 concurrent calls on a
>>>> smaller EC2 (c4.xlarge) size and have not had any issues in either US-East
>>>> or US-West.
>>>>
>>>> Running FS Installed from YUM. CentOS7.
>>>>
>>>> > On Feb 5, 2018, at 5:56 PM, Colin Morelli <colin.morelli at gmail.com>
>>>> wrote:
>>>> >
>>>> > Hey list,
>>>> >
>>>> > I'm running FS on EC2 (I know, I know). Having some issues with
>>>> random packet loss, which I believe almost certainly I have narrowed down
>>>> to timer issues and/or network latency/jitter (seems surprising since I'm
>>>> using c5.xlarge instances).
>>>> >
>>>> > Behavior is that, during a call, brief pauses or notable audio loss
>>>> will occur. This is on high bandwidth links that are otherwise stable.
>>>> Freeswitch logs with max debug spew out "Hot Hit 1" through "Hot Hit 10"
>>>> and eventually "auto-flush catching up 1 packet(s)" in rapid succession
>>>> (usually going through the cycle 4-5 times) before things settle again.
>>>> Obviously that means a minimum of 4-5 audio packets were dropped within the
>>>> span of a second which results in considerable audio artifacting.
>>>> >
>>>> > Changing rtp-timer-name to none, which I understand to perform
>>>> synchronous reads of RTP audio (as opposed to timer-based async reads)
>>>> makes the audio notably smoother. That said, I'm having a hard time
>>>> uncovering the consequences of doing this. Obviously I understand that
>>>> reads will block the RTP thread, but I can't seem to understand the
>>>> potential ramifications of this. Could anyone help clarify?
>>>> >
>>>> > My other question is: assuming "timer while hot" indicates what I
>>>> believe it does (that when the timer hit there was >1 packet in the queue
>>>> to be read), couldn't this issue also just be caused by network jitter, and
>>>> not necessarily just timer inconsistencies?
>>>> >
>>>> > Thanks in advance.
>>>> >
>>>> > Best,
>>>> > Colin
>>>> >
>>>> >
>>>> > ____________________________________________________________
>>>> _____________
>>>> > Professional FreeSWITCH Consulting Services:
>>>> > consulting at freeswitch.org
>>>> > http://www.freeswitchsolutions.com
>>>> >
>>>> > Official FreeSWITCH Sites
>>>> > http://www.freeswitch.org
>>>> > http://confluence.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/free
>>>> switch-users
>>>> > http://www.freeswitch.org
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.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/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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 brians at iptel.co  Tue Feb  6 08:48:22 2018
From: brians at iptel.co (Brian :)
Date: Tue, 6 Feb 2018 08:48:22 +0000
Subject: [Freeswitch-users] rtp-timer-name / timer issues
In-Reply-To: <CAPtU-UqfL-kQNYzPzts84UppsEjhvVbv=wJOpRnxZTCm-d184Q@mail.gmail.com>
References: <CAPtU-UqfL-kQNYzPzts84UppsEjhvVbv=wJOpRnxZTCm-d184Q@mail.gmail.com>
Message-ID: <CAGPQfi98UEZixqVtO=vW2EWgFdhBvHyumtPp6Lh=-7XKa68etA@mail.gmail.com>

Hi Colin,

Depending on what you are doing with Freeswitch setting the rtp-timer
to none can produce all sorts of subtle weirdness. I would advise
against it. 2 things that I remember from our tests with this was lots
of blocked / hung calls that would build and need to be HUPed and also
carriers that would send SDP but no RTP when silence was being sent -
the call wouldn't progress through dialplan - it just got blocked
waiting on RTP>

B

On Tue, Feb 6, 2018 at 12:56 AM, Colin Morelli <colin.morelli at gmail.com> wrote:
> Hey list,
>
> I'm running FS on EC2 (I know, I know). Having some issues with random
> packet loss, which I believe almost certainly I have narrowed down to timer
> issues and/or network latency/jitter (seems surprising since I'm using
> c5.xlarge instances).
>
> Behavior is that, during a call, brief pauses or notable audio loss will
> occur. This is on high bandwidth links that are otherwise stable. Freeswitch
> logs with max debug spew out "Hot Hit 1" through "Hot Hit 10" and eventually
> "auto-flush catching up 1 packet(s)" in rapid succession (usually going
> through the cycle 4-5 times) before things settle again. Obviously that
> means a minimum of 4-5 audio packets were dropped within the span of a
> second which results in considerable audio artifacting.
>
> Changing rtp-timer-name to none, which I understand to perform synchronous
> reads of RTP audio (as opposed to timer-based async reads) makes the audio
> notably smoother. That said, I'm having a hard time uncovering the
> consequences of doing this. Obviously I understand that reads will block the
> RTP thread, but I can't seem to understand the potential ramifications of
> this. Could anyone help clarify?
>
> My other question is: assuming "timer while hot" indicates what I believe it
> does (that when the timer hit there was >1 packet in the queue to be read),
> couldn't this issue also just be caused by network jitter, and not
> necessarily just timer inconsistencies?
>
> Thanks in advance.
>
> Best,
> Colin
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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


From bilaln018 at gmail.com  Tue Feb  6 10:42:27 2018
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Tue, 6 Feb 2018 15:42:27 +0500
Subject: [Freeswitch-users] [Lumenvox][mod_unimrcp][Issue]
Message-ID: <CADSVerBt+r6GAAU+pWEX6aRRO0KKtygEhKoJMAZaeZUwhXe4VQ@mail.gmail.com>

Hi Users,
I am using FreeSWITCH mod_unimrcp to detect VM using the lumenvox,
Below is my configuration,

<action application="set" data="play_and_detect_speech_close_asr=true"/>

<action application="play_and_detect_speech" data="${sound_file_string}
detect:unimrcp:lumenvox
{start-input-timers=false,no-input-timeout=5000,recognition-timeout=5000}builtin:special/amd"
/>

everything works really good, but under testing i stopped the lumenvox SIP
server(called lvmediaserverd server), and when i tried to make a call i was
not able to hear the message that needs to be played.

I need to know any parameter/configuration that can play the sound file
even the lumenvox is down in uni_mrcp(as i don't want to get the service
down if lumenvox server is down, max loss i should get is the VM detection
that will surely stop working, but i want my messages to be delivered to
live users).

Regards
Abbasi
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From colin.morelli at gmail.com  Tue Feb  6 12:09:15 2018
From: colin.morelli at gmail.com (Colin Morelli)
Date: Tue, 06 Feb 2018 12:09:15 +0000
Subject: [Freeswitch-users] rtp-timer-name / timer issues
In-Reply-To: <CAGPQfi98UEZixqVtO=vW2EWgFdhBvHyumtPp6Lh=-7XKa68etA@mail.gmail.com>
References: <CAPtU-UqfL-kQNYzPzts84UppsEjhvVbv=wJOpRnxZTCm-d184Q@mail.gmail.com>
 <CAGPQfi98UEZixqVtO=vW2EWgFdhBvHyumtPp6Lh=-7XKa68etA@mail.gmail.com>
Message-ID: <CAPtU-UoTYL1KvXTAqNYzu=0u_QyM6-5oegdLoytVNUKrj3OfGA@mail.gmail.com>

Appreciate the input, Brian. I’ll definitely try to avoid setting the timer
option.

In other news. I deployed The exact same FS instance (same docker
container) on baremetal last night and it experiences the same issue. So,
virtualization does not appear to be the problem. I just can’t figure out
what else would cause this. I’m sure it’s something simple.
On Tue, Feb 6, 2018 at 3:49 AM Brian : <brians at iptel.co> wrote:

> Hi Colin,
>
> Depending on what you are doing with Freeswitch setting the rtp-timer
> to none can produce all sorts of subtle weirdness. I would advise
> against it. 2 things that I remember from our tests with this was lots
> of blocked / hung calls that would build and need to be HUPed and also
> carriers that would send SDP but no RTP when silence was being sent -
> the call wouldn't progress through dialplan - it just got blocked
> waiting on RTP>
>
> B
>
> On Tue, Feb 6, 2018 at 12:56 AM, Colin Morelli <colin.morelli at gmail.com>
> wrote:
> > Hey list,
> >
> > I'm running FS on EC2 (I know, I know). Having some issues with random
> > packet loss, which I believe almost certainly I have narrowed down to
> timer
> > issues and/or network latency/jitter (seems surprising since I'm using
> > c5.xlarge instances).
> >
> > Behavior is that, during a call, brief pauses or notable audio loss will
> > occur. This is on high bandwidth links that are otherwise stable.
> Freeswitch
> > logs with max debug spew out "Hot Hit 1" through "Hot Hit 10" and
> eventually
> > "auto-flush catching up 1 packet(s)" in rapid succession (usually going
> > through the cycle 4-5 times) before things settle again. Obviously that
> > means a minimum of 4-5 audio packets were dropped within the span of a
> > second which results in considerable audio artifacting.
> >
> > Changing rtp-timer-name to none, which I understand to perform
> synchronous
> > reads of RTP audio (as opposed to timer-based async reads) makes the
> audio
> > notably smoother. That said, I'm having a hard time uncovering the
> > consequences of doing this. Obviously I understand that reads will block
> the
> > RTP thread, but I can't seem to understand the potential ramifications of
> > this. Could anyone help clarify?
> >
> > My other question is: assuming "timer while hot" indicates what I
> believe it
> > does (that when the timer hit there was >1 packet in the queue to be
> read),
> > couldn't this issue also just be caused by network jitter, and not
> > necessarily just timer inconsistencies?
> >
> > Thanks in advance.
> >
> > Best,
> > Colin
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 gmina at connectfirst.com  Tue Feb  6 13:45:32 2018
From: gmina at connectfirst.com (Geoff Mina)
Date: Tue, 6 Feb 2018 06:45:32 -0700
Subject: [Freeswitch-users] rtp-timer-name / timer issues
In-Reply-To: <CAPtU-UoTYL1KvXTAqNYzu=0u_QyM6-5oegdLoytVNUKrj3OfGA@mail.gmail.com>
References: <CAPtU-UqfL-kQNYzPzts84UppsEjhvVbv=wJOpRnxZTCm-d184Q@mail.gmail.com>
 <CAGPQfi98UEZixqVtO=vW2EWgFdhBvHyumtPp6Lh=-7XKa68etA@mail.gmail.com>
 <CAPtU-UoTYL1KvXTAqNYzu=0u_QyM6-5oegdLoytVNUKrj3OfGA@mail.gmail.com>
Message-ID: <C90EFC3B-7F28-4208-9013-279AF231831F@connectfirst.com>

What WebRTC client are you using? Does this happen in all browsers or just one?

> On Feb 6, 2018, at 5:09 AM, Colin Morelli <colin.morelli at gmail.com> wrote:
> 
> Appreciate the input, Brian. I’ll definitely try to avoid setting the timer option.
> 
> In other news. I deployed The exact same FS instance (same docker container) on baremetal last night and it experiences the same issue. So, virtualization does not appear to be the problem. I just can’t figure out what else would cause this. I’m sure it’s something simple.
>> On Tue, Feb 6, 2018 at 3:49 AM Brian : <brians at iptel.co> wrote:
>> Hi Colin,
>> 
>> Depending on what you are doing with Freeswitch setting the rtp-timer
>> to none can produce all sorts of subtle weirdness. I would advise
>> against it. 2 things that I remember from our tests with this was lots
>> of blocked / hung calls that would build and need to be HUPed and also
>> carriers that would send SDP but no RTP when silence was being sent -
>> the call wouldn't progress through dialplan - it just got blocked
>> waiting on RTP>
>> 
>> B
>> 
>> On Tue, Feb 6, 2018 at 12:56 AM, Colin Morelli <colin.morelli at gmail.com> wrote:
>> > Hey list,
>> >
>> > I'm running FS on EC2 (I know, I know). Having some issues with random
>> > packet loss, which I believe almost certainly I have narrowed down to timer
>> > issues and/or network latency/jitter (seems surprising since I'm using
>> > c5.xlarge instances).
>> >
>> > Behavior is that, during a call, brief pauses or notable audio loss will
>> > occur. This is on high bandwidth links that are otherwise stable. Freeswitch
>> > logs with max debug spew out "Hot Hit 1" through "Hot Hit 10" and eventually
>> > "auto-flush catching up 1 packet(s)" in rapid succession (usually going
>> > through the cycle 4-5 times) before things settle again. Obviously that
>> > means a minimum of 4-5 audio packets were dropped within the span of a
>> > second which results in considerable audio artifacting.
>> >
>> > Changing rtp-timer-name to none, which I understand to perform synchronous
>> > reads of RTP audio (as opposed to timer-based async reads) makes the audio
>> > notably smoother. That said, I'm having a hard time uncovering the
>> > consequences of doing this. Obviously I understand that reads will block the
>> > RTP thread, but I can't seem to understand the potential ramifications of
>> > this. Could anyone help clarify?
>> >
>> > My other question is: assuming "timer while hot" indicates what I believe it
>> > does (that when the timer hit there was >1 packet in the queue to be read),
>> > couldn't this issue also just be caused by network jitter, and not
>> > necessarily just timer inconsistencies?
>> >
>> > Thanks in advance.
>> >
>> > Best,
>> > Colin
>> >
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.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
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 colin.morelli at gmail.com  Tue Feb  6 14:04:17 2018
From: colin.morelli at gmail.com (Colin Morelli)
Date: Tue, 6 Feb 2018 09:04:17 -0500
Subject: [Freeswitch-users] rtp-timer-name / timer issues
In-Reply-To: <C90EFC3B-7F28-4208-9013-279AF231831F@connectfirst.com>
References: <CAPtU-UqfL-kQNYzPzts84UppsEjhvVbv=wJOpRnxZTCm-d184Q@mail.gmail.com>
 <CAGPQfi98UEZixqVtO=vW2EWgFdhBvHyumtPp6Lh=-7XKa68etA@mail.gmail.com>
 <CAPtU-UoTYL1KvXTAqNYzu=0u_QyM6-5oegdLoytVNUKrj3OfGA@mail.gmail.com>
 <C90EFC3B-7F28-4208-9013-279AF231831F@connectfirst.com>
Message-ID: <CAPtU-UpcE2PeATJ_rpWTazo8cLQVVnL4s80R6apKGUYWtv0dkQ@mail.gmail.com>

Happens on all browsers.

Just want to clarify my previous message, though. I had a call bridged A ->
B (A is the WebRTC side, B the PSTN). I recorded both legs of the call
individually. On the recording for B, B's audio is clear and smooth. On the
recording for A, B's audio has dropped packets that correspond with the
logs mentioned on FS. Unless I'm misunderstanding something I believe this
should eliminate network/WebRTC/clients as being the issue.

On Tue, Feb 6, 2018 at 8:45 AM, Geoff Mina <gmina at connectfirst.com> wrote:

> What WebRTC client are you using? Does this happen in all browsers or just
> one?
>
>
> On Feb 6, 2018, at 5:09 AM, Colin Morelli <colin.morelli at gmail.com> wrote:
>
> Appreciate the input, Brian. I’ll definitely try to avoid setting the
> timer option.
>
> In other news. I deployed The exact same FS instance (same docker
> container) on baremetal last night and it experiences the same issue. So,
> virtualization does not appear to be the problem. I just can’t figure out
> what else would cause this. I’m sure it’s something simple.
> On Tue, Feb 6, 2018 at 3:49 AM Brian : <brians at iptel.co> wrote:
>
>> Hi Colin,
>>
>> Depending on what you are doing with Freeswitch setting the rtp-timer
>> to none can produce all sorts of subtle weirdness. I would advise
>> against it. 2 things that I remember from our tests with this was lots
>> of blocked / hung calls that would build and need to be HUPed and also
>> carriers that would send SDP but no RTP when silence was being sent -
>> the call wouldn't progress through dialplan - it just got blocked
>> waiting on RTP>
>>
>> B
>>
>> On Tue, Feb 6, 2018 at 12:56 AM, Colin Morelli <colin.morelli at gmail.com>
>> wrote:
>> > Hey list,
>> >
>> > I'm running FS on EC2 (I know, I know). Having some issues with random
>> > packet loss, which I believe almost certainly I have narrowed down to
>> timer
>> > issues and/or network latency/jitter (seems surprising since I'm using
>> > c5.xlarge instances).
>> >
>> > Behavior is that, during a call, brief pauses or notable audio loss will
>> > occur. This is on high bandwidth links that are otherwise stable.
>> Freeswitch
>> > logs with max debug spew out "Hot Hit 1" through "Hot Hit 10" and
>> eventually
>> > "auto-flush catching up 1 packet(s)" in rapid succession (usually going
>> > through the cycle 4-5 times) before things settle again. Obviously that
>> > means a minimum of 4-5 audio packets were dropped within the span of a
>> > second which results in considerable audio artifacting.
>> >
>> > Changing rtp-timer-name to none, which I understand to perform
>> synchronous
>> > reads of RTP audio (as opposed to timer-based async reads) makes the
>> audio
>> > notably smoother. That said, I'm having a hard time uncovering the
>> > consequences of doing this. Obviously I understand that reads will
>> block the
>> > RTP thread, but I can't seem to understand the potential ramifications
>> of
>> > this. Could anyone help clarify?
>> >
>> > My other question is: assuming "timer while hot" indicates what I
>> believe it
>> > does (that when the timer hit there was >1 packet in the queue to be
>> read),
>> > couldn't this issue also just be caused by network jitter, and not
>> > necessarily just timer inconsistencies?
>> >
>> > Thanks in advance.
>> >
>> > Best,
>> > Colin
>> >
>> >
>> >
>> > ____________________________________________________________
>> _____________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 ryharris at airmail.cc  Mon Feb  5 21:08:39 2018
From: ryharris at airmail.cc (Ryan Harris)
Date: Mon, 5 Feb 2018 16:08:39 -0500
Subject: [Freeswitch-users] Stream inbound Freeswitch (RTMP?)
In-Reply-To: <CAMALkh2pQ9u3ZWzWB+PPCMQh-p0mgMcRzXG4VJgiRBYm58ba3A@mail.gmail.com>
References: <CAMALkh2pQ9u3ZWzWB+PPCMQh-p0mgMcRzXG4VJgiRBYm58ba3A@mail.gmail.com>
Message-ID: <3af2c4a7-61b4-71ea-8e1a-51a86bef4a5f@airmail.cc>

On 02/04/2018 02:37 PM, Scott Howell wrote:
> I have an audio/video stream that I already set up, I would like to
> use the audio portion as a "conference" that users can dial into and
> listen live.

I'd go for mod_shout or mod_vlc
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From kevinchi at foxmail.com  Tue Feb  6 04:28:17 2018
From: kevinchi at foxmail.com (=?ISO-8859-1?B?S2V2aW4gQ2hp?=)
Date: Tue, 6 Feb 2018 12:28:17 +0800
Subject: [Freeswitch-users] "fsctl recover" doesn't work on Freeswitch
	V1.6.19
Message-ID: <tencent_02C00030E26255102C8403FB325D82811A09@qq.com>

hello everyone,


The steps of my call recovery test as below.
1.First I added  the line "<param name="track-calls" value="true"/>" to internal.xml.
2.UA called 9664 to play the hold music, and then run "fsctl crash" command.
3.start freeswtich and run "fsctl recover" command.
4.call was interrupted, recover failed.


I opened siptrace before "fsctl recover", the sipflow make me can not understand.
freeswitch send INVITE to UA
freeswitch recive 200 from UA
freeswitch send BYE to UA
freeswitch recive 200 from UA
freeswitch send ACK from UA


My freeswitch is only single server, the version is V1.6.19. I have tried the newer version V1.6.20, but the result was same.
What's the reason? Pls give me some suggestion, thx a lot.


------------------
With regards,

Kevin Chi
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From colin.morelli at gmail.com  Tue Feb  6 15:32:00 2018
From: colin.morelli at gmail.com (Colin Morelli)
Date: Tue, 6 Feb 2018 10:32:00 -0500
Subject: [Freeswitch-users] rtp-timer-name / timer issues
In-Reply-To: <CAPtU-UpcE2PeATJ_rpWTazo8cLQVVnL4s80R6apKGUYWtv0dkQ@mail.gmail.com>
References: <CAPtU-UqfL-kQNYzPzts84UppsEjhvVbv=wJOpRnxZTCm-d184Q@mail.gmail.com>
 <CAGPQfi98UEZixqVtO=vW2EWgFdhBvHyumtPp6Lh=-7XKa68etA@mail.gmail.com>
 <CAPtU-UoTYL1KvXTAqNYzu=0u_QyM6-5oegdLoytVNUKrj3OfGA@mail.gmail.com>
 <C90EFC3B-7F28-4208-9013-279AF231831F@connectfirst.com>
 <CAPtU-UpcE2PeATJ_rpWTazo8cLQVVnL4s80R6apKGUYWtv0dkQ@mail.gmail.com>
Message-ID: <CAPtU-Upa2mLmocuc8zbX5oCvZz+zwmoeuyHCLVraZRTcoMh=eA@mail.gmail.com>

Tested again on a fresh EC2 instances (c5.xlarge) running Debian Jessie
(Kernel 3.16.0-4-amd64), since I believe that's the current recommendation,
with packages installed from the Freeswitch mainline
(version 1.6.20-37-987c9b9~64bit) and vanilla configs. I am still able to
reproduce issues where one side's audio recording drops packets that are
present in the other side. Running out of things to look at here, since I
was able to repro on baremetal as well.

On Tue, Feb 6, 2018 at 9:04 AM, Colin Morelli <colin.morelli at gmail.com>
wrote:

> Happens on all browsers.
>
> Just want to clarify my previous message, though. I had a call bridged A
> -> B (A is the WebRTC side, B the PSTN). I recorded both legs of the call
> individually. On the recording for B, B's audio is clear and smooth. On the
> recording for A, B's audio has dropped packets that correspond with the
> logs mentioned on FS. Unless I'm misunderstanding something I believe this
> should eliminate network/WebRTC/clients as being the issue.
>
> On Tue, Feb 6, 2018 at 8:45 AM, Geoff Mina <gmina at connectfirst.com> wrote:
>
>> What WebRTC client are you using? Does this happen in all browsers or
>> just one?
>>
>>
>> On Feb 6, 2018, at 5:09 AM, Colin Morelli <colin.morelli at gmail.com>
>> wrote:
>>
>> Appreciate the input, Brian. I’ll definitely try to avoid setting the
>> timer option.
>>
>> In other news. I deployed The exact same FS instance (same docker
>> container) on baremetal last night and it experiences the same issue. So,
>> virtualization does not appear to be the problem. I just can’t figure out
>> what else would cause this. I’m sure it’s something simple.
>> On Tue, Feb 6, 2018 at 3:49 AM Brian : <brians at iptel.co> wrote:
>>
>>> Hi Colin,
>>>
>>> Depending on what you are doing with Freeswitch setting the rtp-timer
>>> to none can produce all sorts of subtle weirdness. I would advise
>>> against it. 2 things that I remember from our tests with this was lots
>>> of blocked / hung calls that would build and need to be HUPed and also
>>> carriers that would send SDP but no RTP when silence was being sent -
>>> the call wouldn't progress through dialplan - it just got blocked
>>> waiting on RTP>
>>>
>>> B
>>>
>>> On Tue, Feb 6, 2018 at 12:56 AM, Colin Morelli <colin.morelli at gmail.com>
>>> wrote:
>>> > Hey list,
>>> >
>>> > I'm running FS on EC2 (I know, I know). Having some issues with random
>>> > packet loss, which I believe almost certainly I have narrowed down to
>>> timer
>>> > issues and/or network latency/jitter (seems surprising since I'm using
>>> > c5.xlarge instances).
>>> >
>>> > Behavior is that, during a call, brief pauses or notable audio loss
>>> will
>>> > occur. This is on high bandwidth links that are otherwise stable.
>>> Freeswitch
>>> > logs with max debug spew out "Hot Hit 1" through "Hot Hit 10" and
>>> eventually
>>> > "auto-flush catching up 1 packet(s)" in rapid succession (usually going
>>> > through the cycle 4-5 times) before things settle again. Obviously that
>>> > means a minimum of 4-5 audio packets were dropped within the span of a
>>> > second which results in considerable audio artifacting.
>>> >
>>> > Changing rtp-timer-name to none, which I understand to perform
>>> synchronous
>>> > reads of RTP audio (as opposed to timer-based async reads) makes the
>>> audio
>>> > notably smoother. That said, I'm having a hard time uncovering the
>>> > consequences of doing this. Obviously I understand that reads will
>>> block the
>>> > RTP thread, but I can't seem to understand the potential ramifications
>>> of
>>> > this. Could anyone help clarify?
>>> >
>>> > My other question is: assuming "timer while hot" indicates what I
>>> believe it
>>> > does (that when the timer hit there was >1 packet in the queue to be
>>> read),
>>> > couldn't this issue also just be caused by network jitter, and not
>>> > necessarily just timer inconsistencies?
>>> >
>>> > Thanks in advance.
>>> >
>>> > Best,
>>> > Colin
>>> >
>>> >
>>> >
>>> > ____________________________________________________________
>>> _____________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://confluence.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
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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 david.villasmil.work at gmail.com  Tue Feb  6 16:38:27 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 6 Feb 2018 11:38:27 -0500
Subject: [Freeswitch-users] is this even possible?
Message-ID: <CAFGRPVqAp4KSGTHuR4ob9qcES68r12UoHpzReOtudaam_Z5yTg@mail.gmail.com>

Hello guys,

So i have a conference running on a fs... and i want to move it to another
fs... is this even possible??

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
ᐧ
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From colin.morelli at gmail.com  Tue Feb  6 18:51:53 2018
From: colin.morelli at gmail.com (Colin Morelli)
Date: Tue, 6 Feb 2018 13:51:53 -0500
Subject: [Freeswitch-users] rtp-timer-name / timer issues
In-Reply-To: <CAPtU-Upa2mLmocuc8zbX5oCvZz+zwmoeuyHCLVraZRTcoMh=eA@mail.gmail.com>
References: <CAPtU-UqfL-kQNYzPzts84UppsEjhvVbv=wJOpRnxZTCm-d184Q@mail.gmail.com>
 <CAGPQfi98UEZixqVtO=vW2EWgFdhBvHyumtPp6Lh=-7XKa68etA@mail.gmail.com>
 <CAPtU-UoTYL1KvXTAqNYzu=0u_QyM6-5oegdLoytVNUKrj3OfGA@mail.gmail.com>
 <C90EFC3B-7F28-4208-9013-279AF231831F@connectfirst.com>
 <CAPtU-UpcE2PeATJ_rpWTazo8cLQVVnL4s80R6apKGUYWtv0dkQ@mail.gmail.com>
 <CAPtU-Upa2mLmocuc8zbX5oCvZz+zwmoeuyHCLVraZRTcoMh=eA@mail.gmail.com>
Message-ID: <CAPtU-UpEwtgpNZ0nyjCW3keokjtigaJ3W+zq97HeG5mGYNSHkw@mail.gmail.com>

Alright, it looks like CNG may be to blame here.

We were using bridge_generate_comfort_noise to deal with some provider
issues and it may be responding poorly to even the slightest latency. If my
understanding is correct, bridge_generate_comfort_noise will generate a
silence packet into the media stream transparently to the endpoints. The
result would be that if the timer hits and no audio is available on a
channel, CNG is immediately generated and placed into the stream, rendering
jitter buffers on either end fairly useless, since they're still receiving
consistent audio streams (just some packets may be silent)

Does this sound reasonable? Short of timing issues it's the only thing I
can think of that would cause the audio differences between the A and B leg
given that a dedicated host didn't solve the problem either.

Thanks in advance,
Colin

On Tue, Feb 6, 2018 at 10:32 AM, Colin Morelli <colin.morelli at gmail.com>
wrote:

> Tested again on a fresh EC2 instances (c5.xlarge) running Debian Jessie
> (Kernel 3.16.0-4-amd64), since I believe that's the current
> recommendation, with packages installed from the Freeswitch mainline
> (version 1.6.20-37-987c9b9~64bit) and vanilla configs. I am still able to
> reproduce issues where one side's audio recording drops packets that are
> present in the other side. Running out of things to look at here, since I
> was able to repro on baremetal as well.
>
> On Tue, Feb 6, 2018 at 9:04 AM, Colin Morelli <colin.morelli at gmail.com>
> wrote:
>
>> Happens on all browsers.
>>
>> Just want to clarify my previous message, though. I had a call bridged A
>> -> B (A is the WebRTC side, B the PSTN). I recorded both legs of the call
>> individually. On the recording for B, B's audio is clear and smooth. On the
>> recording for A, B's audio has dropped packets that correspond with the
>> logs mentioned on FS. Unless I'm misunderstanding something I believe this
>> should eliminate network/WebRTC/clients as being the issue.
>>
>> On Tue, Feb 6, 2018 at 8:45 AM, Geoff Mina <gmina at connectfirst.com>
>> wrote:
>>
>>> What WebRTC client are you using? Does this happen in all browsers or
>>> just one?
>>>
>>>
>>> On Feb 6, 2018, at 5:09 AM, Colin Morelli <colin.morelli at gmail.com>
>>> wrote:
>>>
>>> Appreciate the input, Brian. I’ll definitely try to avoid setting the
>>> timer option.
>>>
>>> In other news. I deployed The exact same FS instance (same docker
>>> container) on baremetal last night and it experiences the same issue. So,
>>> virtualization does not appear to be the problem. I just can’t figure out
>>> what else would cause this. I’m sure it’s something simple.
>>> On Tue, Feb 6, 2018 at 3:49 AM Brian : <brians at iptel.co> wrote:
>>>
>>>> Hi Colin,
>>>>
>>>> Depending on what you are doing with Freeswitch setting the rtp-timer
>>>> to none can produce all sorts of subtle weirdness. I would advise
>>>> against it. 2 things that I remember from our tests with this was lots
>>>> of blocked / hung calls that would build and need to be HUPed and also
>>>> carriers that would send SDP but no RTP when silence was being sent -
>>>> the call wouldn't progress through dialplan - it just got blocked
>>>> waiting on RTP>
>>>>
>>>> B
>>>>
>>>> On Tue, Feb 6, 2018 at 12:56 AM, Colin Morelli <colin.morelli at gmail.com>
>>>> wrote:
>>>> > Hey list,
>>>> >
>>>> > I'm running FS on EC2 (I know, I know). Having some issues with random
>>>> > packet loss, which I believe almost certainly I have narrowed down to
>>>> timer
>>>> > issues and/or network latency/jitter (seems surprising since I'm using
>>>> > c5.xlarge instances).
>>>> >
>>>> > Behavior is that, during a call, brief pauses or notable audio loss
>>>> will
>>>> > occur. This is on high bandwidth links that are otherwise stable.
>>>> Freeswitch
>>>> > logs with max debug spew out "Hot Hit 1" through "Hot Hit 10" and
>>>> eventually
>>>> > "auto-flush catching up 1 packet(s)" in rapid succession (usually
>>>> going
>>>> > through the cycle 4-5 times) before things settle again. Obviously
>>>> that
>>>> > means a minimum of 4-5 audio packets were dropped within the span of a
>>>> > second which results in considerable audio artifacting.
>>>> >
>>>> > Changing rtp-timer-name to none, which I understand to perform
>>>> synchronous
>>>> > reads of RTP audio (as opposed to timer-based async reads) makes the
>>>> audio
>>>> > notably smoother. That said, I'm having a hard time uncovering the
>>>> > consequences of doing this. Obviously I understand that reads will
>>>> block the
>>>> > RTP thread, but I can't seem to understand the potential
>>>> ramifications of
>>>> > this. Could anyone help clarify?
>>>> >
>>>> > My other question is: assuming "timer while hot" indicates what I
>>>> believe it
>>>> > does (that when the timer hit there was >1 packet in the queue to be
>>>> read),
>>>> > couldn't this issue also just be caused by network jitter, and not
>>>> > necessarily just timer inconsistencies?
>>>> >
>>>> > Thanks in advance.
>>>> >
>>>> > Best,
>>>> > Colin
>>>> >
>>>> >
>>>> >
>>>> > ____________________________________________________________
>>>> _____________
>>>> > Professional FreeSWITCH Consulting Services:
>>>> > consulting at freeswitch.org
>>>> > http://www.freeswitchsolutions.com
>>>> >
>>>> > Official FreeSWITCH Sites
>>>> > http://www.freeswitch.org
>>>> > http://confluence.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/free
>>>> switch-users
>>>> > http://www.freeswitch.org
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.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/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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 prestonh at gmail.com  Tue Feb  6 19:02:26 2018
From: prestonh at gmail.com (Preston Hagar)
Date: Tue, 6 Feb 2018 13:02:26 -0600
Subject: [Freeswitch-users] Long distance calling in the US
Message-ID: <CAK6zN=0Xaoe0HRt2MiiRX-UbSO2EOHrE67o25rWmhnGeo0vx-A@mail.gmail.com>

Hello all,

We are setting up FreeSwitch to use FreeTDM through a Sangoma A102DE, which
is connected to a PRI.  We are located in Texas in the US, where we are
required to call the full 10 digits (area code and number) for "local"
numbers and then dial 1 followed by the full 10 digits for calls that are
long distance.

Currently, our dialplan is working as expected to make local calls:

 <!-- regex is NANPA +1NxxNxxXXXX E.164 Dialstring -->
  <extension name="tdm-outgoing">
    <condition field="destination_number"
expression="^9?(1{0,1}[2-9]\d\d[2-9]\d{6})$">
      <action application="log" data="WARNING outbound call through TDM to
$1"/>
      <action application="set"
data="effective_caller_id_number=${outbound_caller_id_number}"/>
      <action application="set"
data="effective_caller_id_name=${outbound_caller_id_name}"/>
      <action application="set" data="call_direction=outbound"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="bridge" data="freetdm/1/a/$1"/>
    </condition>
  </extension>

When we dial a number with a 1 in front of it, the 1 is still included in
the connect lines and all the output I can find from the logs or fs_cli.
For example:

2018-02-06 12:52:28.376794 [DEBUG] mod_freetdm.c:1343 Connect outbound
channel FreeTDM/1:1/17138370311

but we receive an automated message from our local PRI provider saying that
we need to dial 1 in front of the number to dial long distance.  I called
into their support and they claim that "everything is normal" and as a
related note, up until about 2 days ago, this PRI was working properly
calling long distance on our old Mitel system.

I'm a bit of a loss as to where to check next.  Any ideas on either what
might be pulling out the 1 when delivering it to the PRI or if there is
some secret extra code the PRI is expecting for long distance I'm not aware
of?

Thanks for the help!
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From steveayre at gmail.com  Tue Feb  6 19:27:29 2018
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 6 Feb 2018 19:27:29 +0000
Subject: [Freeswitch-users] Missing git tag for v1.6.20
Message-ID: <CAFiqYu=2QmDnzaZrcW=0=yqbkreF3KaOZL9xkoN3ebY7svTyGw@mail.gmail.com>

I've just noticed the release announcement for v1.6.20

I can see the tarball linked to from the confluence blog, but it looks like
that version hasn't been tagged in git checking both stash and my local
clone?

Can that be corrected for those of us that build from git please?
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From abaci64 at gmail.com  Tue Feb  6 19:34:48 2018
From: abaci64 at gmail.com (Abaci B)
Date: Tue, 6 Feb 2018 14:34:48 -0500
Subject: [Freeswitch-users] is this even possible?
In-Reply-To: <CAFGRPVqAp4KSGTHuR4ob9qcES68r12UoHpzReOtudaam_Z5yTg@mail.gmail.com>
References: <CAFGRPVqAp4KSGTHuR4ob9qcES68r12UoHpzReOtudaam_Z5yTg@mail.gmail.com>
Message-ID: <CAFmZFiMAEmJcx2z4k=fC5VZp=8a6nV1_U9Adq04t8NhWjDrTqQ@mail.gmail.com>

You can do a sip redirect.

On Tue, Feb 6, 2018 at 11:38 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello guys,
>
> So i have a conference running on a fs... and i want to move it to another
> fs... is this even possible??
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337 <+34%20669%2044%2083%2037>
> ᐧ
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 mike at jerris.com  Tue Feb  6 20:07:33 2018
From: mike at jerris.com (Michael Jerris)
Date: Tue, 6 Feb 2018 15:07:33 -0500
Subject: [Freeswitch-users] Missing git tag for v1.6.20
In-Reply-To: <CAFiqYu=2QmDnzaZrcW=0=yqbkreF3KaOZL9xkoN3ebY7svTyGw@mail.gmail.com>
References: <CAFiqYu=2QmDnzaZrcW=0=yqbkreF3KaOZL9xkoN3ebY7svTyGw@mail.gmail.com>
Message-ID: <458ECA10-3402-4CC5-A6F4-092438AA164F@jerris.com>

sorry about that, my bad.  Its pushed now.

> On Feb 6, 2018, at 2:27 PM, Steven Ayre <steveayre at gmail.com> wrote:
> 
> I've just noticed the release announcement for v1.6.20
> 
> I can see the tarball linked to from the confluence blog, but it looks like that version hasn't been tagged in git checking both stash and my local clone?
> 
> Can that be corrected for those of us that build from git please?



From steveayre at gmail.com  Tue Feb  6 20:20:28 2018
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 6 Feb 2018 20:20:28 +0000
Subject: [Freeswitch-users] Missing git tag for v1.6.20
In-Reply-To: <458ECA10-3402-4CC5-A6F4-092438AA164F@jerris.com>
References: <CAFiqYu=2QmDnzaZrcW=0=yqbkreF3KaOZL9xkoN3ebY7svTyGw@mail.gmail.com>
 <458ECA10-3402-4CC5-A6F4-092438AA164F@jerris.com>
Message-ID: <CAFiqYukUxaCdTNkPmKZk6u3_npOL7-S7Pk=L+VK4EbX+iW4_JQ@mail.gmail.com>

Thanks Mike!

On 6 February 2018 at 20:07, Michael Jerris <mike at jerris.com> wrote:

> sorry about that, my bad.  Its pushed now.
>
> > On Feb 6, 2018, at 2:27 PM, Steven Ayre <steveayre at gmail.com> wrote:
> >
> > I've just noticed the release announcement for v1.6.20
> >
> > I can see the tarball linked to from the confluence blog, but it looks
> like that version hasn't been tagged in git checking both stash and my
> local clone?
> >
> > Can that be corrected for those of us that build from git please?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 scott at tgifriday.com  Tue Feb  6 23:38:45 2018
From: scott at tgifriday.com (Scott Howell)
Date: Tue, 06 Feb 2018 23:38:45 +0000
Subject: [Freeswitch-users] Stream inbound Freeswitch (RTMP?)
In-Reply-To: <3af2c4a7-61b4-71ea-8e1a-51a86bef4a5f@airmail.cc>
References: <CAMALkh2pQ9u3ZWzWB+PPCMQh-p0mgMcRzXG4VJgiRBYm58ba3A@mail.gmail.com>
 <3af2c4a7-61b4-71ea-8e1a-51a86bef4a5f@airmail.cc>
Message-ID: <CAMALkh16eg=YBEcVpFfyLgM+0YOP2tJ9yxAy-9DBW_3b0UmQGQ@mail.gmail.com>

will mod_shout / mod_vlc work for youtube live streams?  We currently
stream to youtube live, so picking up that stream would be perfect, if
either of those work for that..

-Scott

On Tue, Feb 6, 2018 at 1:53 PM Ryan Harris <ryharris at airmail.cc> wrote:

> On 02/04/2018 02:37 PM, Scott Howell wrote:
>
> I have an audio/video stream that I already set up, I would like to use
> the audio portion as a "conference" that users can dial into and listen
> live.
>
>
> I'd go for mod_shout or mod_vlc
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 brian at freeswitch.com  Wed Feb  7 04:30:04 2018
From: brian at freeswitch.com (Brian West)
Date: Wed, 07 Feb 2018 04:30:04 +0000
Subject: [Freeswitch-users] Stream inbound Freeswitch (RTMP?)
In-Reply-To: <CAMALkh16eg=YBEcVpFfyLgM+0YOP2tJ9yxAy-9DBW_3b0UmQGQ@mail.gmail.com>
References: <CAMALkh2pQ9u3ZWzWB+PPCMQh-p0mgMcRzXG4VJgiRBYm58ba3A@mail.gmail.com>
 <3af2c4a7-61b4-71ea-8e1a-51a86bef4a5f@airmail.cc>
 <CAMALkh16eg=YBEcVpFfyLgM+0YOP2tJ9yxAy-9DBW_3b0UmQGQ@mail.gmail.com>
Message-ID: <CAEJMVkCyBwjcL94z+5TCutrva_wgAkJ2p4Z3LNy9QqM7V8NkwQ@mail.gmail.com>

Scott,

Ignore mod_vlc, I sent you a private email, we do this every week for
ClueCon weekly.  Reply and we can discuss.

/b

On Tue, Feb 6, 2018 at 5:41 PM Scott Howell <scott at tgifriday.com> wrote:

> will mod_shout / mod_vlc work for youtube live streams?  We currently
> stream to youtube live, so picking up that stream would be perfect, if
> either of those work for that..
>
> -Scott
>
> On Tue, Feb 6, 2018 at 1:53 PM Ryan Harris <ryharris at airmail.cc> wrote:
>
>> On 02/04/2018 02:37 PM, Scott Howell wrote:
>>
>> I have an audio/video stream that I already set up, I would like to use
>> the audio portion as a "conference" that users can dial into and listen
>> live.
>>
>>
>> I'd go for mod_shout or mod_vlc
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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

-- 

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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From raimundo.perez.cuba at gmail.com  Wed Feb  7 12:47:50 2018
From: raimundo.perez.cuba at gmail.com (=?utf-8?Q?Raimundo_P=C3=A9rez_Nieves?=)
Date: Wed, 7 Feb 2018 13:47:50 +0100
Subject: [Freeswitch-users] Outbound Socket isn't killing each pid
Message-ID: <F5FDDBB3-35B0-4128-8082-EA50E3203D78@gmail.com>

Hi guys, 
Iam running outbound socket in a server, and It start perfect and receive 265 calls, which mean 265 pid created by outbound socket.
The problem is in that exactly moment, the server get memory ram out and get freezes.
I can see each pid of outbound socket there:

Feb  6 21:14:57 test-server kernel: [97024.382843] [16701]     0 16701    10032      846      25        0             0 outbound_socket
Feb  6 21:14:57 test-server kernel: [97024.382845] [16704]     0 16704    26451     2463      27        0             0 outbound_socket
Feb  6 21:14:57 test-server kernel: [97024.382846] [16709]     0 16709    10094      898      23        0             0 outbound_socket
Feb  6 21:14:57 test-server kernel: [97024.382848] [16711]     0 16711    11686     2498      24        0             0 outbound_socket
Feb  6 21:14:57 test-server kernel: [97024.382849] [16715]     0 16715    10067      882      23        0             0 outbound_socket
Feb  6 21:14:57 test-server kernel: [97024.382851] [16720]     0 16720    11686     2498      24        0             0 outbound_socket
Feb  6 21:14:57 test-server kernel: [97024.382852] [16725]     0 16725    11694     2498      24        0             0 outbound_socket
Feb  6 21:14:57 test-server kernel: [97024.382854] [16730]     0 16730    11686     2498      24        0             0 outbound_socket
Feb  6 21:14:57 test-server kernel: [97024.382855] [16736]     0 16736    10094      898      23        0             0 outbound_socket
Feb  6 21:14:57 test-server kernel: [97024.382857] [16738]     0 16738    11688     2498      24        0             0 outbound_socket
Feb  6 21:14:57 test-server kernel: [97024.382858] [16743]     0 16743    11686     2486      24        0             0 outbound_socket
Feb  6 21:14:57 test-server kernel: [97024.382860] [16748]     0 16748    11686     2498      24        0             0 outbound_socket
Feb  6 21:14:57 test-server kernel: [97024.382861] [16754]     0 16754    11694     2488      24        0             0 outbound_socket
Feb  6 21:14:57 test-server kernel: [97024.382863] [16756]     0 16756    11694     2498      24        0             0 outbound_socket
Feb  6 21:14:57 test-server kernel: [97024.382864] [16761]     0 16761    11686     2498      24        0             0 outbound_socket
Feb  6 21:14:57 test-server kernel: [97024.382866] [16766]     0 16766    10085      898      23        0             0 outbound_socket
Feb  6 21:14:57 test-server kernel: [97024.382867] [16771]     0 16771    11688     2489      24        0             0 outbound_socket
Feb  6 21:14:57 test-server kernel: [97024.382868] [16775]     0 16775    11694     2498      24        0             0 outbound_socket
Feb  6 21:14:57 test-server kernel: [97024.382870] [16777]     0 16777    11686     2498      24        0             0 outbound_socket
And continues……

At the end:
Feb  6 21:14:57 test-server kernel: [97024.383243] Out of memory: Kill process 14314 (freeswitch) score 28 or sacrifice child
Feb  6 21:14:57 test-server kernel: [97024.384424] Killed process 14314 (freeswitch) total-vm:1282524kB, anon-rss:58856kB, file-rss:0kB


I use this code.

require ESL;
use IO::Socket::INET;

my $ip = "127.0.0.1";
my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '8083',  Proto => 'tcp',  Listen => 1,  Reuse => 1 );
die "Could not create socket: $!\n" unless $sock;
for(;;) {
  my $new_sock = $sock->accept();
  my $pid = fork();
  if ($pid) {
  print "New child pid $pid created...\n";
    close($new_sock);
    next;
  }
  my $fd = fileno($new_sock);
  my $con = new ESL::ESLconnection($fd);
  my $info = $con->getInfo();
  my $uuidLegB = $info->getHeader("unique-id");
  #This $uuidLegA is a temp variable where I store uuid leg A for post bridge
  my $uuidLegA = $info->getHeader("variable_outbound_caller_id_name");
  my $joinUUID = 'resp'.$uuidLegA.'|'.$uuidLegB.' XML default';
  print("UUID Leg A is $uuidLegA and B is $uuidLegB\n");
  $con->events("plain","all");
  $con->execute("spandsp_start_tone_detect","34");
  my $connectedSession = 1;
  
  while($con->connected() & $connectedSession == 1) {
    my $e = $con->recvEvent();
    my $ev_name = $e->getHeader("Event-Name");
    print("$ev_name\n");
    if($ev_name eq 'DETECTED_TONE'){
      my $tone_name = $e->getHeader("Detected-Tone");
      print "DETECTED_TONE [$tone_name]\n";  
      if ($tone_name eq 'RING_TONE') {
        $command = 'spandsp_stop_tone_detect '.$uuidLegB;
        $con->api($command);
        $con->execute("transfer",$joinUUID);
        $connectedSession = 0;
      } 
    }
  }
  print "BYE\n";
  close($new_sock);
}
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From alihaider.4189 at gmail.com  Wed Feb  7 13:42:58 2018
From: alihaider.4189 at gmail.com (Ali Haider)
Date: Wed, 7 Feb 2018 18:42:58 +0500
Subject: [Freeswitch-users] how to make a dial plane
Message-ID: <CAB2bS9Qt5jD6EZJ2iwmKRRxfD-7qgvWhC6udJJ1sZTSizkqmpA@mail.gmail.com>

Hiiii users
             I want to know how to make a dialplan for pc
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From alex at freeswitch.com  Wed Feb  7 13:49:48 2018
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Wed, 7 Feb 2018 22:49:48 +0900
Subject: [Freeswitch-users] how to make a dial plane
In-Reply-To: <CAB2bS9Qt5jD6EZJ2iwmKRRxfD-7qgvWhC6udJJ1sZTSizkqmpA@mail.gmail.com>
References: <CAB2bS9Qt5jD6EZJ2iwmKRRxfD-7qgvWhC6udJJ1sZTSizkqmpA@mail.gmail.com>
Message-ID: <CAGJLwFg9UH0fZXoFFowV=4d-h_F+mzbk8Z+7pgAt8kdnxSqvXw@mail.gmail.com>

https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan

On Wed, Feb 7, 2018 at 10:42 PM, Ali Haider <alihaider.4189 at gmail.com>
wrote:

> Hiiii users
>              I want to know how to make a dialplan for pc
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

Alex Sibyakin | Support Engineer
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: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.
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From gmaruzz at gmail.com  Wed Feb  7 13:50:22 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 7 Feb 2018 14:50:22 +0100
Subject: [Freeswitch-users] how to make a dial plane
In-Reply-To: <CAB2bS9Qt5jD6EZJ2iwmKRRxfD-7qgvWhC6udJJ1sZTSizkqmpA@mail.gmail.com>
References: <CAB2bS9Qt5jD6EZJ2iwmKRRxfD-7qgvWhC6udJJ1sZTSizkqmpA@mail.gmail.com>
Message-ID: <CALXCt0qNT50G6h4AygMBifK=aPEYOuMhAFyQOPB9rUe4xPnZFw@mail.gmail.com>

Ali,

why, why, why, you send a mail like this?

Study the documentation, use google, and when you have a real question
write to the mailing list.

Is not nice you push us to ignore you, or to answer you something like
"your questions is too much general, blah blah blah".

-giovanni


On 7 February 2018 at 14:42, Ali Haider <alihaider.4189 at gmail.com> wrote:

> Hiiii users
>              I want to know how to make a dialplan for pc
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From infos at madovsky.org  Wed Feb  7 13:58:39 2018
From: infos at madovsky.org (Madovsky)
Date: Wed, 7 Feb 2018 05:58:39 -0800
Subject: [Freeswitch-users] how to make a dial plane
In-Reply-To: <CALXCt0qNT50G6h4AygMBifK=aPEYOuMhAFyQOPB9rUe4xPnZFw@mail.gmail.com>
References: <CAB2bS9Qt5jD6EZJ2iwmKRRxfD-7qgvWhC6udJJ1sZTSizkqmpA@mail.gmail.com>
 <CALXCt0qNT50G6h4AygMBifK=aPEYOuMhAFyQOPB9rUe4xPnZFw@mail.gmail.com>
Message-ID: <4c43e321-d958-54ec-4dc4-59507bfc6e1a@madovsky.org>

This emailist is to talk and support any problem of installing, updating 
and so on.

to learn how to use FS just read the documentation but please don't 
bother the whole emailist subscribers with

your unconsistent email thanks.


On 2/7/2018 5:50 AM, Giovanni Maruzzelli wrote:
> Ali,
>
> why, why, why, you send a mail like this?
>
> Study the documentation, use google, and when you have a real question 
> write to the mailing list.
>
> Is not nice you push us to ignore you, or to answer you something like 
> "your questions is too much general, blah blah blah".
>
> -giovanni
>
>
> On 7 February 2018 at 14:42, Ali Haider <alihaider.4189 at gmail.com 
> <mailto:alihaider.4189 at gmail.com>> wrote:
>
>     Hiiii users
>                  I want to know how to make a dialplan for pc
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com>
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org <http://confluence.freeswitch.org>
>     http://www.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>
>     http://www.freeswitch.org
>
>
>
>
> -- 
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 rmundkowsky at ets.org  Tue Feb  6 21:23:47 2018
From: rmundkowsky at ets.org (Mundkowsky, Robert)
Date: Tue, 6 Feb 2018 21:23:47 +0000
Subject: [Freeswitch-users] Freeswitch support for text messages/commands
 between Verto/webRTC to/from conferenced backend SIP/RTP systems
In-Reply-To: <CY1PR0701MB122514D0CB73497C803D2FCBC3FD0@CY1PR0701MB1225.namprd07.prod.outlook.com>
References: <CY1PR07MB237750C64DF5CDAAFECCF266BEFD0@CY1PR07MB2377.namprd07.prod.outlook.com>
 <CY1PR0701MB122514D0CB73497C803D2FCBC3FD0@CY1PR0701MB1225.namprd07.prod.outlook.com>
Message-ID: <CY1PR07MB2377514FC809253B46BE6259BEFD0@CY1PR07MB2377.namprd07.prod.outlook.com>


We are wondering if FreeSWITCH supports sending and receiving text messages between Verto/webRTC to/from backend SIP/RTP systems that conferenced together?


1)      We want to use this to send commands from the web browser to the backend systems, for example:

a.       "hey backend system, please turn on the ASR to start listening."


2)      We also wonder if FreeSWITCH supports sending commands from the backend to the frontend (web browser), for example:

a.       "hey avatar, here are some visemes to tell you how you should move your lips, arms, ... at the same time as playing this audio"

If both Verto/SIP or webRTC/RTP are supported, is either more ideal?

For instance, if the text/commands are over RTC then maybe we can use one clock for all RTP streams and/or RTP sessions and use the RTP timestamps to synchronize the commands with audio/video.  Or maybe Verto/SIP is better for commands.


I see there are various RFCs for text over RTP and text over RTCDataChannel in webRTC, but not sure if FreeSWITCH supports any of these.

I also see FreeSWITCH mod_verto  support text messages, but not sure if FreeSWITCH sends these to/from SIP (e.g. maybe via SIMPLE SIP) or via RTCDataChannel?


Robert



________________________________

This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited.


Thank you for your compliance.

________________________________
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From bilaln018 at gmail.com  Thu Feb  8 06:11:09 2018
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Thu, 8 Feb 2018 11:11:09 +0500
Subject: [Freeswitch-users] how to make a dial plane
In-Reply-To: <4c43e321-d958-54ec-4dc4-59507bfc6e1a@madovsky.org>
References: <CAB2bS9Qt5jD6EZJ2iwmKRRxfD-7qgvWhC6udJJ1sZTSizkqmpA@mail.gmail.com>
 <CALXCt0qNT50G6h4AygMBifK=aPEYOuMhAFyQOPB9rUe4xPnZFw@mail.gmail.com>
 <4c43e321-d958-54ec-4dc4-59507bfc6e1a@madovsky.org>
Message-ID: <CADSVerC2+gsdzYnTVGd-EuaMtvQNjO-kN5pCG=DdwDiq2srJ3Q@mail.gmail.com>

@giovanni,
Yeah, you know what i did? i took his personal email and i spend few hours
guiding him through the installation, just to make user-list clean.
Ali please read the documentations and come up with real questions.

Regards
Abbasi

On Wed, Feb 7, 2018 at 6:58 PM, Madovsky <infos at madovsky.org> wrote:

> This emailist is to talk and support any problem of installing, updating
> and so on.
>
> to learn how to use FS just read the documentation but please don't bother
> the whole emailist subscribers with
>
> your unconsistent email thanks.
>
>
> On 2/7/2018 5:50 AM, Giovanni Maruzzelli wrote:
>
> Ali,
>
> why, why, why, you send a mail like this?
>
> Study the documentation, use google, and when you have a real question
> write to the mailing list.
>
> Is not nice you push us to ignore you, or to answer you something like
> "your questions is too much general, blah blah blah".
>
> -giovanni
>
>
> On 7 February 2018 at 14:42, Ali Haider <alihaider.4189 at gmail.com> wrote:
>
>> Hiiii users
>>              I want to know how to make a dialplan for pc
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 michael at mailworks.org  Thu Feb  8 09:33:56 2018
From: michael at mailworks.org (Michael Avers)
Date: Thu, 08 Feb 2018 02:33:56 -0700
Subject: [Freeswitch-users] T38 re-invite, 488 Not Acceptable Here,
	Freeswitch 1.6.20
Message-ID: <1518082436.566731.1263750016.385885A8@webmail.messagingengine.com>

Hello,

I understand the T38 behavior changed after 1.6.13. I watched Brian's
video from last year, read the docs, but I just don't see what else I
would need for a very simple scenario for receiving a fax with the
receiving end (ATA) re-inviting T38.
PSTN > FS > HT801 ATA > T38 RE-INVITE > FS > PSTN

The ATA does re-invite, however FS rejects immediately with 488 Not
Acceptable Here. I'm using the following dialplan. I also tried to use
export instead of set, but same result.
<extension name="faxdid1" continue="false">
 <condition field="destination_number" expression="^(4154447777)$">
   <action application="set" data="ignore_early_media=true"/>
   <action application="set" data="fax_enable_t38_request=true"/>
   <action application="set" data="fax_enable_t38=true"/>
   <action application="bridge" data="user/6050 at sip.domain.org"/>
 </condition>
</extension>

Is there anything obvious that I'm missing here?

Thanks
Mike
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From paranoya at paranoya.org  Thu Feb  8 15:02:57 2018
From: paranoya at paranoya.org (Jakub Wojewoda)
Date: Thu, 8 Feb 2018 16:02:57 +0100
Subject: [Freeswitch-users] freeswitch odbc in the core and RDS
In-Reply-To: <CAFGRPVrsHCoUSGfqietuJjm3G_PZRuEUSt_pX1PrRcwnNuRVMg@mail.gmail.com>
References: <CAFGRPVrsHCoUSGfqietuJjm3G_PZRuEUSt_pX1PrRcwnNuRVMg@mail.gmail.com>
Message-ID: <CAMy7UaS7jy6+bk52+EOdo4=w7FsGvihrsedsxGEiLNk2ATRkNA@mail.gmail.com>

Try changing collation of columns to latin1.

Regards

2018-02-05 20:30 GMT+01:00 David Villasmil <david.villasmil.work at gmail.com>:

> Hello guys,
>
> trying to use RDS as a backend for freeswitch core, but when creating the
> tables, RDS comes back with:
>
> [STATE: HY000 CODE 1118 ERROR: [MySQL][ODBC 5.1
> Driver][mysqld-5.6.37-log]Row size too large. The maximum row size for the
> used table type, not counting BLOBs, is 65535. This includes storage
> overhead, check the manual. You have to change some columns to TEXT or BLOBs
>
> for some tables, thus messing up everything
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337 <+34%20669%2044%2083%2037>
> ᐧ
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 michael at mailworks.org  Thu Feb  8 17:08:00 2018
From: michael at mailworks.org (Michael Avers)
Date: Thu, 08 Feb 2018 10:08:00 -0700
Subject: [Freeswitch-users] T38 re-invite, 488 Not Acceptable Here,
	Freeswitch 1.6.20
Message-ID: <1518109680.757010.1264255480.51BA90A7@webmail.messagingengine.com>

Hello,

I understand the T38 behavior changed after 1.6.13. I watched Brian's
video from last year, read the docs, but I just don't see what else I
would need for a very simple scenario for receiving a fax with the
receiving end (ATA) re-inviting T38.
PSTN > FS > HT801 ATA > T38 RE-INVITE > FS > PSTN

The ATA does re-invite, however FS rejects immediately with 488 Not
Acceptable Here. I'm using the following dialplan. I also tried to use
export instead of set, but same result.
<extension name="faxdid1" continue="false">
 <condition field="destination_number" expression="^(4154447777)$">
   <action application="set" data="ignore_early_media=true"/>
   <action application="set" data="fax_enable_t38_request=true"/>
   <action application="set" data="fax_enable_t38=true"/>
   <action application="bridge" data="user/6050 at sip.domain.org"/>
 </condition>
</extension>

Is there anything obvious that I'm missing here?

FreeSWITCH (Version 1.6.20 -37-987c9b9 64bit)



Thanks
Mike
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From bilaln018 at gmail.com  Thu Feb  8 17:43:47 2018
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Thu, 8 Feb 2018 22:43:47 +0500
Subject: [Freeswitch-users] [rtp-timeout-sec][need experts comments]
Message-ID: <CADSVerCfDk74ygoYxmnjJqYJzTtyvW1y7dLvWe8hE36At44DEQ@mail.gmail.com>

Hi Users,

I did checked the default value of rtp timeout is 300 seconds, for me its
very long, i just wanted to know why it is placed that long, for me if a
call is on silent for 30-40 seconds i will hang up that. any down side for
doing this?

Regards
Abbasi
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From asilva at wirelessmundi.com  Fri Feb  9 00:08:03 2018
From: asilva at wirelessmundi.com (antonio)
Date: Fri, 9 Feb 2018 01:08:03 +0100
Subject: [Freeswitch-users] chatplan: control sip messages
Message-ID: <c4b6d2d0-7df8-f877-7304-e89fc25b8e26@wirelessmundi.com>

Hi,

Is it possible to block sip messages between sip clients?

I try to set the action stop in chatplan but it doesn't do anything, the 
message is sent between the two endpoints.

What i want is to be able to control witch accounts can send messages 
and limited destinations, i also want to able receive the messages in a 
common direction, like, chat at domain.local, without having the error the 
warning message "sofia_presence.c:225 Can't find registered user"..


Thanks.


-- 
Saludos / Regards / Cumprimentos
Anónio Silva



From asilva at wirelessmundi.com  Fri Feb  9 11:25:04 2018
From: asilva at wirelessmundi.com (antonio)
Date: Fri, 9 Feb 2018 12:25:04 +0100
Subject: [Freeswitch-users] ping
Message-ID: <ab2ae7ee-2e8b-e04e-cc7c-abb4f2f41ea5@wirelessmundi.com>

hi,

just testing... i reply to some mails and i don't see my answer. Also 
put some question and it seams is not arriving..

-- 
Saludos / Regards / Cumprimentos
Anónio Silva



From mbodbg at gmx.net  Fri Feb  9 11:41:47 2018
From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=)
Date: Fri, 9 Feb 2018 12:41:47 +0100
Subject: [Freeswitch-users] WSS - DTLS handshake takes about 6s in firefox
Message-ID: <9A71A4FA-CA49-40B3-87E6-367ECDC5B3AE@gmx.net>


We're using an application which uses the JSSip library to connect to Freeswitch 1.6.19 to make and receive calls. On Chome, everything is fine - when a call get's connected we can hear the caller from the beginning. However on Firefox we can not hear the caller for the first 5 Seconds. 

If I look in the logs, it seems the DTLS handshake takes more than 5 seconds between Firefox and Freeswitch:

Firefox 57.0.4 on OSX 10.13.3

2018-02-09 11:38:05.613120 [INFO] switch_rtp.c:3581 Activate RTP/RTCP audio DTLS server
2018-02-09 11:38:05.613120 [INFO] switch_rtp.c:3730 Changing audio DTLS state from OFF to HANDSHAKE
2018-02-09 11:38:11.833115 [INFO] switch_rtp.c:3192 Changing audio DTLS state from HANDSHAKE to SETUP <---
2018-02-09 11:38:11.853115 [INFO] switch_rtp.c:3141 Changing audio DTLS state from SETUP to READY

Chrome 62.0.3202.89 on OSX 10.13.3

2018-02-09 11:45:30.233115 [INFO] switch_rtp.c:3581 Activate RTP/RTCP audio DTLS server
2018-02-09 11:45:30.233115 [INFO] switch_rtp.c:3730 Changing audio DTLS state from OFF to HANDSHAKE
2018-02-09 11:45:30.533125 [INFO] switch_rtp.c:3192 Changing audio DTLS state from HANDSHAKE to SETUP
2018-02-09 11:45:30.553117 [INFO] switch_rtp.c:3141 Changing audio DTLS state from SETUP to READY


Is this a general issue in firefox or can it somehow be optimized by changing the freeswitch configuration?

Thanks and regards

Markus

From mario_fs at mgtech.com  Fri Feb  9 17:40:00 2018
From: mario_fs at mgtech.com (Mario)
Date: Fri, 9 Feb 2018 09:40:00 -0800
Subject: [Freeswitch-users] Test email no messages since 2/6
Message-ID: <AD080D20-3995-4DA2-AF9E-F715075A6214@mgtech.com>

Have not gotten FS mail since 2/6 this is a test.


From tculjaga at gmail.com  Fri Feb  9 21:32:31 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 9 Feb 2018 22:32:31 +0100
Subject: [Freeswitch-users] "received=" in path header
Message-ID: <CAJoKKdXk-Y_p8_3ADLu89FqYX52garpZPaqOa3vFfDWbhtnNww@mail.gmail.com>

hi,


i have  UA > Kamailio > Freeswitch

UA is using sip for websockets and registers via kamailio to freeswitch.
When i try to originate a call from FS to the registered endpoint, the call
fails.

freeswitch at FS01> bgapi originate
{origination_caller_id_number=1002}sofia/internal/1001%mydomain &echo()

I think FS is trying to use transport from received parematar instead of
path uri.

any advice how to handle this ?





recv 964 bytes from udp/[192.168.50.60]:5060 at 22:18:57.909700:
   ------------------------------------------------------------------------
   REGISTER sip:192.168.50.60 SIP/2.0
   Via: SIP/2.0/UDP
192.168.50.60;branch=z9hG4bKe929.b6aa43dd1eefa9e4756af7d31d65066e.0
   Via: SIP/2.0/WSS
192.0.2.110;rport=61744;received=192.168.200.77;branch=z9hG4bK3179897
   Max-Forwards: 69
   To: <sip:1001 at 192.168.50.60>
   From: <sip:1001 at 192.168.50.60>;tag=vq30modpgd
   Call-ID: dlnrna9o4ngn25fb9d1vi2
   CSeq: 192 REGISTER
   Authorization: Digest algorithm=MD5, username="1001",
realm="192.168.50.60", nonce="22c52b4b-f795-4caa-bb9b-15bbd87564f7",
uri="sip:192.168.50.60", response="a073277bc9c1fcd681de661cb418d838",
qop=auth, cnonce="euqeokf2d1gr", nc=00000001
   Contact: <sip:1001 at 192.0.2.110
;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:587f4694-7e55-46dc-a858-579a82556a20>";expires=600
   Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
   Supported: path, gruu, 100rel, outbound
   User-Agent: SIP.js/0.7.0 BB
   Content-Length: 0
   Path: <sip:192.168.50.60;lr;received=sip:192.168.200.77:61744
%3Btransport%3Dws>

   ------------------------------------------------------------------------
send 738 bytes to udp/[192.168.50.60]:5060 at 22:18:57.950968:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
192.168.50.60;branch=z9hG4bKe929.b6aa43dd1eefa9e4756af7d31d65066e.0
   Via: SIP/2.0/WSS
192.0.2.110;rport=61744;received=192.168.200.77;branch=z9hG4bK3179897
   From: <sip:1001 at 192.168.50.60>;tag=vq30modpgd
   To: <sip:1001 at 192.168.50.60>;tag=eK8SmZmcU7UBr
   Call-ID: dlnrna9o4ngn25fb9d1vi2
   CSeq: 192 REGISTER
   Contact: <sip:1001 at 192.0.2.110;transport=ws>;expires=600
   Date: Fri, 09 Feb 2018 21:18:57 GMT
   User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Path: <sip:192.168.50.60>;lr;received=sip:192.168.200.77:61744
%3Btransport%3Dws
   Content-Length: 0

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



in freeswitch i the registration:

Call-ID:        dlnrna9o4ngn25fb9d1vi2
User:           1001 at mydomain
Contact:        "" <sip:1001 at 192.0.2.110
;transport=ws;fs_nat=yes;fs_path=sip%3A192.168.50.60%3Blr%3Breceived%3Dsip%3A192.168.200.77%3A61744%3Btransport%3Dws>
Agent:          SIP.js/0.7.0 BB
Status:         Registered(WS-NAT)(unknown) EXP(2018-02-09 22:29:57)
EXPSECS(510)
Ping-Status:    Reachable
Ping-Time:      0.00
Host:           FS01
IP:             192.168.50.60
Port:           5060
Auth-User:      1001
Auth-Realm:     192.168.50.60
MWI-Account:    1001 at mydomain




the register contain a path header with received=<original uri>.

when i try to originate a call to this registered user the call goes
nowhere :=)


freeswitch at FS01> bgapi originate
{origination_caller_id_number=1002}sofia/internal/1001%mydomain &echo()
+OK Job-UUID: 200e2197-a6de-42d9-b4bb-63917223bc53

2018-02-09 22:25:40.500100 [DEBUG] switch_ivr_originate.c:2142 Parsing
global variables
2018-02-09 22:25:40.500100 [NOTICE] switch_channel.c:1104 New Channel
sofia/internal/1001 [327826f9-8a8c-46c7-b3ed-8bf22852df88]
2018-02-09 22:25:40.500100 [DEBUG] mod_sofia.c:4819 (sofia/internal/1001)
State Change CS_NEW -> CS_INIT
2018-02-09 22:25:40.500100 [DEBUG] switch_core_state_machine.c:584
(sofia/internal/1001) Running State Change CS_INIT (Cur 1 Tot 217)
2018-02-09 22:25:40.500100 [DEBUG] switch_core_state_machine.c:627
(sofia/internal/1001) State INIT
2018-02-09 22:25:40.500100 [DEBUG] mod_sofia.c:90 sofia/internal/1001 SOFIA
INIT
2018-02-09 22:25:40.500100 [DEBUG] sofia_glue.c:1264
sip:192.168.50.60;lr;received=sip:192.168.200.77:61744;transport=ws Setting
proxy route to sofia/internal/1001
2018-02-09 22:25:40.500100 [DEBUG] sofia_glue.c:1295 sofia/internal/1001
sending invite version: 1.6.19 git b1b21d0 2017-11-20 16:34:16Z 64bit
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From gmina at connectfirst.com  Sat Feb 10 21:43:40 2018
From: gmina at connectfirst.com (Geoff Mina)
Date: Sat, 10 Feb 2018 14:43:40 -0700
Subject: [Freeswitch-users] Locked, Waiting on external entities
Message-ID: <CAKzROs2ZKLu=xFRPBNn8iyeLrnp51iTHBMJTSzL3WBzHGnhdoQ@mail.gmail.com>

Greetings,
I was wondering if anyone knows what the "Locked, Waiting on external
entities" message indicates.

First, I am using a FusionPBX GUI to play with some basic PBX
functionality.  The flow of my call is an XML transfer from a public to a
private context.  Enabling trusted outside party to dial extensions
directly on the Freeswitch host.

The call comes in on the external SIP profile and transfers to the internal
profile.  The softphone rings and I hit "decline" which sends back a 486
Busy Here to FS.  Freeswitch then hangs in a "Waiting on external entities"
state until a timeout occurs at which point the inbound call is finally
sent to voicemail after an internal 408 timeout.

This is FS 1.6.20 installed from Yum on a CentOS7 box.  Attached is the
full log file with SIP tracing enabled.  Here is the relevant snippet where
the switch is just sitting for 16 seconds doing nothing.

2018-02-10 21:27:11.132820 [DEBUG] switch_core_state_machine.c:610
(sofia/internal/113 at 161.97.193.191:51392) State Change CS_REPORTING ->
CS_DESTROY
*2018-02-10 21:27:11.132820 [DEBUG] switch_core_session.c:1665 Session 87
(sofia/internal/113 at 161.97.193.191:51392 <http://113 at 161.97.193.191:51392>)
Locked, Waiting on external entities*
*2018-02-10 21:27:40.952819 [DEBUG] sofia.c:7084 Channel
sofia/internal/113 at 161.97.193.191:51094 <http://113 at 161.97.193.191:51094>
entering state [terminated][408]*
2018-02-10 21:27:40.952819 [NOTICE] sofia.c:8273 Hangup sofia/internal/
113 at 161.97.193.191:51094 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE]

Just a point in the right direction as to what may cause this would be
appreciated.

Thanks,
Geoff
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recv 1102 bytes from udp/[174.37.181.39]:5060 at 21:27:08.881601:
   ------------------------------------------------------------------------
   INVITE sip:113 at c00-pbx.somedomain.com SIP/2.0
   Record-Route: <sip:174.37.181.39;lr;ftag=as54b5500e>
   Via: SIP/2.0/UDP 174.37.181.39;branch=z9hG4bK1375.9bd5a993470896a1e4e38b2bff4f084b.0
   Via: SIP/2.0/UDP 52.11.183.123:5060;branch=z9hG4bK61968ea6;rport=5060
   From: "5555551212" <sip:5555551212 at somedomain.com>;tag=as54b5500e
   To: <sip:113 at 174.37.181.39:5060>
   Contact: <sip:5555551212 at 52.11.183.123>
   Call-ID: 4f43cc193246d9f07d56ccf80ecb9d09 at somedomain.com
   CSeq: 102 INVITE
   User-Agent: G-Tel v1.0
   Max-Forwards: 69
   Remote-Party-ID: "5555551212" <sip:5555551212 at somedomain.com>;privacy=off;screen=no
   Date: Sat, 10 Feb 2018 21:27:08 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
   Supported: replaces
   P-Account-ID: 99990009
   Content-Type: application/sdp
   Content-Length: 289

   v=0
   o=root 27044 27044 IN IP4 52.11.183.123
   s=session
   c=IN IP4 52.11.183.123
   t=0 0
   m=audio 10466 RTP/AVP 18 0 101
   a=rtpmap:18 G729/8000
   a=fmtp:18 annexb=no
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
send 457 bytes to udp/[174.37.181.39]:5060 at 21:27:08.881851:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 174.37.181.39;branch=z9hG4bK1375.9bd5a993470896a1e4e38b2bff4f084b.0
   Via: SIP/2.0/UDP 52.11.183.123:5060;branch=z9hG4bK61968ea6;rport=5060
   Record-Route: <sip:174.37.181.39;lr;ftag=as54b5500e>
   From: "5555551212" <sip:5555551212 at somedomain.com>;tag=as54b5500e
   To: <sip:113 at 174.37.181.39:5060>
   Call-ID: 4f43cc193246d9f07d56ccf80ecb9d09 at somedomain.com
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH
   Content-Length: 0

   ------------------------------------------------------------------------
2018-02-10 21:27:08.872820 [NOTICE] switch_channel.c:1104 New Channel sofia/external/5555551212 at somedomain.com [2640726c-0ea9-11e8-ad64-53f0f54925c8]
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:584 (sofia/external/5555551212 at somedomain.com) Running State Change CS_NEW (Cur 1 Tot 84)
2018-02-10 21:27:08.872820 [DEBUG] sofia.c:9873 sofia/external/5555551212 at somedomain.com receiving invite from 174.37.181.39:5060 version: 1.6.20  64bit
2018-02-10 21:27:08.872820 [DEBUG] sofia.c:7084 Channel sofia/external/5555551212 at somedomain.com entering state [received][100]
2018-02-10 21:27:08.872820 [DEBUG] sofia.c:7094 Remote SDP:
v=0
o=root 27044 27044 IN IP4 52.11.183.123
s=session
c=IN IP4 52.11.183.123
t=0 0
m=audio 10466 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

