[Freeswitch-users] Broken silence with webrtc
Anthony Minessale
anthony.minessale at gmail.com
Wed Sep 2 00:18:40 MSD 2015
1) Do not interact on the mailing list if you have a Digest Subscription.
This is intended for those who only read.
When you reply in this mode, it breaks the threading of the email. Switch
to the standard subscription.
2) You need to use wireshark to debug. run fsctl debug_level 10 on cli so
you can see additional debug.
if it says no stun for a long time it means you are not getting any data.
3) With a call up, you can get the uuid with "show channels" then
uuid_debug_media <uuid> both on
4) Did you actually revert to the stock configs for all modules? Since you
are the only person with this problem it suggest to examine your
environment. (see #2)
On Tue, Sep 1, 2015 at 3:05 PM, Gary Foreman <gaz.foreman at gmail.com> wrote:
> Ok so on a freshly built Debian install with FS 1.7 I get the exact same
> behaviour.
>
> I managed to stop the broken silence when using the originate command by
> adding absolute_codec_string=PCMA. My thoughts were it could be a
> transcoding thing but inbound calls from the Verto client (which have no
> issue) result in opus > g722 so cant be.
>
> The intermittent audio on verto calls is still a problem though. The most
> common issue is one way audio from the verto client. The recording has no
> audio from the verto side either so I guess the RTP stream isn't being sent.
>
> Does anyone have any pointers to where I can look when I reproduce the
> fault? It takes a while for the issue to occur but once I have the call
> active I need to find where in the chain it is falling down. No errors are
> reported in the Chrome or FS console.
>
> Thanks
>
> On Tue, Sep 1, 2015 at 4:36 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. Re: Broken silence with webrtc (Anthony Minessale)
>>
>>
>> ---------- Forwarded message ----------
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Date: Tue, 1 Sep 2015 10:35:30 -0500
>> Subject: Re: [Freeswitch-users] Broken silence with webrtc
>> Your findings contradict each other so much, I recommend you start over
>> from scratch.
>> Backup your configs. update to the latest master version of FS. If you
>> are on 1.4, nothing new will be done to mitigate webrtc issues. Set up a
>> box with the default configurations and retest.
>>
>> you may also want to try putting <X-PRE-PROCESS cmd="set" data=
>> "suppress_cng=true"/> in vars.xml
>>
>>
>>
>> On Tue, Sep 1, 2015 at 10:25 AM, Ken Rice <krice at freeswitch.org> wrote:
>>
>>> Just enable verto debugging in verto.conf.xml in your configs… it’ll
>>> print it right to the screen
>>>
>>>
>>>
>>>
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gary
>>> Foreman
>>> *Sent:* Tuesday, September 1, 2015 10:13 AM
>>> *To:* freeswitch-users at lists.freeswitch.org
>>>
>>> *Subject:* Re: [Freeswitch-users] Broken silence with webrtc
>>>
>>>
>>>
>>> Ok so the issue has been superseded by intermittent one-way / no audio.
>>> I'm getting it very intermittently (1 in every 30 calls or so) but I'm
>>> struggling to debug it as the traffic is encrypted and wireshark doesn't
>>> see it as rtp stream.
>>>
>>>
>>>
>>> Where is the best place to start debugging verto? I was previously using
>>> sip.js without any audio issues so it seems to be verto specific.
>>>
>>>
>>>
>>> On Tue, Sep 1, 2015 at 12:04 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. Re: Broken silence with webrtc (Stanislav Sinyagin)
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Stanislav Sinyagin <ssinyagin at gmail.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Cc:
>>> Date: Tue, 1 Sep 2015 13:03:25 +0200
>>> Subject: Re: [Freeswitch-users] Broken silence with webrtc
>>>
>>> is it running on a virtual machine?
>>>
>>> I found a strange effect that I could only reproduce in a VM, and never
>>> on physical hardware:
>>> https://freeswitch.org/jira/browse/FS-7805
>>>
>>> under certain load, an originate command triggers a continuous
>>> distortion in another, running and unrelated, channel.
