From william.king at quentustech.com  Wed Jul  1 00:27:48 2015
From: william.king at quentustech.com (William King)
Date: Tue, 30 Jun 2015 13:27:48 -0700
Subject: [Freeswitch-users] External Softphone Best Practices
In-Reply-To: <CA+zebiaoGD5vCZww_HGsaJE0ZOwJ3KnR36GyqqtXFRgHnjz4NA@mail.gmail.com>
References: <CA+zebiaoGD5vCZww_HGsaJE0ZOwJ3KnR36GyqqtXFRgHnjz4NA@mail.gmail.com>
Message-ID: <5592FBC4.7050301@quentustech.com>

If you are going to create a profile specifically to handle remote sip
phones, you might as well setup the profile with TLS, and configure it
to require register auth. Then by the time the call gets to the
dialplan, it will have already had successful authentication.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 6/30/15 12:35 PM, Komar, Jason wrote:
> I have set up a couple of PBXs using FreeSWITCH over the last few years.
> I have learned quite a bit, but am certainly no expert.
> 
> Recently, I installed Bria on my Android cell phone and was using it to
> make calls over FreeSWITCH throughout our building on our local network
> wifi. I wanted to be able to make calls from offsite as well, sometimes
> on wifi and sometimes over cellular data. I setup an external5090
> profile as I read on the wiki (didn't see anything in confluence yet). I
> was able to register through this profile, but outgoing calls from my
> softphone hit the public context and didn't go any further.
> 
> The user_context variable is set to default in my directory entry for
> this user, but that didn't seem to make a difference. I tried two things
> that worked, but am not sure if they open up any holes that would cause
> security problems:
> 
> 1.) If I set the context to default rather than public in the
> external5090 profile, it works, but I am unsure if this is at all secure.
> 
> 2.) If I leave the context as public in the external5090 profile and
> uncomment this section in the public.xml dialplan,
> 
>     <!--
> If you have made it this far lets challenge the caller and if they
> authenticate
> lets try what they dialed in the default context. (commented out by default)
>     -->
> 
>     <extension name="check_auth" continue="true">
>       <condition field="${sip_authorized}" expression="^true$"
> break="never">
> <anti-action application="respond" data="407"/>
>       </condition>
>     </extension>
>     
>     <extension name="transfer_to_default">
>       <condition>
> <action application="transfer" data="${destination_number} XML default"/>
>       </condition>
>     </extension>
> 
> it also works. This one seems the better option, but again, I'm not sure
> so I am asking the opinion of the experts on the list.
> 
> I spent several hours searching and reading everything I could find
> through Google and the mailing list archives, but came up a bit short.
> 
> Thanks in advance for your help.
> 
> Jason
> jkomar at jbox.ca <mailto:jkomar at jbox.ca>
> 
> 
> 
> _________________________________________________________________________
> 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 jkomar at jbox.ca  Wed Jul  1 02:49:52 2015
From: jkomar at jbox.ca (Komar, Jason)
Date: Tue, 30 Jun 2015 16:49:52 -0600
Subject: [Freeswitch-users] External Softphone Best Practices
In-Reply-To: <5592FBC4.7050301@quentustech.com>
References: <CA+zebiaoGD5vCZww_HGsaJE0ZOwJ3KnR36GyqqtXFRgHnjz4NA@mail.gmail.com>
	<5592FBC4.7050301@quentustech.com>
Message-ID: <CA+zebiYd-42jFYrpAFtR7wCoMT_bNf1qWe+Vpw3Teq=f=d5CyA@mail.gmail.com>

Thanks William. I totally missed seeing the auth-calls parameter in the
profile. I set it to true and it works great.

Thanks again,

Jason


On Tue, Jun 30, 2015 at 2:27 PM, William King <william.king at quentustech.com>
wrote:

> If you are going to create a profile specifically to handle remote sip
> phones, you might as well setup the profile with TLS, and configure it
> to require register auth. Then by the time the call gets to the
> dialplan, it will have already had successful authentication.
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
>
> On 6/30/15 12:35 PM, Komar, Jason wrote:
> > I have set up a couple of PBXs using FreeSWITCH over the last few years.
> > I have learned quite a bit, but am certainly no expert.
> >
> > Recently, I installed Bria on my Android cell phone and was using it to
> > make calls over FreeSWITCH throughout our building on our local network
> > wifi. I wanted to be able to make calls from offsite as well, sometimes
> > on wifi and sometimes over cellular data. I setup an external5090
> > profile as I read on the wiki (didn't see anything in confluence yet). I
> > was able to register through this profile, but outgoing calls from my
> > softphone hit the public context and didn't go any further.
> >
> > The user_context variable is set to default in my directory entry for
> > this user, but that didn't seem to make a difference. I tried two things
> > that worked, but am not sure if they open up any holes that would cause
> > security problems:
> >
> > 1.) If I set the context to default rather than public in the
> > external5090 profile, it works, but I am unsure if this is at all secure.
> >
> > 2.) If I leave the context as public in the external5090 profile and
> > uncomment this section in the public.xml dialplan,
> >
> >     <!--
> > If you have made it this far lets challenge the caller and if they
> > authenticate
> > lets try what they dialed in the default context. (commented out by
> default)
> >     -->
> >
> >     <extension name="check_auth" continue="true">
> >       <condition field="${sip_authorized}" expression="^true$"
> > break="never">
> > <anti-action application="respond" data="407"/>
> >       </condition>
> >     </extension>
> >
> >     <extension name="transfer_to_default">
> >       <condition>
> > <action application="transfer" data="${destination_number} XML default"/>
> >       </condition>
> >     </extension>
> >
> > it also works. This one seems the better option, but again, I'm not sure
> > so I am asking the opinion of the experts on the list.
> >
> > I spent several hours searching and reading everything I could find
> > through Google and the mailing list archives, but came up a bit short.
> >
> > Thanks in advance for your help.
> >
> > Jason
> > jkomar at jbox.ca <mailto:jkomar at jbox.ca>
> >
> >
> >
> > _________________________________________________________________________
> > 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 dujinfang at gmail.com  Wed Jul  1 03:43:44 2015
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 1 Jul 2015 07:43:44 +0800
Subject: [Freeswitch-users] Video playback in 1.7
In-Reply-To: <CAKbxfG8xUg+hEoAp76pdhYuji5YA6WkcUJrfEfNpdby2TWx-Zw@mail.gmail.com>
References: <CACOYK=h8effa=OwvVweQT9CC3a4u3M8Ff7mzgaDHQ4di26u9Rw@mail.gmail.com>
	<CALXCt0qf+a1tYQt=Wb0DsdwDUCATwpLoRq7J2MeyHy+aHezwnw@mail.gmail.com>
	<CACOYK=jugw0Z5Z6SXh98P5Xd4gM0carohDJ2-_ftuev=jAAKCA@mail.gmail.com>
	<CAKbxfG8d8XpdRvSCq1ZYut71E5+bWQtPb8QnKXSkt-dHBdNXMw@mail.gmail.com>
	<CACOYK=hE=evF8c+HDjmQP1=2-ZCwSe0GAVkPkcjUCEsrv+vMuA@mail.gmail.com>
	<CAKbxfG_5J6uUmLCYPAUsa47dsSPx2A4JaHycGn-fB56nEFE-SA@mail.gmail.com>
	<CACOYK=hnVpaZNbXXTxaj-tcYBP6q_OL3fVxiggVN7=BUzDDVwg@mail.gmail.com>
	<CAKbxfG8xUg+hEoAp76pdhYuji5YA6WkcUJrfEfNpdby2TWx-Zw@mail.gmail.com>
Message-ID: <CAJs+-5q7xzcetk_ny2XGqMqybWkoH24Mom4OMFT8jFgaeoivhg@mail.gmail.com>