2018-02-10 21:27:08.872820 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G729:18:8000:20:8000:1]/[G7221:115:32000:20:48000:1]
2018-02-10 21:27:08.872820 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G729:18:8000:20:8000:1]/[G7221:107:16000:20:32000:1]
2018-02-10 21:27:08.872820 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G729:18:8000:20:8000:1]/[G722:9:8000:20:64000:1]
2018-02-10 21:27:08.872820 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1]
2018-02-10 21:27:08.872820 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1]
2018-02-10 21:27:08.872820 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G729:18:8000:20:8000:1]/[GSM:3:8000:20:13200:1]
2018-02-10 21:27:08.872820 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G7221:115:32000:20:48000:1]
2018-02-10 21:27:08.872820 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G7221:107:16000:20:32000:1]
2018-02-10 21:27:08.872820 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2018-02-10 21:27:08.872820 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2018-02-10 21:27:08.872820 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2018-02-10 21:27:08.872820 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2018-02-10 21:27:08.872820 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2018-02-10 21:27:08.872820 [DEBUG] switch_core_media.c:4365 Set telephone-event payload to 101 at 8000
2018-02-10 21:27:08.872820 [DEBUG] switch_core_media.c:3061 Set Codec sofia/external/5555551212 at somedomain.com PCMU/8000 20 ms 160 samples 64000 bits 1 channels
2018-02-10 21:27:08.872820 [DEBUG] switch_core_codec.c:111 sofia/external/5555551212 at somedomain.com Original read codec set to PCMU:0
2018-02-10 21:27:08.872820 [DEBUG] switch_core_media.c:4708 Set telephone-event payload to 101 at 8000
2018-02-10 21:27:08.872820 [DEBUG] switch_core_media.c:4767 sofia/external/5555551212 at somedomain.com Set 2833 dtmf send payload to 101 recv payload to 101
2018-02-10 21:27:08.872820 [DEBUG] sofia.c:7507 (sofia/external/5555551212 at somedomain.com) State Change CS_NEW -> CS_INIT
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:603 (sofia/external/5555551212 at somedomain.com) State NEW
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:584 (sofia/external/5555551212 at somedomain.com) Running State Change CS_INIT (Cur 1 Tot 84)
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:627 (sofia/external/5555551212 at somedomain.com) State INIT
2018-02-10 21:27:08.872820 [DEBUG] mod_sofia.c:90 sofia/external/5555551212 at somedomain.com SOFIA INIT
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:40 sofia/external/5555551212 at somedomain.com Standard INIT
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:48 (sofia/external/5555551212 at somedomain.com) State Change CS_INIT -> CS_ROUTING
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:627 (sofia/external/5555551212 at somedomain.com) State INIT going to sleep
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:584 (sofia/external/5555551212 at somedomain.com) Running State Change CS_ROUTING (Cur 1 Tot 84)
2018-02-10 21:27:08.872820 [DEBUG] switch_channel.c:2249 (sofia/external/5555551212 at somedomain.com) Callstate Change DOWN -> RINGING
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:643 (sofia/external/5555551212 at somedomain.com) State ROUTING
2018-02-10 21:27:08.872820 [DEBUG] mod_sofia.c:143 sofia/external/5555551212 at somedomain.com SOFIA ROUTING
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:236 sofia/external/5555551212 at somedomain.com Standard ROUTING
2018-02-10 21:27:08.872820 [INFO] mod_dialplan_xml.c:637 Processing 5555551212 <5555551212>->113 in context public
Dialplan: sofia/external/5555551212 at somedomain.com parsing [public->1000] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [1000] destination_number(113) =~ /^(1000)$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [public->112] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [112] destination_number(113) =~ /^(112)$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [public->113] continue=true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [113] destination_number(113) =~ /^(113)$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com Action set(call_direction=inbound) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(call_direction=inbound)
2018-02-10 21:27:08.872820 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [call_direction]=[inbound]
Dialplan: sofia/external/5555551212 at somedomain.com Action set(domain_uuid=1f2f7a81-1b0f-4d02-87e4-a316e8187e81) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(domain_uuid=1f2f7a81-1b0f-4d02-87e4-a316e8187e81)
2018-02-10 21:27:08.872820 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [domain_uuid]=[1f2f7a81-1b0f-4d02-87e4-a316e8187e81]
Dialplan: sofia/external/5555551212 at somedomain.com Action set(domain_name=c00-pbx.somedomain.com) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(domain_name=c00-pbx.somedomain.com)
2018-02-10 21:27:08.872820 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [domain_name]=[c00-pbx.somedomain.com]
Dialplan: sofia/external/5555551212 at somedomain.com Action transfer(113 XML c00-pbx.somedomain.com)
Dialplan: sofia/external/5555551212 at somedomain.com parsing [public->114] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [114] destination_number(113) =~ /^(114)$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [public->115] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [115] destination_number(113) =~ /^(115)$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [public->116] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [116] destination_number(113) =~ /^(116)$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [public->118] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [118] destination_number(113) =~ /^(118)$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [public->119] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [119] destination_number(113) =~ /^(119)$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [public->2000] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [2000] destination_number(113) =~ /^(2000)$/ break=on-false
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:286 (sofia/external/5555551212 at somedomain.com) State Change CS_ROUTING -> CS_EXECUTE
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:643 (sofia/external/5555551212 at somedomain.com) State ROUTING going to sleep
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:584 (sofia/external/5555551212 at somedomain.com) Running State Change CS_EXECUTE (Cur 1 Tot 84)
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:650 (sofia/external/5555551212 at somedomain.com) State EXECUTE
2018-02-10 21:27:08.872820 [DEBUG] mod_sofia.c:198 sofia/external/5555551212 at somedomain.com SOFIA EXECUTE
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:328 sofia/external/5555551212 at somedomain.com Standard EXECUTE
EXECUTE sofia/external/5555551212 at somedomain.com transfer(113 XML c00-pbx.somedomain.com)
2018-02-10 21:27:08.872820 [DEBUG] switch_ivr.c:2165 (sofia/external/5555551212 at somedomain.com) State Change CS_EXECUTE -> CS_ROUTING
2018-02-10 21:27:08.872820 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/5555551212 at somedomain.com to XML[113 at c00-pbx.somedomain.com]
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:650 (sofia/external/5555551212 at somedomain.com) State EXECUTE going to sleep
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:584 (sofia/external/5555551212 at somedomain.com) Running State Change CS_ROUTING (Cur 1 Tot 84)
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:643 (sofia/external/5555551212 at somedomain.com) State ROUTING
2018-02-10 21:27:08.872820 [DEBUG] mod_sofia.c:143 sofia/external/5555551212 at somedomain.com SOFIA ROUTING
2018-02-10 21:27:08.872820 [DEBUG] switch_core_state_machine.c:236 sofia/external/5555551212 at somedomain.com Standard ROUTING
2018-02-10 21:27:08.872820 [INFO] mod_dialplan_xml.c:637 Processing 5555551212 <5555551212>->113 in context c00-pbx.somedomain.com
2018-02-10 21:27:08.892884 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f6e24050320 Connected.
2018-02-10 21:27:08.892884 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f6e24050320 released.
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->user_exists] continue=true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [user_exists] () =~ // break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com Action set(user_exists=${user_exists id ${destination_number} ${domain_name}}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(user_exists=true)
2018-02-10 21:27:08.892884 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [user_exists]=[true]
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [user_exists] ${user_exists}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com Action set(extension_uuid=${user_data ${destination_number}@${domain_name} var extension_uuid}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(extension_uuid=667ba7bb-fdd8-4a4c-9ac6-11643eaac636)
2018-02-10 21:27:08.892884 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [extension_uuid]=[667ba7bb-fdd8-4a4c-9ac6-11643eaac636]
Dialplan: sofia/external/5555551212 at somedomain.com Action set(hold_music=${user_data ${destination_number}@${domain_name} var hold_music}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(hold_music=local_stream://default)
2018-02-10 21:27:08.892884 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [hold_music]=[local_stream://default]
Dialplan: sofia/external/5555551212 at somedomain.com Action set(forward_all_enabled=${user_data ${destination_number}@${domain_name} var forward_all_enabled}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(forward_all_enabled=)
2018-02-10 21:27:08.912879 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [forward_all_enabled]=[UNDEF]
Dialplan: sofia/external/5555551212 at somedomain.com Action set(forward_all_destination=${user_data ${destination_number}@${domain_name} var forward_all_destination}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(forward_all_destination=)
2018-02-10 21:27:08.912879 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [forward_all_destination]=[UNDEF]
Dialplan: sofia/external/5555551212 at somedomain.com Action set(forward_busy_enabled=${user_data ${destination_number}@${domain_name} var forward_busy_enabled}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(forward_busy_enabled=)
2018-02-10 21:27:08.912879 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [forward_busy_enabled]=[UNDEF]
Dialplan: sofia/external/5555551212 at somedomain.com Action set(forward_busy_destination=${user_data ${destination_number}@${domain_name} var forward_busy_destination}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(forward_busy_destination=)
2018-02-10 21:27:08.912879 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [forward_busy_destination]=[UNDEF]
Dialplan: sofia/external/5555551212 at somedomain.com Action set(forward_no_answer_enabled=${user_data ${destination_number}@${domain_name} var forward_no_answer_enabled}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(forward_no_answer_enabled=)
2018-02-10 21:27:08.912879 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [forward_no_answer_enabled]=[UNDEF]
Dialplan: sofia/external/5555551212 at somedomain.com Action set(forward_no_answer_destination=${user_data ${destination_number}@${domain_name} var forward_no_answer_destination}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(forward_no_answer_destination=)
2018-02-10 21:27:08.912879 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [forward_no_answer_destination]=[UNDEF]
Dialplan: sofia/external/5555551212 at somedomain.com Action set(forward_user_not_registered_enabled=${user_data ${destination_number}@${domain_name} var forward_user_not_registered_enabled}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(forward_user_not_registered_enabled=)
2018-02-10 21:27:08.912879 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [forward_user_not_registered_enabled]=[UNDEF]
Dialplan: sofia/external/5555551212 at somedomain.com Action set(forward_user_not_registered_destination=${user_data ${destination_number}@${domain_name} var forward_user_not_registered_destination}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(forward_user_not_registered_destination=)
2018-02-10 21:27:08.912879 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [forward_user_not_registered_destination]=[UNDEF]
Dialplan: sofia/external/5555551212 at somedomain.com Action set(do_not_disturb=${user_data ${destination_number}@${domain_name} var do_not_disturb}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(do_not_disturb=)
2018-02-10 21:27:08.912879 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [do_not_disturb]=[UNDEF]
Dialplan: sofia/external/5555551212 at somedomain.com Action set(call_timeout=${user_data ${destination_number}@${domain_name} var call_timeout}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(call_timeout=30)
2018-02-10 21:27:08.932884 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [call_timeout]=[30]
Dialplan: sofia/external/5555551212 at somedomain.com Action set(missed_call_app=${user_data ${destination_number}@${domain_name} var missed_call_app}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(missed_call_app=)
2018-02-10 21:27:08.932884 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [missed_call_app]=[UNDEF]
Dialplan: sofia/external/5555551212 at somedomain.com Action set(missed_call_data=${user_data ${destination_number}@${domain_name} var missed_call_data}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(missed_call_data=)
2018-02-10 21:27:08.932884 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [missed_call_data]=[UNDEF]
Dialplan: sofia/external/5555551212 at somedomain.com Action set(toll_allow=${user_data ${destination_number}@${domain_name} var toll_allow}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(toll_allow=)
2018-02-10 21:27:08.932884 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [toll_allow]=[UNDEF]
Dialplan: sofia/external/5555551212 at somedomain.com Action set(call_screen_enabled=${user_data ${destination_number}@${domain_name} var call_screen_enabled}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(call_screen_enabled=false)
2018-02-10 21:27:08.932884 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [call_screen_enabled]=[false]
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->call-direction] continue=true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [call-direction] ${call_direction}(inbound) =~ /^(inbound|outbound|local)$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->variables] continue=true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [variables] () =~ // break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com Action export(origination_callee_id_name=${destination_number})
Dialplan: sofia/external/5555551212 at somedomain.com Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->user_record] continue=true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [user_record] () =~ // break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com Action set(user_record=${user_data ${destination_number}@${domain_name} var user_record}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com set(user_record=)
2018-02-10 21:27:08.932884 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [user_record]=[UNDEF]
Dialplan: sofia/external/5555551212 at somedomain.com Action set(from_user_exists=${user_exists id ${sip_from_user} ${sip_from_host}}) INLINE
2018-02-10 21:27:08.932884 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f6e24050320 Connected.
2018-02-10 21:27:08.932884 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f6e24050320 released.
EXECUTE sofia/external/5555551212 at somedomain.com set(from_user_exists=false)
2018-02-10 21:27:08.932884 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [from_user_exists]=[false]
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [user_record] ${user_exists}(true) =~ /^true$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [user_record] ${user_record}() =~ /^all$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [user_record] ${user_exists}(true) =~ /^true$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [user_record] ${call_direction}(inbound) =~ /^inbound$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [user_record] ${user_record}() =~ /^inbound$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [user_record] ${user_exists}(true) =~ /^true$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [user_record] ${call_direction}(inbound) =~ /^outbound$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [user_record] ${user_record}() =~ /^outbound$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [user_record] ${user_exists}(true) =~ /^true$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [user_record] ${call_direction}(inbound) =~ /^local$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [user_record] ${user_record}() =~ /^local$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [user_record] ${from_user_exists}(false) =~ /^true$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [user_record] ${from_user_exists}(false) =~ /^true$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [user_record] ${from_user_record}() =~ /^all$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [user_record] ${from_user_exists}(false) =~ /^true$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [user_record] ${call_direction}(inbound) =~ /^inbound$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [user_record] ${from_user_record}() =~ /^inbound$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [user_record] ${from_user_exists}(false) =~ /^true$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [user_record] ${call_direction}(inbound) =~ /^outbound$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [user_record] ${from_user_record}() =~ /^outbound$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [user_record] ${from_user_exists}(false) =~ /^true$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [user_record] ${call_direction}(inbound) =~ /^local$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [user_record] ${from_user_record}() =~ /^local$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [user_record] ${record_session}() =~ /^true$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->redial] continue=true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [redial] destination_number(113) =~ /^(redial|\*870)$/ break=on-true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [redial] () =~ // break=never
Dialplan: sofia/external/5555551212 at somedomain.com Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->speed_dial] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [speed_dial] destination_number(113) =~ /^\*0(.*)$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->sipious.011.9-17d] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [sipious.011.9-17d] destination_number(113) =~ /^(011\d{9,17})$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->sipious.10d] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [sipious.10d] destination_number(113) =~ /^(\d{10})$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->sipious.11d] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [sipious.11d] destination_number(113) =~ /^\+?(\d{11})$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->agent_status] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [agent_status] destination_number(113) =~ /^\*22$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->agent_status_id] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [agent_status_id] destination_number(113) =~ /^\*23$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->group-intercept] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [group-intercept] destination_number(113) =~ /^\*8$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->page-extension] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [page-extension] destination_number(113) =~ /^\*8(\d{2,7})$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->call_privacy] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [call_privacy] destination_number(113) =~ /^\*67(\d+)$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->call_return] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [call_return] destination_number(113) =~ /^\*69$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->extension_queue] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [extension_queue] destination_number(113) =~ /^\*800(.*)$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->intercept-ext] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [intercept-ext] destination_number(113) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->intercept-ext-polycom] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [intercept-ext-polycom] destination_number(113) =~ /^\*97(\d+)$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->dx] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [dx] destination_number(113) =~ /^dx$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->att_xfer] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [att_xfer] destination_number(113) =~ /^att_xfer$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->extension-to-voicemail] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [extension-to-voicemail] ${user_exists}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [extension-to-voicemail] username(5555551212) =~ /^5555551212$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [extension-to-voicemail] destination_number(113) =~ /^5555551212$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->send_to_voicemail] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [send_to_voicemail] destination_number(113) =~ /^\*99(\d{2,10})$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->vmain] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [vmain] destination_number(113) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->xfer_vm] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [xfer_vm] destination_number(113) =~ /^xfer_vm$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->is_transfer] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [is_transfer] destination_number(113) =~ /^is_transfer$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->vmain_user] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [vmain_user] destination_number(113) =~ /^\*97$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->Dev-Conference] continue=
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [Dev-Conference] destination_number(113) =~ /^1000$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->Exec-Conference] continue=
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [Exec-Conference] destination_number(113) =~ /^2000$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->cf] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [cf] destination_number(113) =~ /^cf$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->delay_echo] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [delay_echo] destination_number(113) =~ /^\*9195$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->echo] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [echo] destination_number(113) =~ /^\*9196$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->is_zrtp_secure] continue=true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [is_zrtp_secure] ${zrtp_secure_media_confirmed}() =~ /^true$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com ANTI-Action eval(not_secure)
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->milliwatt] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [milliwatt] destination_number(113) =~ /^\*9197$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->is_secure] continue=true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [is_secure] ${sip_via_protocol}(udp) =~ /tls/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->tone_stream] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [tone_stream] destination_number(113) =~ /^\*9198$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->hold_music] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [hold_music] destination_number(113) =~ /^\*9664$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->recordings] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [recordings] destination_number(113) =~ /^\*(732)$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->directory] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [directory] destination_number(113) =~ /^\*411$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->wake-up] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [wake-up] destination_number(113) =~ /^\*(925)$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->valet_park] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [valet_park] destination_number(113) =~ /^(park\+)?(\*59[0-9][0-9])$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [valet_park] ${sip_h_Referred-By}() =~ /sip:(.*)@.*/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [valet_park] destination_number(113) =~ /^(park\+)?(\*59[0-9][0-9])$/ break=never
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [valet_park] destination_number(113) =~ /^(park\+)?(\*59[0-9][0-9])$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->operator] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [operator] destination_number(113) =~ /^0$|^operator$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->operator-forward] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [operator-forward] destination_number(113) =~ /^\*000$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->do-not-disturb] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [do-not-disturb] destination_number(113) =~ /^\*77$/ break=on-true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [do-not-disturb] destination_number(113) =~ /^\*78$|\*363$/ break=on-true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [do-not-disturb] destination_number(113) =~ /^\*79$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->call-forward] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [call-forward] destination_number(113) =~ /^\*72$/ break=on-true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [call-forward] destination_number(113) =~ /^\*73$/ break=on-true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [call-forward] destination_number(113) =~ /^\*74$/ break=on-true
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->call forward all] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [call forward all] ${user_exists}(true) =~ /^true/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [call forward all] ${forward_all_enabled}() =~ /^true/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->follow-me] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [follow-me] destination_number(113) =~ /^\*21$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->clear_sip_auto_answer] continue=true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [clear_sip_auto_answer] ${click_to_call}() =~ /true/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->talking clock date and time] continue=true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [talking clock date and time] destination_number(113) =~ /^\*9172$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->talking clock time] continue=true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [talking clock time] destination_number(113) =~ /^\*9170$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->talking clock date] continue=true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [talking clock date] destination_number(113) =~ /^\*9171$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->call_screen] continue=true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (FAIL) [call_screen] ${call_screen_enabled}(false) =~ /^true$/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->local_extension] continue=true
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [local_extension] ${user_exists}(true) =~ /true/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com Action export(dialed_extension=${destination_number}) INLINE
EXECUTE sofia/external/5555551212 at somedomain.com export(dialed_extension=113)
2018-02-10 21:27:08.932884 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [dialed_extension]=[113]
Dialplan: sofia/external/5555551212 at somedomain.com Action limit(hash ${domain_name} ${destination_number} ${limit_max} ${limit_destination})
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [local_extension] () =~ // break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com Action set(hangup_after_bridge=true)
Dialplan: sofia/external/5555551212 at somedomain.com Action set(continue_on_fail=true)
Dialplan: sofia/external/5555551212 at somedomain.com Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
Dialplan: sofia/external/5555551212 at somedomain.com Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
Dialplan: sofia/external/5555551212 at somedomain.com Action set(called_party_call_group=${user_data(${dialed_extension}@${domain_name} var call_group)})
Dialplan: sofia/external/5555551212 at somedomain.com Action hash(insert/${domain_name}-last_dial/${called_party_call_group}/${uuid})
Dialplan: sofia/external/5555551212 at somedomain.com Action export(domain_name=${domain_name})
Dialplan: sofia/external/5555551212 at somedomain.com Action bridge({originate_timeout=40}user/${destination_number}@${domain_name})
Dialplan: sofia/external/5555551212 at somedomain.com parsing [c00-pbx.somedomain.com->voicemail] continue=false
Dialplan: sofia/external/5555551212 at somedomain.com Regex (PASS) [voicemail] ${user_exists}(true) =~ /true/ break=on-false
Dialplan: sofia/external/5555551212 at somedomain.com Action answer()
Dialplan: sofia/external/5555551212 at somedomain.com Action sleep(1000)
Dialplan: sofia/external/5555551212 at somedomain.com Action set(voicemail_action=save)
Dialplan: sofia/external/5555551212 at somedomain.com Action set(voicemail_id=${destination_number})
Dialplan: sofia/external/5555551212 at somedomain.com Action set(voicemail_profile=default)
Dialplan: sofia/external/5555551212 at somedomain.com Action lua(app.lua voicemail)
2018-02-10 21:27:08.932884 [DEBUG] switch_core_state_machine.c:286 (sofia/external/5555551212 at somedomain.com) State Change CS_ROUTING -> CS_EXECUTE
2018-02-10 21:27:08.932884 [DEBUG] switch_core_state_machine.c:643 (sofia/external/5555551212 at somedomain.com) State ROUTING going to sleep
2018-02-10 21:27:08.932884 [DEBUG] switch_core_state_machine.c:584 (sofia/external/5555551212 at somedomain.com) Running State Change CS_EXECUTE (Cur 1 Tot 84)
2018-02-10 21:27:08.932884 [DEBUG] switch_core_state_machine.c:650 (sofia/external/5555551212 at somedomain.com) State EXECUTE
2018-02-10 21:27:08.932884 [DEBUG] mod_sofia.c:198 sofia/external/5555551212 at somedomain.com SOFIA EXECUTE
2018-02-10 21:27:08.932884 [DEBUG] switch_core_state_machine.c:328 sofia/external/5555551212 at somedomain.com Standard EXECUTE
EXECUTE sofia/external/5555551212 at somedomain.com export(origination_callee_id_name=113)
2018-02-10 21:27:08.932884 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [origination_callee_id_name]=[113]
EXECUTE sofia/external/5555551212 at somedomain.com set(RFC2822_DATE=Sat, 10 Feb 2018 21:27:08 +0000)
2018-02-10 21:27:08.932884 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [RFC2822_DATE]=[Sat, 10 Feb 2018 21:27:08 +0000]
EXECUTE sofia/external/5555551212 at somedomain.com hash(insert/c00-pbx.somedomain.com-last_dial/5555551212/113)
EXECUTE sofia/external/5555551212 at somedomain.com eval(not_secure)
EXECUTE sofia/external/5555551212 at somedomain.com limit(hash c00-pbx.somedomain.com 113  )
2018-02-10 21:27:08.932884 [DEBUG] switch_limit.c:126 incr called: c00-pbx.somedomain.com_113 max:-1, interval:0
2018-02-10 21:27:08.932884 [DEBUG] mod_hash.c:194 Usage for c00-pbx.somedomain.com_113 is now 1
EXECUTE sofia/external/5555551212 at somedomain.com set(hangup_after_bridge=true)
2018-02-10 21:27:08.932884 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [hangup_after_bridge]=[true]
EXECUTE sofia/external/5555551212 at somedomain.com set(continue_on_fail=true)
2018-02-10 21:27:08.932884 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [continue_on_fail]=[true]
EXECUTE sofia/external/5555551212 at somedomain.com hash(insert/c00-pbx.somedomain.com-call_return/113/5555551212)
EXECUTE sofia/external/5555551212 at somedomain.com hash(insert/c00-pbx.somedomain.com-last_dial_ext/113/2640726c-0ea9-11e8-ad64-53f0f54925c8)
EXECUTE sofia/external/5555551212 at somedomain.com set(called_party_call_group=)
2018-02-10 21:27:08.952886 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [called_party_call_group]=[UNDEF]
EXECUTE sofia/external/5555551212 at somedomain.com hash(insert/c00-pbx.somedomain.com-last_dial//2640726c-0ea9-11e8-ad64-53f0f54925c8)
EXECUTE sofia/external/5555551212 at somedomain.com export(domain_name=c00-pbx.somedomain.com)
2018-02-10 21:27:08.952886 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [domain_name]=[c00-pbx.somedomain.com]
EXECUTE sofia/external/5555551212 at somedomain.com bridge({originate_timeout=40}user/113 at c00-pbx.somedomain.com)
2018-02-10 21:27:08.952886 [DEBUG] switch_channel.c:1250 sofia/external/5555551212 at somedomain.com EXPORTING[export_vars] [dialed_extension]=[113] to event
2018-02-10 21:27:08.952886 [DEBUG] switch_channel.c:1250 sofia/external/5555551212 at somedomain.com EXPORTING[export_vars] [origination_callee_id_name]=[113] to event
2018-02-10 21:27:08.952886 [DEBUG] switch_channel.c:1250 sofia/external/5555551212 at somedomain.com EXPORTING[export_vars] [domain_name]=[c00-pbx.somedomain.com] to event
2018-02-10 21:27:08.952886 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables
2018-02-10 21:27:08.952886 [DEBUG] switch_channel.c:1250 sofia/external/5555551212 at somedomain.com EXPORTING[export_vars] [dialed_extension]=[113] to event
2018-02-10 21:27:08.952886 [DEBUG] switch_channel.c:1250 sofia/external/5555551212 at somedomain.com EXPORTING[export_vars] [origination_callee_id_name]=[113] to event
2018-02-10 21:27:08.952886 [DEBUG] switch_channel.c:1250 sofia/external/5555551212 at somedomain.com EXPORTING[export_vars] [domain_name]=[c00-pbx.somedomain.com] to event
2018-02-10 21:27:08.952886 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables
2018-02-10 21:27:08.952886 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/113 at 161.97.193.191:51094 [264c07a8-0ea9-11e8-ad90-53f0f54925c8]
2018-02-10 21:27:08.952886 [DEBUG] mod_sofia.c:4819 (sofia/internal/113 at 161.97.193.191:51094) State Change CS_NEW -> CS_INIT
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/113 at 161.97.193.191:51094) Running State Change CS_INIT (Cur 2 Tot 85)
2018-02-10 21:27:08.952886 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/113 at 161.97.193.191:51337 [264c1838-0ea9-11e8-ad95-53f0f54925c8]
2018-02-10 21:27:08.952886 [DEBUG] mod_sofia.c:4819 (sofia/internal/113 at 161.97.193.191:51337) State Change CS_NEW -> CS_INIT
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:627 (sofia/internal/113 at 161.97.193.191:51094) State INIT
2018-02-10 21:27:08.952886 [DEBUG] mod_sofia.c:90 sofia/internal/113 at 161.97.193.191:51094 SOFIA INIT
2018-02-10 21:27:08.952886 [DEBUG] sofia_glue.c:1264 sip:113 at 161.97.193.191:51094;rinstance=630f1da21036c7ab;transport=tcp Setting proxy route to sofia/internal/113 at 161.97.193.191:51094
2018-02-10 21:27:08.952886 [DEBUG] sofia_glue.c:1295 sofia/internal/113 at 161.97.193.191:51094 sending invite version: 1.6.20  64bit
Local SDP:
v=0
o=FreeSWITCH 1518274048 1518274049 IN IP4 52.25.47.217
s=FreeSWITCH
c=IN IP4 52.25.47.217
t=0 0
m=audio 23980 RTP/AVP 0 102 103 9 8 3 101 13 104 105 106 107
a=rtpmap:0 PCMU/8000
a=rtpmap:102 G7221/32000
a=fmtp:102 bitrate=48000
a=rtpmap:103 G7221/16000
a=fmtp:103 bitrate=32000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:104 telephone-event/32000
a=fmtp:104 0-16
a=rtpmap:106 telephone-event/16000
a=fmtp:106 0-16
a=rtpmap:13 CN/8000
a=rtpmap:105 CN/32000
a=rtpmap:107 CN/16000
a=ptime:20
a=sendrecv

2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:40 sofia/internal/113 at 161.97.193.191:51094 Standard INIT
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/113 at 161.97.193.191:51094) State Change CS_INIT -> CS_ROUTING
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:627 (sofia/internal/113 at 161.97.193.191:51094) State INIT going to sleep
2018-02-10 21:27:08.952886 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/113 at 161.97.193.191:51392 [264c28be-0ea9-11e8-ad9a-53f0f54925c8]
2018-02-10 21:27:08.952886 [DEBUG] mod_sofia.c:4819 (sofia/internal/113 at 161.97.193.191:51392) State Change CS_NEW -> CS_INIT
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/113 at 161.97.193.191:51337) Running State Change CS_INIT (Cur 4 Tot 87)
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/113 at 161.97.193.191:51094) Running State Change CS_ROUTING (Cur 4 Tot 87)
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/113 at 161.97.193.191:51094) State ROUTING
2018-02-10 21:27:08.952886 [DEBUG] mod_sofia.c:143 sofia/internal/113 at 161.97.193.191:51094 SOFIA ROUTING
2018-02-10 21:27:08.952886 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/113 at 161.97.193.191:51094) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/113 at 161.97.193.191:51094) State ROUTING going to sleep
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:627 (sofia/internal/113 at 161.97.193.191:51337) State INIT
2018-02-10 21:27:08.952886 [DEBUG] mod_sofia.c:90 sofia/internal/113 at 161.97.193.191:51337 SOFIA INIT
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/113 at 161.97.193.191:51094) Running State Change CS_CONSUME_MEDIA (Cur 4 Tot 87)
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/113 at 161.97.193.191:51392) Running State Change CS_INIT (Cur 4 Tot 87)
2018-02-10 21:27:08.952886 [DEBUG] sofia.c:7084 Channel sofia/internal/113 at 161.97.193.191:51094 entering state [calling][0]
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/113 at 161.97.193.191:51094) State CONSUME_MEDIA
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/113 at 161.97.193.191:51094) State CONSUME_MEDIA going to sleep
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:627 (sofia/internal/113 at 161.97.193.191:51392) State INIT
2018-02-10 21:27:08.952886 [DEBUG] mod_sofia.c:90 sofia/internal/113 at 161.97.193.191:51392 SOFIA INIT
2018-02-10 21:27:08.952886 [DEBUG] sofia_glue.c:1264 sip:113 at 161.97.193.191:51337;rinstance=3baeca2d38d0500d;transport=tcp Setting proxy route to sofia/internal/113 at 161.97.193.191:51337
2018-02-10 21:27:08.952886 [DEBUG] sofia_glue.c:1295 sofia/internal/113 at 161.97.193.191:51337 sending invite version: 1.6.20  64bit
Local SDP:
v=0
o=FreeSWITCH 1518268504 1518268505 IN IP4 52.25.47.217
s=FreeSWITCH
c=IN IP4 52.25.47.217
t=0 0
m=audio 29524 RTP/AVP 0 102 103 9 8 3 101 13 104 105 106 107
a=rtpmap:0 PCMU/8000
a=rtpmap:102 G7221/32000
a=fmtp:102 bitrate=48000
a=rtpmap:103 G7221/16000
a=fmtp:103 bitrate=32000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:104 telephone-event/32000
a=fmtp:104 0-16
a=rtpmap:106 telephone-event/16000
a=fmtp:106 0-16
a=rtpmap:13 CN/8000
a=rtpmap:105 CN/32000
a=rtpmap:107 CN/16000
a=ptime:20
a=sendrecv

2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:40 sofia/internal/113 at 161.97.193.191:51337 Standard INIT
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/113 at 161.97.193.191:51337) State Change CS_INIT -> CS_ROUTING
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:627 (sofia/internal/113 at 161.97.193.191:51337) State INIT going to sleep
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/113 at 161.97.193.191:51337) Running State Change CS_ROUTING (Cur 4 Tot 87)
2018-02-10 21:27:08.952886 [DEBUG] sofia_glue.c:1264 sip:113 at 161.97.193.191:51392;rinstance=3baeca2d38d0500d;transport=tcp Setting proxy route to sofia/internal/113 at 161.97.193.191:51392
2018-02-10 21:27:08.952886 [DEBUG] sofia_glue.c:1295 sofia/internal/113 at 161.97.193.191:51392 sending invite version: 1.6.20  64bit
Local SDP:
v=0
o=FreeSWITCH 1518274964 1518274965 IN IP4 52.25.47.217
s=FreeSWITCH
c=IN IP4 52.25.47.217
t=0 0
m=audio 23064 RTP/AVP 0 102 103 9 8 3 101 13 104 105 106 107
a=rtpmap:0 PCMU/8000
a=rtpmap:102 G7221/32000
a=fmtp:102 bitrate=48000
a=rtpmap:103 G7221/16000
a=fmtp:103 bitrate=32000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:104 telephone-event/32000
a=fmtp:104 0-16
a=rtpmap:106 telephone-event/16000
a=fmtp:106 0-16
a=rtpmap:13 CN/8000
a=rtpmap:105 CN/32000
a=rtpmap:107 CN/16000
a=ptime:20
a=sendrecv

2018-02-10 21:27:08.952886 [DEBUG] sofia.c:7084 Channel sofia/internal/113 at 161.97.193.191:51337 entering state [calling][0]
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/113 at 161.97.193.191:51337) State ROUTING
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:40 sofia/internal/113 at 161.97.193.191:51392 Standard INIT
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/113 at 161.97.193.191:51392) State Change CS_INIT -> CS_ROUTING
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:627 (sofia/internal/113 at 161.97.193.191:51392) State INIT going to sleep
2018-02-10 21:27:08.952886 [DEBUG] mod_sofia.c:143 sofia/internal/113 at 161.97.193.191:51337 SOFIA ROUTING
2018-02-10 21:27:08.952886 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/113 at 161.97.193.191:51337) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/113 at 161.97.193.191:51337) State ROUTING going to sleep
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/113 at 161.97.193.191:51337) Running State Change CS_CONSUME_MEDIA (Cur 4 Tot 87)
send 1715 bytes to tcp/[161.97.193.191]:51392 at 21:27:08.959845:
   ------------------------------------------------------------------------
   INVITE sip:113 at 161.97.193.191:51392;rinstance=3baeca2d38d0500d;transport=tcp SIP/2.0
   Via: SIP/2.0/TCP 52.25.47.217:5080;rport;branch=z9hG4bK9U4H9c0yB0eme
   Route: <sip:113 at 161.97.193.191:51392>;rinstance=3baeca2d38d0500d;transport=tcp
   Max-Forwards: 67
   From: "5555551212" <sip:5555551212 at c00-pbx.somedomain.com>;tag=Bct4yrNeQ5gmK
   To: <sip:113 at 161.97.193.191:51392;rinstance=3baeca2d38d0500d;transport=tcp>
   Call-ID: fdaa910f-894b-1236-0aa7-02a1db012a3a
   CSeq: 118812950 INVITE
   Contact: <sip:mod_sofia at 52.25.47.217:5080;transport=tcp>
   User-Agent: FreeSWITCH
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 595
   P-Account-ID: 99990009
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/113 at 161.97.193.191:51392) Running State Change CS_ROUTING (Cur 4 Tot 87)
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "5555551212" <sip:5555551212 at c00-pbx.somedomain.com>;party=calling;screen=no;privacy=off

   v=0
   o=FreeSWITCH 1518274964 1518274965 IN IP4 52.25.47.217
   s=FreeSWITCH
   c=IN IP4 52.25.47.217
   t=0 0
   m=audio 23064 RTP/AVP 0 102 103 9 8 3 101 13 104 105 106 107
   a=rtpmap:0 PCMU/8000
   a=rtpmap:102 G7221/32000
   a=fmtp:102 bitrate=48000
   a=rtpmap:103 G7221/16000
   a=fmtp:103 bitrate=32000
   a=rtpmap:9 G722/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=rtpmap:104 telephone-event/32000
   a=fmtp:104 0-16
   a=rtpmap:105 CN/32000
   a=rtpmap:106 telephone-event/16000
   a=fmtp:106 0-16
   a=rtpmap:107 CN/16000
   a=ptime:20
   ------------------------------------------------------------------------
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/113 at 161.97.193.191:51337) State CONSUME_MEDIA
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/113 at 161.97.193.191:51337) State CONSUME_MEDIA going to sleep
2018-02-10 21:27:08.952886 [DEBUG] sofia.c:7084 Channel sofia/internal/113 at 161.97.193.191:51392 entering state [calling][0]
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/113 at 161.97.193.191:51392) State ROUTING
2018-02-10 21:27:08.952886 [DEBUG] mod_sofia.c:143 sofia/internal/113 at 161.97.193.191:51392 SOFIA ROUTING
2018-02-10 21:27:08.952886 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/113 at 161.97.193.191:51392) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/113 at 161.97.193.191:51392) State ROUTING going to sleep
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/113 at 161.97.193.191:51392) Running State Change CS_CONSUME_MEDIA (Cur 4 Tot 87)
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/113 at 161.97.193.191:51392) State CONSUME_MEDIA
2018-02-10 21:27:08.952886 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/113 at 161.97.193.191:51392) State CONSUME_MEDIA going to sleep
recv 336 bytes from tcp/[161.97.193.191]:51392 at 21:27:09.537722:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/TCP 52.25.47.217:5080;rport=5080;branch=z9hG4bK9U4H9c0yB0eme
   To: <sip:113 at 161.97.193.191:51392;rinstance=3baeca2d38d0500d;transport=tcp>
   From: "5555551212" <sip:5555551212 at c00-pbx.somedomain.com>;tag=Bct4yrNeQ5gmK
   Call-ID: fdaa910f-894b-1236-0aa7-02a1db012a3a
   CSeq: 118812950 INVITE
   Content-Length: 0

   ------------------------------------------------------------------------
recv 519 bytes from tcp/[161.97.193.191]:51392 at 21:27:09.618735:
   ------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/TCP 52.25.47.217:5080;rport=5080;branch=z9hG4bK9U4H9c0yB0eme
   Contact: <sip:113 at 161.97.193.191:51392;rinstance=3baeca2d38d0500d;transport=tcp>
   To: "Geoffrey Mina"<sip:113 at 161.97.193.191:51392;rinstance=3baeca2d38d0500d;transport=tcp>;tag=e91e9a69
   From: "5555551212" <sip:5555551212 at c00-pbx.somedomain.com>;tag=Bct4yrNeQ5gmK
   Call-ID: fdaa910f-894b-1236-0aa7-02a1db012a3a
   CSeq: 118812950 INVITE
   User-Agent: Bria 5 release 5.0.3 stamp 88307
   Allow-Events: talk, hold
   Content-Length: 0

   ------------------------------------------------------------------------
2018-02-10 21:27:09.612821 [DEBUG] sofia.c:7084 Channel sofia/internal/113 at 161.97.193.191:51392 entering state [proceeding][180]
2018-02-10 21:27:09.612821 [NOTICE] sofia.c:7192 Ring-Ready sofia/internal/113 at 161.97.193.191:51392!
2018-02-10 21:27:09.612821 [DEBUG] switch_channel.c:3346 (sofia/internal/113 at 161.97.193.191:51392) Callstate Change DOWN -> RINGING
2018-02-10 21:27:09.612821 [INFO] switch_ivr_originate.c:1215 Sending early media
2018-02-10 21:27:09.612821 [DEBUG] switch_core_media.c:6878 AUDIO RTP [sofia/external/5555551212 at somedomain.com] 10.10.1.160 port 25762 -> 52.11.183.123 port 10466 codec: 0 ms: 20
2018-02-10 21:27:09.612821 [DEBUG] switch_rtp.c:4137 Starting timer [soft] 160 bytes per 20ms
2018-02-10 21:27:09.612821 [DEBUG] switch_core_media.c:7180 sofia/external/5555551212 at somedomain.com Set 2833 dtmf send payload to 101
2018-02-10 21:27:09.612821 [DEBUG] switch_core_media.c:7187 sofia/external/5555551212 at somedomain.com Set 2833 dtmf receive payload to 101
2018-02-10 21:27:09.612821 [DEBUG] switch_core_media.c:7210 sofia/external/5555551212 at somedomain.com Set rtp dtmf delay to 40
2018-02-10 21:27:09.612821 [DEBUG] mod_sofia.c:2364 Ring SDP:
v=0
o=FreeSWITCH 1518272267 1518272268 IN IP4 52.25.47.217
s=FreeSWITCH
c=IN IP4 52.25.47.217
t=0 0
m=audio 25762 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

2018-02-10 21:27:09.612821 [NOTICE] mod_sofia.c:2367 Pre-Answer sofia/external/5555551212 at somedomain.com!
2018-02-10 21:27:09.612821 [DEBUG] switch_channel.c:3474 (sofia/external/5555551212 at somedomain.com) Callstate Change RINGING -> EARLY
2018-02-10 21:27:09.612821 [DEBUG] switch_ivr_originate.c:1273 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
2018-02-10 21:27:09.612821 [DEBUG] switch_core_codec.c:223 sofia/external/5555551212 at somedomain.com Push codec L16:100
2018-02-10 21:27:09.612821 [DEBUG] switch_ivr_originate.c:1342 Play Ringback Tone [%(2000,4000,440,480)]
send 1098 bytes to udp/[174.37.181.39]:5060 at 21:27:09.624226:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP 174.37.181.39;branch=z9hG4bK1375.9bd5a993470896a1e4e38b2bff4f084b.0
   Via: SIP/2.0/UDP 52.11.183.123:5060;branch=z9hG4bK61968ea6;rport=5060
   Record-Route: <sip:174.37.181.39;lr;ftag=as54b5500e>
   From: "5555551212" <sip:5555551212 at somedomain.com>;tag=as54b5500e
   To: <sip:113 at 174.37.181.39:5060>;tag=Z3SgKX3H5Fc7a
   Call-ID: 4f43cc193246d9f07d56ccf80ecb9d09 at somedomain.com
   CSeq: 102 INVITE
   Contact: <sip:113 at 52.25.47.217:5060;transport=udp>
   User-Agent: FreeSWITCH
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 220
   Remote-Party-ID: "113" <sip:113 at 174.37.181.39>;party=calling;privacy=off;screen=no

   v=0
   o=FreeSWITCH 1518272267 1518272268 IN IP4 52.25.47.217
   s=FreeSWITCH
   c=IN IP4 52.25.47.217
   t=0 0
   m=audio 25762 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
2018-02-10 21:27:09.632878 [DEBUG] sofia.c:7084 Channel sofia/external/5555551212 at somedomain.com entering state [early][183]
recv 398 bytes from tcp/[161.97.193.191]:51392 at 21:27:11.147223:
   ------------------------------------------------------------------------
   SIP/2.0 486 Busy Here
   Via: SIP/2.0/TCP 52.25.47.217:5080;rport=5080;branch=z9hG4bK9U4H9c0yB0eme
   To: <sip:113 at 161.97.193.191:51392;rinstance=3baeca2d38d0500d;transport=tcp>;tag=e91e9a69
   From: "5555551212" <sip:5555551212 at c00-pbx.somedomain.com>;tag=Bct4yrNeQ5gmK
   Call-ID: fdaa910f-894b-1236-0aa7-02a1db012a3a
   CSeq: 118812950 INVITE
   User-Agent: Bria 5 release 5.0.3 stamp 88307
   Content-Length: 0

   ------------------------------------------------------------------------
send 502 bytes to tcp/[161.97.193.191]:51392 at 21:27:11.147324:
   ------------------------------------------------------------------------
   ACK sip:113 at 161.97.193.191:51392;rinstance=3baeca2d38d0500d;transport=tcp SIP/2.0
   Via: SIP/2.0/TCP 52.25.47.217:5080;rport;branch=z9hG4bK9U4H9c0yB0eme
   Route: <sip:113 at 161.97.193.191:51392>;rinstance=3baeca2d38d0500d;transport=tcp
   Max-Forwards: 67
   From: "5555551212" <sip:5555551212 at c00-pbx.somedomain.com>;tag=Bct4yrNeQ5gmK
   To: <sip:113 at 161.97.193.191:51392;rinstance=3baeca2d38d0500d;transport=tcp>;tag=e91e9a69
   Call-ID: fdaa910f-894b-1236-0aa7-02a1db012a3a
   CSeq: 118812950 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
2018-02-10 21:27:11.132820 [DEBUG] sofia.c:7084 Channel sofia/internal/113 at 161.97.193.191:51392 entering state [terminated][486]
2018-02-10 21:27:11.132820 [NOTICE] sofia.c:8273 Hangup sofia/internal/113 at 161.97.193.191:51392 [CS_CONSUME_MEDIA] [USER_BUSY]
2018-02-10 21:27:11.132820 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/113 at 161.97.193.191:51392) Running State Change CS_HANGUP (Cur 4 Tot 87)
2018-02-10 21:27:11.132820 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/113 at 161.97.193.191:51392) Callstate Change RINGING -> HANGUP
2018-02-10 21:27:11.132820 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/113 at 161.97.193.191:51392) State HANGUP
2018-02-10 21:27:11.132820 [DEBUG] mod_sofia.c:438 Channel sofia/internal/113 at 161.97.193.191:51392 hanging up, cause: USER_BUSY
2018-02-10 21:27:11.132820 [DEBUG] switch_core_state_machine.c:60 sofia/internal/113 at 161.97.193.191:51392 Standard HANGUP, cause: USER_BUSY
2018-02-10 21:27:11.132820 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/113 at 161.97.193.191:51392) State HANGUP going to sleep
2018-02-10 21:27:11.132820 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/113 at 161.97.193.191:51392) State Change CS_HANGUP -> CS_REPORTING
2018-02-10 21:27:11.132820 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/113 at 161.97.193.191:51392) Running State Change CS_REPORTING (Cur 4 Tot 87)
2018-02-10 21:27:11.132820 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/113 at 161.97.193.191:51392) State REPORTING
2018-02-10 21:27:11.132820 [DEBUG] switch_core_state_machine.c:174 sofia/internal/113 at 161.97.193.191:51392 Standard REPORTING, cause: USER_BUSY
2018-02-10 21:27:11.132820 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/113 at 161.97.193.191:51392) State REPORTING going to sleep
2018-02-10 21:27:11.132820 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/113 at 161.97.193.191:51392) State Change CS_REPORTING -> CS_DESTROY
2018-02-10 21:27:11.132820 [DEBUG] switch_core_session.c:1665 Session 87 (sofia/internal/113 at 161.97.193.191:51392) Locked, Waiting on external entities
2018-02-10 21:27:40.952819 [DEBUG] sofia.c:7084 Channel sofia/internal/113 at 161.97.193.191:51094 entering state [terminated][408]
2018-02-10 21:27:40.952819 [NOTICE] sofia.c:8273 Hangup sofia/internal/113 at 161.97.193.191:51094 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE]
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/113 at 161.97.193.191:51094) Running State Change CS_HANGUP (Cur 4 Tot 87)
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/113 at 161.97.193.191:51094) Callstate Change DOWN -> HANGUP
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/113 at 161.97.193.191:51094) State HANGUP
2018-02-10 21:27:40.952819 [DEBUG] mod_sofia.c:438 Channel sofia/internal/113 at 161.97.193.191:51094 hanging up, cause: RECOVERY_ON_TIMER_EXPIRE
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:60 sofia/internal/113 at 161.97.193.191:51094 Standard HANGUP, cause: RECOVERY_ON_TIMER_EXPIRE
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/113 at 161.97.193.191:51094) State HANGUP going to sleep
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/113 at 161.97.193.191:51094) State Change CS_HANGUP -> CS_REPORTING
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/113 at 161.97.193.191:51094) Running State Change CS_REPORTING (Cur 4 Tot 87)
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/113 at 161.97.193.191:51094) State REPORTING
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:174 sofia/internal/113 at 161.97.193.191:51094 Standard REPORTING, cause: RECOVERY_ON_TIMER_EXPIRE
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/113 at 161.97.193.191:51094) State REPORTING going to sleep
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/113 at 161.97.193.191:51094) State Change CS_REPORTING -> CS_DESTROY
2018-02-10 21:27:40.952819 [DEBUG] switch_core_session.c:1665 Session 85 (sofia/internal/113 at 161.97.193.191:51094) Locked, Waiting on external entities
2018-02-10 21:27:40.952819 [DEBUG] sofia.c:7084 Channel sofia/internal/113 at 161.97.193.191:51337 entering state [terminated][408]
2018-02-10 21:27:40.952819 [NOTICE] sofia.c:8273 Hangup sofia/internal/113 at 161.97.193.191:51337 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE]
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/113 at 161.97.193.191:51337) Running State Change CS_HANGUP (Cur 4 Tot 87)
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/113 at 161.97.193.191:51337) Callstate Change DOWN -> HANGUP
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/113 at 161.97.193.191:51337) State HANGUP
2018-02-10 21:27:40.952819 [DEBUG] mod_sofia.c:438 Channel sofia/internal/113 at 161.97.193.191:51337 hanging up, cause: RECOVERY_ON_TIMER_EXPIRE
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:60 sofia/internal/113 at 161.97.193.191:51337 Standard HANGUP, cause: RECOVERY_ON_TIMER_EXPIRE
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/113 at 161.97.193.191:51337) State HANGUP going to sleep
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/113 at 161.97.193.191:51337) State Change CS_HANGUP -> CS_REPORTING
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/113 at 161.97.193.191:51337) Running State Change CS_REPORTING (Cur 4 Tot 87)
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/113 at 161.97.193.191:51337) State REPORTING
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:174 sofia/internal/113 at 161.97.193.191:51337 Standard REPORTING, cause: RECOVERY_ON_TIMER_EXPIRE
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/113 at 161.97.193.191:51337) State REPORTING going to sleep
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/113 at 161.97.193.191:51337) State Change CS_REPORTING -> CS_DESTROY
2018-02-10 21:27:40.952819 [DEBUG] switch_core_session.c:1665 Session 86 (sofia/internal/113 at 161.97.193.191:51337) Locked, Waiting on external entities
2018-02-10 21:27:40.952819 [DEBUG] switch_core_codec.c:248 sofia/external/5555551212 at somedomain.com Restore previous codec PCMU:0.
2018-02-10 21:27:40.952819 [DEBUG] switch_ivr_originate.c:3848 Originate Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE]
2018-02-10 21:27:40.952819 [NOTICE] switch_core_session.c:1683 Session 85 (sofia/internal/113 at 161.97.193.191:51094) Ended
2018-02-10 21:27:40.952819 [NOTICE] switch_core_session.c:1687 Close Channel sofia/internal/113 at 161.97.193.191:51094 [CS_DESTROY]
2018-02-10 21:27:40.952819 [NOTICE] switch_core_session.c:1683 Session 86 (sofia/internal/113 at 161.97.193.191:51337) Ended
2018-02-10 21:27:40.952819 [NOTICE] switch_core_session.c:1687 Close Channel sofia/internal/113 at 161.97.193.191:51337 [CS_DESTROY]
2018-02-10 21:27:40.952819 [NOTICE] switch_core_session.c:1683 Session 87 (sofia/internal/113 at 161.97.193.191:51392) Ended
2018-02-10 21:27:40.952819 [NOTICE] switch_core_session.c:1687 Close Channel sofia/internal/113 at 161.97.193.191:51392 [CS_DESTROY]
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/113 at 161.97.193.191:51337) Running State Change CS_DESTROY (Cur 1 Tot 87)
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/113 at 161.97.193.191:51094) Running State Change CS_DESTROY (Cur 1 Tot 87)
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/113 at 161.97.193.191:51337) State DESTROY
2018-02-10 21:27:40.952819 [DEBUG] mod_sofia.c:343 sofia/internal/113 at 161.97.193.191:51337 SOFIA DESTROY
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:181 sofia/internal/113 at 161.97.193.191:51337 Standard DESTROY
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/113 at 161.97.193.191:51337) State DESTROY going to sleep
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/113 at 161.97.193.191:51094) State DESTROY
2018-02-10 21:27:40.952819 [DEBUG] mod_sofia.c:343 sofia/internal/113 at 161.97.193.191:51094 SOFIA DESTROY
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:181 sofia/internal/113 at 161.97.193.191:51094 Standard DESTROY
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/113 at 161.97.193.191:51094) State DESTROY going to sleep
2018-02-10 21:27:40.952819 [NOTICE] switch_ivr_originate.c:2851 Cannot create outgoing channel of type [user] cause: [RECOVERY_ON_TIMER_EXPIRE]
2018-02-10 21:27:40.952819 [DEBUG] switch_ivr_originate.c:3848 Originate Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE]
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/113 at 161.97.193.191:51392) Running State Change CS_DESTROY (Cur 1 Tot 87)
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/113 at 161.97.193.191:51392) State DESTROY
2018-02-10 21:27:40.952819 [DEBUG] mod_sofia.c:343 sofia/internal/113 at 161.97.193.191:51392 SOFIA DESTROY
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:181 sofia/internal/113 at 161.97.193.191:51392 Standard DESTROY
2018-02-10 21:27:40.952819 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/113 at 161.97.193.191:51392) State DESTROY going to sleep
2018-02-10 21:27:40.952819 [INFO] mod_dptools.c:3436 Originate Failed.  Cause: RECOVERY_ON_TIMER_EXPIRE
EXECUTE sofia/external/5555551212 at somedomain.com answer()
2018-02-10 21:27:40.952819 [DEBUG] switch_core_media.c:6861 Audio params are unchanged for sofia/external/5555551212 at somedomain.com.
2018-02-10 21:27:40.952819 [DEBUG] mod_sofia.c:850 Local SDP sofia/external/5555551212 at somedomain.com:
v=0
o=FreeSWITCH 1518272267 1518272269 IN IP4 52.25.47.217
s=FreeSWITCH
c=IN IP4 52.25.47.217
t=0 0
m=audio 25762 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

2018-02-10 21:27:40.952819 [NOTICE] mod_dptools.c:1312 Channel [sofia/external/5555551212 at somedomain.com] has been answered
send 1059 bytes to udp/[174.37.181.39]:5060 at 21:27:40.963158:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 174.37.181.39;branch=z9hG4bK1375.9bd5a993470896a1e4e38b2bff4f084b.0
   Via: SIP/2.0/UDP 52.11.183.123:5060;branch=z9hG4bK61968ea6;rport=5060
   Record-Route: <sip:174.37.181.39;lr;ftag=as54b5500e>
   From: "5555551212" <sip:5555551212 at somedomain.com>;tag=as54b5500e
   To: <sip:113 at 174.37.181.39:5060>;tag=Z3SgKX3H5Fc7a
   Call-ID: 4f43cc193246d9f07d56ccf80ecb9d09 at somedomain.com
   CSeq: 102 INVITE
   Contact: <sip:113 at 52.25.47.217:5060;transport=udp>
   User-Agent: FreeSWITCH
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 220
   Remote-Party-ID: "113" <sip:113 at 174.37.181.39>;party=calling;privacy=off;screen=no

   v=0
   o=FreeSWITCH 1518272267 1518272268 IN IP4 52.25.47.217
   s=FreeSWITCH
   c=IN IP4 52.25.47.217
   t=0 0
   m=audio 25762 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
2018-02-10 21:27:40.952819 [DEBUG] switch_channel.c:3773 (sofia/external/5555551212 at somedomain.com) Callstate Change EARLY -> ACTIVE
2018-02-10 21:27:40.952819 [DEBUG] sofia.c:7084 Channel sofia/external/5555551212 at somedomain.com entering state [completed][200]
EXECUTE sofia/external/5555551212 at somedomain.com sleep(1000)
recv 597 bytes from udp/[174.37.181.39]:5060 at 21:27:40.981008:
   ------------------------------------------------------------------------
   ACK sip:113 at 52.25.47.217:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 174.37.181.39;branch=z9hG4bK1375.1a5a8dc16597fd9a60f6e198c0e8018e.0
   Via: SIP/2.0/UDP 52.11.183.123:5060;branch=z9hG4bK6e94b06c;rport=5060
   From: "5555551212" <sip:5555551212 at somedomain.com>;tag=as54b5500e
   To: <sip:113 at 174.37.181.39:5060>;tag=Z3SgKX3H5Fc7a
   Contact: <sip:5555551212 at 52.11.183.123>
   Call-ID: 4f43cc193246d9f07d56ccf80ecb9d09 at somedomain.com
   CSeq: 102 ACK
   User-Agent: G-Tel v1.0
   Max-Forwards: 69
   Remote-Party-ID: "5555551212" <sip:5555551212 at somedomain.com>;privacy=off;screen=no
   Content-Length: 0

   ------------------------------------------------------------------------
2018-02-10 21:27:40.972822 [DEBUG] sofia.c:7084 Channel sofia/external/5555551212 at somedomain.com entering state [ready][200]
2018-02-10 21:27:41.052818 [DEBUG] switch_rtp.c:7308 Correct audio ip/port confirmed.
EXECUTE sofia/external/5555551212 at somedomain.com set(voicemail_action=save)
2018-02-10 21:27:41.952820 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [voicemail_action]=[save]
EXECUTE sofia/external/5555551212 at somedomain.com set(voicemail_id=113)
2018-02-10 21:27:41.952820 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [voicemail_id]=[113]
EXECUTE sofia/external/5555551212 at somedomain.com set(voicemail_profile=default)
2018-02-10 21:27:41.952820 [DEBUG] mod_dptools.c:1548 SET sofia/external/5555551212 at somedomain.com [voicemail_profile]=[default]
EXECUTE sofia/external/5555551212 at somedomain.com lua(app.lua voicemail)
2018-02-10 21:27:41.952820 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7f6e24050320 Connected.
EXECUTE sofia/external/5555551212 at somedomain.com unbind_meta_app()
2018-02-10 21:27:41.952820 [INFO] switch_ivr_async.c:4072 UnBound A-Leg: ALL
2018-02-10 21:27:41.952820 [DEBUG] switch_cpp.cpp:745 CoreSession::setVariable('playback_terminators', '#')
EXECUTE sofia/external/5555551212 at somedomain.com playback(silence_stream://200)
2018-02-10 21:27:41.972822 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms
2018-02-10 21:27:42.152818 [DEBUG] switch_ivr_play_say.c:1942 done playing file silence_stream://200
2018-02-10 21:27:42.152818 [DEBUG] switch_core_file.c:342 File /var/lib/freeswitch/storage/voicemail/default/c00-pbx.somedomain.com/113/greeting_1.wav sample rate 16000 doesn't match requested rate 8000
2018-02-10 21:27:42.152818 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms
recv 620 bytes from udp/[174.37.181.39]:5060 at 21:27:45.144259:
   ------------------------------------------------------------------------
   BYE sip:113 at 52.25.47.217:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 174.37.181.39;branch=z9hG4bK2375.b9d921e7f38da83d8c7dcaa0c03cab38.0
   Via: SIP/2.0/UDP 52.11.183.123:5060;branch=z9hG4bK3af11d1c;rport=5060
   From: "5555551212" <sip:5555551212 at somedomain.com>;tag=as54b5500e
   To: <sip:113 at 174.37.181.39:5060>;tag=Z3SgKX3H5Fc7a
   Call-ID: 4f43cc193246d9f07d56ccf80ecb9d09 at somedomain.com
   CSeq: 103 BYE
   User-Agent: G-Tel v1.0
   Max-Forwards: 69
   Remote-Party-ID: "5555551212" <sip:5555551212 at somedomain.com>;privacy=off;screen=no
   X-Asterisk-HangupCause: Unknown
   X-Asterisk-HangupCauseCode: 0
   Content-Length: 0

   ------------------------------------------------------------------------
2018-02-10 21:27:45.152820 [NOTICE] sofia.c:1012 Hangup sofia/external/5555551212 at somedomain.com [CS_EXECUTE] [NORMAL_CLEARING]
2018-02-10 21:27:45.152820 [DEBUG] mod_hash.c:297 Usage for c00-pbx.somedomain.com_113 is now 0
2018-02-10 21:27:45.152820 [DEBUG] switch_ivr_play_say.c:1942 done playing file /var/lib/freeswitch/storage/voicemail/default/c00-pbx.somedomain.com/113/greeting_1.wav
send 538 bytes to udp/[174.37.181.39]:5060 at 21:27:45.162662:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 174.37.181.39;branch=z9hG4bK2375.b9d921e7f38da83d8c7dcaa0c03cab38.0
   Via: SIP/2.0/UDP 52.11.183.123:5060;branch=z9hG4bK3af11d1c;rport=5060
   From: "5555551212" <sip:5555551212 at somedomain.com>;tag=as54b5500e
   To: <sip:113 at 174.37.181.39:5060>;tag=Z3SgKX3H5Fc7a
   Call-ID: 4f43cc193246d9f07d56ccf80ecb9d09 at somedomain.com
   CSeq: 103 BYE
   User-Agent: FreeSWITCH
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
2018-02-10 21:27:45.152820 [NOTICE] switch_cpp.cpp:1365 [voicemail]
2018-02-10 21:27:45.152820 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7f6e24050320 released.
2018-02-10 21:27:45.152820 [DEBUG] switch_cpp.cpp:1112 sofia/external/5555551212 at somedomain.com destroy/unlink session from object
2018-02-10 21:27:45.152820 [DEBUG] switch_core_session.c:2815 sofia/external/5555551212 at somedomain.com skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2018-02-10 21:27:45.152820 [DEBUG] switch_core_state_machine.c:650 (sofia/external/5555551212 at somedomain.com) State EXECUTE going to sleep
2018-02-10 21:27:45.152820 [DEBUG] switch_core_state_machine.c:584 (sofia/external/5555551212 at somedomain.com) Running State Change CS_HANGUP (Cur 1 Tot 87)
2018-02-10 21:27:45.152820 [DEBUG] switch_core_state_machine.c:850 (sofia/external/5555551212 at somedomain.com) Callstate Change ACTIVE -> HANGUP
2018-02-10 21:27:45.152820 [DEBUG] switch_core_state_machine.c:852 (sofia/external/5555551212 at somedomain.com) State HANGUP
2018-02-10 21:27:45.152820 [DEBUG] mod_sofia.c:432 sofia/external/5555551212 at somedomain.com Overriding SIP cause 480 with 408 from the other leg
2018-02-10 21:27:45.152820 [DEBUG] mod_sofia.c:438 Channel sofia/external/5555551212 at somedomain.com hanging up, cause: NORMAL_CLEARING
2018-02-10 21:27:45.152820 [DEBUG] switch_core_state_machine.c:60 sofia/external/5555551212 at somedomain.com Standard HANGUP, cause: NORMAL_CLEARING
2018-02-10 21:27:45.152820 [DEBUG] switch_core_state_machine.c:852 (sofia/external/5555551212 at somedomain.com) State HANGUP going to sleep
2018-02-10 21:27:45.152820 [DEBUG] switch_core_state_machine.c:619 (sofia/external/5555551212 at somedomain.com) State Change CS_HANGUP -> CS_REPORTING
2018-02-10 21:27:45.152820 [DEBUG] switch_core_state_machine.c:584 (sofia/external/5555551212 at somedomain.com) Running State Change CS_REPORTING (Cur 1 Tot 87)
2018-02-10 21:27:45.152820 [DEBUG] switch_core_state_machine.c:938 (sofia/external/5555551212 at somedomain.com) State REPORTING
2018-02-10 21:27:45.172877 [DEBUG] switch_core_state_machine.c:174 sofia/external/5555551212 at somedomain.com Standard REPORTING, cause: NORMAL_CLEARING
2018-02-10 21:27:45.172877 [DEBUG] switch_core_state_machine.c:938 (sofia/external/5555551212 at somedomain.com) State REPORTING going to sleep
2018-02-10 21:27:45.172877 [DEBUG] switch_core_state_machine.c:610 (sofia/external/5555551212 at somedomain.com) State Change CS_REPORTING -> CS_DESTROY
2018-02-10 21:27:45.172877 [DEBUG] switch_core_session.c:1665 Session 84 (sofia/external/5555551212 at somedomain.com) Locked, Waiting on external entities
2018-02-10 21:27:45.172877 [NOTICE] switch_core_session.c:1683 Session 84 (sofia/external/5555551212 at somedomain.com) Ended
2018-02-10 21:27:45.172877 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/5555551212 at somedomain.com [CS_DESTROY]
2018-02-10 21:27:45.172877 [DEBUG] switch_core_state_machine.c:741 (sofia/external/5555551212 at somedomain.com) Running State Change CS_DESTROY (Cur 0 Tot 87)
2018-02-10 21:27:45.172877 [DEBUG] switch_core_state_machine.c:751 (sofia/external/5555551212 at somedomain.com) State DESTROY
2018-02-10 21:27:45.172877 [DEBUG] mod_sofia.c:343 sofia/external/5555551212 at somedomain.com SOFIA DESTROY
2018-02-10 21:27:45.172877 [DEBUG] switch_core_state_machine.c:181 sofia/external/5555551212 at somedomain.com Standard DESTROY
2018-02-10 21:27:45.172877 [DEBUG] switch_core_state_machine.c:751 (sofia/external/5555551212 at somedomain.com) State DESTROY going to sleep
freeswitch at c00-pbx.somedomain.com>

From gmina at connectfirst.com  Sun Feb 11 01:54:59 2018
From: gmina at connectfirst.com (Geoff Mina)
Date: Sun, 11 Feb 2018 01:54:59 +0000
Subject: [Freeswitch-users] Locked, Waiting on external entities
In-Reply-To: <CAKzROs2ZKLu=xFRPBNn8iyeLrnp51iTHBMJTSzL3WBzHGnhdoQ@mail.gmail.com>
References: <CAKzROs2ZKLu=xFRPBNn8iyeLrnp51iTHBMJTSzL3WBzHGnhdoQ@mail.gmail.com>
Message-ID: <CAKzROs0ewV=0EWBtNX+mwjAx0YUenauwFuX3yyuvCrQHqemz4A@mail.gmail.com>

Pretty sure it was a LUA issue. I scrapped the instance and rebuilt it and
the issue went away.

I will just chock it up to FusionPBX config fragility.

On Sat, Feb 10, 2018 at 2:43 PM Geoff Mina <gmina at connectfirst.com> wrote:

> Greetings,
> I was wondering if anyone knows what the "Locked, Waiting on external
> entities" message indicates.
>
> First, I am using a FusionPBX GUI to play with some basic PBX
> functionality.  The flow of my call is an XML transfer from a public to a
> private context.  Enabling trusted outside party to dial extensions
> directly on the Freeswitch host.
>
> The call comes in on the external SIP profile and transfers to the
> internal profile.  The softphone rings and I hit "decline" which sends back
> a 486 Busy Here to FS.  Freeswitch then hangs in a "Waiting on external
> entities" state until a timeout occurs at which point the inbound call is
> finally sent to voicemail after an internal 408 timeout.
>
> This is FS 1.6.20 installed from Yum on a CentOS7 box.  Attached is the
> full log file with SIP tracing enabled.  Here is the relevant snippet where
> the switch is just sitting for 16 seconds doing nothing.
>
> 2018-02-10 21:27:11.132820 [DEBUG] switch_core_state_machine.c:610
> (sofia/internal/113 at 161.97.193.191:51392) State Change CS_REPORTING ->
> CS_DESTROY
> *2018-02-10 21:27:11.132820 [DEBUG] switch_core_session.c:1665 Session 87
> (sofia/internal/113 at 161.97.193.191:51392 <http://113 at 161.97.193.191:51392>)
> Locked, Waiting on external entities*
> *2018-02-10 21:27:40.952819 [DEBUG] sofia.c:7084 Channel
> sofia/internal/113 at 161.97.193.191:51094 <http://113 at 161.97.193.191:51094>
> entering state [terminated][408]*
> 2018-02-10 21:27:40.952819 [NOTICE] sofia.c:8273 Hangup sofia/internal/
> 113 at 161.97.193.191:51094 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE]
>
> Just a point in the right direction as to what may cause this would be
> appreciated.
>
> Thanks,
> Geoff
>
-- 


*GEOFF MINA*Chief Executive Officer
Connect First / Contact Center Solutions, Built Better.

3101 Iris Ave #200, Boulder, CO 80301
720.335.5924 <javascript:void(0);>
gmina at connectfirst.com / www.connectfirst.com <http://connectfirst.com/>

 [image:
https://docs.google.com/uc?export=download&id=0B5b6KnVfm9lJTlFrQzRVUjJ2ZVE&revid=0B5b6KnVfm9lJUXpUMTFEbGJvaktwN1p5ejM3YTFkdWVWNzBzPQ]

 This email and any files transmitted with it are confidential and intended
solely for the use of the individual or entity to whom they are addressed.
If you have received this email in error please notify the system manager.
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From manpower13.cse at gmail.com  Sun Feb 11 15:22:35 2018
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Sun, 11 Feb 2018 20:52:35 +0530
Subject: [Freeswitch-users] Mongo CDR issue with Limit
Message-ID: <CAMRuXGiUe0xYZY6X_Apcr511rXpRGcCtHdBphvLvg-x2Krpc2Q@mail.gmail.com>

HI,

When i use limit in my dialplan i am getting following error  [ERR]
mod_cdr_mongodb.c:413 mongo_insert: (error code 13) , Look like cdr insert
fails to mongodb.


Hope it issue when i use LIMIT ,if i remove limit then its work fine
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From italo at freeswitch.org  Mon Feb 12 03:16:05 2018
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Mon, 12 Feb 2018 03:16:05 +0000
Subject: [Freeswitch-users] Outbound Socket isn't killing each pid
In-Reply-To: <F5FDDBB3-35B0-4128-8082-EA50E3203D78@gmail.com>
References: <F5FDDBB3-35B0-4128-8082-EA50E3203D78@gmail.com>
Message-ID: <CABMX_w8Gqb=DqSSbzbqZ=hW5d=c2hPQ0ZW_MxJbdiATJ04mV9g@mail.gmail.com>

Reports bugs to jira not the mailing list. Also, you did not specify your
freeswitch version.
Em qua, 7 de fev de 2018 às 09:53, Raimundo Pérez Nieves <
raimundo.perez.cuba at gmail.com> escreveu:

> Hi guys,
> Iam running outbound socket in a server, and It start perfect and receive
> 265 calls, which mean 265 pid created by outbound socket.
> The problem is in that exactly moment, the server get memory ram out and
> get freezes.
> I can see each pid of outbound socket there:
>
> Feb  6 21:14:57 test-server kernel: [97024.382843] [16701]     0 16701
> 10032      846      25        0             0 outbound_socket
> Feb  6 21:14:57 test-server kernel: [97024.382845] [16704]     0 16704
> 26451     2463      27        0             0 outbound_socket
> Feb  6 21:14:57 test-server kernel: [97024.382846] [16709]     0 16709
> 10094      898      23        0             0 outbound_socket
> Feb  6 21:14:57 test-server kernel: [97024.382848] [16711]     0 16711
> 11686     2498      24        0             0 outbound_socket
> Feb  6 21:14:57 test-server kernel: [97024.382849] [16715]     0 16715
> 10067      882      23        0             0 outbound_socket
> Feb  6 21:14:57 test-server kernel: [97024.382851] [16720]     0 16720
> 11686     2498      24        0             0 outbound_socket
> Feb  6 21:14:57 test-server kernel: [97024.382852] [16725]     0 16725
> 11694     2498      24        0             0 outbound_socket
> Feb  6 21:14:57 test-server kernel: [97024.382854] [16730]     0 16730
> 11686     2498      24        0             0 outbound_socket
> Feb  6 21:14:57 test-server kernel: [97024.382855] [16736]     0 16736
> 10094      898      23        0             0 outbound_socket
> Feb  6 21:14:57 test-server kernel: [97024.382857] [16738]     0 16738
> 11688     2498      24        0             0 outbound_socket
> Feb  6 21:14:57 test-server kernel: [97024.382858] [16743]     0 16743
> 11686     2486      24        0             0 outbound_socket
> Feb  6 21:14:57 test-server kernel: [97024.382860] [16748]     0 16748
> 11686     2498      24        0             0 outbound_socket
> Feb  6 21:14:57 test-server kernel: [97024.382861] [16754]     0 16754
> 11694     2488      24        0             0 outbound_socket
> Feb  6 21:14:57 test-server kernel: [97024.382863] [16756]     0 16756
> 11694     2498      24        0             0 outbound_socket
> Feb  6 21:14:57 test-server kernel: [97024.382864] [16761]     0 16761
> 11686     2498      24        0             0 outbound_socket
> Feb  6 21:14:57 test-server kernel: [97024.382866] [16766]     0 16766
> 10085      898      23        0             0 outbound_socket
> Feb  6 21:14:57 test-server kernel: [97024.382867] [16771]     0 16771
> 11688     2489      24        0             0 outbound_socket
> Feb  6 21:14:57 test-server kernel: [97024.382868] [16775]     0 16775
> 11694     2498      24        0             0 outbound_socket
> Feb  6 21:14:57 test-server kernel: [97024.382870] [16777]     0 16777
> 11686     2498      24        0             0 outbound_socket
> And continues……
>
> At the end:
> *Feb  6 21:14:57 *test-server* kernel: [97024.383243] Out of memory: Kill
> process 14314 (freeswitch) score 28 or sacrifice child*
> Feb  6 21:14:57 test-server kernel: [97024.384424] Killed process 14314
> (freeswitch) total-vm:1282524kB, anon-rss:58856kB, file-rss:0kB
>
>
> I use this code.
>
> require ESL;
> use IO::Socket::INET;
>
> my $ip = "127.0.0.1";
> my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8083',
> Proto => 'tcp', Listen => 1, Reuse => 1 );
> die "Could not create socket: $!\n" unless $sock;
> for(;;) {
> my $new_sock = $sock->accept();
> my $pid = fork();
> if ($pid) {
>   print "New child pid $pid created...\n";
> close($new_sock);
> next;
> }
> my $fd = fileno($new_sock);
> my $con = new ESL::ESLconnection($fd);
> my $info = $con->getInfo();
> my $uuidLegB = $info->getHeader("unique-id");
>  #This $uuidLegA is a temp variable where I store uuid leg A for post
> bridge
> my $uuidLegA = $info->getHeader("variable_outbound_caller_id_name");
> my $joinUUID = 'resp'.$uuidLegA.'|'.$uuidLegB.' XML default';
> print("UUID Leg A is $uuidLegA and B is $uuidLegB\n");
> $con->events("plain","all");
> $con->execute("spandsp_start_tone_detect","34");
> my $connectedSession = 1;
> while($con->connected() & $connectedSession == 1) {
> my $e = $con->recvEvent();
> my $ev_name = $e->getHeader("Event-Name");
> print("$ev_name\n");
> if($ev_name eq 'DETECTED_TONE'){
> my $tone_name = $e->getHeader("Detected-Tone");
> print "DETECTED_TONE [$tone_name]\n";
> if ($tone_name eq 'RING_TONE') {
> $command = 'spandsp_stop_tone_detect '.$uuidLegB;
> $con->api($command);
> $con->execute("transfer",$joinUUID);
> $connectedSession = 0;
> }
> }
> }
> print "BYE\n";
> close($new_sock);
> }
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 alex at freeswitch.com  Mon Feb 12 12:26:37 2018
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Mon, 12 Feb 2018 21:26:37 +0900
Subject: [Freeswitch-users] test mail
Message-ID: <CAGJLwFgQ1xZXOirSu67hvxRT=X4aqS=6br4u5ssQ-Pn2_8ushw@mail.gmail.com>

Just a test, please ignore it

-- 

Alex Sibyakin | Support Engineer
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: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.
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From krice at freeswitch.org  Mon Feb 12 12:31:55 2018
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 12 Feb 2018 06:31:55 -0600
Subject: [Freeswitch-users] test
Message-ID: <11a301d3a3fd$78a72670$69f57350$@freeswitch.org>

Test 1234

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From gmaruzz at gmail.com  Mon Feb 12 14:03:16 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 12 Feb 2018 15:03:16 +0100
Subject: [Freeswitch-users] test
In-Reply-To: <11a301d3a3fd$78a72670$69f57350$@freeswitch.org>
References: <11a301d3a3fd$78a72670$69f57350$@freeswitch.org>
Message-ID: <CALXCt0osryX3900gYvhyjFfc3zYsjnyAVrRy04impZ_p-9uGAw@mail.gmail.com>

pong


On 12 February 2018 at 13:31, Ken Rice <krice at freeswitch.org> wrote:

> Test 1234
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From gmina at connectfirst.com  Mon Feb 12 14:08:41 2018
From: gmina at connectfirst.com (Geoff Mina)
Date: Mon, 12 Feb 2018 07:08:41 -0700
Subject: [Freeswitch-users] WSS - DTLS handshake takes about 6s in
	firefox
In-Reply-To: <9A71A4FA-CA49-40B3-87E6-367ECDC5B3AE@gmx.net>
References: <9A71A4FA-CA49-40B3-87E6-367ECDC5B3AE@gmx.net>
Message-ID: <B2F1185F-7B00-4527-8CE7-A1ACB8CDFBD6@connectfirst.com>

Can you get a Developer Tool console log from Firefox which includes the SIP?

FireFox has a 5 second timeout on ICE candidate gathering.  

I’m wondering if there is a failure is on the ICE side and FF is returning too many srflx records that need to be tried. 

> On Feb 9, 2018, at 4:41 AM, Markus Bönke <mbodbg at gmx.net> wrote:
> 
> 
> We're using an application which uses the JSSip library to connect to Freeswitch 1.6.19 to make and receive calls. On Chome, everything is fine - when a call get's connected we can hear the caller from the beginning. However on Firefox we can not hear the caller for the first 5 Seconds. 
> 
> If I look in the logs, it seems the DTLS handshake takes more than 5 seconds between Firefox and Freeswitch:
> 
> Firefox 57.0.4 on OSX 10.13.3
> 
> 2018-02-09 11:38:05.613120 [INFO] switch_rtp.c:3581 Activate RTP/RTCP audio DTLS server
> 2018-02-09 11:38:05.613120 [INFO] switch_rtp.c:3730 Changing audio DTLS state from OFF to HANDSHAKE
> 2018-02-09 11:38:11.833115 [INFO] switch_rtp.c:3192 Changing audio DTLS state from HANDSHAKE to SETUP <---
> 2018-02-09 11:38:11.853115 [INFO] switch_rtp.c:3141 Changing audio DTLS state from SETUP to READY
> 
> Chrome 62.0.3202.89 on OSX 10.13.3
> 
> 2018-02-09 11:45:30.233115 [INFO] switch_rtp.c:3581 Activate RTP/RTCP audio DTLS server
> 2018-02-09 11:45:30.233115 [INFO] switch_rtp.c:3730 Changing audio DTLS state from OFF to HANDSHAKE
> 2018-02-09 11:45:30.533125 [INFO] switch_rtp.c:3192 Changing audio DTLS state from HANDSHAKE to SETUP
> 2018-02-09 11:45:30.553117 [INFO] switch_rtp.c:3141 Changing audio DTLS state from SETUP to READY
> 
> 
> Is this a general issue in firefox or can it somehow be optimized by changing the freeswitch configuration?
> 
> Thanks and regards
> 
> Markus
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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


From gmaruzz at gmail.com  Mon Feb 12 14:12:05 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 12 Feb 2018 15:12:05 +0100
Subject: [Freeswitch-users] how to make a dial plane
In-Reply-To: <CADSVerC2+gsdzYnTVGd-EuaMtvQNjO-kN5pCG=DdwDiq2srJ3Q@mail.gmail.com>
References: <CAB2bS9Qt5jD6EZJ2iwmKRRxfD-7qgvWhC6udJJ1sZTSizkqmpA@mail.gmail.com>
 <CALXCt0qNT50G6h4AygMBifK=aPEYOuMhAFyQOPB9rUe4xPnZFw@mail.gmail.com>
 <4c43e321-d958-54ec-4dc4-59507bfc6e1a@madovsky.org>
 <CADSVerC2+gsdzYnTVGd-EuaMtvQNjO-kN5pCG=DdwDiq2srJ3Q@mail.gmail.com>
Message-ID: <CALXCt0qoEMXULQq+cgPSQHf4rmRPr3z1VubzCpyc9P+KPs688Q@mail.gmail.com>

Bilal, more power to you, because your patience seems (to me) supernatural.

It is obvious (to me) you study before asking someone else, because you
value someone else's time, and you don't want to shift the effort from your
own shoulders to the shoulders of someone else.

Also I am (personally, my opinion) totally against guiding others in doing
something.

There is documentation for this, and documentation can be used by million
people, while only a few dozen people can be guided (and guiding them will
do very little good to them).

So, I (personally) believe that is a waste of time and a damage to guide
someone, and that time and energy is better spent in the betterment of
documentation.

-giovanni

On 8 February 2018 at 07:11, Bilal Abbasi <bilaln018 at gmail.com> wrote:

> @giovanni,
> Yeah, you know what i did? i took his personal email and i spend few hours
> guiding him through the installation, just to make user-list clean.
> Ali please read the documentations and come up with real questions.
>
> Regards
> Abbasi
>
> On Wed, Feb 7, 2018 at 6:58 PM, Madovsky <infos at madovsky.org> wrote:
>
>> This emailist is to talk and support any problem of installing, updating
>> and so on.
>>
>> to learn how to use FS just read the documentation but please don't
>> bother the whole emailist subscribers with
>>
>> your unconsistent email thanks.
>>
>>
>> On 2/7/2018 5:50 AM, Giovanni Maruzzelli wrote:
>>
>> Ali,
>>
>> why, why, why, you send a mail like this?
>>
>> Study the documentation, use google, and when you have a real question
>> write to the mailing list.
>>
>> Is not nice you push us to ignore you, or to answer you something like
>> "your questions is too much general, blah blah blah".
>>
>> -giovanni
>>
>>
>> On 7 February 2018 at 14:42, Ali Haider <alihaider.4189 at gmail.com> wrote:
>>
>>> Hiiii users
>>>              I want to know how to make a dialplan for pc
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>>
>>
>>
>>
>> --
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From alexandr.popov at iqoption.com  Mon Feb 12 14:29:34 2018
From: alexandr.popov at iqoption.com (Alexandr Popov)
Date: Mon, 12 Feb 2018 16:29:34 +0200
Subject: [Freeswitch-users] WSS - DTLS handshake takes about 6s in
	firefox
In-Reply-To: <B2F1185F-7B00-4527-8CE7-A1ACB8CDFBD6@connectfirst.com>
References: <9A71A4FA-CA49-40B3-87E6-367ECDC5B3AE@gmx.net>
 <B2F1185F-7B00-4527-8CE7-A1ACB8CDFBD6@connectfirst.com>
Message-ID: <CAKeZRhun4KXq__iQxufdGB2_MeQBB2R-SMCeHaxV45_VrQ2P-w@mail.gmail.com>

JsSip issue - you can reduce ice gathering -- info you can get here

https://github.com/versatica/JsSIP/issues/432

2018-02-12 16:08 GMT+02:00 Geoff Mina <gmina at connectfirst.com>:

> Can you get a Developer Tool console log from Firefox which includes the
> SIP?
>
> FireFox has a 5 second timeout on ICE candidate gathering.
>
> I’m wondering if there is a failure is on the ICE side and FF is returning
> too many srflx records that need to be tried.
>
> > On Feb 9, 2018, at 4:41 AM, Markus Bönke <mbodbg at gmx.net> wrote:
> >
> >
> > We're using an application which uses the JSSip library to connect to
> Freeswitch 1.6.19 to make and receive calls. On Chome, everything is fine -
> when a call get's connected we can hear the caller from the beginning.
> However on Firefox we can not hear the caller for the first 5 Seconds.
> >
> > If I look in the logs, it seems the DTLS handshake takes more than 5
> seconds between Firefox and Freeswitch:
> >
> > Firefox 57.0.4 on OSX 10.13.3
> >
> > 2018-02-09 11:38:05.613120 [INFO] switch_rtp.c:3581 Activate RTP/RTCP
> audio DTLS server
> > 2018-02-09 11:38:05.613120 [INFO] switch_rtp.c:3730 Changing audio DTLS
> state from OFF to HANDSHAKE
> > 2018-02-09 11:38:11.833115 [INFO] switch_rtp.c:3192 Changing audio DTLS
> state from HANDSHAKE to SETUP <---
> > 2018-02-09 11:38:11.853115 [INFO] switch_rtp.c:3141 Changing audio DTLS
> state from SETUP to READY
> >
> > Chrome 62.0.3202.89 on OSX 10.13.3
> >
> > 2018-02-09 11:45:30.233115 [INFO] switch_rtp.c:3581 Activate RTP/RTCP
> audio DTLS server
> > 2018-02-09 11:45:30.233115 [INFO] switch_rtp.c:3730 Changing audio DTLS
> state from OFF to HANDSHAKE
> > 2018-02-09 11:45:30.533125 [INFO] switch_rtp.c:3192 Changing audio DTLS
> state from HANDSHAKE to SETUP
> > 2018-02-09 11:45:30.553117 [INFO] switch_rtp.c:3141 Changing audio DTLS
> state from SETUP to READY
> >
> >
> > Is this a general issue in firefox or can it somehow be optimized by
> changing the freeswitch configuration?
> >
> > Thanks and regards
> >
> > Markus
> > ____________________________________________________________
> _____________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 brian at freeswitch.com  Mon Feb 12 15:53:49 2018
From: brian at freeswitch.com (Brian West)
Date: Mon, 12 Feb 2018 15:53:49 +0000
Subject: [Freeswitch-users] test
In-Reply-To: <CALXCt0osryX3900gYvhyjFfc3zYsjnyAVrRy04impZ_p-9uGAw@mail.gmail.com>
References: <11a301d3a3fd$78a72670$69f57350$@freeswitch.org>
 <CALXCt0osryX3900gYvhyjFfc3zYsjnyAVrRy04impZ_p-9uGAw@mail.gmail.com>
Message-ID: <CAEJMVkD4jPtwvh2BRf7Bg4ARA3riYZAvKZ4+e0BCt-zYrzzWfw@mail.gmail.com>

Test 4321

On Mon, Feb 12, 2018 at 8:20 AM Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> pong
>
>
> On 12 February 2018 at 13:31, Ken Rice <krice at freeswitch.org> wrote:
>
>> Test 1234
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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

-- 

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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From mirkobrankovic at gmail.com  Mon Feb 12 16:19:27 2018
From: mirkobrankovic at gmail.com (Mirko Brankovic)
Date: Mon, 12 Feb 2018 17:19:27 +0100
Subject: [Freeswitch-users] test
In-Reply-To: <CAEJMVkD4jPtwvh2BRf7Bg4ARA3riYZAvKZ4+e0BCt-zYrzzWfw@mail.gmail.com>
References: <11a301d3a3fd$78a72670$69f57350$@freeswitch.org>
 <CALXCt0osryX3900gYvhyjFfc3zYsjnyAVrRy04impZ_p-9uGAw@mail.gmail.com>
 <CAEJMVkD4jPtwvh2BRf7Bg4ARA3riYZAvKZ4+e0BCt-zYrzzWfw@mail.gmail.com>
Message-ID: <CAND18T2Fmdrr5d_cQ+E4hBCjgt0nH0wAhpwEOBEFidj6Jw0UKA@mail.gmail.com>

works  4321 :)

On Mon, Feb 12, 2018 at 4:53 PM, Brian West <brian at freeswitch.com> wrote:

> Test 4321
>
> On Mon, Feb 12, 2018 at 8:20 AM Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> pong
>>
>>
>> On 12 February 2018 at 13:31, Ken Rice <krice at freeswitch.org> wrote:
>>
>>> Test 1234
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>>
>>
>>
>>
>> --
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>
> --
>
> 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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
> color-twitter-96.png]
> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 
Regards,
Mirko
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From s.safarov at gmail.com  Mon Feb 12 17:14:49 2018
From: s.safarov at gmail.com (Sergey Safarov)
Date: Mon, 12 Feb 2018 17:14:49 +0000
Subject: [Freeswitch-users] "received=" in path header
In-Reply-To: <CAJoKKdXk-Y_p8_3ADLu89FqYX52garpZPaqOa3vFfDWbhtnNww@mail.gmail.com>
References: <CAJoKKdXk-Y_p8_3ADLu89FqYX52garpZPaqOa3vFfDWbhtnNww@mail.gmail.com>
Message-ID: <CAHtxdDcDP_3nbCAtMiunOcJB8ZPcNQJroF4-=0VPpxy4q9LLmw@mail.gmail.com>

Need to get registered user contact string FS api.
Using this contact string you can dial webrtc user

пн, 12 февр. 2018 г., 16:54 Tihomir Culjaga <tculjaga at gmail.com>:

> hi,
>
>
> i have  UA > Kamailio > Freeswitch
>
> UA is using sip for websockets and registers via kamailio to freeswitch.
> When i try to originate a call from FS to the registered endpoint, the
> call fails.
>
> freeswitch at FS01> bgapi originate
> {origination_caller_id_number=1002}sofia/internal/1001%mydomain &echo()
>
> I think FS is trying to use transport from received parematar instead of
> path uri.
>
> any advice how to handle this ?
>
>
>
>
>
> recv 964 bytes from udp/[192.168.50.60]:5060 at 22:18:57.909700:
>    ------------------------------------------------------------------------
>    REGISTER sip:192.168.50.60 SIP/2.0
>    Via: SIP/2.0/UDP
> 192.168.50.60;branch=z9hG4bKe929.b6aa43dd1eefa9e4756af7d31d65066e.0
>    Via: SIP/2.0/WSS
> 192.0.2.110;rport=61744;received=192.168.200.77;branch=z9hG4bK3179897
>    Max-Forwards: 69
>    To: <sip:1001 at 192.168.50.60>
>    From: <sip:1001 at 192.168.50.60>;tag=vq30modpgd
>    Call-ID: dlnrna9o4ngn25fb9d1vi2
>    CSeq: 192 REGISTER
>    Authorization: Digest algorithm=MD5, username="1001",
> realm="192.168.50.60", nonce="22c52b4b-f795-4caa-bb9b-15bbd87564f7",
> uri="sip:192.168.50.60", response="a073277bc9c1fcd681de661cb418d838",
> qop=auth, cnonce="euqeokf2d1gr", nc=00000001
>    Contact: <sip:1001 at 192.0.2.110
> ;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:587f4694-7e55-46dc-a858-579a82556a20>";expires=600
>    Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
>    Supported: path, gruu, 100rel, outbound
>    User-Agent: SIP.js/0.7.0 BB
>    Content-Length: 0
>    Path: <sip:192.168.50.60;lr;received=sip:192.168.200.77:61744
> %3Btransport%3Dws>
>
>    ------------------------------------------------------------------------
> send 738 bytes to udp/[192.168.50.60]:5060 at 22:18:57.950968:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/UDP
> 192.168.50.60;branch=z9hG4bKe929.b6aa43dd1eefa9e4756af7d31d65066e.0
>    Via: SIP/2.0/WSS
> 192.0.2.110;rport=61744;received=192.168.200.77;branch=z9hG4bK3179897
>    From: <sip:1001 at 192.168.50.60>;tag=vq30modpgd
>    To: <sip:1001 at 192.168.50.60>;tag=eK8SmZmcU7UBr
>    Call-ID: dlnrna9o4ngn25fb9d1vi2
>    CSeq: 192 REGISTER
>    Contact: <sip:1001 at 192.0.2.110;transport=ws>;expires=600
>    Date: Fri, 09 Feb 2018 21:18:57 GMT
>    User-Agent:
> FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY
>    Supported: timer, path, replaces
>    Path: <sip:192.168.50.60>;lr;received=sip:192.168.200.77:61744
> %3Btransport%3Dws
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
>
>
>
> in freeswitch i the registration:
>
> Call-ID:        dlnrna9o4ngn25fb9d1vi2
> User:           1001 at mydomain
> Contact:        "" <sip:1001 at 192.0.2.110
> ;transport=ws;fs_nat=yes;fs_path=sip%3A192.168.50.60%3Blr%3Breceived%3Dsip%3A192.168.200.77%3A61744%3Btransport%3Dws>
> Agent:          SIP.js/0.7.0 BB
> Status:         Registered(WS-NAT)(unknown) EXP(2018-02-09 22:29:57)
> EXPSECS(510)
> Ping-Status:    Reachable
> Ping-Time:      0.00
> Host:           FS01
> IP:             192.168.50.60
> Port:           5060
> Auth-User:      1001
> Auth-Realm:     192.168.50.60
> MWI-Account:    1001 at mydomain
>
>
>
>
> the register contain a path header with received=<original uri>.
>
> when i try to originate a call to this registered user the call goes
> nowhere :=)
>
>
> freeswitch at FS01> bgapi originate
> {origination_caller_id_number=1002}sofia/internal/1001%mydomain &echo()
> +OK Job-UUID: 200e2197-a6de-42d9-b4bb-63917223bc53
>
> 2018-02-09 22:25:40.500100 [DEBUG] switch_ivr_originate.c:2142 Parsing
> global variables
> 2018-02-09 22:25:40.500100 [NOTICE] switch_channel.c:1104 New Channel
> sofia/internal/1001 [327826f9-8a8c-46c7-b3ed-8bf22852df88]
> 2018-02-09 22:25:40.500100 [DEBUG] mod_sofia.c:4819 (sofia/internal/1001)
> State Change CS_NEW -> CS_INIT
> 2018-02-09 22:25:40.500100 [DEBUG] switch_core_state_machine.c:584
> (sofia/internal/1001) Running State Change CS_INIT (Cur 1 Tot 217)
> 2018-02-09 22:25:40.500100 [DEBUG] switch_core_state_machine.c:627
> (sofia/internal/1001) State INIT
> 2018-02-09 22:25:40.500100 [DEBUG] mod_sofia.c:90 sofia/internal/1001
> SOFIA INIT
> 2018-02-09 22:25:40.500100 [DEBUG] sofia_glue.c:1264
> sip:192.168.50.60;lr;received=sip:192.168.200.77:61744;transport=ws
> Setting proxy route to sofia/internal/1001
> 2018-02-09 22:25:40.500100 [DEBUG] sofia_glue.c:1295 sofia/internal/1001
> sending invite version: 1.6.19 git b1b21d0 2017-11-20 16:34:16Z 64bit
>
>
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 lpopov at blasterphone.com  Mon Feb 12 18:07:44 2018
From: lpopov at blasterphone.com (Lyubo Popov)
Date: Mon, 12 Feb 2018 16:07:44 -0200
Subject: [Freeswitch-users] Set b-leg CallerID using from in a 300
 Multiple Choices reply
In-Reply-To: <B5365ACE-09C6-4821-9BBC-D42C2DE24EC3@tm.net.uk>
References: <B5365ACE-09C6-4821-9BBC-D42C2DE24EC3@tm.net.uk>
Message-ID: <CAMSNS4hVsugSpqcHkYLpcZuamdsdywqDLkHxVB0h9xs+uQXCjA@mail.gmail.com>

Hello Joseph,

I am having the same problem and would like to resurrect this post. Does
someone knows how to do that? I send one CallerID to redirect server where
this is being rewritten and receive another from it in the 300 response and
would like to update this CID before sending to termination (b-leg)

Re.,
L.Popov

<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon>
Virus-free.
www.avast.com
<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link>
<#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

On Tue, Jul 11, 2017 at 4:32 PM, Joseph Waite <joelists at tm.net.uk> wrote:

> Hi Guys
>
> I am trying to use the From field in a 300 Multiple Choices message to set
> the outbound CallerID in the re-directed b-leg
>
> So call comes into FreeSwitch, gets sent out to a SIP Redirect server
> which replies with a 300 Multiple choices.
> I need to pull the number from the From field of this 300 Response and use
> it to set the From field, raid & Paid fields in the re-directed invite.
>
> Any help would be very much appreciated.
>
> Regards
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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




-- 
Atenciosamente,
============================
Lyubo Popov
CEO - BlasterPhone LLC
Tel: 4003-1556 ( Outside Brazil 55 11 4003-1556)
iNum: +883 510001-354111
Website: http://www.blastervoip.com.br/
============================
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From joelists at tm.net.uk  Mon Feb 12 18:20:45 2018
From: joelists at tm.net.uk (Jospeh Waite)
Date: Mon, 12 Feb 2018 18:20:45 +0000
Subject: [Freeswitch-users] Set b-leg CallerID using from in a 300
	Multiple Choices reply
In-Reply-To: <CAMSNS4hVsugSpqcHkYLpcZuamdsdywqDLkHxVB0h9xs+uQXCjA@mail.gmail.com>
References: <B5365ACE-09C6-4821-9BBC-D42C2DE24EC3@tm.net.uk>
 <CAMSNS4hVsugSpqcHkYLpcZuamdsdywqDLkHxVB0h9xs+uQXCjA@mail.gmail.com>
Message-ID: <A80E76FB-5C40-4AC5-BE0D-F04A7B20D4CA@tm.net.uk>

Hi L.Popov

Now this would depend in what format your redirect server is sending the reply back. In my case I’m using Jerasoft VCS which is sending the re-written caller id in a per contact form with src_number= set.

I use the following in my redirect catching dial plan

       <condition field="${sip_redirect_contact_0}" expression="src_number=(.+)\;(.+)$">
            <action application="set" data="effective_caller_id_name=$1"/>
            <action application="set" data="effective_caller_id_number=$1"/>
        </condition>

        <condition field="${sip_redirect_contact_0}" expression="^<(.+)\>(.+)$">
            <action application="bridge" data="{sip_invite_from_uri=sip:${effective_caller_id_number}@${sip_network_ip}}sofia/external/$1" />
        </condition>

This catches the src_number in the first condition and updates the caller_id and then bridges the call in the second.

The one drawback is you would have to replicate this for each response your likely to have in the 300 response. So if it gives 3 different options to send the call to you would have to copy this 3 times, changing the sip_redirect_contact_0 and increasing the number on the end for each one.

Regards

> On 12 Feb 2018, at 18:07, Lyubo Popov <lpopov at blasterphone.com> wrote:
> 
> Hello Joseph,
> 
> I am having the same problem and would like to resurrect this post. Does someone knows how to do that? I send one CallerID to redirect server where this is being rewritten and receive another from it in the 300 response and would like to update this CID before sending to termination (b-leg)
> 
> Re.,
> L.Popov
> 
>  <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon>	Virus-free. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link> <x-msg://8/#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
> 
> On Tue, Jul 11, 2017 at 4:32 PM, Joseph Waite <joelists at tm.net.uk <mailto:joelists at tm.net.uk>> wrote:
> Hi Guys
> 
> I am trying to use the From field in a 300 Multiple Choices message to set the outbound CallerID in the re-directed b-leg
> 
> So call comes into FreeSwitch, gets sent out to a SIP Redirect server which replies with a 300 Multiple choices.
> I need to pull the number from the From field of this 300 Response and use it to set the From field, raid & Paid fields in the re-directed invite.
> 
> Any help would be very much appreciated.
> 
> Regards
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.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>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> -- 
> Atenciosamente,
> ============================
> Lyubo Popov
> CEO - BlasterPhone LLC
> Tel: 4003-1556 ( Outside Brazil 55 11 4003-1556)
> iNum: +883 510001-354111
> Website: http://www.blastervoip.com.br/ <http://www.blastervoip.com.br/>
> ============================
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 lpopov at blasterphone.com  Mon Feb 12 18:31:50 2018
From: lpopov at blasterphone.com (Lyubo Popov)
Date: Mon, 12 Feb 2018 16:31:50 -0200
Subject: [Freeswitch-users] Set b-leg CallerID using from in a 300
 Multiple Choices reply
In-Reply-To: <A80E76FB-5C40-4AC5-BE0D-F04A7B20D4CA@tm.net.uk>
References: <B5365ACE-09C6-4821-9BBC-D42C2DE24EC3@tm.net.uk>
 <CAMSNS4hVsugSpqcHkYLpcZuamdsdywqDLkHxVB0h9xs+uQXCjA@mail.gmail.com>
 <A80E76FB-5C40-4AC5-BE0D-F04A7B20D4CA@tm.net.uk>
Message-ID: <CAMSNS4g=w2P6mDV5OxP867OTwP8Meynsd86=4opn7uCTJekbmQ@mail.gmail.com>

I am also using Jerasoft so I am going to try this out on my setup. Great
help! Thank you!