>>>
>>> It seems to be triggered by insufficient CPU resource at the moment of
>>> the origination.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Tue, Sep 1, 2015 at 12:49 PM, Gary Foreman <gaz.foreman at gmail.com>
>>> wrote:
>>>
>>> I've found that it occurs after any bridge, its not specific to the
>>> originate command.
>>>
>>>
>>>
>>> Would you require a wireshark trace or the output of the freeswitch
>>> console?
>>>
>>>
>>>
>>> The scenario below reproduces the issue ...
>>>
>>>
>>>
>>> Test extension
>>>
>>>
>>>
>>> <extension name="Inbound_Routing">
>>>
>>> <condition field="destination_number" expression="^2003$">
>>>
>>> <action application="export" data="dialed_extension=$1"/>
>>>
>>> <action application="export" data="transfer_ringback=$${uk-ring}"/>
>>>
>>> <action application="set" data="RECORD_TITLE=Title goes here"/>
>>>
>>> <action application="set"
>>> data="rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/>
>>>
>>> <action application="export" data="send_silence_when_idle=false"/>
>>>
>>> <action application="export" data="bridge_generate_comfort_noise=false"/>
>>>
>>> <action application="export" data="rtp_enable_vad_in=false"/>
>>>
>>> <action application="export" data="rtp_enable_vad_out=false"/>
>>>
>>> <action application="export" data="fire_talk_events=true"/>
>>>
>>> <action application="export" data="fire_not_talk_events=true"/>
>>>
>>> <action application="export" data="recording_id=${uuid}"/>
>>>
>>> <action application="set" data="media_bug_answer_req=true"/>
>>>
>>> <action application="answer" />
>>>
>>> <action application="playback"
>>> data="tone_stream://L=100;%(400,200,400,450);%(400,2000,400,450)"/>
>>>
>>> </condition>
>>>
>>> </extension>
>>>
>>>
>>>
>>> I originate a call from a polycom handset using g722 to the extension
>>> above.
>>>
>>> I originate a call using the verto client to the extension above.
>>>
>>>
>>>
>>> I get the uuids of the channels using show channels and use uuid_bridge
>>> [uuid1] [uuid2] to merge the channels.
>>>
>>>
>>>
>>> On Tue, Sep 1, 2015 at 11:30 AM, <
>>> 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. How send in To header anonymous and in P-Asserted-Identity
>>> caller-id-number (Alex Polischuk)
>>> 2. Loop play while wait DTMF digit (Dmitry Mordovin)
>>> 3. Broken silence with webrtc (Gary Foreman)
>>> 4. Re: Broken silence with webrtc (Brian West)
>>> 5. SIP profile not loading (Chris Young)
>>> 6. Playing with conditions (Dmitry Mordovin)
>>> 7. Re: Playing with conditions (Avi Marcus)
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Alex Polischuk <alxpol at gmail.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Cc:
>>> Date: Mon, 31 Aug 2015 17:02:25 +0300
>>> Subject: [Freeswitch-users] How send in To header anonymous and in
>>> P-Asserted-Identity caller-id-number
>>>
>>> Hi all,
>>>
>>>
>>>
>>> How I can define different users and domains in To and P-Asserted-Identity
>>> headers?
>>>
>>>
>>>
>>> Thanks,
>>>
>>> Alex
>>>
>>>
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Dmitry Mordovin <d.mordovin at dwide.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Cc:
>>> Date: Mon, 31 Aug 2015 18:24:08 +0400
>>> Subject: [Freeswitch-users] Loop play while wait DTMF digit
>>>
>>> Hello
>>>
>>> This example play conf-pin.wav and wait DTMF.
>>>
>>>
>>> <extension name="play_and_get_digits example">
>>>
>>> <condition field="destination_number" expression="^(1888)$">
>>>
>>> <action application="play_and_get_digits" data="2 5 3 7000 # $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav foobar \d+"/>
>>>
>>> <action application="log" data="CRIT ${foobar}"/>
>>>
>>> </condition>
>>>
>>> </extension>
>>>
>>>
>>> Is it possible to play WAV file in infinity loop and wait user DTMF?