yes I can confirm it also works with av:///tmp/test.mkv and .flv etc.

On Wed, Jul 1, 2015 at 2:45 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Depending on what ext you have and the proper build of mod_av I can't
> comment.
>
> I can confirm with the properly installed debian packages for jessie that
> at least mp4 files work either by reference to .mp4 files or by using
> av:///path/to/file.mp4
>
>
>
> On Tue, Jun 30, 2015 at 12:46 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
>> av:///tmp/somefile.ext crashed the daemon, I didn't yet find the time to
>> analyze it and file a jira.
>>  On Jun 30, 2015 7:43 PM, "Anthony Minessale" <
>> anthony.minessale at gmail.com> wrote:
>>
>>> mod_av is not an endpoint its a codec and file format module.
>>>
>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:213 Adding
>>> Codec H264 99 H264 Video 90000hz 0ms (VBR)
>>>
>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:292 Adding
>>> Application 'record_av'
>>>
>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:338 Adding
>>> API Function 'av_format'
>>>
>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:338 Adding
>>> API Function 'av_codec'
>>>
>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390 Adding
>>> File Format 'av'
>>>
>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390 Adding
>>> File Format 'rtmp'
>>>
>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390 Adding
>>> File Format 'mp4'
>>>
>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390 Adding
>>> File Format 'mov'
>>>
>>>
>>> It was av:// not avformat:// i was remembering the old version.
>>>
>>>
>>> if av:///tmp/somefile.ext works then it can be added to mod_avformat_load
>>> function in avformat.c:1949 ish  to register the extension for convenience.
>>>
>>>
>>>
>>>
>>>
>>> On Tue, Jun 30, 2015 at 2:07 AM, Stanislav Sinyagin <ssinyagin at gmail.com
>>> > wrote:
>>>
>>>> mod_av doesn't seem to be registering an endpoint, so prefixing
>>>> avformat:// or av:// does not help. I'll have a closer look later and
>>>> probably open a Jira
>>>>
>>>> On Tue, Jun 30, 2015 at 1:18 AM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> Webm has its own module.  Av and vlc both have broken webm at the time
>>>>> of coding.
>>>>>
>>>>>
>>>>> mod_vlc can play other formats but they are not registered in the
>>>>> module by file exten however you can use vlc:// syntax.
>>>>>
>>>>> mp4 is the safest bet because it works in mod_av which is more stable
>>>>> than vlc.  More formats can be added to mod_av as well but I don't remember
>>>>> if its as easy as avformat://<path>
>>>>>
>>>>> We don't have any choosing best format etc.  It's not going to be a
>>>>> point of focus to squeeze performance out of stuff like that in this stage
>>>>> of development.
>>>>>
>>>>> On Monday, June 29, 2015, Stanislav Sinyagin <ssinyagin at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> by the way, is there a way for playback to select a best matching
>>>>>> encoding, like it does with audio sample rates?
>>>>>>
>>>>>> On Tue, Jun 30, 2015 at 12:56 AM, Giovanni Maruzzelli <
>>>>>> gmaruzz at gmail.com> wrote:
>>>>>> > h264 I believe is supported...
>>>>>> >
>>>>>> > On Tue, Jun 30, 2015 at 12:48 AM, Stanislav Sinyagin <
>>>>>> ssinyagin at gmail.com>
>>>>>> > wrote:
>>>>>> >>
>>>>>> >> the newest 1.7 freeswitch successfully played an .mp4 file with
>>>>>> >> "playback" application, and the picture was sent to an VP8 client
>>>>>> >> (linphone on Android).
>>>>>> >>
>>>>>> >> The playback took about 20% CPU usage on a Xeon core -- probably
>>>>>> >> because of resising work. The source file was taken from
>>>>>> >> http://www.quirksmode.org/html5/tests/video.html
>>>>>> >>
>>>>>> >>
>>>>>> >> Question: what other file formats are supported?
>>>>>> >>
>>>>>> >> I tried .ogv and .webm, but I got "Invalid file format" error.
>>>>>> >>
>>>>>> >> thanks
>>>>>> >>
>>>>>> >>
>>>>>> _________________________________________________________________________
>>>>>> >> 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
>>>>>> > Cell : +39-347-2665618
>>>>>> >
>>>>>> >
>>>>>> _________________________________________________________________________
>>>>>> > 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
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> 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
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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
>>>>
>>>
>>>
>>>
>>> --
>>> 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
>>>
>>> _________________________________________________________________________
>>> 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
>>
>
>
>
> --
> 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
>
> _________________________________________________________________________
> 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 ssinyagin at gmail.com  Wed Jul  1 03:51:02 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 1 Jul 2015 01:51:02 +0200
Subject: [Freeswitch-users] Video playback in 1.7
In-Reply-To: <CAKbxfG8xUg+hEoAp76pdhYuji5YA6WkcUJrfEfNpdby2TWx-Zw@mail.gmail.com>
References: <CACOYK=h8effa=OwvVweQT9CC3a4u3M8Ff7mzgaDHQ4di26u9Rw@mail.gmail.com>
	<CALXCt0qf+a1tYQt=Wb0DsdwDUCATwpLoRq7J2MeyHy+aHezwnw@mail.gmail.com>
	<CACOYK=jugw0Z5Z6SXh98P5Xd4gM0carohDJ2-_ftuev=jAAKCA@mail.gmail.com>
	<CAKbxfG8d8XpdRvSCq1ZYut71E5+bWQtPb8QnKXSkt-dHBdNXMw@mail.gmail.com>
	<CACOYK=hE=evF8c+HDjmQP1=2-ZCwSe0GAVkPkcjUCEsrv+vMuA@mail.gmail.com>
	<CAKbxfG_5J6uUmLCYPAUsa47dsSPx2A4JaHycGn-fB56nEFE-SA@mail.gmail.com>
	<CACOYK=hnVpaZNbXXTxaj-tcYBP6q_OL3fVxiggVN7=BUzDDVwg@mail.gmail.com>
	<CAKbxfG8xUg+hEoAp76pdhYuji5YA6WkcUJrfEfNpdby2TWx-Zw@mail.gmail.com>
Message-ID: <CACOYK=jsR4xDPz-SoZ-YKY-J17BL+_ryCZiCso0xkO+dCvNdpw@mail.gmail.com>