Re.,
L.Popov

<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon>
Virus-free.
www.avast.com
<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link>
<#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

On Mon, Feb 12, 2018 at 4:20 PM, Jospeh Waite <joelists at tm.net.uk> wrote:

> Hi L.Popov
>
> Now this would depend in what format your redirect server is sending the
> reply back. In my case I’m using Jerasoft VCS which is sending the
> re-written caller id in a per contact form with src_number= set.
>
> I use the following in my redirect catching dial plan
>
>        <condition field="${sip_redirect_contact_0}"
> expression="src_number=(.+)\;(.+)$">
>             <action application="set" data="effective_caller_id_name=$1"/>
>             <action application="set" data="effective_caller_id_
> number=$1"/>
>         </condition>
>
>         <condition field="${sip_redirect_contact_0}"
> expression="^<(.+)\>(.+)$">
>             <action application="bridge" data="{sip_invite_from_uri=
> sip:${effective_caller_id_number}@${sip_network_ip}}sofia/external/$1" />
>         </condition>
>
> This catches the src_number in the first condition and updates the
> caller_id and then bridges the call in the second.
>
> The one drawback is you would have to replicate this for each response
> your likely to have in the 300 response. So if it gives 3 different options
> to send the call to you would have to copy this 3 times, changing the
> sip_redirect_contact_0 and increasing the number on the end for each one.
>
> Regards
>
> On 12 Feb 2018, at 18:07, Lyubo Popov <lpopov at blasterphone.com> wrote:
>
> Hello Joseph,
>
> I am having the same problem and would like to resurrect this post. Does
> someone knows how to do that? I send one CallerID to redirect server where
> this is being rewritten and receive another from it in the 300 response and
> would like to update this CID before sending to termination (b-leg)
>
> Re.,
> L.Popov
>
>
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon> Virus-free.
> www.avast.com
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link>
>
> On Tue, Jul 11, 2017 at 4:32 PM, Joseph Waite <joelists at tm.net.uk> wrote:
>
>> Hi Guys
>>
>> I am trying to use the From field in a 300 Multiple Choices message to
>> set the outbound CallerID in the re-directed b-leg
>>
>> So call comes into FreeSwitch, gets sent out to a SIP Redirect server
>> which replies with a 300 Multiple choices.
>> I need to pull the number from the From field of this 300 Response and
>> use it to set the From field, raid & Paid fields in the re-directed invite.
>>
>> Any help would be very much appreciated.
>>
>> Regards
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>
>
>
>
> --
> Atenciosamente,
> ============================
> Lyubo Popov
> CEO - BlasterPhone LLC
> Tel: 4003-1556 ( Outside Brazil 55 11 4003-1556)
> iNum: +883 510001-354111 <+883%20510%20001%20354%20111>
> Website: http://www.blastervoip.com.br/
> ============================
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 
Atenciosamente,
============================
Lyubo Popov
CEO - BlasterPhone LLC
Tel: 4003-1556 ( Outside Brazil 55 11 4003-1556)
iNum: +883 510001-354111
Website: http://www.blastervoip.com.br/
============================
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From joelists at tm.net.uk  Mon Feb 12 18:39:00 2018
From: joelists at tm.net.uk (Jospeh Waite)
Date: Mon, 12 Feb 2018 18:39:00 +0000
Subject: [Freeswitch-users] Set b-leg CallerID using from in a 300
	Multiple Choices reply
In-Reply-To: <CAMSNS4g=w2P6mDV5OxP867OTwP8Meynsd86=4opn7uCTJekbmQ@mail.gmail.com>
References: <B5365ACE-09C6-4821-9BBC-D42C2DE24EC3@tm.net.uk>
 <CAMSNS4hVsugSpqcHkYLpcZuamdsdywqDLkHxVB0h9xs+uQXCjA@mail.gmail.com>
 <A80E76FB-5C40-4AC5-BE0D-F04A7B20D4CA@tm.net.uk>
 <CAMSNS4g=w2P6mDV5OxP867OTwP8Meynsd86=4opn7uCTJekbmQ@mail.gmail.com>
Message-ID: <FC48A2C6-108F-4624-9CE4-952D634540DF@tm.net.uk>

Hi L.Popov

Any other queries then please let me know.

I am working on a full config for FS with very good Jerasoft integration.

I have a custom collector which makes routing to local extensions and also domain based call routing work very well.

Soon as it’s finished I will let you know.

Regards
> On 12 Feb 2018, at 18:31, Lyubo Popov <lpopov at blasterphone.com> wrote:
> 
> I am also using Jerasoft so I am going to try this out on my setup. Great help! Thank you!
> 
> Re.,
> L.Popov
> 
>  <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon>	Virus-free. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link> <x-msg://11/#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
> 
> On Mon, Feb 12, 2018 at 4:20 PM, Jospeh Waite <joelists at tm.net.uk <mailto:joelists at tm.net.uk>> wrote:
> Hi L.Popov
> 
> Now this would depend in what format your redirect server is sending the reply back. In my case I’m using Jerasoft VCS which is sending the re-written caller id in a per contact form with src_number= set.
> 
> I use the following in my redirect catching dial plan
> 
>        <condition field="${sip_redirect_contact_0}" expression="src_number=(.+)\;(.+)$">
>             <action application="set" data="effective_caller_id_name=$1"/>
>             <action application="set" data="effective_caller_id_number=$1"/>
>         </condition>
> 
>         <condition field="${sip_redirect_contact_0}" expression="^<(.+)\>(.+)$">
>             <action application="bridge" data="{sip_invite_from_uri=sip:${effective_caller_id_number}@${sip_network_ip}}sofia/external/$1" />
>         </condition>
> 
> This catches the src_number in the first condition and updates the caller_id and then bridges the call in the second.
> 
> The one drawback is you would have to replicate this for each response your likely to have in the 300 response. So if it gives 3 different options to send the call to you would have to copy this 3 times, changing the sip_redirect_contact_0 and increasing the number on the end for each one.
> 
> Regards
> 
>> On 12 Feb 2018, at 18:07, Lyubo Popov <lpopov at blasterphone.com <mailto:lpopov at blasterphone.com>> wrote:
>> 
>> Hello Joseph,
>> 
>> I am having the same problem and would like to resurrect this post. Does someone knows how to do that? I send one CallerID to redirect server where this is being rewritten and receive another from it in the 300 response and would like to update this CID before sending to termination (b-leg)
>> 
>> Re.,
>> L.Popov
>> 
>>  <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon>	Virus-free. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link> <>
>> 
>> On Tue, Jul 11, 2017 at 4:32 PM, Joseph Waite <joelists at tm.net.uk <mailto:joelists at tm.net.uk>> wrote:
>> Hi Guys
>> 
>> I am trying to use the From field in a 300 Multiple Choices message to set the outbound CallerID in the re-directed b-leg
>> 
>> So call comes into FreeSwitch, gets sent out to a SIP Redirect server which replies with a 300 Multiple choices.
>> I need to pull the number from the From field of this 300 Response and use it to set the From field, raid & Paid fields in the re-directed invite.
>> 
>> Any help would be very much appreciated.
>> 
>> Regards
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.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>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> 
>> 
>> -- 
>> Atenciosamente,
>> ============================
>> Lyubo Popov
>> CEO - BlasterPhone LLC
>> Tel: 4003-1556 ( Outside Brazil 55 11 4003-1556)
>> iNum: +883 510001-354111 <tel:+883%20510%20001%20354%20111>
>> Website: http://www.blastervoip.com.br/ <http://www.blastervoip.com.br/>
>> ============================
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.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>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.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>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Atenciosamente,
> ============================
> Lyubo Popov
> CEO - BlasterPhone LLC
> Tel: 4003-1556 ( Outside Brazil 55 11 4003-1556)
> iNum: +883 510001-354111
> Website: http://www.blastervoip.com.br/ <http://www.blastervoip.com.br/>
> ============================
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 lpopov at blasterphone.com  Mon Feb 12 18:58:13 2018
From: lpopov at blasterphone.com (Lyubo Popov)
Date: Mon, 12 Feb 2018 16:58:13 -0200
Subject: [Freeswitch-users] Set b-leg CallerID using from in a 300
 Multiple Choices reply
In-Reply-To: <FC48A2C6-108F-4624-9CE4-952D634540DF@tm.net.uk>
References: <B5365ACE-09C6-4821-9BBC-D42C2DE24EC3@tm.net.uk>
 <CAMSNS4hVsugSpqcHkYLpcZuamdsdywqDLkHxVB0h9xs+uQXCjA@mail.gmail.com>
 <A80E76FB-5C40-4AC5-BE0D-F04A7B20D4CA@tm.net.uk>
 <CAMSNS4g=w2P6mDV5OxP867OTwP8Meynsd86=4opn7uCTJekbmQ@mail.gmail.com>
 <FC48A2C6-108F-4624-9CE4-952D634540DF@tm.net.uk>
Message-ID: <CAMSNS4jL9Ty9NFAWO3Dgts-CFYcWCazaAP7kp-hGVYeqNL3=eQ@mail.gmail.com>

I am grateful for your help Joseph! I am new in FS with VCS and using the
default configs provided by Jerasoft but I see the possibilities are
endless and I may abandon MVTS (my current SBC) for good if I can manage to
configure the FS the way I need it.

Re.
Lyubo Popov

On Mon, Feb 12, 2018 at 4:39 PM, Jospeh Waite <joelists at tm.net.uk> wrote:

> Hi L.Popov
>
> Any other queries then please let me know.
>
> I am working on a full config for FS with very good Jerasoft integration.
>
> I have a custom collector which makes routing to local extensions and also
> domain based call routing work very well.
>
> Soon as it’s finished I will let you know.
>
> Regards
>
> On 12 Feb 2018, at 18:31, Lyubo Popov <lpopov at blasterphone.com> wrote:
>
> I am also using Jerasoft so I am going to try this out on my setup. Great
> help! Thank you!
>
> Re.,
> L.Popov
>
>
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon> Virus-free.
> www.avast.com
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link>
>
> On Mon, Feb 12, 2018 at 4:20 PM, Jospeh Waite <joelists at tm.net.uk> wrote:
>
>> Hi L.Popov
>>
>> Now this would depend in what format your redirect server is sending the
>> reply back. In my case I’m using Jerasoft VCS which is sending the
>> re-written caller id in a per contact form with src_number= set.
>>
>> I use the following in my redirect catching dial plan
>>
>>        <condition field="${sip_redirect_contact_0}"
>> expression="src_number=(.+)\;(.+)$">
>>             <action application="set" data="effective_caller_id_name
>> =$1"/>
>>             <action application="set" data="effective_caller_id_numb
>> er=$1"/>
>>         </condition>
>>
>>         <condition field="${sip_redirect_contact_0}"
>> expression="^<(.+)\>(.+)$">
>>             <action application="bridge" data="{sip_invite_from_uri=sip:$
>> {effective_caller_id_number}@${sip_network_ip}}sofia/external/$1" />
>>         </condition>
>>
>> This catches the src_number in the first condition and updates the
>> caller_id and then bridges the call in the second.
>>
>> The one drawback is you would have to replicate this for each response
>> your likely to have in the 300 response. So if it gives 3 different options
>> to send the call to you would have to copy this 3 times, changing the
>> sip_redirect_contact_0 and increasing the number on the end for each one.
>>
>> Regards
>>
>> On 12 Feb 2018, at 18:07, Lyubo Popov <lpopov at blasterphone.com> wrote:
>>
>> Hello Joseph,
>>
>> I am having the same problem and would like to resurrect this post. Does
>> someone knows how to do that? I send one CallerID to redirect server where
>> this is being rewritten and receive another from it in the 300 response and
>> would like to update this CID before sending to termination (b-leg)
>>
>> Re.,
>> L.Popov
>>
>>
>> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon> Virus-free.
>> www.avast.com
>> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link>
>>
>> On Tue, Jul 11, 2017 at 4:32 PM, Joseph Waite <joelists at tm.net.uk> wrote:
>>
>>> Hi Guys
>>>
>>> I am trying to use the From field in a 300 Multiple Choices message to
>>> set the outbound CallerID in the re-directed b-leg
>>>
>>> So call comes into FreeSwitch, gets sent out to a SIP Redirect server
>>> which replies with a 300 Multiple choices.
>>> I need to pull the number from the From field of this 300 Response and
>>> use it to set the From field, raid & Paid fields in the re-directed invite.
>>>
>>> Any help would be very much appreciated.
>>>
>>> Regards
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>
>>
>>
>>
>> --
>> Atenciosamente,
>> ============================
>> Lyubo Popov
>> CEO - BlasterPhone LLC
>> Tel: 4003-1556 ( Outside Brazil 55 11 4003-1556)
>> iNum: +883 510001-354111 <+883%20510%20001%20354%20111>
>> Website: http://www.blastervoip.com.br/
>> ============================
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
>
> --
> Atenciosamente,
> ============================
> Lyubo Popov
> CEO - BlasterPhone LLC
> Tel: 4003-1556 ( Outside Brazil 55 11 4003-1556)
> iNum: +883 510001-354111 <+883%20510%20001%20354%20111>
> Website: http://www.blastervoip.com.br/
> ============================
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 
Atenciosamente,
============================
Lyubo Popov
CEO - BlasterPhone LLC
Tel: 4003-1556 ( Outside Brazil 55 11 4003-1556)
iNum: +883 510001-354111
Website: http://www.blastervoip.com.br/
============================
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From dujinfang at gmail.com  Tue Feb 13 01:04:56 2018
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 13 Feb 2018 09:04:56 +0800
Subject: [Freeswitch-users] test
In-Reply-To: <CAND18T2Fmdrr5d_cQ+E4hBCjgt0nH0wAhpwEOBEFidj6Jw0UKA@mail.gmail.com>
References: <11a301d3a3fd$78a72670$69f57350$@freeswitch.org>
 <CALXCt0osryX3900gYvhyjFfc3zYsjnyAVrRy04impZ_p-9uGAw@mail.gmail.com>
 <CAEJMVkD4jPtwvh2BRf7Bg4ARA3riYZAvKZ4+e0BCt-zYrzzWfw@mail.gmail.com>
 <CAND18T2Fmdrr5d_cQ+E4hBCjgt0nH0wAhpwEOBEFidj6Jw0UKA@mail.gmail.com>
Message-ID: <CAJs+-5rwZxp+o1--RM4vyHDEsUp61W+xfOWLtr=k+fY+HVQdrQ@mail.gmail.com>

7.

On Tue, Feb 13, 2018 at 12:19 AM, Mirko Brankovic <mirkobrankovic at gmail.com>
wrote:

> works  4321 :)
>
> On Mon, Feb 12, 2018 at 4:53 PM, Brian West <brian at freeswitch.com> wrote:
>
>> Test 4321
>>
>> On Mon, Feb 12, 2018 at 8:20 AM Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>
>>> pong
>>>
>>>
>>> On 12 February 2018 at 13:31, Ken Rice <krice at freeswitch.org> wrote:
>>>
>>>> Test 1234
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.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
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> OpenTelecom.IT
>>> cell: +39 347 266 56 18
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>
>> --
>>
>> 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 <(918)%20424-9378>
>>
>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>
>> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
>> color-twitter-96.png]
>> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
>
> --
> Regards,
> Mirko
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn
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From dujinfang at gmail.com  Tue Feb 13 01:06:54 2018
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 13 Feb 2018 09:06:54 +0800
Subject: [Freeswitch-users] Freeswitch support for text
 messages/commands between Verto/webRTC to/from conferenced backend SIP/RTP
 systems
In-Reply-To: <CY1PR07MB2377514FC809253B46BE6259BEFD0@CY1PR07MB2377.namprd07.prod.outlook.com>
References: <CY1PR07MB237750C64DF5CDAAFECCF266BEFD0@CY1PR07MB2377.namprd07.prod.outlook.com>
 <CY1PR0701MB122514D0CB73497C803D2FCBC3FD0@CY1PR0701MB1225.namprd07.prod.outlook.com>
 <CY1PR07MB2377514FC809253B46BE6259BEFD0@CY1PR07MB2377.namprd07.prod.outlook.com>
Message-ID: <CAJs+-5rMdp7h8HTKoKd=tBCPg3V0BRHFiaYuaJ0Xfj+OnxXYhQ@mail.gmail.com>

I think the verto demo has text .

On Wed, Feb 7, 2018 at 5:23 AM, Mundkowsky, Robert <rmundkowsky at ets.org>
wrote:

>
>
> We are wondering if FreeSWITCH supports sending and receiving text
> messages between Verto/webRTC to/from backend SIP/RTP systems that
> conferenced together?
>
>
>
> 1)      We want to use this to send commands from the web browser to the
> backend systems, for example:
>
> a.       “hey backend system, please turn on the ASR to start listening.”
>
>
>
> 2)      We also wonder if FreeSWITCH supports sending commands from the
> backend to the frontend (web browser), for example:
>
> a.       “hey avatar, here are some visemes to tell you how you should
> move your lips, arms, … at the same time as playing this audio”
>
>
>
> If both Verto/SIP or webRTC/RTP are supported, is either more ideal?
>
>
>
> For instance, if the text/commands are over RTC then maybe we can use one
> clock for all RTP streams and/or RTP sessions and use the RTP timestamps to
> synchronize the commands with audio/video.  Or maybe Verto/SIP is better
> for commands.
>
>
>
>
>
> I see there are various RFCs for text over RTP and text over
> RTCDataChannel in webRTC, but not sure if FreeSWITCH supports any of these.
>
>
>
> I also see FreeSWITCH mod_verto  support text messages, but not sure if
> FreeSWITCH sends these to/from SIP (e.g. maybe via SIMPLE SIP) or via
> RTCDataChannel?
>
>
>
>
>
> Robert
>
>
>
>
>
> ------------------------------
>
> This e-mail and any files transmitted with it may contain privileged or
> confidential information. It is solely for use by the individual for whom
> it is intended, even if addressed incorrectly. If you received this e-mail
> in error, please notify the sender; do not disclose, copy, distribute, or
> take any action in reliance on the contents of this information; and delete
> it from your system. Any other use of this e-mail is prohibited.
>
> Thank you for your compliance.
> ------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn
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From dujinfang at gmail.com  Tue Feb 13 01:10:19 2018
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 13 Feb 2018 09:10:19 +0800
Subject: [Freeswitch-users] "fsctl recover" doesn't work on Freeswitch
	V1.6.19
In-Reply-To: <tencent_02C00030E26255102C8403FB325D82811A09@qq.com>
References: <tencent_02C00030E26255102C8403FB325D82811A09@qq.com>
Message-ID: <CAJs+-5rAeE5+NTZHZVJrc7yKcgJNJNR+OdW2PT28R+SOnL9LLQ@mail.gmail.com>

hard to tell if you don't post logs to pastebin. but I can confirm master
works.

On Tue, Feb 6, 2018 at 12:28 PM, Kevin Chi <kevinchi at foxmail.com> wrote:

> hello everyone,
>
> The steps of my call recovery test as below.
> 1.First I added  the line "<param name="track-calls" value="true"/>" to
> internal.xml.
> 2.UA called 9664 to play the hold music, and then run "fsctl crash"
> command.
> 3.start freeswtich and run "fsctl recover" command.
> 4.call was interrupted, recover failed.
>
> I opened siptrace before "fsctl recover", the sipflow make me can not
> understand.
> freeswitch send INVITE to UA
> freeswitch recive 200 from UA
> freeswitch send BYE to UA
> freeswitch recive 200 from UA
> freeswitch send ACK from UA
>
> My freeswitch is only single server, the version is V1.6.19. I have tried
> the newer version V1.6.20, but the result was same.
> What's the reason? Pls give me some suggestion, thx a lot.
>
> ------------------
> With regards,
> Kevin Chi
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn
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From ko at sv01.de  Wed Feb  7 21:49:55 2018
From: ko at sv01.de (Kevin Olbrich)
Date: Wed, 7 Feb 2018 22:49:55 +0100
Subject: [Freeswitch-users] Syslog forwarding instead of file
Message-ID: <CA+gLzy-1QArDmQNz2xAGmkBhGULRXzuuT0auNZdeA3L9Zkf3KQ@mail.gmail.com>

 Hi!

We are currently testing Graylog2 to merge logs from several UC nodes.
Is Freeswitch able to send syslog messages to a remote server?

Kind regards,
Kevin
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From info at gbc.fi  Mon Feb 12 17:30:48 2018
From: info at gbc.fi (GBC Networks Oy)
Date: Mon, 12 Feb 2018 19:30:48 +0200
Subject: [Freeswitch-users] FS fails to restart when TLS endpoints are
	registered (Solaris 11.3)
Message-ID: <CACPMAjYkLvguNorEUbp64DAfZVE2LPrH3TsTkf_bijpx=8=C4Q@mail.gmail.com>

We're running FS on Solaris 11.3 and it is working perfectly, except that
we're unable to restart FS without deleting the TLS registrations from
sofia_reg_internal.db.
Our FS version is FreeSWITCH Version
1.9.0+git~20171010T231940Z~5940b04d4e~64bit (git 5940b04 2017-10-10
23:19:40Z 64bit)

Here are the symptoms:
1.    Service stops FS with fs_cli -x "fsctl shutdown now"
2.    FS stops and the service is restarted
3.    FS starts up and fs_cli is working, but ” sofia status” is empty
The console shows:
freeswitch at fs> 2018-02-12 17:06:23.875991 [ERR] sofia.c:3221 Error Creating
SIP UA for profile: internal
(sip:mod_sofia at 123.123.123.123:5060;maddr=123.123.123.123;transport=udp,tcp)
ATTEMPT 3 (RETRY IN 5
SEC)

2018-02-12 17:06:23.875991 [ERR] sofia.c:3231 Error Creating SIP UA for
profile: internal
(sip:mod_sofia at 123.123.123.123:5060;maddr=123.123.123.123;transport=udp,tcp)

The likely causes for this
are:

1) Another application is already listening on the specified
address.

2) The IP the profile is attempting to bind to is not local to this system.

Note: the IP is correct and nothing is listening on the port.

4.    Next we stop FS and restart
This time the console shows:
2018-02-12 13:33:51.132676 [INFO] switch_core_sqldb.c:1693 sofia:internal
Starting SQL thread.
2018-02-12 13:33:51.132770 [DEBUG] sofia.c:3373 Starting thread for internal
2018-02-12 13:33:51.132777 [DEBUG] sofia.c:3020 Launching worker thread for
internal
2018-02-12 13:33:51.138255 [NOTICE] sofia_reg.c:3398 Added gateway
'5e4ac47f-6ce4-462d-9247-565711fe3d4e' to profile 'internal'
2018-02-12 13:33:51.138405 [NOTICE] sofia_reg.c:3398 Added gateway
'b2563fad-0051-4067-abc3-6a431e05481e' to profile 'internal'
2018-02-12 13:33:51.247731 [DEBUG] freeswitch_lua.cpp:372 DBH handle
125b300 Connected.
2018-02-12 13:33:51.254128 [DEBUG] freeswitch_lua.cpp:401 DBH handle
125b300 released.
Assertion failed: orq->orq_queue, file nta.c, line 8101

5.    sqlite3 sofia_reg_internal.db "DELETE from sip_registrations WHERE
status LIKE '%TLS%'"
6.    FS starts normally


Here is a registration (non-TLS) that doesn’t break sofia:
# sqlite3 -header sofia_reg_internal.db "SELECT * from sip_registrations"
call_id|sip_user|sip_host|presence_hosts|contact|status|ping_status|ping_count|ping_time|force_ping|rpid|expires|ping_expires|user_agent|server_user|server_host|profile_name|hostname|network_ip|network_port|sip_username|sip_realm|mwi_user|mwi_host|orig_server_host|orig_hostname|sub_host
313531383139323735393532323132-m5mp9qlsdu27|USERNAME|fs.domain.com||"FS"
<sip:USERNAME at REMOTE-IP
:2054>|Registered(UDP)|Reachable|3|39824|1|unknown|1518455533|1518453577|snom320/
8.7.5.35|USERNAME||internal|fs|REMOTE-IP|2054|USERNAME|fs.domain.com
|USERNAME|fs.domain.com||fs|fs.domain.com

Here is a registration (TLS) that breaks the restart:
# sqlite3 -header sofia_reg_internal.db "SELECT * from
sip_registrations"
call_id|sip_user|sip_host|presence_hosts|contact|status|ping_status|ping_count|ping_time|force_ping|rpid|expires|ping_expires|user_agent|server_user|server_host|profile_name|hostname|network_ip|network_port|sip_username|sip_realm|mwi_user|mwi_host|orig_server_host|orig_hostname|sub_host
313531383139323735393532323132-m5mp9qlsdu27|USERNAME|fs.domain.com||"FS"
<sip:USERNAME at 192.168.6.236:2170
;transport=tls;fs_nat=yes;fs_path=sip%3AUSERNAME%40REMOTE-IP%3A2170%3Btransport%3Dtls>|Registered(TLS-NAT)|Reachable|0||1|unknown|1518457302|1518453664|snom320/
8.7.5.35|USERNAME||internal|fs|REMOTE-IP|2170|USERNAME|fs.domain.com
|USERNAME|fs.domain.com||fs|fs.domain.com


Best regards,
Mikko
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From rfmundkowsky at yahoo.com  Mon Feb 12 18:29:38 2018
From: rfmundkowsky at yahoo.com (robert mundkowsky)
Date: Mon, 12 Feb 2018 18:29:38 +0000 (UTC)
Subject: [Freeswitch-users] Freeswitch support for text messages/commands
 between Verto/webRTC to/from conferenced backend SIP/RTP systems
In-Reply-To: <141202963.679846.1518127154064@mail.yahoo.com>
References: <141202963.679846.1518127154064.ref@mail.yahoo.com>
 <141202963.679846.1518127154064@mail.yahoo.com>
Message-ID: <1868694664.1281193.1518460178550@mail.yahoo.com>



We are wondering if FreeSWITCH supports sending andreceiving text messages between Verto/webRTC to/from backend SIP/RTP systemsthat conferenced together?

 

1)     We want to use this to send commands from theweb browser to the backend systems, for example:

a.       “heybackend system, please turn on the ASR to start listening.”

 

2)     We also wonder if FreeSWITCH supports sendingcommands from the backend to the frontend (web browser), for example:

a.       “heyavatar, here are some visemes to tell you how you should move your lips, arms,… at the same time as playing this audio”

 

If both Verto/SIP or webRTC/RTP are supported for sending text, is eithermore ideal?  

 
For instance, if the text/commandsare over RTC then maybe we can use one clock for all RTP streams and/or RTPsessions and use the RTP timestamps to synchronize the commands withaudio/video.  Or maybe Verto/SIP is better for commands.
 

 

I see there are various RFCs for text over RTP and text over webRTC, but not sure if FreeSWITCH supports any of these.

 

I also see FreeSWITCH mod_verto  support text messages,but not sure if FreeSWITCH sends these to/from SIP (e.g. maybe via SIMPLE SIP)or via text over RTP?

 

 

Robert

 

  
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From alex at freeswitch.com  Tue Feb 13 08:23:06 2018
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Tue, 13 Feb 2018 17:23:06 +0900
Subject: [Freeswitch-users] FS fails to restart when TLS endpoints are
 registered (Solaris 11.3)
In-Reply-To: <CACPMAjYkLvguNorEUbp64DAfZVE2LPrH3TsTkf_bijpx=8=C4Q@mail.gmail.com>
References: <CACPMAjYkLvguNorEUbp64DAfZVE2LPrH3TsTkf_bijpx=8=C4Q@mail.gmail.com>
Message-ID: <CAGJLwFjcYZwzRUg+RyMqCK_d+=UDjEaFRnGoZ7ZWaDfhc0fheA@mail.gmail.com>

It looks like Sofia is not releasing ip on Solaris. If so it's a bug and it
should be reported to Sofia's bug tracker or Solaris or both. If you can
reproduce this problem on supported OS (Debian 8 at this moment) you can
open an issue on FreeSWITCH Jira.

Consider to read this first:
https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA

On Tue, Feb 13, 2018 at 2:30 AM, GBC Networks Oy <info at gbc.fi> wrote:

> We're running FS on Solaris 11.3 and it is working perfectly, except that
> we're unable to restart FS without deleting the TLS registrations from
> sofia_reg_internal.db.
> Our FS version is FreeSWITCH Version 1.9.0+git~20171010T231940Z~5940b04d4e~64bit
> (git 5940b04 2017-10-10 23:19:40Z 64bit)
>
> Here are the symptoms:
> 1.    Service stops FS with fs_cli -x "fsctl shutdown now"
> 2.    FS stops and the service is restarted
> 3.    FS starts up and fs_cli is working, but ” sofia status” is empty
> The console shows:
> freeswitch at fs> 2018-02-12 17:06:23.875991 [ERR] sofia.c:3221 Error
> Creating SIP UA for profile: internal (sip:mod_sofia at 123.123.123.
> 123:5060;maddr=123.123.123.123;transport=udp,tcp) ATTEMPT 3 (RETRY IN 5
> SEC)
>
> 2018-02-12 17:06:23.875991 [ERR] sofia.c:3231 Error Creating SIP UA for
> profile: internal (sip:mod_sofia at 123.123.123.123:5060;maddr=123.123.123.
> 123;transport=udp,tcp)
>
>
> The likely causes for this are:
>
>
> 1) Another application is already listening on the specified
> address.
>
> 2) The IP the profile is attempting to bind to is not local to this system.
>
> Note: the IP is correct and nothing is listening on the port.
>
> 4.    Next we stop FS and restart
> This time the console shows:
> 2018-02-12 13:33:51.132676 [INFO] switch_core_sqldb.c:1693 sofia:internal
> Starting SQL thread.
> 2018-02-12 13:33:51.132770 [DEBUG] sofia.c:3373 Starting thread for
> internal
> 2018-02-12 13:33:51.132777 [DEBUG] sofia.c:3020 Launching worker thread
> for internal
> 2018-02-12 13:33:51.138255 [NOTICE] sofia_reg.c:3398 Added gateway
> '5e4ac47f-6ce4-462d-9247-565711fe3d4e' to profile 'internal'
> 2018-02-12 13:33:51.138405 [NOTICE] sofia_reg.c:3398 Added gateway
> 'b2563fad-0051-4067-abc3-6a431e05481e' to profile 'internal'
> 2018-02-12 13:33:51.247731 [DEBUG] freeswitch_lua.cpp:372 DBH handle
> 125b300 Connected.
> 2018-02-12 13:33:51.254128 [DEBUG] freeswitch_lua.cpp:401 DBH handle
> 125b300 released.
> Assertion failed: orq->orq_queue, file nta.c, line 8101
>
> 5.    sqlite3 sofia_reg_internal.db "DELETE from sip_registrations WHERE
> status LIKE '%TLS%'"
> 6.    FS starts normally
>
>
> Here is a registration (non-TLS) that doesn’t break sofia:
> # sqlite3 -header sofia_reg_internal.db "SELECT * from sip_registrations"
> call_id|sip_user|sip_host|presence_hosts|contact|status|
> ping_status|ping_count|ping_time|force_ping|rpid|expires|
> ping_expires|user_agent|server_user|server_host|
> profile_name|hostname|network_ip|network_port|sip_username|
> sip_realm|mwi_user|mwi_host|orig_server_host|orig_hostname|sub_host
> 313531383139323735393532323132-m5mp9qlsdu27|USERNAME|fs.domain.com||"FS"
> <sip:USERNAME at REMOTE-IP:2054>|Registered(UDP)|Reachable|3|
> 39824|1|unknown|1518455533|1518453577|snom320/8.7.5.35|
> USERNAME||internal|fs|REMOTE-IP|2054|USERNAME|fs.domain.com|USERNAME|
> fs.domain.com||fs|fs.domain.com
>
> Here is a registration (TLS) that breaks the restart:
> # sqlite3 -header sofia_reg_internal.db "SELECT * from
> sip_registrations"
> call_id|sip_user|sip_host|presence_hosts|contact|status|
> ping_status|ping_count|ping_time|force_ping|rpid|expires|
> ping_expires|user_agent|server_user|server_host|
> profile_name|hostname|network_ip|network_port|sip_username|
> sip_realm|mwi_user|mwi_host|orig_server_host|orig_hostname|sub_host
> 313531383139323735393532323132-m5mp9qlsdu27|USERNAME|fs.domain.com||"FS" <
> sip:USERNAME at 192.168.6.236:2170;transport=tls;fs_nat=yes;
> fs_path=sip%3AUSERNAME%40REMOTE-IP%3A2170%3Btransport%3Dtls>|Registered(
> TLS-NAT)|Reachable|0||1|unknown|1518457302|1518453664|snom320/8.7.5.35
> |USERNAME||internal|fs|REMOTE-IP|2170|USERNAME|fs.domain.com|USERNAME|
> fs.domain.com||fs|fs.domain.com
>
>
> Best regards,
> Mikko
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

Alex Sibyakin | Support Engineer
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: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.
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From asilva at wirelessmundi.com  Tue Feb 13 08:35:54 2018
From: asilva at wirelessmundi.com (antonio)
Date: Tue, 13 Feb 2018 09:35:54 +0100
Subject: [Freeswitch-users] chatplan: control sip messages
In-Reply-To: <c4b6d2d0-7df8-f877-7304-e89fc25b8e26@wirelessmundi.com>
References: <c4b6d2d0-7df8-f877-7304-e89fc25b8e26@wirelessmundi.com>
Message-ID: <cf135c51-7972-ec9e-8054-585775847f10@wirelessmundi.com>

hi, can this be done?


On 02/09/2018 01:08 AM, antonio wrote:
> Hi,
>
> Is it possible to block sip messages between sip clients?
>
> I try to set the action stop in chatplan but it doesn't do anything, 
> the message is sent between the two endpoints.
>
> What i want is to be able to control witch accounts can send messages 
> and limited destinations, i also want to able receive the messages in 
> a common direction, like, chat at domain.local, without having the error 
> the warning message "sofia_presence.c:225 Can't find registered user"..
>
>
> Thanks.
>
>

-- 
Saludos / Regards / Cumprimentos
Anónio Silva

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From scott at tgifriday.com  Tue Feb 13 12:07:00 2018
From: scott at tgifriday.com (Scott Howell)
Date: Tue, 13 Feb 2018 12:07:00 +0000
Subject: [Freeswitch-users] test
In-Reply-To: <CAJs+-5rwZxp+o1--RM4vyHDEsUp61W+xfOWLtr=k+fY+HVQdrQ@mail.gmail.com>
References: <11a301d3a3fd$78a72670$69f57350$@freeswitch.org>
 <CALXCt0osryX3900gYvhyjFfc3zYsjnyAVrRy04impZ_p-9uGAw@mail.gmail.com>
 <CAEJMVkD4jPtwvh2BRf7Bg4ARA3riYZAvKZ4+e0BCt-zYrzzWfw@mail.gmail.com>
 <CAND18T2Fmdrr5d_cQ+E4hBCjgt0nH0wAhpwEOBEFidj6Jw0UKA@mail.gmail.com>
 <CAJs+-5rwZxp+o1--RM4vyHDEsUp61W+xfOWLtr=k+fY+HVQdrQ@mail.gmail.com>
Message-ID: <CAMALkh06Zy64tPWD-pNVK_VesrvX+FY+sTyKOLxsnEJF2Y5D9w@mail.gmail.com>

42

On Mon, Feb 12, 2018, 11:20 PM Seven Du <dujinfang at gmail.com> wrote:

> 7.
>
> On Tue, Feb 13, 2018 at 12:19 AM, Mirko Brankovic <
> mirkobrankovic at gmail.com> wrote:
>
>> works  4321 :)
>>
>> On Mon, Feb 12, 2018 at 4:53 PM, Brian West <brian at freeswitch.com> wrote:
>>
>>> Test 4321
>>>
>>> On Mon, Feb 12, 2018 at 8:20 AM Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>
>>>> pong
>>>>
>>>>
>>>> On 12 February 2018 at 13:31, Ken Rice <krice at freeswitch.org> wrote:
>>>>
>>>>> Test 1234
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.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
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> OpenTelecom.IT
>>>> cell: +39 347 266 56 18
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.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
>>>
>>> --
>>>
>>> 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 <(918)%20424-9378>
>>>
>>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>>
>>> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
>>> color-twitter-96.png]
>>> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>>
>>
>>
>>
>> --
>> Regards,
>> Mirko
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 gmaruzz at gmail.com  Tue Feb 13 12:18:13 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 13 Feb 2018 13:18:13 +0100
Subject: [Freeswitch-users] test
In-Reply-To: <CAMALkh06Zy64tPWD-pNVK_VesrvX+FY+sTyKOLxsnEJF2Y5D9w@mail.gmail.com>
References: <11a301d3a3fd$78a72670$69f57350$@freeswitch.org>
 <CALXCt0osryX3900gYvhyjFfc3zYsjnyAVrRy04impZ_p-9uGAw@mail.gmail.com>
 <CAEJMVkD4jPtwvh2BRf7Bg4ARA3riYZAvKZ4+e0BCt-zYrzzWfw@mail.gmail.com>
 <CAND18T2Fmdrr5d_cQ+E4hBCjgt0nH0wAhpwEOBEFidj6Jw0UKA@mail.gmail.com>
 <CAJs+-5rwZxp+o1--RM4vyHDEsUp61W+xfOWLtr=k+fY+HVQdrQ@mail.gmail.com>
 <CAMALkh06Zy64tPWD-pNVK_VesrvX+FY+sTyKOLxsnEJF2Y5D9w@mail.gmail.com>
Message-ID: <CALXCt0rKRWnyDdei3ucvM6gqCBhjZAPwwYT78nUAyijjQnD9Fg@mail.gmail.com>

On Feb 13, 2018 1:07 PM, "Scott Howell" <scott at tgifriday.com> wrote:

42

:D



On Mon, Feb 12, 2018, 11:20 PM Seven Du <dujinfang at gmail.com> wrote:

> 7.
>
> On Tue, Feb 13, 2018 at 12:19 AM, Mirko Brankovic <
> mirkobrankovic at gmail.com> wrote:
>
>> works  4321 :)
>>
>> On Mon, Feb 12, 2018 at 4:53 PM, Brian West <brian at freeswitch.com> wrote:
>>
>>> Test 4321
>>>
>>> On Mon, Feb 12, 2018 at 8:20 AM Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>
>>>> pong
>>>>
>>>>
>>>> On 12 February 2018 at 13:31, Ken Rice <krice at freeswitch.org> wrote:
>>>>
>>>>> Test 1234
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.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
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> OpenTelecom.IT
>>>> cell: +39 347 266 56 18
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.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
>>>
>>> --
>>>
>>> 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 <(918)%20424-9378>
>>>
>>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>>
>>> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
>>> color-twitter-96.png]
>>> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>>
>>
>>
>>
>> --
>> Regards,
>> Mirko
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.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 info at gbc.fi  Tue Feb 13 13:06:00 2018
From: info at gbc.fi (GBC Networks Oy)
Date: Tue, 13 Feb 2018 15:06:00 +0200
Subject: [Freeswitch-users] FS fails to restart when TLS endpoints are
 registered (Solaris 11.3)
Message-ID: <CACPMAjaosi-WJ1JpZdzXZrT33kVFVgbXK3G1-JtxWXkO+g6adw@mail.gmail.com>

Thanks Alexey for your help in diagnosing the problem.
I dug further into the TCP stack and found out we are having 2 separate
restart problems.

The first problem is not that the socket is failing to be released, but
apparently Sofia is trying to open the port without the SO_REUSEADDR
option, which means FS must wait for all connection to close first.
So as long as there are any TCP connections to those ports in the TIME_WAIT
state, FS will not start Sofia.
I tested with netcat and that program can bind to the port regardless of
the state of old connections, so I suspect this is a compile error.

The second problem is that the TLS and TCP registrations in
sofia_reg_internal.db cause FS to throw the assertion error “Assertion
failed: orq->orq_queue, file nta.c, line 8101”.
UDP registrations are fine.
Does FS try to continue these registrations assuming that the connection is
still up or reopen a connection to the same port?
The assert line is found in nta.c’s outgoing_create function. It seems that
the outgoing message queue is uninitialized, so the assertion error is
likely more a symptom of some other problem, instead of being the cause of
the problem.
How should I go about debugging the outgoing message queue?


Thanks in advance,
Mikko

---------- Edelleenlähetetty viesti ----------

> From: Alexey Sibyakin <alex at freeswitch.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Tue, 13 Feb 2018 17:23:06 +0900
> Subject: Re: [Freeswitch-users] FS fails to restart when TLS endpoints are
> registered (Solaris 11.3)
> It looks like Sofia is not releasing ip on Solaris. If so it's a bug and
> it should be reported to Sofia's bug tracker or Solaris or both. If you can
> reproduce this problem on supported OS (Debian 8 at this moment) you can
> open an issue on FreeSWITCH Jira.
>
> Consider to read this first: https://freeswitch.org/
> confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA
>
> On Tue, Feb 13, 2018 at 2:30 AM, GBC Networks Oy <info at gbc.fi> wrote:
>
>> We're running FS on Solaris 11.3 and it is working perfectly, except that
>> we're unable to restart FS without deleting the TLS registrations from
>> sofia_reg_internal.db.
>> Our FS version is FreeSWITCH Version 1.9.0+git~20171010T231940Z~5940b04d4e~64bit
>> (git 5940b04 2017-10-10 23:19:40Z 64bit)
>>
>> Here are the symptoms:
>> 1.    Service stops FS with fs_cli -x "fsctl shutdown now"
>> 2.    FS stops and the service is restarted
>> 3.    FS starts up and fs_cli is working, but ” sofia status” is empty
>> The console shows:
>> freeswitch at fs> 2018-02-12 17:06:23.875991 [ERR] sofia.c:3221 Error
>> Creating SIP UA for profile: internal (sip:mod_sofia at 123.123.123.123
>> :5060;maddr=123.123.123.123;transport=udp,tcp) ATTEMPT 3 (RETRY IN 5
>> SEC)
>>
>> 2018-02-12 17:06:23.875991 [ERR] sofia.c:3231 Error Creating SIP UA for
>> profile: internal (sip:mod_sofia at 123.123.123.123
>> :5060;maddr=123.123.123.123;transport=udp,tcp)
>>
>>
>>
>> The likely causes for this are:
>>
>>
>> 1) Another application is already listening on the specified
>> address.
>>
>> 2) The IP the profile is attempting to bind to is not local to this
>> system.
>>
>> Note: the IP is correct and nothing is listening on the port.
>>
>> 4.    Next we stop FS and restart
>> This time the console shows:
>> 2018-02-12 13:33:51.132676 [INFO] switch_core_sqldb.c:1693 sofia:internal
>> Starting SQL thread.
>> 2018-02-12 13:33:51.132770 [DEBUG] sofia.c:3373 Starting thread for
>> internal
>> 2018-02-12 13:33:51.132777 [DEBUG] sofia.c:3020 Launching worker thread
>> for internal
>> 2018-02-12 13:33:51.138255 [NOTICE] sofia_reg.c:3398 Added gateway
>> '5e4ac47f-6ce4-462d-9247-565711fe3d4e' to profile 'internal'
>> 2018-02-12 13:33:51.138405 [NOTICE] sofia_reg.c:3398 Added gateway
>> 'b2563fad-0051-4067-abc3-6a431e05481e' to profile 'internal'
>> 2018-02-12 13:33:51.247731 [DEBUG] freeswitch_lua.cpp:372 DBH handle
>> 125b300 Connected.
>> 2018-02-12 13:33:51.254128 [DEBUG] freeswitch_lua.cpp:401 DBH handle
>> 125b300 released.
>> Assertion failed: orq->orq_queue, file nta.c, line 8101
>>
>> 5.    sqlite3 sofia_reg_internal.db "DELETE from sip_registrations WHERE
>> status LIKE '%TLS%'"
>> 6.    FS starts normally
>>
>>
>> Here is a registration (non-TLS) that doesn’t break sofia:
>> # sqlite3 -header sofia_reg_internal.db "SELECT * from sip_registrations"
>> call_id|sip_user|sip_host|presence_hosts|contact|status|ping
>> _status|ping_count|ping_time|force_ping|rpid|expires|ping_
>> expires|user_agent|server_user|server_host|profile_name|
>> hostname|network_ip|network_port|sip_username|sip_realm|
>> mwi_user|mwi_host|orig_server_host|orig_hostname|sub_host
>> 313531383139323735393532323132-m5mp9qlsdu27|USERNAME|fs.domain.com||"FS"
>> <sip:USERNAME at REMOTE-IP:2054>|Registered(UDP)|Reachable|3|39
>> 824|1|unknown|1518455533|1518453577|snom320/8.7.5.35|USERNAM
>> E||internal|fs|REMOTE-IP|2054|USERNAME|fs.domain.com|USERNAME|
>> fs.domain.com||fs|fs.domain.com
>>
>> Here is a registration (TLS) that breaks the restart:
>> # sqlite3 -header sofia_reg_internal.db "SELECT * from
>> sip_registrations"
>> call_id|sip_user|sip_host|presence_hosts|contact|status|ping
>> _status|ping_count|ping_time|force_ping|rpid|expires|ping_
>> expires|user_agent|server_user|server_host|profile_name|
>> hostname|network_ip|network_port|sip_username|sip_realm|
>> mwi_user|mwi_host|orig_server_host|orig_hostname|sub_host
>> 313531383139323735393532323132-m5mp9qlsdu27|USERNAME|fs.domain.com||"FS"
>> <sip:USERNAME at 192.168.6.236:2170;transport=tls;fs_nat=yes;fs
>> _path=sip%3AUSERNAME%40REMOTE-IP%3A2170%3Btransport%3Dtls>|
>> Registered(TLS-NAT)|Reachable|0||1|unknown|1518457302|1518453664|snom320/
>> 8.7.5.35|USERNAME||internal|fs|REMOTE-IP|2170|USERNAME|fs.domain.com
>> |USERNAME|fs.domain.com||fs|fs.domain.com
>>
>>
>> Best regards,
>> Mikko
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
>
> --
>
> Alex Sibyakin | Support Engineer
> 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: alex at freeswitch.com
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
> Need commercial support? Contact sales at freeswitch.com for details.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From basmo13 at yahoo.co.uk  Tue Feb 13 15:00:01 2018
From: basmo13 at yahoo.co.uk (Sonnie)
Date: Tue, 13 Feb 2018 15:00:01 +0000 (UTC)
Subject: [Freeswitch-users] Originate with Dialplan
References: <120013994.268226.1518534001458.ref@mail.yahoo.com>
Message-ID: <120013994.268226.1518534001458@mail.yahoo.com>

 I have a somewhat newbie question 
I have a current scenerio were a user registers with freeswitch and makes a call out - Works Perfect!
<include>  <user id="1000">    <!-- CGRateS config sample.     Make default 1000 user prepaid by adding cgr_reqtype variable to your user -->    <params>      <param name="password" value="$${default_password}"/>      <param name="vm-password" value="1000"/>    </params>    <variables>      <variable name="toll_allow" value="domestic,international,local"/>      <variable name="accountcode" value="1000"/>      <variable name="user_context" value="default"/>      <variable name="effective_caller_id_name" value="Extension 1000"/>      <variable name="effective_caller_id_number" value="1000"/>      <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>      <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>      <variable name="callgroup" value="techsupport"/>      <variable name="cgr_reqtype" value="*prepaid"/>    </variables>  </user></include>
and then in the default dialplan, i have this
    <!--- CGRateS related extensions -->    <!-- Place all requests into park app. Notify must be empty so we do not loop. Order Lcr computation -->    <extension name="CGRateS_Auth">      <condition field="${cgr_notify}" expression="^$">        <!---<action application="set" data="cgr_computelcr=true" />-->        <action application="park" />      </condition>    </extension>
Now, i would love to send a call out with the  originate command I've tried this - originate {origination_caller_id_number='123456789',ignore_early_media=true,originate_timeout=45}sofia/gateway/provider_one/7774262712 &playback(/usr/share/freeswitch/sounds/en/us/callie/voicemail/8000/vm-hello.wav) default CGRates_Auth
That sends the call out and then hangs it up(expectedly)

Is there a way i can replicate my working scenerio(maybe pass in those channel variable) but with the originate command? I'm naively thinking, maybe sending the call to a dialplan first before sending it to a gateway?

Please pardon me if the answers are obvious.
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From igorolhovskiy at gmail.com  Tue Feb 13 16:36:28 2018
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Tue, 13 Feb 2018 18:36:28 +0200
Subject: [Freeswitch-users] Calls recording issue
Message-ID: <b462b754-0a38-4524-95c4-41e0d55b0f60@Spark>

Hi!
I’ve faced an issue with call recording.
I’m recording with uuid_record in wav (also tried with mp3) with these legs

A: WebRTC/SIPJS/Opus at 8000
B: SIP/PCMA

In this case I get clear recording only on leg B. Leg A is sounds like tape on SLOW (really SLOW) playing. Mostly from leg A i hear smth like (A-R-R….!)
In a case if leg A is SIP with any codec, it’s ok.

FS 1.6.20

Could it be a bug in FS?

Regards, Igor
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From ryharris at airmail.cc  Tue Feb 13 19:22:30 2018
From: ryharris at airmail.cc (Ryan Harris)
Date: Tue, 13 Feb 2018 14:22:30 -0500
Subject: [Freeswitch-users] Originate with Dialplan
In-Reply-To: <120013994.268226.1518534001458@mail.yahoo.com>
References: <120013994.268226.1518534001458.ref@mail.yahoo.com>
 <120013994.268226.1518534001458@mail.yahoo.com>
Message-ID: <e57efc9d-44af-f475-3167-c75fdfe605a8@airmail.cc>

On 02/13/2018 10:00 AM, Sonnie wrote:
> That sends the call out and then hangs it up(expectedly)

It's probably playing the audio before you get a chance to pick up. Try
seeing if the behavior is the same when you &park() or play something
longer, like music.


From michael at mailworks.org  Wed Feb 14 07:51:52 2018
From: michael at mailworks.org (Michael Avers)
Date: Wed, 14 Feb 2018 00:51:52 -0700
Subject: [Freeswitch-users] DTMF relay in 3-way call
Message-ID: <1518594712.3324578.1270289312.3B4B56FA@webmail.messagingengine.com>

Hello,

I have the following scenario:

Deskphone dials PSTN Destination A. Once answered, deskphone 3-ways a second PSTN Destination B. From that point, DTMF pressed by Destination A needs to be heard by B. Is there a simple way to have that DTMF relay over from A to B?

Currently the 3-way is happening on the deskphone (Yealink T42). Is there perhaps a different way they should be connected?

Thanks
Mike


From ssoni at ucdavis.edu  Tue Feb 13 19:41:19 2018
From: ssoni at ucdavis.edu (Shivang Soni)
Date: Tue, 13 Feb 2018 11:41:19 -0800
Subject: [Freeswitch-users] Calls recording issue
In-Reply-To: <b462b754-0a38-4524-95c4-41e0d55b0f60@Spark>
References: <b462b754-0a38-4524-95c4-41e0d55b0f60@Spark>
Message-ID: <CAMVzwA4-R9n20LowbPTg190zzDSBV6zmHhcnN3pa8Acey7R5Xw@mail.gmail.com>

Hi,
I am Shivang Soni. I am currently working on a web development project
which requires accessing the microphone. I can able to access microphone if
the application runs on my local system but cannot able to access
microphone when running the same application on our college interaction
server. So I tried to explore about Freeswitch by coming to know it is
possible through that. As I am new to FreeSWITCH, it would be great if you
can guide me on how to resolve this problem using the free switch. I have
currently installed free-switch on the Debian 8 OS on AWS. As I am not
familiar with the free switch I am also curious to know how to keep my
particular application on FreeSWITCH so that it can access the microphone.

I am subscribed to mailing list but don't know any page where to post, how
to create new thread..

Looking forward to your response,
Regards,
Shivang Soni.


On Tue, Feb 13, 2018 at 8:36 AM, Igor Olhovskiy <igorolhovskiy at gmail.com>
wrote:

> Hi!
> I’ve faced an issue with call recording.
> I’m recording with uuid_record in wav (also tried with mp3) with these legs
>
> A: WebRTC/SIPJS/Opus at 8000
> B: SIP/PCMA
>
> In this case I get clear recording only on leg B. Leg A is sounds like
> tape on SLOW (really SLOW) playing. Mostly from leg A i hear smth like
> (A-R-R….!)
> In a case if leg A is SIP with any codec, it’s ok.
>
> FS 1.6.20
>
> Could it be a bug in FS?
>
> Regards, Igor
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 basmo13 at yahoo.co.uk  Wed Feb 14 08:56:19 2018
From: basmo13 at yahoo.co.uk (Sonnie)
Date: Wed, 14 Feb 2018 08:56:19 +0000 (UTC)
Subject: [Freeswitch-users] Originate with Dialplan
References: <709942155.993344.1518598579275.ref@mail.yahoo.com>
Message-ID: <709942155.993344.1518598579275@mail.yahoo.com>

 Hello,
I have a current scenerio were a user registers with freeswitch and makes a call out - Works Perfect!
<include>  <user id="1000">    <!-- CGRateS config sample.     Make default 1000 user prepaid by adding cgr_reqtype variable to your user -->    <params>      <param name="password" value="$${default_password}"/>      <param name="vm-password" value="1000"/>    </params>    <variables>      <variable name="toll_allow" value="domestic,international,local"/>      <variable name="accountcode" value="1000"/>      <variable name="user_context" value="default"/>      <variable name="effective_caller_id_name" value="Extension 1000"/>      <variable name="effective_caller_id_number" value="1000"/>      <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>      <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>      <variable name="callgroup" value="techsupport"/>      <variable name="cgr_reqtype" value="*prepaid"/>    </variables>  </user></include>
and then in the default dialplan, i have this
    <!--- CGRateS related extensions -->    <!-- Place all requests into park app. Notify must be empty so we do not loop. Order Lcr computation -->    <extension name="CGRateS_Auth">      <condition field="${cgr_notify}" expression="^$">        <!---<action application="set" data="cgr_computelcr=true" />-->        <action application="park" />      </condition>    </extension>
Now, i would love to send a call out with the  originate command I've tried this - originate {origination_caller_id_number='123456789',ignore_early_media=true,originate_timeout=45}sofia/gateway/provider_one/7774262712 &playback(/usr/share/freeswitch/sounds/en/us/callie/voicemail/8000/vm-hello.wav) default CGRates_Auth
That sends the call out and then hangs it up(expectedly)

Is there a way i can replicate my working scenerio(maybe pass in those channel variable) but with the originate command? I'm naively thinking, maybe sending the call to a dialplan first before sending it to a gateway?

Please pardon me if the answers are obvious.
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From rmundkowsky at ets.org  Tue Feb 13 20:40:27 2018
From: rmundkowsky at ets.org (Mundkowsky, Robert)
Date: Tue, 13 Feb 2018 20:40:27 +0000
Subject: [Freeswitch-users] Freeswitch support for text
 messages/commands between Verto/webRTC to/from conferenced backend SIP/RTP
 systems
References: <CY1PR07MB237750C64DF5CDAAFECCF266BEFD0@CY1PR07MB2377.namprd07.prod.outlook.com>
 <CY1PR0701MB122514D0CB73497C803D2FCBC3FD0@CY1PR0701MB1225.namprd07.prod.outlook.com>
 <CY1PR07MB2377514FC809253B46BE6259BEFD0@CY1PR07MB2377.namprd07.prod.outlook.com>
 <CAJs+-5rMdp7h8HTKoKd=tBCPg3V0BRHFiaYuaJ0Xfj+OnxXYhQ@mail.gmail.com> 
Message-ID: <CY1PR07MB23777B904D603C86D784B43BBEF60@CY1PR07MB2377.namprd07.prod.outlook.com>

Well,

I read up on this in the FreeSWITCH 1.8 book, seems VERTO webRTC clients can send text messages to SIP/RTP clients via messages out of band from the main audio/video stream.   For SIP/RTP clients seems they get messages via SIP SIMPLE hence they are over SIP.

And FreeSWITCH modules that support the FreeSWITCH Chat API can inter-communicate (mod_dptools: chat supports sip, verto, and maybe jingles protocol, mod_sms supports sms,…).

I did not see anything about Message Session Relay Protocol, text over RTP, nor RTCDataChannel, so I assume the text messages likely need to be small.

Robert Mundkowsky

From: Mundkowsky, Robert
Sent: Monday, February 12, 2018 11:02 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: RE: [Freeswitch-users] Freeswitch support for text messages/commands between Verto/webRTC to/from conferenced backend SIP/RTP systems

Yes, but where does the text go for SIP/RTP users?

Robert Mundkowsky

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Seven Du
Sent: Monday, February 12, 2018 8:07 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Freeswitch support for text messages/commands between Verto/webRTC to/from conferenced backend SIP/RTP systems

I think the verto demo has text .

On Wed, Feb 7, 2018 at 5:23 AM, Mundkowsky, Robert <rmundkowsky at ets.org<mailto:rmundkowsky at ets.org>> wrote:

We are wondering if FreeSWITCH supports sending and receiving text messages between Verto/webRTC to/from backend SIP/RTP systems that conferenced together?


1)      We want to use this to send commands from the web browser to the backend systems, for example:

a.       “hey backend system, please turn on the ASR to start listening.”


2)      We also wonder if FreeSWITCH supports sending commands from the backend to the frontend (web browser), for example:

a.       “hey avatar, here are some visemes to tell you how you should move your lips, arms, … at the same time as playing this audio”

If both Verto/SIP or webRTC/RTP are supported, is either more ideal?

For instance, if the text/commands are over RTC then maybe we can use one clock for all RTP streams and/or RTP sessions and use the RTP timestamps to synchronize the commands with audio/video.  Or maybe Verto/SIP is better for commands.


I see there are various RFCs for text over RTP and text over RTCDataChannel in webRTC, but not sure if FreeSWITCH supports any of these.

I also see FreeSWITCH mod_verto  support text messages, but not sure if FreeSWITCH sends these to/from SIP (e.g. maybe via SIMPLE SIP) or via RTCDataChannel?


Robert



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From tculjaga at gmail.com  Wed Feb 14 10:00:21 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 14 Feb 2018 11:00:21 +0100
Subject: [Freeswitch-users] "received=" in path header
In-Reply-To: <CAHtxdDcDP_3nbCAtMiunOcJB8ZPcNQJroF4-=0VPpxy4q9LLmw@mail.gmail.com>
References: <CAJoKKdXk-Y_p8_3ADLu89FqYX52garpZPaqOa3vFfDWbhtnNww@mail.gmail.com>
 <CAHtxdDcDP_3nbCAtMiunOcJB8ZPcNQJroF4-=0VPpxy4q9LLmw@mail.gmail.com>
Message-ID: <CAJoKKdU4nGQa5juSNN9zvi3knUuv0tfqbcF6fXeDeVa0tFLUBg@mail.gmail.com>

hi Sergey,


i tried this before and let me nowhere :=)


sofia_contact internal/1001 at mydomain
sofia/internal/sip:1001 at 192.0.2.77
;transport=ws;fs_nat=yes;fs_path=sip%3A192.168.50.60%3Blr%3Breceived%3Dsip%3A192.168.200.77%3A63316%3Btransport%3Dws


originate example:

bgapi expand originate ${sofia_contact(internal/1001 at imydomain)} &echo()
+OK Job-UUID: 86f2dd6b-a3d8-4208-8f1b-e4def81cd2a7

2018-02-14 10:49:19.200026 [DEBUG] switch_ivr_originate.c:2142 Parsing
global variables
2018-02-14 10:49:19.200026 [NOTICE] switch_channel.c:1104 New Channel
sofia/internal/1001 at 192.0.2.77 [c6605a6c-78c8-4553-a1f7-d4e59793727d]
2018-02-14 10:49:19.200026 [DEBUG] mod_sofia.c:4819 (sofia/internal/
1001 at 192.0.2.77) State Change CS_NEW -> CS_INIT
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:584
(sofia/internal/1001 at 192.0.2.77) Running State Change CS_INIT (Cur 1 Tot
334)
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:627
(sofia/internal/1001 at 192.0.2.77) State INIT
2018-02-14 10:49:19.200026 [DEBUG] mod_sofia.c:90 sofia/internal/
1001 at 192.0.2.77 SOFIA INIT
2018-02-14 10:49:19.200026 [DEBUG] sofia_glue.c:1264
sip:192.168.50.60;lr;received=sip:192.168.200.77:63316;transport=ws Setting
proxy route to sofia/internal/1001 at 192.0.2.77
2018-02-14 10:49:19.200026 [DEBUG] sofia_glue.c:1295 sofia/internal/
1001 at 192.0.2.77 sending invite version: 1.6.19 git b1b21d0 2017-11-20
16:34:16Z 64bit
Local SDP:
v=0
o=FreeSWITCH 1518569171 1518569172 IN IP4 192.168.50.65
s=FreeSWITCH
c=IN IP4 192.168.50.65
t=0 0
a=msid-semantic: WMS bXi5EYQcfFSgTMzERN0T6iPThuyNRKxT
m=audio 32588 RTP/SAVPF 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fingerprint:sha-256
09:E7:A0:05:C2:9D:7E:8F:C2:AF:FE:9C:BB:67:0F:F0:4A:E2:E0:33:D0:F5:D5:F7:23:6C:3D:BA:1E:27:C5:65
a=setup:actpass
a=rtcp-mux
a=rtcp:32588 IN IP4 192.168.50.65
a=ssrc:1855748415 cname:xGP2uv4A0NHKp6e6
a=ssrc:1855748415 msid:bXi5EYQcfFSgTMzERN0T6iPThuyNRKxT a0
a=ssrc:1855748415 mslabel:bXi5EYQcfFSgTMzERN0T6iPThuyNRKxT
a=ssrc:1855748415 label:bXi5EYQcfFSgTMzERN0T6iPThuyNRKxTa0
a=ice-ufrag:K5QGdvqlPOYxomgM
a=ice-pwd:SVEdhzD8cbIr8NoIJ4fNpvY5
a=candidate:5270270800 1 udp 659136 192.168.50.65 32588 typ host generation
0
a=candidate:5270270800 2 udp 659136 192.168.50.65 32588 typ host generation
0
a=ptime:20
a=sendrecv

2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:40
sofia/internal/1001 at 192.0.2.77 Standard INIT
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:48
(sofia/internal/1001 at 192.0.2.77) State Change CS_INIT -> CS_ROUTING
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:627
(sofia/internal/1001 at 192.0.2.77) State INIT going to sleep
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:584
(sofia/internal/1001 at 192.0.2.77) Running State Change CS_ROUTING (Cur 1 Tot
334)
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:643
(sofia/internal/1001 at 192.0.2.77) State ROUTING
2018-02-14 10:49:19.200026 [DEBUG] mod_sofia.c:143 sofia/internal/
1001 at 192.0.2.77 SOFIA ROUTING
2018-02-14 10:49:19.200026 [DEBUG] switch_ivr_originate.c:67
(sofia/internal/1001 at 192.0.2.77) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:643
(sofia/internal/1001 at 192.0.2.77) State ROUTING going to sleep
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:584
(sofia/internal/1001 at 192.0.2.77) Running State Change CS_CONSUME_MEDIA (Cur
1 Tot 334)
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:662
(sofia/internal/1001 at 192.0.2.77) State CONSUME_MEDIA
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:662
(sofia/internal/1001 at 192.0.2.77) State CONSUME_MEDIA going to sleep
2018-02-14 10:49:19.200026 [DEBUG] sofia.c:7084 Channel sofia/internal/
1001 at 192.0.2.77 entering state [calling][0]
2018-02-14 10:49:19.200026 [DEBUG] sofia.c:7084 Channel sofia/internal/
1001 at 192.0.2.77 entering state [terminated][503]
2018-02-14 10:49:19.200026 [NOTICE] sofia.c:8273 Hangup sofia/internal/
1001 at 192.0.2.77 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:584
(sofia/internal/1001 at 192.0.2.77) Running State Change CS_HANGUP (Cur 1 Tot
334)
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:850
(sofia/internal/1001 at 192.0.2.77) Callstate Change DOWN -> HANGUP
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:852
(sofia/internal/1001 at 192.0.2.77) State HANGUP
2018-02-14 10:49:19.200026 [DEBUG] mod_sofia.c:438 Channel sofia/internal/
1001 at 192.0.2.77 hanging up, cause: NORMAL_TEMPORARY_FAILURE
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:60
sofia/internal/1001 at 192.0.2.77 Standard HANGUP, cause:
NORMAL_TEMPORARY_FAILURE
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:852
(sofia/internal/1001 at 192.0.2.77) State HANGUP going to sleep
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:619
(sofia/internal/1001 at 192.0.2.77) State Change CS_HANGUP -> CS_REPORTING
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:584
(sofia/internal/1001 at 192.0.2.77) Running State Change CS_REPORTING (Cur 1
Tot 334)
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:938
(sofia/internal/1001 at 192.0.2.77) State REPORTING
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:174
sofia/internal/1001 at 192.0.2.77 Standard REPORTING, cause:
NORMAL_TEMPORARY_FAILURE
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:938
(sofia/internal/1001 at 192.0.2.77) State REPORTING going to sleep
2018-02-14 10:49:19.200026 [DEBUG] switch_core_state_machine.c:610
(sofia/internal/1001 at 192.0.2.77) State Change CS_REPORTING -> CS_DESTROY
2018-02-14 10:49:19.200026 [DEBUG] switch_core_session.c:1665 Session 334
(sofia/internal/1001 at 192.0.2.77) Locked, Waiting on external entities
2018-02-14 10:49:19.220029 [DEBUG] switch_ivr_originate.c:3848 Originate
Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE]
2018-02-14 10:49:19.220029 [NOTICE] switch_core_session.c:1683 Session 334
(sofia/internal/1001 at 192.0.2.77) Ended
2018-02-14 10:49:19.220029 [NOTICE] switch_core_session.c:1687 Close
Channel sofia/internal/1001 at 192.0.2.77 [CS_DESTROY]
2018-02-14 10:49:19.220029 [DEBUG] switch_core_state_machine.c:741
(sofia/internal/1001 at 192.0.2.77) Running State Change CS_DESTROY (Cur 0 Tot
334)
2018-02-14 10:49:19.220029 [DEBUG] switch_core_state_machine.c:751
(sofia/internal/1001 at 192.0.2.77) State DESTROY
2018-02-14 10:49:19.220029 [DEBUG] mod_sofia.c:343 sofia/internal/
1001 at 192.0.2.77 SOFIA DESTROY
2018-02-14 10:49:19.220029 [DEBUG] switch_core_state_machine.c:181
sofia/internal/1001 at 192.0.2.77 Standard DESTROY
2018-02-14 10:49:19.220029 [DEBUG] switch_core_state_machine.c:751
(sofia/internal/1001 at 192.0.2.77) State DESTROY going to sleep




the workaround i found is to set received_format in kamilio to be
ip~port~protoid

modparam("path", "received_format", 1)


by doing this i get a contact that FS "knows" how to route:

sofia_contact internal/1001 at mydomain
sofia/internal/sip:1001 at 192.0.2.77
;transport=ws;fs_nat=yes;fs_path=sip%3A192.168.50.60%3Blr%3Breceived%3D192.168.200.77~63931~6

i was unable to find any documentation on how FS behaves when we get
received parametar present.


T.



On 12 February 2018 at 18:14, Sergey Safarov <s.safarov at gmail.com> wrote:

> Need to get registered user contact string FS api.
> Using this contact string you can dial webrtc user
>
> пн, 12 февр. 2018 г., 16:54 Tihomir Culjaga <tculjaga at gmail.com>:
>
>> hi,
>>
>>
>> i have  UA > Kamailio > Freeswitch
>>
>> UA is using sip for websockets and registers via kamailio to freeswitch.
>> When i try to originate a call from FS to the registered endpoint, the
>> call fails.
>>
>> freeswitch at FS01> bgapi originate {origination_caller_id_number=
>> 1002}sofia/internal/1001%mydomain &echo()
>>
>> I think FS is trying to use transport from received parematar instead of
>> path uri.
>>
>> any advice how to handle this ?
>>
>>
>>
>>
>>
>> recv 964 bytes from udp/[192.168.50.60]:5060 at 22:18:57.909700:
>>    -----------------------------------------------------------
>> -------------
>>    REGISTER sip:192.168.50.60 SIP/2.0
>>    Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKe929.
>> b6aa43dd1eefa9e4756af7d31d65066e.0
>>    Via: SIP/2.0/WSS 192.0.2.110;rport=61744;received=192.168.200.77;
>> branch=z9hG4bK3179897
>>    Max-Forwards: 69
>>    To: <sip:1001 at 192.168.50.60>
>>    From: <sip:1001 at 192.168.50.60>;tag=vq30modpgd
>>    Call-ID: dlnrna9o4ngn25fb9d1vi2
>>    CSeq: 192 REGISTER
>>    Authorization: Digest algorithm=MD5, username="1001",
>> realm="192.168.50.60", nonce="22c52b4b-f795-4caa-bb9b-15bbd87564f7",
>> uri="sip:192.168.50.60", response="a073277bc9c1fcd681de661cb418d838",
>> qop=auth, cnonce="euqeokf2d1gr", nc=00000001
>>    Contact: <sip:1001 at 192.0.2.110;transport=ws>;reg-id=1;+sip.
>> instance="<urn:uuid:587f4694-7e55-46dc-a858-579a82556a20>";expires=600
>>    Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
>>    Supported: path, gruu, 100rel, outbound
>>    User-Agent: SIP.js/0.7.0 BB
>>    Content-Length: 0
>>    Path: <sip:192.168.50.60;lr;received=sip:192.168.200.77:
>> 61744%3Btransport%3Dws>
>>
>>    -----------------------------------------------------------
>> -------------
>> send 738 bytes to udp/[192.168.50.60]:5060 at 22:18:57.950968:
>>    -----------------------------------------------------------
>> -------------
>>    SIP/2.0 200 OK
>>    Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKe929.
>> b6aa43dd1eefa9e4756af7d31d65066e.0
>>    Via: SIP/2.0/WSS 192.0.2.110;rport=61744;received=192.168.200.77;
>> branch=z9hG4bK3179897
>>    From: <sip:1001 at 192.168.50.60>;tag=vq30modpgd
>>    To: <sip:1001 at 192.168.50.60>;tag=eK8SmZmcU7UBr
>>    Call-ID: dlnrna9o4ngn25fb9d1vi2
>>    CSeq: 192 REGISTER
>>    Contact: <sip:1001 at 192.0.2.110;transport=ws>;expires=600
>>    Date: Fri, 09 Feb 2018 21:18:57 GMT
>>    User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~
>> b1b21d0695~64bit
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY
>>    Supported: timer, path, replaces
>>    Path: <sip:192.168.50.60>;lr;received=sip:192.168.200.77:
>> 61744%3Btransport%3Dws
>>    Content-Length: 0
>>
>>    -----------------------------------------------------------
>> -------------
>>
>>
>>
>> in freeswitch i the registration:
>>
>> Call-ID:        dlnrna9o4ngn25fb9d1vi2
>> User:           1001 at mydomain
>> Contact:        "" <sip:1001 at 192.0.2.110;transport=ws;fs_nat=yes;fs_
>> path=sip%3A192.168.50.60%3Blr%3Breceived%3Dsip%3A192.168.
>> 200.77%3A61744%3Btransport%3Dws>
>> Agent:          SIP.js/0.7.0 BB
>> Status:         Registered(WS-NAT)(unknown) EXP(2018-02-09 22:29:57)
>> EXPSECS(510)
>> Ping-Status:    Reachable
>> Ping-Time:      0.00
>> Host:           FS01
>> IP:             192.168.50.60
>> Port:           5060
>> Auth-User:      1001
>> Auth-Realm:     192.168.50.60
>> MWI-Account:    1001 at mydomain
>>
>>
>>
>>
>> the register contain a path header with received=<original uri>.
>>
>> when i try to originate a call to this registered user the call goes
>> nowhere :=)
>>
>>
>> freeswitch at FS01> bgapi originate {origination_caller_id_number=
>> 1002}sofia/internal/1001%mydomain &echo()
>> +OK Job-UUID: 200e2197-a6de-42d9-b4bb-63917223bc53
>>
>> 2018-02-09 22:25:40.500100 [DEBUG] switch_ivr_originate.c:2142 Parsing
>> global variables
>> 2018-02-09 22:25:40.500100 [NOTICE] switch_channel.c:1104 New Channel
>> sofia/internal/1001 [327826f9-8a8c-46c7-b3ed-8bf22852df88]
>> 2018-02-09 22:25:40.500100 [DEBUG] mod_sofia.c:4819 (sofia/internal/1001)
>> State Change CS_NEW -> CS_INIT
>> 2018-02-09 22:25:40.500100 [DEBUG] switch_core_state_machine.c:584
>> (sofia/internal/1001) Running State Change CS_INIT (Cur 1 Tot 217)
>> 2018-02-09 22:25:40.500100 [DEBUG] switch_core_state_machine.c:627
>> (sofia/internal/1001) State INIT
>> 2018-02-09 22:25:40.500100 [DEBUG] mod_sofia.c:90 sofia/internal/1001
>> SOFIA INIT
>> 2018-02-09 22:25:40.500100 [DEBUG] sofia_glue.c:1264
>> sip:192.168.50.60;lr;received=sip:192.168.200.77:61744;transport=ws
>> Setting proxy route to sofia/internal/1001
>> 2018-02-09 22:25:40.500100 [DEBUG] sofia_glue.c:1295 sofia/internal/1001
>> sending invite version: 1.6.19 git b1b21d0 2017-11-20 16:34:16Z 64bit
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 dragos.oancea at vonage.com  Wed Feb 14 14:00:36 2018
From: dragos.oancea at vonage.com (Oancea, Dragos)
Date: Wed, 14 Feb 2018 14:00:36 +0000
Subject: [Freeswitch-users] Calls recording issue
In-Reply-To: <CAMVzwA4-R9n20LowbPTg190zzDSBV6zmHhcnN3pa8Acey7R5Xw@mail.gmail.com>
References: <b462b754-0a38-4524-95c4-41e0d55b0f60@Spark>
 <CAMVzwA4-R9n20LowbPTg190zzDSBV6zmHhcnN3pa8Acey7R5Xw@mail.gmail.com>
Message-ID: <CAOsE5hjShExSfUAjCNOYO0=PDC476sT6xfFH0EEdwSRC8K0CLw@mail.gmail.com>

Igor, there's not such codec string as "OPUS at 8000" , it's "OPUS at 8000h" .
Don't forget the "h" , otherwise it will load the default OPUS at 48 khz.
Is it possible you're playing the file at a different sample rate than the
one with which is recorded ?
Try also setting this channel variable before starting the recording, like
here below or by using uuid_setvar:

<action application="set" data="record_sample_rate=8000"/>

Regards,
Dragos


On Tue, Feb 13, 2018 at 7:41 PM, Shivang Soni <ssoni at ucdavis.edu> wrote:

> Hi,
> I am Shivang Soni. I am currently working on a web development project
> which requires accessing the microphone. I can able to access microphone if
> the application runs on my local system but cannot able to access
> microphone when running the same application on our college interaction
> server. So I tried to explore about Freeswitch by coming to know it is
> possible through that. As I am new to FreeSWITCH, it would be great if you
> can guide me on how to resolve this problem using the free switch. I have
> currently installed free-switch on the Debian 8 OS on AWS. As I am not
> familiar with the free switch I am also curious to know how to keep my
> particular application on FreeSWITCH so that it can access the microphone.
>
> I am subscribed to mailing list but don't know any page where to post, how
> to create new thread..
>
> Looking forward to your response,
> Regards,
> Shivang Soni.
>
>
> On Tue, Feb 13, 2018 at 8:36 AM, Igor Olhovskiy <igorolhovskiy at gmail.com>
> wrote:
>
>> Hi!
>> I’ve faced an issue with call recording.
>> I’m recording with uuid_record in wav (also tried with mp3) with these
>> legs
>>
>> A: WebRTC/SIPJS/Opus at 8000
>> B: SIP/PCMA
>>
>> In this case I get clear recording only on leg B. Leg A is sounds like
>> tape on SLOW (really SLOW) playing. Mostly from leg A i hear smth like
>> (A-R-R….!)
>> In a case if leg A is SIP with any codec, it’s ok.
>>
>> FS 1.6.20
>>
>> Could it be a bug in FS?
>>
>> Regards, Igor
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 hunterj91 at hotmail.com  Wed Feb 14 14:01:07 2018
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Wed, 14 Feb 2018 14:01:07 +0000
Subject: [Freeswitch-users] Mod Verto - Chat Version 1.6.19
Message-ID: <DB4PR02MB033634006BD31E5A27528B1DBDF50@DB4PR02MB0336.eurprd02.prod.outlook.com>

Hi Guys,

I am testing verto on version 1.6.19 and it works great for Video and Audio, however when using the

function setupChat I am seeing some strange behaviour.

I am using extension 1002 and I connect to conference conference_1.

I then send a chat message (to test functionality) and it sends it fine, however in the chat history  window the same message is repeated, I assume by the server, and in the console I see;

2018-02-14 13:54:06.369878 [ALERT] mod_verto.c:604 WRITE 82.8.193.240:55456 [{
        "jsonrpc":      "2.0",
        "id":   3450,
        "method":       "verto.info",
        "params":       {
                "msg":  {
                        "from": "1002 at 172.31.12.49",
                        "to":   "1002 at 172.31.12.49",
                        "body": "hello",
                        "from_user":    "1002",
                        "from_host":    "172.31.12.49",
                        "from_full":    "1002",
                        "from_msg_name":        "Jon H",
                        "from_msg_number":      "Consumer123456"
                }
        }
}]
2018-02-14 13:54:06.369878 [ALERT] mod_verto.c:604 WRITE 82.8.193.240:55456 [{
        "jsonrpc":      "2.0",
        "id":   3451,
        "method":       "verto.info",
        "params":       {
                "msg":  {
                        "from": "conference_1 at 172.31.12.49",
                        "to":   "1002 at 172.31.12.49",
                        "body": "Hello, you said: ",
                        "to_user":      "1002",
                        "to_host":      "172.31.12.49",
                        "to_full":      "1002",
                        "to_msg_name":  "Jon H",
                        "to_msg_number":        "Consumer123456"
                }
        }
}]
2018-02-14 13:54:06.369878 [ALERT] mod_verto.c:604 WRITE 82.8.193.240:55456 [{
        "jsonrpc":      "2.0",
        "id":   3452,
        "method":       "verto.info",
        "params":       {
                "msg":  {
                        "from": "1002 at 172.31.12.49",
                        "to":   "1002 at 172.31.12.49",
                        "body": "Hello, you said: ",
                        "to_user":      "1002",
                        "to_host":      "172.31.12.49",
                        "to_full":      "1002",
                        "to_msg_name":  "JonH",
                        "to_msg_number":        "Consumer123456"
                }
        }
}]

Where are these messages coming with "Hello, you said:" from?

Is this a configuration issue on my part?

Many thanks!

Jon

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From hunterj91 at hotmail.com  Wed Feb 14 16:45:26 2018
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Wed, 14 Feb 2018 16:45:26 +0000
Subject: [Freeswitch-users] Mod Verto - Chat Version 1.6.19
In-Reply-To: <DB4PR02MB033624A47B625FDEC2368C64BDF50@DB4PR02MB0336.eurprd02.prod.outlook.com>
References: <DB4PR02MB033634006BD31E5A27528B1DBDF50@DB4PR02MB0336.eurprd02.prod.outlook.com>
 <CAEJMVkDgTPcUeJW_isNXSV1EG5zCh0Br4KiWWDoKzVYDLrCR3Q@mail.gmail.com>
 <DB4PR02MB0336E21808A9E598A8933EBDBDF50@DB4PR02MB0336.eurprd02.prod.outlook.com>,
 <CAEJMVkCAOj+GwgvnasaYuqyZGdiLJavr870L27uQQk8yxS+0eA@mail.gmail.com>,
 <DB4PR02MB033624A47B625FDEC2368C64BDF50@DB4PR02MB0336.eurprd02.prod.outlook.com>
Message-ID: <DB4PR02MB0336379FBA485EC8BF3D5C08BDF50@DB4PR02MB0336.eurprd02.prod.outlook.com>


Sorry Brian  I missed off the users list!

Apologies !

Jon

________________________________
From: Jonathan Hunter <hunterj91 at hotmail.com>
Sent: 14 February 2018 16:44
To: Brian West
Subject: Re: [Freeswitch-users] Mod Verto - Chat Version 1.6.19


Hi Brian,


No problem glad we are on same page now 😊


I am just testing with the mod_verto demo and I was getting the same issue, however it appears to be due to mod_sms being loaded, as this sends back;


 <action application="reply" data="Hello, you said: ${_body}"/>


However what is best practice when I just want to use chat within verto having subscribed to live events?


Should I make sure mod_sms and so on are not enabled?


Thanks!


Jon


________________________________
From: brian at freeswitch.org <brian at freeswitch.org> on behalf of Brian West <brian at freeswitch.com>
Sent: 14 February 2018 16:00
To: Jonathan Hunter
Subject: Re: [Freeswitch-users] Mod Verto - Chat Version 1.6.19

Cool, I thought you were the Jonathan Hunter that works at TimeWarner, Its always interesting to see how larger companies use FreeSWITCH.

There may need to be some work done in this area to get it working like you want, can you show me what you've tried so far?

/b


On Wed, Feb 14, 2018 at 9:28 AM, Jonathan Hunter <hunterj91 at hotmail.com<mailto:hunterj91 at hotmail.com>> wrote:

Hi Brian,


Thanks for the response! Not sure what you mean by TW/Spectrum? :)


I am building a webrtc client to use with verto, and I am testing and have same issue when using the verto demo version, using verto.js on my server.


Just wondered where these chat messages are being generated from and if I can stop them or if its configuration.


Happy to provide any further detail!


Thanks


Jon


________________________________
From: brian at freeswitch.org<mailto:brian at freeswitch.org> <brian at freeswitch.org<mailto:brian at freeswitch.org>> on behalf of Brian West <brian at freeswitch.com<mailto:brian at freeswitch.com>>
Sent: 14 February 2018 14:16
To: hunterj91 at hotmail.com<mailto:hunterj91 at hotmail.com>
Subject: Re: [Freeswitch-users] Mod Verto - Chat Version 1.6.19

What exactly is TW/Spectrum doing with FreeSWITCH?

/b


On Wed, Feb 14, 2018 at 9:01 AM, Jonathan Hunter <hunterj91 at hotmail.com<mailto:hunterj91 at hotmail.com>> wrote:

Hi Guys,

I am testing verto on version 1.6.19 and it works great for Video and Audio, however when using the

function setupChat I am seeing some strange behaviour.

I am using extension 1002 and I connect to conference conference_1.

I then send a chat message (to test functionality) and it sends it fine, however in the chat history  window the same message is repeated, I assume by the server, and in the console I see;

2018-02-14 13:54:06.369878 [ALERT] mod_verto.c:604 WRITE 82.8.193.240:55456<http://82.8.193.240:55456> [{
        "jsonrpc":      "2.0",
        "id":   3450,
        "method":       "verto.info<http://verto.info>",
        "params":       {
                "msg":  {
                        "from": "1002 at 172.31.12.49<mailto:1002 at 172.31.12.49>",
                        "to":   "1002 at 172.31.12.49<mailto:1002 at 172.31.12.49>",
                        "body": "hello",
                        "from_user":    "1002",
                        "from_host":    "172.31.12.49",
                        "from_full":    "1002",
                        "from_msg_name":        "Jon H",
                        "from_msg_number":      "Consumer123456"
                }
        }
}]
2018-02-14 13:54:06.369878 [ALERT] mod_verto.c:604 WRITE 82.8.193.240:55456<http://82.8.193.240:55456> [{
        "jsonrpc":      "2.0",
        "id":   3451,
        "method":       "verto.info<http://verto.info>",
        "params":       {
                "msg":  {
                        "from": "conference_1 at 172.31.12.49<mailto:conference_1 at 172.31.12.49>",
                        "to":   "1002 at 172.31.12.49<mailto:1002 at 172.31.12.49>",
                        "body": "Hello, you said: ",
                        "to_user":      "1002",
                        "to_host":      "172.31.12.49",
                        "to_full":      "1002",
                        "to_msg_name":  "Jon H",
                        "to_msg_number":        "Consumer123456"
                }
        }
}]
2018-02-14 13:54:06.369878 [ALERT] mod_verto.c:604 WRITE 82.8.193.240:55456<http://82.8.193.240:55456> [{
        "jsonrpc":      "2.0",
        "id":   3452,
        "method":       "verto.info<http://verto.info>",
        "params":       {
                "msg":  {
                        "from": "1002 at 172.31.12.49<mailto:1002 at 172.31.12.49>",
                        "to":   "1002 at 172.31.12.49<mailto:1002 at 172.31.12.49>",
                        "body": "Hello, you said: ",
                        "to_user":      "1002",
                        "to_host":      "172.31.12.49",
                        "to_full":      "1002",
                        "to_msg_name":  "JonH",
                        "to_msg_number":        "Consumer123456"
                }
        }
}]

Where are these messages coming with "Hello, you said:" from?

Is this a configuration issue on my part?

Many thanks!

Jon


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<http://freeswitchsolutions.com>

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

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



--
[https://docs.google.com/uc?export=download&id=1xswZRZyVDo0WQhaemK47pU266yzDRmi0&revid=0B2xnT7i45ngrMTVKM1dpSHZIN28zU0QzbW9xeVF6RXFyRHhBPQ]

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<tel:(918)%20424-9378>

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

[color-facebook-96.png]<https://www.facebook.com/freeswitch/>[color-twitter-96.png]<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>



--
[https://docs.google.com/uc?export=download&id=1xswZRZyVDo0WQhaemK47pU266yzDRmi0&revid=0B2xnT7i45ngrMTVKM1dpSHZIN28zU0QzbW9xeVF6RXFyRHhBPQ]

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/>

[color-facebook-96.png]<https://www.facebook.com/freeswitch/>[color-twitter-96.png]<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From davidwaf at gmail.com  Wed Feb 14 19:12:58 2018
From: davidwaf at gmail.com (David Wafula)
Date: Wed, 14 Feb 2018 21:12:58 +0200
Subject: [Freeswitch-users] How to unregister verto user
Message-ID: <CAK9Ws2foDzCr_m_B3FsJSwuq3ZChpSwALUAbF7b6BnUEB_oKYg@mail.gmail.com>

Hi all,
I have a verto user who keeps closing browser window without logging out
and i need to force unregistration periodically.
What command can be used to accomplish this?

The only commands am able to use are:
verto status/xmlstatus

Kind regards

--
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From gregor at infomedia.si  Wed Feb 14 20:48:09 2018
From: gregor at infomedia.si (Gregor Nanger)
Date: Wed, 14 Feb 2018 21:48:09 +0100
Subject: [Freeswitch-users] How to unregister verto user
In-Reply-To: <CAK9Ws2foDzCr_m_B3FsJSwuq3ZChpSwALUAbF7b6BnUEB_oKYg@mail.gmail.com>
References: <CAK9Ws2foDzCr_m_B3FsJSwuq3ZChpSwALUAbF7b6BnUEB_oKYg@mail.gmail.com>
Message-ID: <CAC7ttFHwsG80vVKMr8Jc8k-9T6SrEE2_XNHRSB4me80oVo8caw@mail.gmail.com>

As far as I know, you can't. Session will expire eventualy.