>>>
>>> And how can I check DTMF input after user entered DTMF?
>>>
>>> In dialpeer, like.
>>> If ${DTMF} = 1 then bridge to XXX
>>> If ${DTMF} = 2 then play file and finish session
>>>
>>>
>>>
>>> Thank you.
>>> Dmitry
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Gary Foreman <gaz.foreman at gmail.com>
>>> To: freeswitch-users at lists.freeswitch.org
>>> Cc:
>>> Date: Mon, 31 Aug 2015 20:36:59 +0100
>>> Subject: [Freeswitch-users] Broken silence with webrtc
>>>
>>> Hi all,
>>>
>>>
>>>
>>> Hoping someone can point me in the right direction because after several
>>> hours I'm out of ideas.
>>>
>>>
>>>
>>> I'm having an issue where the 2nd leg of a call that is bridged to a
>>> verto endpoint has broken silence. When there is no sound on the line the
>>> audio goes completely silent (all background noise is dropped) but around
>>> every third of a second it repeatedly cuts back in for a fraction, then
>>> goes completely silent again. This only happens when the call is created
>>> using the originate command.
>>>
>>>
>>>
>>> Can anyone give me an idea of where to look next? I have a wireshark
>>> trace that during playback shows the audio cutting out periodically on the
>>> 2nd leg during periods of silence. The 1st leg is using webrtc encryption
>>> and I cant decode the stream.
>>>
>>>
>>>
>>> Thanks in advance!
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Brian West <brian at freeswitch.org>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Cc:
>>> Date: Mon, 31 Aug 2015 14:40:17 -0500
>>> Subject: Re: [Freeswitch-users] Broken silence with webrtc
>>>
>>> Please provide logs and samples of how you originate this, sounds like
>>> Voice Activity Detection possibly.
>>>
>>>
>>>
>>> On Mon, Aug 31, 2015 at 2:36 PM, Gary Foreman <gaz.foreman at gmail.com>
>>> wrote:
>>>
>>> Hi all,
>>>
>>>
>>>
>>> Hoping someone can point me in the right direction because after several
>>> hours I'm out of ideas.
>>>
>>>
>>>
>>> I'm having an issue where the 2nd leg of a call that is bridged to a
>>> verto endpoint has broken silence. When there is no sound on the line the
>>> audio goes completely silent (all background noise is dropped) but around
>>> every third of a second it repeatedly cuts back in for a fraction, then
>>> goes completely silent again. This only happens when the call is created
>>> using the originate command.
>>>
>>>
>>>
>>> Can anyone give me an idea of where to look next? I have a wireshark
>>> trace that during playback shows the audio cutting out periodically on the
>>> 2nd leg during periods of silence. The 1st leg is using webrtc encryption
>>> and I cant decode the stream.
>>>
>>>
>>>
>>> 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*
>>> brian at freeswitch.org
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Chris Young <Chris.Young at enghouse.com>
>>> To: "freeswitch-users at lists.freeswitch.org" <
>>> freeswitch-users at lists.freeswitch.org>
>>> Cc:
>>> Date: Tue, 1 Sep 2015 07:31:00 +0000
>>> Subject: [Freeswitch-users] SIP profile not loading
>>>
>>> Hi all,
>>>
>>>
>>>
>>> Recently, we've begun experiencing a strange problem whereby the first
>>> SIP profile to be loaded gets 'stuck' and never actually completes its
>>> initialisation. This always seems to affect the first profile only, so if I
>>> have profiles named (for example):
>>>
>>>
>>>
>>> dummy.xml
>>>
>>> external.xml
>>>
>>> internal.xml
>>>
>>>
>>>
>>> then 'dummy' would fail to load but 'external' and 'internal' would be
>>> fine. No error messages are output to the logs but preliminary
>>> investigation suggests that the profile thread is becoming blocked for some
>>> reason. The specified IP address is valid and available and there are no
>>> other processes using the requested port. FreeSWITCH comes up successfully
>>> but 'sofia status' shows only 'external' and 'internal'. At this point, I
>>> can use 'sofia profile dummy start' and the profile loads correctly so it
>>> appears to be valid.