I made a few more tests: the original file has AAC audio, and avconv needs
"-strict experimental" option to process that. Probably that's why
FreeSWITCH crashes. After I converted the video to 320x240 and MP3 audio, I
get a different error:

2015-07-01 01:46:22.701171 [ERR] avformat.c:1136 Failed to initialize the
resampling context

same error if I need to produce 48kHz OPUS or 8kHZ G711.

VLC plays back both the original and converted videos so far.

I'll play around with it during the week.




On Tue, Jun 30, 2015 at 8:45 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Depending on what ext you have and the proper build of mod_av I can't
> comment.
>
> I can confirm with the properly installed debian packages for jessie that
> at least mp4 files work either by reference to .mp4 files or by using
> av:///path/to/file.mp4
>
>
>
> On Tue, Jun 30, 2015 at 12:46 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
>> av:///tmp/somefile.ext crashed the daemon, I didn't yet find the time to
>> analyze it and file a jira.
>>  On Jun 30, 2015 7:43 PM, "Anthony Minessale" <
>> anthony.minessale at gmail.com> wrote:
>>
>>> mod_av is not an endpoint its a codec and file format module.
>>>
>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:213 Adding
>>> Codec H264 99 H264 Video 90000hz 0ms (VBR)
>>>
>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:292 Adding
>>> Application 'record_av'
>>>
>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:338 Adding
>>> API Function 'av_format'
>>>
>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:338 Adding
>>> API Function 'av_codec'
>>>
>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390 Adding
>>> File Format 'av'
>>>
>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390 Adding
>>> File Format 'rtmp'
>>>
>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390 Adding
>>> File Format 'mp4'
>>>
>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390 Adding
>>> File Format 'mov'
>>>
>>>
>>> It was av:// not avformat:// i was remembering the old version.
>>>
>>>
>>> if av:///tmp/somefile.ext works then it can be added to mod_avformat_load
>>> function in avformat.c:1949 ish  to register the extension for convenience.
>>>
>>>
>>>
>>>
>>>
>>> On Tue, Jun 30, 2015 at 2:07 AM, Stanislav Sinyagin <ssinyagin at gmail.com
>>> > wrote:
>>>
>>>> mod_av doesn't seem to be registering an endpoint, so prefixing
>>>> avformat:// or av:// does not help. I'll have a closer look later and
>>>> probably open a Jira
>>>>
>>>> On Tue, Jun 30, 2015 at 1:18 AM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> Webm has its own module.  Av and vlc both have broken webm at the time
>>>>> of coding.
>>>>>
>>>>>
>>>>> mod_vlc can play other formats but they are not registered in the
>>>>> module by file exten however you can use vlc:// syntax.
>>>>>
>>>>> mp4 is the safest bet because it works in mod_av which is more stable
>>>>> than vlc.  More formats can be added to mod_av as well but I don't remember
>>>>> if its as easy as avformat://<path>
>>>>>
>>>>> We don't have any choosing best format etc.  It's not going to be a
>>>>> point of focus to squeeze performance out of stuff like that in this stage
>>>>> of development.
>>>>>
>>>>> On Monday, June 29, 2015, Stanislav Sinyagin <ssinyagin at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> by the way, is there a way for playback to select a best matching
>>>>>> encoding, like it does with audio sample rates?
>>>>>>
>>>>>> On Tue, Jun 30, 2015 at 12:56 AM, Giovanni Maruzzelli <
>>>>>> gmaruzz at gmail.com> wrote:
>>>>>> > h264 I believe is supported...
>>>>>> >
>>>>>> > On Tue, Jun 30, 2015 at 12:48 AM, Stanislav Sinyagin <
>>>>>> ssinyagin at gmail.com>
>>>>>> > wrote:
>>>>>> >>
>>>>>> >> the newest 1.7 freeswitch successfully played an .mp4 file with
>>>>>> >> "playback" application, and the picture was sent to an VP8 client
>>>>>> >> (linphone on Android).
>>>>>> >>
>>>>>> >> The playback took about 20% CPU usage on a Xeon core -- probably
>>>>>> >> because of resising work. The source file was taken from
>>>>>> >> http://www.quirksmode.org/html5/tests/video.html
>>>>>> >>
>>>>>> >>
>>>>>> >> Question: what other file formats are supported?
>>>>>> >>
>>>>>> >> I tried .ogv and .webm, but I got "Invalid file format" error.
>>>>>> >>
>>>>>> >> thanks
>>>>>> >>
>>>>>> >>
>>>>>> _________________________________________________________________________
>>>>>> >> 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
>>>>>> > Cell : +39-347-2665618
>>>>>> >
>>>>>> >
>>>>>> _________________________________________________________________________
>>>>>> > 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
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> 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
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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
>>>>
>>>
>>>
>>>
>>> --
>>> 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
>>>
>>> _________________________________________________________________________
>>> 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
>>
>
>
>
> --
> 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
>
> _________________________________________________________________________
> 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 freeswitch.org  Wed Jul  1 04:54:54 2015
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 01 Jul 2015 00:54:54 +0000
Subject: [Freeswitch-users] FLOSS Weekly featuring FreeSWITCH!
Message-ID: <55933a5e7fbda_957e36b3204102@resque-worker-high.1.mail>