2018-02-14 20:12 GMT+01:00 David Wafula <davidwaf at gmail.com>:

> Hi all,
> I have a verto user who keeps closing browser window without logging out
> and i need to force unregistration periodically.
> What command can be used to accomplish this?
>
> The only commands am able to use are:
> verto status/xmlstatus
>
> Kind regards
>
> --
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
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From tculjaga at gmail.com  Wed Feb 14 21:56:12 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 14 Feb 2018 22:56:12 +0100
Subject: [Freeswitch-users] How to unregister verto user
In-Reply-To: <CAC7ttFHwsG80vVKMr8Jc8k-9T6SrEE2_XNHRSB4me80oVo8caw@mail.gmail.com>
References: <CAK9Ws2foDzCr_m_B3FsJSwuq3ZChpSwALUAbF7b6BnUEB_oKYg@mail.gmail.com>
 <CAC7ttFHwsG80vVKMr8Jc8k-9T6SrEE2_XNHRSB4me80oVo8caw@mail.gmail.com>
Message-ID: <CAJoKKdXcO-EEGKUaPczKo6LL_LE6qJ_v=-RYsKkLX+F-cJa2Hw@mail.gmail.com>

simply


var verto;

...
...
verto = new $.verto(confg)

...
...


verto.logout()









On 14 February 2018 at 21:48, Gregor Nanger <gregor at infomedia.si> wrote:

> As far as I know, you can't. Session will expire eventualy.
>
> 2018-02-14 20:12 GMT+01:00 David Wafula <davidwaf at gmail.com>:
>
>> Hi all,
>> I have a verto user who keeps closing browser window without logging out
>> and i need to force unregistration periodically.
>> What command can be used to accomplish this?
>>
>> The only commands am able to use are:
>> verto status/xmlstatus
>>
>> Kind regards
>>
>> --
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
>
> --
> Gregor Nanger
>
> *CTO*
> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
> • www.infomedia.si
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 tculjaga at gmail.com  Wed Feb 14 21:56:59 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 14 Feb 2018 22:56:59 +0100
Subject: [Freeswitch-users] How to unregister verto user
In-Reply-To: <CAJoKKdXcO-EEGKUaPczKo6LL_LE6qJ_v=-RYsKkLX+F-cJa2Hw@mail.gmail.com>
References: <CAK9Ws2foDzCr_m_B3FsJSwuq3ZChpSwALUAbF7b6BnUEB_oKYg@mail.gmail.com>
 <CAC7ttFHwsG80vVKMr8Jc8k-9T6SrEE2_XNHRSB4me80oVo8caw@mail.gmail.com>
 <CAJoKKdXcO-EEGKUaPczKo6LL_LE6qJ_v=-RYsKkLX+F-cJa2Hw@mail.gmail.com>
Message-ID: <CAJoKKdXHHgM0sShRFFETDREzmXUz+D4UniiTbun-VCDf9Sfx5w@mail.gmail.com>

freeswitch at bladeranger> verto status
                     Name          Type
   Data      State
=================================================================================================
           nexios-cc-ipv4       profile                     wss:
192.168.5.150:8082      RUNNING
nexios-cc-ipv4::agent3/nexios at 192.168.5.150     client
  95.178.240.214:51297      CONN_REG (WSS)
=================================================================================================
1 profile , 1 client

2018-02-14 22:53:07.886865 [DEBUG] mod_verto.c:1882 BAD READ -1000
2018-02-14 22:53:07.886865 [DEBUG] mod_verto.c:2025 95.178.240.214:51297
Ending client thread.
2018-02-14 22:53:07.886865 [DEBUG] mod_verto.c:2033 95.178.240.214:51297
Thread ended
freeswitch at bladeranger>
freeswitch at bladeranger> verto status
                     Name          Type
   Data      State
=================================================================================================
           nexios-cc-ipv4       profile                     wss:
192.168.5.150:8082      RUNNING
=================================================================================================
1 profile , 0 clients




On 14 February 2018 at 22:56, Tihomir Culjaga <tculjaga at gmail.com> wrote:

>
> simply
>
>
> var verto;
>
> ...
> ...
> verto = new $.verto(confg)
>
> ...
> ...
>
>
> verto.logout()
>
>
>
>
>
>
>
>
>
> On 14 February 2018 at 21:48, Gregor Nanger <gregor at infomedia.si> wrote:
>
>> As far as I know, you can't. Session will expire eventualy.
>>
>> 2018-02-14 20:12 GMT+01:00 David Wafula <davidwaf at gmail.com>:
>>
>>> Hi all,
>>> I have a verto user who keeps closing browser window without logging out
>>> and i need to force unregistration periodically.
>>> What command can be used to accomplish this?
>>>
>>> The only commands am able to use are:
>>> verto status/xmlstatus
>>>
>>> Kind regards
>>>
>>> --
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>>
>>
>>
>>
>> --
>> Gregor Nanger
>>
>> *CTO*
>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>> • www.infomedia.si
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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 gregor at infomedia.si  Thu Feb 15 09:28:00 2018
From: gregor at infomedia.si (Gregor Nanger)
Date: Thu, 15 Feb 2018 10:28:00 +0100
Subject: [Freeswitch-users] How to unregister verto user
In-Reply-To: <CAJoKKdXHHgM0sShRFFETDREzmXUz+D4UniiTbun-VCDf9Sfx5w@mail.gmail.com>
References: <CAK9Ws2foDzCr_m_B3FsJSwuq3ZChpSwALUAbF7b6BnUEB_oKYg@mail.gmail.com>
 <CAC7ttFHwsG80vVKMr8Jc8k-9T6SrEE2_XNHRSB4me80oVo8caw@mail.gmail.com>
 <CAJoKKdXcO-EEGKUaPczKo6LL_LE6qJ_v=-RYsKkLX+F-cJa2Hw@mail.gmail.com>
 <CAJoKKdXHHgM0sShRFFETDREzmXUz+D4UniiTbun-VCDf9Sfx5w@mail.gmail.com>
Message-ID: <CAC7ttFHAdaZ2nsAkT7MzOi5RuWNO=hbSLPLdC8VbLeYqfdQeKw@mail.gmail.com>

Tihomir, is there any command to unregister on server side?

David, as Tihomir suggests, you can capture onunload event in JS and
unregister via javascript.


2018-02-14 22:56 GMT+01:00 Tihomir Culjaga <tculjaga at gmail.com>:

> freeswitch at bladeranger> verto status
>                      Name          Type
>    Data      State
> ============================================================
> =====================================
>            nexios-cc-ipv4       profile                     wss:
> 192.168.5.150:8082      RUNNING
> nexios-cc-ipv4::agent3/nexios at 192.168.5.150     client
>     95.178.240.214:51297      CONN_REG (WSS)
> ============================================================
> =====================================
> 1 profile , 1 client
>
> 2018-02-14 22:53:07.886865 [DEBUG] mod_verto.c:1882 BAD READ -1000
> 2018-02-14 22:53:07.886865 [DEBUG] mod_verto.c:2025 95.178.240.214:51297
> Ending client thread.
> 2018-02-14 22:53:07.886865 [DEBUG] mod_verto.c:2033 95.178.240.214:51297
> Thread ended
> freeswitch at bladeranger>
> freeswitch at bladeranger> verto status
>                      Name          Type
>    Data      State
> ============================================================
> =====================================
>            nexios-cc-ipv4       profile                     wss:
> 192.168.5.150:8082      RUNNING
> ============================================================
> =====================================
> 1 profile , 0 clients
>
>
>
>
> On 14 February 2018 at 22:56, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
>>
>> simply
>>
>>
>> var verto;
>>
>> ...
>> ...
>> verto = new $.verto(confg)
>>
>> ...
>> ...
>>
>>
>> verto.logout()
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On 14 February 2018 at 21:48, Gregor Nanger <gregor at infomedia.si> wrote:
>>
>>> As far as I know, you can't. Session will expire eventualy.
>>>
>>> 2018-02-14 20:12 GMT+01:00 David Wafula <davidwaf at gmail.com>:
>>>
>>>> Hi all,
>>>> I have a verto user who keeps closing browser window without logging
>>>> out and i need to force unregistration periodically.
>>>> What command can be used to accomplish this?
>>>>
>>>> The only commands am able to use are:
>>>> verto status/xmlstatus
>>>>
>>>> Kind regards
>>>>
>>>> --
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.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/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Gregor Nanger
>>>
>>> *CTO*
>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>>> • www.infomedia.si
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
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From tculjaga at gmail.com  Thu Feb 15 14:23:07 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 15 Feb 2018 15:23:07 +0100
Subject: [Freeswitch-users] How to unregister verto user
In-Reply-To: <CAC7ttFHAdaZ2nsAkT7MzOi5RuWNO=hbSLPLdC8VbLeYqfdQeKw@mail.gmail.com>
References: <CAK9Ws2foDzCr_m_B3FsJSwuq3ZChpSwALUAbF7b6BnUEB_oKYg@mail.gmail.com>
 <CAC7ttFHwsG80vVKMr8Jc8k-9T6SrEE2_XNHRSB4me80oVo8caw@mail.gmail.com>
 <CAJoKKdXcO-EEGKUaPczKo6LL_LE6qJ_v=-RYsKkLX+F-cJa2Hw@mail.gmail.com>
 <CAJoKKdXHHgM0sShRFFETDREzmXUz+D4UniiTbun-VCDf9Sfx5w@mail.gmail.com>
 <CAC7ttFHAdaZ2nsAkT7MzOi5RuWNO=hbSLPLdC8VbLeYqfdQeKw@mail.gmail.com>
Message-ID: <CAJoKKdVo_WCeVCnWO8iCmqrSLRVBzkJaj-DYSeoS1M9AeBnbmw@mail.gmail.com>

no, there is nothing you can do on server side :(

i was thinking to introduce it in mod verto

On 15 February 2018 at 10:28, Gregor Nanger <gregor at infomedia.si> wrote:

> Tihomir, is there any command to unregister on server side?
>
> David, as Tihomir suggests, you can capture onunload event in JS and
> unregister via javascript.
>
>
> 2018-02-14 22:56 GMT+01:00 Tihomir Culjaga <tculjaga at gmail.com>:
>
>> freeswitch at bladeranger> verto status
>>                      Name          Type
>>      Data      State
>> ============================================================
>> =====================================
>>            nexios-cc-ipv4       profile                     wss:
>> 192.168.5.150:8082      RUNNING
>> nexios-cc-ipv4::agent3/nexios at 192.168.5.150     client
>>       95.178.240.214:51297      CONN_REG (WSS)
>> ============================================================
>> =====================================
>> 1 profile , 1 client
>>
>> 2018-02-14 22:53:07.886865 [DEBUG] mod_verto.c:1882 BAD READ -1000
>> 2018-02-14 22:53:07.886865 [DEBUG] mod_verto.c:2025 95.178.240.214:51297
>> Ending client thread.
>> 2018-02-14 22:53:07.886865 [DEBUG] mod_verto.c:2033 95.178.240.214:51297
>> Thread ended
>> freeswitch at bladeranger>
>> freeswitch at bladeranger> verto status
>>                      Name          Type
>>      Data      State
>> ============================================================
>> =====================================
>>            nexios-cc-ipv4       profile                     wss:
>> 192.168.5.150:8082      RUNNING
>> ============================================================
>> =====================================
>> 1 profile , 0 clients
>>
>>
>>
>>
>> On 14 February 2018 at 22:56, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>>
>>>
>>> simply
>>>
>>>
>>> var verto;
>>>
>>> ...
>>> ...
>>> verto = new $.verto(confg)
>>>
>>> ...
>>> ...
>>>
>>>
>>> verto.logout()
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On 14 February 2018 at 21:48, Gregor Nanger <gregor at infomedia.si> wrote:
>>>
>>>> As far as I know, you can't. Session will expire eventualy.
>>>>
>>>> 2018-02-14 20:12 GMT+01:00 David Wafula <davidwaf at gmail.com>:
>>>>
>>>>> Hi all,
>>>>> I have a verto user who keeps closing browser window without logging
>>>>> out and i need to force unregistration periodically.
>>>>> What command can be used to accomplish this?
>>>>>
>>>>> The only commands am able to use are:
>>>>> verto status/xmlstatus
>>>>>
>>>>> Kind regards
>>>>>
>>>>> --
>>>>>
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.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/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Gregor Nanger
>>>>
>>>> *CTO*
>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>>>> • www.infomedia.si
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.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/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
>
> --
> Gregor Nanger
>
> *CTO*
> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
> • www.infomedia.si
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 rmundkowsky at ets.org  Wed Feb 14 22:09:04 2018
From: rmundkowsky at ets.org (Mundkowsky, Robert)
Date: Wed, 14 Feb 2018 22:09:04 +0000
Subject: [Freeswitch-users] How to unregister verto user
In-Reply-To: <CAJoKKdXcO-EEGKUaPczKo6LL_LE6qJ_v=-RYsKkLX+F-cJa2Hw@mail.gmail.com>
References: <CAK9Ws2foDzCr_m_B3FsJSwuq3ZChpSwALUAbF7b6BnUEB_oKYg@mail.gmail.com>
 <CAC7ttFHwsG80vVKMr8Jc8k-9T6SrEE2_XNHRSB4me80oVo8caw@mail.gmail.com>
 <CAJoKKdXcO-EEGKUaPczKo6LL_LE6qJ_v=-RYsKkLX+F-cJa2Hw@mail.gmail.com>
Message-ID: <CY1PR07MB2377EDDCFE5ADE5609376504BEF50@CY1PR07MB2377.namprd07.prod.outlook.com>

In the old days, web browsers did not have an event for window getting closed.  I suspect that is still true, so you do not have an event to trigger this javascript code you mentioned.

Robert

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir Culjaga
Sent: Wednesday, February 14, 2018 4:56 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] How to unregister verto user


simply


var verto;

...
...
verto = new $.verto(confg)

...
...


verto.logout()









On 14 February 2018 at 21:48, Gregor Nanger <gregor at infomedia.si<mailto:gregor at infomedia.si>> wrote:
As far as I know, you can't. Session will expire eventualy.


2018-02-14 20:12 GMT+01:00 David Wafula <davidwaf at gmail.com<mailto:davidwaf at gmail.com>>:
Hi all,
I have a verto user who keeps closing browser window without logging out and i need to force unregistration periodically.
What command can be used to accomplish this?

The only commands am able to use are:
verto status/xmlstatus

Kind regards

--


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639751357&sdata=CcbH%2BpF8NcktyJIZ%2Bk5TMOVLaKKKsQ%2FUDLfvAUQDw9Q%3D&reserved=0>

Official FreeSWITCH Sites
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FreeSWITCH-users mailing list
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--
Gregor Nanger


CTO
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.infomedia.si&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=3vN9VFnMCEkuOcYyAEGfDgiFkyDqrRLJVKmzqt0tgkw%3D&reserved=0>


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=eLKFdLAeGvA9zv9OFT6WARumrpW7Pr5INhdLXAadqS4%3D&reserved=0>

Official FreeSWITCH Sites
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From douglas.davenport at gmail.com  Thu Feb 15 16:14:30 2018
From: douglas.davenport at gmail.com (Douglas Davenport)
Date: Thu, 15 Feb 2018 11:14:30 -0500
Subject: [Freeswitch-users] T38 re-invite, 488 Not Acceptable Here,
 Freeswitch 1.6.20
In-Reply-To: <1518082436.566731.1263750016.385885A8@webmail.messagingengine.com>
References: <1518082436.566731.1263750016.385885A8@webmail.messagingengine.com>
Message-ID: <CAOsHgtvS5X-tr8P-r8xXTV91ORqzo7bCedYKHtps1rpbDFZPxg@mail.gmail.com>

If you are trying to recieve a fax you should be calling rxfax not bridging
to a user.

<action application="rxfax" />

On Thu, Feb 8, 2018 at 4:33 AM, Michael Avers <michael at mailworks.org> wrote:

> Hello,
>
> I understand the T38 behavior changed after 1.6.13. I watched Brian's
> video from last year, read the docs, but I just don't see what else I would
> need for a very simple scenario for receiving a fax with the receiving end
> (ATA) re-inviting T38.
>
> PSTN > FS > HT801 ATA > T38 RE-INVITE > FS > PSTN
>
> The ATA does re-invite, however FS rejects immediately with 488 Not
> Acceptable Here. I'm using the following dialplan. I also tried to use
> export instead of set, but same result.
>
> <extension name="faxdid1" continue="false">
>  <condition field="destination_number" expression="^(4154447777
> <(415)%20444-7777>)$">
>    <action application="set" data="ignore_early_media=true"/>
>    <action application="set" data="fax_enable_t38_request=true"/>
>    <action application="set" data="fax_enable_t38=true"/>
>    <action application="bridge" data="user/6050 at sip.domain.org"/>
>  </condition>
> </extension>
>
> Is there anything obvious that I'm missing here?
>
> Thanks
> Mike
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 brian at freeswitch.com  Thu Feb 15 18:40:02 2018
From: brian at freeswitch.com (Brian West)
Date: Thu, 15 Feb 2018 18:40:02 +0000
Subject: [Freeswitch-users] How to unregister verto user
In-Reply-To: <CY1PR07MB2377EDDCFE5ADE5609376504BEF50@CY1PR07MB2377.namprd07.prod.outlook.com>
References: <CAK9Ws2foDzCr_m_B3FsJSwuq3ZChpSwALUAbF7b6BnUEB_oKYg@mail.gmail.com>
 <CAC7ttFHwsG80vVKMr8Jc8k-9T6SrEE2_XNHRSB4me80oVo8caw@mail.gmail.com>
 <CAJoKKdXcO-EEGKUaPczKo6LL_LE6qJ_v=-RYsKkLX+F-cJa2Hw@mail.gmail.com>
 <CY1PR07MB2377EDDCFE5ADE5609376504BEF50@CY1PR07MB2377.namprd07.prod.outlook.com>
Message-ID: <CAEJMVkAkJmFhdgzAa8xRneUaMS2ypPgFJMDMt4nGdKkYz5jF4g@mail.gmail.com>

When the socket disconnects it should unregister them, what exactly is the
issue your having?

/b

On Thu, Feb 15, 2018 at 11:39 AM Mundkowsky, Robert <rmundkowsky at ets.org>
wrote:

> In the old days, web browsers did not have an event for window getting
> closed.  I suspect that is still true, so you do not have an event to
> trigger this javascript code you mentioned.
>
>
>
> Robert
>
>
>
> *From:* FreeSWITCH-users [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tihomir
> Culjaga
> *Sent:* Wednesday, February 14, 2018 4:56 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] How to unregister verto user
>
>
>
>
>
> simply
>
>
>
>
>
> var verto;
>
>
>
> ...
>
> ...
>
> verto = new $.verto(confg)
>
>
>
> ...
>
> ...
>
>
>
>
>
> verto.logout()
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> On 14 February 2018 at 21:48, Gregor Nanger <gregor at infomedia.si> wrote:
>
> As far as I know, you can't. Session will expire eventualy.
>
>
>
> 2018-02-14 20:12 GMT+01:00 David Wafula <davidwaf at gmail.com>:
>
> Hi all,
>
> I have a verto user who keeps closing browser window without logging out
> and i need to force unregistration periodically.
>
> What command can be used to accomplish this?
>
>
>
> The only commands am able to use are:
>
> verto status/xmlstatus
>
>
>
> Kind regards
>
>
>
> --
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639751357&sdata=CcbH%2BpF8NcktyJIZ%2Bk5TMOVLaKKKsQ%2FUDLfvAUQDw9Q%3D&reserved=0>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639751357&sdata=9Wm6rFEQgBXnz9HNo%2Fxc3IMFSaG%2FpK77r2H4Vj%2FsZLU%3D&reserved=0>
> http://confluence.freeswitch.org
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=KlSM7XhL1Ejl1jtoKYKg2ospXA6GhN22zAM9trQeYnE%3D&reserved=0>
> http://www.cluecon.com
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=o0zN0nAx4fKo%2Fc%2F7036WuO4m0VjmG8vWo%2B969P5CaD8%3D&reserved=0>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=SaLZXzXEHJ3kQdj6opvrEzbqYc1X6%2Bwlc402FuDaC2k%3D&reserved=0>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=LXfaaJRsN3QbszvuAUm3cVwP5qWyiB6dTt4acP%2FCxXg%3D&reserved=0>
> http://www.freeswitch.org
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>
>
>
>
>
> --
>
> *Gregor Nanger*
>
>
>
> *CTO*
> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
> • www.infomedia.si
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.infomedia.si&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=3vN9VFnMCEkuOcYyAEGfDgiFkyDqrRLJVKmzqt0tgkw%3D&reserved=0>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=eLKFdLAeGvA9zv9OFT6WARumrpW7Pr5INhdLXAadqS4%3D&reserved=0>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
> http://confluence.freeswitch.org
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=KlSM7XhL1Ejl1jtoKYKg2ospXA6GhN22zAM9trQeYnE%3D&reserved=0>
> http://www.cluecon.com
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=o0zN0nAx4fKo%2Fc%2F7036WuO4m0VjmG8vWo%2B969P5CaD8%3D&reserved=0>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=SaLZXzXEHJ3kQdj6opvrEzbqYc1X6%2Bwlc402FuDaC2k%3D&reserved=0>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=LXfaaJRsN3QbszvuAUm3cVwP5qWyiB6dTt4acP%2FCxXg%3D&reserved=0>
> http://www.freeswitch.org
> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>
>
>
> ------------------------------
>
> This e-mail and any files transmitted with it may contain privileged or
> confidential information. It is solely for use by the individual for whom
> it is intended, even if addressed incorrectly. If you received this e-mail
> in error, please notify the sender; do not disclose, copy, distribute, or
> take any action in reliance on the contents of this information; and delete
> it from your system. Any other use of this e-mail is prohibited.
>
> Thank you for your compliance.
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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

-- 

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/>

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From gregor at infomedia.si  Thu Feb 15 18:51:50 2018
From: gregor at infomedia.si (Gregor Nanger)
Date: Thu, 15 Feb 2018 18:51:50 +0000
Subject: [Freeswitch-users] How to unregister verto user
In-Reply-To: <CAEJMVkAkJmFhdgzAa8xRneUaMS2ypPgFJMDMt4nGdKkYz5jF4g@mail.gmail.com>
References: <CAK9Ws2foDzCr_m_B3FsJSwuq3ZChpSwALUAbF7b6BnUEB_oKYg@mail.gmail.com>
 <CAC7ttFHwsG80vVKMr8Jc8k-9T6SrEE2_XNHRSB4me80oVo8caw@mail.gmail.com>
 <CAJoKKdXcO-EEGKUaPczKo6LL_LE6qJ_v=-RYsKkLX+F-cJa2Hw@mail.gmail.com>
 <CY1PR07MB2377EDDCFE5ADE5609376504BEF50@CY1PR07MB2377.namprd07.prod.outlook.com>
 <CAEJMVkAkJmFhdgzAa8xRneUaMS2ypPgFJMDMt4nGdKkYz5jF4g@mail.gmail.com>
Message-ID: <CAC7ttFHbhxMvYnSASdorzokT7Ob2Mwduh77w_Rs++2ng49gqtw@mail.gmail.com>

Yes, it is still the same. You have on unload event, but this captures
everything, refresh, go to other link and even close. It's better than
nothing.

On Thu, Feb 15, 2018, 19:43 Brian West <brian at freeswitch.com> wrote:

> When the socket disconnects it should unregister them, what exactly is the
> issue your having?
>
> /b
>
> On Thu, Feb 15, 2018 at 11:39 AM Mundkowsky, Robert <rmundkowsky at ets.org>
> wrote:
>
>> In the old days, web browsers did not have an event for window getting
>> closed.  I suspect that is still true, so you do not have an event to
>> trigger this javascript code you mentioned.
>>
>>
>>
>> Robert
>>
>>
>>
>> *From:* FreeSWITCH-users [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tihomir
>> Culjaga
>> *Sent:* Wednesday, February 14, 2018 4:56 PM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* Re: [Freeswitch-users] How to unregister verto user
>>
>>
>>
>>
>>
>> simply
>>
>>
>>
>>
>>
>> var verto;
>>
>>
>>
>> ...
>>
>> ...
>>
>> verto = new $.verto(confg)
>>
>>
>>
>> ...
>>
>> ...
>>
>>
>>
>>
>>
>> verto.logout()
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On 14 February 2018 at 21:48, Gregor Nanger <gregor at infomedia.si> wrote:
>>
>> As far as I know, you can't. Session will expire eventualy.
>>
>>
>>
>> 2018-02-14 20:12 GMT+01:00 David Wafula <davidwaf at gmail.com>:
>>
>> Hi all,
>>
>> I have a verto user who keeps closing browser window without logging out
>> and i need to force unregistration periodically.
>>
>> What command can be used to accomplish this?
>>
>>
>>
>> The only commands am able to use are:
>>
>> verto status/xmlstatus
>>
>>
>>
>> Kind regards
>>
>>
>>
>> --
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639751357&sdata=CcbH%2BpF8NcktyJIZ%2Bk5TMOVLaKKKsQ%2FUDLfvAUQDw9Q%3D&reserved=0>
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639751357&sdata=9Wm6rFEQgBXnz9HNo%2Fxc3IMFSaG%2FpK77r2H4Vj%2FsZLU%3D&reserved=0>
>> http://confluence.freeswitch.org
>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=KlSM7XhL1Ejl1jtoKYKg2ospXA6GhN22zAM9trQeYnE%3D&reserved=0>
>> http://www.cluecon.com
>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=o0zN0nAx4fKo%2Fc%2F7036WuO4m0VjmG8vWo%2B969P5CaD8%3D&reserved=0>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=SaLZXzXEHJ3kQdj6opvrEzbqYc1X6%2Bwlc402FuDaC2k%3D&reserved=0>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=LXfaaJRsN3QbszvuAUm3cVwP5qWyiB6dTt4acP%2FCxXg%3D&reserved=0>
>> http://www.freeswitch.org
>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>>
>>
>>
>>
>>
>> --
>>
>> *Gregor Nanger*
>>
>>
>>
>> *CTO*
>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>> • www.infomedia.si
>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.infomedia.si&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=3vN9VFnMCEkuOcYyAEGfDgiFkyDqrRLJVKmzqt0tgkw%3D&reserved=0>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=eLKFdLAeGvA9zv9OFT6WARumrpW7Pr5INhdLXAadqS4%3D&reserved=0>
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>> http://confluence.freeswitch.org
>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=KlSM7XhL1Ejl1jtoKYKg2ospXA6GhN22zAM9trQeYnE%3D&reserved=0>
>> http://www.cluecon.com
>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=o0zN0nAx4fKo%2Fc%2F7036WuO4m0VjmG8vWo%2B969P5CaD8%3D&reserved=0>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=SaLZXzXEHJ3kQdj6opvrEzbqYc1X6%2Bwlc402FuDaC2k%3D&reserved=0>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=LXfaaJRsN3QbszvuAUm3cVwP5qWyiB6dTt4acP%2FCxXg%3D&reserved=0>
>> http://www.freeswitch.org
>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>>
>>
>>
>> ------------------------------
>>
>> This e-mail and any files transmitted with it may contain privileged or
>> confidential information. It is solely for use by the individual for whom
>> it is intended, even if addressed incorrectly. If you received this e-mail
>> in error, please notify the sender; do not disclose, copy, distribute, or
>> take any action in reliance on the contents of this information; and delete
>> it from your system. Any other use of this e-mail is prohibited.
>>
>> Thank you for your compliance.
>> ------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>
> --
>
> 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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
> color-twitter-96.png]
> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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

-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
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From michael at mailworks.org  Thu Feb 15 19:38:58 2018
From: michael at mailworks.org (Michael Avers)
Date: Thu, 15 Feb 2018 12:38:58 -0700
Subject: [Freeswitch-users] T38 re-invite, 488 Not Acceptable Here,
 Freeswitch 1.6.20
In-Reply-To: <CAOsHgtvS5X-tr8P-r8xXTV91ORqzo7bCedYKHtps1rpbDFZPxg@mail.gmail.com>
References: <1518082436.566731.1263750016.385885A8@webmail.messagingengine.com>
 <CAOsHgtvS5X-tr8P-r8xXTV91ORqzo7bCedYKHtps1rpbDFZPxg@mail.gmail.com>
Message-ID: <1518723538.1972790.1272279528.5EE92770@webmail.messagingengine.com>

I'm trying to get a registered ATA to send/receive


On Thu, Feb 15, 2018, at 9:14 AM, Douglas Davenport wrote:
> If you are trying to recieve a fax you should be calling rxfax not
> bridging to a user.> 
> <action application="rxfax" />
> 
> On Thu, Feb 8, 2018 at 4:33 AM, Michael Avers
> <michael at mailworks.org> wrote:>> __
>> 
>> Hello,
>> 
>> I understand the T38 behavior changed after 1.6.13. I watched Brian's
>> video from last year, read the docs, but I just don't see what else I
>> would need for a very simple scenario for receiving a fax with the
>> receiving end (ATA) re-inviting T38.>> 
>> PSTN > FS > HT801 ATA > T38 RE-INVITE > FS > PSTN
>> 
>> The ATA does re-invite, however FS rejects immediately with 488 Not
>> Acceptable Here. I'm using the following dialplan. I also tried to
>> use export instead of set, but same result.>> 
>> <extension name="faxdid1" continue="false">
>>  <condition field="destination_number"
>>  expression="^(4154447777[1])$">>>    <action application="set" data="ignore_early_media=true"/>
>>    <action application="set" data="fax_enable_t38_request=true"/>
>>    <action application="set" data="fax_enable_t38=true"/>
>>    <action application="bridge" data="user/6050 at sip.domain.org"/>
>>  </condition>
>> </extension>
>> 
>> Is there anything obvious that I'm missing here?
>> 
>> 
>> Thanks
>> Mike
>> 
>> __________________________________________________________________-
>> _______>>  Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>>  Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
> ___________________________________________________________________-
> ________> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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


Links:

  1. tel:(415)%20444-7777
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From brian at freeswitch.com  Thu Feb 15 19:43:53 2018
From: brian at freeswitch.com (Brian West)
Date: Thu, 15 Feb 2018 14:43:53 -0500
Subject: [Freeswitch-users] T38 re-invite, 488 Not Acceptable Here,
 Freeswitch 1.6.20
In-Reply-To: <1518723538.1972790.1272279528.5EE92770@webmail.messagingengine.com>
References: <1518082436.566731.1263750016.385885A8@webmail.messagingengine.com>
 <CAOsHgtvS5X-tr8P-r8xXTV91ORqzo7bCedYKHtps1rpbDFZPxg@mail.gmail.com>
 <1518723538.1972790.1272279528.5EE92770@webmail.messagingengine.com>
Message-ID: <CAEJMVkC6h=W3CvfzMPo1kPvxHr1q3g3SLhzvwmdKjhLi+=9w4Q@mail.gmail.com>

global_setvar t38_passthru=true

/b


On Thu, Feb 15, 2018 at 2:38 PM, Michael Avers <michael at mailworks.org>
wrote:

> I'm trying to get a registered ATA to send/receive
>
>
> On Thu, Feb 15, 2018, at 9:14 AM, Douglas Davenport wrote:
>
> If you are trying to recieve a fax you should be calling rxfax not
> bridging to a user.
>
> <action application="rxfax" />
>
> On Thu, Feb 8, 2018 at 4:33 AM, Michael Avers <michael at mailworks.org>
> wrote:
>
>
>
> Hello,
>
> I understand the T38 behavior changed after 1.6.13. I watched Brian's
> video from last year, read the docs, but I just don't see what else I would
> need for a very simple scenario for receiving a fax with the receiving end
> (ATA) re-inviting T38.
>
> PSTN > FS > HT801 ATA > T38 RE-INVITE > FS > PSTN
>
> The ATA does re-invite, however FS rejects immediately with 488 Not
> Acceptable Here. I'm using the following dialplan. I also tried to use
> export instead of set, but same result.
>
> <extension name="faxdid1" continue="false">
>  <condition field="destination_number" expression="^(4154447777
> <(415)%20444-7777>)$">
>    <action application="set" data="ignore_early_media=true"/>
>    <action application="set" data="fax_enable_t38_request=true"/>
>    <action application="set" data="fax_enable_t38=true"/>
>    <action application="bridge" data="user/6050 at sip.domain.org"/>
>  </condition>
> </extension>
>
> Is there anything obvious that I'm missing here?
>
>
> Thanks
> Mike
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
> *_________________________________________________________________________*
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From olegstolyar at gmail.com  Thu Feb 15 23:31:28 2018
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Thu, 15 Feb 2018 15:31:28 -0800
Subject: [Freeswitch-users] Media Timeout disconnect with Verto
Message-ID: <CAC8JMOL6n7KJFFqb=aUxgsfiJrKx01M-0A9GCp=T_93xHhxdMQ@mail.gmail.com>

Hi guys,

what is the verto equivalent of rtp-timeout-sec param?  How do I control
when to disconnect a verto call when there is no media coming from the
browser client?
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From tculjaga at gmail.com  Fri Feb 16 07:59:35 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 16 Feb 2018 08:59:35 +0100
Subject: [Freeswitch-users] How to unregister verto user
In-Reply-To: <CAC7ttFHbhxMvYnSASdorzokT7Ob2Mwduh77w_Rs++2ng49gqtw@mail.gmail.com>
References: <CAK9Ws2foDzCr_m_B3FsJSwuq3ZChpSwALUAbF7b6BnUEB_oKYg@mail.gmail.com>
 <CAC7ttFHwsG80vVKMr8Jc8k-9T6SrEE2_XNHRSB4me80oVo8caw@mail.gmail.com>
 <CAJoKKdXcO-EEGKUaPczKo6LL_LE6qJ_v=-RYsKkLX+F-cJa2Hw@mail.gmail.com>
 <CY1PR07MB2377EDDCFE5ADE5609376504BEF50@CY1PR07MB2377.namprd07.prod.outlook.com>
 <CAEJMVkAkJmFhdgzAa8xRneUaMS2ypPgFJMDMt4nGdKkYz5jF4g@mail.gmail.com>
 <CAC7ttFHbhxMvYnSASdorzokT7Ob2Mwduh77w_Rs++2ng49gqtw@mail.gmail.com>
Message-ID: <CAJoKKdWa4-+HnwuRrhc5CW_WuiZwnah91Sf3MuCVWWaR8P8z1A@mail.gmail.com>

just call logout on your logout button click... otherwise wss will timeout
on browser close / crash eventually

On 15 February 2018 at 19:51, Gregor Nanger <gregor at infomedia.si> wrote:

> Yes, it is still the same. You have on unload event, but this captures
> everything, refresh, go to other link and even close. It's better than
> nothing.
>
>
> On Thu, Feb 15, 2018, 19:43 Brian West <brian at freeswitch.com> wrote:
>
>> When the socket disconnects it should unregister them, what exactly is
>> the issue your having?
>>
>> /b
>>
>> On Thu, Feb 15, 2018 at 11:39 AM Mundkowsky, Robert <rmundkowsky at ets.org>
>> wrote:
>>
>>> In the old days, web browsers did not have an event for window getting
>>> closed.  I suspect that is still true, so you do not have an event to
>>> trigger this javascript code you mentioned.
>>>
>>>
>>>
>>> Robert
>>>
>>>
>>>
>>> *From:* FreeSWITCH-users [mailto:freeswitch-users-
>>> bounces at lists.freeswitch.org] *On Behalf Of *Tihomir Culjaga
>>> *Sent:* Wednesday, February 14, 2018 4:56 PM
>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> *Subject:* Re: [Freeswitch-users] How to unregister verto user
>>>
>>>
>>>
>>>
>>>
>>> simply
>>>
>>>
>>>
>>>
>>>
>>> var verto;
>>>
>>>
>>>
>>> ...
>>>
>>> ...
>>>
>>> verto = new $.verto(confg)
>>>
>>>
>>>
>>> ...
>>>
>>> ...
>>>
>>>
>>>
>>>
>>>
>>> verto.logout()
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On 14 February 2018 at 21:48, Gregor Nanger <gregor at infomedia.si> wrote:
>>>
>>> As far as I know, you can't. Session will expire eventualy.
>>>
>>>
>>>
>>> 2018-02-14 20:12 GMT+01:00 David Wafula <davidwaf at gmail.com>:
>>>
>>> Hi all,
>>>
>>> I have a verto user who keeps closing browser window without logging out
>>> and i need to force unregistration periodically.
>>>
>>> What command can be used to accomplish this?
>>>
>>>
>>>
>>> The only commands am able to use are:
>>>
>>> verto status/xmlstatus
>>>
>>>
>>>
>>> Kind regards
>>>
>>>
>>>
>>> --
>>>
>>>
>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639751357&sdata=CcbH%2BpF8NcktyJIZ%2Bk5TMOVLaKKKsQ%2FUDLfvAUQDw9Q%3D&reserved=0>
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639751357&sdata=9Wm6rFEQgBXnz9HNo%2Fxc3IMFSaG%2FpK77r2H4Vj%2FsZLU%3D&reserved=0>
>>> http://confluence.freeswitch.org
>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=KlSM7XhL1Ejl1jtoKYKg2ospXA6GhN22zAM9trQeYnE%3D&reserved=0>
>>> http://www.cluecon.com
>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=o0zN0nAx4fKo%2Fc%2F7036WuO4m0VjmG8vWo%2B969P5CaD8%3D&reserved=0>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=SaLZXzXEHJ3kQdj6opvrEzbqYc1X6%2Bwlc402FuDaC2k%3D&reserved=0>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=LXfaaJRsN3QbszvuAUm3cVwP5qWyiB6dTt4acP%2FCxXg%3D&reserved=0>
>>> http://www.freeswitch.org
>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>>>
>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Gregor Nanger*
>>>
>>>
>>>
>>> *CTO*
>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>>> • www.infomedia.si
>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.infomedia.si&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=3vN9VFnMCEkuOcYyAEGfDgiFkyDqrRLJVKmzqt0tgkw%3D&reserved=0>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=eLKFdLAeGvA9zv9OFT6WARumrpW7Pr5INhdLXAadqS4%3D&reserved=0>
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>>> http://confluence.freeswitch.org
>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=KlSM7XhL1Ejl1jtoKYKg2ospXA6GhN22zAM9trQeYnE%3D&reserved=0>
>>> http://www.cluecon.com
>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=o0zN0nAx4fKo%2Fc%2F7036WuO4m0VjmG8vWo%2B969P5CaD8%3D&reserved=0>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=SaLZXzXEHJ3kQdj6opvrEzbqYc1X6%2Bwlc402FuDaC2k%3D&reserved=0>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=LXfaaJRsN3QbszvuAUm3cVwP5qWyiB6dTt4acP%2FCxXg%3D&reserved=0>
>>> http://www.freeswitch.org
>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>>>
>>>
>>>
>>> ------------------------------
>>>
>>> This e-mail and any files transmitted with it may contain privileged or
>>> confidential information. It is solely for use by the individual for whom
>>> it is intended, even if addressed incorrectly. If you received this e-mail
>>> in error, please notify the sender; do not disclose, copy, distribute, or
>>> take any action in reliance on the contents of this information; and delete
>>> it from your system. Any other use of this e-mail is prohibited.
>>>
>>> Thank you for your compliance.
>>> ------------------------------
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>
>> --
>>
>> 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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
>> color-twitter-96.png]
>> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>
> --
> Gregor Nanger
>
> *CTO*
> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
> • www.infomedia.si
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 alexandr.popov at iqoption.com  Fri Feb 16 09:35:47 2018
From: alexandr.popov at iqoption.com (Alexandr Popov)
Date: Fri, 16 Feb 2018 11:35:47 +0200
Subject: [Freeswitch-users] How to unregister verto user
In-Reply-To: <CAJoKKdWa4-+HnwuRrhc5CW_WuiZwnah91Sf3MuCVWWaR8P8z1A@mail.gmail.com>
References: <CAK9Ws2foDzCr_m_B3FsJSwuq3ZChpSwALUAbF7b6BnUEB_oKYg@mail.gmail.com>
 <CAC7ttFHwsG80vVKMr8Jc8k-9T6SrEE2_XNHRSB4me80oVo8caw@mail.gmail.com>
 <CAJoKKdXcO-EEGKUaPczKo6LL_LE6qJ_v=-RYsKkLX+F-cJa2Hw@mail.gmail.com>
 <CY1PR07MB2377EDDCFE5ADE5609376504BEF50@CY1PR07MB2377.namprd07.prod.outlook.com>
 <CAEJMVkAkJmFhdgzAa8xRneUaMS2ypPgFJMDMt4nGdKkYz5jF4g@mail.gmail.com>
 <CAC7ttFHbhxMvYnSASdorzokT7Ob2Mwduh77w_Rs++2ng49gqtw@mail.gmail.com>
 <CAJoKKdWa4-+HnwuRrhc5CW_WuiZwnah91Sf3MuCVWWaR8P8z1A@mail.gmail.com>
Message-ID: <CAKeZRhvy0EA4Qh=Z89JWFP17jBPGm4=83=ON1oBLrYWJoO1tEw@mail.gmail.com>

Brian, sometimes it could stuck.

2018-02-16 9:59 GMT+02:00 Tihomir Culjaga <tculjaga at gmail.com>:

> just call logout on your logout button click... otherwise wss will timeout
> on browser close / crash eventually
>
> On 15 February 2018 at 19:51, Gregor Nanger <gregor at infomedia.si> wrote:
>
>> Yes, it is still the same. You have on unload event, but this captures
>> everything, refresh, go to other link and even close. It's better than
>> nothing.
>>
>>
>> On Thu, Feb 15, 2018, 19:43 Brian West <brian at freeswitch.com> wrote:
>>
>>> When the socket disconnects it should unregister them, what exactly is
>>> the issue your having?
>>>
>>> /b
>>>
>>> On Thu, Feb 15, 2018 at 11:39 AM Mundkowsky, Robert <rmundkowsky at ets.org>
>>> wrote:
>>>
>>>> In the old days, web browsers did not have an event for window getting
>>>> closed.  I suspect that is still true, so you do not have an event to
>>>> trigger this javascript code you mentioned.
>>>>
>>>>
>>>>
>>>> Robert
>>>>
>>>>
>>>>
>>>> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc
>>>> es at lists.freeswitch.org] *On Behalf Of *Tihomir Culjaga
>>>> *Sent:* Wednesday, February 14, 2018 4:56 PM
>>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> *Subject:* Re: [Freeswitch-users] How to unregister verto user
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> simply
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> var verto;
>>>>
>>>>
>>>>
>>>> ...
>>>>
>>>> ...
>>>>
>>>> verto = new $.verto(confg)
>>>>
>>>>
>>>>
>>>> ...
>>>>
>>>> ...
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> verto.logout()
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On 14 February 2018 at 21:48, Gregor Nanger <gregor at infomedia.si>
>>>> wrote:
>>>>
>>>> As far as I know, you can't. Session will expire eventualy.
>>>>
>>>>
>>>>
>>>> 2018-02-14 20:12 GMT+01:00 David Wafula <davidwaf at gmail.com>:
>>>>
>>>> Hi all,
>>>>
>>>> I have a verto user who keeps closing browser window without logging
>>>> out and i need to force unregistration periodically.
>>>>
>>>> What command can be used to accomplish this?
>>>>
>>>>
>>>>
>>>> The only commands am able to use are:
>>>>
>>>> verto status/xmlstatus
>>>>
>>>>
>>>>
>>>> Kind regards
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639751357&sdata=CcbH%2BpF8NcktyJIZ%2Bk5TMOVLaKKKsQ%2FUDLfvAUQDw9Q%3D&reserved=0>
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639751357&sdata=9Wm6rFEQgBXnz9HNo%2Fxc3IMFSaG%2FpK77r2H4Vj%2FsZLU%3D&reserved=0>
>>>> http://confluence.freeswitch.org
>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=KlSM7XhL1Ejl1jtoKYKg2ospXA6GhN22zAM9trQeYnE%3D&reserved=0>
>>>> http://www.cluecon.com
>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=o0zN0nAx4fKo%2Fc%2F7036WuO4m0VjmG8vWo%2B969P5CaD8%3D&reserved=0>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=SaLZXzXEHJ3kQdj6opvrEzbqYc1X6%2Bwlc402FuDaC2k%3D&reserved=0>
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>>>> freeswitch-users
>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=LXfaaJRsN3QbszvuAUm3cVwP5qWyiB6dTt4acP%2FCxXg%3D&reserved=0>
>>>> http://www.freeswitch.org
>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> *Gregor Nanger*
>>>>
>>>>
>>>>
>>>> *CTO*
>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>>>> • www.infomedia.si
>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.infomedia.si&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=3vN9VFnMCEkuOcYyAEGfDgiFkyDqrRLJVKmzqt0tgkw%3D&reserved=0>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=eLKFdLAeGvA9zv9OFT6WARumrpW7Pr5INhdLXAadqS4%3D&reserved=0>
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>>>> http://confluence.freeswitch.org
>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=KlSM7XhL1Ejl1jtoKYKg2ospXA6GhN22zAM9trQeYnE%3D&reserved=0>
>>>> http://www.cluecon.com
>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=o0zN0nAx4fKo%2Fc%2F7036WuO4m0VjmG8vWo%2B969P5CaD8%3D&reserved=0>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=SaLZXzXEHJ3kQdj6opvrEzbqYc1X6%2Bwlc402FuDaC2k%3D&reserved=0>
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>>>> freeswitch-users
>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=LXfaaJRsN3QbszvuAUm3cVwP5qWyiB6dTt4acP%2FCxXg%3D&reserved=0>
>>>> http://www.freeswitch.org
>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>>>>
>>>>
>>>>
>>>> ------------------------------
>>>>
>>>> This e-mail and any files transmitted with it may contain privileged or
>>>> confidential information. It is solely for use by the individual for whom
>>>> it is intended, even if addressed incorrectly. If you received this e-mail
>>>> in error, please notify the sender; do not disclose, copy, distribute, or
>>>> take any action in reliance on the contents of this information; and delete
>>>> it from your system. Any other use of this e-mail is prohibited.
>>>>
>>>> Thank you for your compliance.
>>>> ------------------------------
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.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
>>>
>>> --
>>>
>>> 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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
>>> color-twitter-96.png]
>>> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>
>> --
>> Gregor Nanger
>>
>> *CTO*
>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>> • www.infomedia.si
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 davidwaf at gmail.com  Fri Feb 16 09:40:49 2018
From: davidwaf at gmail.com (David Wafula)
Date: Fri, 16 Feb 2018 11:40:49 +0200
Subject: [Freeswitch-users] How to unregister verto user
In-Reply-To: <CAKeZRhvy0EA4Qh=Z89JWFP17jBPGm4=83=ON1oBLrYWJoO1tEw@mail.gmail.com>
References: <CAK9Ws2foDzCr_m_B3FsJSwuq3ZChpSwALUAbF7b6BnUEB_oKYg@mail.gmail.com>
 <CAC7ttFHwsG80vVKMr8Jc8k-9T6SrEE2_XNHRSB4me80oVo8caw@mail.gmail.com>
 <CAJoKKdXcO-EEGKUaPczKo6LL_LE6qJ_v=-RYsKkLX+F-cJa2Hw@mail.gmail.com>
 <CY1PR07MB2377EDDCFE5ADE5609376504BEF50@CY1PR07MB2377.namprd07.prod.outlook.com>
 <CAEJMVkAkJmFhdgzAa8xRneUaMS2ypPgFJMDMt4nGdKkYz5jF4g@mail.gmail.com>
 <CAC7ttFHbhxMvYnSASdorzokT7Ob2Mwduh77w_Rs++2ng49gqtw@mail.gmail.com>
 <CAJoKKdWa4-+HnwuRrhc5CW_WuiZwnah91Sf3MuCVWWaR8P8z1A@mail.gmail.com>
 <CAKeZRhvy0EA4Qh=Z89JWFP17jBPGm4=83=ON1oBLrYWJoO1tEw@mail.gmail.com>
Message-ID: <CAK9Ws2fntCyfkrTDLbtACWqb-n7pBw9T2=3ds9+9Y5THf3B14A@mail.gmail.com>

I have tested ...if the user closes browser without explicitly logging out,
the session is never terminated and it seems the user stays in the
conference forever.

On Fri, Feb 16, 2018 at 11:35 AM, Alexandr Popov <
alexandr.popov at iqoption.com> wrote:

> Brian, sometimes it could stuck.
>
> 2018-02-16 9:59 GMT+02:00 Tihomir Culjaga <tculjaga at gmail.com>:
>
>> just call logout on your logout button click... otherwise wss will
>> timeout on browser close / crash eventually
>>
>> On 15 February 2018 at 19:51, Gregor Nanger <gregor at infomedia.si> wrote:
>>
>>> Yes, it is still the same. You have on unload event, but this captures
>>> everything, refresh, go to other link and even close. It's better than
>>> nothing.
>>>
>>>
>>> On Thu, Feb 15, 2018, 19:43 Brian West <brian at freeswitch.com> wrote:
>>>
>>>> When the socket disconnects it should unregister them, what exactly is
>>>> the issue your having?
>>>>
>>>> /b
>>>>
>>>> On Thu, Feb 15, 2018 at 11:39 AM Mundkowsky, Robert <
>>>> rmundkowsky at ets.org> wrote:
>>>>
>>>>> In the old days, web browsers did not have an event for window getting
>>>>> closed.  I suspect that is still true, so you do not have an event to
>>>>> trigger this javascript code you mentioned.
>>>>>
>>>>>
>>>>>
>>>>> Robert
>>>>>
>>>>>
>>>>>
>>>>> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc
>>>>> es at lists.freeswitch.org] *On Behalf Of *Tihomir Culjaga
>>>>> *Sent:* Wednesday, February 14, 2018 4:56 PM
>>>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>> *Subject:* Re: [Freeswitch-users] How to unregister verto user
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> simply
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> var verto;
>>>>>
>>>>>
>>>>>
>>>>> ...
>>>>>
>>>>> ...
>>>>>
>>>>> verto = new $.verto(confg)
>>>>>
>>>>>
>>>>>
>>>>> ...
>>>>>
>>>>> ...
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> verto.logout()
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On 14 February 2018 at 21:48, Gregor Nanger <gregor at infomedia.si>
>>>>> wrote:
>>>>>
>>>>> As far as I know, you can't. Session will expire eventualy.
>>>>>
>>>>>
>>>>>
>>>>> 2018-02-14 20:12 GMT+01:00 David Wafula <davidwaf at gmail.com>:
>>>>>
>>>>> Hi all,
>>>>>
>>>>> I have a verto user who keeps closing browser window without logging
>>>>> out and i need to force unregistration periodically.
>>>>>
>>>>> What command can be used to accomplish this?
>>>>>
>>>>>
>>>>>
>>>>> The only commands am able to use are:
>>>>>
>>>>> verto status/xmlstatus
>>>>>
>>>>>
>>>>>
>>>>> Kind regards
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639751357&sdata=CcbH%2BpF8NcktyJIZ%2Bk5TMOVLaKKKsQ%2FUDLfvAUQDw9Q%3D&reserved=0>
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639751357&sdata=9Wm6rFEQgBXnz9HNo%2Fxc3IMFSaG%2FpK77r2H4Vj%2FsZLU%3D&reserved=0>
>>>>> http://confluence.freeswitch.org
>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=KlSM7XhL1Ejl1jtoKYKg2ospXA6GhN22zAM9trQeYnE%3D&reserved=0>
>>>>> http://www.cluecon.com
>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=o0zN0nAx4fKo%2Fc%2F7036WuO4m0VjmG8vWo%2B969P5CaD8%3D&reserved=0>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=SaLZXzXEHJ3kQdj6opvrEzbqYc1X6%2Bwlc402FuDaC2k%3D&reserved=0>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=LXfaaJRsN3QbszvuAUm3cVwP5qWyiB6dTt4acP%2FCxXg%3D&reserved=0>
>>>>> http://www.freeswitch.org
>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>> *Gregor Nanger*
>>>>>
>>>>>
>>>>>
>>>>> *CTO*
>>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>>>>> • www.infomedia.si
>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.infomedia.si&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=3vN9VFnMCEkuOcYyAEGfDgiFkyDqrRLJVKmzqt0tgkw%3D&reserved=0>
>>>>>
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=eLKFdLAeGvA9zv9OFT6WARumrpW7Pr5INhdLXAadqS4%3D&reserved=0>
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>>>>> http://confluence.freeswitch.org
>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=KlSM7XhL1Ejl1jtoKYKg2ospXA6GhN22zAM9trQeYnE%3D&reserved=0>
>>>>> http://www.cluecon.com
>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=o0zN0nAx4fKo%2Fc%2F7036WuO4m0VjmG8vWo%2B969P5CaD8%3D&reserved=0>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=SaLZXzXEHJ3kQdj6opvrEzbqYc1X6%2Bwlc402FuDaC2k%3D&reserved=0>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=LXfaaJRsN3QbszvuAUm3cVwP5qWyiB6dTt4acP%2FCxXg%3D&reserved=0>
>>>>> http://www.freeswitch.org
>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------
>>>>>
>>>>> This e-mail and any files transmitted with it may contain privileged
>>>>> or confidential information. It is solely for use by the individual for
>>>>> whom it is intended, even if addressed incorrectly. If you received this
>>>>> e-mail in error, please notify the sender; do not disclose, copy,
>>>>> distribute, or take any action in reliance on the contents of this
>>>>> information; and delete it from your system. Any other use of this e-mail
>>>>> is prohibited.
>>>>>
>>>>> Thank you for your compliance.
>>>>> ------------------------------
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.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/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>
>>>> --
>>>>
>>>> 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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
>>>> color-twitter-96.png]
>>>> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.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/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>
>>> --
>>> Gregor Nanger
>>>
>>> *CTO*
>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>>> • www.infomedia.si
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 
David W
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From brian at freeswitch.com  Fri Feb 16 12:48:45 2018
From: brian at freeswitch.com (Brian West)
Date: Fri, 16 Feb 2018 07:48:45 -0500
Subject: [Freeswitch-users] How to unregister verto user
In-Reply-To: <CAK9Ws2fntCyfkrTDLbtACWqb-n7pBw9T2=3ds9+9Y5THf3B14A@mail.gmail.com>
References: <CAK9Ws2foDzCr_m_B3FsJSwuq3ZChpSwALUAbF7b6BnUEB_oKYg@mail.gmail.com>
 <CAC7ttFHwsG80vVKMr8Jc8k-9T6SrEE2_XNHRSB4me80oVo8caw@mail.gmail.com>
 <CAJoKKdXcO-EEGKUaPczKo6LL_LE6qJ_v=-RYsKkLX+F-cJa2Hw@mail.gmail.com>
 <CY1PR07MB2377EDDCFE5ADE5609376504BEF50@CY1PR07MB2377.namprd07.prod.outlook.com>
 <CAEJMVkAkJmFhdgzAa8xRneUaMS2ypPgFJMDMt4nGdKkYz5jF4g@mail.gmail.com>
 <CAC7ttFHbhxMvYnSASdorzokT7Ob2Mwduh77w_Rs++2ng49gqtw@mail.gmail.com>
 <CAJoKKdWa4-+HnwuRrhc5CW_WuiZwnah91Sf3MuCVWWaR8P8z1A@mail.gmail.com>
 <CAKeZRhvy0EA4Qh=Z89JWFP17jBPGm4=83=ON1oBLrYWJoO1tEw@mail.gmail.com>
 <CAK9Ws2fntCyfkrTDLbtACWqb-n7pBw9T2=3ds9+9Y5THf3B14A@mail.gmail.com>
Message-ID: <CAEJMVkAAEe3-dEjTxrwPhRj=7urEorgpFQw+UZGhuNBZmM2ZNQ@mail.gmail.com>

Please file a JIRA.

/b


On Fri, Feb 16, 2018 at 4:40 AM, David Wafula <davidwaf at gmail.com> wrote:

> I have tested ...if the user closes browser without explicitly logging
> out, the session is never terminated and it seems the user stays in the
> conference forever.
>
> On Fri, Feb 16, 2018 at 11:35 AM, Alexandr Popov <
> alexandr.popov at iqoption.com> wrote:
>
>> Brian, sometimes it could stuck.
>>
>> 2018-02-16 9:59 GMT+02:00 Tihomir Culjaga <tculjaga at gmail.com>:
>>
>>> just call logout on your logout button click... otherwise wss will
>>> timeout on browser close / crash eventually
>>>
>>> On 15 February 2018 at 19:51, Gregor Nanger <gregor at infomedia.si> wrote:
>>>
>>>> Yes, it is still the same. You have on unload event, but this captures
>>>> everything, refresh, go to other link and even close. It's better than
>>>> nothing.
>>>>
>>>>
>>>> On Thu, Feb 15, 2018, 19:43 Brian West <brian at freeswitch.com> wrote:
>>>>
>>>>> When the socket disconnects it should unregister them, what exactly is
>>>>> the issue your having?
>>>>>
>>>>> /b
>>>>>
>>>>> On Thu, Feb 15, 2018 at 11:39 AM Mundkowsky, Robert <
>>>>> rmundkowsky at ets.org> wrote:
>>>>>
>>>>>> In the old days, web browsers did not have an event for window
>>>>>> getting closed.  I suspect that is still true, so you do not have an event
>>>>>> to trigger this javascript code you mentioned.
>>>>>>
>>>>>>
>>>>>>
>>>>>> Robert
>>>>>>
>>>>>>
>>>>>>
>>>>>> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc
>>>>>> es at lists.freeswitch.org] *On Behalf Of *Tihomir Culjaga
>>>>>> *Sent:* Wednesday, February 14, 2018 4:56 PM
>>>>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>>> *Subject:* Re: [Freeswitch-users] How to unregister verto user
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> simply
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> var verto;
>>>>>>
>>>>>>
>>>>>>
>>>>>> ...
>>>>>>
>>>>>> ...
>>>>>>
>>>>>> verto = new $.verto(confg)
>>>>>>
>>>>>>
>>>>>>
>>>>>> ...
>>>>>>
>>>>>> ...
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> verto.logout()
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On 14 February 2018 at 21:48, Gregor Nanger <gregor at infomedia.si>
>>>>>> wrote:
>>>>>>
>>>>>> As far as I know, you can't. Session will expire eventualy.
>>>>>>
>>>>>>
>>>>>>
>>>>>> 2018-02-14 20:12 GMT+01:00 David Wafula <davidwaf at gmail.com>:
>>>>>>
>>>>>> Hi all,
>>>>>>
>>>>>> I have a verto user who keeps closing browser window without logging
>>>>>> out and i need to force unregistration periodically.
>>>>>>
>>>>>> What command can be used to accomplish this?
>>>>>>
>>>>>>
>>>>>>
>>>>>> The only commands am able to use are:
>>>>>>
>>>>>> verto status/xmlstatus
>>>>>>
>>>>>>
>>>>>>
>>>>>> Kind regards
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639751357&sdata=CcbH%2BpF8NcktyJIZ%2Bk5TMOVLaKKKsQ%2FUDLfvAUQDw9Q%3D&reserved=0>
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639751357&sdata=9Wm6rFEQgBXnz9HNo%2Fxc3IMFSaG%2FpK77r2H4Vj%2FsZLU%3D&reserved=0>
>>>>>> http://confluence.freeswitch.org
>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=KlSM7XhL1Ejl1jtoKYKg2ospXA6GhN22zAM9trQeYnE%3D&reserved=0>
>>>>>> http://www.cluecon.com
>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=o0zN0nAx4fKo%2Fc%2F7036WuO4m0VjmG8vWo%2B969P5CaD8%3D&reserved=0>
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=SaLZXzXEHJ3kQdj6opvrEzbqYc1X6%2Bwlc402FuDaC2k%3D&reserved=0>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>> switch-users
>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=LXfaaJRsN3QbszvuAUm3cVwP5qWyiB6dTt4acP%2FCxXg%3D&reserved=0>
>>>>>> http://www.freeswitch.org
>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>> *Gregor Nanger*
>>>>>>
>>>>>>
>>>>>>
>>>>>> *CTO*
>>>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>>>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>>>>>> • www.infomedia.si
>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.infomedia.si&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=3vN9VFnMCEkuOcYyAEGfDgiFkyDqrRLJVKmzqt0tgkw%3D&reserved=0>
>>>>>>
>>>>>>
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=eLKFdLAeGvA9zv9OFT6WARumrpW7Pr5INhdLXAadqS4%3D&reserved=0>
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>>>>>> http://confluence.freeswitch.org
>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=KlSM7XhL1Ejl1jtoKYKg2ospXA6GhN22zAM9trQeYnE%3D&reserved=0>
>>>>>> http://www.cluecon.com
>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=o0zN0nAx4fKo%2Fc%2F7036WuO4m0VjmG8vWo%2B969P5CaD8%3D&reserved=0>
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=SaLZXzXEHJ3kQdj6opvrEzbqYc1X6%2Bwlc402FuDaC2k%3D&reserved=0>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>> switch-users
>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=LXfaaJRsN3QbszvuAUm3cVwP5qWyiB6dTt4acP%2FCxXg%3D&reserved=0>
>>>>>> http://www.freeswitch.org
>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>>>>>>
>>>>>>
>>>>>>
>>>>>> ------------------------------
>>>>>>
>>>>>> This e-mail and any files transmitted with it may contain privileged
>>>>>> or confidential information. It is solely for use by the individual for
>>>>>> whom it is intended, even if addressed incorrectly. If you received this
>>>>>> e-mail in error, please notify the sender; do not disclose, copy,
>>>>>> distribute, or take any action in reliance on the contents of this
>>>>>> information; and delete it from your system. Any other use of this e-mail
>>>>>> is prohibited.
>>>>>>
>>>>>> Thank you for your compliance.
>>>>>> ------------------------------
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.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/free
>>>>>> switch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>> --
>>>>>
>>>>> 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 <(918)%20424-9378>
>>>>>
>>>>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>>>>
>>>>> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
>>>>> color-twitter-96.png]
>>>>> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.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/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>
>>>> --
>>>> Gregor Nanger
>>>>
>>>> *CTO*
>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>>>> • www.infomedia.si
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.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/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
>
> --
> David W
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From chad at apartmentlines.com  Fri Feb 16 16:02:20 2018
From: chad at apartmentlines.com (Chad Phillips)
Date: Fri, 16 Feb 2018 08:02:20 -0800
Subject: [Freeswitch-users] Media Timeout disconnect with Verto
In-Reply-To: <CAC8JMOL6n7KJFFqb=aUxgsfiJrKx01M-0A9GCp=T_93xHhxdMQ@mail.gmail.com>
References: <CAC8JMOL6n7KJFFqb=aUxgsfiJrKx01M-0A9GCp=T_93xHhxdMQ@mail.gmail.com>
Message-ID: <CAJ0sghQzuWN0AhCVB1hBeDCT3W5ZwoFBum4S4OLSeOv9ALU5pQ@mail.gmail.com>

In verto.conf, I have:

  <settings>
    <param name="detach-timeout-sec" value="10"/>
  </settings>

That hangs up the call on the server side 10 seconds after disconnect, if
no reconnect happens.

On Thu, Feb 15, 2018 at 3:31 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> Hi guys,
>
> what is the verto equivalent of rtp-timeout-sec param?  How do I control
> when to disconnect a verto call when there is no media coming from the
> browser client?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 brian at freeswitch.com  Fri Feb 16 16:12:23 2018
From: brian at freeswitch.com (Brian West)
Date: Fri, 16 Feb 2018 11:12:23 -0500
Subject: [Freeswitch-users] Media Timeout disconnect with Verto
In-Reply-To: <CAJ0sghQzuWN0AhCVB1hBeDCT3W5ZwoFBum4S4OLSeOv9ALU5pQ@mail.gmail.com>
References: <CAC8JMOL6n7KJFFqb=aUxgsfiJrKx01M-0A9GCp=T_93xHhxdMQ@mail.gmail.com>
 <CAJ0sghQzuWN0AhCVB1hBeDCT3W5ZwoFBum4S4OLSeOv9ALU5pQ@mail.gmail.com>
Message-ID: <CAEJMVkBNCuyM0Z90HwcOS6h0VyhjoasW70XDuJxh4v+UERadQA@mail.gmail.com>

I think some are saying this doesn't work so a JIRA probably need to be
filed if that is the case.

/b


On Fri, Feb 16, 2018 at 11:02 AM, Chad Phillips <chad at apartmentlines.com>
wrote:

> In verto.conf, I have:
>
>   <settings>
>     <param name="detach-timeout-sec" value="10"/>
>   </settings>
>
> That hangs up the call on the server side 10 seconds after disconnect, if
> no reconnect happens.
>
> On Thu, Feb 15, 2018 at 3:31 PM, Oleg Stolyar <olegstolyar at gmail.com>
> wrote:
>
>> Hi guys,
>>
>> what is the verto equivalent of rtp-timeout-sec param?  How do I control
>> when to disconnect a verto call when there is no media coming from the
>> browser client?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From olegstolyar at gmail.com  Fri Feb 16 16:37:34 2018
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Fri, 16 Feb 2018 08:37:34 -0800
Subject: [Freeswitch-users] Media Timeout disconnect with Verto
In-Reply-To: <CAEJMVkBNCuyM0Z90HwcOS6h0VyhjoasW70XDuJxh4v+UERadQA@mail.gmail.com>
References: <CAC8JMOL6n7KJFFqb=aUxgsfiJrKx01M-0A9GCp=T_93xHhxdMQ@mail.gmail.com>
 <CAJ0sghQzuWN0AhCVB1hBeDCT3W5ZwoFBum4S4OLSeOv9ALU5pQ@mail.gmail.com>
 <CAEJMVkBNCuyM0Z90HwcOS6h0VyhjoasW70XDuJxh4v+UERadQA@mail.gmail.com>
Message-ID: <CAC8JMO+0GzWyBK3vLaTCCQ99CqvUUJv7WCZcMvzNLBBLEx2GPA@mail.gmail.com>

Actually it does work for me.  I just put it in the wrong place before. I
put it under profile instead of under settings.  Sorry about that.

On Fri, Feb 16, 2018 at 8:12 AM, Brian West <brian at freeswitch.com> wrote:

> I think some are saying this doesn't work so a JIRA probably need to be
> filed if that is the case.
>
> /b
>
>
> On Fri, Feb 16, 2018 at 11:02 AM, Chad Phillips <chad at apartmentlines.com>
> wrote:
>
>> In verto.conf, I have:
>>
>>   <settings>
>>     <param name="detach-timeout-sec" value="10"/>
>>   </settings>
>>
>> That hangs up the call on the server side 10 seconds after disconnect, if
>> no reconnect happens.
>>
>> On Thu, Feb 15, 2018 at 3:31 PM, Oleg Stolyar <olegstolyar at gmail.com>
>> wrote:
>>
>>> Hi guys,
>>>
>>> what is the verto equivalent of rtp-timeout-sec param?  How do I control
>>> when to disconnect a verto call when there is no media coming from the
>>> browser client?
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
>
> --
>
> 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 <(918)%20424-9378>
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
> color-twitter-96.png]
> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 me at nevian.org  Fri Feb 16 16:44:10 2018
From: me at nevian.org (Serge S. Yuriev)
Date: Fri, 16 Feb 2018 19:44:10 +0300
Subject: [Freeswitch-users] loopback_initial_codec
Message-ID: <5f5d6476-5022-e947-1e9e-48ee215f679e@nevian.org>

Hello,

Using FS release 1.6.20

I'm trying to originate call via loopback with preset codec
originate {loopback_initial_codec=PCMA}loopback/12550/internal  &park()

But codec ignored and set to L16.

2018-02-16 18:51:48.294389 [DEBUG] switch_ivr_originate.c:2142 Parsing 
global variables
2018-02-16 18:51:48.294389 [NOTICE] switch_channel.c:1104 New Channel 
loopback/12550/internal-a [9de6c978-4ef8-4f2d-a7d8-f630bebe13af]
2018-02-16 18:51:48.294389 [DEBUG] mod_loopback.c:158 
loopback/12550/internal-a setup codec L16/8000/20
2018-02-16 18:51:48.294389 [NOTICE] switch_channel.c:1102 Rename Channel 
loopback/12550/internal-a->loopback/12550-a 
[9de6c978-4ef8-4f2d-a7d8-f630bebe13af]
2018-02-16 18:51:48.294389 [DEBUG] mod_loopback.c:1174 
(loopback/12550-a) State Change CS_NEW -> CS_INIT
2018-02-16 18:51:48.294389 [DEBUG] mod_loopback.c:601 loopback/12550-a 
CHANNEL KILL
2018-02-16 18:51:48.294389 [DEBUG] switch_core_state_machine.c:584 
(loopback/12550-a) Running State Change CS_INIT (Cur 1 Tot 1230)
2018-02-16 18:51:48.294389 [DEBUG] switch_core_state_machine.c:627 
(loopback/12550-a) State INIT
2018-02-16 18:51:48.294389 [NOTICE] switch_channel.c:1104 New Channel 
loopback/12550-b [e945099e-2ed9-4c49-9d48-58a25756fb08]
2018-02-16 18:51:48.294389 [DEBUG] mod_loopback.c:158 loopback/12550-b 
setup codec L16/8000/20

-- 
Serge S. Yuriev
Lead VoIP engineer



From infos at madovsky.org  Sat Feb 17 13:10:53 2018
From: infos at madovsky.org (Madovsky)
Date: Sat, 17 Feb 2018 05:10:53 -0800
Subject: [Freeswitch-users] about mod_av
Message-ID: <2640b5f9-e5a8-41bc-4965-8b7eec893421@madovsky.org>

Hi folks,

is mod_av compile with ffmpeg or libav?

also what the last version ffmpeg/libav supported by mod_av to compile 
correctly?

thanks

Franck



From alexanderhenryperkins at gmail.com  Sat Feb 17 15:42:51 2018
From: alexanderhenryperkins at gmail.com (Alexander Perkins)
Date: Sat, 17 Feb 2018 09:42:51 -0600
Subject: [Freeswitch-users] Freeswitch Tables
Message-ID: <CALLkTp1dXmrt4aXBhUvEH7R0usVZyXtnnxgHOVq1QohW+w3qGg@mail.gmail.com>

Hi.  Does anybody have documentation regarding what these tables are for?
I've looked online, but cannot find it.

[image: Inline image 1]

Thanks!
Alex
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From gmaruzz at gmail.com  Sat Feb 17 18:04:15 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 17 Feb 2018 19:04:15 +0100
Subject: [Freeswitch-users] Freeswitch Tables
In-Reply-To: <CALLkTp1dXmrt4aXBhUvEH7R0usVZyXtnnxgHOVq1QohW+w3qGg@mail.gmail.com>
References: <CALLkTp1dXmrt4aXBhUvEH7R0usVZyXtnnxgHOVq1QohW+w3qGg@mail.gmail.com>
Message-ID: <CALXCt0pVDWgGboZpzG5AQN6seuToyHaWn-jt9H6yN3RxoXhhdg@mail.gmail.com>

On 17 February 2018 at 16:42, Alexander Perkins <
alexanderhenryperkins at gmail.com> wrote:

> Hi.  Does anybody have documentation regarding what these tables are for?
> I've looked online, but cannot find it.
>
> [image: Inline image 1]
>
>
those are used by FS when you put core in external database.

they substitute tables in /usr/local/freeswitch/db/

Eg, check
https://freeswitch.org/confluence/display/FREESWITCH/PostgreSQL+in+the+core




> Thanks!
> Alex
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From sdevoy at bizfocused.com  Sun Feb 18 16:28:06 2018
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Sun, 18 Feb 2018 16:28:06 +0000
Subject: [Freeswitch-users] NAT port problem
Message-ID: <BN6PR01MB25617C7C3DF58C04AAAE0F1CBBC90@BN6PR01MB2561.prod.exchangelabs.com>

HI all,

I have a NAT problem I can't seem to get past.  The phone is NATed.  From my location with a FIOS router it works fine.  But the client recently upgraded to a new SonicWall router and splat, this device failed.  There are about 10 Cisco phones at this location working fine.  This phone is a Panasonic KX-TGP550.

I have poured over the sip logs and identified the problem.  The sip packets come from the router IP and port 15xxx but are returned to that IP and the phone's source port 5060 instead of 15xxx.

I have tried every NDLB feature I can find, but not every combination. The responses always go back to the wrong port.

Any help would be greatly appreciated.

Thanks,
Sean

Sample sip dump:
    ------------------------------------------------------------------------
recv 492 bytes from udp/[98.204.241.22]:15183 at 23:16:33.764217:
   ------------------------------------------------------------------------
     REGISTER sip:XYZ.bizfocused.com:5060 SIP/2.0
     Via: SIP/2.0/UDP 192.168.2.109:5074;branch=z9hG4bKb6be703b
     Max-Forwards: 70
     To: <sip:34 at XYZ.bizfocused.com>
     From: <sip:34 at XYZ.bizfocused.com>;tag=3263080642
     Call-ID: 97742e94-d1dc14228bad9baadd810080f041d179 at 192.168.2.109<mailto:97742e94-d1dc14228bad9baadd810080f041d179 at 192.168.2.109>
     CSeq: 1 REGISTER
     Contact: <sip:34 at 192.168.2.109:5074>
     Expires: 3600
     Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,NOTIFY,REFER,UPDATE
     User-Agent: Panasonic_KX-TGP550T04/13.33 (0080f041d179)
     Content-Length: 0

    ------------------------------------------------------------------------
send 669 bytes to udp/[98.204.241.22]:5074 at 23:16:33.764394:
   ------------------------------------------------------------------------
     SIP/2.0 401 Unauthorized


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From brian at freeswitch.com  Sun Feb 18 16:50:33 2018
From: brian at freeswitch.com (Brian West)
Date: Sun, 18 Feb 2018 11:50:33 -0500
Subject: [Freeswitch-users] NAT port problem
In-Reply-To: <BN6PR01MB25617C7C3DF58C04AAAE0F1CBBC90@BN6PR01MB2561.prod.exchangelabs.com>
References: <BN6PR01MB25617C7C3DF58C04AAAE0F1CBBC90@BN6PR01MB2561.prod.exchangelabs.com>
Message-ID: <CAEJMVkAJ_DuSOZtXnwfLp5evKHv9Nr26TPYUj=BCExDuX7x8hw@mail.gmail.com>

Enable rport, The issue was from 15183 and we respond to 5074, so the
device never receives the 401.

Nat 101

/b


On Sun, Feb 18, 2018 at 11:28 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> HI all,
>
>
>
> I have a NAT problem I can’t seem to get past.  The phone is NATed.  From
> my location with a FIOS router it works fine.  But the client recently
> upgraded to a new SonicWall router and splat, this device failed.  There
> are about 10 Cisco phones at this location working fine.  This phone is a
> Panasonic KX-TGP550.
>
>
>
> I have poured over the sip logs and identified the problem.  *The sip
> packets come from the router IP and port 15xxx but are returned to that IP
> and the phone’s source port 5060 instead of 15xxx.*
>
>
>
> I have tried every NDLB feature I can find, but not every combination. The
> responses always go back to the wrong port.
>
>
>
> Any help would be greatly appreciated.
>
>
>
> Thanks,
>
> Sean
>
>
>
> Sample sip dump:
>
>     ------------------------------------------------------------
> ------------
>
> recv 492 bytes from *udp/[98.204.241.22]:15183* at 23:16:33.764217:
>
>    ------------------------------------------------------------
> ------------
>
>      REGISTER sip:XYZ.bizfocused.com:5060 SIP/2.0
>
>      Via: SIP/2.0/UDP 192.168.2.109:5074;branch=z9hG4bKb6be703b
>
>      Max-Forwards: 70
>
>      To: <sip:34 at XYZ.bizfocused.com>
>
>      From: <sip:34 at XYZ.bizfocused.com>;tag=3263080642
>
>      Call-ID: 97742e94-d1dc14228bad9baadd810080f041d179 at 192.168.2.109
>
>      CSeq: 1 REGISTER
>
>      Contact: <sip:34 at 192.168.2.109:5074>
>
>      Expires: 3600
>
>      Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,NOTIFY,REFER,UPDATE
>
>      User-Agent: Panasonic_KX-TGP550T04/13.33 (0080f041d179)
>
>      Content-Length: 0
>
>
>
>     --------------------------------------------------------
> ----------------
>
> send 669 bytes to *udp/[98.204.241.22]:5074* at 23:16:33.764394:
>
>    ------------------------------------------------------------
> ------------
>
>      SIP/2.0 401 Unauthorized
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From sdevoy at bizfocused.com  Sun Feb 18 22:43:17 2018
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Sun, 18 Feb 2018 22:43:17 +0000
Subject: [Freeswitch-users] NAT port problem
In-Reply-To: <CAEJMVkAJ_DuSOZtXnwfLp5evKHv9Nr26TPYUj=BCExDuX7x8hw@mail.gmail.com>
References: <BN6PR01MB25617C7C3DF58C04AAAE0F1CBBC90@BN6PR01MB2561.prod.exchangelabs.com>
 <CAEJMVkAJ_DuSOZtXnwfLp5evKHv9Nr26TPYUj=BCExDuX7x8hw@mail.gmail.com>
Message-ID: <BN6PR01MB2561BC0DB10D03F3284FB4E2BBC90@BN6PR01MB2561.prod.exchangelabs.com>

Thanks Brian.  I should have been more clear.  That was exactly my diagnosis and attempted fix using this line:
<param name="NDLB-force-rport" value="true"/>
in the <param> section of the <user> in the Directory file has no effect.

That means one of two things.  I am putting it in the wrong file or I must bite the bullet and get FS up to the current version.

Thanks again.

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Sunday, February 18, 2018 11:51 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] NAT port problem

Enable rport, The issue was from 15183 and we respond to 5074, so the device never receives the 401.

Nat 101

/b


On Sun, Feb 18, 2018 at 11:28 AM, Sean Devoy <sdevoy at bizfocused.com<mailto:sdevoy at bizfocused.com>> wrote:
HI all,

I have a NAT problem I can’t seem to get past.  The phone is NATed.  From my location with a FIOS router it works fine.  But the client recently upgraded to a new SonicWall router and splat, this device failed.  There are about 10 Cisco phones at this location working fine.  This phone is a Panasonic KX-TGP550.

I have poured over the sip logs and identified the problem.  The sip packets come from the router IP and port 15xxx but are returned to that IP and the phone’s source port 5060 instead of 15xxx.

I have tried every NDLB feature I can find, but not every combination. The responses always go back to the wrong port.

Any help would be greatly appreciated.

Thanks,
Sean

Sample sip dump:
    ------------------------------------------------------------------------
recv 492 bytes from udp/[98.204.241.22]:15183 at 23:16:33.764217:
   ------------------------------------------------------------------------
     REGISTER sip:XYZ.bizfocused.com:5060 SIP/2.0
     Via: SIP/2.0/UDP 192.168.2.109:5074;branch=z9hG4bKb6be703b
     Max-Forwards: 70
     To: <sip:34 at XYZ.bizfocused.com>
     From: <sip:34 at XYZ.bizfocused.com>;tag=3263080642
     Call-ID: 97742e94-d1dc14228bad9baadd810080f041d179 at 192.168.2.109<mailto:97742e94-d1dc14228bad9baadd810080f041d179 at 192.168.2.109>
     CSeq: 1 REGISTER
     Contact: <sip:34 at 192.168.2.109:5074>
     Expires: 3600
     Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,NOTIFY,REFER,UPDATE
     User-Agent: Panasonic_KX-TGP550T04/13.33 (0080f041d179)
     Content-Length: 0

    ------------------------------------------------------------------------
send 669 bytes to udp/[98.204.241.22]:5074 at 23:16:33.764394:
   ------------------------------------------------------------------------
     SIP/2.0 401 Unauthorized



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org



--
[https://docs.google.com/uc?export=download&id=1xswZRZyVDo0WQhaemK47pU266yzDRmi0&revid=0B2xnT7i45ngrMTVKM1dpSHZIN28zU0QzbW9xeVF6RXFyRHhBPQ]

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/>

[color-facebook-96.png]<https://www.facebook.com/freeswitch/>[color-twitter-96.png]<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From kaduww at gmail.com  Mon Feb 19 12:10:23 2018
From: kaduww at gmail.com (Carlos Eduardo)
Date: Mon, 19 Feb 2018 09:10:23 -0300
Subject: [Freeswitch-users] NAT port problem
In-Reply-To: <BN6PR01MB2561BC0DB10D03F3284FB4E2BBC90@BN6PR01MB2561.prod.exchangelabs.com>
References: <BN6PR01MB25617C7C3DF58C04AAAE0F1CBBC90@BN6PR01MB2561.prod.exchangelabs.com>
 <CAEJMVkAJ_DuSOZtXnwfLp5evKHv9Nr26TPYUj=BCExDuX7x8hw@mail.gmail.com>
 <BN6PR01MB2561BC0DB10D03F3284FB4E2BBC90@BN6PR01MB2561.prod.exchangelabs.com>
Message-ID: <CAB=EcebwT4VdsycMCdTO=Qxizxhq7J5XP3do71=h3aP15rPQWA@mail.gmail.com>

It can be an ALG problem (
https://www.voip-info.org/wiki/view/Routers+SIP+ALG). Try running another
sip profile with a different sip port, 5070 for example, and register the
phone using this port.

2018-02-18 19:43 GMT-03:00 Sean Devoy <sdevoy at bizfocused.com>:

> Thanks Brian.  I should have been more clear.  That was exactly my
> diagnosis and attempted fix using this line:
>
> <param name="NDLB-force-rport" value="true"/>
>
> in the <param> section of the <user> in the Directory file has no effect.
>
>
>
> That means one of two things.  I am putting it in the wrong file or I must
> bite the bullet and get FS up to the current version.
>
>
>
> Thanks again.
>
>
>
> *From:* FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] *On Behalf Of *Brian West
> *Sent:* Sunday, February 18, 2018 11:51 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] NAT port problem
>
>
>
> Enable rport, The issue was from 15183 and we respond to 5074, so the
> device never receives the 401.
>
>
>
> Nat 101
>
>
>
> /b
>
>
>
>
>
> On Sun, Feb 18, 2018 at 11:28 AM, Sean Devoy <sdevoy at bizfocused.com>
> wrote:
>
> HI all,
>
>
>
> I have a NAT problem I can’t seem to get past.  The phone is NATed.  From
> my location with a FIOS router it works fine.  But the client recently
> upgraded to a new SonicWall router and splat, this device failed.  There
> are about 10 Cisco phones at this location working fine.  This phone is a
> Panasonic KX-TGP550.
>
>
>
> I have poured over the sip logs and identified the problem.  *The sip
> packets come from the router IP and port 15xxx but are returned to that IP
> and the phone’s source port 5060 instead of 15xxx.*
>
>
>
> I have tried every NDLB feature I can find, but not every combination. The
> responses always go back to the wrong port.
>
>
>
> Any help would be greatly appreciated.
>
>
>
> Thanks,
>
> Sean
>
>
>
> Sample sip dump:
>
>     ------------------------------------------------------------
> ------------
>
> recv 492 bytes from *udp/[98.204.241.22]:15183* at 23:16:33.764217:
>
>    ------------------------------------------------------------
> ------------
>
>      REGISTER sip:XYZ.bizfocused.com:5060 SIP/2.0
>
>      Via: SIP/2.0/UDP 192.168.2.109:5074;branch=z9hG4bKb6be703b
>
>      Max-Forwards: 70
>
>      To: <sip:34 at XYZ.bizfocused.com>
>
>      From: <sip:34 at XYZ.bizfocused.com>;tag=3263080642 <(32)%206308-0642>
>
>      Call-ID: 97742e94-d1dc14228bad9baadd810080f041d179 at 192.168.2.109
>
>      CSeq: 1 REGISTER
>
>      Contact: <sip:34 at 192.168.2.109:5074>
>
>      Expires: 3600
>
>      Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,NOTIFY,REFER,UPDATE
>
>      User-Agent: Panasonic_KX-TGP550T04/13.33 (0080f041d179)
>
>      Content-Length: 0
>
>
>
>     --------------------------------------------------------
> ----------------
>
> send 669 bytes to * udp/[98.204.241.22]:5074* at 23:16:33.764394:
>
>    ------------------------------------------------------------
> ------------
>
>      SIP/2.0 401 Unauthorized
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
>
>
>
>
> --
>
> 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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
> color-twitter-96.png]
> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 
*Carlos E. Wagner*
*Tecnólogo em Telecomunicações, OCP, dCAA*

*E-mail:* *kaduww at gmail.com <kaduww at gmail.com>*
*Fone:* +55 48 9981-0894
*Skype:* carlos.e.wagner
www.blogdovoip.com
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From brian at freeswitch.com  Mon Feb 19 13:07:38 2018
From: brian at freeswitch.com (Brian West)
Date: Mon, 19 Feb 2018 07:07:38 -0600
Subject: [Freeswitch-users] NAT port problem
In-Reply-To: <CAB=EcebwT4VdsycMCdTO=Qxizxhq7J5XP3do71=h3aP15rPQWA@mail.gmail.com>
References: <BN6PR01MB25617C7C3DF58C04AAAE0F1CBBC90@BN6PR01MB2561.prod.exchangelabs.com>
 <CAEJMVkAJ_DuSOZtXnwfLp5evKHv9Nr26TPYUj=BCExDuX7x8hw@mail.gmail.com>
 <BN6PR01MB2561BC0DB10D03F3284FB4E2BBC90@BN6PR01MB2561.prod.exchangelabs.com>
 <CAB=EcebwT4VdsycMCdTO=Qxizxhq7J5XP3do71=h3aP15rPQWA@mail.gmail.com>
Message-ID: <CAEJMVkC8sRm-O0HJdEUoaN_SSP8fmK9BVPaw4QEtqsLDib2SSQ@mail.gmail.com>

In this case its probably not an ALG, you should enable rport in your
device and NOT force it with the NDLB param, if your device doesn't support
it, throw that device in the trash or take a hammer to it, and buy a device
from a manufacture that actually made an effort to implement SIP.