>>>
>>>
>>>
>>> Has anybody else seen this kind of behaviour or know what could be
>>> causing it?
>>>
>>>
>>>
>>> Many thanks,
>>>
>>> Chris
>>>
>>>
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Dmitry Mordovin <d.mordovin at dwide.com>
>>> To: "freeswitch-users at lists.freeswitch.org" <
>>> freeswitch-users at lists.freeswitch.org>
>>> Cc:
>>> Date: Tue, 01 Sep 2015 12:56:08 +0400
>>> Subject: [Freeswitch-users] Playing with conditions
>>> Hello
>>>
>>> <section name="dialplan" description="Local call">
>>> <context name="public">
>>> <extension name="local_1">
>>>
>>> <condition field="destination_number" expression="^(\d{4})$">
>>> <action application="answer"/>
>>>
>>> <action application="play_and_get_digits" data="1 1 3 7000 #
>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav /invalid.wav
>>> foobar \d+"/>
>>> <action application="log" data="CRIT ${foobar}"/>
>>>
>>> <condition field="${foobar}" expression="1$">
>>> <action application="log" data="CRIT
>>> --------------1------------------"/>
>>> <anti-action application="log" data="CRIT --------------not
>>> 1------------------"/>
>>> </condition>
>>>
>>> <action application="log" data="CRIT ---- last ${foobar}"/>
>>> </condition>
>>>
>>> </extension>
>>> </context>
>>> </section>
>>>
>>> Why inner condition not works?
>>>
>>>
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Avi Marcus <avi at avimarcus.net>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Cc:
>>> Date: Tue, 1 Sep 2015 10:30:00 +0000
>>> Subject: Re: [Freeswitch-users] Playing with conditions
>>>
>>> Short answer: Each extension only has 1 set of conditions.
>>>
>>> The condition evaluating foobar is run *before* it gets set.
>>>
>>> After the play_and_get_digits you should transfer/execute_extension to
>>> a new extension that will evaluate ${foobar}
>>>
>>>
>>> -Avi Marcus
>>> BestFone
>>>
>>>
>>>
>>> On Tue, Sep 1, 2015 at 11:56 AM, Dmitry Mordovin <d.mordovin at dwide.com>
>>> wrote:
>>>
>>> Hello
>>>
>>> <section name="dialplan" description="Local call">
>>> <context name="public">
>>> <extension name="local_1">
>>>
>>> <condition field="destination_number" expression="^(\d{4})$">
>>> <action application="answer"/>
>>>
>>> <action application="play_and_get_digits" data="1 1 3 7000 #
>>> $${base_dir}/sounds/en/us/callie/conference/8000/conf-pin.wav
>>> /invalid.wav foobar \d+"/>
>>> <action application="log" data="CRIT ${foobar}"/>
>>>
>>> <condition field="${foobar}" expression="1$">
>>> <action application="log" data="CRIT
>>> --------------1------------------"/>
>>> <anti-action application="log" data="CRIT --------------not
>>> 1------------------"/>
>>> </condition>
>>>
>>> <action application="log" data="CRIT ---- last ${foobar}"/>
>>> </condition>
>>>
>>> </extension>
>>> </context>
>>> </section>
>>>
>>> Why inner condition not works?
>>>
>>> _________________________________________________________________________
>>> 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
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> 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
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> 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 ♬ @anthmfs ♬ @FreeSWITCH ♬
>>
>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞
>> http://twitter.com/FreeSWITCH
>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+
>> <http://freeswitch.org/g+>*
>>
>> ClueCon Weekly Development Call
>> ☎ sip:888 at conference.freeswitch.org ☎ +19193869900
>>
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>
>> _______________________________________________
>> FreeSWITCH-users mailing 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
>
--
Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞
http://twitter.com/FreeSWITCH
☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+
<http://freeswitch.org/g+>*
ClueCon Weekly Development Call
☎ sip:888 at conference.freeswitch.org ☎ +19193869900
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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