New Post on freeswitch.org from Kathleen
check it out at http://ift.tt/1LGHh66
FLOSS Weekly featuring FreeSWITCH!
Anthony Minessale and Michael Jerris will be featured on FLOSS Weekly tomorrow to talk about FreeSWITCH! Go check it out!

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From mike at jerris.com  Wed Jul  1 04:54:53 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 30 Jun 2015 20:54:53 -0400
Subject: [Freeswitch-users] Video playback in 1.7
In-Reply-To: <CACOYK=jsR4xDPz-SoZ-YKY-J17BL+_ryCZiCso0xkO+dCvNdpw@mail.gmail.com>
References: <CACOYK=h8effa=OwvVweQT9CC3a4u3M8Ff7mzgaDHQ4di26u9Rw@mail.gmail.com>
	<CALXCt0qf+a1tYQt=Wb0DsdwDUCATwpLoRq7J2MeyHy+aHezwnw@mail.gmail.com>
	<CACOYK=jugw0Z5Z6SXh98P5Xd4gM0carohDJ2-_ftuev=jAAKCA@mail.gmail.com>
	<CAKbxfG8d8XpdRvSCq1ZYut71E5+bWQtPb8QnKXSkt-dHBdNXMw@mail.gmail.com>
	<CACOYK=hE=evF8c+HDjmQP1=2-ZCwSe0GAVkPkcjUCEsrv+vMuA@mail.gmail.com>
	<CAKbxfG_5J6uUmLCYPAUsa47dsSPx2A4JaHycGn-fB56nEFE-SA@mail.gmail.com>
	<CACOYK=hnVpaZNbXXTxaj-tcYBP6q_OL3fVxiggVN7=BUzDDVwg@mail.gmail.com>
	<CAKbxfG8xUg+hEoAp76pdhYuji5YA6WkcUJrfEfNpdby2TWx-Zw@mail.gmail.com>
	<CACOYK=jsR4xDPz-SoZ-YKY-J17BL+_ryCZiCso0xkO+dCvNdpw@mail.gmail.com>
Message-ID: <1C0C06B1-187F-4D2D-A305-76763F975890@jerris.com>

That error happens when avresample_open call fails.  This is going to be some sort of issue with how libav was built as this is known working.