/b


On Mon, Feb 19, 2018 at 6:10 AM, Carlos Eduardo <kaduww at gmail.com> wrote:

> It can be an ALG problem (https://www.voip-info.org/
> wiki/view/Routers+SIP+ALG). Try running another sip profile with a
> different sip port, 5070 for example, and register the phone using this
> port.
>
> 2018-02-18 19:43 GMT-03:00 Sean Devoy <sdevoy at bizfocused.com>:
>
>> Thanks Brian.  I should have been more clear.  That was exactly my
>> diagnosis and attempted fix using this line:
>>
>> <param name="NDLB-force-rport" value="true"/>
>>
>> in the <param> section of the <user> in the Directory file has no effect.
>>
>>
>>
>> That means one of two things.  I am putting it in the wrong file or I
>> must bite the bullet and get FS up to the current version.
>>
>>
>>
>> Thanks again.
>>
>>
>>
>> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc
>> es at lists.freeswitch.org] *On Behalf Of *Brian West
>> *Sent:* Sunday, February 18, 2018 11:51 AM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* Re: [Freeswitch-users] NAT port problem
>>
>>
>>
>> Enable rport, The issue was from 15183 and we respond to 5074, so the
>> device never receives the 401.
>>
>>
>>
>> Nat 101
>>
>>
>>
>> /b
>>
>>
>>
>>
>>
>> On Sun, Feb 18, 2018 at 11:28 AM, Sean Devoy <sdevoy at bizfocused.com>
>> wrote:
>>
>> HI all,
>>
>>
>>
>> I have a NAT problem I can’t seem to get past.  The phone is NATed.  From
>> my location with a FIOS router it works fine.  But the client recently
>> upgraded to a new SonicWall router and splat, this device failed.  There
>> are about 10 Cisco phones at this location working fine.  This phone is a
>> Panasonic KX-TGP550.
>>
>>
>>
>> I have poured over the sip logs and identified the problem.  *The sip
>> packets come from the router IP and port 15xxx but are returned to that IP
>> and the phone’s source port 5060 instead of 15xxx.*
>>
>>
>>
>> I have tried every NDLB feature I can find, but not every combination.
>> The responses always go back to the wrong port.
>>
>>
>>
>> Any help would be greatly appreciated.
>>
>>
>>
>> Thanks,
>>
>> Sean
>>
>>
>>
>> Sample sip dump:
>>
>>     ------------------------------------------------------------
>> ------------
>>
>> recv 492 bytes from *udp/[98.204.241.22]:15183* at 23:16:33.764217:
>>
>>    ------------------------------------------------------------
>> ------------
>>
>>      REGISTER sip:XYZ.bizfocused.com:5060 SIP/2.0
>>
>>      Via: SIP/2.0/UDP 192.168.2.109:5074;branch=z9hG4bKb6be703b
>>
>>      Max-Forwards: 70
>>
>>      To: <sip:34 at XYZ.bizfocused.com>
>>
>>      From: <sip:34 at XYZ.bizfocused.com>;tag=3263080642 <(32)%206308-0642>
>>
>>      Call-ID: 97742e94-d1dc14228bad9baadd810080f041d179 at 192.168.2.109
>>
>>      CSeq: 1 REGISTER
>>
>>      Contact: <sip:34 at 192.168.2.109:5074>
>>
>>      Expires: 3600
>>
>>      Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,NOTIFY,REFER,UPDATE
>>
>>      User-Agent: Panasonic_KX-TGP550T04/13.33 (0080f041d179)
>>
>>      Content-Length: 0
>>
>>
>>
>>     --------------------------------------------------------
>> ----------------
>>
>> send 669 bytes to * udp/[98.204.241.22]:5074* at 23:16:33.764394:
>>
>>    ------------------------------------------------------------
>> ------------
>>
>>      SIP/2.0 401 Unauthorized
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>>
>>
>>
>>
>> --
>>
>> 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 <(918)%20424-9378>
>>
>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>
>> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
>> color-twitter-96.png]
>> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
>
> --
> *Carlos E. Wagner*
> *Tecnólogo em Telecomunicações, OCP, dCAA*
>
> *E-mail:* *kaduww at gmail.com <kaduww at gmail.com>*
> *Fone:* +55 48 9981-0894 <+55%2048%209981-0894>
> *Skype:* carlos.e.wagner
> www.blogdovoip.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From tculjaga at gmail.com  Mon Feb 19 20:42:57 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 19 Feb 2018 21:42:57 +0100
Subject: [Freeswitch-users] "fsctl recover" doesn't work on Freeswitch
	V1.6.19
In-Reply-To: <CAJs+-5rAeE5+NTZHZVJrc7yKcgJNJNR+OdW2PT28R+SOnL9LLQ@mail.gmail.com>
References: <tencent_02C00030E26255102C8403FB325D82811A09@qq.com>
 <CAJs+-5rAeE5+NTZHZVJrc7yKcgJNJNR+OdW2PT28R+SOnL9LLQ@mail.gmail.com>
Message-ID: <CAJoKKdV=7RGjcg2=puB6JzzE8CdEOs1EJxLsWGitzRVQu6gstg@mail.gmail.com>

im running 1.6.19 and it does work!

you have to configure database in the core, set switch name to same value
on both left and right FS, set same domain in both left and right FS.

switch.conf.xml
  switchname
  core-recovery-db-dsn
  core-db-dsn

<profile-name>.conf.xml
  track-calls


vars.xml
  domain





On 13 February 2018 at 02:10, Seven Du <dujinfang at gmail.com> wrote:

> hard to tell if you don't post logs to pastebin. but I can confirm master
> works.
>
> On Tue, Feb 6, 2018 at 12:28 PM, Kevin Chi <kevinchi at foxmail.com> wrote:
>
>> hello everyone,
>>
>> The steps of my call recovery test as below.
>> 1.First I added  the line "<param name="track-calls" value="true"/>" to
>> internal.xml.
>> 2.UA called 9664 to play the hold music, and then run "fsctl crash"
>> command.
>> 3.start freeswtich and run "fsctl recover" command.
>> 4.call was interrupted, recover failed.
>>
>> I opened siptrace before "fsctl recover", the sipflow make me can not
>> understand.
>> freeswitch send INVITE to UA
>> freeswitch recive 200 from UA
>> freeswitch send BYE to UA
>> freeswitch recive 200 from UA
>> freeswitch send ACK from UA
>>
>> My freeswitch is only single server, the version is V1.6.19. I have tried
>> the newer version V1.6.20, but the result was same.
>> What's the reason? Pls give me some suggestion, thx a lot.
>>
>> ------------------
>> With regards,
>> Kevin Chi
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 brian at freeswitch.com  Mon Feb 19 21:58:37 2018
From: brian at freeswitch.com (Brian West)
Date: Mon, 19 Feb 2018 15:58:37 -0600
Subject: [Freeswitch-users] "fsctl recover" doesn't work on Freeswitch
	V1.6.19
In-Reply-To: <CAJoKKdV=7RGjcg2=puB6JzzE8CdEOs1EJxLsWGitzRVQu6gstg@mail.gmail.com>
References: <tencent_02C00030E26255102C8403FB325D82811A09@qq.com>
 <CAJs+-5rAeE5+NTZHZVJrc7yKcgJNJNR+OdW2PT28R+SOnL9LLQ@mail.gmail.com>
 <CAJoKKdV=7RGjcg2=puB6JzzE8CdEOs1EJxLsWGitzRVQu6gstg@mail.gmail.com>
Message-ID: <CAEJMVkAiJ2a99Z8dpxJsjpVZGAR5Y-OW6z-dFOFUUofR12OEYw@mail.gmail.com>

Guessing TCP calls.

/b


On Mon, Feb 19, 2018 at 2:42 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> im running 1.6.19 and it does work!
>
> you have to configure database in the core, set switch name to same value
> on both left and right FS, set same domain in both left and right FS.
>
> switch.conf.xml
>   switchname
>   core-recovery-db-dsn
>   core-db-dsn
>
> <profile-name>.conf.xml
>   track-calls
>
>
> vars.xml
>   domain
>
>
>
>
>
> On 13 February 2018 at 02:10, Seven Du <dujinfang at gmail.com> wrote:
>
>> hard to tell if you don't post logs to pastebin. but I can confirm master
>> works.
>>
>> On Tue, Feb 6, 2018 at 12:28 PM, Kevin Chi <kevinchi at foxmail.com> wrote:
>>
>>> hello everyone,
>>>
>>> The steps of my call recovery test as below.
>>> 1.First I added  the line "<param name="track-calls" value="true"/>" to
>>> internal.xml.
>>> 2.UA called 9664 to play the hold music, and then run "fsctl crash"
>>> command.
>>> 3.start freeswtich and run "fsctl recover" command.
>>> 4.call was interrupted, recover failed.
>>>
>>> I opened siptrace before "fsctl recover", the sipflow make me can not
>>> understand.
>>> freeswitch send INVITE to UA
>>> freeswitch recive 200 from UA
>>> freeswitch send BYE to UA
>>> freeswitch recive 200 from UA
>>> freeswitch send ACK from UA
>>>
>>> My freeswitch is only single server, the version is V1.6.19. I have
>>> tried the newer version V1.6.20, but the result was same.
>>> What's the reason? Pls give me some suggestion, thx a lot.
>>>
>>> ------------------
>>> With regards,
>>> Kevin Chi
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>>
>>
>>
>>
>> --
>> About: http://about.me/dujinfang
>> Blog: http://www.dujinfang.com
>> Proj:  http://www.freeswitch.org.cn
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From italo at freeswitch.org  Tue Feb 20 02:58:56 2018
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Tue, 20 Feb 2018 02:58:56 +0000
Subject: [Freeswitch-users] How to unregister verto user
In-Reply-To: <CAEJMVkAAEe3-dEjTxrwPhRj=7urEorgpFQw+UZGhuNBZmM2ZNQ@mail.gmail.com>
References: <CAK9Ws2foDzCr_m_B3FsJSwuq3ZChpSwALUAbF7b6BnUEB_oKYg@mail.gmail.com>
 <CAC7ttFHwsG80vVKMr8Jc8k-9T6SrEE2_XNHRSB4me80oVo8caw@mail.gmail.com>
 <CAJoKKdXcO-EEGKUaPczKo6LL_LE6qJ_v=-RYsKkLX+F-cJa2Hw@mail.gmail.com>
 <CY1PR07MB2377EDDCFE5ADE5609376504BEF50@CY1PR07MB2377.namprd07.prod.outlook.com>
 <CAEJMVkAkJmFhdgzAa8xRneUaMS2ypPgFJMDMt4nGdKkYz5jF4g@mail.gmail.com>
 <CAC7ttFHbhxMvYnSASdorzokT7Ob2Mwduh77w_Rs++2ng49gqtw@mail.gmail.com>
 <CAJoKKdWa4-+HnwuRrhc5CW_WuiZwnah91Sf3MuCVWWaR8P8z1A@mail.gmail.com>
 <CAKeZRhvy0EA4Qh=Z89JWFP17jBPGm4=83=ON1oBLrYWJoO1tEw@mail.gmail.com>
 <CAK9Ws2fntCyfkrTDLbtACWqb-n7pBw9T2=3ds9+9Y5THf3B14A@mail.gmail.com>
 <CAEJMVkAAEe3-dEjTxrwPhRj=7urEorgpFQw+UZGhuNBZmM2ZNQ@mail.gmail.com>
Message-ID: <CABMX_w-pj81PA3W9b2w9mARAuV6iT9gTEKEBD9Rxzu-wUaNXAA@mail.gmail.com>

There's a default timeout of 120 seconds, that's the time to expire this
detached session. Are you sure it still show the sessions after that time?

Run Fs_cli -x verto status after that time and see
Em sex, 16 de fev de 2018 às 09:51, Brian West <brian at freeswitch.com>
escreveu:

> Please file a JIRA.
>
> /b
>
>
> On Fri, Feb 16, 2018 at 4:40 AM, David Wafula <davidwaf at gmail.com> wrote:
>
>> I have tested ...if the user closes browser without explicitly logging
>> out, the session is never terminated and it seems the user stays in the
>> conference forever.
>>
>> On Fri, Feb 16, 2018 at 11:35 AM, Alexandr Popov <
>> alexandr.popov at iqoption.com> wrote:
>>
>>> Brian, sometimes it could stuck.
>>>
>>> 2018-02-16 9:59 GMT+02:00 Tihomir Culjaga <tculjaga at gmail.com>:
>>>
>>>> just call logout on your logout button click... otherwise wss will
>>>> timeout on browser close / crash eventually
>>>>
>>>> On 15 February 2018 at 19:51, Gregor Nanger <gregor at infomedia.si>
>>>> wrote:
>>>>
>>>>> Yes, it is still the same. You have on unload event, but this captures
>>>>> everything, refresh, go to other link and even close. It's better than
>>>>> nothing.
>>>>>
>>>>>
>>>>> On Thu, Feb 15, 2018, 19:43 Brian West <brian at freeswitch.com> wrote:
>>>>>
>>>>>> When the socket disconnects it should unregister them, what exactly
>>>>>> is the issue your having?
>>>>>>
>>>>>> /b
>>>>>>
>>>>>> On Thu, Feb 15, 2018 at 11:39 AM Mundkowsky, Robert <
>>>>>> rmundkowsky at ets.org> wrote:
>>>>>>
>>>>>>> In the old days, web browsers did not have an event for window
>>>>>>> getting closed.  I suspect that is still true, so you do not have an event
>>>>>>> to trigger this javascript code you mentioned.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Robert
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> *From:* FreeSWITCH-users [mailto:
>>>>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tihomir
>>>>>>> Culjaga
>>>>>>> *Sent:* Wednesday, February 14, 2018 4:56 PM
>>>>>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>>>> *Subject:* Re: [Freeswitch-users] How to unregister verto user
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> simply
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> var verto;
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> ...
>>>>>>>
>>>>>>> ...
>>>>>>>
>>>>>>> verto = new $.verto(confg)
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> ...
>>>>>>>
>>>>>>> ...
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> verto.logout()
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On 14 February 2018 at 21:48, Gregor Nanger <gregor at infomedia.si>
>>>>>>> wrote:
>>>>>>>
>>>>>>> As far as I know, you can't. Session will expire eventualy.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> 2018-02-14 20:12 GMT+01:00 David Wafula <davidwaf at gmail.com>:
>>>>>>>
>>>>>>> Hi all,
>>>>>>>
>>>>>>> I have a verto user who keeps closing browser window without logging
>>>>>>> out and i need to force unregistration periodically.
>>>>>>>
>>>>>>> What command can be used to accomplish this?
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> The only commands am able to use are:
>>>>>>>
>>>>>>> verto status/xmlstatus
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Kind regards
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639751357&sdata=CcbH%2BpF8NcktyJIZ%2Bk5TMOVLaKKKsQ%2FUDLfvAUQDw9Q%3D&reserved=0>
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639751357&sdata=9Wm6rFEQgBXnz9HNo%2Fxc3IMFSaG%2FpK77r2H4Vj%2FsZLU%3D&reserved=0>
>>>>>>> http://confluence.freeswitch.org
>>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=KlSM7XhL1Ejl1jtoKYKg2ospXA6GhN22zAM9trQeYnE%3D&reserved=0>
>>>>>>> http://www.cluecon.com
>>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=o0zN0nAx4fKo%2Fc%2F7036WuO4m0VjmG8vWo%2B969P5CaD8%3D&reserved=0>
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=SaLZXzXEHJ3kQdj6opvrEzbqYc1X6%2Bwlc402FuDaC2k%3D&reserved=0>
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=LXfaaJRsN3QbszvuAUm3cVwP5qWyiB6dTt4acP%2FCxXg%3D&reserved=0>
>>>>>>> http://www.freeswitch.org
>>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>>
>>>>>>> *Gregor Nanger*
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> *CTO*
>>>>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>>>>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>>>>>>> • www.infomedia.si
>>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.infomedia.si&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=3vN9VFnMCEkuOcYyAEGfDgiFkyDqrRLJVKmzqt0tgkw%3D&reserved=0>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=eLKFdLAeGvA9zv9OFT6WARumrpW7Pr5INhdLXAadqS4%3D&reserved=0>
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>>>>>>> http://confluence.freeswitch.org
>>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=KlSM7XhL1Ejl1jtoKYKg2ospXA6GhN22zAM9trQeYnE%3D&reserved=0>
>>>>>>> http://www.cluecon.com
>>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=o0zN0nAx4fKo%2Fc%2F7036WuO4m0VjmG8vWo%2B969P5CaD8%3D&reserved=0>
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=SaLZXzXEHJ3kQdj6opvrEzbqYc1X6%2Bwlc402FuDaC2k%3D&reserved=0>
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=LXfaaJRsN3QbszvuAUm3cVwP5qWyiB6dTt4acP%2FCxXg%3D&reserved=0>
>>>>>>> http://www.freeswitch.org
>>>>>>> <https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C642bb9e3a4e54b09996d08d573f5f85f%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636542422639907611&sdata=ipq5LdFuk%2BZWP6%2F25T5fPCiwqgAAFMKvU9FR1bWr4h4%3D&reserved=0>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> ------------------------------
>>>>>>>
>>>>>>> This e-mail and any files transmitted with it may contain privileged
>>>>>>> or confidential information. It is solely for use by the individual for
>>>>>>> whom it is intended, even if addressed incorrectly. If you received this
>>>>>>> e-mail in error, please notify the sender; do not disclose, copy,
>>>>>>> distribute, or take any action in reliance on the contents of this
>>>>>>> information; and delete it from your system. Any other use of this e-mail
>>>>>>> is prohibited.
>>>>>>>
>>>>>>> Thank you for your compliance.
>>>>>>> ------------------------------
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.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
>>>>>>
>>>>>> --
>>>>>>
>>>>>> 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 <(918)%20424-9378>
>>>>>>
>>>>>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>>>>>
>>>>>> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
>>>>>> color-twitter-96.png]
>>>>>> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.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
>>>>>
>>>>> --
>>>>> Gregor Nanger
>>>>>
>>>>> *CTO*
>>>>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>>>>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>>>>> • www.infomedia.si
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.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
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.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
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>>
>>
>>
>>
>> --
>> David W
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
>
> --
>
> 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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
> color-twitter-96.png]
> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 cong.wang.itsherpa at gmail.com  Tue Feb 20 09:08:11 2018
From: cong.wang.itsherpa at gmail.com (=?utf-8?B?546L6IGh?=)
Date: Tue, 20 Feb 2018 18:08:11 +0900
Subject: [Freeswitch-users] The time on FreeSWITCH is faster than server time
Message-ID: <AA810E17-392C-43C8-9E46-A916EADBC7F1@gmail.com>

Hey all,

Recently I noticed a clock problem on my FS server, the time in fs_cli had about 10sec faster than server time(correct time), so I executed sync_clock to correct it.
This server had run about 1 month, and I wonder if I need to make sync_clock per month, and how to fix this clock problem.

Regards.

From shaun.stokes at itec-support.co.uk  Tue Feb 20 09:41:55 2018
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Tue, 20 Feb 2018 09:41:55 +0000
Subject: [Freeswitch-users] The time on FreeSWITCH is faster than server
 time
In-Reply-To: <AA810E17-392C-43C8-9E46-A916EADBC7F1@gmail.com>
References: <AA810E17-392C-43C8-9E46-A916EADBC7F1@gmail.com>
Message-ID: <1519119875310.774@itec-support.co.uk>

If you're using FreeSWITCH in a virtual environment we've had similar issues when using FreeSWITCH on a Hyper-V VM, the time on the VM seems to run at a slightly difference pace. When the VM syncs with the host FreeSWITCH doesn't pick this up.

We found using sync_clock fixed this which we run daily as a scheduled task which executes the command 'fsctl sync_clock_when_idle' via Event Socket API, also disabled the time sync between the VM and switched to NTP.

It would be useful if FreeSWITCH would execute this command every 24 hours out of the box, I can't think of any circumstances where this behaviour wouldn't be desired.

Shaun
________________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of 王聡 <cong.wang.itsherpa at gmail.com>
Sent: 20 February 2018 09:08
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] The time on FreeSWITCH is faster than server time

Hey all,

Recently I noticed a clock problem on my FS server, the time in fs_cli had about 10sec faster than server time(correct time), so I executed sync_clock to correct it.
This server had run about 1 month, and I wonder if I need to make sync_clock per month, and how to fix this clock problem.

Regards.
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.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

______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
______________________________________________________________________

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______________________________________________________________________


From brian at freeswitch.com  Tue Feb 20 12:11:58 2018
From: brian at freeswitch.com (Brian West)
Date: Tue, 20 Feb 2018 06:11:58 -0600
Subject: [Freeswitch-users] The time on FreeSWITCH is faster than server
	time
In-Reply-To: <1519119875310.774@itec-support.co.uk>
References: <AA810E17-392C-43C8-9E46-A916EADBC7F1@gmail.com>
 <1519119875310.774@itec-support.co.uk>
Message-ID: <CAEJMVkDS_7XyZR=foAnY74gofjrF50hE8YWq7p0UnToYfOe27A@mail.gmail.com>

10 seconds in a month is rather good to hear.

On Tue, Feb 20, 2018 at 3:41 AM, Shaun Stokes <
shaun.stokes at itec-support.co.uk> wrote:

> If you're using FreeSWITCH in a virtual environment we've had similar
> issues when using FreeSWITCH on a Hyper-V VM, the time on the VM seems to
> run at a slightly difference pace. When the VM syncs with the host
> FreeSWITCH doesn't pick this up.
>
> We found using sync_clock fixed this which we run daily as a scheduled
> task which executes the command 'fsctl sync_clock_when_idle' via Event
> Socket API, also disabled the time sync between the VM and switched to NTP.
>
> It would be useful if FreeSWITCH would execute this command every 24 hours
> out of the box, I can't think of any circumstances where this behaviour
> wouldn't be desired.
>
> Shaun
> ________________________________________
> From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on
> behalf of 王聡 <cong.wang.itsherpa at gmail.com>
> Sent: 20 February 2018 09:08
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] The time on FreeSWITCH is faster than server
> time
>
> Hey all,
>
> Recently I noticed a clock problem on my FS server, the time in fs_cli had
> about 10sec faster than server time(correct time), so I executed sync_clock
> to correct it.
> This server had run about 1 month, and I wonder if I need to make
> sync_clock per month, and how to fix this clock problem.
>
> Regards.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus
> Scanning Service.
> ______________________________________________________________________
>
> ______________________________________________________________________
> This message has been checked for all known viruses by  MessageLabs Virus
> Scanning Service.
> ______________________________________________________________________
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From bmcandrews at remotelink.com  Tue Feb 20 15:03:52 2018
From: bmcandrews at remotelink.com (Bill McAndrews)
Date: Tue, 20 Feb 2018 09:03:52 -0600
Subject: [Freeswitch-users] Stop Muted tone_stream When Unmuted
Message-ID: <CAJC2a=oU1SrkSPVrbKM0412q9KYU-JapGJv96MMFjCEJxSD-CA@mail.gmail.com>

Hello,

I am having trouble with users who mute themselves in conference but don't
realize they have. These incidents can be from a minute or two to several
hours. We want to play an intermittent beep tone in the background to
remind them they are muted.

I set muted-sound to to a continuous tone_stream loop which serves the
purpose. The problem is that when unmuted, the tone continues. I have
unmuted-sound set and I've tried uuid_break as well as other sound related
commands but the tone continues.

Is there a way to stop a tone_stream that I'm missing? If not, is there a
way to code the tone stream in muted-sound that will stop when the channel
is unmuted?

FreeSWITCH Version 1.6.20-37-987c9b9~64bit (-37-987c9b9 64bit) on Debian 8

Thank-you in advance.
Bill

-- 
Bill McAndrews

*Infrastructure Administrator*
*Direct: 331-684-8823*
RemoteLink.com <http://www.remotelink.com/>
RemoteLink.com.ph <http://www.remotelink.com.ph/>

<http://www.remotelink.com/>

<http://www.facebook.com/remotelink> <http://www.twitter.com/remotelinkinc>
<http://www.linkedin.com/company/147838?trk=tyah>
<http://www.rltalkscloud.wordpress.com/feed>

*Specialties:*

*Cloud Computing, SugarCRM, Google Apps for Business, SugarCRM 6.0
Quickbooks Integration, Telecommunications, SIP Networks, Java, Php
Development, Search Marketing SEO/SEM, Social Media Coaching*
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From mickael at winlux.fr  Fri Feb 16 16:33:34 2018
From: mickael at winlux.fr (Mickael Hubert)
Date: Fri, 16 Feb 2018 17:33:34 +0100
Subject: [Freeswitch-users] Send RTP to external server
Message-ID: <CA+fRVSWZxqe3VxmaS8TD0TmbUbmbh-=6NoKcyWVPm_+bGJ8b1A@mail.gmail.com>

Hi list,
I want to record each call through freeswitch. But i want record only
caller (SSRC 1) OR callee (SSRC 2) voice (not both).

I read about SIPREC, Jack, etc ... not interesting

Do you have a idea for me please ?

Thanks in advance
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From mwhite41 at gmail.com  Mon Feb 19 05:16:04 2018
From: mwhite41 at gmail.com (Mark White)
Date: Mon, 19 Feb 2018 15:16:04 +1000
Subject: [Freeswitch-users] Javascript - session_in_hangup_hook: How?
Message-ID: <CAE=3BVzHgncNzxvL6NXXY=gVE=4DwEfO9r9gwdUHrGEk8=rQPA@mail.gmail.com>

Hi - I'm starting to use Freeswitch to receive inbound faxes - which works
brilliantly - but I am trying to get my head around how to launch some .js
code after hangup, i.e. to deal with the fax file itself.

Specifically, I'm expecting that I'm able to access session data in the
script by setting session_in_hangup_hook=true, as below.

Now, invoking .js files seems to work fine - either in the dialpan itself,
or via the hangup hook. But what I can't seem to understand is how the
session data is accessible in the invoked javascript. I'm getting the
following error on the FS console from hangup.js:

Exception: ReferenceError: request is not defined (near: "var env =
request.dumpENV("text");")

XML Dialplan and hangup.js code fragments below. Any clues as to what I'm
doing wrong?

(Environment is a current 1.6, Debian 8, installed via FS packages - not
from source. mod_v8 is loaded - the scripts are executing, apparently - but
I'm stuck trying to find the example code to access the session variables).

Any help appreciated,

Mark

-----

Dialplan XML:

<extension name="fax_receive">

  <condition field="destination_number" expression="^5551212$">

    <action application="set" data="session_in_hangup_hook=true"/>

    <action application="set" data="api_hangup_hook=jsrun
/home/mark/hangup.js"/>

    <action application="answer" />

    <action application="javascript" data="/home/mark/postfax.js"/>

    <action application="playback" data="silence_stream://2000"/>

    <action application="set" data="fax_enable_t38_request=true"/>

    <action application="set" data="fax_enable_t38=true"/>

    <action application="rxfax" data="/tmp/FAX-${uuid}.tif"/>

    <action application="hangup"/>

  </condition>

</extension>



// hangup.js code fragment

console_log("notice","Inside the hangup.js script");  // This executes, I
get the console message post-hangup


// Get and log all available session variables.

var env;

env = request.dumpENV("text");


console_log(env);
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From pierre at couderc.eu  Tue Feb 20 07:07:34 2018
From: pierre at couderc.eu (Pierre Couderc)
Date: Tue, 20 Feb 2018 08:07:34 +0100
Subject: [Freeswitch-users] freze when installing Freeswitch
Message-ID: <da62c7c9-d75f-0fd2-1c86-a69ceb96d02e@couderc.eu>

I am trying to install my first Freeswitch nearly in debian 8 :

I have followed 
https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie :

wget -O - 
https://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | 
apt-key add -
echo "deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie 
main" > /etc/apt/sources.list.d/freeswitch.list
apt-get update && apt-get install -y freeswitch-meta-all

It starts well but does not give back the prompt :

Setting up tzdata-java (2017c-0+deb8u1) ...
Setting up sgml-base (1.26+nmu4) ...
Setting up vdpau-va-driver:amd64 (0.7.4-3) ...
Setting up libtxc-dxtn-s2tc0:amd64 (0~git20131104-1.1) ...
update-alternatives: using 
/usr/lib/x86_64-linux-gnu/libtxc_dxtn_s2tc.so.0 to provide 
/usr/lib/x86_64-linux-gnu/libtxc_dxtn.so (libtxc-dxtn-x86_64-linux-gnu) 
in auto mode
Setting up file (1:5.22+15-2+deb8u3) ...
Setting up at-spi2-core (2.14.0-1) ...
Setting up policykit-1 (0.105-15~deb8u2) ...
Removed symlink /run/systemd/system/polkitd.service.

---> never returns the prompt here.


if I try on another console :

root at test:~# fs_cli -rRS
[ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error]
[INFO] fs_cli.c:1665 main() Retrying
[ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error]
[INFO] fs_cli.c:1665 main() Retrying
...

Please note that my debian 8 is a LXD container inside a debian 9 host.

My question is : what should I do ? How to retry ?


Pierre Couderc




From me at nevian.org  Tue Feb 20 16:17:03 2018
From: me at nevian.org (Serge S. Yuriev)
Date: Tue, 20 Feb 2018 19:17:03 +0300
Subject: [Freeswitch-users] loopback_initial_codec
In-Reply-To: <5f5d6476-5022-e947-1e9e-48ee215f679e@nevian.org>
References: <5f5d6476-5022-e947-1e9e-48ee215f679e@nevian.org>
Message-ID: <7329ed62-1f44-85c8-2e4b-427291f415d2@nevian.org>

Hi,

Anyone? Maybe I do it wrong?

On 16/02/18 19:44, Serge S. Yuriev wrote:
> Hello,
> 
> Using FS release 1.6.20
> 
> I'm trying to originate call via loopback with preset codec
> originate {loopback_initial_codec=PCMA}loopback/12550/internal  &park()
> 
> But codec ignored and set to L16.
> 
> 2018-02-16 18:51:48.294389 [DEBUG] switch_ivr_originate.c:2142 Parsing 
> global variables
> 2018-02-16 18:51:48.294389 [NOTICE] switch_channel.c:1104 New Channel 
> loopback/12550/internal-a [9de6c978-4ef8-4f2d-a7d8-f630bebe13af]
> 2018-02-16 18:51:48.294389 [DEBUG] mod_loopback.c:158 
> loopback/12550/internal-a setup codec L16/8000/20
> 2018-02-16 18:51:48.294389 [NOTICE] switch_channel.c:1102 Rename Channel 
> loopback/12550/internal-a->loopback/12550-a 
> [9de6c978-4ef8-4f2d-a7d8-f630bebe13af]
> 2018-02-16 18:51:48.294389 [DEBUG] mod_loopback.c:1174 
> (loopback/12550-a) State Change CS_NEW -> CS_INIT
> 2018-02-16 18:51:48.294389 [DEBUG] mod_loopback.c:601 loopback/12550-a 
> CHANNEL KILL
> 2018-02-16 18:51:48.294389 [DEBUG] switch_core_state_machine.c:584 
> (loopback/12550-a) Running State Change CS_INIT (Cur 1 Tot 1230)
> 2018-02-16 18:51:48.294389 [DEBUG] switch_core_state_machine.c:627 
> (loopback/12550-a) State INIT
> 2018-02-16 18:51:48.294389 [NOTICE] switch_channel.c:1104 New Channel 
> loopback/12550-b [e945099e-2ed9-4c49-9d48-58a25756fb08]
> 2018-02-16 18:51:48.294389 [DEBUG] mod_loopback.c:158 loopback/12550-b 
> setup codec L16/8000/20
> 

-- 
Serge S. Yuriev
Lead VoIP engineer



From mike at jerris.com  Tue Feb 20 19:02:32 2018
From: mike at jerris.com (Michael Jerris)
Date: Tue, 20 Feb 2018 14:02:32 -0500
Subject: [Freeswitch-users] [rtp-timeout-sec][need experts comments]
In-Reply-To: <CADSVerCfDk74ygoYxmnjJqYJzTtyvW1y7dLvWe8hE36At44DEQ@mail.gmail.com>
References: <CADSVerCfDk74ygoYxmnjJqYJzTtyvW1y7dLvWe8hE36At44DEQ@mail.gmail.com>
Message-ID: <99066E27-A723-4585-9948-D5AC2456E6ED@jerris.com>

rtp timeouts generally are a bad reason to tear down a call unless its actually stuck.  The problem with shorter times is sometimes with dtx and hold you can see it hangup on you.  With proper session timers there is typically no reason for rtp timers

> On Feb 8, 2018, at 12:43 PM, Bilal Abbasi <bilaln018 at gmail.com> wrote:
> 
> Hi Users,
> 
> I did checked the default value of rtp timeout is 300 seconds, for me its very long, i just wanted to know why it is placed that long, for me if a call is on silent for 30-40 seconds i will hang up that. any down side for doing this?
> 
> Regards
> Abbasi



From mike at jerris.com  Tue Feb 20 20:03:33 2018
From: mike at jerris.com (Michael Jerris)
Date: Tue, 20 Feb 2018 15:03:33 -0500
Subject: [Freeswitch-users] Syslog forwarding instead of file
In-Reply-To: <CA+gLzy-1QArDmQNz2xAGmkBhGULRXzuuT0auNZdeA3L9Zkf3KQ@mail.gmail.com>
References: <CA+gLzy-1QArDmQNz2xAGmkBhGULRXzuuT0auNZdeA3L9Zkf3KQ@mail.gmail.com>
Message-ID: <D53D9F2A-28FD-42A0-A500-971A899B9C53@jerris.com>

./src/mod/loggers/mod_graylog2
> On Feb 7, 2018, at 4:49 PM, Kevin Olbrich <ko at sv01.de> wrote:
> 
> Hi!
> 
> We are currently testing Graylog2 to merge logs from several UC nodes.
> Is Freeswitch able to send syslog messages to a remote server?
> 
> Kind regards,
> Kevin

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From sdevoy at bizfocused.com  Tue Feb 20 20:19:48 2018
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 20 Feb 2018 20:19:48 +0000
Subject: [Freeswitch-users] NAT port problem
In-Reply-To: <CAEJMVkC8sRm-O0HJdEUoaN_SSP8fmK9BVPaw4QEtqsLDib2SSQ@mail.gmail.com>
References: <BN6PR01MB25617C7C3DF58C04AAAE0F1CBBC90@BN6PR01MB2561.prod.exchangelabs.com>
 <CAEJMVkAJ_DuSOZtXnwfLp5evKHv9Nr26TPYUj=BCExDuX7x8hw@mail.gmail.com>
 <BN6PR01MB2561BC0DB10D03F3284FB4E2BBC90@BN6PR01MB2561.prod.exchangelabs.com>
 <CAB=EcebwT4VdsycMCdTO=Qxizxhq7J5XP3do71=h3aP15rPQWA@mail.gmail.com>
 <CAEJMVkC8sRm-O0HJdEUoaN_SSP8fmK9BVPaw4QEtqsLDib2SSQ@mail.gmail.com>
Message-ID: <BN6PR01MB25614520AD6BDF1A0C92762BBBCF0@BN6PR01MB2561.prod.exchangelabs.com>

LOL!  Exactly.  This device is, shall we say, VERY BASIC.  It worked with the old router. When they upgraded to a SONICWALL I cannot get it to work. I have learned my lesson. In the future I will only allow a quality ATA and DECT phone for this type of solution. For the record this is a Panasonic KX-TGP550.

I tried alternate local port numbers.  I tried all the ALG settings on the SonicWALL.  I am upgrading from 1.2 <hangs head in shame> to 1.6 in hopes that NDLB rport will fix it.

Thanks to all who replied.

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Monday, February 19, 2018 8:08 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] NAT port problem

In this case its probably not an ALG, you should enable rport in your device and NOT force it with the NDLB param, if your device doesn't support it, throw that device in the trash or take a hammer to it, and buy a device from a manufacture that actually made an effort to implement SIP.

/b


On Mon, Feb 19, 2018 at 6:10 AM, Carlos Eduardo <kaduww at gmail.com<mailto:kaduww at gmail.com>> wrote:
It can be an ALG problem (https://www.voip-info.org/wiki/view/Routers+SIP+ALG). Try running another sip profile with a different sip port, 5070 for example, and register the phone using this port.

2018-02-18 19:43 GMT-03:00 Sean Devoy <sdevoy at bizfocused.com<mailto:sdevoy at bizfocused.com>>:
Thanks Brian.  I should have been more clear.  That was exactly my diagnosis and attempted fix using this line:
<param name="NDLB-force-rport" value="true"/>
in the <param> section of the <user> in the Directory file has no effect.

That means one of two things.  I am putting it in the wrong file or I must bite the bullet and get FS up to the current version.

Thanks again.

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Brian West
Sent: Sunday, February 18, 2018 11:51 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] NAT port problem

Enable rport, The issue was from 15183 and we respond to 5074, so the device never receives the 401.

Nat 101

/b


On Sun, Feb 18, 2018 at 11:28 AM, Sean Devoy <sdevoy at bizfocused.com<mailto:sdevoy at bizfocused.com>> wrote:
HI all,

I have a NAT problem I can’t seem to get past.  The phone is NATed.  From my location with a FIOS router it works fine.  But the client recently upgraded to a new SonicWall router and splat, this device failed.  There are about 10 Cisco phones at this location working fine.  This phone is a Panasonic KX-TGP550.

I have poured over the sip logs and identified the problem.  The sip packets come from the router IP and port 15xxx but are returned to that IP and the phone’s source port 5060 instead of 15xxx.

I have tried every NDLB feature I can find, but not every combination. The responses always go back to the wrong port.

Any help would be greatly appreciated.

Thanks,
Sean

Sample sip dump:
    ------------------------------------------------------------------------
recv 492 bytes from udp/[98.204.241.22]:15183 at 23:16:33.764217:
   ------------------------------------------------------------------------
     REGISTER sip:XYZ.bizfocused.com:5060 SIP/2.0
     Via: SIP/2.0/UDP 192.168.2.109:5074;branch=z9hG4bKb6be703b
     Max-Forwards: 70
     To: <sip:34 at XYZ.bizfocused.com>
     From: <sip:34 at XYZ.bizfocused.com>;tag=3263080642<tel:(32)%206308-0642>
     Call-ID: 97742e94-d1dc14228bad9baadd810080f041d179 at 192.168.2.109<mailto:97742e94-d1dc14228bad9baadd810080f041d179 at 192.168.2.109>
     CSeq: 1 REGISTER
     Contact: <sip:34 at 192.168.2.109:5074>
     Expires: 3600
     Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,NOTIFY,REFER,UPDATE
     User-Agent: Panasonic_KX-TGP550T04/13.33 (0080f041d179)
     Content-Length: 0

    ------------------------------------------------------------------------
send 669 bytes to udp/[98.204.241.22]:5074 at 23:16:33.764394:
   ------------------------------------------------------------------------
     SIP/2.0 401 Unauthorized



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org



--
[https://docs.google.com/uc?export=download&id=1xswZRZyVDo0WQhaemK47pU266yzDRmi0&revid=0B2xnT7i45ngrMTVKM1dpSHZIN28zU0QzbW9xeVF6RXFyRHhBPQ]

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<tel:(918)%20424-9378>

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

[color-facebook-96.png]<https://www.facebook.com/freeswitch/>[color-twitter-96.png]<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org



--
Carlos E. Wagner
Tecnólogo em Telecomunicações, OCP, dCAA
E-mail: kaduww at gmail.com<mailto:kaduww at gmail.com>
Fone: +55 48 9981-0894<tel:+55%2048%209981-0894>
Skype: carlos.e.wagner
www.blogdovoip.com<http://www.blogdovoip.com>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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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http://www.freeswitch.org



--
[https://docs.google.com/uc?export=download&id=1xswZRZyVDo0WQhaemK47pU266yzDRmi0&revid=0B2xnT7i45ngrMTVKM1dpSHZIN28zU0QzbW9xeVF6RXFyRHhBPQ]

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/>

[color-facebook-96.png]<https://www.facebook.com/freeswitch/>[color-twitter-96.png]<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From brian at freeswitch.com  Tue Feb 20 23:50:28 2018
From: brian at freeswitch.com (Brian West)
Date: Tue, 20 Feb 2018 17:50:28 -0600
Subject: [Freeswitch-users] Send RTP to external server
In-Reply-To: <CA+fRVSWZxqe3VxmaS8TD0TmbUbmbh-=6NoKcyWVPm_+bGJ8b1A@mail.gmail.com>
References: <CA+fRVSWZxqe3VxmaS8TD0TmbUbmbh-=6NoKcyWVPm_+bGJ8b1A@mail.gmail.com>
Message-ID: <CAEJMVkD2AMvYmh2w9sMJWFzbXeJ5nNr2fM9sOttiZmT8BcGSQQ@mail.gmail.com>

you can already do this without SIPREC in freeswitch. By setting the
RECORD_READ_ONLY or RECORD_WRITE_ONLY variables.

/b


On Fri, Feb 16, 2018 at 10:33 AM, Mickael Hubert <mickael at winlux.fr> wrote:

> Hi list,
> I want to record each call through freeswitch. But i want record only
> caller (SSRC 1) OR callee (SSRC 2) voice (not both).
>
> I read about SIPREC, Jack, etc ... not interesting
>
> Do you have a idea for me please ?
>
> Thanks in advance
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From brian at freeswitch.com  Tue Feb 20 23:52:54 2018
From: brian at freeswitch.com (Brian West)
Date: Tue, 20 Feb 2018 17:52:54 -0600
Subject: [Freeswitch-users] NAT port problem
In-Reply-To: <BN6PR01MB25614520AD6BDF1A0C92762BBBCF0@BN6PR01MB2561.prod.exchangelabs.com>
References: <BN6PR01MB25617C7C3DF58C04AAAE0F1CBBC90@BN6PR01MB2561.prod.exchangelabs.com>
 <CAEJMVkAJ_DuSOZtXnwfLp5evKHv9Nr26TPYUj=BCExDuX7x8hw@mail.gmail.com>
 <BN6PR01MB2561BC0DB10D03F3284FB4E2BBC90@BN6PR01MB2561.prod.exchangelabs.com>
 <CAB=EcebwT4VdsycMCdTO=Qxizxhq7J5XP3do71=h3aP15rPQWA@mail.gmail.com>
 <CAEJMVkC8sRm-O0HJdEUoaN_SSP8fmK9BVPaw4QEtqsLDib2SSQ@mail.gmail.com>
 <BN6PR01MB25614520AD6BDF1A0C92762BBBCF0@BN6PR01MB2561.prod.exchangelabs.com>
Message-ID: <CAEJMVkAy+AAoWXp2+=JfStjxTomsbWxh3ZKX-LW7SpNx+bYv4w@mail.gmail.com>

That device is far from basic, the admin manual is like a dictionary, you
may not see the settings in the UI but I know that device has tons of
options.

/b


On Tue, Feb 20, 2018 at 2:19 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> LOL!  Exactly.  This device is, shall we say, VERY BASIC.  It worked with
> the old router. When they upgraded to a SONICWALL I cannot get it to work.
> I have learned my lesson. In the future I will only allow a quality ATA and
> DECT phone for this type of solution. For the record this is a Panasonic
> KX-TGP550.
>
>
>
> I tried alternate local port numbers.  I tried all the ALG settings on the
> SonicWALL.  I am upgrading from 1.2 <hangs head in shame> to 1.6 in hopes
> that NDLB rport will fix it.
>
>
>
> Thanks to all who replied.
>
>
>
> *From:* FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] *On Behalf Of *Brian West
> *Sent:* Monday, February 19, 2018 8:08 AM
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] NAT port problem
>
>
>
> In this case its probably not an ALG, you should enable rport in your
> device and NOT force it with the NDLB param, if your device doesn't support
> it, throw that device in the trash or take a hammer to it, and buy a device
> from a manufacture that actually made an effort to implement SIP.
>
>
>
> /b
>
>
>
>
>
> On Mon, Feb 19, 2018 at 6:10 AM, Carlos Eduardo <kaduww at gmail.com> wrote:
>
> It can be an ALG problem (https://www.voip-info.org/
> wiki/view/Routers+SIP+ALG). Try running another sip profile with a
> different sip port, 5070 for example, and register the phone using this
> port.
>
>
>
> 2018-02-18 19:43 GMT-03:00 Sean Devoy <sdevoy at bizfocused.com>:
>
> Thanks Brian.  I should have been more clear.  That was exactly my
> diagnosis and attempted fix using this line:
>
> <param name="NDLB-force-rport" value="true"/>
>
> in the <param> section of the <user> in the Directory file has no effect.
>
>
>
> That means one of two things.  I am putting it in the wrong file or I must
> bite the bullet and get FS up to the current version.
>
>
>
> Thanks again.
>
>
>
> *From:* FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] *On Behalf Of *Brian West
> *Sent:* Sunday, February 18, 2018 11:51 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] NAT port problem
>
>
>
> Enable rport, The issue was from 15183 and we respond to 5074, so the
> device never receives the 401.
>
>
>
> Nat 101
>
>
>
> /b
>
>
>
>
>
> On Sun, Feb 18, 2018 at 11:28 AM, Sean Devoy <sdevoy at bizfocused.com>
> wrote:
>
> HI all,
>
>
>
> I have a NAT problem I can’t seem to get past.  The phone is NATed.  From
> my location with a FIOS router it works fine.  But the client recently
> upgraded to a new SonicWall router and splat, this device failed.  There
> are about 10 Cisco phones at this location working fine.  This phone is a
> Panasonic KX-TGP550.
>
>
>
> I have poured over the sip logs and identified the problem.  *The sip
> packets come from the router IP and port 15xxx but are returned to that IP
> and the phone’s source port 5060 instead of 15xxx.*
>
>
>
> I have tried every NDLB feature I can find, but not every combination. The
> responses always go back to the wrong port.
>
>
>
> Any help would be greatly appreciated.
>
>
>
> Thanks,
>
> Sean
>
>
>
> Sample sip dump:
>
>     ------------------------------------------------------------
> ------------
>
> recv 492 bytes from *udp/[98.204.241.22]:15183* at 23:16:33.764217:
>
>    ------------------------------------------------------------
> ------------
>
>      REGISTER sip:XYZ.bizfocused.com:5060 SIP/2.0
>
>      Via: SIP/2.0/UDP 192.168.2.109:5074;branch=z9hG4bKb6be703b
>
>      Max-Forwards: 70
>
>      To: <sip:34 at XYZ.bizfocused.com>
>
>      From: <sip:34 at XYZ.bizfocused.com>;tag=3263080642 <(32)%206308-0642>
>
>      Call-ID: 97742e94-d1dc14228bad9baadd810080f041d179 at 192.168.2.109
>
>      CSeq: 1 REGISTER
>
>      Contact: <sip:34 at 192.168.2.109:5074>
>
>      Expires: 3600
>
>      Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,NOTIFY,REFER,UPDATE
>
>      User-Agent: Panasonic_KX-TGP550T04/13.33 (0080f041d179)
>
>      Content-Length: 0
>
>
>
>     --------------------------------------------------------
> ----------------
>
> send 669 bytes to * udp/[98.204.241.22]:5074* at 23:16:33.764394:
>
>    ------------------------------------------------------------
> ------------
>
>      SIP/2.0 401 Unauthorized
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
>
>
>
>
> --
>
> 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 <(918)%20424-9378>
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
> color-twitter-96.png]
> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
>
>
>
> --
>
> *Carlos E. Wagner*
>
> *Tecnólogo em Telecomunicações, OCP, dCAA*
>
> *E-mail:* *kaduww at gmail.com <kaduww at gmail.com>*
>
> *Fone:* +55 48 9981-0894 <+55%2048%209981-0894>
> *Skype:* carlos.e.wagner
>
> www.blogdovoip.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
>
>
>
>
> --
>
> 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 <(918)%20424-9378>
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
> color-twitter-96.png]
> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From kaduww at gmail.com  Wed Feb 21 02:22:52 2018
From: kaduww at gmail.com (Carlos Eduardo)
Date: Tue, 20 Feb 2018 23:22:52 -0300
Subject: [Freeswitch-users] loopback_initial_codec
In-Reply-To: <7329ed62-1f44-85c8-2e4b-427291f415d2@nevian.org>
References: <5f5d6476-5022-e947-1e9e-48ee215f679e@nevian.org>
 <7329ed62-1f44-85c8-2e4b-427291f415d2@nevian.org>
Message-ID: <CAB=EceaUTppurQtyyswE1kSJQ85rLzONj1H567oOqE0phjVu1w@mail.gmail.com>

Hey,

Try using absolute_codec_string variable.

2018-02-20 13:17 GMT-03:00 Serge S. Yuriev <me at nevian.org>:

> Hi,
>
> Anyone? Maybe I do it wrong?
>
>
> On 16/02/18 19:44, Serge S. Yuriev wrote:
>
>> Hello,
>>
>> Using FS release 1.6.20
>>
>> I'm trying to originate call via loopback with preset codec
>> originate {loopback_initial_codec=PCMA}loopback/12550/internal  &park()
>>
>> But codec ignored and set to L16.
>>
>> 2018-02-16 18:51:48.294389 [DEBUG] switch_ivr_originate.c:2142 Parsing
>> global variables
>> 2018-02-16 18:51:48.294389 [NOTICE] switch_channel.c:1104 New Channel
>> loopback/12550/internal-a [9de6c978-4ef8-4f2d-a7d8-f630bebe13af]
>> 2018-02-16 18:51:48.294389 [DEBUG] mod_loopback.c:158
>> loopback/12550/internal-a setup codec L16/8000/20
>> 2018-02-16 18:51:48.294389 [NOTICE] switch_channel.c:1102 Rename Channel
>> loopback/12550/internal-a->loopback/12550-a
>> [9de6c978-4ef8-4f2d-a7d8-f630bebe13af]
>> 2018-02-16 18:51:48.294389 [DEBUG] mod_loopback.c:1174 (loopback/12550-a)
>> State Change CS_NEW -> CS_INIT
>> 2018-02-16 18:51:48.294389 [DEBUG] mod_loopback.c:601 loopback/12550-a
>> CHANNEL KILL
>> 2018-02-16 18:51:48.294389 [DEBUG] switch_core_state_machine.c:584
>> (loopback/12550-a) Running State Change CS_INIT (Cur 1 Tot 1230)
>> 2018-02-16 18:51:48.294389 [DEBUG] switch_core_state_machine.c:627
>> (loopback/12550-a) State INIT
>> 2018-02-16 18:51:48.294389 [NOTICE] switch_channel.c:1104 New Channel
>> loopback/12550-b [e945099e-2ed9-4c49-9d48-58a25756fb08]
>> 2018-02-16 18:51:48.294389 [DEBUG] mod_loopback.c:158 loopback/12550-b
>> setup codec L16/8000/20
>>
>>
> --
> Serge S. Yuriev
> Lead VoIP engineer
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 
*Carlos E. Wagner*
*Tecnólogo em Telecomunicações, OCP, dCAA*

*E-mail:* *kaduww at gmail.com <kaduww at gmail.com>*
*Fone:* +55 48 9981-0894
*Skype:* carlos.e.wagner
www.blogdovoip.com
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From netcentrica at gmail.com  Wed Feb 21 11:28:47 2018
From: netcentrica at gmail.com (Adam Raszynski)
Date: Wed, 21 Feb 2018 12:28:47 +0100
Subject: [Freeswitch-users] How to bridge session to user in valet_park
Message-ID: <CAJWEF2bXgY7UKPSOBTMa=9G7wN6GjavOU8mcuNX14NMmwZ7JBA@mail.gmail.com>

Hi

I have user put in valet_park waiting

In the meantime I execute lua script to create new session to external
number

I have uuid of this user in local variable, script is like this:

local uuid = "here uuid got from valet_info api call"
session1 = freeswitch.Session(dialstring);
if (session1:ready() and session1:answered()) then
     api:executeString("uuid_broadcast " .. uuid .. "
say::en\\snumber\\spronounced\\s1 aleg");
     api:executeString("uuid_bridge " .. uuid .. " " .. session1.uuid);
end

But it works sometimes. Most of time I get CS_RESET after uuid_bridge API
call

What I'm doing wrong here?

Maybe there is some other method of bridging my session1 with user waiting
in valet_park?

Can someone help here

Kind Regards
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From anthony.minessale at gmail.com  Tue Feb 20 16:30:53 2018
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 20 Feb 2018 16:30:53 +0000
Subject: [Freeswitch-users] The time on FreeSWITCH is faster than server
	time
In-Reply-To: <CAEJMVkDS_7XyZR=foAnY74gofjrF50hE8YWq7p0UnToYfOe27A@mail.gmail.com>
References: <AA810E17-392C-43C8-9E46-A916EADBC7F1@gmail.com>
 <1519119875310.774@itec-support.co.uk>
 <CAEJMVkDS_7XyZR=foAnY74gofjrF50hE8YWq7p0UnToYfOe27A@mail.gmail.com>
Message-ID: <CAKbxfG9er5VNxyv51arEtubuFpp=GkkB_avJTEMsRef=+fmK4Q@mail.gmail.com>

The reason it does not stay synced is because if the time changes mid call
it would break cdrs.

I think there is some param or command line switch to sync to system time
but its generally a bad idea.


On Tue, Feb 20, 2018 at 7:13 AM Brian West <brian at freeswitch.com> wrote:

> 10 seconds in a month is rather good to hear.
>
> On Tue, Feb 20, 2018 at 3:41 AM, Shaun Stokes <
> shaun.stokes at itec-support.co.uk> wrote:
>
>> If you're using FreeSWITCH in a virtual environment we've had similar
>> issues when using FreeSWITCH on a Hyper-V VM, the time on the VM seems to
>> run at a slightly difference pace. When the VM syncs with the host
>> FreeSWITCH doesn't pick this up.
>>
>> We found using sync_clock fixed this which we run daily as a scheduled
>> task which executes the command 'fsctl sync_clock_when_idle' via Event
>> Socket API, also disabled the time sync between the VM and switched to NTP.
>>
>> It would be useful if FreeSWITCH would execute this command every 24
>> hours out of the box, I can't think of any circumstances where this
>> behaviour wouldn't be desired.
>>
>> Shaun
>> ________________________________________
>> From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org>
>> on behalf of 王聡 <cong.wang.itsherpa at gmail.com>
>> Sent: 20 February 2018 09:08
>> To: FreeSWITCH Users Help
>> Subject: [Freeswitch-users] The time on FreeSWITCH is faster than server
>> time
>>
>> Hey all,
>>
>> Recently I noticed a clock problem on my FS server, the time in fs_cli
>> had about 10sec faster than server time(correct time), so I executed
>> sync_clock to correct it.
>> This server had run about 1 month, and I wonder if I need to make
>> sync_clock per month, and how to fix this clock problem.
>>
>> Regards.
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>> ______________________________________________________________________
>> This message has been checked for all known viruses by MessageLabs Virus
>> Scanning Service.
>> ______________________________________________________________________
>>
>> ______________________________________________________________________
>> This message has been checked for all known viruses by  MessageLabs Virus
>> Scanning Service.
>> ______________________________________________________________________
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
>
> --
>
> 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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
> color-twitter-96.png]
> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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

-- 
Anthony Minessale II
Founder, FreeSWITCH.
http://freeswitch.com


https://youtu.be/l_hOxzCt6X4
https://www.youtube.com/watch?v=oAxXgyx5jUw
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From randomdev4 at gmail.com  Tue Feb 20 19:04:11 2018
From: randomdev4 at gmail.com (Tim Smith)
Date: Tue, 20 Feb 2018 19:04:11 +0000
Subject: [Freeswitch-users] Need help getting mod_voicemail going (cuts off
	after greeting)
Message-ID: <CA+HuS5HK+gZ0Nz3xgmS0wcaQ0MRa+60fn91DEwXD13JdNV=64w@mail.gmail.com>

Hi,

With the config below, the handset rings as expected, then voicemail
picks up, but then all i head is "The person at extension goodbye", in
other words, freeswitch hangs up before giving me the chance to leave
a message.

Inbound dialplan for DID:

<include>
  <extension name="TEST_inbound">
    <condition field="destination_number" expression="^(000000)$">
     <action application="set" data="domain_name=$${domain}"/>
        <action application="export"
data="alert_info=<http://localhost/>;info=P1"/>
        <action application="export" data="nolocal:rtp_secure_media=true"/>
    <action application="set" data="hangup_after_bridge=true" />
    <action application="bridge"
data="{originate_timeout=5}[leg_timeout=5]${sofia_contact(internal/2003)}"/>
    <action application="answer"/>
    <action application="voicemail" data="default 5000"/>
    </condition>
  </extension>
</include>

Voicemail directory entry:

<include>
  <user id="5000">
    <params>
      <param name="a1-hash" value="aaaaaaaa" />
      <param name="vm-password" value="5000"/>
      <param name="manage-presence" value="true"/>
    </params>
  </user>
</include>




Pastebin of the call's dying moments :


From rmundkowsky at ets.org  Tue Feb 20 19:13:40 2018
From: rmundkowsky at ets.org (Mundkowsky, Robert)
Date: Tue, 20 Feb 2018 19:13:40 +0000
Subject: [Freeswitch-users] [rtp-timeout-sec][need experts comments]
In-Reply-To: <99066E27-A723-4585-9948-D5AC2456E6ED@jerris.com>
References: <CADSVerCfDk74ygoYxmnjJqYJzTtyvW1y7dLvWe8hE36At44DEQ@mail.gmail.com>
 <99066E27-A723-4585-9948-D5AC2456E6ED@jerris.com>
Message-ID: <CY1PR07MB2377795C1B544DEA264DF1E3BECF0@CY1PR07MB2377.namprd07.prod.outlook.com>

Sorry for the naïve question but are RTP packets with "silence" sent when on Hold?  Or are the RTP packets actually stopped.

Is the following ICE connection status a good way to check if the other side is down?

https://stackoverflow.com/questions/21233828/detecting-that-the-peers-browser-was-closed-in-a-webrtc-videochat


Robert Mundkowsky

-----Original Message-----
From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Tuesday, February 20, 2018 2:03 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] [rtp-timeout-sec][need experts comments]

rtp timeouts generally are a bad reason to tear down a call unless its actually stuck.  The problem with shorter times is sometimes with dtx and hold you can see it hangup on you.  With proper session timers there is typically no reason for rtp timers

> On Feb 8, 2018, at 12:43 PM, Bilal Abbasi <bilaln018 at gmail.com> wrote:
>
> Hi Users,
>
> I did checked the default value of rtp timeout is 300 seconds, for me its very long, i just wanted to know why it is placed that long, for me if a call is on silent for 30-40 seconds i will hang up that. any down side for doing this?
>
> Regards
> Abbasi


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C4e635ec412de47e7760908d57894e4a9%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636547503251976723&sdata=j2lgkGvCKJu4TSC2MKPnWSQQp%2BCewckwwuu7FydzFzI%3D&reserved=0

Official FreeSWITCH Sites
https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C4e635ec412de47e7760908d57894e4a9%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636547503251976723&sdata=vOHhwZO8IRDYXRZcFZq2EWf0V%2BzIhrWyBMGmctd5D0Q%3D&reserved=0
https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C4e635ec412de47e7760908d57894e4a9%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636547503251976723&sdata=TQ92r%2BWALOLOXTuUtM%2FhAXP2TjycYdQo0BAd3jO61Vw%3D&reserved=0
https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7C4e635ec412de47e7760908d57894e4a9%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636547503251976723&sdata=y3zso%2FRuNOqHBuDn03bJ5ut7tXPhvFP1JIiGjf3Jo%2F4%3D&reserved=0

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C4e635ec412de47e7760908d57894e4a9%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636547503251976723&sdata=ZzafTTnLAfDgrx8C6gUX2RcAegIvfZz5REC65hIyK4Q%3D&reserved=0
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C4e635ec412de47e7760908d57894e4a9%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636547503251976723&sdata=vOHhwZO8IRDYXRZcFZq2EWf0V%2BzIhrWyBMGmctd5D0Q%3D&reserved=0

________________________________

This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited.


Thank you for your compliance.

________________________________

From evan at corpwest.com  Tue Feb 20 20:37:42 2018
From: evan at corpwest.com (Evan P. Hall)
Date: Tue, 20 Feb 2018 20:37:42 +0000
Subject: [Freeswitch-users] NAT port problem
In-Reply-To: <BN6PR01MB25614520AD6BDF1A0C92762BBBCF0@BN6PR01MB2561.prod.exchangelabs.com>
References: <BN6PR01MB25617C7C3DF58C04AAAE0F1CBBC90@BN6PR01MB2561.prod.exchangelabs.com>
 <CAEJMVkAJ_DuSOZtXnwfLp5evKHv9Nr26TPYUj=BCExDuX7x8hw@mail.gmail.com>
 <BN6PR01MB2561BC0DB10D03F3284FB4E2BBC90@BN6PR01MB2561.prod.exchangelabs.com>
 <CAB=EcebwT4VdsycMCdTO=Qxizxhq7J5XP3do71=h3aP15rPQWA@mail.gmail.com>
 <CAEJMVkC8sRm-O0HJdEUoaN_SSP8fmK9BVPaw4QEtqsLDib2SSQ@mail.gmail.com>
 <BN6PR01MB25614520AD6BDF1A0C92762BBBCF0@BN6PR01MB2561.prod.exchangelabs.com>
Message-ID: <D68D2C39A2B8BB45B4E6029C9E4FEB200372BAF72B@MB01.hmc.corpwest.com>

We had this exact problem with these Panasonic phones.  We had a customer with 2 units that seemed identical to me, and one had this behavior and one didn’t. I tried resets, firmware updates, etc., but no hope for the one unit.  If it helps, the unit that is still actually working at their office shows useragent string of  Panasonic_KX-TGP550T04/12.58.

-Evan

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean Devoy
Sent: Tuesday, February 20, 2018 12:20 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] NAT port problem

LOL!  Exactly.  This device is, shall we say, VERY BASIC.  It worked with the old router. When they upgraded to a SONICWALL I cannot get it to work. I have learned my lesson. In the future I will only allow a quality ATA and DECT phone for this type of solution. For the record this is a Panasonic KX-TGP550.

I tried alternate local port numbers.  I tried all the ALG settings on the SonicWALL.  I am upgrading from 1.2 <hangs head in shame> to 1.6 in hopes that NDLB rport will fix it.

Thanks to all who replied.

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Monday, February 19, 2018 8:08 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] NAT port problem

In this case its probably not an ALG, you should enable rport in your device and NOT force it with the NDLB param, if your device doesn't support it, throw that device in the trash or take a hammer to it, and buy a device from a manufacture that actually made an effort to implement SIP.

/b


On Mon, Feb 19, 2018 at 6:10 AM, Carlos Eduardo <kaduww at gmail.com<mailto:kaduww at gmail.com>> wrote:
It can be an ALG problem (https://www.voip-info.org/wiki/view/Routers+SIP+ALG). Try running another sip profile with a different sip port, 5070 for example, and register the phone using this port.

2018-02-18 19:43 GMT-03:00 Sean Devoy <sdevoy at bizfocused.com<mailto:sdevoy at bizfocused.com>>:
Thanks Brian.  I should have been more clear.  That was exactly my diagnosis and attempted fix using this line:
<param name="NDLB-force-rport" value="true"/>
in the <param> section of the <user> in the Directory file has no effect.

That means one of two things.  I am putting it in the wrong file or I must bite the bullet and get FS up to the current version.

Thanks again.

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Brian West
Sent: Sunday, February 18, 2018 11:51 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] NAT port problem

Enable rport, The issue was from 15183 and we respond to 5074, so the device never receives the 401.

Nat 101

/b


On Sun, Feb 18, 2018 at 11:28 AM, Sean Devoy <sdevoy at bizfocused.com<mailto:sdevoy at bizfocused.com>> wrote:
HI all,

I have a NAT problem I can’t seem to get past.  The phone is NATed.  From my location with a FIOS router it works fine.  But the client recently upgraded to a new SonicWall router and splat, this device failed.  There are about 10 Cisco phones at this location working fine.  This phone is a Panasonic KX-TGP550.

I have poured over the sip logs and identified the problem.  The sip packets come from the router IP and port 15xxx but are returned to that IP and the phone’s source port 5060 instead of 15xxx.

I have tried every NDLB feature I can find, but not every combination. The responses always go back to the wrong port.

Any help would be greatly appreciated.

Thanks,
Sean

Sample sip dump:
    ------------------------------------------------------------------------
recv 492 bytes from udp/[98.204.241.22]:15183 at 23:16:33.764217:
   ------------------------------------------------------------------------
     REGISTER sip:XYZ.bizfocused.com:5060 SIP/2.0
     Via: SIP/2.0/UDP 192.168.2.109:5074;branch=z9hG4bKb6be703b
     Max-Forwards: 70
     To: <sip:34 at XYZ.bizfocused.com>
     From: <sip:34 at XYZ.bizfocused.com>;tag=3263080642<tel:(32)%206308-0642>
     Call-ID: 97742e94-d1dc14228bad9baadd810080f041d179 at 192.168.2.109<mailto:97742e94-d1dc14228bad9baadd810080f041d179 at 192.168.2.109>
     CSeq: 1 REGISTER
     Contact: <sip:34 at 192.168.2.109:5074>
     Expires: 3600
     Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,NOTIFY,REFER,UPDATE
     User-Agent: Panasonic_KX-TGP550T04/13.33 (0080f041d179)
     Content-Length: 0

    ------------------------------------------------------------------------
send 669 bytes to udp/[98.204.241.22]:5074 at 23:16:33.764394:
   ------------------------------------------------------------------------
     SIP/2.0 401 Unauthorized



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
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http://confluence.freeswitch.org
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FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
[https://docs.google.com/uc?export=download&id=1xswZRZyVDo0WQhaemK47pU266yzDRmi0&revid=0B2xnT7i45ngrMTVKM1dpSHZIN28zU0QzbW9xeVF6RXFyRHhBPQ]

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<tel:(918)%20424-9378>

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

[color-facebook-96.png]<https://www.facebook.com/freeswitch/>[color-twitter-96.png]<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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--
Carlos E. Wagner
Tecnólogo em Telecomunicações, OCP, dCAA
E-mail: kaduww at gmail.com<mailto:kaduww at gmail.com>
Fone: +55 48 9981-0894<tel:+55%2048%209981-0894>
Skype: carlos.e.wagner
www.blogdovoip.com<http://www.blogdovoip.com>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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--
[https://docs.google.com/uc?export=download&id=1xswZRZyVDo0WQhaemK47pU266yzDRmi0&revid=0B2xnT7i45ngrMTVKM1dpSHZIN28zU0QzbW9xeVF6RXFyRHhBPQ]

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/>

[color-facebook-96.png]<https://www.facebook.com/freeswitch/>[color-twitter-96.png]<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From umair at tezrosolutions.com  Tue Feb 20 22:13:35 2018
From: umair at tezrosolutions.com (Muhammad Umair)
Date: Tue, 20 Feb 2018 17:13:35 -0500
Subject: [Freeswitch-users] CALL DOES NOT EXIST error in mod_verto
Message-ID: <791F2C1E-4DCE-4D9F-8065-BBCB34BE1AA5@tezrosolutions.com>

Hi,

I have built Freeswitch on window 8 (64bit) using Visual Studio. Every time I try to join a conference it gives me following error in Javascript:

INVALID METHOD OR NON-EXISTANT CALL REFERENCE IGNORED verto.clientReady

I have attached FS logs as well.


Any ideas?

- Umair

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From b631093f-779b-4d67-9ffe-5f6d5b1d3f8a at protonmail.ch  Tue Feb 20 22:18:13 2018
From: b631093f-779b-4d67-9ffe-5f6d5b1d3f8a at protonmail.ch (Bob Smith)
Date: Tue, 20 Feb 2018 17:18:13 -0500
Subject: [Freeswitch-users] mod_voicemail configuration issues
Message-ID: <kgo0vJqMooJv4okrB4zqy9h-2r8SXQMZU3RB7VgIAD_Oobst-l4IoOcxc56ypst1krYdyobME4MxMxyIgaOrPmcsAqp8CrCgL_9GP4Y1H2A=@protonmail.ch>

Hi,

With the config below, the handset rings as expected, then voicemail
picks up, but then all i head is "The person at extension goodbye", in
other words, freeswitch hangs up before giving me the chance to leave
a message.

Inbound dialplan for DID:

<include>
  <extension name="TEST_inbound">
    <condition field="destination_number" expression="^(000000)$">
     <action application="set" data="domain_name=$${domain}"/>
        <action application="export"
data="alert_info=<http://localhost/>;info=P1"/>
        <action application="export" data="nolocal:rtp_secure_media=true"/>
    <action application="set" data="hangup_after_bridge=true" />
    <action application="bridge"
data="{originate_timeout=5}[leg_timeout=5]${sofia_contact(internal/2003)}"/>
    <action application="answer"/>
    <action application="voicemail" data="default 5000"/>
    </condition>
  </extension>
</include>

Voicemail directory entry:

<include>
  <user id="5000">
    <params>
      <param name="a1-hash" value="aaaaaaaa" />
      <param name="vm-password" value="5000"/>
      <param name="manage-presence" value="true"/>
    </params>
  </user>
</include>

Pastebin of the call's dying moments :

https://pastebin.com/NqurmFJv
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From jeff at ugnd.org  Wed Feb 21 12:35:11 2018
From: jeff at ugnd.org (Jeff Pyle)
Date: Wed, 21 Feb 2018 07:35:11 -0500
Subject: [Freeswitch-users] access incoming Alert-Info header in dial plan
Message-ID: <CAPhW+0JckjB0QpDMBJF3LtocMPP-6tnd7X_XeLc+r-nhhVWH-A@mail.gmail.com>

Hello,

Is there any way to access the Alert-Info header of an incoming SIP call?
I've tried ${sip_h_Alert-Info} but there's nothing in it.  In fact, running
the info application doesn't show Alert-Info's contents anywhere.


Regards,
Jeff
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From umair at tezrosolutions.com  Wed Feb 21 16:04:11 2018
From: umair at tezrosolutions.com (Muhammad Umair)
Date: Wed, 21 Feb 2018 11:04:11 -0500
Subject: [Freeswitch-users] Passthrough conference video frame rate
Message-ID: <ED21FD45-54F7-4E46-8C4A-AC4E18C7AAE5@tezrosolutions.com>

Hi,

I am wondering what is the video frame rate in conference passthrough mode. Any ideas?




From vma at vallimamod.org  Wed Feb 21 16:42:16 2018
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Wed, 21 Feb 2018 17:42:16 +0100
Subject: [Freeswitch-users] access incoming Alert-Info header in dial
	plan
In-Reply-To: <CAPhW+0JckjB0QpDMBJF3LtocMPP-6tnd7X_XeLc+r-nhhVWH-A@mail.gmail.com>
References: <CAPhW+0JckjB0QpDMBJF3LtocMPP-6tnd7X_XeLc+r-nhhVWH-A@mail.gmail.com>
Message-ID: <2AFEE56A-0030-4E33-B488-8486BB603254@vallimamod.org>

Hi,

According to the source, the corresponding channel var is ${alert_info} and the event header, Alert-Info.
Hope this helps.

Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sipsolutions.fr
.

> On 21 Feb 2018, at 13:35, Jeff Pyle <jeff at ugnd.org> wrote:
> 
> Hello,
> 
> Is there any way to access the Alert-Info header of an incoming SIP call?  I've tried ${sip_h_Alert-Info} but there's nothing in it.  In fact, running the info application doesn't show Alert-Info's contents anywhere.
> 
> 
> Regards,
> Jeff
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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



From jungleboogie0 at gmail.com  Wed Feb 21 17:38:39 2018
From: jungleboogie0 at gmail.com (Jungle Boogie)
Date: Wed, 21 Feb 2018 09:38:39 -0800
Subject: [Freeswitch-users] mod_voicemail configuration issues
In-Reply-To: <kgo0vJqMooJv4okrB4zqy9h-2r8SXQMZU3RB7VgIAD_Oobst-l4IoOcxc56ypst1krYdyobME4MxMxyIgaOrPmcsAqp8CrCgL_9GP4Y1H2A=@protonmail.ch>
References: <kgo0vJqMooJv4okrB4zqy9h-2r8SXQMZU3RB7VgIAD_Oobst-l4IoOcxc56ypst1krYdyobME4MxMxyIgaOrPmcsAqp8CrCgL_9GP4Y1H2A=@protonmail.ch>
Message-ID: <20180221173839.GA26261@puffer.in.lylie.net>

On Tue 20 Feb 2018  5:18 PM, Bob Smith wrote:
> Hi,
> 
> With the config below, the handset rings as expected, then voicemail
> picks up, but then all i head is "The person at extension goodbye", in
> other words, freeswitch hangs up before giving me the chance to leave
> a message.

Does this happen if you call the extension directly?
I've seen this before, but I don't remember the cause of it. 
Is it affecting other voicemail boxes?

Do you have a sip trace of the issue?


From b631093f-779b-4d67-9ffe-5f6d5b1d3f8a at protonmail.ch  Wed Feb 21 19:07:20 2018
From: b631093f-779b-4d67-9ffe-5f6d5b1d3f8a at protonmail.ch (Bob Smith)
Date: Wed, 21 Feb 2018 14:07:20 -0500
Subject: [Freeswitch-users] mod_voicemail configuration issues
In-Reply-To: <20180221173839.GA26261@puffer.in.lylie.net>
References: <kgo0vJqMooJv4okrB4zqy9h-2r8SXQMZU3RB7VgIAD_Oobst-l4IoOcxc56ypst1krYdyobME4MxMxyIgaOrPmcsAqp8CrCgL_9GP4Y1H2A=@protonmail.ch>
 <20180221173839.GA26261@puffer.in.lylie.net>
Message-ID: <1Dhn2Nj3unF3XIapFBF0Zam6cCieL1ZrMZENJHPE81PWoe5JhEiMP7eWK0QJ8Uj5VNBisf_GxK_glPG0fH1dNse6YD7CQHdiOUVABZ37gEY=@protonmail.ch>

Hi,

Thanks for your reply.

SIP trace here:
https://pastebin.com/H4VWd7Xe

This SIP trace was with the config below (which still does not work, but is the culmination of too many hours in the company of Mr Google finding various parameters to work with - I have also tried the two configs presented in the docs and they do not work either https://freeswitch.org/confluence/display/FREESWITCH/mod_voicemail) 

<include>
  <extension name="FOOBAR_inbound">
    <condition field="destination_number" expression="^(00000000000000)$">
     <action application="set" data="domain_name=$${domain}"/> 
        <action application="export" data="alert_info=<http://localhost/>;info=P1"/>
        <action application="export" data="nolocal:rtp_secure_media=true"/>
    <action application="set" data="hangup_after_bridge=true" />
    <action application="set" data="continue_on_fail=true"/>
    <action application="set" data="ringback=${us-ring}"/>
    <action application="set" data="uuid_bridge_continue_on_cancel=true"/>
    <action application="bridge" data="{ignore_early_media=ring_ready}{originate_timeout=5}[leg_timeout=5]${sofia_contact(internal/2003)}"/>
    <action application="answer"/>
    <action application="voicemail" data="default 5000"/>
    <action application="hangup"/>
    </condition>
  </extension>
</include>
​

-------- Original Message --------
 On February 21, 2018 5:38 PM, Jungle Boogie <jungleboogie0 at gmail.com> wrote:

>On Tue 20 Feb 2018  5:18 PM, Bob Smith wrote:
>>Hi,
>>With the config below, the handset rings as expected, then voicemail
>> picks up, but then all i head is "The person at extension goodbye", in
>> other words, freeswitch hangs up before giving me the chance to leave
>> a message.
>>
> Does this happen if you call the extension directly?
> I've seen this before, but I don't remember the cause of it.
> Is it affecting other voicemail boxes?
>
> Do you have a sip trace of the issue?
>
>
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://confluence.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
>



From basmo13 at yahoo.co.uk  Wed Feb 21 19:48:47 2018
From: basmo13 at yahoo.co.uk (Sonnie)
Date: Wed, 21 Feb 2018 19:48:47 +0000 (UTC)
Subject: [Freeswitch-users] Upload file in Lua background
References: <616482861.4777902.1519242527600.ref@mail.yahoo.com>
Message-ID: <616482861.4777902.1519242527600@mail.yahoo.com>

 I'm writing some IVR in Lua  and trying to execute some code while the main program continues execution.    I've tried       ...    -- aws is AWS Command Line Interface    io.popen("aws s3 cp " ..  s3_src .. " ".. s3_dst)    os.execute("aws s3 cp " ..  s3_src .. " ".. s3_dst)    ... -- program execution continues    
Even tried  passing it off to a shell script(`s3_upload`)
    os.execute("s3_upload " ..  s3_src .." " .. s3_dst)    io.popen("s3_upload " ..  s3_src .. " ".. s3_dst)
I've tried 
 session:execute("bgsystem","s3_upload " ..  s3_src .. " ".. s3_dst )


If i use the Lua Interactive shell to run these commands independently, it works.
would appreciate any assistance.
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From basmo13 at yahoo.co.uk  Wed Feb 21 19:51:52 2018
From: basmo13 at yahoo.co.uk (Sonnie)
Date: Wed, 21 Feb 2018 19:51:52 +0000 (UTC)
Subject: [Freeswitch-users] Upload file in background - Lua
References: <1793295444.4758145.1519242712215.ref@mail.yahoo.com>
Message-ID: <1793295444.4758145.1519242712215@mail.yahoo.com>

I'm writing some IVR in Lua  and trying to execute some code while the main program continues execution.    I've tried       ...    -- aws is AWS Command Line Interface    io.popen("aws s3 cp " ..  s3_src .. " ".. s3_dst)    os.execute("aws s3 cp " ..  s3_src .. " ".. s3_dst)    ... -- program execution continues    
Even tried  passing it off to a shell script(`s3_upload`)
    os.execute("s3_upload " ..  s3_src .." " .. s3_dst)    io.popen("s3_upload " ..  s3_src .. " ".. s3_dst)
I've tried 
 session:execute("bgsystem"," s3_upload " ..  s3_src .. " ".. s3_dst )


If i use the Lua Interactive shell to run these commands independently, it works.
would appreciate any assistance. 
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From mike at jerris.com  Wed Feb 21 21:04:51 2018
From: mike at jerris.com (Michael Jerris)
Date: Wed, 21 Feb 2018 16:04:51 -0500
Subject: [Freeswitch-users] about mod_av
In-Reply-To: <2640b5f9-e5a8-41bc-4965-8b7eec893421@madovsky.org>
References: <2640b5f9-e5a8-41bc-4965-8b7eec893421@madovsky.org>
Message-ID: <D2A64BC0-8B67-40EE-B577-E4AE132CB1CC@jerris.com>

compiles with both.  Not sure about the very latest versions but I know it compiles with what is in deb9, which is fairly new.

> On Feb 17, 2018, at 8:10 AM, Madovsky <infos at madovsky.org> wrote:
> 
> Hi folks,
> 
> is mod_av compile with ffmpeg or libav?
> 
> also what the last version ffmpeg/libav supported by mod_av to compile correctly?
> 
> thanks
> 
> Franck
> 



From mike at jerris.com  Wed Feb 21 21:11:05 2018
From: mike at jerris.com (Michael Jerris)
Date: Wed, 21 Feb 2018 16:11:05 -0500
Subject: [Freeswitch-users] Freeswitch Tables
In-Reply-To: <CALXCt0pVDWgGboZpzG5AQN6seuToyHaWn-jt9H6yN3RxoXhhdg@mail.gmail.com>
References: <CALLkTp1dXmrt4aXBhUvEH7R0usVZyXtnnxgHOVq1QohW+w3qGg@mail.gmail.com>
 <CALXCt0pVDWgGboZpzG5AQN6seuToyHaWn-jt9H6yN3RxoXhhdg@mail.gmail.com>
Message-ID: <7C149F8D-C9B4-47C0-9AAE-FD562E55FB19@jerris.com>

what they are for:

aliases: fs_cli command aliases
calls: list of currently active calls for show calls
channels: list of currently active channels for show channels
complete: tab complete data for fs_cli
db_data: for the db application data storage
group_data: for the group command data storages
interfaces: for the show interfaces command
nat: active nat mappings 
recovery: state data for call recovery
registrations: central data repo for storing registration data
sip_authentication: storage of sip nonce info
sip_dialogs: sip current dialog info, used for presence
sip_presence: used for presence state
sip_registrations: used for data for active sip reg
sip_shared_appearance_*: used for another kind of sip presence
sip_subscriptions: to store active subscriptions
tasks: for the task scheduler

> On Feb 17, 2018, at 1:04 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> 
> 
> 
> On 17 February 2018 at 16:42, Alexander Perkins <alexanderhenryperkins at gmail.com <mailto:alexanderhenryperkins at gmail.com>> wrote:
> Hi.  Does anybody have documentation regarding what these tables are for?  I've looked online, but cannot find it.
> 
> <image.png>
> 
> 
> those are used by FS when you put core in external database.
> 
> they substitute tables in /usr/local/freeswitch/db/
> 
> Eg, check https://freeswitch.org/confluence/display/FREESWITCH/PostgreSQL+in+the+core <https://freeswitch.org/confluence/display/FREESWITCH/PostgreSQL+in+the+core>
> 

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From mike at jerris.com  Wed Feb 21 23:07:11 2018
From: mike at jerris.com (Michael Jerris)
Date: Wed, 21 Feb 2018 18:07:11 -0500
Subject: [Freeswitch-users] freze when installing Freeswitch
In-Reply-To: <da62c7c9-d75f-0fd2-1c86-a69ceb96d02e@couderc.eu>
References: <da62c7c9-d75f-0fd2-1c86-a69ceb96d02e@couderc.eu>
Message-ID: <017CF0D9-5321-430F-BA0D-087E3B80E866@jerris.com>

this is something blocking in a non freeswitch package postinstall hooks.  You’d have to research issues installing policykit.  Its a dep of some lib, i don’t know about it off hand and haven’t seen this issue before.


> On Feb 20, 2018, at 2:07 AM, Pierre Couderc <pierre at couderc.eu> wrote:
> 
> I am trying to install my first Freeswitch nearly in debian 8 :
> 
> I have followed https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie :
> 
> wget -O - https://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | apt-key add -
> echo "deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie main" > /etc/apt/sources.list.d/freeswitch.list
> apt-get update && apt-get install -y freeswitch-meta-all
> 
> It starts well but does not give back the prompt :
> 
> Setting up tzdata-java (2017c-0+deb8u1) ...
> Setting up sgml-base (1.26+nmu4) ...
> Setting up vdpau-va-driver:amd64 (0.7.4-3) ...
> Setting up libtxc-dxtn-s2tc0:amd64 (0~git20131104-1.1) ...
> update-alternatives: using /usr/lib/x86_64-linux-gnu/libtxc_dxtn_s2tc.so.0 to provide /usr/lib/x86_64-linux-gnu/libtxc_dxtn.so (libtxc-dxtn-x86_64-linux-gnu) in auto mode
> Setting up file (1:5.22+15-2+deb8u3) ...
> Setting up at-spi2-core (2.14.0-1) ...
> Setting up policykit-1 (0.105-15~deb8u2) ...
> Removed symlink /run/systemd/system/polkitd.service.
> 
> ---> never returns the prompt here.
> 
> 
> if I try on another console :
> 
> root at test:~# fs_cli -rRS
> [ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error]
> [INFO] fs_cli.c:1665 main() Retrying
> [ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error]
> [INFO] fs_cli.c:1665 main() Retrying
> ...
> 
> Please note that my debian 8 is a LXD container inside a debian 9 host.
> 
> My question is : what should I do ? How to retry ?
> 
> 
> Pierre Couderc
> 



From jeff at ugnd.org  Wed Feb 21 21:14:10 2018
From: jeff at ugnd.org (Jeff Pyle)
Date: Wed, 21 Feb 2018 16:14:10 -0500
Subject: [Freeswitch-users] access incoming Alert-Info header in dial
	plan
In-Reply-To: <2AFEE56A-0030-4E33-B488-8486BB603254@vallimamod.org>
References: <CAPhW+0JckjB0QpDMBJF3LtocMPP-6tnd7X_XeLc+r-nhhVWH-A@mail.gmail.com>
 <2AFEE56A-0030-4E33-B488-8486BB603254@vallimamod.org>
Message-ID: <CAPhW+0K1hDS=eQ+3rSGX0kds=OaGLWCuLG=uZFtna29iV=MXMQ@mail.gmail.com>

Hi,

That variable is not there / empty.  I checked explicitly with a log
statement.  It's also not in the info dump.

I looked at the source also, and from what I could tell I can write
${alert_info} to create an Alert-Info header for the B-leg, but not read an
incoming Alert-Info on the A-leg.


- Jeff



On Wed, Feb 21, 2018 at 11:42 AM, Vallimamod Abdullah <vma at vallimamod.org>
wrote:

> Hi,
>
> According to the source, the corresponding channel var is ${alert_info}
> and the event header, Alert-Info.
> Hope this helps.
>
> Best Regards,
> --
> Vallimamod Abdullah
> SIP Solutions
> vma at sipsolutions.fr
>
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From larry.hemenway at gmail.com  Wed Feb 21 23:09:09 2018
From: larry.hemenway at gmail.com (Larry Hemenway)
Date: Wed, 21 Feb 2018 17:09:09 -0600
Subject: [Freeswitch-users] Receiving audio despite SDP with a=sendonly
Message-ID: <CAN=Q5SBZ9iyR1HFFxK0GycJFSC=u5aW5=N_bgXV5tsq39DCsvA@mail.gmail.com>

Hello,

Is there a way to establish a call with one-way audio on a call from the start?

I'm currently sending a request with the following SDP - (note a=sendonly):

v=0
o=Larry 2890844526 2890844526 IN IP4 127.0.0.1
s= My Session
c=IN IP4 172.22.112.1
t=0 0
m=audio 49170 RTP/AVP 0 8
a=sendonly
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

FreeSWITCH is sending back the right response (note the a=recvonly):

v=0
o=FreeSWITCH 1519171720 1519171721 IN IP4 172.17.0.2
s=FreeSWITCH
c=IN IP4 172.17.0.2
t=0 0
m=audio 21768 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=recvonly
a=ptime:20

But I'm still receiving audio from FreeSWITCH. It is only after I
resend the SDP via a reinvite that the audio stops being sent from
FreeSWITCH.

Larry


From mike at jerris.com  Wed Feb 21 23:15:47 2018
From: mike at jerris.com (Michael Jerris)
Date: Wed, 21 Feb 2018 18:15:47 -0500
Subject: [Freeswitch-users] How to bridge session to user in valet_park
In-Reply-To: <CAJWEF2bXgY7UKPSOBTMa=9G7wN6GjavOU8mcuNX14NMmwZ7JBA@mail.gmail.com>
References: <CAJWEF2bXgY7UKPSOBTMa=9G7wN6GjavOU8mcuNX14NMmwZ7JBA@mail.gmail.com>
Message-ID: <5512AA68-B457-4CBD-9001-AA5B2D4B141C@jerris.com>

why this complicated process instead of just bridging the call?

> On Feb 21, 2018, at 6:28 AM, Adam Raszynski <netcentrica at gmail.com> wrote:
> 
> Hi
> 
> I have user put in valet_park waiting
> 
> In the meantime I execute lua script to create new session to external number
> 
> I have uuid of this user in local variable, script is like this:
> 
> local uuid = "here uuid got from valet_info api call"
> session1 = freeswitch.Session(dialstring);
> if (session1:ready() and session1:answered()) then
>      api:executeString("uuid_broadcast " .. uuid .. " say::en\\snumber\\spronounced\\s1 aleg");
>      api:executeString("uuid_bridge " .. uuid .. " " .. session1.uuid);
> end
> 
> But it works sometimes. Most of time I get CS_RESET after uuid_bridge API call
> 
> What I'm doing wrong here?
> 
> Maybe there is some other method of bridging my session1 with user waiting in valet_park?