> On Jun 30, 2015, at 7:51 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> 
> I made a few more tests: the original file has AAC audio, and avconv needs "-strict experimental" option to process that. Probably that's why FreeSWITCH crashes. After I converted the video to 320x240 and MP3 audio, I get a different error: 
> 
> 2015-07-01 01:46:22.701171 [ERR] avformat.c:1136 Failed to initialize the resampling context
> 
> same error if I need to produce 48kHz OPUS or 8kHZ G711.
> 
> VLC plays back both the original and converted videos so far.
> 
> I'll play around with it during the week.
> 
> 
> 
> 
> On Tue, Jun 30, 2015 at 8:45 PM, Anthony Minessale <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
> Depending on what ext you have and the proper build of mod_av I can't comment.
> 
> I can confirm with the properly installed debian packages for jessie that at least mp4 files work either by reference to .mp4 files or by using av:///path/to/file.mp4
> 
> 
> 
> On Tue, Jun 30, 2015 at 12:46 PM, Stanislav Sinyagin <ssinyagin at gmail.com <mailto:ssinyagin at gmail.com>> wrote:
> av:///tmp/somefile.ext crashed the daemon, I didn't yet find the time to analyze it and file a jira.
> 
> On Jun 30, 2015 7:43 PM, "Anthony Minessale" <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
> mod_av is not an endpoint its a codec and file format module.
> 
> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:213 Adding Codec H264 99 H264 Video 90000hz 0ms (VBR)
> 
> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:292 Adding Application 'record_av'
> 
> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:338 Adding API Function 'av_format'
> 
> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:338 Adding API Function 'av_codec'
> 
> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390 Adding File Format 'av'
> 
> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390 Adding File Format 'rtmp'
> 
> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390 Adding File Format 'mp4'
> 
> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390 Adding File Format 'mov'
> 
> 
> 
> It was av:// not avformat:// i was remembering the old version.
> 
> 
> 
> if av:///tmp/somefile.ext works then it can be added to mod_avformat_load function in avformat.c:1949 ish  to register the extension for convenience.
> 
>  
> 
> 
> 
> On Tue, Jun 30, 2015 at 2:07 AM, Stanislav Sinyagin <ssinyagin at gmail.com <mailto:ssinyagin at gmail.com>> wrote:
> mod_av doesn't seem to be registering an endpoint, so prefixing avformat:// or av:// does not help. I'll have a closer look later and probably open a Jira
> 
> On Tue, Jun 30, 2015 at 1:18 AM, Anthony Minessale <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
> Webm has its own module.  Av and vlc both have broken webm at the time of coding.
> 
> 
> mod_vlc can play other formats but they are not registered in the module by file exten however you can use vlc:// syntax.
> 
> mp4 is the safest bet because it works in mod_av which is more stable than vlc.  More formats can be added to mod_av as well but I don't remember if its as easy as avformat://<path>
> 
> We don't have any choosing best format etc.  It's not going to be a point of focus to squeeze performance out of stuff like that in this stage of development.
> 
> On Monday, June 29, 2015, Stanislav Sinyagin <ssinyagin at gmail.com <mailto:ssinyagin at gmail.com>> wrote:
> by the way, is there a way for playback to select a best matching
> encoding, like it does with audio sample rates?
> 
> On Tue, Jun 30, 2015 at 12:56 AM, Giovanni Maruzzelli <gmaruzz at gmail.com <>> wrote:
> > h264 I believe is supported...
> >
> > On Tue, Jun 30, 2015 at 12:48 AM, Stanislav Sinyagin <ssinyagin at gmail.com <>>
> > wrote:
> >>
> >> the newest 1.7 freeswitch successfully played an .mp4 file with
> >> "playback" application, and the picture was sent to an VP8 client
> >> (linphone on Android).
> >>
> >> The playback took about 20% CPU usage on a Xeon core -- probably
> >> because of resising work. The source file was taken from
> >> http://www.quirksmode.org/html5/tests/video.html <http://www.quirksmode.org/html5/tests/video.html>
> >>
> >>
> >> Question: what other file formats are supported?
> >>
> >> I tried .ogv and .webm, but I got "Invalid file format" error.
> >>
> >> thanks
> >>
> >> _________________________________________________________________________
> >> 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 <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/>
> >
> >
> >
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618 <tel:%2B39-347-2665618>
> >
> > _________________________________________________________________________
> > 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 <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 <>
> 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 <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/>
> 
> 
> -- 
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
> 
> ? http://freeswitch.org/ <http://freeswitch.org/>  ? http://cluecon.com/ <http://cluecon.com/>  ? http://twitter.com/FreeSWITCH <http://twitter.com/FreeSWITCH>
> ? irc.freenode.net <http://irc.freenode.net/> #freeswitch ? http://freeswitch.org/g+ <http://freeswitch.org/g+>
> 
> ClueCon Weekly Development Call 
> ? sip:888 at conference.freeswitch.org <mailto:sip%3A888 at conference.freeswitch.org>  ? +19193869900 <tel:%2B19193869900> 
> 
> https://www.youtube.com/watch?v=9XXgW34t40s <https://www.youtube.com/watch?v=9XXgW34t40s>
> https://www.youtube.com/watch?v=NLaDpGQuZDA <https://www.youtube.com/watch?v=NLaDpGQuZDA>
> 
> 
> _________________________________________________________________________
> 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/>
> 
> 
> 
> -- 
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
> 
> ? http://freeswitch.org/ <http://freeswitch.org/>  ? http://cluecon.com/ <http://cluecon.com/>  ? http://twitter.com/FreeSWITCH <http://twitter.com/FreeSWITCH>
> ? irc.freenode.net <http://irc.freenode.net/> #freeswitch ? http://freeswitch.org/g+ <http://freeswitch.org/g+>
> 
> ClueCon Weekly Development Call 
> ? sip:888 at conference.freeswitch.org <mailto:sip%3A888 at conference.freeswitch.org>  ? +19193869900 <tel:%2B19193869900> 
> 
> https://www.youtube.com/watch?v=9XXgW34t40s <https://www.youtube.com/watch?v=9XXgW34t40s>
> https://www.youtube.com/watch?v=NLaDpGQuZDA <https://www.youtube.com/watch?v=NLaDpGQuZDA>
> 
> _________________________________________________________________________
> 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/>
> 
> 
> 
> -- 
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
> 
> ? http://freeswitch.org/ <http://freeswitch.org/>  ? http://cluecon.com/ <http://cluecon.com/>  ? http://twitter.com/FreeSWITCH <http://twitter.com/FreeSWITCH>
> ? irc.freenode.net <http://irc.freenode.net/> #freeswitch ? http://freeswitch.org/g+ <http://freeswitch.org/g+>
> 
> ClueCon Weekly Development Call 
> ? sip:888 at conference.freeswitch.org <mailto:sip%3A888 at conference.freeswitch.org>  ? +19193869900 <tel:%2B19193869900> 
> 
> https://www.youtube.com/watch?v=9XXgW34t40s <https://www.youtube.com/watch?v=9XXgW34t40s>
> https://www.youtube.com/watch?v=NLaDpGQuZDA <https://www.youtube.com/watch?v=NLaDpGQuZDA>
> 
> _________________________________________________________________________
> 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 anthony.minessale at gmail.com  Wed Jul  1 05:17:28 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 30 Jun 2015 20:17:28 -0500
Subject: [Freeswitch-users] Video playback in 1.7
In-Reply-To: <1C0C06B1-187F-4D2D-A305-76763F975890@jerris.com>
References: <CACOYK=h8effa=OwvVweQT9CC3a4u3M8Ff7mzgaDHQ4di26u9Rw@mail.gmail.com>
	<CALXCt0qf+a1tYQt=Wb0DsdwDUCATwpLoRq7J2MeyHy+aHezwnw@mail.gmail.com>
	<CACOYK=jugw0Z5Z6SXh98P5Xd4gM0carohDJ2-_ftuev=jAAKCA@mail.gmail.com>
	<CAKbxfG8d8XpdRvSCq1ZYut71E5+bWQtPb8QnKXSkt-dHBdNXMw@mail.gmail.com>
	<CACOYK=hE=evF8c+HDjmQP1=2-ZCwSe0GAVkPkcjUCEsrv+vMuA@mail.gmail.com>
	<CAKbxfG_5J6uUmLCYPAUsa47dsSPx2A4JaHycGn-fB56nEFE-SA@mail.gmail.com>
	<CACOYK=hnVpaZNbXXTxaj-tcYBP6q_OL3fVxiggVN7=BUzDDVwg@mail.gmail.com>
	<CAKbxfG8xUg+hEoAp76pdhYuji5YA6WkcUJrfEfNpdby2TWx-Zw@mail.gmail.com>
	<CACOYK=jsR4xDPz-SoZ-YKY-J17BL+_ryCZiCso0xkO+dCvNdpw@mail.gmail.com>
	<1C0C06B1-187F-4D2D-A305-76763F975890@jerris.com>
Message-ID: <CAKbxfG8028tSL-8wj-=qcNQ3xk08xM7uwCqXg-OSM_f=JFc5GQ@mail.gmail.com>

Like I said, you need the precise versions we have detailed in our Debian
jessie packaging.   It does not work on older versions of libav*
If you don't want to use jessie you need to see the versions of everything
we use and manually build it all and its full chain of cross depends.


On Tue, Jun 30, 2015 at 7:54 PM, Michael Jerris <mike at jerris.com> wrote:

> That error happens when avresample_open call fails.  This is going to be
> some sort of issue with how libav was built as this is known working.
>
> On Jun 30, 2015, at 7:51 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
> I made a few more tests: the original file has AAC audio, and avconv needs
> "-strict experimental" option to process that. Probably that's why
> FreeSWITCH crashes. After I converted the video to 320x240 and MP3 audio, I
> get a different error:
>
> 2015-07-01 01:46:22.701171 [ERR] avformat.c:1136 Failed to initialize the
> resampling context
>
> same error if I need to produce 48kHz OPUS or 8kHZ G711.
>
> VLC plays back both the original and converted videos so far.
>
> I'll play around with it during the week.
>
>
>
>
> On Tue, Jun 30, 2015 at 8:45 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Depending on what ext you have and the proper build of mod_av I can't
>> comment.
>>
>> I can confirm with the properly installed debian packages for jessie that
>> at least mp4 files work either by reference to .mp4 files or by using
>> av:///path/to/file.mp4
>>
>>
>>
>> On Tue, Jun 30, 2015 at 12:46 PM, Stanislav Sinyagin <ssinyagin at gmail.com
>> > wrote:
>>
>>> av:///tmp/somefile.ext crashed the daemon, I didn't yet find the time
>>> to analyze it and file a jira.
>>> On Jun 30, 2015 7:43 PM, "Anthony Minessale" <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> mod_av is not an endpoint its a codec and file format module.
>>>>
>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:213 Adding
>>>> Codec H264 99 H264 Video 90000hz 0ms (VBR)
>>>>
>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:292 Adding
>>>> Application 'record_av'
>>>>
>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:338 Adding
>>>> API Function 'av_format'
>>>>
>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:338 Adding
>>>> API Function 'av_codec'
>>>>
>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390 Adding
>>>> File Format 'av'
>>>>
>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390 Adding
>>>> File Format 'rtmp'
>>>>
>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390 Adding
>>>> File Format 'mp4'
>>>>
>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390 Adding
>>>> File Format 'mov'
>>>>
>>>>
>>>> It was av:// not avformat:// i was remembering the old version.
>>>>
>>>>
>>>> if av:///tmp/somefile.ext works then it can be added to mod_avformat_load
>>>> function in avformat.c:1949 ish  to register the extension for convenience.
>>>>
>>>>
>>>>
>>>>
>>>> On Tue, Jun 30, 2015 at 2:07 AM, Stanislav Sinyagin <
>>>> ssinyagin at gmail.com> wrote:
>>>>
>>>>> mod_av doesn't seem to be registering an endpoint, so prefixing
>>>>> avformat:// or av:// does not help. I'll have a closer look later and
>>>>> probably open a Jira
>>>>>
>>>>> On Tue, Jun 30, 2015 at 1:18 AM, Anthony Minessale <
>>>>> anthony.minessale at gmail.com> wrote:
>>>>>
>>>>>> Webm has its own module.  Av and vlc both have broken webm at the
>>>>>> time of coding.
>>>>>>
>>>>>>
>>>>>> mod_vlc can play other formats but they are not registered in the
>>>>>> module by file exten however you can use vlc:// syntax.
>>>>>>
>>>>>> mp4 is the safest bet because it works in mod_av which is more stable
>>>>>> than vlc.  More formats can be added to mod_av as well but I don't remember
>>>>>> if its as easy as avformat://<path>
>>>>>>
>>>>>> We don't have any choosing best format etc.  It's not going to be a
>>>>>> point of focus to squeeze performance out of stuff like that in this stage
>>>>>> of development.
>>>>>>
>>>>>> On Monday, June 29, 2015, Stanislav Sinyagin <ssinyagin at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> by the way, is there a way for playback to select a best matching
>>>>>>> encoding, like it does with audio sample rates?
>>>>>>>
>>>>>>> On Tue, Jun 30, 2015 at 12:56 AM, Giovanni Maruzzelli <
>>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>> > h264 I believe is supported...
>>>>>>> >
>>>>>>> > On Tue, Jun 30, 2015 at 12:48 AM, Stanislav Sinyagin <
>>>>>>> ssinyagin at gmail.com>
>>>>>>> > wrote:
>>>>>>> >>
>>>>>>> >> the newest 1.7 freeswitch successfully played an .mp4 file with
>>>>>>> >> "playback" application, and the picture was sent to an VP8 client
>>>>>>> >> (linphone on Android).
>>>>>>> >>
>>>>>>> >> The playback took about 20% CPU usage on a Xeon core -- probably
>>>>>>> >> because of resising work. The source file was taken from
>>>>>>> >> http://www.quirksmode.org/html5/tests/video.html
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> Question: what other file formats are supported?
>>>>>>> >>
>>>>>>> >> I tried .ogv and .webm, but I got "Invalid file format" error.
>>>>>>> >>
>>>>>>> >> thanks
>>>>>>> >>
>>>>>>> >>
>>>>>>> _________________________________________________________________________
>>>>>>> >> 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
>>>>>>> > Cell : +39-347-2665618
>>>>>>> >
>>>>>>> >
>>>>>>> _________________________________________________________________________
>>>>>>> > 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
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> 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
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> 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
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> 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
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>>>
>>
>>
>>
>> --
>> 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
>>
>> _________________________________________________________________________
>> 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
>