From mike at jerris.com  Wed Feb 21 23:17:57 2018
From: mike at jerris.com (Michael Jerris)
Date: Wed, 21 Feb 2018 18:17:57 -0500
Subject: [Freeswitch-users] [rtp-timeout-sec][need experts comments]
In-Reply-To: <CY1PR07MB2377795C1B544DEA264DF1E3BECF0@CY1PR07MB2377.namprd07.prod.outlook.com>
References: <CADSVerCfDk74ygoYxmnjJqYJzTtyvW1y7dLvWe8hE36At44DEQ@mail.gmail.com>
 <99066E27-A723-4585-9948-D5AC2456E6ED@jerris.com>
 <CY1PR07MB2377795C1B544DEA264DF1E3BECF0@CY1PR07MB2377.namprd07.prod.outlook.com>
Message-ID: <1113B8A3-D5BC-4A9D-8CB6-474F561322C2@jerris.com>

it depends on the endpoint.  There is no reason to send packets when there is silence but many endpoints do anyways.  Due to this, people start depending on rtp timeouts for things.  in ice setups even if there is no rtp there is typically some ice exchange still.  Still none of this is the best way to see a connection dropped, sip times or signaling based methods are better.

> On Feb 20, 2018, at 2:13 PM, Mundkowsky, Robert <rmundkowsky at ets.org> wrote:
> 
> Sorry for the naïve question but are RTP packets with "silence" sent when on Hold?  Or are the RTP packets actually stopped.
> 
> Is the following ICE connection status a good way to check if the other side is down?
> 
> https://stackoverflow.com/questions/21233828/detecting-that-the-peers-browser-was-closed-in-a-webrtc-videochat
> 
> 
> Robert Mundkowsky
> 
> -----Original Message-----
> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
> Sent: Tuesday, February 20, 2018 2:03 PM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] [rtp-timeout-sec][need experts comments]
> 
> rtp timeouts generally are a bad reason to tear down a call unless its actually stuck.  The problem with shorter times is sometimes with dtx and hold you can see it hangup on you.  With proper session timers there is typically no reason for rtp timers
> 
>> On Feb 8, 2018, at 12:43 PM, Bilal Abbasi <bilaln018 at gmail.com> wrote:
>> 
>> Hi Users,
>> 
>> I did checked the default value of rtp timeout is 300 seconds, for me its very long, i just wanted to know why it is placed that long, for me if a call is on silent for 30-40 seconds i will hang up that. any down side for doing this?
>> 



From mike at jerris.com  Wed Feb 21 23:19:48 2018
From: mike at jerris.com (Michael Jerris)
Date: Wed, 21 Feb 2018 18:19:48 -0500
Subject: [Freeswitch-users] Need help getting mod_voicemail going (cuts
 off after greeting)
In-Reply-To: <CA+HuS5HK+gZ0Nz3xgmS0wcaQ0MRa+60fn91DEwXD13JdNV=64w@mail.gmail.com>
References: <CA+HuS5HK+gZ0Nz3xgmS0wcaQ0MRa+60fn91DEwXD13JdNV=64w@mail.gmail.com>
Message-ID: <412FCEF7-8AED-498C-99E7-8371993689E4@jerris.com>

the logs probably has more info about whats going on.

> On Feb 20, 2018, at 2:04 PM, Tim Smith <randomdev4 at gmail.com> wrote:
> 
> Hi,
> 
> With the config below, the handset rings as expected, then voicemail
> picks up, but then all i head is "The person at extension goodbye", in
> other words, freeswitch hangs up before giving me the chance to leave
> a message.
> 
> Inbound dialplan for DID:
> 
> <include>
>  <extension name="TEST_inbound">
>    <condition field="destination_number" expression="^(000000)$">
>     <action application="set" data="domain_name=$${domain}"/>
>        <action application="export"
> data="alert_info=<http://localhost/>;info=P1"/>
>        <action application="export" data="nolocal:rtp_secure_media=true"/>
>    <action application="set" data="hangup_after_bridge=true" />
>    <action application="bridge"
> data="{originate_timeout=5}[leg_timeout=5]${sofia_contact(internal/2003)}"/>
>    <action application="answer"/>
>    <action application="voicemail" data="default 5000"/>
>    </condition>
>  </extension>
> </include>
> 
> Voicemail directory entry:
> 
> <include>
>  <user id="5000">
>    <params>
>      <param name="a1-hash" value="aaaaaaaa" />
>      <param name="vm-password" value="5000"/>
>      <param name="manage-presence" value="true"/>
>    </params>
>  </user>
> </include>
> 
> 



From mike at jerris.com  Wed Feb 21 23:22:12 2018
From: mike at jerris.com (Michael Jerris)
Date: Wed, 21 Feb 2018 18:22:12 -0500
Subject: [Freeswitch-users] Passthrough conference video frame rate
In-Reply-To: <ED21FD45-54F7-4E46-8C4A-AC4E18C7AAE5@tezrosolutions.com>
References: <ED21FD45-54F7-4E46-8C4A-AC4E18C7AAE5@tezrosolutions.com>
Message-ID: <3BB3EDAD-47D1-47FF-8D46-A91CE7336740@jerris.com>

whatever the rate of the sending device is.

> On Feb 21, 2018, at 11:04 AM, Muhammad Umair <umair at tezrosolutions.com> wrote:
> 
> Hi,
> 
> I am wondering what is the video frame rate in conference passthrough mode. Any ideas?
> 



From mike at jerris.com  Wed Feb 21 23:23:25 2018
From: mike at jerris.com (Michael Jerris)
Date: Wed, 21 Feb 2018 18:23:25 -0500
Subject: [Freeswitch-users] Upload file in Lua background
In-Reply-To: <616482861.4777902.1519242527600@mail.yahoo.com>
References: <616482861.4777902.1519242527600.ref@mail.yahoo.com>
 <616482861.4777902.1519242527600@mail.yahoo.com>
Message-ID: <DF8BE28D-CA6A-482E-A5A3-523FC52A376E@jerris.com>

to do it in the background it needs to run in a different thread.  Sometime with bgapi executing the command to do the upload passing the right params should work.

> On Feb 21, 2018, at 2:48 PM, Sonnie <basmo13 at yahoo.co.uk> wrote:
> 
>  I'm writing some IVR in Lua  and trying to execute some code while the main program continues execution. 
>    
> I've tried 
>   
>     ...
>     -- aws is AWS Command Line Interface
>     io.popen("aws s3 cp " ..  s3_src .. " ".. s3_dst)
>     os.execute("aws s3 cp " ..  s3_src .. " ".. s3_dst)
>     ... -- program execution continues
>     
> 
> Even tried  passing it off to a shell script(`s3_upload`)
> 
>     os.execute("s3_upload " ..  s3_src .." " .. s3_dst)
>     io.popen("s3_upload " ..  s3_src .. " ".. s3_dst)
> 
> I've tried 
> 
>  session:execute("bgsystem","s3_upload " ..  s3_src .. " ".. s3_dst )
> 
> 
> If i use the Lua Interactive shell to run these commands independently, it works.
> 
> would appreciate any assistance.

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From vineet.verma at bics.com  Thu Feb 22 00:11:06 2018
From: vineet.verma at bics.com (vineet)
Date: Wed, 21 Feb 2018 17:11:06 -0700 (MST)
Subject: [Freeswitch-users] RTP is not getting passed by freeswitch
Message-ID: <1519258266914-0.post@n2.nabble.com>

Dears,

I am facing some issue with my freeswitch that that its getting RTP from
remote SBC but but it is not passing the RTP to B party after bridging the
call.
signaling is working fine.

also using TCPDUMP 
I see that freeswitch is looping the RTP to its own IP and port.

Can some one Please help me to resolve this issue.

Thanks,
Vineet





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


From gmaruzz at gmail.com  Thu Feb 22 09:05:37 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 22 Feb 2018 10:05:37 +0100
Subject: [Freeswitch-users] Fwd: confirm d512ce407ddf84b1aee26e096771
Message-ID: <CALXCt0qxD=8oRb91dAHAtQDi=VD6hAVCbenStFAMrt4d4_Ki6A@mail.gmail.com>

List problems again?


---------- Forwarded message ----------
From: <freeswitch-users-request at lists.freeswitch.org>
Date: 22 February 2018 at 01:44
Subject: confirm d512ce407ddf84b1aee26e0967
To: gmaruzz at gmail.com


Your membership in the mailing list FreeSWITCH-users has been disabled
due to excessive bounces The last bounce received from you was dated
22-Feb-2018.  You will not get any more messages from this list until
you re-enable your membership.  You will receive 3 more reminders like
this before your membership in the list is deleted.

To re-enable your membership, you can simply respond to this message
(leaving the Subject: line intact), or visit the confirmation page at

    http://lists.freeswitch.org/mailman/confirm/freeswitch-users/
d512ce407ddf84b1aee
<http://lists.freeswitch.org/mailman/confirm/freeswitch-users/d512ce407ddf84b1aee26e09677135e861863512>


You can also visit your membership page at

    http://lists.freeswitch.org/mailman/options/freeswitch-
users/gmaruzz%40gmail.com


On your membership page, you can change various delivery options such
as your email address and whether you get digests or not.  As a
reminder, your membership password is

    xxxxx

If you have any questions or problems, you can contact the list owner
at

    freeswitch-users-owner at lists.freeswitch.org



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From krice at freeswitch.org  Thu Feb 22 09:14:08 2018
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 22 Feb 2018 03:14:08 -0600
Subject: [Freeswitch-users] Fwd: confirm d512ce407ddf84b1aee26e096771
In-Reply-To: <CALXCt0qxD=8oRb91dAHAtQDi=VD6hAVCbenStFAMrt4d4_Ki6A@mail.gmail.com>
References: <CALXCt0qxD=8oRb91dAHAtQDi=VD6hAVCbenStFAMrt4d4_Ki6A@mail.gmail.com>
Message-ID: <04e901d3abbd$7f063c00$7d12b400$@freeswitch.org>

Nope this happens on google hosted accounts all the time. Mailman uses the SMTP protocol fully, google doesn’t allow multiple domains in the same transation, so if say krice at freeswitch.org <mailto:krice at freeswitch.org>  and gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>  are sent in the same transaction even tho they have the same MX host, google will reject any that don’t match the first domain.

 

Just reply to it and you’ll reset the counter.

 

I have to do this a couple of times a month

K

 

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli
Sent: Thursday, February 22, 2018 3:06 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Fwd: confirm d512ce407ddf84b1aee26e096771

 

 

List problems again?

 

---------- Forwarded message ----------
From: <freeswitch-users-request at lists.freeswitch.org <mailto:freeswitch-users-request at lists.freeswitch.org> >
Date: 22 February 2018 at 01:44
Subject: confirm d512ce407ddf84b1aee26e0967
To: gmaruzz at gmail.com <mailto:gmaruzz at gmail.com> 


Your membership in the mailing list FreeSWITCH-users has been disabled
due to excessive bounces The last bounce received from you was dated
22-Feb-2018.  You will not get any more messages from this list until
you re-enable your membership.  You will receive 3 more reminders like
this before your membership in the list is deleted.

To re-enable your membership, you can simply respond to this message
(leaving the Subject: line intact), or visit the confirmation page at

    http://lists.freeswitch.org/mailman/confirm/freeswitch-users/d512ce407ddf84b1aee <http://lists.freeswitch.org/mailman/confirm/freeswitch-users/d512ce407ddf84b1aee26e09677135e861863512> 


You can also visit your membership page at

    http://lists.freeswitch.org/mailman/options/freeswitch-users/gmaruzz%40gmail.com


On your membership page, you can change various delivery options such
as your email address and whether you get digests or not.  As a
reminder, your membership password is

    xxxxx


If you have any questions or problems, you can contact the list owner
at

    freeswitch-users-owner at lists.freeswitch.org <mailto:freeswitch-users-owner at lists.freeswitch.org> 




-- 


Sincerely,

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

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From netcentrica at gmail.com  Thu Feb 22 10:17:29 2018
From: netcentrica at gmail.com (Adam Raszynski)
Date: Thu, 22 Feb 2018 11:17:29 +0100
Subject: [Freeswitch-users] How to bridge session to user in valet_park
In-Reply-To: <5512AA68-B457-4CBD-9001-AA5B2D4B141C@jerris.com>
References: <CAJWEF2bXgY7UKPSOBTMa=9G7wN6GjavOU8mcuNX14NMmwZ7JBA@mail.gmail.com>
 <5512AA68-B457-4CBD-9001-AA5B2D4B141C@jerris.com>
Message-ID: <CAJWEF2bL8JEu1uXUv3BOdsQ713AaL7FJCsgFbpbwYcppD6zApw@mail.gmail.com>

Hi

First user dials-in and waits in valet_park, all done in dialplan. Second
call leg is a kind of click-to-call, initiated over http request, so I use
lua script

I tried, but session.bridge() does not seem to accept uuid

Can you please post some example how can I bridge this user waiting in
valet_park with new call originated from lua script?

2018-02-22 0:15 GMT+01:00 Michael Jerris <mike at jerris.com>:

> why this complicated process instead of just bridging the call?
>
> > On Feb 21, 2018, at 6:28 AM, Adam Raszynski <netcentrica at gmail.com>
> wrote:
> >
> > Hi
> >
> > I have user put in valet_park waiting
> >
> > In the meantime I execute lua script to create new session to external
> number
> >
> > I have uuid of this user in local variable, script is like this:
> >
> > local uuid = "here uuid got from valet_info api call"
> > session1 = freeswitch.Session(dialstring);
> > if (session1:ready() and session1:answered()) then
> >      api:executeString("uuid_broadcast " .. uuid .. "
> say::en\\snumber\\spronounced\\s1 aleg");
> >      api:executeString("uuid_bridge " .. uuid .. " " .. session1.uuid);
> > end
> >
> > But it works sometimes. Most of time I get CS_RESET after uuid_bridge
> API call
> >
> > What I'm doing wrong here?
> >
> > Maybe there is some other method of bridging my session1 with user
> waiting in valet_park?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 gmaruzz at gmail.com  Thu Feb 22 10:45:07 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 22 Feb 2018 11:45:07 +0100
Subject: [Freeswitch-users] Fwd: confirm d512ce407ddf84b1aee26e096771
In-Reply-To: <04e901d3abbd$7f063c00$7d12b400$@freeswitch.org>
References: <CALXCt0qxD=8oRb91dAHAtQDi=VD6hAVCbenStFAMrt4d4_Ki6A@mail.gmail.com>
 <04e901d3abbd$7f063c00$7d12b400$@freeswitch.org>
Message-ID: <CALXCt0p4iYak5WXHn9Aa0gYE_v82ZArk5AWes-FFiN6Te39w7w@mail.gmail.com>

Thanks a lot, Ken!



On 22 February 2018 at 10:14, Ken Rice <krice at freeswitch.org> wrote:

> Nope this happens on google hosted accounts all the time. Mailman uses the
> SMTP protocol fully, google doesn’t allow multiple domains in the same
> transation, so if say krice at freeswitch.org and gmaruzz at gmail.com are sent
> in the same transaction even tho they have the same MX host, google will
> reject any that don’t match the first domain.
>
>
>
> Just reply to it and you’ll reset the counter.
>
>
>
> I have to do this a couple of times a month
>
> K
>
>
>
> *From:* FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] *On Behalf Of *Giovanni Maruzzelli
> *Sent:* Thursday, February 22, 2018 3:06 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Fwd: confirm d512ce407ddf84b1aee26e096771
>
>
>
>
>
> List problems again?
>
>
>
> ---------- Forwarded message ----------
> From: <freeswitch-users-request at lists.freeswitch.org>
> Date: 22 February 2018 at 01:44
> Subject: confirm d512ce407ddf84b1aee26e0967
> To: gmaruzz at gmail.com
>
>
> Your membership in the mailing list FreeSWITCH-users has been disabled
> due to excessive bounces The last bounce received from you was dated
> 22-Feb-2018.  You will not get any more messages from this list until
> you re-enable your membership.  You will receive 3 more reminders like
> this before your membership in the list is deleted.
>
> To re-enable your membership, you can simply respond to this message
> (leaving the Subject: line intact), or visit the confirmation page at
>
>     http://lists.freeswitch.org/mailman/confirm/freeswitch-
> users/d512ce407ddf84b1aee
> <http://lists.freeswitch.org/mailman/confirm/freeswitch-users/d512ce407ddf84b1aee26e09677135e861863512>
>
>
> You can also visit your membership page at
>
>     http://lists.freeswitch.org/mailman/options/freeswitch-
> users/gmaruzz%40gmail.com
>
>
> On your membership page, you can change various delivery options such
> as your email address and whether you get digests or not.  As a
> reminder, your membership password is
>
>     xxxxx
>
>
> If you have any questions or problems, you can contact the list owner
> at
>
>     freeswitch-users-owner at lists.freeswitch.org
>
>
>
>
> --
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From brian at freeswitch.com  Thu Feb 22 12:59:27 2018
From: brian at freeswitch.com (Brian West)
Date: Thu, 22 Feb 2018 06:59:27 -0600
Subject: [Freeswitch-users] Fwd: confirm d512ce407ddf84b1aee26e096771
In-Reply-To: <CALXCt0p4iYak5WXHn9Aa0gYE_v82ZArk5AWes-FFiN6Te39w7w@mail.gmail.com>
References: <CALXCt0qxD=8oRb91dAHAtQDi=VD6hAVCbenStFAMrt4d4_Ki6A@mail.gmail.com>
 <04e901d3abbd$7f063c00$7d12b400$@freeswitch.org>
 <CALXCt0p4iYak5WXHn9Aa0gYE_v82ZArk5AWes-FFiN6Te39w7w@mail.gmail.com>
Message-ID: <CAEJMVkCgM82LK+_PM73LMAc+2TpBzt8D2qgn1+rijgFkLcbt8A@mail.gmail.com>

https://support.google.com/mail/answer/2451690

This is most likely the culprit, I'm going to implement this today.

/b


On Thu, Feb 22, 2018 at 4:45 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

>
> Thanks a lot, Ken!
>
>
>
> On 22 February 2018 at 10:14, Ken Rice <krice at freeswitch.org> wrote:
>
>> Nope this happens on google hosted accounts all the time. Mailman uses
>> the SMTP protocol fully, google doesn’t allow multiple domains in the same
>> transation, so if say krice at freeswitch.org and gmaruzz at gmail.com are
>> sent in the same transaction even tho they have the same MX host, google
>> will reject any that don’t match the first domain.
>>
>>
>>
>> Just reply to it and you’ll reset the counter.
>>
>>
>>
>> I have to do this a couple of times a month
>>
>> K
>>
>>
>>
>> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc
>> es at lists.freeswitch.org] *On Behalf Of *Giovanni Maruzzelli
>> *Sent:* Thursday, February 22, 2018 3:06 AM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* [Freeswitch-users] Fwd: confirm d512ce407ddf84b1aee26e096771
>>
>>
>>
>>
>>
>> List problems again?
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: <freeswitch-users-request at lists.freeswitch.org>
>> Date: 22 February 2018 at 01:44
>> Subject: confirm d512ce407ddf84b1aee26e0967
>> To: gmaruzz at gmail.com
>>
>>
>> Your membership in the mailing list FreeSWITCH-users has been disabled
>> due to excessive bounces The last bounce received from you was dated
>> 22-Feb-2018.  You will not get any more messages from this list until
>> you re-enable your membership.  You will receive 3 more reminders like
>> this before your membership in the list is deleted.
>>
>> To re-enable your membership, you can simply respond to this message
>> (leaving the Subject: line intact), or visit the confirmation page at
>>
>>     http://lists.freeswitch.org/mailman/confirm/freeswitch-users
>> /d512ce407ddf84b1aee
>> <http://lists.freeswitch.org/mailman/confirm/freeswitch-users/d512ce407ddf84b1aee26e09677135e861863512>
>>
>>
>> You can also visit your membership page at
>>
>>     http://lists.freeswitch.org/mailman/options/freeswitch-users
>> /gmaruzz%40gmail.com
>>
>>
>> On your membership page, you can change various delivery options such
>> as your email address and whether you get digests or not.  As a
>> reminder, your membership password is
>>
>>     xxxxx
>>
>>
>> If you have any questions or problems, you can contact the list owner
>> at
>>
>>     freeswitch-users-owner at lists.freeswitch.org
>>
>>
>>
>>
>> --
>>
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From mike at jerris.com  Thu Feb 22 18:33:07 2018
From: mike at jerris.com (Michael Jerris)
Date: Thu, 22 Feb 2018 13:33:07 -0500
Subject: [Freeswitch-users] How to bridge session to user in valet_park
In-Reply-To: <CAJWEF2bL8JEu1uXUv3BOdsQ713AaL7FJCsgFbpbwYcppD6zApw@mail.gmail.com>
References: <CAJWEF2bXgY7UKPSOBTMa=9G7wN6GjavOU8mcuNX14NMmwZ7JBA@mail.gmail.com>
 <5512AA68-B457-4CBD-9001-AA5B2D4B141C@jerris.com>
 <CAJWEF2bL8JEu1uXUv3BOdsQ713AaL7FJCsgFbpbwYcppD6zApw@mail.gmail.com>
Message-ID: <2E7BF515-3152-4ED1-AD67-940B30245C8C@jerris.com>

instead of doing a different originate, just transfer the parked call to the dial plan to bridge to the destination?

> On Feb 22, 2018, at 5:17 AM, Adam Raszynski <netcentrica at gmail.com> wrote:
> 
> Hi
> 
> First user dials-in and waits in valet_park, all done in dialplan. Second call leg is a kind of click-to-call, initiated over http request, so I use lua script
> 
> I tried, but session.bridge() does not seem to accept uuid
> 
> Can you please post some example how can I bridge this user waiting in valet_park with new call originated from lua script? 
> 
> 2018-02-22 0:15 GMT+01:00 Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>:
> why this complicated process instead of just bridging the call?
> 
> > On Feb 21, 2018, at 6:28 AM, Adam Raszynski <netcentrica at gmail.com <mailto:netcentrica at gmail.com>> wrote:
> >
> > Hi
> >
> > I have user put in valet_park waiting
> >
> > In the meantime I execute lua script to create new session to external number
> >
> > I have uuid of this user in local variable, script is like this:
> >
> > local uuid = "here uuid got from valet_info api call"
> > session1 = freeswitch.Session(dialstring);
> > if (session1:ready() and session1:answered()) then
> >      api:executeString("uuid_broadcast " .. uuid .. " say::en\\snumber\\spronounced\\s1 aleg");
> >      api:executeString("uuid_bridge " .. uuid .. " " .. session1.uuid);
> > end
> >
> > But it works sometimes. Most of time I get CS_RESET after uuid_bridge API call
> >
> > What I'm doing wrong here?
> >
> > Maybe there is some other method of bridging my session1 with user waiting in valet_park?

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From t90fpe at outlook.com  Thu Feb 22 16:33:59 2018
From: t90fpe at outlook.com (Fred Pettersson)
Date: Thu, 22 Feb 2018 16:33:59 +0000
Subject: [Freeswitch-users] My 2 cents - online community
Message-ID: <VI1PR0902MB2143D7DDD51D903F6D9BCD33D2CD0@VI1PR0902MB2143.eurprd09.prod.outlook.com>

Look I don't wanna start a fire here and have full respect for the team behind FreeSWITCH.

It's 2018 right? I am talking about Evolution. Why doesn't the FreeSWITCH community have a more up to date online community board? I am talking about the mailing lists, why not use a tool to help the community a bit more and make it grow?

Today I think it's hard to get a view of all the questions/answers and the mailing list archive is not that user friendly. If people more easy can search/browse for answers then less questions will be generated. I don't see a big and viable community as FreeSWITCH deserves IMO and I am certain of getting a online community board will increase the community in that sense. This is one of the first things I look for when getting a new component or trying to find help. I am still a beginner of FreeSWITCH even though I've followed the project for years and I know a community board would help me and most likely others.

It's easy to scope/categorize the content of a community board and to allow certain discussions of possible configuration errors vs bugs etc, as a novice like my self I have a hard time to spot this and when I read answers like "bugs are not to discussed on the list - file a Jira". If you, like me, don't even know if this is a mistake by myself and get this kind of answer I really hesitate to ask.

All I want is the best for FreeSWITCH. What do you think?

/Fred

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From mike at jerris.com  Thu Feb 22 19:24:39 2018
From: mike at jerris.com (Michael Jerris)
Date: Thu, 22 Feb 2018 14:24:39 -0500
Subject: [Freeswitch-users] My 2 cents - online community
In-Reply-To: <VI1PR0902MB2143D7DDD51D903F6D9BCD33D2CD0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
References: <VI1PR0902MB2143D7DDD51D903F6D9BCD33D2CD0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
Message-ID: <3EE95868-81A5-45C9-AFB4-6396E0E783FE@jerris.com>

The plan is to migrate to Mailman 3 when its ready, but the last few passes at it, they were not ready to migrate from older installs.  If thats ready, we can move that up the infrastructure priority list.  If anyone wants to do the research on if thats ready, i’d be happy to hear the findings.


> On Feb 22, 2018, at 11:33 AM, Fred Pettersson <t90fpe at outlook.com> wrote:
> 
> Look I don't wanna start a fire here and have full respect for the team behind FreeSWITCH. 
> 
> It's 2018 right? I am talking about Evolution. Why doesn't the FreeSWITCH community have a more up to date online community board? I am talking about the mailing lists, why not use a tool to help the community a bit more and make it grow?
> 
> Today I think it's hard to get a view of all the questions/answers and the mailing list archive is not that user friendly. If people more easy can search/browse for answers then less questions will be generated. I don't see a big and viable community as FreeSWITCH deserves IMO and I am certain of getting a online community board will increase the community in that sense. This is one of the first things I look for when getting a new component or trying to find help. I am still a beginner of FreeSWITCH even though I've followed the project for years and I know a community board would help me and most likely others. 
> 
> It's easy to scope/categorize the content of a community board and to allow certain discussions of possible configuration errors vs bugs etc, as a novice like my self I have a hard time to spot this and when I read answers like "bugs are not to discussed on the list - file a Jira". If you, like me, don't even know if this is a mistake by myself and get this kind of answer I really hesitate to ask.
> 
> All I want is the best for FreeSWITCH. What do you think?   
> 
> /Fred

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From fs at szmidt.org  Thu Feb 22 19:31:46 2018
From: fs at szmidt.org (fs)
Date: Thu, 22 Feb 2018 14:31:46 -0500
Subject: [Freeswitch-users] My 2 cents - online community
In-Reply-To: <VI1PR0902MB2143D7DDD51D903F6D9BCD33D2CD0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
References: <VI1PR0902MB2143D7DDD51D903F6D9BCD33D2CD0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
Message-ID: <37d1d669-e96f-67f7-7713-2910e57e6325@szmidt.org>

On 02/22/2018 11:33, Fred Pettersson wrote:
> Look I don't wanna start a fire here and have full respect for the team
> behind FreeSWITCH.
> 
> It's 2018 right? I am talking about Evolution. Why doesn't the
> FreeSWITCH community have a more up to date online community board? I am
> talking about the mailing lists, why not use a tool to help the
> community a bit more and make it grow?
As a user who's not delved very deep into it myself, and someone who
truly appreciates the fact that FS is properly designed and developed,
which is why _I_ use it rather than *, but as such I never end up
spending much time consecutive time on it, and in this long running
sentence, will always appreciate things that makes it easy for a novice
to get started.

Open source software, often driven by individual interests, certainly
can have issues with getting people past the initial install. FS for
example, have some excellent videos to help you get started! Then if you
want to know more they would not mind, I'm sure, if you paid them for
support so they can have some income.

I've walked down the path of offering Open Source s/w and charged for
support etc. It can be profitable, but it can also be a pain in the butt
with not enough cash flow to become really viable.

Often times someone will suggest some improvement, as you have, and the
reply can easily be "Great, do it!". I'm not suggesting that I'm talking
for this group in the least, but I have seen it first hand and know how
it can be.


---
fs

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From mail at paulzillmann.de  Thu Feb 22 19:51:19 2018
From: mail at paulzillmann.de (Paul Zillmann)
Date: Thu, 22 Feb 2018 20:51:19 +0100
Subject: [Freeswitch-users] My 2 cents - online community
In-Reply-To: <VI1PR0902MB2143D7DDD51D903F6D9BCD33D2CD0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
References: <VI1PR0902MB2143D7DDD51D903F6D9BCD33D2CD0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
Message-ID: <785d549a-526d-4306-dee2-780946c1c0de@paulzillmann.de>

Hello Fred,


I think Google provides a good way of querying the mailing-list, given 
that the subject-line is good and the content isn't just a random 
"Hiiiiiiiii - how to solve command"

Many projects in the unix-world including operating systems rely heavily 
on mailing-list communication. And that one works fine.

Could you state more precise what you mean by an online community board? 
In my understanding that would be a forum.


Some users aren't able to use the search functions in such forums either.

By now I can't see a benefit of an "online community board".


Besides the mailing-list you may participate in the weekly conference 
call and ClueCon weekly.

Aside from that: don't be afraid to ask on the list and try to apply the 
basic rules you'll find in many online forums: good titles, be 
respectful, and put some effort into your text.


Sincerely,

Paul


Am 22.02.2018 um 17:33 schrieb Fred Pettersson:
> Look I don't wanna start a fire here and have full respect for the 
> team behind FreeSWITCH.
>
> It's 2018 right? I am talking about Evolution. Why doesn't the 
> FreeSWITCH community have a more up to date online community board? I 
> am talking about the mailing lists, why not use a tool to help the 
> community a bit more and make it grow?
>
> Today I think it's hard to get a view of all the questions/answers and 
> the mailing list archive is not that user friendly. If people more 
> easy can search/browse for answers then less questions will be 
> generated. I don't see a big and viable community as FreeSWITCH 
> deserves IMO and I am certain of getting a online community board will 
> increase the community in that sense. This is one of the first things 
> I look for when getting a new component or trying to find help. I am 
> still a beginner of FreeSWITCH even though I've followed the project 
> for years and I know a community board would help me and most likely 
> others.
>
> It's easy to scope/categorize the content of a community board and to 
> allow certain discussions of possible configuration errors vs bugs 
> etc, as a novice like my self I have a hard time to spot this and when 
> I read answers like "bugs are not to discussed on the list - file a 
> Jira". If you, like me, don't even know if this is a mistake by myself 
> and get this kind of answer I really hesitate to ask.
>
> All I want is the best for FreeSWITCH. What do you think?
>
> /Fred
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 krice at tollfreegateway.com  Thu Feb 22 20:35:01 2018
From: krice at tollfreegateway.com (krice at tollfreegateway.com)
Date: Thu, 22 Feb 2018 14:35:01 -0600
Subject: [Freeswitch-users] My 2 cents - online community
In-Reply-To: <785d549a-526d-4306-dee2-780946c1c0de@paulzillmann.de>
References: <VI1PR0902MB2143D7DDD51D903F6D9BCD33D2CD0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
 <785d549a-526d-4306-dee2-780946c1c0de@paulzillmann.de>
Message-ID: <099401d3ac1c$9d33b630$d79b2290$@tollfreegateway.com>

As Paul just said, lists.freeswitch.org contains a complete archive of the
mailing list. This site is heavily indexed by google.

 

A hint for searching it is when you google use the
"site:lists.freeswitch.org" or "site:freeswitch.org" to restrict your google
queries. This should return anything in the archive relevant to your actual
query string.

 

Have fun!

K

 

From: FreeSWITCH-users
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul
Zillmann
Sent: Thursday, February 22, 2018 1:51 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] My 2 cents - online community

 

Hello Fred,

 

I think Google provides a good way of querying the mailing-list, given that
the subject-line is good and the content isn't just a random "Hiiiiiiiii -
how to solve command"

Many projects in the unix-world including operating systems rely heavily on
mailing-list communication. And that one works fine.

Could you state more precise what you mean by an online community board? In
my understanding that would be a forum.

 

Some users aren't able to use the search functions in such forums either.

By now I can't see a benefit of an "online community board".

 

Besides the mailing-list you may participate in the weekly conference call
and ClueCon weekly.

Aside from that: don't be afraid to ask on the list and try to apply the
basic rules you'll find in many online forums: good titles, be respectful,
and put some effort into your text.

 

Sincerely,

Paul

 

Am 22.02.2018 um 17:33 schrieb Fred Pettersson:

Look I don't wanna start a fire here and have full respect for the team
behind FreeSWITCH. 

It's 2018 right? I am talking about Evolution. Why doesn't the FreeSWITCH
community have a more up to date online community board? I am talking about
the mailing lists, why not use a tool to help the community a bit more and
make it grow?

Today I think it's hard to get a view of all the questions/answers and the
mailing list archive is not that user friendly. If people more easy can
search/browse for answers then less questions will be generated. I don't see
a big and viable community as FreeSWITCH deserves IMO and I am certain of
getting a online community board will increase the community in that sense.
This is one of the first things I look for when getting a new component or
trying to find help. I am still a beginner of FreeSWITCH even though I've
followed the project for years and I know a community board would help me
and most likely others. 

It's easy to scope/categorize the content of a community board and to allow
certain discussions of possible configuration errors vs bugs etc, as a
novice like my self I have a hard time to spot this and when I read answers
like "bugs are not to discussed on the list - file a Jira". If you, like me,
don't even know if this is a mistake by myself and get this kind of answer I
really hesitate to ask.

All I want is the best for FreeSWITCH. What do you think?   

/Fred

 






_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com
 
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org

 

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From pierre at couderc.eu  Thu Feb 22 21:49:35 2018
From: pierre at couderc.eu (nous)
Date: Thu, 22 Feb 2018 22:49:35 +0100
Subject: [Freeswitch-users] freze when installing Freeswitch
In-Reply-To: <017CF0D9-5321-430F-BA0D-087E3B80E866@jerris.com>
References: <da62c7c9-d75f-0fd2-1c86-a69ceb96d02e@couderc.eu>
 <017CF0D9-5321-430F-BA0D-087E3B80E866@jerris.com>
Message-ID: <4c5b101d-491e-74ee-ba9a-57c3b2dc1797@couderc.eu>

Thank you very much.


On 22/02/2018 00:07, Michael Jerris wrote:
> this is something blocking in a non freeswitch package postinstall hooks.  You’d have to research issues installing policykit.  Its a dep of some lib, i don’t know about it off hand and haven’t seen this issue before.
>
>
>> On Feb 20, 2018, at 2:07 AM, Pierre Couderc <pierre at couderc.eu> wrote:
>>
>> I am trying to install my first Freeswitch nearly in debian 8 :
>>
>> I have followed https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie :
>>
>> wget -O - https://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | apt-key add -
>> echo "deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie main" > /etc/apt/sources.list.d/freeswitch.list
>> apt-get update && apt-get install -y freeswitch-meta-all
>>
>> It starts well but does not give back the prompt :
>>
>> Setting up tzdata-java (2017c-0+deb8u1) ...
>> Setting up sgml-base (1.26+nmu4) ...
>> Setting up vdpau-va-driver:amd64 (0.7.4-3) ...
>> Setting up libtxc-dxtn-s2tc0:amd64 (0~git20131104-1.1) ...
>> update-alternatives: using /usr/lib/x86_64-linux-gnu/libtxc_dxtn_s2tc.so.0 to provide /usr/lib/x86_64-linux-gnu/libtxc_dxtn.so (libtxc-dxtn-x86_64-linux-gnu) in auto mode
>> Setting up file (1:5.22+15-2+deb8u3) ...
>> Setting up at-spi2-core (2.14.0-1) ...
>> Setting up policykit-1 (0.105-15~deb8u2) ...
>> Removed symlink /run/systemd/system/polkitd.service.
>>
>> ---> never returns the prompt here.
>>
>>
>> if I try on another console :
>>
>> root at test:~# fs_cli -rRS
>> [ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error]
>> [INFO] fs_cli.c:1665 main() Retrying
>> [ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error]
>> [INFO] fs_cli.c:1665 main() Retrying
>> ...
>>
>> Please note that my debian 8 is a LXD container inside a debian 9 host.
>>
>> My question is : what should I do ? How to retry ?
>>
>>
>> Pierre Couderc
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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



From pierre at couderc.eu  Thu Feb 22 21:51:08 2018
From: pierre at couderc.eu (nous)
Date: Thu, 22 Feb 2018 22:51:08 +0100
Subject: [Freeswitch-users] Fwd: confirm d512ce407ddf84b1aee26e096771
In-Reply-To: <CALXCt0qxD=8oRb91dAHAtQDi=VD6hAVCbenStFAMrt4d4_Ki6A@mail.gmail.com>
References: <CALXCt0qxD=8oRb91dAHAtQDi=VD6hAVCbenStFAMrt4d4_Ki6A@mail.gmail.com>
Message-ID: <47081f1e-7614-07a2-8323-b8dfd90b1c6d@couderc.eu>

It seems ok now. Thank you.


On 22/02/2018 10:05, Giovanni Maruzzelli wrote:
>
> List problems again?
>
>
> ---------- Forwarded message ----------
> From: <freeswitch-users-request at lists.freeswitch.org 
> <mailto:freeswitch-users-request at lists.freeswitch.org>>
> Date: 22 February 2018 at 01:44
> Subject: confirm d512ce407ddf84b1aee26e0967
> To: gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>
>
>
> Your membership in the mailing list FreeSWITCH-users has been disabled
> due to excessive bounces The last bounce received from you was dated
> 22-Feb-2018.  You will not get any more messages from this list until
> you re-enable your membership.  You will receive 3 more reminders like
> this before your membership in the list is deleted.
>
> To re-enable your membership, you can simply respond to this message
> (leaving the Subject: line intact), or visit the confirmation page at
>
> http://lists.freeswitch.org/mailman/confirm/freeswitch-users/d512ce407ddf84b1aee 
> <http://lists.freeswitch.org/mailman/confirm/freeswitch-users/d512ce407ddf84b1aee26e09677135e861863512>
>
>
> You can also visit your membership page at
>
> http://lists.freeswitch.org/mailman/options/freeswitch-users/gmaruzz%40gmail.com 
> <http://lists.freeswitch.org/mailman/options/freeswitch-users/gmaruzz%40gmail.com>
>
>
> On your membership page, you can change various delivery options such
> as your email address and whether you get digests or not. As a
> reminder, your membership password is
>
>     xxxxx
>
> If you have any questions or problems, you can contact the list owner
> at
>
> freeswitch-users-owner at lists.freeswitch.org 
> <mailto:freeswitch-users-owner at lists.freeswitch.org>
>
>
>
> -- 
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 b631093f-779b-4d67-9ffe-5f6d5b1d3f8a at protonmail.ch  Thu Feb 22 20:15:56 2018
From: b631093f-779b-4d67-9ffe-5f6d5b1d3f8a at protonmail.ch (Bob Smith)
Date: Thu, 22 Feb 2018 15:15:56 -0500
Subject: [Freeswitch-users] Lua question (re: FollowMe Dialplan Example from
	the docs)
Message-ID: <lL7Qbo1q4G4E_xf2PoXnRUk2-JQ-sQ5xRdeP2osSAOBxI-d6pfcuxaSMrNMX35k4Kq14-CwZ0i9JjmNsqiIp-8h3Ib5cTRVRtzN1jGxuzHc=@protonmail.ch>

Hello,

The docs (https://freeswitch.org/confluence/display/FREESWITCH/Dialplan+FollowMe) have an interesting example, the snappily titled "FollowMe Dialplan Example 6".

This looks interesting. But I have one concern, and being a Lua newbie, I don't know how to resolve it.

My reading of the script is that it does not take care of the somewhat obvious problem of what happens if the cell phone goes to voicemail ?

My understanding of the script as presented is that it would just sit there forever waiting for a reply ?

Could a resident Lua guru kindly enlighten this newbie as to how I could timeout the session ?
Thanks !

Bob
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From gmaruzz at gmail.com  Fri Feb 23 08:26:23 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 23 Feb 2018 09:26:23 +0100
Subject: [Freeswitch-users] My 2 cents - online community
In-Reply-To: <099401d3ac1c$9d33b630$d79b2290$@tollfreegateway.com>
References: <VI1PR0902MB2143D7DDD51D903F6D9BCD33D2CD0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
 <785d549a-526d-4306-dee2-780946c1c0de@paulzillmann.de>
 <099401d3ac1c$9d33b630$d79b2290$@tollfreegateway.com>
Message-ID: <CALXCt0qGuU2v9MP=D9+9yDk1fROZ91BN6vDQpO3rsbOHpro1dA@mail.gmail.com>

also, don't forget there are 3 FreeSWITCH books out there, each of them
available in ebook, pdf and printed paper format...

If you can't buy them, you can at least ask a friend in the know to provide
them to you.

-giovanni


On 22 February 2018 at 21:35, <krice at tollfreegateway.com> wrote:

> As Paul just said, lists.freeswitch.org contains a complete archive of
> the mailing list. This site is heavily indexed by google.
>
>
>
> A hint for searching it is when you google use the “site:
> lists.freeswitch.org” or “site:freeswitch.org” to restrict your google
> queries. This should return anything in the archive relevant to your actual
> query string.
>
>
>
> Have fun!
>
> K
>
>
>
> *From:* FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] *On Behalf Of *Paul Zillmann
> *Sent:* Thursday, February 22, 2018 1:51 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] My 2 cents - online community
>
>
>
> Hello Fred,
>
>
>
> I think Google provides a good way of querying the mailing-list, given
> that the subject-line is good and the content isn't just a random
> "Hiiiiiiiii - how to solve command"
>
> Many projects in the unix-world including operating systems rely heavily
> on mailing-list communication. And that one works fine.
>
> Could you state more precise what you mean by an online community board?
> In my understanding that would be a forum.
>
>
>
> Some users aren't able to use the search functions in such forums either.
>
> By now I can't see a benefit of an "online community board".
>
>
>
> Besides the mailing-list you may participate in the weekly conference call
> and ClueCon weekly.
>
> Aside from that: don't be afraid to ask on the list and try to apply the
> basic rules you'll find in many online forums: good titles, be respectful,
> and put some effort into your text.
>
>
>
> Sincerely,
>
> Paul
>
>
>
> Am 22.02.2018 um 17:33 schrieb Fred Pettersson:
>
> Look I don't wanna start a fire here and have full respect for the team
> behind FreeSWITCH.
>
> It's 2018 right? I am talking about Evolution. Why doesn't the FreeSWITCH
> community have a more up to date online community board? I am talking about
> the mailing lists, why not use a tool to help the community a bit more and
> make it grow?
>
> Today I think it's hard to get a view of all the questions/answers and the
> mailing list archive is not that user friendly. If people more easy can
> search/browse for answers then less questions will be generated. I don't
> see a big and viable community as FreeSWITCH deserves IMO and I am certain
> of getting a online community board will increase the community in that
> sense. This is one of the first things I look for when getting a new
> component or trying to find help. I am still a beginner of FreeSWITCH even
> though I've followed the project for years and I know a community board
> would help me and most likely others.
>
> It's easy to scope/categorize the content of a community board and to
> allow certain discussions of possible configuration errors vs bugs etc, as
> a novice like my self I have a hard time to spot this and when I read
> answers like "bugs are not to discussed on the list - file a Jira". If you,
> like me, don't even know if this is a mistake by myself and get this kind
> of answer I really hesitate to ask.
>
> All I want is the best for FreeSWITCH. What do you think?
>
> /Fred
>
>
>
>
>
>
> _________________________________________________________________________
>
> Professional FreeSWITCH Consulting Services:
>
> consulting at freeswitch.org
>
> http://www.freeswitchsolutions.com
>
>
>
> Official FreeSWITCH Sites
>
> http://www.freeswitch.org
>
> http://confluence.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From larry.hemenway at gmail.com  Fri Feb 23 14:48:30 2018
From: larry.hemenway at gmail.com (Larry Hemenway)
Date: Fri, 23 Feb 2018 08:48:30 -0600
Subject: [Freeswitch-users] Receiving audio despite SDP with a=sendonly
In-Reply-To: <CAN=Q5SBZ9iyR1HFFxK0GycJFSC=u5aW5=N_bgXV5tsq39DCsvA@mail.gmail.com>
References: <CAN=Q5SBZ9iyR1HFFxK0GycJFSC=u5aW5=N_bgXV5tsq39DCsvA@mail.gmail.com>
Message-ID: <CAN=Q5SB8SJ+dpn930PPrjc5H--mZRoOHpx8-m9=B_DztsycyUQ@mail.gmail.com>

It looks like FreeSWITCH does not support one-way audio at all. We have a
strong need for this functionality if we choose to use FreeSWITCH. If I
were to investigate and implement this, is it something that the community
would be interested in? I'm a bit worried about diving into this because if
FreeSWITCH honors the sdp direction it could potentially break some legacy
applications that rely on current functionality.

Also, any tips on where to start would be appreciated.

More info on my investigation:

I got a little smarter in how I search the mailing list and came across
this, so it sounds like this is not supported functionality.

http://freeswitch-users.2379917.n2.nabble.com/sendonly-attribute-ignored-td5933886.html

I also ran some more experiments. The reinvite with a=sendonly results in a
hold event, so I verified audio is not passed in either direction despite
the SDP direction attribute.

I'm currently doing these experiments with a conference. We would like to
use it as a sort of intercom function, but our security group is strongly
discouraging us from making a product that sends audio to all the endpoints
even if its silence, so the conference relate API, from their perspective,
isn't a solution.

Larry



On Wed, Feb 21, 2018 at 5:09 PM, Larry Hemenway <larry.hemenway at gmail.com>
wrote:

> Hello,
>
> Is there a way to establish a call with one-way audio on a call from the
> start?
>
> I'm currently sending a request with the following SDP - (note a=sendonly):
>
> v=0
> o=Larry 2890844526 2890844526 IN IP4 127.0.0.1
> s= My Session
> c=IN IP4 172.22.112.1
> t=0 0
> m=audio 49170 RTP/AVP 0 8
> a=sendonly
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
>
> FreeSWITCH is sending back the right response (note the a=recvonly):
>
> v=0
> o=FreeSWITCH 1519171720 1519171721 IN IP4 172.17.0.2
> s=FreeSWITCH
> c=IN IP4 172.17.0.2
> t=0 0
> m=audio 21768 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=recvonly
> a=ptime:20
>
> But I'm still receiving audio from FreeSWITCH. It is only after I
> resend the SDP via a reinvite that the audio stops being sent from
> FreeSWITCH.
>
> Larry
>
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From hawkins at hawkinsegroup.com  Fri Feb 23 16:23:39 2018
From: hawkins at hawkinsegroup.com (Don Hawkins)
Date: Fri, 23 Feb 2018 10:23:39 -0600
Subject: [Freeswitch-users] Receiving audio despite SDP with a=sendonly
In-Reply-To: <CAN=Q5SB8SJ+dpn930PPrjc5H--mZRoOHpx8-m9=B_DztsycyUQ@mail.gmail.com>
References: <CAN=Q5SBZ9iyR1HFFxK0GycJFSC=u5aW5=N_bgXV5tsq39DCsvA@mail.gmail.com>
 <CAN=Q5SB8SJ+dpn930PPrjc5H--mZRoOHpx8-m9=B_DztsycyUQ@mail.gmail.com>
Message-ID: <CAO6gFMxn75=c1-XhcGqvZFxEmCzXpO5oys7YnYTg8xCHn4Zujw@mail.gmail.com>

Without knowing all the details of your use case, my suggestion is to put
the callers in a conference room and mute one member until desired.


Sent from my NationPCS Galaxy Note 5

On Feb 23, 2018 10:14 AM, "Larry Hemenway" <larry.hemenway at gmail.com> wrote:

> It looks like FreeSWITCH does not support one-way audio at all. We have a
> strong need for this functionality if we choose to use FreeSWITCH. If I
> were to investigate and implement this, is it something that the community
> would be interested in? I'm a bit worried about diving into this because if
> FreeSWITCH honors the sdp direction it could potentially break some legacy
> applications that rely on current functionality.
>
> Also, any tips on where to start would be appreciated.
>
> More info on my investigation:
>
> I got a little smarter in how I search the mailing list and came across
> this, so it sounds like this is not supported functionality.
>
> http://freeswitch-users.2379917.n2.nabble.com/sendonly-attribute-ignored-
> td5933886.html
>
> I also ran some more experiments. The reinvite with a=sendonly results in
> a hold event, so I verified audio is not passed in either direction despite
> the SDP direction attribute.
>
> I'm currently doing these experiments with a conference. We would like to
> use it as a sort of intercom function, but our security group is strongly
> discouraging us from making a product that sends audio to all the endpoints
> even if its silence, so the conference relate API, from their perspective,
> isn't a solution.
>
> Larry
>
>
>
> On Wed, Feb 21, 2018 at 5:09 PM, Larry Hemenway <larry.hemenway at gmail.com>
> wrote:
>
>> Hello,
>>
>> Is there a way to establish a call with one-way audio on a call from the
>> start?
>>
>> I'm currently sending a request with the following SDP - (note
>> a=sendonly):
>>
>> v=0
>> o=Larry 2890844526 2890844526 IN IP4 127.0.0.1
>> s= My Session
>> c=IN IP4 172.22.112.1
>> t=0 0
>> m=audio 49170 RTP/AVP 0 8
>> a=sendonly
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>>
>> FreeSWITCH is sending back the right response (note the a=recvonly):
>>
>> v=0
>> o=FreeSWITCH 1519171720 1519171721 IN IP4 172.17.0.2
>> s=FreeSWITCH
>> c=IN IP4 172.17.0.2
>> t=0 0
>> m=audio 21768 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=recvonly
>> a=ptime:20
>>
>> But I'm still receiving audio from FreeSWITCH. It is only after I
>> resend the SDP via a reinvite that the audio stops being sent from
>> FreeSWITCH.
>>
>> Larry
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 t90fpe at outlook.com  Fri Feb 23 18:23:33 2018
From: t90fpe at outlook.com (Fred Pettersson)
Date: Fri, 23 Feb 2018 18:23:33 +0000
Subject: [Freeswitch-users] My 2 cents - online community
In-Reply-To: <CALXCt0qGuU2v9MP=D9+9yDk1fROZ91BN6vDQpO3rsbOHpro1dA@mail.gmail.com>
References: <VI1PR0902MB2143D7DDD51D903F6D9BCD33D2CD0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
 <785d549a-526d-4306-dee2-780946c1c0de@paulzillmann.de>
 <099401d3ac1c$9d33b630$d79b2290$@tollfreegateway.com>,
 <CALXCt0qGuU2v9MP=D9+9yDk1fROZ91BN6vDQpO3rsbOHpro1dA@mail.gmail.com>
Message-ID: <VI1PR0902MB21437D62F6CEC1DFA88750F4D2CC0@VI1PR0902MB2143.eurprd09.prod.outlook.com>

Thank you all for feedback. Remember this was just my 2-cents. I just want to highlight that I am not saying mailing-list communication is not working but there are better options IMO.

Just to clarify - I am talking about an online forum instead of the mailing list. To justify the old way just because some user can't use the search in an online forum is just for the wrong reason. How many can't use them today or even know it's there today. It's a shopping window. I know about the FS books, I've bought three of them (paper). They are good but they do not cover everything. I know about Confluence, ClueCon Weekly, I use YouTube to watch them etc. To be fair I think you are missing my point here :) I think there is room for a good place to have discussions with others in an easy to use online tool (forum could serve that).

You know what's most "scary" about the whole thing? That's non of you does not even consider or elaborates about a change of this but tries to guide me how/were instead but I am talking about the community. Where is the spirit? :)

As I said I don't want to start a fire here, I want the best for FreeSWITCH and make the community to grow. I honestly don't think mailman is the most efficient tool in terms of growing the community. I rest my case I don't want to start a big thing.

Regards,
Fred


________________________________
Från: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> för Giovanni Maruzzelli <gmaruzz at gmail.com>
Skickat: den 23 februari 2018 09:26:23
Till: FreeSWITCH Users Help
Ämne: Re: [Freeswitch-users] My 2 cents - online community

also, don't forget there are 3 FreeSWITCH books out there, each of them available in ebook, pdf and printed paper format...

If you can't buy them, you can at least ask a friend in the know to provide them to you.

-giovanni


On 22 February 2018 at 21:35, <krice at tollfreegateway.com<mailto:krice at tollfreegateway.com>> wrote:

As Paul just said, lists.freeswitch.org<http://lists.freeswitch.org> contains a complete archive of the mailing list. This site is heavily indexed by google.



A hint for searching it is when you google use the “site:lists.freeswitch.org<http://lists.freeswitch.org>” or “site:freeswitch.org<http://freeswitch.org>” to restrict your google queries. This should return anything in the archive relevant to your actual query string.



Have fun!

K



From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Paul Zillmann
Sent: Thursday, February 22, 2018 1:51 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] My 2 cents - online community



Hello Fred,



I think Google provides a good way of querying the mailing-list, given that the subject-line is good and the content isn't just a random "Hiiiiiiiii - how to solve command"

Many projects in the unix-world including operating systems rely heavily on mailing-list communication. And that one works fine.

Could you state more precise what you mean by an online community board? In my understanding that would be a forum.



Some users aren't able to use the search functions in such forums either.

By now I can't see a benefit of an "online community board".



Besides the mailing-list you may participate in the weekly conference call and ClueCon weekly.

Aside from that: don't be afraid to ask on the list and try to apply the basic rules you'll find in many online forums: good titles, be respectful, and put some effort into your text.



Sincerely,

Paul



Am 22.02.2018 um 17:33 schrieb Fred Pettersson:

Look I don't wanna start a fire here and have full respect for the team behind FreeSWITCH.

It's 2018 right? I am talking about Evolution. Why doesn't the FreeSWITCH community have a more up to date online community board? I am talking about the mailing lists, why not use a tool to help the community a bit more and make it grow?

Today I think it's hard to get a view of all the questions/answers and the mailing list archive is not that user friendly. If people more easy can search/browse for answers then less questions will be generated. I don't see a big and viable community as FreeSWITCH deserves IMO and I am certain of getting a online community board will increase the community in that sense. This is one of the first things I look for when getting a new component or trying to find help. I am still a beginner of FreeSWITCH even though I've followed the project for years and I know a community board would help me and most likely others.

It's easy to scope/categorize the content of a community board and to allow certain discussions of possible configuration errors vs bugs etc, as a novice like my self I have a hard time to spot this and when I read answers like "bugs are not to discussed on the list - file a Jira". If you, like me, don't even know if this is a mistake by myself and get this kind of answer I really hesitate to ask.

All I want is the best for FreeSWITCH. What do you think?

/Fred






_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org<mailto:consulting at freeswitch.org>

http://www.freeswitchsolutions.com



Official FreeSWITCH Sites

http://www.freeswitch.org

http://confluence.freeswitch.org

http://www.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

http://www.freeswitch.org



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org



--

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From keith at rhizomatica.org  Fri Feb 23 13:03:37 2018
From: keith at rhizomatica.org (Keith Whyte)
Date: Fri, 23 Feb 2018 14:03:37 +0100
Subject: [Freeswitch-users] FS 1.6.20 console tab completion missing?
Message-ID: <388c74a1-d994-e20e-c53a-43bf1ac06651@rhizomatica.org>

Seems like a minor thing, but I didn't find any reference on the mailing
list:


After starting up freeswitch, I don't get tab completion for the console
command itself in fs_cli, UNTIL I do reload mod_console.
Do other people have the same thing? I notice it because I'm used to
doing the like of console loglevel x..

This is FS 1.6.20-37-987c9b9~64bit installed from files.freeswitch.org
packages.

It is Debian 9.0... installed by doing a manual `dpkg -i
libssl1.0.0_1.0.1t-1+deb8u7_amd64.deb` (could be relevant,I guess)

Thanks!!!

K/







From mike at jerris.com  Fri Feb 23 19:27:02 2018
From: mike at jerris.com (Michael Jerris)
Date: Fri, 23 Feb 2018 14:27:02 -0500
Subject: [Freeswitch-users] My 2 cents - online community
In-Reply-To: <VI1PR0902MB21437D62F6CEC1DFA88750F4D2CC0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
References: <VI1PR0902MB2143D7DDD51D903F6D9BCD33D2CD0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
 <785d549a-526d-4306-dee2-780946c1c0de@paulzillmann.de>
 <099401d3ac1c$9d33b630$d79b2290$@tollfreegateway.com>
 <CALXCt0qGuU2v9MP=D9+9yDk1fROZ91BN6vDQpO3rsbOHpro1dA@mail.gmail.com>
 <VI1PR0902MB21437D62F6CEC1DFA88750F4D2CC0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
Message-ID: <45DED6CD-40F7-4F49-A1E6-668E256E98DE@jerris.com>

You are not starting a fire at all.  I understood you to be looking at wanting some better sort of forum.  Mailman 3 as i said is the planned solution to this as it does a decent job combining a forum and mailing list together into one concept, in a way where we don’t have to monitor two different things separately and you get the best exposure.  I totally understood what you were proposing, was letting you know what the blockers to moving forward on this are, and putting it out there a way that you could help the community by moving this process forward.  Please jump in and assist in moving this process forward if you are willing and able.  Thanks as always we appreciate the feedback about how we can foster a better community.

Mike


> On Feb 23, 2018, at 1:23 PM, Fred Pettersson <t90fpe at outlook.com> wrote:
> 
> Thank you all for feedback. Remember this was just my 2-cents. I just want to highlight that I am not saying mailing-list communication is not working but there are better options IMO.
>  
> Just to clarify - I am talking about an online forum instead of the mailing list. To justify the old way just because some user can't use the search in an online forum is just for the wrong reason. How many can't use them today or even know it's there today. It's a shopping window. I know about the FS books, I've bought three of them (paper). They are good but they do not cover everything. I know about Confluence, ClueCon Weekly, I use YouTube to watch them etc. To be fair I think you are missing my point here :) I think there is room for a good place to have discussions with others in an easy to use online tool (forum could serve that).
> 
> You know what's most "scary" about the whole thing? That's non of you does not even consider or elaborates about a change of this but tries to guide me how/were instead but I am talking about the community. Where is the spirit? :)
> 
> As I said I don't want to start a fire here, I want the best for FreeSWITCH and make the community to grow. I honestly don't think mailman is the most efficient tool in terms of growing the community. I rest my case I don't want to start a big thing.
> 
> Regards,
> Fred
> 
> Från: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>> för Giovanni Maruzzelli <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>>
> Skickat: den 23 februari 2018 09:26:23
> Till: FreeSWITCH Users Help
> Ämne: Re: [Freeswitch-users] My 2 cents - online community
>  
> also, don't forget there are 3 FreeSWITCH books out there, each of them available in ebook, pdf and printed paper format...
> 
> If you can't buy them, you can at least ask a friend in the know to provide them to you.
> 
> -giovanni
> 
> 
> On 22 February 2018 at 21:35, <krice at tollfreegateway.com <mailto:krice at tollfreegateway.com>> wrote:
> As Paul just said, lists.freeswitch.org <http://lists.freeswitch.org/> contains a complete archive of the mailing list. This site is heavily indexed by google.
>  
> A hint for searching it is when you google use the “site:lists.freeswitch.org <http://lists.freeswitch.org/>” or “site:freeswitch.org <http://freeswitch.org/>” to restrict your google queries. This should return anything in the archive relevant to your actual query string.
>  
> Have fun!
> K
>  
> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Paul Zillmann
> Sent: Thursday, February 22, 2018 1:51 PM
> To: freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] My 2 cents - online community
>  
> Hello Fred,
>  
> I think Google provides a good way of querying the mailing-list, given that the subject-line is good and the content isn't just a random "Hiiiiiiiii - how to solve command"
> Many projects in the unix-world including operating systems rely heavily on mailing-list communication. And that one works fine.
> Could you state more precise what you mean by an online community board? In my understanding that would be a forum.
>  
> Some users aren't able to use the search functions in such forums either.
> By now I can't see a benefit of an "online community board".
>  
> Besides the mailing-list you may participate in the weekly conference call and ClueCon weekly.
> Aside from that: don't be afraid to ask on the list and try to apply the basic rules you'll find in many online forums: good titles, be respectful, and put some effort into your text.
>  
> Sincerely,
> Paul
>  
> Am 22.02.2018 um 17:33 schrieb Fred Pettersson:
> Look I don't wanna start a fire here and have full respect for the team behind FreeSWITCH. 
> 
> It's 2018 right? I am talking about Evolution. Why doesn't the FreeSWITCH community have a more up to date online community board? I am talking about the mailing lists, why not use a tool to help the community a bit more and make it grow?
> 
> Today I think it's hard to get a view of all the questions/answers and the mailing list archive is not that user friendly. If people more easy can search/browse for answers then less questions will be generated. I don't see a big and viable community as FreeSWITCH deserves IMO and I am certain of getting a online community board will increase the community in that sense. This is one of the first things I look for when getting a new component or trying to find help. I am still a beginner of FreeSWITCH even though I've followed the project for years and I know a community board would help me and most likely others. 
> 
> It's easy to scope/categorize the content of a community board and to allow certain discussions of possible configuration errors vs bugs etc, as a novice like my self I have a hard time to spot this and when I read answers like "bugs are not to discussed on the list - file a Jira". If you, like me, don't even know if this is a mistake by myself and get this kind of answer I really hesitate to ask.
> 
> All I want is the best for FreeSWITCH. What do you think?   
> 
> /Fred
>  
> 
> 
> 

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From jungleboogie0 at gmail.com  Fri Feb 23 20:25:00 2018
From: jungleboogie0 at gmail.com (Jungle Boogie)
Date: Fri, 23 Feb 2018 12:25:00 -0800
Subject: [Freeswitch-users] My 2 cents - online community
In-Reply-To: <VI1PR0902MB2143D7DDD51D903F6D9BCD33D2CD0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
References: <VI1PR0902MB2143D7DDD51D903F6D9BCD33D2CD0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
Message-ID: <20180223202500.GA33388@puffer.in.lylie.net>

On Thu 22 Feb 2018  4:33 PM, Fred Pettersson wrote:
> Look I don't wanna start a fire here and have full respect for the team behind FreeSWITCH.
> 
> It's 2018 right? I am talking about Evolution. Why doesn't the FreeSWITCH community have a more up to date online community board? I am talking about the mailing lists, why not use a tool to help the community a bit more and make it grow?

Have you joined the hipchat channel? https://hipchat.freeswitch.org

Forums are nice, if the participation is good. Otherwise, it's wasted developer
time setting it up.


From infos at madovsky.org  Fri Feb 23 20:35:50 2018
From: infos at madovsky.org (Madovsky)
Date: Fri, 23 Feb 2018 12:35:50 -0800
Subject: [Freeswitch-users] My 2 cents - online community
In-Reply-To: <VI1PR0902MB21437D62F6CEC1DFA88750F4D2CC0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
References: <VI1PR0902MB2143D7DDD51D903F6D9BCD33D2CD0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
 <785d549a-526d-4306-dee2-780946c1c0de@paulzillmann.de>
 <099401d3ac1c$9d33b630$d79b2290$@tollfreegateway.com>
 <CALXCt0qGuU2v9MP=D9+9yDk1fROZ91BN6vDQpO3rsbOHpro1dA@mail.gmail.com>
 <VI1PR0902MB21437D62F6CEC1DFA88750F4D2CC0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
Message-ID: <983e6acb-b1e1-6139-40bd-e357166b2683@madovsky.org>

 > just because some user can't use the search in an online forum is 
just for the wrong reason

could you elaborate?


if you need to discuss there is a fantastic freeswitch IRC channel ;)

today the choice to communicate is not based on old or new, it's just a 
question of taste....


On 2/23/2018 10:23 AM, Fred Pettersson wrote:
> Thank you all for feedback. Remember this was just my 2-cents. I just 
> want to highlight that I am not saying mailing-list communication is 
> not working but there are better options IMO.
>
> Just to clarify - I am talking about an online forum instead of the 
> mailing list. To justify the old way just because some user can't use 
> the search in an online forum is just for the wrong reason. How many 
> can't use them today or even know it's there today. It's a shopping 
> window. I know about the FS books, I've bought three of them (paper). 
> They are good but they do not cover everything. I know about 
> Confluence, ClueCon Weekly, I use YouTube to watch them etc. To be 
> fair I think you are missing my point here :) I think there is room 
> for a good place to have discussions with others in an easy to use 
> online tool (forum could serve that).
>
> You know what's most "scary" about the whole thing? That's non of you 
> does not even consider or elaborates about a change of this but tries 
> to guide me how/were instead but I am talking about the community. 
> Where is the spirit? :)
>
> As I said I don't want to start a fire here, I want the best for 
> FreeSWITCH and make the community to grow. I honestly don't think 
> mailman is the most efficient tool in terms of growing the community. 
> I rest my case I don't want to start a big thing.
>
> Regards,
> Fred
>
> ------------------------------------------------------------------------
> *Från:* FreeSWITCH-users 
> <freeswitch-users-bounces at lists.freeswitch.org> för Giovanni 
> Maruzzelli <gmaruzz at gmail.com>
> *Skickat:* den 23 februari 2018 09:26:23
> *Till:* FreeSWITCH Users Help
> *Ämne:* Re: [Freeswitch-users] My 2 cents - online community
> also, don't forget there are 3 FreeSWITCH books out there, each of 
> them available in ebook, pdf and printed paper format...
>
> If you can't buy them, you can at least ask a friend in the know to 
> provide them to you.
>
> -giovanni
>
>
> On 22 February 2018 at 21:35, <krice at tollfreegateway.com 
> <mailto:krice at tollfreegateway.com>> wrote:
>
>     As Paul just said, lists.freeswitch.org
>     <http://lists.freeswitch.org> contains a complete archive of the
>     mailing list. This site is heavily indexed by google.
>
>     A hint for searching it is when you google use the
>     “site:lists.freeswitch.org <http://lists.freeswitch.org>” or
>     “site:freeswitch.org <http://freeswitch.org>” to restrict your
>     google queries. This should return anything in the archive
>     relevant to your actual query string.
>
>     Have fun!
>
>     K
>
>     *From:*FreeSWITCH-users
>     [mailto:freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf
>     Of *Paul Zillmann
>     *Sent:* Thursday, February 22, 2018 1:51 PM
>     *To:* freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     *Subject:* Re: [Freeswitch-users] My 2 cents - online community
>
>     Hello Fred,
>
>     I think Google provides a good way of querying the mailing-list,
>     given that the subject-line is good and the content isn't just a
>     random "Hiiiiiiiii - how to solve command"
>
>     Many projects in the unix-world including operating systems rely
>     heavily on mailing-list communication. And that one works fine.
>
>     Could you state more precise what you mean by an online community
>     board? In my understanding that would be a forum.
>
>     Some users aren't able to use the search functions in such forums
>     either.
>
>     By now I can't see a benefit of an "online community board".
>
>     Besides the mailing-list you may participate in the weekly
>     conference call and ClueCon weekly.
>
>     Aside from that: don't be afraid to ask on the list and try to
>     apply the basic rules you'll find in many online forums: good
>     titles, be respectful, and put some effort into your text.
>
>     Sincerely,
>
>     Paul
>
>     Am 22.02.2018 um 17:33 schrieb Fred Pettersson:
>
>         Look I don't wanna start a fire here and have full respect for
>         the team behind FreeSWITCH.
>
>         It's 2018 right? I am talking about Evolution. Why doesn't the
>         FreeSWITCH community have a more up to date online community
>         board? I am talking about the mailing lists, why not use a
>         tool to help the community a bit more and make it grow?
>
>         Today I think it's hard to get a view of all the
>         questions/answers and the mailing list archive is not that
>         user friendly. If people more easy can search/browse for
>         answers then less questions will be generated. I don't see a
>         big and viable community as FreeSWITCH deserves IMO and I am
>         certain of getting a online community board will increase the
>         community in that sense. This is one of the first things I
>         look for when getting a new component or trying to find help.
>         I am still a beginner of FreeSWITCH even though I've followed
>         the project for years and I know a community board would help
>         me and most likely others.
>
>         It's easy to scope/categorize the content of a community board
>         and to allow certain discussions of possible configuration
>         errors vs bugs etc, as a novice like my self I have a hard
>         time to spot this and when I read answers like "bugs are not
>         to discussed on the list - file a Jira". If you, like me,
>         don't even know if this is a mistake by myself and get this
>         kind of answer I really hesitate to ask.
>
>         All I want is the best for FreeSWITCH. What do you think?
>
>         /Fred
>
>
>
>
>         _________________________________________________________________________
>
>         Professional FreeSWITCH Consulting Services:
>
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>
>         http://www.freeswitchsolutions.com
>         <http://www.freeswitchsolutions.com>
>
>           
>
>         Official FreeSWITCH Sites
>
>         http://www.freeswitch.org
>
>         http://confluence.freeswitch.org
>         <http://confluence.freeswitch.org>
>
>         http://www.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>
>
>         http://www.freeswitch.org
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com>
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org <http://confluence.freeswitch.org>
>     http://www.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>
>     http://www.freeswitch.org
>
>
>
>
> -- 
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 domrumsey at hotmail.com  Fri Feb 23 20:57:41 2018
From: domrumsey at hotmail.com (Dom Rumsey)
Date: Fri, 23 Feb 2018 20:57:41 +0000
Subject: [Freeswitch-users] New Question for FreeSWITCH Users
Message-ID: <CY1PR16MB0316AFD0D71388AFDF2AB1BAACCC0@CY1PR16MB0316.namprd16.prod.outlook.com>

Hi



I’ve created a multiway video chat application using FreeSWITCH v1.6.20 and I’m having a specific problem with the video streams.



Basically, I am using conference mode pass through and it’s working great unless one of the attendees is vmuted. If the attendee who is vmuted speaks it freezes the video feed of the last person who was speaking. It’s as if the canvas wants to show the video of the speaker but realises that there is no video to share so just keeps showing the last frame in the canvas.



I have tried adding the parameters video-mute-exit-canvas and video-required-for-canvas but to no avail. They didn’t seem to do anything, so I just want to make sure I am not missing something. Ideally the video of the last person sharing video will continue to show in the canvas while the user who is vmuted speaks. Is there any way to achieve this? Any help would be much appreciated, I’ve hunted through the mailing lists and couldn’t find anything that addresses this specific challenge.



Thanks in advance for your help.

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From mike at jerris.com  Fri Feb 23 21:53:29 2018
From: mike at jerris.com (Michael Jerris)
Date: Fri, 23 Feb 2018 16:53:29 -0500
Subject: [Freeswitch-users] New Question for FreeSWITCH Users
In-Reply-To: <CY1PR16MB0316AFD0D71388AFDF2AB1BAACCC0@CY1PR16MB0316.namprd16.prod.outlook.com>
References: <CY1PR16MB0316AFD0D71388AFDF2AB1BAACCC0@CY1PR16MB0316.namprd16.prod.outlook.com>
Message-ID: <4B45CD71-B83B-4D2A-8B19-F3AD62D55F93@jerris.com>

All of the canvas related features are features of mux mode.  There are basically no features available in passthrough mode, its just a left over legacy feature from before we had real video support.  It doesn’t work very well and is not very useful if you want reliable video and any features for manipulating video content.



> On Feb 23, 2018, at 3:57 PM, Dom Rumsey <domrumsey at hotmail.com> wrote:
> 
> Hi
>  
> I’ve created a multiway video chat application using FreeSWITCH v1.6.20 and I’m having a specific problem with the video streams.
>  
> Basically, I am using conference mode pass through and it’s working great unless one of the attendees is vmuted. If the attendee who is vmuted speaks it freezes the video feed of the last person who was speaking. It’s as if the canvas wants to show the video of the speaker but realises that there is no video to share so just keeps showing the last frame in the canvas.

Thats exactly what it is doing.

>  
> I have tried adding the parameters video-mute-exit-canvas and video-required-for-canvas but to no avail. They didn’t seem to do anything, so I just want to make sure I am not missing something. Ideally the video of the last person sharing video will continue to show in the canvas while the user who is vmuted speaks. Is there any way to achieve this? Any help would be much appreciated, I’ve hunted through the mailing lists and couldn’t find anything that addresses this specific challenge.

Features like this require mux mode

>  
> Thanks in advance for your help.


Mike
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From domrumsey at hotmail.com  Sat Feb 24 09:41:36 2018
From: domrumsey at hotmail.com (Dom Rumsey)
Date: Sat, 24 Feb 2018 09:41:36 +0000
Subject: [Freeswitch-users] New Question for FreeSWITCH Users
In-Reply-To: <4B45CD71-B83B-4D2A-8B19-F3AD62D55F93@jerris.com>
References: <CY1PR16MB0316AFD0D71388AFDF2AB1BAACCC0@CY1PR16MB0316.namprd16.prod.outlook.com>,
 <4B45CD71-B83B-4D2A-8B19-F3AD62D55F93@jerris.com>
Message-ID: <CY1PR16MB031654E7D5AD648D534B2F69ACC30@CY1PR16MB0316.namprd16.prod.outlook.com>

Got it. Thanks Mike.


The machine I'm running doesn't have loads of CPU and earlier in the project I played around with MUX, but noticed it took a lot of CPU. I can swap it out for a different machine, however any tips on what I can do to reduce stress on CPU? I guess lowering the fps will help. Anything else you suggest?


Cheers.


________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Michael Jerris <mike at jerris.com>
Sent: Friday, February 23, 2018 9:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] New Question for FreeSWITCH Users

All of the canvas related features are features of mux mode.  There are basically no features available in passthrough mode, its just a left over legacy feature from before we had real video support.  It doesn’t work very well and is not very useful if you want reliable video and any features for manipulating video content.



On Feb 23, 2018, at 3:57 PM, Dom Rumsey <domrumsey at hotmail.com<mailto:domrumsey at hotmail.com>> wrote:

Hi



I’ve created a multiway video chat application using FreeSWITCH v1.6.20 and I’m having a specific problem with the video streams.



Basically, I am using conference mode pass through and it’s working great unless one of the attendees is vmuted. If the attendee who is vmuted speaks it freezes the video feed of the last person who was speaking. It’s as if the canvas wants to show the video of the speaker but realises that there is no video to share so just keeps showing the last frame in the canvas.

Thats exactly what it is doing.




I have tried adding the parameters video-mute-exit-canvas and video-required-for-canvas but to no avail. They didn’t seem to do anything, so I just want to make sure I am not missing something. Ideally the video of the last person sharing video will continue to show in the canvas while the user who is vmuted speaks. Is there any way to achieve this? Any help would be much appreciated, I’ve hunted through the mailing lists and couldn’t find anything that addresses this specific challenge.

Features like this require mux mode




Thanks in advance for your help.


Mike
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From b631093f-779b-4d67-9ffe-5f6d5b1d3f8a at protonmail.ch  Sat Feb 24 18:35:04 2018
From: b631093f-779b-4d67-9ffe-5f6d5b1d3f8a at protonmail.ch (Bob Smith)
Date: Sat, 24 Feb 2018 13:35:04 -0500
Subject: [Freeswitch-users] Is it possible to "impersonate" a Freeswitch
	directory user ?
Message-ID: <VlVSlht2yUwXBAQtEI3ejoOsz5huIN2wEU669h67o4aBxMWDW6wowWruqLpu_E58Zj_XwUWCC6K1et13kl8RsHgUIR02Qqn5QoH-DcfqspQ=@protonmail.ch>

Hello,

Let's say I'm trying to setup a follow-me type scenario on inbound DDI.

This is all of course fairly easily done by means of a bridge.  But I have one problem.

Let's say I have a few important variables that sofia needs, and these are stored on a per-user basis in their directory entry, e.g. :

"/etc/freeswitch/directory/default/2022.xml" might have params such as :
<variable name="privacy" value="yes"/>
<variable name="effective_caller_id_name" value="foobar"/>
<variable name="effective_caller_id_number" value="00000"/>

Is it then possible to "impersonate the user" or in other words refer to those parameters when calling sofia from, say, "/etc/freeswitch/dialplan/public/25_MYDDI.xml".

i.e. when i call :
<action application="bridge" data="...blah...,sofia/default/MYCELLNUMBERGOESHERE"/>
sofia picks up / gets given privacy/effective_caller_id_number etc. preferences/settings

I don't really want to end up with duplicate variables all over the place, and storing them in the directory entry seems the most sensible ?
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From ryharris at airmail.cc  Sun Feb 25 13:58:59 2018
From: ryharris at airmail.cc (Ryan Harris)
Date: Sun, 25 Feb 2018 08:58:59 -0500
Subject: [Freeswitch-users] Lua question (re: FollowMe Dialplan Example
 from the docs)
In-Reply-To: <lL7Qbo1q4G4E_xf2PoXnRUk2-JQ-sQ5xRdeP2osSAOBxI-d6pfcuxaSMrNMX35k4Kq14-CwZ0i9JjmNsqiIp-8h3Ib5cTRVRtzN1jGxuzHc=@protonmail.ch>
References: <lL7Qbo1q4G4E_xf2PoXnRUk2-JQ-sQ5xRdeP2osSAOBxI-d6pfcuxaSMrNMX35k4Kq14-CwZ0i9JjmNsqiIp-8h3Ib5cTRVRtzN1jGxuzHc=@protonmail.ch>
Message-ID: <a9ae73a4-15a6-e0f2-584b-e1dd6fdd9da7@airmail.cc>

On 02/22/2018 03:15 PM, Bob Smith wrote:
> My reading of the script is that it does not take care of the somewhat
> obvious problem of what happens if the cell phone goes to voicemail ?
>
> My understanding of the script as presented is that it would just sit
> there forever waiting for a reply ?
>

I haven't actually used that script, but it shouldn't sit there forever.
Take a look at what each of the arguments mean:

https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:playAndGetDigits

In the example, session:playAndGetDigits will wait 2 seconds for an
entry and then increment the attempts if unsuccessful. On the third
unsuccessful attempt, digits will be "". Since digits won't be "1" or
"2" the script will session:hangup("NO_ANSWER").

If whatever picks up the call can't dial digits, the call should hangup
after 6 seconds.



From ryharris at airmail.cc  Sun Feb 25 14:14:20 2018
From: ryharris at airmail.cc (Ryan Harris)
Date: Sun, 25 Feb 2018 09:14:20 -0500
Subject: [Freeswitch-users] Is it possible to "impersonate" a Freeswitch
 directory user ?
In-Reply-To: <VlVSlht2yUwXBAQtEI3ejoOsz5huIN2wEU669h67o4aBxMWDW6wowWruqLpu_E58Zj_XwUWCC6K1et13kl8RsHgUIR02Qqn5QoH-DcfqspQ=@protonmail.ch>
References: <VlVSlht2yUwXBAQtEI3ejoOsz5huIN2wEU669h67o4aBxMWDW6wowWruqLpu_E58Zj_XwUWCC6K1et13kl8RsHgUIR02Qqn5QoH-DcfqspQ=@protonmail.ch>
Message-ID: <5d7f6979-fbe3-4932-b762-6de18e270456@airmail.cc>

On 02/24/2018 01:35 PM, Bob Smith wrote:
> Let's say I have a few important variables that sofia needs, and these
> are stored on a per-user basis in their directory entry, e.g. :
>
> "/etc/freeswitch/directory/default/2022.xml" might have params such as :
> <variable name="privacy" value="yes"/>
> <variable name="effective_caller_id_name" value="foobar"/>
> <variable name="effective_caller_id_number" value="00000"/>
>
> Is it then possible to "impersonate the user" or in other words refer
> to those parameters when calling sofia from, say,
> "/etc/freeswitch/dialplan/public/25_MYDDI.xml".

https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-user_data


From brian at freeswitch.com  Mon Feb 26 14:11:52 2018
From: brian at freeswitch.com (Brian West)
Date: Mon, 26 Feb 2018 08:11:52 -0600
Subject: [Freeswitch-users] Is it possible to "impersonate" a Freeswitch
 directory user ?
In-Reply-To: <5d7f6979-fbe3-4932-b762-6de18e270456@airmail.cc>
References: <VlVSlht2yUwXBAQtEI3ejoOsz5huIN2wEU669h67o4aBxMWDW6wowWruqLpu_E58Zj_XwUWCC6K1et13kl8RsHgUIR02Qqn5QoH-DcfqspQ=@protonmail.ch>
 <5d7f6979-fbe3-4932-b762-6de18e270456@airmail.cc>
Message-ID: <CAEJMVkCLf0z4XWbm4AAFzQayghyvLTY8j6tcy9WPb_+=LygCVA@mail.gmail.com>

<action application="set_user" data="user@${domain_name}"/>


On Sun, Feb 25, 2018 at 8:14 AM, Ryan Harris <ryharris at airmail.cc> wrote:

> On 02/24/2018 01:35 PM, Bob Smith wrote:
> > Let's say I have a few important variables that sofia needs, and these
> > are stored on a per-user basis in their directory entry, e.g. :
> >
> > "/etc/freeswitch/directory/default/2022.xml" might have params such as :
> > <variable name="privacy" value="yes"/>
> > <variable name="effective_caller_id_name" value="foobar"/>
> > <variable name="effective_caller_id_number" value="00000"/>
> >
> > Is it then possible to "impersonate the user" or in other words refer
> > to those parameters when calling sofia from, say,
> > "/etc/freeswitch/dialplan/public/25_MYDDI.xml".
>
> https://freeswitch.org/confluence/display/FREESWITCH/
> mod_commands#mod_commands-user_data
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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




-- 

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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From alihaider.4189 at gmail.com  Mon Feb 26 15:00:17 2018
From: alihaider.4189 at gmail.com (Ali Haider)
Date: Mon, 26 Feb 2018 15:00:17 +0000
Subject: [Freeswitch-users] [ERROR] fs_cli.c:1659 main() Error
 Connecting [Socket Connection Error]
In-Reply-To: <2EDEB65D-F7A3-404B-86A4-B9B47F275C05@gmail.com>
References: <2EDEB65D-F7A3-404B-86A4-B9B47F275C05@gmail.com>
Message-ID: <CAB2bS9Tvy+D-uaoh-5h=tqt4M-XV_oW+1OH08QMeJySdhOkccw@mail.gmail.com>

[ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error]
Usage: fs_cli [-H <host>] [-P <port>] [-p <se
how you can ressolve this error kindly tell me please

On 24 January 2018 at 17:09, Raimundo Pérez Nieves <
raimundo.perez.cuba at gmail.com> wrote:

> Yes I checked. Somebody told me a great solution, it was start freeswitch
> going to usr/bin and executing ./freeswitch.
> That start freeswitch in console mode and shows you every problem. It was
> just an extension tag unclosed.
> Thanks anyway
>
> Enviado desde mi iPhone
>
> El 24/01/2018, a la(s) 18:05, Bilal Abbasi <bilaln018 at gmail.com> escribió:
>
> Does the freeswitch process started?
> check that.
>
> On Wed, Jan 24, 2018 at 8:50 PM, Raimundo Pérez Nieves <
> raimundo.perez.cuba at gmail.com> wrote:
>
>> Hi guys, I put shutdown in freeswitch but did't change nothing and when
>> I fs_cli I get this error [ERROR] fs_cli.c:1659 main() Error Connecting
>> [Socket Connection Error]
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 wsimon at stratusvideo.com  Mon Feb 26 15:24:34 2018
From: wsimon at stratusvideo.com (William Simon)
Date: Mon, 26 Feb 2018 15:24:34 +0000
Subject: [Freeswitch-users] My 2 cents - online community
In-Reply-To: <45DED6CD-40F7-4F49-A1E6-668E256E98DE@jerris.com>
References: <VI1PR0902MB2143D7DDD51D903F6D9BCD33D2CD0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
 <785d549a-526d-4306-dee2-780946c1c0de@paulzillmann.de>
 <099401d3ac1c$9d33b630$d79b2290$@tollfreegateway.com>
 <CALXCt0qGuU2v9MP=D9+9yDk1fROZ91BN6vDQpO3rsbOHpro1dA@mail.gmail.com>
 <VI1PR0902MB21437D62F6CEC1DFA88750F4D2CC0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
 <45DED6CD-40F7-4F49-A1E6-668E256E98DE@jerris.com>
Message-ID: <5AD6E643-B0F9-4AEF-827F-707ED11BA046@stratusvideo.com>

Would love an option to upvote / downvote so that great solutions can be highlighted and "haiiii i have error please help fix" can be downvoted to invisibility.

On Feb 23, 2018, at 2:27 PM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:

You are not starting a fire at all.  I understood you to be looking at wanting some better sort of forum.  Mailman 3 as i said is the planned solution to this as it does a decent job combining a forum and mailing list together into one concept, in a way where we don’t have to monitor two different things separately and you get the best exposure.  I totally understood what you were proposing, was letting you know what the blockers to moving forward on this are, and putting it out there a way that you could help the community by moving this process forward.  Please jump in and assist in moving this process forward if you are willing and able.  Thanks as always we appreciate the feedback about how we can foster a better community.

Mike


On Feb 23, 2018, at 1:23 PM, Fred Pettersson <t90fpe at outlook.com<mailto:t90fpe at outlook.com>> wrote:


Thank you all for feedback. Remember this was just my 2-cents. I just want to highlight that I am not saying mailing-list communication is not working but there are better options IMO.

Just to clarify - I am talking about an online forum instead of the mailing list. To justify the old way just because some user can't use the search in an online forum is just for the wrong reason. How many can't use them today or even know it's there today. It's a shopping window. I know about the FS books, I've bought three of them (paper). They are good but they do not cover everything. I know about Confluence, ClueCon Weekly, I use YouTube to watch them etc. To be fair I think you are missing my point here :) I think there is room for a good place to have discussions with others in an easy to use online tool (forum could serve that).

You know what's most "scary" about the whole thing? That's non of you does not even consider or elaborates about a change of this but tries to guide me how/were instead but I am talking about the community. Where is the spirit? :)

As I said I don't want to start a fire here, I want the best for FreeSWITCH and make the community to grow. I honestly don't think mailman is the most efficient tool in terms of growing the community. I rest my case I don't want to start a big thing.

Regards,
Fred


________________________________
Från: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> för Giovanni Maruzzelli <gmaruzz at gmail.com<mailto:gmaruzz at gmail.com>>
Skickat: den 23 februari 2018 09:26:23
Till: FreeSWITCH Users Help
Ämne: Re: [Freeswitch-users] My 2 cents - online community

also, don't forget there are 3 FreeSWITCH books out there, each of them available in ebook, pdf and printed paper format...

If you can't buy them, you can at least ask a friend in the know to provide them to you.

-giovanni


On 22 February 2018 at 21:35, <krice at tollfreegateway.com<mailto:krice at tollfreegateway.com>> wrote:
As Paul just said, lists.freeswitch.org<http://lists.freeswitch.org/> contains a complete archive of the mailing list. This site is heavily indexed by google.

A hint for searching it is when you google use the “site:lists.freeswitch.org<http://lists.freeswitch.org/>” or “site:freeswitch.org<http://freeswitch.org/>” to restrict your google queries. This should return anything in the archive relevant to your actual query string.

Have fun!
K

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Paul Zillmann
Sent: Thursday, February 22, 2018 1:51 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] My 2 cents - online community

Hello Fred,

I think Google provides a good way of querying the mailing-list, given that the subject-line is good and the content isn't just a random "Hiiiiiiiii - how to solve command"
Many projects in the unix-world including operating systems rely heavily on mailing-list communication. And that one works fine.
Could you state more precise what you mean by an online community board? In my understanding that would be a forum.

Some users aren't able to use the search functions in such forums either.
By now I can't see a benefit of an "online community board".

Besides the mailing-list you may participate in the weekly conference call and ClueCon weekly.
Aside from that: don't be afraid to ask on the list and try to apply the basic rules you'll find in many online forums: good titles, be respectful, and put some effort into your text.

Sincerely,
Paul

Am 22.02.2018 um 17:33 schrieb Fred Pettersson:
Look I don't wanna start a fire here and have full respect for the team behind FreeSWITCH.

It's 2018 right? I am talking about Evolution. Why doesn't the FreeSWITCH community have a more up to date online community board? I am talking about the mailing lists, why not use a tool to help the community a bit more and make it grow?

Today I think it's hard to get a view of all the questions/answers and the mailing list archive is not that user friendly. If people more easy can search/browse for answers then less questions will be generated. I don't see a big and viable community as FreeSWITCH deserves IMO and I am certain of getting a online community board will increase the community in that sense. This is one of the first things I look for when getting a new component or trying to find help. I am still a beginner of FreeSWITCH even though I've followed the project for years and I know a community board would help me and most likely others.

It's easy to scope/categorize the content of a community board and to allow certain discussions of possible configuration errors vs bugs etc, as a novice like my self I have a hard time to spot this and when I read answers like "bugs are not to discussed on the list - file a Jira". If you, like me, don't even know if this is a mistake by myself and get this kind of answer I really hesitate to ask.

All I want is the best for FreeSWITCH. What do you think?

/Fred





_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.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



“The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”
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From brian at freeswitch.com  Mon Feb 26 18:20:33 2018
From: brian at freeswitch.com (Brian West)
Date: Mon, 26 Feb 2018 12:20:33 -0600
Subject: [Freeswitch-users] My 2 cents - online community
In-Reply-To: <5AD6E643-B0F9-4AEF-827F-707ED11BA046@stratusvideo.com>
References: <VI1PR0902MB2143D7DDD51D903F6D9BCD33D2CD0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
 <785d549a-526d-4306-dee2-780946c1c0de@paulzillmann.de>
 <099401d3ac1c$9d33b630$d79b2290$@tollfreegateway.com>
 <CALXCt0qGuU2v9MP=D9+9yDk1fROZ91BN6vDQpO3rsbOHpro1dA@mail.gmail.com>
 <VI1PR0902MB21437D62F6CEC1DFA88750F4D2CC0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
 <45DED6CD-40F7-4F49-A1E6-668E256E98DE@jerris.com>
 <5AD6E643-B0F9-4AEF-827F-707ED11BA046@stratusvideo.com>
Message-ID: <CAEJMVkB45n1X_VM9iOOuU=WQptcZuFDeacQuF4xwUdXrSa99Zg@mail.gmail.com>

Kinda like reddit.com/r/freeswitch? :0

On Mon, Feb 26, 2018 at 9:24 AM, William Simon <wsimon at stratusvideo.com>
wrote:

> Would love an option to upvote / downvote so that great solutions can be
> highlighted and "haiiii i have error please help fix" can be downvoted to
> invisibility.
>
> On Feb 23, 2018, at 2:27 PM, Michael Jerris <mike at jerris.com> wrote:
>
> You are not starting a fire at all.  I understood you to be looking at
> wanting some better sort of forum.  Mailman 3 as i said is the planned
> solution to this as it does a decent job combining a forum and mailing list
> together into one concept, in a way where we don’t have to monitor two
> different things separately and you get the best exposure.  I totally
> understood what you were proposing, was letting you know what the blockers
> to moving forward on this are, and putting it out there a way that you
> could help the community by moving this process forward.  Please jump in
> and assist in moving this process forward if you are willing and able.
> Thanks as always we appreciate the feedback about how we can foster a
> better community.
>
> Mike
>
>
> On Feb 23, 2018, at 1:23 PM, Fred Pettersson <t90fpe at outlook.com> wrote:
>
> Thank you all for feedback. Remember this was just my 2-cents. I just want
> to highlight that I am not saying mailing-list communication is not working
> but there are better options IMO.
>
> Just to clarify - I am talking about an online forum instead of the
> mailing list. To justify the old way just because some user can't use the
> search in an online forum is just for the wrong reason. How many can't use
> them today or even know it's there today. It's a shopping window. I know
> about the FS books, I've bought three of them (paper). They are good but
> they do not cover everything. I know about Confluence, ClueCon Weekly, I
> use YouTube to watch them etc. To be fair I think you are missing my point
> here :) I think there is room for a good place to have discussions with
> others in an easy to use online tool (forum could serve that).
>
> You know what's most "scary" about the whole thing? That's non of you does
> not even consider or elaborates about a change of this but tries to guide
> me how/were instead but I am talking about the community. Where is the
> spirit? :)
>
> As I said I don't want to start a fire here, I want the best for
> FreeSWITCH and make the community to grow. I honestly don't think mailman
> is the most efficient tool in terms of growing the community. I rest my
> case I don't want to start a big thing.
>
> Regards,
> Fred
>
> ------------------------------
> *Från:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org>
> för Giovanni Maruzzelli <gmaruzz at gmail.com>
> *Skickat:* den 23 februari 2018 09:26:23
> *Till:* FreeSWITCH Users Help
> *Ämne:* Re: [Freeswitch-users] My 2 cents - online community
>
> also, don't forget there are 3 FreeSWITCH books out there, each of them
> available in ebook, pdf and printed paper format...
>
> If you can't buy them, you can at least ask a friend in the know to
> provide them to you.
>
> -giovanni
>
>
> On 22 February 2018 at 21:35, <krice at tollfreegateway.com> wrote:
>
> As Paul just said, lists.freeswitch.org contains a complete archive of
> the mailing list. This site is heavily indexed by google.
>
> A hint for searching it is when you google use the “site:
> lists.freeswitch.org” or “site:freeswitch.org” to restrict your google
> queries. This should return anything in the archive relevant to your actual
> query string.
>
> Have fun!
> K
>
> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc
> es at lists.freeswitch.org] *On Behalf Of *Paul Zillmann
> *Sent:* Thursday, February 22, 2018 1:51 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] My 2 cents - online community
>
> Hello Fred,
>
> I think Google provides a good way of querying the mailing-list, given
> that the subject-line is good and the content isn't just a random
> "Hiiiiiiiii - how to solve command"
> Many projects in the unix-world including operating systems rely heavily
> on mailing-list communication. And that one works fine.
> Could you state more precise what you mean by an online community board?
> In my understanding that would be a forum.
>
> Some users aren't able to use the search functions in such forums either.
> By now I can't see a benefit of an "online community board".
>
> Besides the mailing-list you may participate in the weekly conference call
> and ClueCon weekly.
> Aside from that: don't be afraid to ask on the list and try to apply the
> basic rules you'll find in many online forums: good titles, be respectful,
> and put some effort into your text.
>
> Sincerely,
> Paul
>
> Am 22.02.2018 um 17:33 schrieb Fred Pettersson:
>
> Look I don't wanna start a fire here and have full respect for the team
> behind FreeSWITCH.
>
> It's 2018 right? I am talking about Evolution. Why doesn't the FreeSWITCH
> community have a more up to date online community board? I am talking about
> the mailing lists, why not use a tool to help the community a bit more and
> make it grow?
>
> Today I think it's hard to get a view of all the questions/answers and the
> mailing list archive is not that user friendly. If people more easy can
> search/browse for answers then less questions will be generated. I don't
> see a big and viable community as FreeSWITCH deserves IMO and I am certain
> of getting a online community board will increase the community in that
> sense. This is one of the first things I look for when getting a new
> component or trying to find help. I am still a beginner of FreeSWITCH even
> though I've followed the project for years and I know a community board
> would help me and most likely others.
>
> It's easy to scope/categorize the content of a community board and to
> allow certain discussions of possible configuration errors vs bugs etc, as
> a novice like my self I have a hard time to spot this and when I read
> answers like "bugs are not to discussed on the list - file a Jira". If you,
> like me, don't even know if this is a mistake by myself and get this kind
> of answer I really hesitate to ask.
>
> All I want is the best for FreeSWITCH. What do you think?
>
> /Fred
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
>
>
>
> “The information transmitted is intended only for the person or entity to
> which it is addressed and may contain proprietary, business-confidential
> and/or privileged material. If you are not the intended recipient of this
> message you are hereby notified that any use, review, retransmission,
> dissemination, distribution, reproduction or any action taken in reliance
> upon this message is prohibited. If you received this in error, please
> contact the sender and delete the material from any computer.”
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From kathleen at freeswitch.com  Mon Feb 26 20:12:58 2018
From: kathleen at freeswitch.com (Kathleen King)
Date: Mon, 26 Feb 2018 12:12:58 -0800
Subject: [Freeswitch-users] ClueCon Weekly with Dan Jenkins!
Message-ID: <CAOStUPZ41bsgLYhWuDnBGYt1DQz1jJ09VUcMeQMjP6EU8H7ghg@mail.gmail.com>

Hey guys!

Dan Jenkins will be joining us this Wednesday to talk about CommCon! You
can dial 888 at https://conference.freeswitch.org/vc/ or watch it live
here: https://youtu.be/1dk1fJ4xAbU



See you there!


Kathleen King | Public Relations / Administrative Assistant

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

Email: Kathleen at freeswitch.com

Mobile: 703-859-3757

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

[image: color-facebook-96.png] <https://www.facebook.com/freeswitch/> [image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
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From gregor at infomedia.si  Mon Feb 26 20:16:34 2018
From: gregor at infomedia.si (Gregor Nanger)
Date: Mon, 26 Feb 2018 21:16:34 +0100
Subject: [Freeswitch-users] My 2 cents - online community
In-Reply-To: <CAEJMVkB45n1X_VM9iOOuU=WQptcZuFDeacQuF4xwUdXrSa99Zg@mail.gmail.com>
References: <VI1PR0902MB2143D7DDD51D903F6D9BCD33D2CD0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
 <785d549a-526d-4306-dee2-780946c1c0de@paulzillmann.de>
 <099401d3ac1c$9d33b630$d79b2290$@tollfreegateway.com>
 <CALXCt0qGuU2v9MP=D9+9yDk1fROZ91BN6vDQpO3rsbOHpro1dA@mail.gmail.com>
 <VI1PR0902MB21437D62F6CEC1DFA88750F4D2CC0@VI1PR0902MB2143.eurprd09.prod.outlook.com>
 <45DED6CD-40F7-4F49-A1E6-668E256E98DE@jerris.com>
 <5AD6E643-B0F9-4AEF-827F-707ED11BA046@stratusvideo.com>
 <CAEJMVkB45n1X_VM9iOOuU=WQptcZuFDeacQuF4xwUdXrSa99Zg@mail.gmail.com>
Message-ID: <CAC7ttFFxcXH=S2CsnXv0y=nEyyutFUz6U+Z=_7OmX6eNVkBs2g@mail.gmail.com>

Only think that bothers me is that this solution now is filling my inbox

2018-02-26 19:20 GMT+01:00 Brian West <brian at freeswitch.com>:

> Kinda like reddit.com/r/freeswitch? :0
>
> On Mon, Feb 26, 2018 at 9:24 AM, William Simon <wsimon at stratusvideo.com>
> wrote:
>
>> Would love an option to upvote / downvote so that great solutions can be
>> highlighted and "haiiii i have error please help fix" can be downvoted to
>> invisibility.
>>
>> On Feb 23, 2018, at 2:27 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>> You are not starting a fire at all.  I understood you to be looking at
>> wanting some better sort of forum.  Mailman 3 as i said is the planned
>> solution to this as it does a decent job combining a forum and mailing list
>> together into one concept, in a way where we don’t have to monitor two
>> different things separately and you get the best exposure.  I totally
>> understood what you were proposing, was letting you know what the blockers
>> to moving forward on this are, and putting it out there a way that you
>> could help the community by moving this process forward.  Please jump in
>> and assist in moving this process forward if you are willing and able.
>> Thanks as always we appreciate the feedback about how we can foster a
>> better community.
>>
>> Mike
>>
>>
>> On Feb 23, 2018, at 1:23 PM, Fred Pettersson <t90fpe at outlook.com> wrote:
>>
>> Thank you all for feedback. Remember this was just my 2-cents. I just
>> want to highlight that I am not saying mailing-list communication is not
>> working but there are better options IMO.
>>
>> Just to clarify - I am talking about an online forum instead of the
>> mailing list. To justify the old way just because some user can't use the
>> search in an online forum is just for the wrong reason. How many can't use
>> them today or even know it's there today. It's a shopping window. I know
>> about the FS books, I've bought three of them (paper). They are good but
>> they do not cover everything. I know about Confluence, ClueCon Weekly, I
>> use YouTube to watch them etc. To be fair I think you are missing my point
>> here :) I think there is room for a good place to have discussions with
>> others in an easy to use online tool (forum could serve that).
>>
>> You know what's most "scary" about the whole thing? That's non of you
>> does not even consider or elaborates about a change of this but tries to
>> guide me how/were instead but I am talking about the community. Where is
>> the spirit? :)
>>
>> As I said I don't want to start a fire here, I want the best for
>> FreeSWITCH and make the community to grow. I honestly don't think mailman
>> is the most efficient tool in terms of growing the community. I rest my
>> case I don't want to start a big thing.
>>
>> Regards,
>> Fred
>>
>> ------------------------------
>> *Från:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org>
>> för Giovanni Maruzzelli <gmaruzz at gmail.com>
>> *Skickat:* den 23 februari 2018 09:26:23
>> *Till:* FreeSWITCH Users Help
>> *Ämne:* Re: [Freeswitch-users] My 2 cents - online community
>>
>> also, don't forget there are 3 FreeSWITCH books out there, each of them
>> available in ebook, pdf and printed paper format...
>>
>> If you can't buy them, you can at least ask a friend in the know to
>> provide them to you.
>>
>> -giovanni
>>
>>
>> On 22 February 2018 at 21:35, <krice at tollfreegateway.com> wrote:
>>
>> As Paul just said, lists.freeswitch.org contains a complete archive of
>> the mailing list. This site is heavily indexed by google.
>>
>> A hint for searching it is when you google use the “site:
>> lists.freeswitch.org” or “site:freeswitch.org” to restrict your google
>> queries. This should return anything in the archive relevant to your actual
>> query string.
>>
>> Have fun!
>> K
>>
>> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc
>> es at lists.freeswitch.org] *On Behalf Of *Paul Zillmann
>> *Sent:* Thursday, February 22, 2018 1:51 PM
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] My 2 cents - online community
>>
>> Hello Fred,
>>
>> I think Google provides a good way of querying the mailing-list, given
>> that the subject-line is good and the content isn't just a random
>> "Hiiiiiiiii - how to solve command"
>> Many projects in the unix-world including operating systems rely heavily
>> on mailing-list communication. And that one works fine.
>> Could you state more precise what you mean by an online community board?
>> In my understanding that would be a forum.
>>
>> Some users aren't able to use the search functions in such forums either.
>> By now I can't see a benefit of an "online community board".
>>
>> Besides the mailing-list you may participate in the weekly conference
>> call and ClueCon weekly.
>> Aside from that: don't be afraid to ask on the list and try to apply the
>> basic rules you'll find in many online forums: good titles, be respectful,
>> and put some effort into your text.
>>
>> Sincerely,
>> Paul
>>
>> Am 22.02.2018 um 17:33 schrieb Fred Pettersson:
>>
>> Look I don't wanna start a fire here and have full respect for the team
>> behind FreeSWITCH.
>>
>> It's 2018 right? I am talking about Evolution. Why doesn't the FreeSWITCH
>> community have a more up to date online community board? I am talking about
>> the mailing lists, why not use a tool to help the community a bit more and
>> make it grow?
>>
>> Today I think it's hard to get a view of all the questions/answers and
>> the mailing list archive is not that user friendly. If people more easy can
>> search/browse for answers then less questions will be generated. I don't
>> see a big and viable community as FreeSWITCH deserves IMO and I am certain
>> of getting a online community board will increase the community in that
>> sense. This is one of the first things I look for when getting a new
>> component or trying to find help. I am still a beginner of FreeSWITCH even
>> though I've followed the project for years and I know a community board
>> would help me and most likely others.
>>
>> It's easy to scope/categorize the content of a community board and to
>> allow certain discussions of possible configuration errors vs bugs etc, as
>> a novice like my self I have a hard time to spot this and when I read
>> answers like "bugs are not to discussed on the list - file a Jira". If you,
>> like me, don't even know if this is a mistake by myself and get this kind
>> of answer I really hesitate to ask.
>>
>> All I want is the best for FreeSWITCH. What do you think?
>>
>> /Fred
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>>
>>
>>
>> “The information transmitted is intended only for the person or entity to
>> which it is addressed and may contain proprietary, business-confidential
>> and/or privileged material. If you are not the intended recipient of this
>> message you are hereby notified that any use, review, retransmission,
>> dissemination, distribution, reproduction or any action taken in reliance
>> upon this message is prohibited. If you received this in error, please
>> contact the sender and delete the material from any computer.”
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
>
> --
>
> 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: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
> color-twitter-96.png]
> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
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From domrumsey at hotmail.com  Mon Feb 26 20:57:34 2018
From: domrumsey at hotmail.com (Dom Rumsey)
Date: Mon, 26 Feb 2018 20:57:34 +0000
Subject: [Freeswitch-users] Conference mux mode video flickering
Message-ID: <CY1PR16MB03161C0EC0B809D96D0B4A5FACC10@CY1PR16MB0316.namprd16.prod.outlook.com>

Hi guys


Thanks for your help previously.


I'm now using MUX mode for video and I'm getting a problem whereby a conference with 3 or more participants keeps flickering video streams on the floor (we are using video-muxing-personal-canvas parameter). It's as if it can't decide who to show. Below is my conference profile:



<profile name="cp">

      <param name="domain" value="$${domain}"/>

      <param name="rate" value="8000"/>

      <param name="video-mode" value="mux"/>

      <param name="video-layout-name" value="1x1"/>

      <param name="interval" value="20"/>

      <param name="caller-controls" value="default"/>

      <param name="energy-level" value="0"/>

      <param name="video-auto-floor-msec" value="3000" />

      <param name="video_no_video_avatar_png" value="/usr/local/freeswitch/conf/images/video-muted.png" />

      <param name="video_mute_png" value="/usr/local/freeswitch/conf/images/video-muted.png" />

      <param name="conference-flags" value="audio-always|livearray-sync|livearray-json-status|video-mute-exit-canvas|video-muxing-personal-canvas"/> <!-- video-bridge-first-two|video-muxing-personal-canvas -->

      <param name="max-members" value="25"/>

      <param name="sound-prefix" value="/usr/local/freeswitch/conf/sounds/"/>

      <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>

      <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>

      <param name="inbound-late-negotiation" value="false"/>

    </profile>

Any pointers on where I'm going wrong would be really appreciated.

Thank you


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From tculjaga at gmail.com  Tue Feb 27 09:52:25 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 27 Feb 2018 10:52:25 +0100
Subject: [Freeswitch-users] test
Message-ID: <CAJoKKdUZ6N9fGiPgnNvquffXgJEL_ykdUShY0H6VPnavyVY5nA@mail.gmail.com>

test123
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From tculjaga at gmail.com  Tue Feb 27 10:50:27 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 27 Feb 2018 11:50:27 +0100
Subject: [Freeswitch-users] NAT / UDP hole punching issue
Message-ID: <CAJoKKdV+YrJ1CuK9FaA5ddT8pf2LyuwGkax4xxLXygdY6hDs4g@mail.gmail.com>

hi,


I have "no audio" issue with TLS and i hope someone could help as Im
getting crazy ... literally :(

my setup is like this:

Phone <> NAT <> INTERNET <> NAT <FreeSWITCH>

FreeSWITCH version: 1.6.12~64bit ( 64bit)

I have a separate profile configured for TLS:

    <param name="rtp-ip" value="192.168.100.60"/>
    <param name="sip-ip" value="192.168.100.60"/>
    <param name="apply-nat-acl" value="rfc1918"/>

    <param name="ext-sip-ip" value="stun:stun.freeswitch.org"/>
    <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/>

    <!--<param name="aggressive-nat-detection" value="true"/>-->

   <param name="tls-only" value="true"/>
   <param name="tls-sip-port" value="15061"/>



=================================================================================================
Name                    tls-public
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_tls-public
Pres Hosts              192.168.100.60,192.168.100.60
Dialplan                XML
Context                 public
Challenge Realm         auto_from
RTP-IP                  192.168.100.60
Ext-RTP-IP              stun:stun.freeswitch.org
SIP-IP                  192.168.100.60
Ext-SIP-IP              85.114.41.180
TLS-URL                 sip:mod_sofia at 85.114.41.180:15061
TLS-BIND-URL            sips:mod_sofia at 85.114.41.180:15061
;maddr=192.168.100.60;transport=tls
WS-BIND-URL             sip:mod_sofia at 192.168.100.60:5066;transport=ws
WSS-BIND-URL            sips:mod_sofia at 192.168.100.60:7443;transport=wss
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               PCMA
CODECS OUT              PCMA
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                true
PROXY-MEDIA             false
ZRTP-PASSTHRU           false
AGGRESSIVENAT           false
CALLS-IN                0
FAILED-CALLS-IN         0
CALLS-OUT               2
FAILED-CALLS-OUT        2
REGISTRATIONS           0



i manage to register the phone with no problems but when i call the phone i
get no audio;

bgapi expand originate ${sofia_contact(tls-profile/agent2/
nexios at 192.168.100.60)} &echo()



FS sends the invite as:


SDP in INVITE message from FS

   v=0
   o=FreeSWITCH 1519708899 1519708900 IN IP4 *85.114.41.180*
   s=FreeSWITCH
   c=IN IP4 *85.114.41.180*
   t=0 0
   m=audio *17480* RTP/AVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20


SIP Client responds with:

   SDP in 200 OK from the client


   v=0
   o=- 3728718779 3728718780 IN IP4 *213.147.96.240*
   s=pjmedia
   b=AS:84
   t=0 0
   a=X-nat:0
   m=audio *4002 *RTP/AVP 8 101
   c=IN IP4 213.147.96.240
   b=TIAS:64000
   a=rtcp:4003 IN IP4 *213.147.96.240*
   a=sendrecv
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16



So the UDP stream is: client( *4002 * ) <> ( *17480* )FS

when i sniff the traffic (on both sides client/FS) using wireshark, i see
RTP packets ( src:4002, dst:17480 ) leaving the client towards FS, but I
don't see RTP packets ( src:17480, dst:4002 ) from FS side leaving towards
the client.


so my question, of course, is why FS is not sending RTP packets to the
IP:PORT notified in 200 OK SDP from the client ? Did i miss some needed
configuration ?


in FS logs i see  *192.168.100.60 port 17480 -> 213.147.96.240 port 4002*
but nothing is actually being sent out from FS

2018-02-27 11:13:02.733376 [DEBUG] sofia.c:6965 Channel
sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 entering state
[ready][200]
2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4311 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4366 Audio Codec
Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4227 Set
telephone-event payload to 101 at 8000
2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:3021 Set Codec
sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 PCMA/8000 20 ms 160
samples 64000 bits 1 channels
2018-02-27 11:13:02.733376 [DEBUG] switch_core_codec.c:111
sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Original read codec
set to PCMA:8
2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4572 Set
telephone-event payload to 101 at 8000
2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4631
sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf send
payload to 101 recv payload to 101
2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6588 AUDIO RTP
[sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551] *192.168.100.60
port 17480 -> 213.147.96.240 port 4002 *codec: 8 ms: 20
2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:3875 Starting timer [soft]
160 bytes per 20ms
2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6806 Activating RTCP
PORT 4003
2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:4261 RTCP send rate is:
5000 and packet rate is: 20000 Remote Port: 4003
2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:2534 Setting RTCP remote
addr to 213.147.96.240:4003 2
2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6887
sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf send
payload to 101
2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6894
sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf
receive payload to 101
2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6917
sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set rtp dtmf delay
to 40
2018-02-27 11:13:02.733376 [NOTICE] sofia.c:8023 Channel
[sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551] has been answered
2018-02-27 11:13:02.733376 [DEBUG] switch_channel.c:3770
(sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) Callstate Change
RINGING -> ACTIVE
2018-02-27 11:13:02.753388 [DEBUG] switch_ivr_originate.c:3686 Originate
Resulted in Success: [sofia/tls-public/sip:agent2/
nexios at 213.147.96.240:10551]
2018-02-27 11:13:02.753388 [INFO] switch_channel.c:3127
sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Flipping CID from
"" <0000000000> to "Outbound Call" <nexios>
2018-02-27 11:13:02.753388 [DEBUG] mod_commands.c:4788
(sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) State Change
CS_CONSUME_MEDIA -> CS_EXECUTE
2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:584
(sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) Running State
Change CS_EXECUTE
2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:650
(sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) State EXECUTE
2018-02-27 11:13:02.753388 [DEBUG] mod_sofia.c:198
sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 SOFIA EXECUTE
2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:328
sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Standard EXECUTE
EXECUTE sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 echo()



Regards,
Tihomir.
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From gmaruzz at gmail.com  Tue Feb 27 11:15:34 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 27 Feb 2018 12:15:34 +0100
Subject: [Freeswitch-users] test
In-Reply-To: <CAJoKKdUZ6N9fGiPgnNvquffXgJEL_ykdUShY0H6VPnavyVY5nA@mail.gmail.com>
References: <CAJoKKdUZ6N9fGiPgnNvquffXgJEL_ykdUShY0H6VPnavyVY5nA@mail.gmail.com>
Message-ID: <CALXCt0onLHwWDt6LevjYsfAmwD_zemPASXBK57GitXUrE_UQKw@mail.gmail.com>

ack


On 27 February 2018 at 10:52, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> test123
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From udy786 at gmail.com  Tue Feb 27 12:33:43 2018
From: udy786 at gmail.com (Uday kumar)
Date: Tue, 27 Feb 2018 18:03:43 +0530
Subject: [Freeswitch-users] Test email. No email received after 23-Feb.
Message-ID: <CAOfqg5YTmMj97pEcT0_6OEQfdqwW8gW6yr1UyJYhUcMOCtcb+w@mail.gmail.com>

Testing email. No email received after 24-Feb.


-- 
Thanks & Regard
Uday.
Mobile:- +91-9377579349
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From cong.wang.itsherpa at gmail.com  Tue Feb 27 12:57:31 2018
From: cong.wang.itsherpa at gmail.com (=?UTF-8?B?546L6IGh?=)
Date: Tue, 27 Feb 2018 12:57:31 +0000
Subject: [Freeswitch-users] Test email. No email received after 23-Feb.
In-Reply-To: <CAOfqg5YTmMj97pEcT0_6OEQfdqwW8gW6yr1UyJYhUcMOCtcb+w@mail.gmail.com>
References: <CAOfqg5YTmMj97pEcT0_6OEQfdqwW8gW6yr1UyJYhUcMOCtcb+w@mail.gmail.com>
Message-ID: <CANhUmQsgcpDHYEsxV+WT8GbHA0SFOL_+P8xy_yL39CZG79RAZw@mail.gmail.com>

pong

Uday kumar <udy786 at gmail.com> 于 2018年2月27日周二 21:35写道：

> Testing email. No email received after 24-Feb.
>
>
> --
> Thanks & Regard
> Uday.
> Mobile:- +91-9377579349
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 francesco at delagarda.com  Mon Feb 26 14:16:46 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Mon, 26 Feb 2018 15:16:46 +0100
Subject: [Freeswitch-users] Clarifications on Grandstream HT503 as FXO
	gateway
Message-ID: <011901d3af0c$705f2bb0$511d8310$@delagarda.com>

Goodmorning to you all!

I am trying to configure a HT503 as a FXO gateway to my freeswitch 1.6

192.168.0.201 is the ip of my HT503

 

I have set the outgoing by creating the following file in
/etc/freeswitch/sip_profiles/external/ht503.xml :

 

<include>

 

                <gateway name="ht503">

                               <param name="realm"
value="192.168.0.201:5062"/>

                               <param name="proxy"
value="192.168.0.201:5062"/>

                               <param name="expire-seconds" value="800"/>

                               <param name="register" value="false"/>

                               <param name="retry-seconds" value="60"/>

                               <param name="context" value="public"/>

                </gateway>

 

</include>

 

And by creating the following in
/etc/freeswitch/dialplan/default/01_custom.xml

 

<include>

                <extension name="messagenet-outbound">

                               <condition field="destination_number"
expression="^\d{8,}$">

                                               <action application="set"
data="effective_caller_id_name=${outbound_caller_id_name}"/>

                                               <action application="set"
data="effective_caller_id_number=${outbound_caller_id_number}"/>

                                               <action application="set"
data="hangup_after_bridge=true"/>

                                               <action application="bridge"
data="sofia/gateway/ht503/${destination_number}@192.168.0.201:5062"/>


                               </condition>

                </extension>

</include>

 

I have 2 Problems:

 

1.	I am using javascript scripting, and I am trying to schedule some
events using:

                var sessOut = new Session("sofia/gateway/ht503/" +
dialedNum + "@192.168.0.201:5062");

                sessOut.execute("set", "execute_on_answer=sched_hangup +20
alloted_timeout")

 

But whereas using a sip provider, the hangup is scheduled correctly, using
the HT it appears that the "execute_on_answer" never happens!

 

2.	Despite reading and searching for days, I have found NO clear
instructions on how to configure the HT503 for INCOMING calls

 

Thank you all in advance

Francesco

 

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From francesco at delagarda.com  Tue Feb 27 07:20:25 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Tue, 27 Feb 2018 08:20:25 +0100
Subject: [Freeswitch-users] Conference mux mode video flickering
In-Reply-To: <CY1PR16MB03161C0EC0B809D96D0B4A5FACC10@CY1PR16MB0316.namprd16.prod.outlook.com>
References: <CY1PR16MB03161C0EC0B809D96D0B4A5FACC10@CY1PR16MB0316.namprd16.prod.outlook.com>
Message-ID: <002101d3af9b$7109b690$531d23b0$@delagarda.com>

Hi there Dom.

I’m very new to FS but I had a hell of a time getting mux to work. In the
end I went through every possible “video-layout-name” I could find (they are
defined in separate xml).

IN the end helped me a lot:

https://books.google.it/books?id=turUDQAAQBAJ
<https://books.google.it/books?id=turUDQAAQBAJ&pg=PA153&lpg=PA153&dq=freeswi
tch+conference+video-mode+mux+layout+zoom&source=bl&ots=Q429H6TX6S&sig=CSI-W
2SCAB8_zBruJ75KdCLB6YM&hl=en&sa=X&ved=0ahUKEwjy2pmCx8XZAhWGKlAKHYTzBTEQ6AEIL
TAB#v=onepage&q=freeswitch%20conference%20video-mode%20mux%20layout%20zoom&f
=false>
&pg=PA153&lpg=PA153&dq=freeswitch+conference+video-mode+mux+layout+zoom&sour
ce=bl&ots=Q429H6TX6S&sig=CSI-W2SCAB8_zBruJ75KdCLB6YM&hl=en&sa=X&ved=0ahUKEwj
y2pmCx8XZAhWGKlAKHYTzBTEQ6AEILTAB#v=onepage&q=freeswitch%20conference%20vide
o-mode%20mux%20layout%20zoom&f=false

Sorry I cant help more.

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On
Behalf Of Dom Rumsey
Sent: lunedì 26 febbraio 2018 21:58
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Conference mux mode video flickering

 

Hi guys

 

Thanks for your help previously. 

 

I'm now using MUX mode for video and I'm getting a problem whereby a
conference with 3 or more participants keeps flickering video streams on the
floor (we are using video-muxing-personal-canvas parameter). It's as if it
can't decide who to show. Below is my conference profile:

 

<profile name="cp">

      <param name="domain" value="$${domain}"/>

      <param name="rate" value="8000"/>

      <param name="video-mode" value="mux"/>

      <param name="video-layout-name" value="1x1"/>

      <param name="interval" value="20"/>

      <param name="caller-controls" value="default"/>

      <param name="energy-level" value="0"/>

      <param name="video-auto-floor-msec" value="3000" />

      <param name="video_no_video_avatar_png"
value="/usr/local/freeswitch/conf/images/video-muted.png" />

      <param name="video_mute_png"
value="/usr/local/freeswitch/conf/images/video-muted.png" />

      <param name="conference-flags"
value="audio-always|livearray-sync|livearray-json-status|video-mute-exit-can
vas|video-muxing-personal-canvas"/> <!--
video-bridge-first-two|video-muxing-personal-canvas -->

      <param name="max-members" value="25"/>

      <param name="sound-prefix"
value="/usr/local/freeswitch/conf/sounds/"/>

      <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>

      <param name="exit-sound"
value="tone_stream://%(500,0,300,200,100,50,25)"/>

      <param name="inbound-late-negotiation" value="false"/>

    </profile>

 

Any pointers on where I'm going wrong would be really appreciated.

 

Thank you

 

 

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From vineet.verma at bics.com  Tue Feb 27 16:10:49 2018
From: vineet.verma at bics.com (vineet)
Date: Tue, 27 Feb 2018 09:10:49 -0700 (MST)
Subject: [Freeswitch-users] bind_local is not working for xmd_cdr.conf
Message-ID: <1519747849804-0.post@n2.nabble.com>

Dears,

I am trying to POST the xml cdr generated by free which to a particular url
defined in “xmd_cdr.conf”
But my http request is getting blocked on firewall. because the source IP
used in while sending the request is not correct.

I tried to change the source IP by modifying the following line in
xml_curl.conf.xml But it did not helped.


Can some please help me to change the source IP while sending the http POST
for xml_cdr?

Thanks,
Vineet




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


From mike at jerris.com  Tue Feb 27 16:24:46 2018
From: mike at jerris.com (Michael Jerris)
Date: Tue, 27 Feb 2018 11:24:46 -0500
Subject: [Freeswitch-users] Conference mux mode video flickering
In-Reply-To: <CY1PR16MB03161C0EC0B809D96D0B4A5FACC10@CY1PR16MB0316.namprd16.prod.outlook.com>
References: <CY1PR16MB03161C0EC0B809D96D0B4A5FACC10@CY1PR16MB0316.namprd16.prod.outlook.com>
Message-ID: <095A5D88-7540-4675-A9C2-F9E0B7028AEB@jerris.com>

The flickering sounds like an old fixed bug we had with overlap and zoom layouts.  Are you using old code?

> On Feb 26, 2018, at 3:57 PM, Dom Rumsey <domrumsey at hotmail.com> wrote:
> 
> Hi guys
> 
> Thanks for your help previously.
> 
> I'm now using MUX mode for video and I'm getting a problem whereby a conference with 3 or more participants keeps flickering video streams on the floor (we are using video-muxing-personal-canvas parameter). It's as if it can't decide who to show. Below is my conference profile:
>  
> <profile name="cp">
>       <param name="domain" value="$${domain}"/>
>       <param name="rate" value="8000"/>
>       <param name="video-mode" value="mux"/>
>       <param name="video-layout-name" value="1x1"/>
>       <param name="interval" value="20"/>
>       <param name="caller-controls" value="default"/>
>       <param name="energy-level" value="0"/>
>       <param name="video-auto-floor-msec" value="3000" />
>       <param name="video_no_video_avatar_png" value="/usr/local/freeswitch/conf/images/video-muted.png" />
>       <param name="video_mute_png" value="/usr/local/freeswitch/conf/images/video-muted.png" />
>       <param name="conference-flags" value="audio-always|livearray-sync|livearray-json-status|video-mute-exit-canvas|video-muxing-personal-canvas"/> <!-- video-bridge-first-two|video-muxing-personal-canvas -->
>       <param name="max-members" value="25"/>
>       <param name="sound-prefix" value="/usr/local/freeswitch/conf/sounds/"/>
>       <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
>       <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
>       <param name="inbound-late-negotiation" value="false"/>
>     </profile>
> 
> Any pointers on where I'm going wrong would be really appreciated.
> 
> Thank you
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.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>
> http://www.freeswitch.org <http://www.freeswitch.org/>
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From domrumsey at hotmail.com  Tue Feb 27 16:53:06 2018
From: domrumsey at hotmail.com (Dom Rumsey)
Date: Tue, 27 Feb 2018 16:53:06 +0000
Subject: [Freeswitch-users] Conference mux mode video flickering
In-Reply-To: <095A5D88-7540-4675-A9C2-F9E0B7028AEB@jerris.com>
References: <CY1PR16MB03161C0EC0B809D96D0B4A5FACC10@CY1PR16MB0316.namprd16.prod.outlook.com>,
 <095A5D88-7540-4675-A9C2-F9E0B7028AEB@jerris.com>
Message-ID: <CY1PR16MB0316954B4FDBEE99FC634D74ACC00@CY1PR16MB0316.namprd16.prod.outlook.com>


Thanks Mike. I'm using 1.6.20.

________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Michael Jerris <mike at jerris.com>
Sent: Tuesday, February 27, 2018 4:24 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conference mux mode video flickering

The flickering sounds like an old fixed bug we had with overlap and zoom layouts.  Are you using old code?

On Feb 26, 2018, at 3:57 PM, Dom Rumsey <domrumsey at hotmail.com<mailto:domrumsey at hotmail.com>> wrote:

Hi guys

Thanks for your help previously.

I'm now using MUX mode for video and I'm getting a problem whereby a conference with 3 or more participants keeps flickering video streams on the floor (we are using video-muxing-personal-canvas parameter). It's as if it can't decide who to show. Below is my conference profile:



<profile name="cp">
      <param name="domain" value="$${domain}"/>
      <param name="rate" value="8000"/>
      <param name="video-mode" value="mux"/>
      <param name="video-layout-name" value="1x1"/>
      <param name="interval" value="20"/>
      <param name="caller-controls" value="default"/>
      <param name="energy-level" value="0"/>
      <param name="video-auto-floor-msec" value="3000" />
      <param name="video_no_video_avatar_png" value="/usr/local/freeswitch/conf/images/video-muted.png" />
      <param name="video_mute_png" value="/usr/local/freeswitch/conf/images/video-muted.png" />
      <param name="conference-flags" value="audio-always|livearray-sync|livearray-json-status|video-mute-exit-canvas|video-muxing-personal-canvas"/> <!-- video-bridge-first-two|video-muxing-personal-canvas -->
      <param name="max-members" value="25"/>
      <param name="sound-prefix" value="/usr/local/freeswitch/conf/sounds/"/>
      <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
      <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
      <param name="inbound-late-negotiation" value="false"/>
    </profile>

Any pointers on where I'm going wrong would be really appreciated.

Thank you


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>

Official FreeSWITCH Sites
http://www.freeswitch.org<http://www.freeswitch.org/>
[https://freeswitch.org/theme/img/freeswitch-social.png]<http://www.freeswitch.org/>

FreeSWITCH<http://www.freeswitch.org/>
www.freeswitch.org
FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications.



http://confluence.freeswitch.org<http://confluence.freeswitch.org/>
http://www.cluecon.com<http://www.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
http://www.freeswitch.org<http://www.freeswitch.org/>

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From mike at jerris.com  Tue Feb 27 17:00:01 2018
From: mike at jerris.com (Michael Jerris)
Date: Tue, 27 Feb 2018 12:00:01 -0500
Subject: [Freeswitch-users] Conference mux mode video flickering
In-Reply-To: <CY1PR16MB0316954B4FDBEE99FC634D74ACC00@CY1PR16MB0316.namprd16.prod.outlook.com>
References: <CY1PR16MB03161C0EC0B809D96D0B4A5FACC10@CY1PR16MB0316.namprd16.prod.outlook.com>
 <095A5D88-7540-4675-A9C2-F9E0B7028AEB@jerris.com>
 <CY1PR16MB0316954B4FDBEE99FC634D74ACC00@CY1PR16MB0316.namprd16.prod.outlook.com>
Message-ID: <FEE1F3A6-81CD-4E88-8472-BB9ECA64DC3E@jerris.com>

only other thing i can think of would be some sort of resource starvation, cpu or network.  Does it do the same without personal canvas?

> On Feb 27, 2018, at 11:53 AM, Dom Rumsey <domrumsey at hotmail.com> wrote:
> 
> 
> Thanks Mike. I'm using 1.6.20.
> 
> From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>> on behalf of Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>
> Sent: Tuesday, February 27, 2018 4:24 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Conference mux mode video flickering
>  
> The flickering sounds like an old fixed bug we had with overlap and zoom layouts.  Are you using old code?
> 
>> On Feb 26, 2018, at 3:57 PM, Dom Rumsey <domrumsey at hotmail.com <mailto:domrumsey at hotmail.com>> wrote:
>> 
>> Hi guys
>> 
>> Thanks for your help previously.
>> 
>> I'm now using MUX mode for video and I'm getting a problem whereby a conference with 3 or more participants keeps flickering video streams on the floor (we are using video-muxing-personal-canvas parameter). It's as if it can't decide who to show. Below is my conference profile:
>>  
>> <profile name="cp">
>>       <param name="domain" value="$${domain}"/>
>>       <param name="rate" value="8000"/>
>>       <param name="video-mode" value="mux"/>
>>       <param name="video-layout-name" value="1x1"/>
>>       <param name="interval" value="20"/>
>>       <param name="caller-controls" value="default"/>
>>       <param name="energy-level" value="0"/>
>>       <param name="video-auto-floor-msec" value="3000" />
>>       <param name="video_no_video_avatar_png" value="/usr/local/freeswitch/conf/images/video-muted.png" />
>>       <param name="video_mute_png" value="/usr/local/freeswitch/conf/images/video-muted.png" />
>>       <param name="conference-flags" value="audio-always|livearray-sync|livearray-json-status|video-mute-exit-canvas|video-muxing-personal-canvas"/> <!-- video-bridge-first-two|video-muxing-personal-canvas -->
>>       <param name="max-members" value="25"/>
>>       <param name="sound-prefix" value="/usr/local/freeswitch/conf/sounds/"/>
>>       <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
>>       <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
>>       <param name="inbound-late-negotiation" value="false"/>
>>     </profile>
>> 
>> Any pointers on where I'm going wrong would be really appreciated.
>> 
>> Thank you
>> 

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From kathleen at freeswitch.com  Tue Feb 27 18:28:49 2018
From: kathleen at freeswitch.com (Kathleen King)
Date: Tue, 27 Feb 2018 10:28:49 -0800
Subject: [Freeswitch-users] Passthrough conference video frame rate
In-Reply-To: <ED21FD45-54F7-4E46-8C4A-AC4E18C7AAE5@tezrosolutions.com>
References: <ED21FD45-54F7-4E46-8C4A-AC4E18C7AAE5@tezrosolutions.com>
Message-ID: <CAOStUPbLJ4nuKToQVUhy6BdA8v_9FVRGt6-KtCNgrN3gUyC7wQ@mail.gmail.com>

Hello,

We have chosen your question to be answered on ClueCon Weekly in the
community corner section. If you would like to call in you can join us
Wednesday at 12:00pm central US time at
https://conference.freeswitch.org/vc/ and dial 888 or watch it on Youtube
at https://youtu.be/1dk1fJ4xAbU


Kathleen King | Public Relations / Administrative Assistant

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

Email: Kathleen at freeswitch.com

Mobile: 703-859-3757

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

[image: color-facebook-96.png] <https://www.facebook.com/freeswitch/> [image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>

On Wed, Feb 21, 2018 at 8:04 AM, Muhammad Umair <umair at tezrosolutions.com>
wrote:

> Hi,
>
> I am wondering what is the video frame rate in conference passthrough
> mode. Any ideas?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 tudor.gabriell at gmail.com  Tue Feb 27 17:37:24 2018
From: tudor.gabriell at gmail.com (tudorgabriell Tudor)
Date: Tue, 27 Feb 2018 19:37:24 +0200
Subject: [Freeswitch-users] Add user to conference from verto communicator
Message-ID: <CAFLFN6SExeaEku1SqnPHv-et0S3+ULwp+Vy4r-V5pF7qdjQH2A@mail.gmail.com>

Hello,

Please tell me if there is a way to add a new participant to conference
from verto communicator?
I would like to dial a number from verto communicator after a conference
has started to add this as a new participant to the conference.

Thank you,
Tudor
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From b631093f-779b-4d67-9ffe-5f6d5b1d3f8a at protonmail.ch  Tue Feb 27 20:53:29 2018
From: b631093f-779b-4d67-9ffe-5f6d5b1d3f8a at protonmail.ch (Bob Smith)
Date: Tue, 27 Feb 2018 15:53:29 -0500
Subject: [Freeswitch-users] Lua question (re: FollowMe Dialplan Example
	from the docs)
In-Reply-To: <a9ae73a4-15a6-e0f2-584b-e1dd6fdd9da7@airmail.cc>
References: <lL7Qbo1q4G4E_xf2PoXnRUk2-JQ-sQ5xRdeP2osSAOBxI-d6pfcuxaSMrNMX35k4Kq14-CwZ0i9JjmNsqiIp-8h3Ib5cTRVRtzN1jGxuzHc=@protonmail.ch>
 <a9ae73a4-15a6-e0f2-584b-e1dd6fdd9da7@airmail.cc>
Message-ID: <BWP7UOpkyET7uQqp1Acxd7x2jquuQNX8V_mda-eps8ToOyDTzy9b4AnZ4m715y2zQRQ74aqHMaFXlbvUewJaB2iArXmkH-S_wMfyZnXkwEs=@protonmail.ch>

A belated thank you to Ryan.   A tweak of the params seems to have done the trick.

‐‐‐‐‐‐‐ Original Message ‐‐‐‐‐‐‐

On February 25, 2018 1:58 PM, Ryan Harris <ryharris at airmail.cc> wrote:

> On 02/22/2018 03:15 PM, Bob Smith wrote:
> 
> > My reading of the script is that it does not take care of the somewhat
> > 
> > obvious problem of what happens if the cell phone goes to voicemail ?
> > 
> > My understanding of the script as presented is that it would just sit
> > 
> > there forever waiting for a reply ?
> 
> I haven't actually used that script, but it shouldn't sit there forever.
> 
> Take a look at what each of the arguments mean:
> 
> https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:playAndGetDigits
> 
> In the example, session:playAndGetDigits will wait 2 seconds for an
> 
> entry and then increment the attempts if unsuccessful. On the third
> 
> unsuccessful attempt, digits will be "". Since digits won't be "1" or
> 
> "2" the script will session:hangup("NO_ANSWER").
> 
> If whatever picks up the call can't dial digits, the call should hangup
> 
> after 6 seconds.
> 
> Professional FreeSWITCH Consulting Services:
> 
> consulting at freeswitch.org
> 
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> 
> http://www.freeswitch.org
> 
> http://confluence.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




From b631093f-779b-4d67-9ffe-5f6d5b1d3f8a at protonmail.ch  Tue Feb 27 20:57:33 2018
From: b631093f-779b-4d67-9ffe-5f6d5b1d3f8a at protonmail.ch (Bob Smith)
Date: Tue, 27 Feb 2018 15:57:33 -0500
Subject: [Freeswitch-users] BLF on Yealink (public "ping" to Kevin
	Wormington).
Message-ID: <NYClqlEUwOALzYCfgM95rI6U1lOVgvQGZycHx52zcaPZ04oQCewAfibwWSaPeQtGL7TEgsvz8eQACPTT3Mbjreu9Dxjr46flVh8oXdpl9tA=@protonmail.ch>

Hi,

I've sent this to list just incase anyone else wants to chip in.

I Googled up an old post from Mr Wormington (August 16 2013 @ 16:09, subject "BLF").

In it, Kevin describes how he has encountered behaviour with Yealink where the BLF remains green despite the subject of the BLF being no longer registered.

There seems to have been very little in public follow up to Kevin's post at the time, so I was wondering if anyone could enlighten me as to what the solution is, because I am facing the same quandry.

Thanks !
Bob
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From w8hdkim at gmail.com  Tue Feb 27 21:02:16 2018
From: w8hdkim at gmail.com (Kim Culhan)
Date: Tue, 27 Feb 2018 16:02:16 -0500
Subject: [Freeswitch-users] Clarifications on Grandstream HT503 as FXO
	gateway
Message-ID: <CAKZxVQVvF04Dh6Q+_RM5HDWNUDYCun5=E5Nz6cebZpxvByc02A@mail.gmail.com>

On Mon, February 26, 2018 9:16 am, Francesco Facco de Lagarda wrote:
> Goodmorning to you all!
>
> I am trying to configure a HT503 as a FXO gateway to my freeswitch 1.6

On the Grandstream configuration web page, take a look at the Basic tab at
the top.

At the bottom the Basic page there is a section:

' *Unconditional Call Forward to VOIP:*

The '503 will initiate a call to the extension number you specify in 'User
ID'
when the FXO rings.

Hope this helps.

-kim
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From fs at szmidt.org  Tue Feb 27 22:07:19 2018
From: fs at szmidt.org (fs)
Date: Tue, 27 Feb 2018 17:07:19 -0500
Subject: [Freeswitch-users] BLF on Yealink (public "ping" to Kevin
 Wormington).
In-Reply-To: <NYClqlEUwOALzYCfgM95rI6U1lOVgvQGZycHx52zcaPZ04oQCewAfibwWSaPeQtGL7TEgsvz8eQACPTT3Mbjreu9Dxjr46flVh8oXdpl9tA=@protonmail.ch>
References: <NYClqlEUwOALzYCfgM95rI6U1lOVgvQGZycHx52zcaPZ04oQCewAfibwWSaPeQtGL7TEgsvz8eQACPTT3Mbjreu9Dxjr46flVh8oXdpl9tA=@protonmail.ch>
Message-ID: <1717f0d4-a3f5-fb93-a17e-d1c1dd6dbd53@szmidt.org>

On 02/27/2018 15:57, Bob Smith wrote:
> Hi,
> 
> I've sent this to list just incase anyone else wants to chip in.
> 
> I Googled up an old post from Mr Wormington (August 16 2013 @ 16:09,
> subject "BLF").
> 
> In it, Kevin describes how he has encountered behaviour with Yealink
> where the BLF remains green despite the subject of the BLF being no
> longer registered.
> 
> There seems to have been very little in public follow up to Kevin's post
> at the time, so I was wondering if anyone could enlighten me as to what
> the solution is, because I am facing the same quandry.

Oh no. I'm just looking to buy the T48G. Which model do you have?


---
fs


From larry.hemenway at gmail.com  Tue Feb 27 23:06:09 2018
From: larry.hemenway at gmail.com (Larry Hemenway)
Date: Tue, 27 Feb 2018 17:06:09 -0600
Subject: [Freeswitch-users] Receiving audio despite SDP with a=sendonly
In-Reply-To: <CAN=Q5SB8SJ+dpn930PPrjc5H--mZRoOHpx8-m9=B_DztsycyUQ@mail.gmail.com>
References: <CAN=Q5SBZ9iyR1HFFxK0GycJFSC=u5aW5=N_bgXV5tsq39DCsvA@mail.gmail.com>
 <CAN=Q5SB8SJ+dpn930PPrjc5H--mZRoOHpx8-m9=B_DztsycyUQ@mail.gmail.com>
Message-ID: <CAN=Q5SDHaaM-vGYZmpO-H62-rCaT24HrhGaOx2qoxF3B5=AUhg@mail.gmail.com>

I'm back to taking a look at this and have a (perhaps) naive view on
how it could be done. I think I could do something like this:

- In the dial plan set a channel variable that indicates the channel
is one-way-audio capable.
- In switch_core_session_write_frame, there would be a conditional
something like this:
        if ((variable_audio_media_flow == recvonly) && (the
one-way-audio-capable variable == true))
           ... do nothing
        else
           ... do work ...
        endif

In this way legacy behavior is preserved, but there is a way to
accommodate one-way audio. With a clever dial plan you could allow it
for some legs and not others.

If anyone has any thoughts on side effects this might have, let me
know. I plan on prototyping it this week if I can find the time.

Larry




On Fri, Feb 23, 2018 at 8:48 AM, Larry Hemenway
<larry.hemenway at gmail.com> wrote:
> It looks like FreeSWITCH does not support one-way audio at all. We have a
> strong need for this functionality if we choose to use FreeSWITCH. If I were
> to investigate and implement this, is it something that the community would
> be interested in? I'm a bit worried about diving into this because if
> FreeSWITCH honors the sdp direction it could potentially break some legacy
> applications that rely on current functionality.
>
> Also, any tips on where to start would be appreciated.
>
> More info on my investigation:
>
> I got a little smarter in how I search the mailing list and came across
> this, so it sounds like this is not supported functionality.
>
> http://freeswitch-users.2379917.n2.nabble.com/sendonly-attribute-ignored-td5933886.html
>
> I also ran some more experiments. The reinvite with a=sendonly results in a
> hold event, so I verified audio is not passed in either direction despite
> the SDP direction attribute.
>
> I'm currently doing these experiments with a conference. We would like to
> use it as a sort of intercom function, but our security group is strongly
> discouraging us from making a product that sends audio to all the endpoints
> even if its silence, so the conference relate API, from their perspective,
> isn't a solution.
>
> Larry
>
>
>
> On Wed, Feb 21, 2018 at 5:09 PM, Larry Hemenway <larry.hemenway at gmail.com>
> wrote:
>>
>> Hello,
>>
>> Is there a way to establish a call with one-way audio on a call from the
>> start?
>>
>> I'm currently sending a request with the following SDP - (note
>> a=sendonly):
>>
>> v=0
>> o=Larry 2890844526 2890844526 IN IP4 127.0.0.1
>> s= My Session
>> c=IN IP4 172.22.112.1
>> t=0 0
>> m=audio 49170 RTP/AVP 0 8
>> a=sendonly
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>>
>> FreeSWITCH is sending back the right response (note the a=recvonly):
>>
>> v=0
>> o=FreeSWITCH 1519171720 1519171721 IN IP4 172.17.0.2
>> s=FreeSWITCH
>> c=IN IP4 172.17.0.2
>> t=0 0
>> m=audio 21768 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=recvonly
>> a=ptime:20
>>
>> But I'm still receiving audio from FreeSWITCH. It is only after I
>> resend the SDP via a reinvite that the audio stops being sent from
>> FreeSWITCH.
>>
>> Larry
>
>


From tculjaga at gmail.com  Wed Feb 28 08:01:23 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 28 Feb 2018 09:01:23 +0100
Subject: [Freeswitch-users] NAT / UDP hole punching issue
In-Reply-To: <CAJoKKdV+YrJ1CuK9FaA5ddT8pf2LyuwGkax4xxLXygdY6hDs4g@mail.gmail.com>
References: <CAJoKKdV+YrJ1CuK9FaA5ddT8pf2LyuwGkax4xxLXygdY6hDs4g@mail.gmail.com>
Message-ID: <CAJoKKdX8y51AUDQHDGLWF98Yt5A2YyAtHwgf+bcoY=fZ7cWT0g@mail.gmail.com>

does anyone have a clue ? :=)

On 27 February 2018 at 11:50, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> hi,
>
>
> I have "no audio" issue with TLS and i hope someone could help as Im
> getting crazy ... literally :(
>
> my setup is like this:
>
> Phone <> NAT <> INTERNET <> NAT <FreeSWITCH>
>
> FreeSWITCH version: 1.6.12~64bit ( 64bit)
>
> I have a separate profile configured for TLS:
>
>     <param name="rtp-ip" value="192.168.100.60"/>
>     <param name="sip-ip" value="192.168.100.60"/>
>     <param name="apply-nat-acl" value="rfc1918"/>
>
>     <param name="ext-sip-ip" value="stun:stun.freeswitch.org"/>
>     <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/>
>
>     <!--<param name="aggressive-nat-detection" value="true"/>-->
>
>    <param name="tls-only" value="true"/>
>    <param name="tls-sip-port" value="15061"/>
>
>
>
> ============================================================
> =====================================
> Name                    tls-public
> Domain Name             N/A
> Auto-NAT                false
> DBName                  sofia_reg_tls-public
> Pres Hosts              192.168.100.60,192.168.100.60
> Dialplan                XML
> Context                 public
> Challenge Realm         auto_from
> RTP-IP                  192.168.100.60
> Ext-RTP-IP              stun:stun.freeswitch.org
> SIP-IP                  192.168.100.60
> Ext-SIP-IP              85.114.41.180
> TLS-URL                 sip:mod_sofia at 85.114.41.180:15061
> TLS-BIND-URL            sips:mod_sofia at 85.114.41.180:
> 15061;maddr=192.168.100.60;transport=tls
> WS-BIND-URL             sip:mod_sofia at 192.168.100.60:5066;transport=ws
> WSS-BIND-URL            sips:mod_sofia at 192.168.100.60:7443;transport=wss
> HOLD-MUSIC              local_stream://moh
> OUTBOUND-PROXY          N/A
> CODECS IN               PCMA
> CODECS OUT              PCMA
> TEL-EVENT               101
> DTMF-MODE               rfc2833
> CNG                     13
> SESSION-TO              0
> MAX-DIALOG              0
> NOMEDIA                 false
> LATE-NEG                true
> PROXY-MEDIA             false
> ZRTP-PASSTHRU           false
> AGGRESSIVENAT           false
> CALLS-IN                0
> FAILED-CALLS-IN         0
> CALLS-OUT               2
> FAILED-CALLS-OUT        2
> REGISTRATIONS           0
>
>
>
> i manage to register the phone with no problems but when i call the phone
> i get no audio;
>
> bgapi expand originate ${sofia_contact(tls-profile/agent2/
> nexios at 192.168.100.60)} &echo()
>
>
>
> FS sends the invite as:
>
>
> SDP in INVITE message from FS
>
>    v=0
>    o=FreeSWITCH 1519708899 1519708900 IN IP4 *85.114.41.180*
>    s=FreeSWITCH
>    c=IN IP4 *85.114.41.180*
>    t=0 0
>    m=audio *17480* RTP/AVP 8 101
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>
>
> SIP Client responds with:
>
>    SDP in 200 OK from the client
>
>
>    v=0
>    o=- 3728718779 3728718780 IN IP4 *213.147.96.240*
>    s=pjmedia
>    b=AS:84
>    t=0 0
>    a=X-nat:0
>    m=audio *4002 *RTP/AVP 8 101
>    c=IN IP4 213.147.96.240
>    b=TIAS:64000
>    a=rtcp:4003 IN IP4 *213.147.96.240*
>    a=sendrecv
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>
>
>
> So the UDP stream is: client( *4002 * ) <> ( *17480* )FS
>
> when i sniff the traffic (on both sides client/FS) using wireshark, i see
> RTP packets ( src:4002, dst:17480 ) leaving the client towards FS, but I
> don't see RTP packets ( src:17480, dst:4002 ) from FS side leaving
> towards the client.
>
>
> so my question, of course, is why FS is not sending RTP packets to the
> IP:PORT notified in 200 OK SDP from the client ? Did i miss some needed
> configuration ?
>
>
> in FS logs i see  *192.168.100.60 port 17480 -> 213.147.96.240 port 4002*
> but nothing is actually being sent out from FS
>
> 2018-02-27 11:13:02.733376 [DEBUG] sofia.c:6965 Channel
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 entering state
> [ready][200]
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4311 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4366 Audio Codec
> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4227 Set
> telephone-event payload to 101 at 8000
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:3021 Set Codec
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 PCMA/8000 20 ms
> 160 samples 64000 bits 1 channels
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_codec.c:111
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Original read
> codec set to PCMA:8
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4572 Set
> telephone-event payload to 101 at 8000
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4631
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf
> send payload to 101 recv payload to 101
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6588 AUDIO RTP
> [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551] *192.168.100.60
> port 17480 -> 213.147.96.240 port 4002 *codec: 8 ms: 20
> 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:3875 Starting timer [soft]
> 160 bytes per 20ms
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6806 Activating
> RTCP PORT 4003
> 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:4261 RTCP send rate is:
> 5000 and packet rate is: 20000 Remote Port: 4003
> 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:2534 Setting RTCP remote
> addr to 213.147.96.240:4003 2
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6887
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf
> send payload to 101
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6894
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf
> receive payload to 101
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6917
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set rtp dtmf
> delay to 40
> 2018-02-27 11:13:02.733376 [NOTICE] sofia.c:8023 Channel
> [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551] has been
> answered
> 2018-02-27 11:13:02.733376 [DEBUG] switch_channel.c:3770
> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) Callstate
> Change RINGING -> ACTIVE
> 2018-02-27 11:13:02.753388 [DEBUG] switch_ivr_originate.c:3686 Originate
> Resulted in Success: [sofia/tls-public/sip:agent2/n
> exios at 213.147.96.240:10551]
> 2018-02-27 11:13:02.753388 [INFO] switch_channel.c:3127
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Flipping CID from
> "" <0000000000> to "Outbound Call" <nexios>
> 2018-02-27 11:13:02.753388 [DEBUG] mod_commands.c:4788
> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) State Change
> CS_CONSUME_MEDIA -> CS_EXECUTE
> 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:584
> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) Running State
> Change CS_EXECUTE
> 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:650
> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) State EXECUTE
> 2018-02-27 11:13:02.753388 [DEBUG] mod_sofia.c:198
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 SOFIA EXECUTE
> 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:328
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Standard EXECUTE
> EXECUTE sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 echo()
>
>
>
> Regards,
> Tihomir.
>
>
>
>
>
>
>
>
>
>
>
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From francesco at delagarda.com  Wed Feb 28 08:18:07 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Wed, 28 Feb 2018 09:18:07 +0100
Subject: [Freeswitch-users] NAT / UDP hole punching issue
In-Reply-To: <CAJoKKdX8y51AUDQHDGLWF98Yt5A2YyAtHwgf+bcoY=fZ7cWT0g@mail.gmail.com>
References: <CAJoKKdV+YrJ1CuK9FaA5ddT8pf2LyuwGkax4xxLXygdY6hDs4g@mail.gmail.com>
 <CAJoKKdX8y51AUDQHDGLWF98Yt5A2YyAtHwgf+bcoY=fZ7cWT0g@mail.gmail.com>
Message-ID: <018801d3b06c$aac462f0$004d28d0$@delagarda.com>

Check your RTP ports .. in the fs config and the port forwarding on firewalls. 

Also, (two cent’s worth), I had a lot of problems with rtp (video and audio) using VErto.. in the end I read that if you don’t specify a stun server, by default it uses google’s.. I don’t know if its applicable in this case, but you never know!

 

Good luck!

F



 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Tihomir Culjaga
Sent: mercoledì 28 febbraio 2018 09:01
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] NAT / UDP hole punching issue

 

does anyone have a clue ? :=)

 

On 27 February 2018 at 11:50, Tihomir Culjaga <tculjaga at gmail.com <mailto:tculjaga at gmail.com> > wrote:

hi, 

 

 

I have "no audio" issue with TLS and i hope someone could help as Im getting crazy ... literally :(

 

my setup is like this:

 

Phone <> NAT <> INTERNET <> NAT <FreeSWITCH>

 

FreeSWITCH version: 1.6.12~64bit ( 64bit)

 

I have a separate profile configured for TLS:

 

    <param name="rtp-ip" value="192.168.100.60"/>

    <param name="sip-ip" value="192.168.100.60"/>

    <param name="apply-nat-acl" value="rfc1918"/>

 

    <param name="ext-sip-ip" value="stun:stun.freeswitch.org <http://stun.freeswitch.org> "/>

    <param name="ext-rtp-ip" value="stun:stun.freeswitch.org <http://stun.freeswitch.org> "/>

 

    <!--<param name="aggressive-nat-detection" value="true"/>-->

 

   <param name="tls-only" value="true"/> 

   <param name="tls-sip-port" value="15061"/>

 

 


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

Name                    tls-public

Domain Name             N/A

Auto-NAT                false

DBName                  sofia_reg_tls-public

Pres Hosts              192.168.100.60,192.168.100.60

Dialplan                XML

Context                 public

Challenge Realm         auto_from

RTP-IP                  192.168.100.60

Ext-RTP-IP              stun:stun.freeswitch.org <http://stun.freeswitch.org> 

SIP-IP                  192.168.100.60

Ext-SIP-IP              85.114.41.180

TLS-URL                 sip:mod_sofia at 85.114.41.180:15061 <http://sip:mod_sofia at 85.114.41.180:15061> 

TLS-BIND-URL            sips:mod_sofia at 85.114.41.180 <mailto:sips%3Amod_sofia at 85.114.41.180> :15061;maddr=192.168.100.60;transport=tls

WS-BIND-URL             sip:mod_sofia at 192.168.100.60 <mailto:sip%3Amod_sofia at 192.168.100.60> :5066;transport=ws

WSS-BIND-URL            sips:mod_sofia at 192.168.100.60 <mailto:sips%3Amod_sofia at 192.168.100.60> :7443;transport=wss

HOLD-MUSIC              local_stream://moh

OUTBOUND-PROXY          N/A

CODECS IN               PCMA

CODECS OUT              PCMA

TEL-EVENT               101

DTMF-MODE               rfc2833

CNG                     13

SESSION-TO              0

MAX-DIALOG              0

NOMEDIA                 false

LATE-NEG                true

PROXY-MEDIA             false

ZRTP-PASSTHRU           false

AGGRESSIVENAT           false

CALLS-IN                0

FAILED-CALLS-IN         0

CALLS-OUT               2

FAILED-CALLS-OUT        2

REGISTRATIONS           0

 

 


i manage to register the phone with no problems but when i call the phone i get no audio;

 

bgapi expand originate ${sofia_contact(tls-profile/agent2/nexios at 192.168.100.60 <mailto:nexios at 192.168.100.60> )} &echo()

 

 

 

FS sends the invite as:

 

 

SDP in INVITE message from FS

   

   v=0

   o=FreeSWITCH 1519708899 1519708900 IN IP4 85.114.41.180

   s=FreeSWITCH

   c=IN IP4 85.114.41.180

   t=0 0

   m=audio 17480 RTP/AVP 8 101

   a=rtpmap:8 PCMA/8000

   a=rtpmap:101 telephone-event/8000

   a=fmtp:101 0-16

   a=ptime:20

 

 

SIP Client responds with:

 

   SDP in 200 OK from the client

 

   

   v=0

   o=- 3728718779 3728718780 IN IP4 213.147.96.240

   s=pjmedia

   b=AS:84

   t=0 0

   a=X-nat:0

   m=audio 4002 RTP/AVP 8 101

   c=IN IP4 213.147.96.240

   b=TIAS:64000

   a=rtcp:4003 IN IP4 213.147.96.240

   a=sendrecv

   a=rtpmap:8 PCMA/8000

   a=rtpmap:101 telephone-event/8000

   a=fmtp:101 0-16

 

 

 

So the UDP stream is: client( 4002  ) <> ( 17480 )FS

 

when i sniff the traffic (on both sides client/FS) using wireshark, i see RTP packets ( src:4002, dst:17480 ) leaving the client towards FS, but I don't see RTP packets ( src:17480, dst:4002 ) from FS side leaving towards the client.

 

 

so my question, of course, is why FS is not sending RTP packets to the IP:PORT notified in 200 OK SDP from the client ? Did i miss some needed configuration ?

 

 

in FS logs i see  192.168.100.60 port 17480 -> 213.147.96.240 port 4002 but nothing is actually being sent out from FS

 

2018-02-27 11:13:02.733376 [DEBUG] sofia.c:6965 Channel sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  entering state [ready][200]

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4311 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4366 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4227 Set telephone-event payload to 101 at 8000

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:3021 Set Codec sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  PCMA/8000 20 ms 160 samples 64000 bits 1 channels

2018-02-27 11:13:02.733376 [DEBUG] switch_core_codec.c:111 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Original read codec set to PCMA:8

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4572 Set telephone-event payload to 101 at 8000

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4631 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Set 2833 dtmf send payload to 101 recv payload to 101

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6588 AUDIO RTP [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ] 192.168.100.60 port 17480 -> 213.147.96.240 port 4002 codec: 8 ms: 20

2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:3875 Starting timer [soft] 160 bytes per 20ms

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6806 Activating RTCP PORT 4003

2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:4261 RTCP send rate is: 5000 and packet rate is: 20000 Remote Port: 4003

2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:2534 Setting RTCP remote addr to 213.147.96.240:4003 <http://213.147.96.240:4003>  2

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6887 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Set 2833 dtmf send payload to 101

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6894 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Set 2833 dtmf receive payload to 101

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6917 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Set rtp dtmf delay to 40

2018-02-27 11:13:02.733376 [NOTICE] sofia.c:8023 Channel [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ] has been answered

2018-02-27 11:13:02.733376 [DEBUG] switch_channel.c:3770 (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ) Callstate Change RINGING -> ACTIVE

2018-02-27 11:13:02.753388 [DEBUG] switch_ivr_originate.c:3686 Originate Resulted in Success: [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ]

2018-02-27 11:13:02.753388 [INFO] switch_channel.c:3127 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Flipping CID from "" <0000000000> to "Outbound Call" <nexios>

2018-02-27 11:13:02.753388 [DEBUG] mod_commands.c:4788 (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ) State Change CS_CONSUME_MEDIA -> CS_EXECUTE

2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:584 (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ) Running State Change CS_EXECUTE

2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:650 (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ) State EXECUTE

2018-02-27 11:13:02.753388 [DEBUG] mod_sofia.c:198 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  SOFIA EXECUTE

2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:328 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Standard EXECUTE

EXECUTE sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  echo()

 

 

 

Regards,

Tihomir.

 

 

 

 

 

 

 

 

 

 

 

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From vineet.verma at bics.com  Wed Feb 28 08:18:56 2018
From: vineet.verma at bics.com (vineet)
Date: Wed, 28 Feb 2018 01:18:56 -0700 (MST)
Subject: [Freeswitch-users] Freeswitch session limit in freeswitch
Message-ID: <1519805936705-0.post@n2.nabble.com>

Dears 
I am experiencing that freeswitch is not able to handle more than 500
sessions even I have configured the switch.conf.xml with 5000 sessions.

Can you please help me ?

Thanks,
vineet

  <settings>
    
    

    
    

    
    

    
    
    


    
    
    
    

    
    

    
    
    
    
    
    




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


From francesco at delagarda.com  Wed Feb 28 08:19:21 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Wed, 28 Feb 2018 09:19:21 +0100
Subject: [Freeswitch-users] Clarifications on Grandstream HT503 as
	FXO	gateway
In-Reply-To: <CAKZxVQVvF04Dh6Q+_RM5HDWNUDYCun5=E5Nz6cebZpxvByc02A@mail.gmail.com>
References: <CAKZxVQVvF04Dh6Q+_RM5HDWNUDYCun5=E5Nz6cebZpxvByc02A@mail.gmail.com>
Message-ID: <019501d3b06c$d6a91c80$83fb5580$@delagarda.com>

Thanks kim!

 

.. and ofc I can set the extension to some group to make multiple phones ring, right? 

 

But do I need to configure the HT503 as an extension?

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Kim Culhan
Sent: martedì 27 febbraio 2018 22:02
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Clarifications on Grandstream HT503 as FXO gateway

 

On Mon, February 26, 2018 9:16 am, Francesco Facco de Lagarda wrote:

> Goodmorning to you all!

> 

> I am trying to configure a HT503 as a FXO gateway to my freeswitch 1.6

 

On the Grandstream configuration web page, take a look at the Basic tab at the top.

 

At the bottom the Basic page there is a section:

 

' Unconditional Call Forward to VOIP:

 

The '503 will initiate a call to the extension number you specify in 'User ID'

when the FXO rings.

 

Hope this helps.

 

-kim

 

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From francesco at delagarda.com  Wed Feb 28 08:42:22 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Wed, 28 Feb 2018 09:42:22 +0100
Subject: [Freeswitch-users] javascript scheduling multiple events
	excecute_on_answer
Message-ID: <01bc01d3b070$0f3b0bf0$2db123d0$@delagarda.com>

I am developing a calling platform using javascript.

 

I have code that calculates how long THAT user is allowed to call THAT
number for.

 

I am trying to schedule 2 events,

1.	1 min before time ends, that plays a message "you have 1 minute left
.."
2.	The actual hangup when the time expires..

 

Despite a zillion tests I have been unable to schedule BOTH events.. 

 

This is my code: for simplicity's sake I have set call time to 120 secs,
with warning at 60:

 

if (session.ready()) {

 

/***

Get user, number, etc.  code omitted for simplicity

**/

 

     var sessOut = new Session("sofia/gateway/ht503/" +  dialedNum +
"@192.168.0.201:5062");

     var totTime = 60;

     sessOut.execute("set", "execute_on_answer=sched_hangup +120
alloted_timeout")

     sessOut.execute("set", "execute_on_answer=sched_broadcast +60
playback::" + soundDir + "one_min_left.wav both");

 

     if (sessOut.ready()) {

          bridge(session, sessOut);

     }

     sessOut.hangup();

     session.hangup();

}

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From tculjaga at gmail.com  Wed Feb 28 08:57:00 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 28 Feb 2018 09:57:00 +0100
Subject: [Freeswitch-users] NAT / UDP hole punching issue
In-Reply-To: <018801d3b06c$aac462f0$004d28d0$@delagarda.com>
References: <CAJoKKdV+YrJ1CuK9FaA5ddT8pf2LyuwGkax4xxLXygdY6hDs4g@mail.gmail.com>
 <CAJoKKdX8y51AUDQHDGLWF98Yt5A2YyAtHwgf+bcoY=fZ7cWT0g@mail.gmail.com>
 <018801d3b06c$aac462f0$004d28d0$@delagarda.com>
Message-ID: <CAJoKKdUfshOJx609KKLyLNZuD0uCUWOoGzPXUPLeMVTKDdhseQ@mail.gmail.com>

RTP ports are defined. When i do a port forward for my RTP range i get my
RTP audio working i guess due to rtp-auto-adjust feature on FS... but it
should work without port forwarding.

here simply FS is not starting to send RTP traffic to the client even if it
notified its public IP:PORT in SDP on 200 OK.

i see FS contacting a STUN server, getting the public IP:PORT and than ...
doesn't send any RTP traffic towards the client... this is what its bugging
me.

T.

On 28 February 2018 at 09:18, Francesco Facco de Lagarda <
francesco at delagarda.com> wrote:

> Check your RTP ports .. in the fs config and the port forwarding on
> firewalls.
>
> Also, (two cent’s worth), I had a lot of problems with rtp (video and
> audio) using VErto.. in the end I read that if you don’t specify a stun
> server, by default it uses google’s.. I don’t know if its applicable in
> this case, but you never know!
>
>
>
> Good luck!
>
> F
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Tihomir Culjaga
> *Sent:* mercoledì 28 febbraio 2018 09:01
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] NAT / UDP hole punching issue
>
>
>
> does anyone have a clue ? :=)
>
>
>
> On 27 February 2018 at 11:50, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
> hi,
>
>
>
>
>
> I have "no audio" issue with TLS and i hope someone could help as Im
> getting crazy ... literally :(
>
>
>
> my setup is like this:
>
>
>
> Phone <> NAT <> INTERNET <> NAT <FreeSWITCH>
>
>
>
> FreeSWITCH version: 1.6.12~64bit ( 64bit)
>
>
>
> I have a separate profile configured for TLS:
>
>
>
>     <param name="rtp-ip" value="192.168.100.60"/>
>
>     <param name="sip-ip" value="192.168.100.60"/>
>
>     <param name="apply-nat-acl" value="rfc1918"/>
>
>
>
>     <param name="ext-sip-ip" value="stun:stun.freeswitch.org"/>
>
>     <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/>
>
>
>
>     <!--<param name="aggressive-nat-detection" value="true"/>-->
>
>
>
>    <param name="tls-only" value="true"/>
>
>    <param name="tls-sip-port" value="15061"/>
>
>
>
>
>
>
> ============================================================
> =====================================
>
> Name                    tls-public
>
> Domain Name             N/A
>
> Auto-NAT                false
>
> DBName                  sofia_reg_tls-public
>
> Pres Hosts              192.168.100.60,192.168.100.60
>
> Dialplan                XML
>
> Context                 public
>
> Challenge Realm         auto_from
>
> RTP-IP                  192.168.100.60
>
> Ext-RTP-IP              stun:stun.freeswitch.org
>
> SIP-IP                  192.168.100.60
>
> Ext-SIP-IP              85.114.41.180
>
> TLS-URL                 sip:mod_sofia at 85.114.41.180:15061
>
> TLS-BIND-URL            sips:mod_sofia at 85.114.41.180:
> 15061;maddr=192.168.100.60;transport=tls
>
> WS-BIND-URL             sip:mod_sofia at 192.168.100.60:5066;transport=ws
>
> WSS-BIND-URL            sips:mod_sofia at 192.168.100.60:7443;transport=wss
>
> HOLD-MUSIC              local_stream://moh
>
> OUTBOUND-PROXY          N/A
>
> CODECS IN               PCMA
>
> CODECS OUT              PCMA
>
> TEL-EVENT               101
>
> DTMF-MODE               rfc2833
>
> CNG                     13
>
> SESSION-TO              0
>
> MAX-DIALOG              0
>
> NOMEDIA                 false
>
> LATE-NEG                true
>
> PROXY-MEDIA             false
>
> ZRTP-PASSTHRU           false
>
> AGGRESSIVENAT           false
>
> CALLS-IN                0
>
> FAILED-CALLS-IN         0
>
> CALLS-OUT               2
>
> FAILED-CALLS-OUT        2
>
> REGISTRATIONS           0
>
>
>
>
>
>
> i manage to register the phone with no problems but when i call the phone
> i get no audio;
>
>
>
> bgapi expand originate ${sofia_contact(tls-profile/agent2/
> nexios at 192.168.100.60)} &echo()
>
>
>
>
>
>
>
> FS sends the invite as:
>
>
>
>
>
> SDP in INVITE message from FS
>
>
>
>    v=0
>
>    o=FreeSWITCH 1519708899 1519708900 IN IP4 *85.114.41.180*
>
>    s=FreeSWITCH
>
>    c=IN IP4 *85.114.41.180*
>
>    t=0 0
>
>    m=audio *17480* RTP/AVP 8 101
>
>    a=rtpmap:8 PCMA/8000
>
>    a=rtpmap:101 telephone-event/8000
>
>    a=fmtp:101 0-16
>
>    a=ptime:20
>
>
>
>
>
> SIP Client responds with:
>
>
>
>    SDP in 200 OK from the client
>
>
>
>
>
>    v=0
>
>    o=- 3728718779 3728718780 IN IP4 *213.147.96.240*
>
>    s=pjmedia
>
>    b=AS:84
>
>    t=0 0
>
>    a=X-nat:0
>
>    m=audio *4002 *RTP/AVP 8 101
>
>    c=IN IP4 213.147.96.240
>
>    b=TIAS:64000
>
>    a=rtcp:4003 IN IP4 *213.147.96.240*
>
>    a=sendrecv
>
>    a=rtpmap:8 PCMA/8000
>
>    a=rtpmap:101 telephone-event/8000
>
>    a=fmtp:101 0-16
>
>
>
>
>
>
>
> So the UDP stream is: client( *4002 * ) <> ( *17480* )FS
>
>
>
> when i sniff the traffic (on both sides client/FS) using wireshark, i see
> RTP packets ( src:4002, dst:17480 ) leaving the client towards FS, but I
> don't see RTP packets ( src:17480, dst:4002 ) from FS side leaving
> towards the client.
>
>
>
>
>
> so my question, of course, is why FS is not sending RTP packets to the
> IP:PORT notified in 200 OK SDP from the client ? Did i miss some needed
> configuration ?
>
>
>
>
>
> in FS logs i see  *192.168.100.60 port 17480 -> 213.147.96.240 port 4002*
> but nothing is actually being sent out from FS
>
>
>
> 2018-02-27 11:13:02.733376 [DEBUG] sofia.c:6965 Channel
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 entering state
> [ready][200]
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4311 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4366 Audio Codec
> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4227 Set
> telephone-event payload to 101 at 8000
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:3021 Set Codec
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 PCMA/8000 20 ms
> 160 samples 64000 bits 1 channels
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_codec.c:111
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Original read
> codec set to PCMA:8
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4572 Set
> telephone-event payload to 101 at 8000
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4631
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf
> send payload to 101 recv payload to 101
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6588 AUDIO RTP
> [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551] *192.168.100.60
> port 17480 -> 213.147.96.240 port 4002 *codec: 8 ms: 20
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:3875 Starting timer [soft]
> 160 bytes per 20ms
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6806 Activating
> RTCP PORT 4003
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:4261 RTCP send rate is:
> 5000 and packet rate is: 20000 Remote Port: 4003
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:2534 Setting RTCP remote
> addr to 213.147.96.240:4003 2
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6887
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf
> send payload to 101
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6894
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf
> receive payload to 101
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6917
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set rtp dtmf
> delay to 40
>
> 2018-02-27 11:13:02.733376 [NOTICE] sofia.c:8023 Channel
> [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551] has been
> answered
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_channel.c:3770
> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) Callstate
> Change RINGING -> ACTIVE
>
> 2018-02-27 11:13:02.753388 [DEBUG] switch_ivr_originate.c:3686 Originate
> Resulted in Success: [sofia/tls-public/sip:agent2/n
> exios at 213.147.96.240:10551]
>
> 2018-02-27 11:13:02.753388 [INFO] switch_channel.c:3127
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Flipping CID from
> "" <0000000000> to "Outbound Call" <nexios>
>
> 2018-02-27 11:13:02.753388 [DEBUG] mod_commands.c:4788
> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) State Change
> CS_CONSUME_MEDIA -> CS_EXECUTE
>
> 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:584
> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) Running State
> Change CS_EXECUTE
>
> 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:650
> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) State EXECUTE
>
> 2018-02-27 11:13:02.753388 [DEBUG] mod_sofia.c:198
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 SOFIA EXECUTE
>
> 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:328
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Standard EXECUTE
>
> EXECUTE sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 echo()
>
>
>
>
>
>
>
> Regards,
>
> Tihomir.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 chandranraviram at gmail.com  Wed Feb 28 09:25:31 2018
From: chandranraviram at gmail.com (Raviram Chandran)
Date: Wed, 28 Feb 2018 14:55:31 +0530
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 140, Issue 93
In-Reply-To: <mailman.484.1519808252.19961.freeswitch-users@lists.freeswitch.org>
References: <mailman.484.1519808252.19961.freeswitch-users@lists.freeswitch.org>
Message-ID: <CADqhVLo5E=zTvTVVhqa0wWW4EGRtzw8t01SPN084G=5YVSj_DQ@mail.gmail.com>

Hi Vineet,

We got our telephony solutions developed by an Indian company (StarTele
Logic) and its working very well, we are running more then 1000 concurrent
calls. I am not sure how they developed that but you should talk them..may
be they can help you out.

All the best.

Ram.

On Wed, Feb 28, 2018 at 2:27 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:

> Send FreeSWITCH-users mailing list submissions to
>         freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
>         freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
>         freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
>    1. Freeswitch session limit in freeswitch (vineet)
>    2. Re: Clarifications on Grandstream HT503 as        FXO     gateway
>       (Francesco Facco de Lagarda)
>    3. javascript scheduling multiple events     excecute_on_answer
>       (Francesco Facco de Lagarda)
>    4. Re: NAT / UDP hole punching issue (Tihomir Culjaga)
>
>
> ---------- Forwarded message ----------
> From: vineet <vineet.verma at bics.com>
> To: freeswitch-users at lists.freeswitch.org
> Cc:
> Bcc:
> Date: Wed, 28 Feb 2018 01:18:56 -0700 (MST)
> Subject: [Freeswitch-users] Freeswitch session limit in freeswitch
> Dears
> I am experiencing that freeswitch is not able to handle more than 500
> sessions even I have configured the switch.conf.xml with 5000 sessions.
>
> Can you please help me ?
>
> Thanks,
> vineet
>
>   <settings>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> --
> Sent from: http://freeswitch-users.2379917.n2.nabble.com/
>
>
>
>
> ---------- Forwarded message ----------
> From: Francesco Facco de Lagarda <francesco at delagarda.com>
> To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Wed, 28 Feb 2018 09:19:21 +0100
> Subject: Re: [Freeswitch-users] Clarifications on Grandstream HT503 as FXO
> gateway
>
> Thanks kim!
>
>
>
> .. and ofc I can set the extension to some group to make multiple phones
> ring, right?
>
>
>
> But do I need to configure the HT503 as an extension?
>
>
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Kim Culhan
> *Sent:* martedì 27 febbraio 2018 22:02
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Clarifications on Grandstream HT503 as
> FXO gateway
>
>
>
> On Mon, February 26, 2018 9:16 am, Francesco Facco de Lagarda wrote:
>
> > Goodmorning to you all!
>
> >
>
> > I am trying to configure a HT503 as a FXO gateway to my freeswitch 1.6
>
>
>
> On the Grandstream configuration web page, take a look at the Basic tab at
> the top.
>
>
>
> At the bottom the Basic page there is a section:
>
>
>
> ' *Unconditional Call Forward to VOIP:*
>
>
>
> The '503 will initiate a call to the extension number you specify in 'User
> ID'
>
> when the FXO rings.
>
>
>
> Hope this helps.
>
>
>
> -kim
>
>
>
>
> ---------- Forwarded message ----------
> From: Francesco Facco de Lagarda <francesco at delagarda.com>
> To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Wed, 28 Feb 2018 09:42:22 +0100
> Subject: [Freeswitch-users] javascript scheduling multiple events
> excecute_on_answer
>
> I am developing a calling platform using javascript.
>
>
>
> I have code that calculates how long THAT user is allowed to call THAT
> number for.
>
>
>
> I am trying to schedule 2 events,
>
>    1. 1 min before time ends, that plays a message “you have 1 minute
>    left ..”
>    2. The actual hangup when the time expires..
>
>
>
> Despite a zillion tests I have been unable to schedule BOTH events..
>
>
>
> This is my code: for simplicity’s sake I have set call time to 120 secs,
> with warning at 60:
>
>
>
> if (session.ready()) {
>
>
>
> /***
>
> Get user, number, etc…  code omitted for simplicity
>
> **/
>
>
>
>      var sessOut = new Session("sofia/gateway/ht503/" +  dialedNum + "@
> 192.168.0.201:5062");
>
>      var totTime = 60;
>
>      sessOut.execute("set", "execute_on_answer=sched_hangup +120
> alloted_timeout")
>
>      sessOut.execute("set", "execute_on_answer=sched_broadcast +60
> playback::" + soundDir + "one_min_left.wav both");
>
>
>
>      if (sessOut.ready()) {
>
>           bridge(session, sessOut);
>
>      }
>
>      sessOut.hangup();
>
>      session.hangup();
>
> }
>
>
> ---------- Forwarded message ----------
> From: Tihomir Culjaga <tculjaga at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Wed, 28 Feb 2018 09:57:00 +0100
> Subject: Re: [Freeswitch-users] NAT / UDP hole punching issue
> RTP ports are defined. When i do a port forward for my RTP range i get my
> RTP audio working i guess due to rtp-auto-adjust feature on FS... but it
> should work without port forwarding.
>
> here simply FS is not starting to send RTP traffic to the client even if
> it notified its public IP:PORT in SDP on 200 OK.
>
> i see FS contacting a STUN server, getting the public IP:PORT and than ...
> doesn't send any RTP traffic towards the client... this is what its bugging
> me.
>
> T.
>
> On 28 February 2018 at 09:18, Francesco Facco de Lagarda <
> francesco at delagarda.com> wrote:
>
>> Check your RTP ports .. in the fs config and the port forwarding on
>> firewalls.
>>
>> Also, (two cent’s worth), I had a lot of problems with rtp (video and
>> audio) using VErto.. in the end I read that if you don’t specify a stun
>> server, by default it uses google’s.. I don’t know if its applicable in
>> this case, but you never know!
>>
>>
>>
>> Good luck!
>>
>> F
>>
>>
>>
>> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
>> Behalf Of *Tihomir Culjaga
>> *Sent:* mercoledì 28 febbraio 2018 09:01
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* Re: [Freeswitch-users] NAT / UDP hole punching issue
>>
>>
>>
>> does anyone have a clue ? :=)
>>
>>
>>
>> On 27 February 2018 at 11:50, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>>
>> hi,
>>
>>
>>
>>
>>
>> I have "no audio" issue with TLS and i hope someone could help as Im
>> getting crazy ... literally :(
>>
>>
>>
>> my setup is like this:
>>
>>
>>
>> Phone <> NAT <> INTERNET <> NAT <FreeSWITCH>
>>
>>
>>
>> FreeSWITCH version: 1.6.12~64bit ( 64bit)
>>
>>
>>
>> I have a separate profile configured for TLS:
>>
>>
>>
>>     <param name="rtp-ip" value="192.168.100.60"/>
>>
>>     <param name="sip-ip" value="192.168.100.60"/>
>>
>>     <param name="apply-nat-acl" value="rfc1918"/>
>>
>>
>>
>>     <param name="ext-sip-ip" value="stun:stun.freeswitch.org"/>
>>
>>     <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/>
>>
>>
>>
>>     <!--<param name="aggressive-nat-detection" value="true"/>-->
>>
>>
>>
>>    <param name="tls-only" value="true"/>
>>
>>    <param name="tls-sip-port" value="15061"/>
>>
>>
>>
>>
>>
>>
>> ============================================================
>> =====================================
>>
>> Name                    tls-public
>>
>> Domain Name             N/A
>>
>> Auto-NAT                false
>>
>> DBName                  sofia_reg_tls-public
>>
>> Pres Hosts              192.168.100.60,192.168.100.60
>>
>> Dialplan                XML
>>
>> Context                 public
>>
>> Challenge Realm         auto_from
>>
>> RTP-IP                  192.168.100.60
>>
>> Ext-RTP-IP              stun:stun.freeswitch.org
>>
>> SIP-IP                  192.168.100.60
>>
>> Ext-SIP-IP              85.114.41.180
>>
>> TLS-URL                 sip:mod_sofia at 85.114.41.180:15061
>>
>> TLS-BIND-URL            sips:mod_sofia at 85.114.41.180:1
>> 5061;maddr=192.168.100.60;transport=tls
>>
>> WS-BIND-URL             sip:mod_sofia at 192.168.100.60:5066;transport=ws
>>
>> WSS-BIND-URL            sips:mod_sofia at 192.168.100.60:7443;transport=wss
>>
>> HOLD-MUSIC              local_stream://moh
>>
>> OUTBOUND-PROXY          N/A
>>
>> CODECS IN               PCMA
>>
>> CODECS OUT              PCMA
>>
>> TEL-EVENT               101
>>
>> DTMF-MODE               rfc2833
>>
>> CNG                     13
>>
>> SESSION-TO              0
>>
>> MAX-DIALOG              0
>>
>> NOMEDIA                 false
>>
>> LATE-NEG                true
>>
>> PROXY-MEDIA             false
>>
>> ZRTP-PASSTHRU           false
>>
>> AGGRESSIVENAT           false
>>
>> CALLS-IN                0
>>
>> FAILED-CALLS-IN         0
>>
>> CALLS-OUT               2
>>
>> FAILED-CALLS-OUT        2
>>
>> REGISTRATIONS           0
>>
>>
>>
>>
>>
>>
>> i manage to register the phone with no problems but when i call the phone
>> i get no audio;
>>
>>
>>
>> bgapi expand originate ${sofia_contact(tls-profile/agent2/
>> nexios at 192.168.100.60)} &echo()
>>
>>
>>
>>
>>
>>
>>
>> FS sends the invite as:
>>
>>
>>
>>
>>
>> SDP in INVITE message from FS
>>
>>
>>
>>    v=0
>>
>>    o=FreeSWITCH 1519708899 1519708900 IN IP4 *85.114.41.180*
>>
>>    s=FreeSWITCH
>>
>>    c=IN IP4 *85.114.41.180*
>>
>>    t=0 0
>>
>>    m=audio *17480* RTP/AVP 8 101
>>
>>    a=rtpmap:8 PCMA/8000
>>
>>    a=rtpmap:101 telephone-event/8000
>>
>>    a=fmtp:101 0-16
>>
>>    a=ptime:20
>>
>>
>>
>>
>>
>> SIP Client responds with:
>>
>>
>>
>>    SDP in 200 OK from the client
>>
>>
>>
>>
>>
>>    v=0
>>
>>    o=- 3728718779 3728718780 IN IP4 *213.147.96.240*
>>
>>    s=pjmedia
>>
>>    b=AS:84
>>
>>    t=0 0
>>
>>    a=X-nat:0
>>
>>    m=audio *4002 *RTP/AVP 8 101
>>
>>    c=IN IP4 213.147.96.240
>>
>>    b=TIAS:64000
>>
>>    a=rtcp:4003 IN IP4 *213.147.96.240*
>>
>>    a=sendrecv
>>
>>    a=rtpmap:8 PCMA/8000
>>
>>    a=rtpmap:101 telephone-event/8000
>>
>>    a=fmtp:101 0-16
>>
>>
>>
>>
>>
>>
>>
>> So the UDP stream is: client( *4002 * ) <> ( *17480* )FS
>>
>>
>>
>> when i sniff the traffic (on both sides client/FS) using wireshark, i see
>> RTP packets ( src:4002, dst:17480 ) leaving the client towards FS, but I
>> don't see RTP packets ( src:17480, dst:4002 ) from FS side leaving
>> towards the client.
>>
>>
>>
>>
>>
>> so my question, of course, is why FS is not sending RTP packets to the
>> IP:PORT notified in 200 OK SDP from the client ? Did i miss some needed
>> configuration ?
>>
>>
>>
>>
>>
>> in FS logs i see  *192.168.100.60 port 17480 -> 213.147.96.240 port 4002*
>> but nothing is actually being sent out from FS
>>
>>
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] sofia.c:6965 Channel
>> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 entering state
>> [ready][200]
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4311 Audio Codec
>> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4366 Audio Codec
>> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4227 Set
>> telephone-event payload to 101 at 8000
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:3021 Set Codec
>> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 PCMA/8000 20 ms
>> 160 samples 64000 bits 1 channels
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_codec.c:111
>> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Original read
>> codec set to PCMA:8
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4572 Set
>> telephone-event payload to 101 at 8000
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4631
>> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf
>> send payload to 101 recv payload to 101
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6588 AUDIO RTP
>> [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551] *192.168.100.60
>> port 17480 -> 213.147.96.240 port 4002 *codec: 8 ms: 20
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:3875 Starting timer
>> [soft] 160 bytes per 20ms
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6806 Activating
>> RTCP PORT 4003
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:4261 RTCP send rate is:
>> 5000 and packet rate is: 20000 Remote Port: 4003
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:2534 Setting RTCP remote
>> addr to 213.147.96.240:4003 2
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6887
>> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf
>> send payload to 101
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6894
>> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf
>> receive payload to 101
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6917
>> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set rtp dtmf
>> delay to 40
>>
>> 2018-02-27 11:13:02.733376 [NOTICE] sofia.c:8023 Channel
>> [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551] has been
>> answered
>>
>> 2018-02-27 11:13:02.733376 [DEBUG] switch_channel.c:3770
>> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) Callstate
>> Change RINGING -> ACTIVE
>>
>> 2018-02-27 11:13:02.753388 [DEBUG] switch_ivr_originate.c:3686 Originate
>> Resulted in Success: [sofia/tls-public/sip:agent2/n
>> exios at 213.147.96.240:10551]
>>
>> 2018-02-27 11:13:02.753388 [INFO] switch_channel.c:3127
>> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Flipping CID
>> from "" <0000000000> to "Outbound Call" <nexios>
>>
>> 2018-02-27 11:13:02.753388 [DEBUG] mod_commands.c:4788
>> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) State Change
>> CS_CONSUME_MEDIA -> CS_EXECUTE
>>
>> 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:584
>> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) Running State
>> Change CS_EXECUTE
>>
>> 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:650
>> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) State EXECUTE
>>
>> 2018-02-27 11:13:02.753388 [DEBUG] mod_sofia.c:198
>> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 SOFIA EXECUTE
>>
>> 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:328
>> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Standard EXECUTE
>>
>> EXECUTE sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 echo()
>>
>>
>>
>>
>>
>>
>>
>> Regards,
>>
>> Tihomir.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing 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 francesco at delagarda.com  Wed Feb 28 09:34:44 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Wed, 28 Feb 2018 10:34:44 +0100
Subject: [Freeswitch-users] NAT / UDP hole punching issue
In-Reply-To: <CAJoKKdUfshOJx609KKLyLNZuD0uCUWOoGzPXUPLeMVTKDdhseQ@mail.gmail.com>
References: <CAJoKKdV+YrJ1CuK9FaA5ddT8pf2LyuwGkax4xxLXygdY6hDs4g@mail.gmail.com>
 <CAJoKKdX8y51AUDQHDGLWF98Yt5A2YyAtHwgf+bcoY=fZ7cWT0g@mail.gmail.com>
 <018801d3b06c$aac462f0$004d28d0$@delagarda.com>
 <CAJoKKdUfshOJx609KKLyLNZuD0uCUWOoGzPXUPLeMVTKDdhseQ@mail.gmail.com>
Message-ID: <01da01d3b077$5fe24ee0$1fa6eca0$@delagarda.com>

Sorry to hear Tihomir.

 

Is there ANYWAY you can test everything locally on same network (i.e. without stun, nat, etc..) maybe the issue isn’t with stun!

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Tihomir Culjaga
Sent: mercoledì 28 febbraio 2018 09:57
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] NAT / UDP hole punching issue

 

RTP ports are defined. When i do a port forward for my RTP range i get my RTP audio working i guess due to rtp-auto-adjust feature on FS... but it should work without port forwarding.

 

here simply FS is not starting to send RTP traffic to the client even if it notified its public IP:PORT in SDP on 200 OK.

 

i see FS contacting a STUN server, getting the public IP:PORT and than ... doesn't send any RTP traffic towards the client... this is what its bugging me.

 

T.

 

On 28 February 2018 at 09:18, Francesco Facco de Lagarda <francesco at delagarda.com <mailto:francesco at delagarda.com> > wrote:

Check your RTP ports .. in the fs config and the port forwarding on firewalls. 

Also, (two cent’s worth), I had a lot of problems with rtp (video and audio) using VErto.. in the end I read that if you don’t specify a stun server, by default it uses google’s.. I don’t know if its applicable in this case, but you never know!

 

Good luck!

F

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> > On Behalf Of Tihomir Culjaga
Sent: mercoledì 28 febbraio 2018 09:01
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: Re: [Freeswitch-users] NAT / UDP hole punching issue

 

does anyone have a clue ? :=)

 

On 27 February 2018 at 11:50, Tihomir Culjaga <tculjaga at gmail.com <mailto:tculjaga at gmail.com> > wrote:

hi, 

 

 

I have "no audio" issue with TLS and i hope someone could help as Im getting crazy ... literally :(

 

my setup is like this:

 

Phone <> NAT <> INTERNET <> NAT <FreeSWITCH>

 

FreeSWITCH version: 1.6.12~64bit ( 64bit)

 

I have a separate profile configured for TLS:

 

    <param name="rtp-ip" value="192.168.100.60"/>

    <param name="sip-ip" value="192.168.100.60"/>

    <param name="apply-nat-acl" value="rfc1918"/>

 

    <param name="ext-sip-ip" value="stun:stun.freeswitch.org <http://stun.freeswitch.org> "/>

    <param name="ext-rtp-ip" value="stun:stun.freeswitch.org <http://stun.freeswitch.org> "/>

 

    <!--<param name="aggressive-nat-detection" value="true"/>-->

 

   <param name="tls-only" value="true"/> 

   <param name="tls-sip-port" value="15061"/>

 

 


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

Name                    tls-public

Domain Name             N/A

Auto-NAT                false

DBName                  sofia_reg_tls-public

Pres Hosts              192.168.100.60,192.168.100.60

Dialplan                XML

Context                 public

Challenge Realm         auto_from

RTP-IP                  192.168.100.60

Ext-RTP-IP              stun:stun.freeswitch.org <http://stun.freeswitch.org> 

SIP-IP                  192.168.100.60

Ext-SIP-IP              85.114.41.180

TLS-URL                 sip:mod_sofia at 85.114.41.180:15061 <http://sip:mod_sofia at 85.114.41.180:15061> 

TLS-BIND-URL            sips:mod_sofia at 85.114.41.180 <mailto:sips%3Amod_sofia at 85.114.41.180> :15061;maddr=192.168.100.60;transport=tls

WS-BIND-URL             sip:mod_sofia at 192.168.100.60 <mailto:sip%3Amod_sofia at 192.168.100.60> :5066;transport=ws

WSS-BIND-URL            sips:mod_sofia at 192.168.100.60 <mailto:sips%3Amod_sofia at 192.168.100.60> :7443;transport=wss

HOLD-MUSIC              local_stream://moh

OUTBOUND-PROXY          N/A

CODECS IN               PCMA

CODECS OUT              PCMA

TEL-EVENT               101

DTMF-MODE               rfc2833

CNG                     13

SESSION-TO              0

MAX-DIALOG              0

NOMEDIA                 false

LATE-NEG                true

PROXY-MEDIA             false

ZRTP-PASSTHRU           false

AGGRESSIVENAT           false

CALLS-IN                0

FAILED-CALLS-IN         0

CALLS-OUT               2

FAILED-CALLS-OUT        2

REGISTRATIONS           0

 

 


i manage to register the phone with no problems but when i call the phone i get no audio;

 

bgapi expand originate ${sofia_contact(tls-profile/agent2/nexios at 192.168.100.60 <mailto:nexios at 192.168.100.60> )} &echo()

 

 

 

FS sends the invite as:

 

 

SDP in INVITE message from FS

   

   v=0

   o=FreeSWITCH 1519708899 1519708900 IN IP4 85.114.41.180

   s=FreeSWITCH

   c=IN IP4 85.114.41.180

   t=0 0

   m=audio 17480 RTP/AVP 8 101

   a=rtpmap:8 PCMA/8000

   a=rtpmap:101 telephone-event/8000

   a=fmtp:101 0-16

   a=ptime:20

 

 

SIP Client responds with:

 

   SDP in 200 OK from the client

 

   

   v=0

   o=- 3728718779 3728718780 IN IP4 213.147.96.240

   s=pjmedia

   b=AS:84

   t=0 0

   a=X-nat:0

   m=audio 4002 RTP/AVP 8 101

   c=IN IP4 213.147.96.240

   b=TIAS:64000

   a=rtcp:4003 IN IP4 213.147.96.240

   a=sendrecv

   a=rtpmap:8 PCMA/8000

   a=rtpmap:101 telephone-event/8000

   a=fmtp:101 0-16

 

 

 

So the UDP stream is: client( 4002  ) <> ( 17480 )FS

 

when i sniff the traffic (on both sides client/FS) using wireshark, i see RTP packets ( src:4002, dst:17480 ) leaving the client towards FS, but I don't see RTP packets ( src:17480, dst:4002 ) from FS side leaving towards the client.

 

 

so my question, of course, is why FS is not sending RTP packets to the IP:PORT notified in 200 OK SDP from the client ? Did i miss some needed configuration ?

 

 

in FS logs i see  192.168.100.60 port 17480 -> 213.147.96.240 port 4002 but nothing is actually being sent out from FS

 

2018-02-27 11:13:02.733376 [DEBUG] sofia.c:6965 Channel sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  entering state [ready][200]

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4311 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4366 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4227 Set telephone-event payload to 101 at 8000

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:3021 Set Codec sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  PCMA/8000 20 ms 160 samples 64000 bits 1 channels

2018-02-27 11:13:02.733376 [DEBUG] switch_core_codec.c:111 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Original read codec set to PCMA:8

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4572 Set telephone-event payload to 101 at 8000

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4631 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Set 2833 dtmf send payload to 101 recv payload to 101

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6588 AUDIO RTP [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ] 192.168.100.60 port 17480 -> 213.147.96.240 port 4002 codec: 8 ms: 20

2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:3875 Starting timer [soft] 160 bytes per 20ms

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6806 Activating RTCP PORT 4003

2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:4261 RTCP send rate is: 5000 and packet rate is: 20000 Remote Port: 4003

2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:2534 Setting RTCP remote addr to 213.147.96.240:4003 <http://213.147.96.240:4003>  2

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6887 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Set 2833 dtmf send payload to 101

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6894 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Set 2833 dtmf receive payload to 101

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6917 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Set rtp dtmf delay to 40

2018-02-27 11:13:02.733376 [NOTICE] sofia.c:8023 Channel [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ] has been answered

2018-02-27 11:13:02.733376 [DEBUG] switch_channel.c:3770 (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ) Callstate Change RINGING -> ACTIVE

2018-02-27 11:13:02.753388 [DEBUG] switch_ivr_originate.c:3686 Originate Resulted in Success: [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ]

2018-02-27 11:13:02.753388 [INFO] switch_channel.c:3127 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Flipping CID from "" <0000000000> to "Outbound Call" <nexios>

2018-02-27 11:13:02.753388 [DEBUG] mod_commands.c:4788 (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ) State Change CS_CONSUME_MEDIA -> CS_EXECUTE

2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:584 (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ) Running State Change CS_EXECUTE

2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:650 (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ) State EXECUTE

2018-02-27 11:13:02.753388 [DEBUG] mod_sofia.c:198 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  SOFIA EXECUTE

2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:328 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Standard EXECUTE

EXECUTE sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  echo()

 

 

 

Regards,

Tihomir.

 

 

 

 

 

 

 

 

 

 

 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org

 

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From francesco at delagarda.com  Wed Feb 28 09:37:01 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Wed, 28 Feb 2018 10:37:01 +0100
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 140, Issue 93
In-Reply-To: <CADqhVLo5E=zTvTVVhqa0wWW4EGRtzw8t01SPN084G=5YVSj_DQ@mail.gmail.com>
References: <mailman.484.1519808252.19961.freeswitch-users@lists.freeswitch.org>
 <CADqhVLo5E=zTvTVVhqa0wWW4EGRtzw8t01SPN084G=5YVSj_DQ@mail.gmail.com>
Message-ID: <01ee01d3b077$b13cb960$13b62c20$@delagarda.com>

Yes, seems like India is the place!

I used ecosmob to my greatest satisfaction , I can give contact details. You can buy 8 hours support for a good price.

My contact is: Krunal Patel krunal.patel at ecosmob.com <mailto:krunal.patel at ecosmob.com> 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Raviram Chandran
Sent: mercoledì 28 febbraio 2018 10:26
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 140, Issue 93

 

Hi Vineet,

 

We got our telephony solutions developed by an Indian company (StarTele Logic) and its working very well, we are running more then 1000 concurrent calls. I am not sure how they developed that but you should talk them..may be they can help you out.

 

All the best.

 

Ram.

 

On Wed, Feb 28, 2018 at 2:27 PM, <freeswitch-users-request at lists.freeswitch.org <mailto:freeswitch-users-request at lists.freeswitch.org> > wrote:

Send FreeSWITCH-users mailing list submissions to
        freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> 

To subscribe or unsubscribe via the World Wide Web, visit
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to
        freeswitch-users-request at lists.freeswitch.org <mailto:freeswitch-users-request at lists.freeswitch.org> 

You can reach the person managing the list at
        freeswitch-users-owner at lists.freeswitch.org <mailto:freeswitch-users-owner at lists.freeswitch.org> 

When replying, please edit your Subject line so it is more specific
than "Re: Contents of FreeSWITCH-users digest..."

Today's Topics:

   1. Freeswitch session limit in freeswitch (vineet)
   2. Re: Clarifications on Grandstream HT503 as        FXO     gateway
      (Francesco Facco de Lagarda)
   3. javascript scheduling multiple events     excecute_on_answer
      (Francesco Facco de Lagarda)
   4. Re: NAT / UDP hole punching issue (Tihomir Culjaga)


---------- Forwarded message ----------
From: vineet <vineet.verma at bics.com <mailto:vineet.verma at bics.com> >
To: freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> 
Cc: 
Bcc: 
Date: Wed, 28 Feb 2018 01:18:56 -0700 (MST)
Subject: [Freeswitch-users] Freeswitch session limit in freeswitch
Dears
I am experiencing that freeswitch is not able to handle more than 500
sessions even I have configured the switch.conf.xml with 5000 sessions.

Can you please help me ?

Thanks,
vineet

  <settings>
































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




---------- Forwarded message ----------
From: Francesco Facco de Lagarda <francesco at delagarda.com <mailto:francesco at delagarda.com> >
To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Cc: 
Bcc: 
Date: Wed, 28 Feb 2018 09:19:21 +0100
Subject: Re: [Freeswitch-users] Clarifications on Grandstream HT503 as FXO gateway

Thanks kim!

 

.. and ofc I can set the extension to some group to make multiple phones ring, right? 

 

But do I need to configure the HT503 as an extension?

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> > On Behalf Of Kim Culhan
Sent: martedì 27 febbraio 2018 22:02
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: Re: [Freeswitch-users] Clarifications on Grandstream HT503 as FXO gateway

 

On Mon, February 26, 2018 9:16 am, Francesco Facco de Lagarda wrote:

> Goodmorning to you all!

> 

> I am trying to configure a HT503 as a FXO gateway to my freeswitch 1.6

 

On the Grandstream configuration web page, take a look at the Basic tab at the top.

 

At the bottom the Basic page there is a section:

 

' Unconditional Call Forward to VOIP:

 

The '503 will initiate a call to the extension number you specify in 'User ID'

when the FXO rings.

 

Hope this helps.

 

-kim

 



---------- Forwarded message ----------
From: Francesco Facco de Lagarda <francesco at delagarda.com <mailto:francesco at delagarda.com> >
To: "'FreeSWITCH Users Help'" <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Cc: 
Bcc: 
Date: Wed, 28 Feb 2018 09:42:22 +0100
Subject: [Freeswitch-users] javascript scheduling multiple events excecute_on_answer

I am developing a calling platform using javascript.

 

I have code that calculates how long THAT user is allowed to call THAT number for.

 

I am trying to schedule 2 events,

1.	1 min before time ends, that plays a message “you have 1 minute left ..”
2.	The actual hangup when the time expires..

 

Despite a zillion tests I have been unable to schedule BOTH events.. 

 

This is my code: for simplicity’s sake I have set call time to 120 secs, with warning at 60:

 

if (session.ready()) {

 

/***

Get user, number, etc…  code omitted for simplicity

**/

 

     var sessOut = new Session("sofia/gateway/ht503/" +  dialedNum + "@192.168.0.201:5062 <http://192.168.0.201:5062> ");

     var totTime = 60;

     sessOut.execute("set", "execute_on_answer=sched_hangup +120 alloted_timeout")

     sessOut.execute("set", "execute_on_answer=sched_broadcast +60 playback::" + soundDir + "one_min_left.wav both");

 

     if (sessOut.ready()) {

          bridge(session, sessOut);

     }

     sessOut.hangup();

     session.hangup();

}



---------- Forwarded message ----------
From: Tihomir Culjaga <tculjaga at gmail.com <mailto:tculjaga at gmail.com> >
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Cc: 
Bcc: 
Date: Wed, 28 Feb 2018 09:57:00 +0100
Subject: Re: [Freeswitch-users] NAT / UDP hole punching issue

RTP ports are defined. When i do a port forward for my RTP range i get my RTP audio working i guess due to rtp-auto-adjust feature on FS... but it should work without port forwarding.

 

here simply FS is not starting to send RTP traffic to the client even if it notified its public IP:PORT in SDP on 200 OK.

 

i see FS contacting a STUN server, getting the public IP:PORT and than ... doesn't send any RTP traffic towards the client... this is what its bugging me.

 

T.

 

On 28 February 2018 at 09:18, Francesco Facco de Lagarda <francesco at delagarda.com <mailto:francesco at delagarda.com> > wrote:

Check your RTP ports .. in the fs config and the port forwarding on firewalls. 

Also, (two cent’s worth), I had a lot of problems with rtp (video and audio) using VErto.. in the end I read that if you don’t specify a stun server, by default it uses google’s.. I don’t know if its applicable in this case, but you never know!

 

Good luck!

F

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> > On Behalf Of Tihomir Culjaga
Sent: mercoledì 28 febbraio 2018 09:01
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: Re: [Freeswitch-users] NAT / UDP hole punching issue

 

does anyone have a clue ? :=)

 

On 27 February 2018 at 11:50, Tihomir Culjaga <tculjaga at gmail.com <mailto:tculjaga at gmail.com> > wrote:

hi, 

 

 

I have "no audio" issue with TLS and i hope someone could help as Im getting crazy ... literally :(

 

my setup is like this:

 

Phone <> NAT <> INTERNET <> NAT <FreeSWITCH>

 

FreeSWITCH version: 1.6.12~64bit ( 64bit)

 

I have a separate profile configured for TLS:

 

    <param name="rtp-ip" value="192.168.100.60"/>

    <param name="sip-ip" value="192.168.100.60"/>

    <param name="apply-nat-acl" value="rfc1918"/>

 

    <param name="ext-sip-ip" value="stun:stun.freeswitch.org <http://stun.freeswitch.org> "/>

    <param name="ext-rtp-ip" value="stun:stun.freeswitch.org <http://stun.freeswitch.org> "/>

 

    <!--<param name="aggressive-nat-detection" value="true"/>-->

 

   <param name="tls-only" value="true"/> 

   <param name="tls-sip-port" value="15061"/>

 

 


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

Name                    tls-public

Domain Name             N/A

Auto-NAT                false

DBName                  sofia_reg_tls-public

Pres Hosts              192.168.100.60,192.168.100.60

Dialplan                XML

Context                 public

Challenge Realm         auto_from

RTP-IP                  192.168.100.60

Ext-RTP-IP              stun:stun.freeswitch.org <http://stun.freeswitch.org> 

SIP-IP                  192.168.100.60

Ext-SIP-IP              85.114.41.180

TLS-URL                 sip:mod_sofia at 85.114.41.180:15061 <http://sip:mod_sofia at 85.114.41.180:15061> 

TLS-BIND-URL            sips:mod_sofia at 85.114.41.180 <mailto:sips%3Amod_sofia at 85.114.41.180> :15061;maddr=192.168.100.60;transport=tls

WS-BIND-URL             sip:mod_sofia at 192.168.100.60 <mailto:sip%3Amod_sofia at 192.168.100.60> :5066;transport=ws

WSS-BIND-URL            sips:mod_sofia at 192.168.100.60 <mailto:sips%3Amod_sofia at 192.168.100.60> :7443;transport=wss

HOLD-MUSIC              local_stream://moh

OUTBOUND-PROXY          N/A

CODECS IN               PCMA

CODECS OUT              PCMA

TEL-EVENT               101

DTMF-MODE               rfc2833

CNG                     13

SESSION-TO              0

MAX-DIALOG              0

NOMEDIA                 false

LATE-NEG                true

PROXY-MEDIA             false

ZRTP-PASSTHRU           false

AGGRESSIVENAT           false

CALLS-IN                0

FAILED-CALLS-IN         0

CALLS-OUT               2

FAILED-CALLS-OUT        2

REGISTRATIONS           0

 

 


i manage to register the phone with no problems but when i call the phone i get no audio;

 

bgapi expand originate ${sofia_contact(tls-profile/agent2/nexios at 192.168.100.60 <mailto:nexios at 192.168.100.60> )} &echo()

 

 

 

FS sends the invite as:

 

 

SDP in INVITE message from FS

   

   v=0

   o=FreeSWITCH 1519708899 1519708900 IN IP4 85.114.41.180

   s=FreeSWITCH

   c=IN IP4 85.114.41.180

   t=0 0

   m=audio 17480 RTP/AVP 8 101

   a=rtpmap:8 PCMA/8000

   a=rtpmap:101 telephone-event/8000

   a=fmtp:101 0-16

   a=ptime:20

 

 

SIP Client responds with:

 

   SDP in 200 OK from the client

 

   

   v=0

   o=- 3728718779 3728718780 IN IP4 213.147.96.240

   s=pjmedia

   b=AS:84

   t=0 0

   a=X-nat:0

   m=audio 4002 RTP/AVP 8 101

   c=IN IP4 213.147.96.240

   b=TIAS:64000

   a=rtcp:4003 IN IP4 213.147.96.240

   a=sendrecv

   a=rtpmap:8 PCMA/8000

   a=rtpmap:101 telephone-event/8000

   a=fmtp:101 0-16

 

 

 

So the UDP stream is: client( 4002  ) <> ( 17480 )FS

 

when i sniff the traffic (on both sides client/FS) using wireshark, i see RTP packets ( src:4002, dst:17480 ) leaving the client towards FS, but I don't see RTP packets ( src:17480, dst:4002 ) from FS side leaving towards the client.

 

 

so my question, of course, is why FS is not sending RTP packets to the IP:PORT notified in 200 OK SDP from the client ? Did i miss some needed configuration ?

 

 

in FS logs i see  192.168.100.60 port 17480 -> 213.147.96.240 port 4002 but nothing is actually being sent out from FS

 

2018-02-27 11:13:02.733376 [DEBUG] sofia.c:6965 Channel sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  entering state [ready][200]

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4311 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4366 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4227 Set telephone-event payload to 101 at 8000

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:3021 Set Codec sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  PCMA/8000 20 ms 160 samples 64000 bits 1 channels

2018-02-27 11:13:02.733376 [DEBUG] switch_core_codec.c:111 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Original read codec set to PCMA:8

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4572 Set telephone-event payload to 101 at 8000

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4631 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Set 2833 dtmf send payload to 101 recv payload to 101

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6588 AUDIO RTP [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ] 192.168.100.60 port 17480 -> 213.147.96.240 port 4002 codec: 8 ms: 20

2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:3875 Starting timer [soft] 160 bytes per 20ms

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6806 Activating RTCP PORT 4003

2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:4261 RTCP send rate is: 5000 and packet rate is: 20000 Remote Port: 4003

2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:2534 Setting RTCP remote addr to 213.147.96.240:4003 <http://213.147.96.240:4003>  2

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6887 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Set 2833 dtmf send payload to 101

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6894 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Set 2833 dtmf receive payload to 101

2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6917 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Set rtp dtmf delay to 40

2018-02-27 11:13:02.733376 [NOTICE] sofia.c:8023 Channel [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ] has been answered

2018-02-27 11:13:02.733376 [DEBUG] switch_channel.c:3770 (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ) Callstate Change RINGING -> ACTIVE

2018-02-27 11:13:02.753388 [DEBUG] switch_ivr_originate.c:3686 Originate Resulted in Success: [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ]

2018-02-27 11:13:02.753388 [INFO] switch_channel.c:3127 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Flipping CID from "" <0000000000> to "Outbound Call" <nexios>

2018-02-27 11:13:02.753388 [DEBUG] mod_commands.c:4788 (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ) State Change CS_CONSUME_MEDIA -> CS_EXECUTE

2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:584 (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ) Running State Change CS_EXECUTE

2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:650 (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551> ) State EXECUTE

2018-02-27 11:13:02.753388 [DEBUG] mod_sofia.c:198 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  SOFIA EXECUTE

2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:328 sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  Standard EXECUTE

EXECUTE sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 <http://nexios at 213.147.96.240:10551>  echo()

 

 

 

Regards,

Tihomir.

 

 

 

 

 

 

 

 

 

 

 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.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
http://www.freeswitch.org

 


_______________________________________________
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
http://www.freeswitch.org

 

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From b631093f-779b-4d67-9ffe-5f6d5b1d3f8a at protonmail.ch  Wed Feb 28 09:55:37 2018
From: b631093f-779b-4d67-9ffe-5f6d5b1d3f8a at protonmail.ch (Bob Smith)
Date: Wed, 28 Feb 2018 04:55:37 -0500
Subject: [Freeswitch-users] BLF on Yealink (public "ping" to Kevin
	Wormington).
In-Reply-To: <1717f0d4-a3f5-fb93-a17e-d1c1dd6dbd53@szmidt.org>
References: <NYClqlEUwOALzYCfgM95rI6U1lOVgvQGZycHx52zcaPZ04oQCewAfibwWSaPeQtGL7TEgsvz8eQACPTT3Mbjreu9Dxjr46flVh8oXdpl9tA=@protonmail.ch>
 <1717f0d4-a3f5-fb93-a17e-d1c1dd6dbd53@szmidt.org>
Message-ID: <D657UY1wKBt0T-9QQK9sEJ1Ell8TTEyzYSdaVTves8quVOnWOKRlFk3o1IhWsTvO6s9CStnsjneDqXJt2l3Qud3RkcBKUrCqX-hC-2aSO14=@protonmail.ch>

I've got a couple, the T46S and the T42S.  Yealink were selling cut-price demo units for a limited period when they launched the T4x series so I took the opportunity ! 

Overall not bad phones, seem to behave themselves and they do the OPUS codec which is an added bonus (limited to a choice of 8 or 16Khz, but still, better than many manufacturers who don't do OPUS at all !).

The Web GUI does suffer from the odd "lost in translation" moment, but nothing too serious, you can normally figure out what its trying to tell you without much of a struggle.

Other than this BLF "issue", the only other bugbear I'm trying to figure out at the moment is how to reprogram the Voicemail button so it can call a shared mailbox (at the moment I'm using a work-around of programming a soft-key).  That might be something that needs to be done via config file rather than GUI .... 

Have not experimented with auto-provisioning yet, that's somewhere next but other priorities have got in the way !


‐‐‐‐‐‐‐ Original Message ‐‐‐‐‐‐‐

> Oh no. I'm just looking to buy the T48G. Which model do you have?
> 
> 
> ---------------------------------------------------------------------
> 
> fs
> 
> Professional FreeSWITCH Consulting Services:
> 
> consulting at freeswitch.org
> 
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> 
> http://www.freeswitch.org
> 
> http://confluence.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




From domrumsey at hotmail.com  Wed Feb 28 10:07:07 2018
From: domrumsey at hotmail.com (Dom Rumsey)
Date: Wed, 28 Feb 2018 10:07:07 +0000
Subject: [Freeswitch-users] Conference mux mode video flickering
In-Reply-To: <FEE1F3A6-81CD-4E88-8472-BB9ECA64DC3E@jerris.com>
References: <CY1PR16MB03161C0EC0B809D96D0B4A5FACC10@CY1PR16MB0316.namprd16.prod.outlook.com>
 <095A5D88-7540-4675-A9C2-F9E0B7028AEB@jerris.com>
 <CY1PR16MB0316954B4FDBEE99FC634D74ACC00@CY1PR16MB0316.namprd16.prod.outlook.com>
 <FEE1F3A6-81CD-4E88-8472-BB9ECA64DC3E@jerris.com>
Message-ID: <CY1PR16MB031610EAA849027514220EE0ACC70@CY1PR16MB0316.namprd16.prod.outlook.com>

No it doesn’t. So in a 3 way chat, for the person speaking, it shows their own video feed back to themself. The other two people (who are listening) see a flicker of the video feeds from both the speaker and the other listener. It's as if there's a paradox, I just can't work out what it is.

Looked at CPU and we still have +70% left. Network is also fine, got loads of bandwidth left.

Thanks
On Feb 27, 2018, at 5:01 PM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
only other thing i can think of would be some sort of resource starvation, cpu or network.  Does it do the same without personal canvas?

On Feb 27, 2018, at 11:53 AM, Dom Rumsey < domrumsey at hotmail.com<mailto:domrumsey at hotmail.com>> wrote:


Thanks Mike. I'm using 1.6.20.

________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> on behalf of Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>>
Sent: Tuesday, February 27, 2018 4:24 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conference mux mode video flickering

The flickering sounds like an old fixed bug we had with overlap and zoom layouts.  Are you using old code?

On Feb 26, 2018, at 3:57 PM, Dom Rumsey < domrumsey at hotmail.com<mailto:domrumsey at hotmail.com>> wrote:

Hi guys

Thanks for your help previously.

I'm now using MUX mode for video and I'm getting   a problem whereby a conference with 3 or more participants keeps flickering video streams on the floor (we are using video-muxing-personal-canvas parameter). It's as if it can't decide who to show. Below is my conference profile:



< profile   name = "cp" >
       < param   name = "domain"   value = "$${domain}" />
       < param   name = "rate"   value = "8000" />
       < param   name = "video-mode"   value = "mux" />
       < param   name = "video-layout-name"   value = "1x1" />
       < param   name = "interval"   value = "20" />
       < param   name = "caller-controls"   value = "default" />
       < param   name = "energy-level"   value = "0" />
       < param   name = "video-auto-floor-msec"   value = "3000"   />
       < param   name = "video_no_video_avatar_png"   value = "/usr/local/freeswitch/conf/images/video-muted.png"   />
       < param   name = "video_mute_png"   value = "/usr/local/freeswitch/conf/images/video-muted.png"   />
       < param   name = "conference-flags"   value = "audio-always|livearray-sync|livearray-json-status|video-mute-exit-canvas|video-muxing-personal-canvas" />   <!-- video-bridge-first-two|video-muxing-personal-canvas -->
       < param   name = "max-members"   value = "25" />
       < param   name = "sound-prefix"   value = "/usr/local/freeswitch/conf/sounds/" />
       < param   name = "enter-sound"   value = "tone_stream://%(200,0,500,600,700)" />
       < param   name = "exit-sound"   value = "tone_stream://%(500,0,300,200,100,50,25)" />
       < param   name = "inbound-late-negotiation"   value = "false" />
     </ profile >

Any pointers on where I'm going wrong would be really appreciated.

Thank you


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From tculjaga at gmail.com  Wed Feb 28 10:11:17 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 28 Feb 2018 11:11:17 +0100
Subject: [Freeswitch-users] NAT / UDP hole punching issue
In-Reply-To: <01da01d3b077$5fe24ee0$1fa6eca0$@delagarda.com>
References: <CAJoKKdV+YrJ1CuK9FaA5ddT8pf2LyuwGkax4xxLXygdY6hDs4g@mail.gmail.com>
 <CAJoKKdX8y51AUDQHDGLWF98Yt5A2YyAtHwgf+bcoY=fZ7cWT0g@mail.gmail.com>
 <018801d3b06c$aac462f0$004d28d0$@delagarda.com>
 <CAJoKKdUfshOJx609KKLyLNZuD0uCUWOoGzPXUPLeMVTKDdhseQ@mail.gmail.com>
 <01da01d3b077$5fe24ee0$1fa6eca0$@delagarda.com>
Message-ID: <CAJoKKdVyqcVsqYehtoqO7FdMRCov-b1Pz+fSpZ4-VQgKPMvXuQ@mail.gmail.com>

Nope, this is a real life scenario and just the fact that it works when i
port forward the RTP range on the FS side proves everything network related
is working just fine.

My question here goes to FS team ....why FS is not sending any RTP packets
to a remote client even when the client advertises the public IP:PORT in
SDP..
Both client and FS uses STUN ( in network capture i see them communicate
with their stun servers respectively ). The client sends RTP toward FS
while FS does not.

how do i debug rtp forwarding on FS itself ... is there any debug i can
turn on ?





On 28 February 2018 at 10:34, Francesco Facco de Lagarda <
francesco at delagarda.com> wrote:

> Sorry to hear Tihomir.
>
>
>
> Is there ANYWAY you can test everything locally on same network (i.e.
> without stun, nat, etc..) maybe the issue isn’t with stun!
>
>
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Tihomir Culjaga
> *Sent:* mercoledì 28 febbraio 2018 09:57
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] NAT / UDP hole punching issue
>
>
>
> RTP ports are defined. When i do a port forward for my RTP range i get my
> RTP audio working i guess due to rtp-auto-adjust feature on FS... but it
> should work without port forwarding.
>
>
>
> here simply FS is not starting to send RTP traffic to the client even if
> it notified its public IP:PORT in SDP on 200 OK.
>
>
>
> i see FS contacting a STUN server, getting the public IP:PORT and than ...
> doesn't send any RTP traffic towards the client... this is what its bugging
> me.
>
>
>
> T.
>
>
>
> On 28 February 2018 at 09:18, Francesco Facco de Lagarda <
> francesco at delagarda.com> wrote:
>
> Check your RTP ports .. in the fs config and the port forwarding on
> firewalls.
>
> Also, (two cent’s worth), I had a lot of problems with rtp (video and
> audio) using VErto.. in the end I read that if you don’t specify a stun
> server, by default it uses google’s.. I don’t know if its applicable in
> this case, but you never know!
>
>
>
> Good luck!
>
> F
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Tihomir Culjaga
> *Sent:* mercoledì 28 febbraio 2018 09:01
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] NAT / UDP hole punching issue
>
>
>
> does anyone have a clue ? :=)
>
>
>
> On 27 February 2018 at 11:50, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
> hi,
>
>
>
>
>
> I have "no audio" issue with TLS and i hope someone could help as Im
> getting crazy ... literally :(
>
>
>
> my setup is like this:
>
>
>
> Phone <> NAT <> INTERNET <> NAT <FreeSWITCH>
>
>
>
> FreeSWITCH version: 1.6.12~64bit ( 64bit)
>
>
>
> I have a separate profile configured for TLS:
>
>
>
>     <param name="rtp-ip" value="192.168.100.60"/>
>
>     <param name="sip-ip" value="192.168.100.60"/>
>
>     <param name="apply-nat-acl" value="rfc1918"/>
>
>
>
>     <param name="ext-sip-ip" value="stun:stun.freeswitch.org"/>
>
>     <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/>
>
>
>
>     <!--<param name="aggressive-nat-detection" value="true"/>-->
>
>
>
>    <param name="tls-only" value="true"/>
>
>    <param name="tls-sip-port" value="15061"/>
>
>
>
>
>
>
> ============================================================
> =====================================
>
> Name                    tls-public
>
> Domain Name             N/A
>
> Auto-NAT                false
>
> DBName                  sofia_reg_tls-public
>
> Pres Hosts              192.168.100.60,192.168.100.60
>
> Dialplan                XML
>
> Context                 public
>
> Challenge Realm         auto_from
>
> RTP-IP                  192.168.100.60
>
> Ext-RTP-IP              stun:stun.freeswitch.org
>
> SIP-IP                  192.168.100.60
>
> Ext-SIP-IP              85.114.41.180
>
> TLS-URL                 sip:mod_sofia at 85.114.41.180:15061
>
> TLS-BIND-URL            sips:mod_sofia at 85.114.41.180:
> 15061;maddr=192.168.100.60;transport=tls
>
> WS-BIND-URL             sip:mod_sofia at 192.168.100.60:5066;transport=ws
>
> WSS-BIND-URL            sips:mod_sofia at 192.168.100.60:7443;transport=wss
>
> HOLD-MUSIC              local_stream://moh
>
> OUTBOUND-PROXY          N/A
>
> CODECS IN               PCMA
>
> CODECS OUT              PCMA
>
> TEL-EVENT               101
>
> DTMF-MODE               rfc2833
>
> CNG                     13
>
> SESSION-TO              0
>
> MAX-DIALOG              0
>
> NOMEDIA                 false
>
> LATE-NEG                true
>
> PROXY-MEDIA             false
>
> ZRTP-PASSTHRU           false
>
> AGGRESSIVENAT           false
>
> CALLS-IN                0
>
> FAILED-CALLS-IN         0
>
> CALLS-OUT               2
>
> FAILED-CALLS-OUT        2
>
> REGISTRATIONS           0
>
>
>
>
>
>
> i manage to register the phone with no problems but when i call the phone
> i get no audio;
>
>
>
> bgapi expand originate ${sofia_contact(tls-profile/agent2/
> nexios at 192.168.100.60)} &echo()
>
>
>
>
>
>
>
> FS sends the invite as:
>
>
>
>
>
> SDP in INVITE message from FS
>
>
>
>    v=0
>
>    o=FreeSWITCH 1519708899 1519708900 IN IP4 *85.114.41.180*
>
>    s=FreeSWITCH
>
>    c=IN IP4 *85.114.41.180*
>
>    t=0 0
>
>    m=audio *17480* RTP/AVP 8 101
>
>    a=rtpmap:8 PCMA/8000
>
>    a=rtpmap:101 telephone-event/8000
>
>    a=fmtp:101 0-16
>
>    a=ptime:20
>
>
>
>
>
> SIP Client responds with:
>
>
>
>    SDP in 200 OK from the client
>
>
>
>
>
>    v=0
>
>    o=- 3728718779 3728718780 IN IP4 *213.147.96.240*
>
>    s=pjmedia
>
>    b=AS:84
>
>    t=0 0
>
>    a=X-nat:0
>
>    m=audio *4002 *RTP/AVP 8 101
>
>    c=IN IP4 213.147.96.240
>
>    b=TIAS:64000
>
>    a=rtcp:4003 IN IP4 *213.147.96.240*
>
>    a=sendrecv
>
>    a=rtpmap:8 PCMA/8000
>
>    a=rtpmap:101 telephone-event/8000
>
>    a=fmtp:101 0-16
>
>
>
>
>
>
>
> So the UDP stream is: client( *4002 * ) <> ( *17480* )FS
>
>
>
> when i sniff the traffic (on both sides client/FS) using wireshark, i see
> RTP packets ( src:4002, dst:17480 ) leaving the client towards FS, but I
> don't see RTP packets ( src:17480, dst:4002 ) from FS side leaving
> towards the client.
>
>
>
>
>
> so my question, of course, is why FS is not sending RTP packets to the
> IP:PORT notified in 200 OK SDP from the client ? Did i miss some needed
> configuration ?
>
>
>
>
>
> in FS logs i see  *192.168.100.60 port 17480 -> 213.147.96.240 port 4002*
> but nothing is actually being sent out from FS
>
>
>
> 2018-02-27 11:13:02.733376 [DEBUG] sofia.c:6965 Channel
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 entering state
> [ready][200]
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4311 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4366 Audio Codec
> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4227 Set
> telephone-event payload to 101 at 8000
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:3021 Set Codec
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 PCMA/8000 20 ms
> 160 samples 64000 bits 1 channels
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_codec.c:111
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Original read
> codec set to PCMA:8
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4572 Set
> telephone-event payload to 101 at 8000
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4631
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf
> send payload to 101 recv payload to 101
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6588 AUDIO RTP
> [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551] *192.168.100.60
> port 17480 -> 213.147.96.240 port 4002 *codec: 8 ms: 20
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:3875 Starting timer [soft]
> 160 bytes per 20ms
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6806 Activating
> RTCP PORT 4003
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:4261 RTCP send rate is:
> 5000 and packet rate is: 20000 Remote Port: 4003
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:2534 Setting RTCP remote
> addr to 213.147.96.240:4003 2
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6887
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf
> send payload to 101
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6894
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set 2833 dtmf
> receive payload to 101
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6917
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Set rtp dtmf
> delay to 40
>
> 2018-02-27 11:13:02.733376 [NOTICE] sofia.c:8023 Channel
> [sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551] has been
> answered
>
> 2018-02-27 11:13:02.733376 [DEBUG] switch_channel.c:3770
> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) Callstate
> Change RINGING -> ACTIVE
>
> 2018-02-27 11:13:02.753388 [DEBUG] switch_ivr_originate.c:3686 Originate
> Resulted in Success: [sofia/tls-public/sip:agent2/n
> exios at 213.147.96.240:10551]
>
> 2018-02-27 11:13:02.753388 [INFO] switch_channel.c:3127
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Flipping CID from
> "" <0000000000> to "Outbound Call" <nexios>
>
> 2018-02-27 11:13:02.753388 [DEBUG] mod_commands.c:4788
> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) State Change
> CS_CONSUME_MEDIA -> CS_EXECUTE
>
> 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:584
> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) Running State
> Change CS_EXECUTE
>
> 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:650
> (sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551) State EXECUTE
>
> 2018-02-27 11:13:02.753388 [DEBUG] mod_sofia.c:198
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 SOFIA EXECUTE
>
> 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:328
> sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 Standard EXECUTE
>
> EXECUTE sofia/tls-public/sip:agent2/nexios at 213.147.96.240:10551 echo()
>
>
>
>
>
>
>
> Regards,
>
> Tihomir.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 shaun.stokes at itec-support.co.uk  Wed Feb 28 13:15:44 2018
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Wed, 28 Feb 2018 13:15:44 +0000
Subject: [Freeswitch-users] Freeswitch session limit in freeswitch
In-Reply-To: <1519805936705-0.post@n2.nabble.com>
References: <1519805936705-0.post@n2.nabble.com>
Message-ID: <1519823930027.59095@itec-support.co.uk>

Hi Vineet,

Have you checked you're not hitting your sessions to second limit?

max-sessions
- Limits the total number of concurrent channels on your FreeSWITCH system.

sessions-per-second
- Throttling mechanism, the switch will only create this many channels at most, per second.

Which version of FreeSWITCH are you using? There was a bug related to this fixed some time ago:
https://freeswitch.org/jira/browse/FS-4670

What are the symptoms when you hit 500 sessions, is the call being rejected if so what does FreeSWITCH respond with (i.e. SIP 486, 582 etc) or does FreeSWITCH not respond?

What do you see in the FreeSWITCH logs when you hit 500 sessions?

If you've recently increased switch.conf.xml to 5000 sessions you would need to restart FreeSWITCH for your changes to take effect.

Thanks,
Shaun
________________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of vineet <vineet.verma at bics.com>
Sent: 28 February 2018 08:18
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Freeswitch session limit in freeswitch

Dears
I am experiencing that freeswitch is not able to handle more than 500
sessions even I have configured the switch.conf.xml with 5000 sessions.

Can you please help me ?

Thanks,
vineet

  <settings>
































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

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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From kaiduanx at yahoo.ca  Wed Feb 28 15:02:10 2018
From: kaiduanx at yahoo.ca (kaiduan xie)
Date: Wed, 28 Feb 2018 15:02:10 +0000 (UTC)
Subject: [Freeswitch-users] DTMF negotiation
References: <510252213.6154862.1519830130424.ref@mail.yahoo.com>
Message-ID: <510252213.6154862.1519830130424@mail.yahoo.com>

Hi,
We are encountering the following DTMF issue,
FreeSwitch adds telephone-event support in INVITE to SBC, SBC replies back without telephone-event support in 200. 
However FreeSwitch still sends out DTMF in RFC 2833 event.
I think FreeSwitch should send out DTMF inband instead of 2833 event, what is the configuration for this?
Many thanks for help,
/Kaiduan
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From asims1979 at hotmail.com  Wed Feb 28 01:32:45 2018
From: asims1979 at hotmail.com (Andrew)
Date: Wed, 28 Feb 2018 01:32:45 +0000
Subject: [Freeswitch-users] Fax Transmit Rate - force to 9600?
Message-ID: <PS2P216MB0035D1FAF69B625B7A582DE5ACC70@PS2P216MB0035.KORP216.PROD.OUTLOOK.COM>

Hi all,


We've been trying to get our fax implementation as reliable as possible using FS; we've had some great results with inbound faxes with the following items set in the incoming dial-plan:


action application ="answer"/>
action application="playback" data="silence_stream://2000"/>
action    set    ignore_early_media=true
action    set    absolute_codec_string='PCMU,PCMA'
action    set    fax_enable_t38=true
action    set    fax_verbose=true
action    set    fax_use_ecm=true
action    set    disable-v17=true
action    set    fax_v17_disabled=true
action    set    fax_disable_v17=true
action    set    fax_enable_t38_request=true
application    rxfax    /tmp/FAX-IN-${uuid}.tif

application    hangup

The one thing that we have identified with this configuration is that 99% of inbound faxes come through fine when the fax transfer rate is negotiated at 9600 bps (as a result of the fax_v17_disabled=true setting).


However, we still see quite a few incoming faxes reported as coming in with a fax transfer rate of 14400 bps, which have a 100% fail rate.


I understand that the two ends need to negotiate a rate; but how do we force (or only offer) the rate of 9600 or lower? My understanding is that the fax_v17_disabled=true setting should cover that? My first thought was that the remote end was forcing the rate at 14400, regardless of what we were sending back in terms of fax capabilities.


Not sure if it helps, but the two error messages we receive when faxes fail due to the 14400 rate are:

- Unexpected DCN while waiting for DCS or DIS, and
- The call dropped prematurely

My thought was, If we could find a way to get all faxes coming in at 9600, we'd be able to improve the reliability.

Looking through the wiki and previous list messages, the fax_disable_v17=true setting seems to be the documented approach, but I'm not sure why we're still seeing faxes come in at a higher rate.

Any advice on this or are we missing something with regards to locking in the lower speed?

cheers,
A

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From mickael at winlux.fr  Wed Feb 28 15:16:27 2018
From: mickael at winlux.fr (Mickael Hubert)
Date: Wed, 28 Feb 2018 16:16:27 +0100
Subject: [Freeswitch-users] Send RTP to external server
In-Reply-To: <CAEJMVkD2AMvYmh2w9sMJWFzbXeJ5nNr2fM9sOttiZmT8BcGSQQ@mail.gmail.com>
References: <CA+fRVSWZxqe3VxmaS8TD0TmbUbmbh-=6NoKcyWVPm_+bGJ8b1A@mail.gmail.com>
 <CAEJMVkD2AMvYmh2w9sMJWFzbXeJ5nNr2fM9sOttiZmT8BcGSQQ@mail.gmail.com>
Message-ID: <CA+fRVSWmhdgW1jS_jiMgc9XvZ_7dtpr1RJOVZgQfDUUrva1cyg@mail.gmail.com>

Hi
thanks a lot for your answer.

But I want to send stream to external server, not record in audio file. i
can use record to "capture" the voice, but to stream it, it's more
complicated ;)

thanks in advance

2018-02-21 0:50 GMT+01:00 Brian West <brian at freeswitch.com>:

> you can already do this without SIPREC in freeswitch. By setting the
> RECORD_READ_ONLY or RECORD_WRITE_ONLY variables.
>
> /b
>
>
> On Fri, Feb 16, 2018 at 10:33 AM, Mickael Hubert <mickael at winlux.fr>
> wrote:
>
>> Hi list,
>> I want to record each call through freeswitch. But i want record only
>> caller (SSRC 1) OR callee (SSRC 2) voice (not both).
>>
>> I read about SIPREC, Jack, etc ... not interesting
>>
>> Do you have a idea for me please ?
>>
>> Thanks in advance
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
>
> --
>
> 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 <(918)%20424-9378>
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
> color-twitter-96.png]
> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 vma at vallimamod.org  Wed Feb 28 17:34:49 2018
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Wed, 28 Feb 2018 18:34:49 +0100
Subject: [Freeswitch-users] Fax Transmit Rate - force to 9600?
In-Reply-To: <PS2P216MB0035D1FAF69B625B7A582DE5ACC70@PS2P216MB0035.KORP216.PROD.OUTLOOK.COM>
References: <PS2P216MB0035D1FAF69B625B7A582DE5ACC70@PS2P216MB0035.KORP216.PROD.OUTLOOK.COM>
Message-ID: <DACBB411-9DCC-4422-88C6-8DC4D443E26D@vallimamod.org>

Hi,

From the error message, it looks like the remote fax drops the call when it sees no match between capabilities (it's the receiving fax who sends the preamble with its capabilities first.)
According to the source code, with v17 disabled, the max bitrate over t38 is 9600, else it's 14400. So I guess you need to enable v17 to communicate with faxes with min bitrates greater than 9600.

Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sipsolutions.fr
.

> On 28 Feb 2018, at 02:32, Andrew <asims1979 at hotmail.com> wrote:
> 
> Hi all,
> 
> We've been trying to get our fax implementation as reliable as possible using FS; we've had some great results with inbound faxes with the following items set in the incoming dial-plan:
> 
> action application ="answer"/>
> action application="playback" data="silence_stream://2000"/>
> action    set    ignore_early_media=true
> action    set    absolute_codec_string='PCMU,PCMA'
> action    set    fax_enable_t38=true
> action    set    fax_verbose=true
> action    set    fax_use_ecm=true
> action    set    disable-v17=true
> action    set    fax_v17_disabled=true
> action    set    fax_disable_v17=true
> action    set    fax_enable_t38_request=true
> application    rxfax    /tmp/FAX-IN-${uuid}.tif
> application    hangup
> 
> The one thing that we have identified with this configuration is that 99% of inbound faxes come through fine when the fax transfer rate is negotiated at 9600 bps (as a result of thefax_v17_disabled=true setting).
> 
> However, we still see quite a few incoming faxes reported as coming in with a fax transfer rate of 14400 bps, which have a 100% fail rate.
> 
> I understand that the two ends need to negotiate a rate; but how do we force (or only offer) the rate of 9600 or lower? My understanding is that the fax_v17_disabled=true setting should cover that? My first thought was that the remote end was forcing the rate at 14400, regardless of what we were sending back in terms of fax capabilities.
> 
> Not sure if it helps, but the two error messages we receive when faxes fail due to the 14400 rate are:
> - Unexpected DCN while waiting for DCS or DIS, and
> - The call dropped prematurely 
> 
> My thought was, If we could find a way to get all faxes coming in at 9600, we'd be able to improve the reliability. 
> 
> Looking through the wiki and previous list messages, the fax_disable_v17=true setting seems to be the documented approach, but I'm not sure why we're still seeing faxes come in at a higher rate.
> 
> Any advice on this or are we missing something with regards to locking in the lower speed?
> 
> cheers,
> A
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.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>
> http://www.freeswitch.org <http://www.freeswitch.org/>
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From mike at jerris.com  Wed Feb 28 17:57:32 2018
From: mike at jerris.com (Michael Jerris)
Date: Wed, 28 Feb 2018 12:57:32 -0500
Subject: [Freeswitch-users] Conference mux mode video flickering
In-Reply-To: <CY1PR16MB031610EAA849027514220EE0ACC70@CY1PR16MB0316.namprd16.prod.outlook.com>
References: <CY1PR16MB03161C0EC0B809D96D0B4A5FACC10@CY1PR16MB0316.namprd16.prod.outlook.com>
 <095A5D88-7540-4675-A9C2-F9E0B7028AEB@jerris.com>
 <CY1PR16MB0316954B4FDBEE99FC634D74ACC00@CY1PR16MB0316.namprd16.prod.outlook.com>
 <FEE1F3A6-81CD-4E88-8472-BB9ECA64DC3E@jerris.com>
 <CY1PR16MB031610EAA849027514220EE0ACC70@CY1PR16MB0316.namprd16.prod.outlook.com>
Message-ID: <35A8243B-2258-4ADE-9CF2-F360F72BCFDA@jerris.com>

look at individual cores, are any maxing out?

> On Feb 28, 2018, at 5:07 AM, Dom Rumsey <domrumsey at hotmail.com> wrote:
> 
> No it doesn’t. So in a 3 way chat, for the person speaking, it shows their own video feed back to themself. The other two people (who are listening) see a flicker of the video feeds from both the speaker and the other listener. It's as if there's a paradox, I just can't work out what it is.
> 
> Looked at CPU and we still have +70% left. Network is also fine, got loads of bandwidth left.
> 
> Thanks
> On Feb 27, 2018, at 5:01 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> only other thing i can think of would be some sort of resource starvation, cpu or network.  Does it do the same without personal canvas?
> 
>> On Feb 27, 2018, at 11:53 AM, Dom Rumsey < domrumsey at hotmail.com <mailto:domrumsey at hotmail.com>> wrote:
>> 
>> 
>> Thanks Mike. I'm using 1.6.20. 
>> 
>> From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>> on behalf of Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>
>> Sent: Tuesday, February 27, 2018 4:24 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Conference mux mode video flickering
>>  
>> The flickering sounds like an old fixed bug we had with overlap and zoom layouts.  Are you using old code?
>> 
>>> On Feb 26, 2018, at 3:57 PM, Dom Rumsey <domrumsey at hotmail.com <mailto:domrumsey at hotmail.com>> wrote:
>>> 
>>> Hi guys
>>> 
>>> Thanks for your help previously.
>>> 
>>> I'm now using MUX mode for video and I'm getting   a problem whereby a conference with 3 or more participants keeps flickering video streams on the floor (we are using video-muxing-personal-canvas parameter). It's as if it can't decide who to show. Below is my conference profile:
>>>  
>>> < profile   name = "cp" >
>>>        < param   name = "domain"   value = "$${domain}" />
>>>        < param   name = "rate"   value = "8000" />
>>>        < param   name = "video-mode"   value = "mux" />
>>>        < param   name = "video-layout-name"   value = "1x1" />
>>>        < param   name = "interval"   value = "20" />
>>>        < param   name = "caller-controls"   value = "default" />
>>>        < param   name = "energy-level"   value = "0" />
>>>        < param   name = "video-auto-floor-msec"   value = "3000"   />
>>>        < param   name = "video_no_video_avatar_png"   value = "/usr/local/freeswitch/conf/images/video-muted.png"   />
>>>        < param   name = "video_mute_png"   value = "/usr/local/freeswitch/conf/images/video-muted.png"   />
>>>        < param   name = "conference-flags"   value = "audio-always|livearray-sync|livearray-json-status|video-mute-exit-canvas|video-muxing-personal-canvas" />   <!-- video-bridge-first-two|video-muxing-personal-canvas -->
>>>        < param   name = "max-members"   value = "25" />
>>>        < param   name = "sound-prefix"   value = "/usr/local/freeswitch/conf/sounds/" />
>>>        < param   name = "enter-sound"   value = "tone_stream://%(200,0,500,600,700)" />
>>>        < param   name = "exit-sound"   value = "tone_stream://%(500,0,300,200,100,50,25)" />
>>>        < param   name = "inbound-late-negotiation"   value = "false" />
>>>      </ profile >
>>> 
>>> Any pointers on where I'm going wrong would be really appreciated.
>>> 
>>> Thank you
>>> 

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From tculjaga at gmail.com  Wed Feb 28 18:47:16 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 28 Feb 2018 19:47:16 +0100
Subject: [Freeswitch-users] Send RTP to external server
In-Reply-To: <CA+fRVSWmhdgW1jS_jiMgc9XvZ_7dtpr1RJOVZgQfDUUrva1cyg@mail.gmail.com>
References: <CA+fRVSWZxqe3VxmaS8TD0TmbUbmbh-=6NoKcyWVPm_+bGJ8b1A@mail.gmail.com>
 <CAEJMVkD2AMvYmh2w9sMJWFzbXeJ5nNr2fM9sOttiZmT8BcGSQQ@mail.gmail.com>
 <CA+fRVSWmhdgW1jS_jiMgc9XvZ_7dtpr1RJOVZgQfDUUrva1cyg@mail.gmail.com>
Message-ID: <CAJoKKdVN9edESx--AyyQ8RY+=CN=90JQ_rLo0oKL7bbP-WZK7Q@mail.gmail.com>

why do you want to stream... isn't it enough just to rsync the file once
the call is finished ?

On 28 February 2018 at 16:16, Mickael Hubert <mickael at winlux.fr> wrote:

> Hi
> thanks a lot for your answer.
>
> But I want to send stream to external server, not record in audio file. i
> can use record to "capture" the voice, but to stream it, it's more
> complicated ;)
>
> thanks in advance
>
> 2018-02-21 0:50 GMT+01:00 Brian West <brian at freeswitch.com>:
>
>> you can already do this without SIPREC in freeswitch. By setting the
>> RECORD_READ_ONLY or RECORD_WRITE_ONLY variables.
>>
>> /b
>>
>>
>> On Fri, Feb 16, 2018 at 10:33 AM, Mickael Hubert <mickael at winlux.fr>
>> wrote:
>>
>>> Hi list,
>>> I want to record each call through freeswitch. But i want record only
>>> caller (SSRC 1) OR callee (SSRC 2) voice (not both).
>>>
>>> I read about SIPREC, Jack, etc ... not interesting
>>>
>>> Do you have a idea for me please ?
>>>
>>> Thanks in advance
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>>
>>
>>
>>
>> --
>>
>> 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 <(918)%20424-9378>
>>
>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>
>> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
>> color-twitter-96.png]
>> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 domrumsey at hotmail.com  Wed Feb 28 19:06:48 2018
From: domrumsey at hotmail.com (Dom Rumsey)
Date: Wed, 28 Feb 2018 19:06:48 +0000
Subject: [Freeswitch-users] Conference mux mode video flickering
In-Reply-To: <35A8243B-2258-4ADE-9CF2-F360F72BCFDA@jerris.com>
References: <CY1PR16MB03161C0EC0B809D96D0B4A5FACC10@CY1PR16MB0316.namprd16.prod.outlook.com>
 <095A5D88-7540-4675-A9C2-F9E0B7028AEB@jerris.com>
 <CY1PR16MB0316954B4FDBEE99FC634D74ACC00@CY1PR16MB0316.namprd16.prod.outlook.com>
 <FEE1F3A6-81CD-4E88-8472-BB9ECA64DC3E@jerris.com>
 <CY1PR16MB031610EAA849027514220EE0ACC70@CY1PR16MB0316.namprd16.prod.outlook.com>
 <35A8243B-2258-4ADE-9CF2-F360F72BCFDA@jerris.com>
Message-ID: <CY1PR16MB0316A5EDBB71A8C3D4E625F2ACC70@CY1PR16MB0316.namprd16.prod.outlook.com>

Nope, not making out the core.

I think it's code related. The speaker sees themselves, which shouldn't happen. Then, it's as if it can't decide if the listeners should be watching the speaker, or the other listener.
On Feb 28, 2018, at 5:59 PM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
look at individual cores, are any maxing out?

On Feb 28, 2018, at 5:07 AM, Dom Rumsey < domrumsey at hotmail.com<mailto:domrumsey at hotmail.com>> wrote:

No it doesn’t. So in a 3 way chat, for the person speaking, it shows their own video feed back to themself. The other two people (who are listening) see a flicker of the video feeds from both the speaker and the other listener. It's as if there's a paradox, I just can't work out what it is.

Looked at CPU and we still have +70% left. Network is also fine, got loads of bandwidth left.

Thanks
On Feb 27, 2018, at 5:01 PM, Michael Jerris < mike at jerris.com<mailto:mike at jerris.com>> wrote:
only other thing i can think of would be some sort of resource starvation, cpu or network.  Does it do the same without personal canvas?

On Feb 27, 2018, at 11:53 AM, Dom Rumsey < domrumsey at hotmail.com<mailto:domrumsey at hotmail.com>> wrote:


Thanks Mike. I'm using 1.6.20.

________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> on behalf of Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>>
Sent: Tuesday, February 27, 2018 4:24 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conference mux mode video flickering

The flickering sounds like an old fixed bug we had with overlap and zoom layouts.  Are you using old code?

On Feb 26, 2018, at 3:57 PM, Dom Rumsey < domrumsey at hotmail.com<mailto:domrumsey at hotmail.com>> wrote:

Hi guys

Thanks for your help previously.

I'm now using MUX mode for video and I'm getting   a problem whereby a conference with 3 or more participants keeps flickering video streams on the floor (we are using video-muxing-personal-canvas parameter). It's as if it can't decide who to show. Below is my conference profile:



< profile   name = "cp" >
       < param   name = "domain"   value = "$${domain}" />
       < param   name = "rate"   value = "8000" />
       < param   name = "video-mode"   value = "mux" />
       < param   name = "video-layout-name"   value = "1x1" />
       < param   name = "interval"   value = "20" />
       < param   name = "caller-controls"   value = "default" />
       < param   name = "energy-level"   value = "0" />
       < param   name = "video-auto-floor-msec"   value = "3000"   />
       < param   name = "video_no_video_avatar_png"   value = "/usr/local/freeswitch/conf/images/video-muted.png"   />
       < param   name = "video_mute_png"   value = "/usr/local/freeswitch/conf/images/video-muted.png"   />
       < param   name = "conference-flags"   value = "audio-always|livearray-sync|livearray-json-status|video-mute-exit-canvas|video-muxing-personal-canvas" />   <!-- video-bridge-first-two|video-muxing-personal-canvas -->
       < param   name = "max-members"   value = "25" />
       < param   name = "sound-prefix"   value = "/usr/local/freeswitch/conf/sounds/" />
       < param   name = "enter-sound"   value = "tone_stream://%(200,0,500,600,700)" />
       < param   name = "exit-sound"   value = "tone_stream://%(500,0,300,200,100,50,25)" />
       < param   name = "inbound-late-negotiation"   value = "false" />
     </ profile >

Any pointers on where I'm going wrong would be really appreciated.

Thank you


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From mickael at winlux.fr  Wed Feb 28 19:53:58 2018
From: mickael at winlux.fr (Mickael Hubert)
Date: Wed, 28 Feb 2018 20:53:58 +0100
Subject: [Freeswitch-users] Send RTP to external server
In-Reply-To: <CAJoKKdVN9edESx--AyyQ8RY+=CN=90JQ_rLo0oKL7bbP-WZK7Q@mail.gmail.com>
References: <CA+fRVSWZxqe3VxmaS8TD0TmbUbmbh-=6NoKcyWVPm_+bGJ8b1A@mail.gmail.com>
 <CAEJMVkD2AMvYmh2w9sMJWFzbXeJ5nNr2fM9sOttiZmT8BcGSQQ@mail.gmail.com>
 <CA+fRVSWmhdgW1jS_jiMgc9XvZ_7dtpr1RJOVZgQfDUUrva1cyg@mail.gmail.com>
 <CAJoKKdVN9edESx--AyyQ8RY+=CN=90JQ_rLo0oKL7bbP-WZK7Q@mail.gmail.com>
Message-ID: <CA+fRVSV6bSfDAfVyxdGXTZ=J2x59X2kYaAZCFi2WTPjaYhnyGg@mail.gmail.com>

Hi,
I must stream the voice to specific app (other server).
This app processes the voice in live, not after end of call.

2018-02-28 19:47 GMT+01:00 Tihomir Culjaga <tculjaga at gmail.com>:

> why do you want to stream... isn't it enough just to rsync the file once
> the call is finished ?
>
> On 28 February 2018 at 16:16, Mickael Hubert <mickael at winlux.fr> wrote:
>
>> Hi
>> thanks a lot for your answer.
>>
>> But I want to send stream to external server, not record in audio file. i
>> can use record to "capture" the voice, but to stream it, it's more
>> complicated ;)
>>
>> thanks in advance
>>
>> 2018-02-21 0:50 GMT+01:00 Brian West <brian at freeswitch.com>:
>>
>>> you can already do this without SIPREC in freeswitch. By setting the
>>> RECORD_READ_ONLY or RECORD_WRITE_ONLY variables.
>>>
>>> /b
>>>
>>>
>>> On Fri, Feb 16, 2018 at 10:33 AM, Mickael Hubert <mickael at winlux.fr>
>>> wrote:
>>>
>>>> Hi list,
>>>> I want to record each call through freeswitch. But i want record only
>>>> caller (SSRC 1) OR callee (SSRC 2) voice (not both).
>>>>
>>>> I read about SIPREC, Jack, etc ... not interesting
>>>>
>>>> Do you have a idea for me please ?
>>>>
>>>> Thanks in advance
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.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/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> 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 <(918)%20424-9378>
>>>
>>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>>
>>> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image:
>>> color-twitter-96.png]
>>> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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 joelists at tm.net.uk  Wed Feb 28 23:33:15 2018
From: joelists at tm.net.uk (Joseph Waite)
Date: Wed, 28 Feb 2018 23:33:15 +0000
Subject: [Freeswitch-users] Cdr files not using correct template in
	master.csv but does if use separate files
Message-ID: <F4FF4C01-309E-4C45-8237-2E60A738F796@tm.net.uk>

Hi Guys

Im wondering if I’m missing something or if this is expected behaviour?

I have 3 different templates and i’m using account code variables in my dial plan to tell FS to use a specific template for certain calls.

If I have master-only set to true then it ignores the templates and all cdr’s use the default template.

If I disable master-only, all cdr’s are written to master.csv and all use the default template, however I also get a file for each account code and this contains only the cdr’s for that account code and they use the correct template.

Surely all cdr’s should use the template that matches the account code if there in the master or not?

Im just wanting to check this before I file a Jira.

FreeSWITCH Version 1.6.19+git~20171025T192126Z~86184a2f73~64bit (git 86184a2 2017-10-25 19:21:26Z 64bit)

I have not tried on latest Master.

Regards
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From asims1979 at hotmail.com  Wed Feb 28 23:39:27 2018
From: asims1979 at hotmail.com (Andrew)
Date: Wed, 28 Feb 2018 23:39:27 +0000
Subject: [Freeswitch-users] Fax Transmit Rate - force to 9600?
In-Reply-To: <DACBB411-9DCC-4422-88C6-8DC4D443E26D@vallimamod.org>
References: <PS2P216MB0035D1FAF69B625B7A582DE5ACC70@PS2P216MB0035.KORP216.PROD.OUTLOOK.COM>,
 <DACBB411-9DCC-4422-88C6-8DC4D443E26D@vallimamod.org>
Message-ID: <PSXP216MB00391A8B381090E0D02CA910ACC70@PSXP216MB0039.KORP216.PROD.OUTLOOK.COM>

Thanks Vallimamod, that gives me some additional info we weren't aware of (re: preamble and t38 max bitrate).


I noticed you mentioned disabling v17; which would allow the bitrate to increase to 14400, which is where we're seeing the problem. We want to force/limit the max bitrate under all circumstances (t38 or t30 over g711) to 9600, which we know works.

Some additional info;

Doing some further digging; the problem doesn't appear to specifically lie with t38; from what we can see some fax calls come in via t30 over g711 which also fail with a bit rate of 14400.

Given the preamble info you mentioned, could it be we're sending the sending fax our capabilities (i.e. max bitrate = 9600) and it's ignoring this and sends it across at 14400? I would assume that most, if not all faxes have the capability to send at 9600 if requested?

Given our issue relates to incoming (i.e we're the receiving end), is there anything else we can do within FS to force the lower bit rate?

cheers,
A



________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Vallimamod Abdullah <vma at vallimamod.org>
Sent: Thursday, 1 March 2018 4:34 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Fax Transmit Rate - force to 9600?

Hi,

>From the error message, it looks like the remote fax drops the call when it sees no match between capabilities (it's the receiving fax who sends the preamble with its capabilities first.)
According to the source code, with v17 disabled, the max bitrate over t38 is 9600, else it's 14400. So I guess you need to enable v17 to communicate with faxes with min bitrates greater than 9600.

Best Regards,
--
Vallimamod Abdullah
SIP Solutions
vma at sipsolutions.fr<mailto:vma at sipsolutions.fr>
.

On 28 Feb 2018, at 02:32, Andrew <asims1979 at hotmail.com<mailto:asims1979 at hotmail.com>> wrote:

Hi all,

We've been trying to get our fax implementation as reliable as possible using FS; we've had some great results with inbound faxes with the following items set in the incoming dial-plan:


action application ="answer"/>
action application="playback" data="silence_stream://2000"/>
action    set    ignore_early_media=true
action    set    absolute_codec_string='PCMU,PCMA'
action    set    fax_enable_t38=true
action    set    fax_verbose=true
action    set    fax_use_ecm=true
action    set    disable-v17=true
action    set    fax_v17_disabled=true
action    set    fax_disable_v17=true
action    set    fax_enable_t38_request=true
application    rxfax    /tmp/FAX-IN-${uuid}.tif

application    hangup

The one thing that we have identified with this configuration is that 99% of inbound faxes come through fine when the fax transfer rate is negotiated at 9600 bps (as a result of thefax_v17_disabled=true setting).

However, we still see quite a few incoming faxes reported as coming in with a fax transfer rate of 14400 bps, which have a 100% fail rate.

I understand that the two ends need to negotiate a rate; but how do we force (or only offer) the rate of 9600 or lower? My understanding is that the fax_v17_disabled=true setting should cover that? My first thought was that the remote end was forcing the rate at 14400, regardless of what we were sending back in terms of fax capabilities.

Not sure if it helps, but the two error messages we receive when faxes fail due to the 14400 rate are:

- Unexpected DCN while waiting for DCS or DIS, and
- The call dropped prematurely

My thought was, If we could find a way to get all faxes coming in at 9600, we'd be able to improve the reliability.

Looking through the wiki and previous list messages, the fax_disable_v17=true setting seems to be the documented approach, but I'm not sure why we're still seeing faxes come in at a higher rate.

Any advice on this or are we missing something with regards to locking in the lower speed?

cheers,
A

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>

Official FreeSWITCH Sites
http://www.freeswitch.org<http://www.freeswitch.org/>
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FreeSWITCH<http://www.freeswitch.org/>
www.freeswitch.org
FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications.


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From mike at jerris.com  Wed Feb 28 23:53:45 2018
From: mike at jerris.com (Michael Jerris)
Date: Wed, 28 Feb 2018 18:53:45 -0500
Subject: [Freeswitch-users] Fax Transmit Rate - force to 9600?
In-Reply-To: <PSXP216MB00391A8B381090E0D02CA910ACC70@PSXP216MB0039.KORP216.PROD.OUTLOOK.COM>
References: <PS2P216MB0035D1FAF69B625B7A582DE5ACC70@PS2P216MB0035.KORP216.PROD.OUTLOOK.COM>
 <DACBB411-9DCC-4422-88C6-8DC4D443E26D@vallimamod.org>
 <PSXP216MB00391A8B381090E0D02CA910ACC70@PSXP216MB0039.KORP216.PROD.OUTLOOK.COM>
Message-ID: <48681EF6-1692-415D-B525-4CF366820DA7@jerris.com>

its not uncommon for 14400 faxes to fail over g711.  should have better luck with t38.

> On Feb 28, 2018, at 6:39 PM, Andrew <asims1979 at hotmail.com> wrote:
> 
> Thanks Vallimamod, that gives me some additional info we weren't aware of (re: preamble and t38 max bitrate).
> 
> I noticed you mentioned disabling v17; which would allow the bitrate to increase to 14400, which is where we're seeing the problem. We want to force/limit the max bitrate under all circumstances (t38 or t30 over g711) to 9600, which we know works.
> 
> Some additional info;
> 
> Doing some further digging; the problem doesn't appear to specifically lie with t38; from what we can see some fax calls come in via t30 over g711 which also fail with a bit rate of 14400. 
> 
> Given the preamble info you mentioned, could it be we're sending the sending fax our capabilities (i.e. max bitrate = 9600) and it's ignoring this and sends it across at 14400? I would assume that most, if not all faxes have the capability to send at 9600 if requested?
> 
> Given our issue relates to incoming (i.e we're the receiving end), is there anything else we can do within FS to force the lower bit rate?
> 
> cheers,
> A
> 
> 
>  
> From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>> on behalf of Vallimamod Abdullah <vma at vallimamod.org <mailto:vma at vallimamod.org>>
> Sent: Thursday, 1 March 2018 4:34 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Fax Transmit Rate - force to 9600?
>  
> Hi,
> 
> From the error message, it looks like the remote fax drops the call when it sees no match between capabilities (it's the receiving fax who sends the preamble with its capabilities first.)
> According to the source code, with v17 disabled, the max bitrate over t38 is 9600, else it's 14400. So I guess you need to enable v17 to communicate with faxes with min bitrates greater than 9600.
> 
> Best Regards,
> -- 
> Vallimamod Abdullah
> SIP Solutions
> vma at sipsolutions.fr <mailto:vma at sipsolutions.fr>
> .
> 
>> On 28 Feb 2018, at 02:32, Andrew <asims1979 at hotmail.com <mailto:asims1979 at hotmail.com>> wrote:
>> 
>> Hi all,
>> 
>> We've been trying to get our fax implementation as reliable as possible using FS; we've had some great results with inbound faxes with the following items set in the incoming dial-plan:
>> 
>> action application ="answer"/>
>> action application="playback" data="silence_stream://2000"/>
>> action    set    ignore_early_media=true
>> action    set    absolute_codec_string='PCMU,PCMA'
>> action    set    fax_enable_t38=true
>> action    set    fax_verbose=true
>> action    set    fax_use_ecm=true
>> action    set    disable-v17=true
>> action    set    fax_v17_disabled=true
>> action    set    fax_disable_v17=true
>> action    set    fax_enable_t38_request=true
>> application    rxfax    /tmp/FAX-IN-${uuid}.tif
>> application    hangup
>> 
>> The one thing that we have identified with this configuration is that 99% of inbound faxes come through fine when the fax transfer rate is negotiated at 9600 bps (as a result of thefax_v17_disabled=true setting).
>> 
>> However, we still see quite a few incoming faxes reported as coming in with a fax transfer rate of 14400 bps, which have a 100% fail rate.
>> 
>> I understand that the two ends need to negotiate a rate; but how do we force (or only offer) the rate of 9600 or lower? My understanding is that the fax_v17_disabled=true setting should cover that? My first thought was that the remote end was forcing the rate at 14400, regardless of what we were sending back in terms of fax capabilities.
>> 
>> Not sure if it helps, but the two error messages we receive when faxes fail due to the 14400 rate are:
>> - Unexpected DCN while waiting for DCS or DIS, and
>> - The call dropped prematurely 
>> 
>> My thought was, If we could find a way to get all faxes coming in at 9600, we'd be able to improve the reliability. 
>> 
>> Looking through the wiki and previous list messages, the fax_disable_v17=true setting seems to be the documented approach, but I'm not sure why we're still seeing faxes come in at a higher rate.
>> 
>> Any advice on this or are we missing something with regards to locking in the lower speed?
>> 
>> cheers,
>> A
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
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>> 
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>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>  <http://www.freeswitch.org/>	
>> FreeSWITCH <http://www.freeswitch.org/>
>> www.freeswitch.org <http://www.freeswitch.org/>
>> FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications.
>> 
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>>  <http://www.cluecon.com/>	
>> ClueCon Telephony and WebRTC Developer's conference <http://www.cluecon.com/>
>> www.cluecon.com <http://www.cluecon.com/>
>> ClueCon is an annual event bringing together all of the open source Telephony and WebRTC developers to collaberate on the latest technology in the communications industry.
>> 
>> 
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>> www.freeswitch.org <http://www.freeswitch.org/>
>> FreeSWITCH is an open-source media application designed to support popular protools such as SIP and WebRTC and provides a platform to develop voice and video applications.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
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> 
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From rmundkowsky at ets.org  Wed Feb 28 19:17:01 2018
From: rmundkowsky at ets.org (Mundkowsky, Robert)
Date: Wed, 28 Feb 2018 19:17:01 +0000
Subject: [Freeswitch-users] Send RTP to external server
In-Reply-To: <CAJoKKdVN9edESx--AyyQ8RY+=CN=90JQ_rLo0oKL7bbP-WZK7Q@mail.gmail.com>
References: <CA+fRVSWZxqe3VxmaS8TD0TmbUbmbh-=6NoKcyWVPm_+bGJ8b1A@mail.gmail.com>
 <CAEJMVkD2AMvYmh2w9sMJWFzbXeJ5nNr2fM9sOttiZmT8BcGSQQ@mail.gmail.com>
 <CA+fRVSWmhdgW1jS_jiMgc9XvZ_7dtpr1RJOVZgQfDUUrva1cyg@mail.gmail.com>
 <CAJoKKdVN9edESx--AyyQ8RY+=CN=90JQ_rLo0oKL7bbP-WZK7Q@mail.gmail.com>
Message-ID: <CY1PR07MB23771896AA8A186E0D283560BEC70@CY1PR07MB2377.namprd07.prod.outlook.com>

It is actually fairly common to want to stream audio/video to somewhere to process the data in real time, rather than waiting until recording is finished.

For example, if want to convert audio to text (ASR) then you do not want to wait until the conference is over before you start the ASR.  Imagine, your ASR takes 1 minute to convert 1 minute of audio then if a conference is 30 minutes long, you would have to wait another 30 minutes for the ASR to finish before you could do something with the text.

I believe you can use gstreamer (https://gstreamer.freedesktop.org/) to handle receiving the RTP from a FreeSWITCH conference.  I think it allows you to sample frames if you want to get frames from video granted I think you are only interested in audio, but I think you still have to develop a daemon that understand SIP/RTP in order to talk to FreeSWITCH.  Another approach might be to create a plugin for the Unimrcp (MRCPv2 server http://www.unimrcp.org/ ) and talk to the FreeSWITCH using their mrcp module and tight that to the conference call.  You might be able to hack the Kaldi plugin for Unimrcp to do what you want.

Robert

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir Culjaga
Sent: Wednesday, February 28, 2018 1:47 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Send RTP to external server

why do you want to stream... isn't it enough just to rsync the file once the call is finished ?

On 28 February 2018 at 16:16, Mickael Hubert <mickael at winlux.fr<mailto:mickael at winlux.fr>> wrote:
Hi
thanks a lot for your answer.
But I want to send stream to external server, not record in audio file. i can use record to "capture" the voice, but to stream it, it's more complicated ;)
thanks in advance

2018-02-21 0:50 GMT+01:00 Brian West <brian at freeswitch.com<mailto:brian at freeswitch.com>>:
you can already do this without SIPREC in freeswitch. By setting the RECORD_READ_ONLY or RECORD_WRITE_ONLY variables.

/b


On Fri, Feb 16, 2018 at 10:33 AM, Mickael Hubert <mickael at winlux.fr<mailto:mickael at winlux.fr>> wrote:
Hi list,
I want to record each call through freeswitch. But i want record only caller (SSRC 1) OR callee (SSRC 2) voice (not both).

I read about SIPREC, Jack, etc ... not interesting

Do you have a idea for me please ?

Thanks in advance

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Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitchsolutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C4f0c6786eb0f40725f8a08d57edbe173%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636554405236219602&sdata=6wPzLzP8i%2FDBCXxEyl%2F7xIPJ2hYQy9C2NNGXSiRvfKc%3D&reserved=0>

Official FreeSWITCH Sites
http://www.freeswitch.org<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C4f0c6786eb0f40725f8a08d57edbe173%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636554405236219602&sdata=36oG%2FMWaCRXZ%2BfGlz79y3Uv80cWbJvOvf3UtJrUZgKU%3D&reserved=0>
http://confluence.freeswitch.org<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fconfluence.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C4f0c6786eb0f40725f8a08d57edbe173%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636554405236219602&sdata=Fvt9Vq6L9eMPg6cfmpWZURJdemUPRGBOTkXsCKDC5yk%3D&reserved=0>
http://www.cluecon.com<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.cluecon.com&data=02%7C01%7Crmundkowsky%40ets.org%7C4f0c6786eb0f40725f8a08d57edbe173%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636554405236219602&sdata=Qe%2F2MQhf1lJmX1SfvyqVqxdnaxIdYzffbuf7eNPrSr0%3D&reserved=0>

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<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C4f0c6786eb0f40725f8a08d57edbe173%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636554405236219602&sdata=mdeOaiFaQ3UDDrIaOowtRBUXi4QcX5qlplMJAEP6eHM%3D&reserved=0>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users&data=02%7C01%7Crmundkowsky%40ets.org%7C4f0c6786eb0f40725f8a08d57edbe173%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636554405236219602&sdata=KqGoGGxqpgJFj494jDK%2FcI3kEsImyIWHdBZnnRW7FX4%3D&reserved=0>
http://www.freeswitch.org<https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.freeswitch.org&data=02%7C01%7Crmundkowsky%40ets.org%7C4f0c6786eb0f40725f8a08d57edbe173%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636554405236219602&sdata=36oG%2FMWaCRXZ%2BfGlz79y3Uv80cWbJvOvf3UtJrUZgKU%3D&reserved=0>


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