-- 
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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From wiebittewas at googlemail.com  Wed Jul  1 07:11:57 2015
From: wiebittewas at googlemail.com (wiebittewas)
Date: Wed, 01 Jul 2015 05:11:57 +0200
Subject: [Freeswitch-users] no display-update after transfer
Message-ID: <55935A7D.2010701@gmail.com>

when using transfer on the phones, the call will be successfully transfered to the other phone, but it doesn't display the orinal caller-id, but keeps instead that id of the transfering phone.

as I can see on the net, the refer-to info from the transfering phone to the server looks good:

	REFER sip:mod_sofia at 192.168.1.1:5060 SIP/2.0
	Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK850879522;rport
	From: <sip:24 at 192.168.1.3:5060>;tag=27328813
	To: "27" <sip:27 at pbx.priv>;tag=Net73KDB46X6r
	Call-ID: 33f2a7ee-9a39-1233-eab8-7674fbcada00
	CSeq: 77522256 REFER
	Contact: <sip:24 at 192.168.1.3:5060>
	Max-Forwards: 70
	Supported: replaces, path, timer
	Refer-To: <sip:20 at pbx.priv?Replaces=1999853991-5060-136%40BJC.BGI.CBI.EC%3Bto-tag%3DQ0cS79ejyracg%3Bfrom-tag%3D219451532>
	Referred-By: <sip:24 at pbx.priv>
	Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
	Content-Length: 0

and freeswitch sends an ACCEPTED-msg for this REFER



in the freeswitch.log I found entries like:

[ALERT] sofia.c:1198 sofia_update_callee_id() sofia/internal/u620 at 192.168.1.4:5060 Same Callee ID "Outbound Call" <20>

which leads to the situation, that no display-update will be send.

unfortunately the REFER-handling of freeswitch doesn't seem to offer any possibility to intervene for debugging the available variables, so I have no idea, why freeswitch sees equal callee-ids.

any suggestions how to look deeper or how to modifier the variables so that the update will be sent?

regards

w.


From nandy1925 at gmail.com  Wed Jul  1 07:52:11 2015
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Wed, 1 Jul 2015 11:52:11 +0800
Subject: [Freeswitch-users] So you wanna setup your own CA for
	WSS/SSL/TLS?
In-Reply-To: <CAEJMVkCdzFZrzfxuMXoGa=P1LVYcAOJJgKqLccZGBTdi2z0=ug@mail.gmail.com>
References: <CAEJMVkDH0tfmToSr=QooRf8GKum+Ru5WUAq6Lu1fDEXdpt06dg@mail.gmail.com>
	<53D2A7A7.9040400@quentustech.com>
	<CAEJMVkCdzFZrzfxuMXoGa=P1LVYcAOJJgKqLccZGBTdi2z0=ug@mail.gmail.com>
Message-ID: <CAHKqMb72n4oOpScq7-JgYV0OAO+b=GuD1GoeYAhKYkAc8+a0sA@mail.gmail.com>

Hi Brian,

I used your script to generate the certificates to test mod_verto in an
intranet setup. Questions on your script:

1) Is 4096 bits required? Or 2048 bits will work, too?
2) Examining certs/wss.pem, there should be a  <chain> certificate at the
end. But the script, inputs only 2 - *.crt and *.key. What should be the
3rd?

Tks,
/Nandy


On Sat, Jul 26, 2014 at 2:59 AM, Brian West <brian at freeswitch.org> wrote:

> I've corrected the how-to and put it in tree:
>
>
> https://stash.freeswitch.org/projects/FS/repos/freeswitch/browse/docs/how_to_make_your_own_ca_correctly.txt?raw
>
> Importing the ca.crt into your system keychain for it to be trusted is
> left to the end user to figure out.  If you can't do that step then you'll
> kinda be SOL, I know on my Mac I just open ca.crt and it does the import
> for me... Windows I suspect is similar as for Linux NO CLUE.
>
>
> On Fri, Jul 25, 2014 at 1:53 PM, William King <
> william.king at quentustech.com> wrote:
>
>> One correction inline, and did you have any luck getting chrome to work
>> with the custom CA?
>>
>> William King
>> Senior Engineer
>> Quentus Technologies, INC
>> 1037 NE 65th St Suite 273
>> Seattle, WA 98115
>> Main:   (877) 211-9337
>> Office: (206) 388-4772
>> Cell:   (253) 686-5518
>> william.king at quentustech.com
>>
>> On 07/25/2014 08:12 AM, Brian West wrote:
>> > Someone should probably turn this into a nice how-to:
>> >
>> > Here is how I did it.
>> >
>> > wget http://www.openssl.org/contrib/ssl.ca-0.1.tar.gz
>> > tar zxfv ssl.ca-0.1.tar.gz
>> > cd ssl.ca-0.1/
>> > perl -i -pe 's/md5/sha1/g' *.sh
>> > perl -i -pe 's/2048/2048/g' *.sh
>> This is a noop. I assume it was suppose to be /2048/4096/ or /1024/2048/
>> > ./new-root-ca.sh
>> > ./new-server-cert.sh self.bkw.org <http://self.bkw.org>
>> > ./sign-server-cert.sh self.bkw.org <http://self.bkw.org>
>> > cat self.bkw.org.crt self.bkw.org.key >
>> /usr/local/freeswitch/certs/wss.pem
>> >
>> > Setup Apache:
>> >
>> > default-ssl:
>> >
>> > SSLCertificateFile    /usr/local/freeswitch/certs/wss.pem
>> > SSLCertificateKeyFile /usr/local/freeswitch/certs/wss.pem
>> > SSLCertificateChainFile /usr/local/freeswitch/certs/wss.pem
>> >
>> > Setup Sofia TLS:
>> >
>> > cat self.bkw.org.crt self.bkw.org.key >
>> > /usr/local/freeswitch/certs/agent.pem
>> > cat ca.crt > /usr/local/freeswitch/certs/cafile.pem
>> >
>> > vars.xml:
>> >
>> > <X-PRE-PROCESScmd="set"data="internal_ssl_enable=true"/>
>> > <X-PRE-PROCESScmd="set"data="external_ssl_enable=true"/>
>> >
>> > Restart FreeSWITCH.
>> >
>> > Now make sure your system has ca.crt imported so it will trust your new
>> > found hotness.
>> >
>> > TEST:
>> >
>> > openssl s_client -connect self.bkw.org:443 <http://self.bkw.org:443>
>> > openssl s_client -connect self.bkw.org:8082 <http://self.bkw.org:8082>
>> >
>> >
>> > Depending on what you've setup you'll see:
>> >
>> > subject=/C=US/ST=Oklahoma/L=McAlester/O=Tonka Truck/OU=Secure Web
>> > Server/CN=self.bkw.org/emailAddress=brian at bkw.org
>> > <http://self.bkw.org/emailAddress=brian at bkw.org>
>> >
>> > issuer=/C=US/ST=Oklahoma/L=McAlester/O=Whizzzzzzy Bang
>> > Bang/OU=Certification Services Division/CN=WBB Root
>> > CA/emailAddress=brian at bkw.org <mailto:brian at bkw.org>
>> >
>> > Or there abouts.
>> >
>> > --
>> >
>> > */Brian West/*
>> > brian at freeswitch.org <mailto:brian at freeswitch.org>
>> >
>> >
>> > */Twitter: @FreeSWITCH , @briankwest/*
>> > http://www.freeswitchbook.com
>> > http://www.freeswitchcookbook.com
>> >
>> > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>> >
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>> > http://www.cudatel.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>> http://www.cudatel.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ssinyagin at gmail.com  Wed Jul  1 09:19:30 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 1 Jul 2015 07:19:30 +0200
Subject: [Freeswitch-users] Video playback in 1.7
In-Reply-To: <CAKbxfG8028tSL-8wj-=qcNQ3xk08xM7uwCqXg-OSM_f=JFc5GQ@mail.gmail.com>
References: <CACOYK=h8effa=OwvVweQT9CC3a4u3M8Ff7mzgaDHQ4di26u9Rw@mail.gmail.com>
	<CALXCt0qf+a1tYQt=Wb0DsdwDUCATwpLoRq7J2MeyHy+aHezwnw@mail.gmail.com>
	<CACOYK=jugw0Z5Z6SXh98P5Xd4gM0carohDJ2-_ftuev=jAAKCA@mail.gmail.com>
	<CAKbxfG8d8XpdRvSCq1ZYut71E5+bWQtPb8QnKXSkt-dHBdNXMw@mail.gmail.com>
	<CACOYK=hE=evF8c+HDjmQP1=2-ZCwSe0GAVkPkcjUCEsrv+vMuA@mail.gmail.com>
	<CAKbxfG_5J6uUmLCYPAUsa47dsSPx2A4JaHycGn-fB56nEFE-SA@mail.gmail.com>
	<CACOYK=hnVpaZNbXXTxaj-tcYBP6q_OL3fVxiggVN7=BUzDDVwg@mail.gmail.com>
	<CAKbxfG8xUg+hEoAp76pdhYuji5YA6WkcUJrfEfNpdby2TWx-Zw@mail.gmail.com>
	<CACOYK=jsR4xDPz-SoZ-YKY-J17BL+_ryCZiCso0xkO+dCvNdpw@mail.gmail.com>
	<1C0C06B1-187F-4D2D-A305-76763F975890@jerris.com>
	<CAKbxfG8028tSL-8wj-=qcNQ3xk08xM7uwCqXg-OSM_f=JFc5GQ@mail.gmail.com>
Message-ID: <CACOYK=iwPiRRC6uMDiDm83MusFDLpipogmDnCJ5y6Xbd-u6q-w@mail.gmail.com>

This was all on a freshly installed Jessie, with your 1.7 debs. I will have
a closer look and document the issue later, now I only had time for a quick
check.
On Jul 1, 2015 3:18 AM, "Anthony Minessale" <anthony.minessale at gmail.com>
wrote:

> Like I said, you need the precise versions we have detailed in our Debian
> jessie packaging.   It does not work on older versions of libav*
> If you don't want to use jessie you need to see the versions of everything
> we use and manually build it all and its full chain of cross depends.
>
>
> On Tue, Jun 30, 2015 at 7:54 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> That error happens when avresample_open call fails.  This is going to be
>> some sort of issue with how libav was built as this is known working.
>>
>> On Jun 30, 2015, at 7:51 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
>> wrote:
>>
>> I made a few more tests: the original file has AAC audio, and avconv
>> needs "-strict experimental" option to process that. Probably that's why
>> FreeSWITCH crashes. After I converted the video to 320x240 and MP3 audio, I
>> get a different error:
>>
>> 2015-07-01 01:46:22.701171 [ERR] avformat.c:1136 Failed to initialize the
>> resampling context
>>
>> same error if I need to produce 48kHz OPUS or 8kHZ G711.
>>
>> VLC plays back both the original and converted videos so far.
>>
>> I'll play around with it during the week.
>>
>>
>>
>>
>> On Tue, Jun 30, 2015 at 8:45 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> Depending on what ext you have and the proper build of mod_av I can't
>>> comment.
>>>
>>> I can confirm with the properly installed debian packages for jessie
>>> that at least mp4 files work either by reference to .mp4 files or by using
>>> av:///path/to/file.mp4
>>>
>>>
>>>
>>> On Tue, Jun 30, 2015 at 12:46 PM, Stanislav Sinyagin <
>>> ssinyagin at gmail.com> wrote:
>>>
>>>> av:///tmp/somefile.ext crashed the daemon, I didn't yet find the time
>>>> to analyze it and file a jira.
>>>> On Jun 30, 2015 7:43 PM, "Anthony Minessale" <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> mod_av is not an endpoint its a codec and file format module.
>>>>>
>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:213
>>>>> Adding Codec H264 99 H264 Video 90000hz 0ms (VBR)
>>>>>
>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:292
>>>>> Adding Application 'record_av'
>>>>>
>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:338
>>>>> Adding API Function 'av_format'
>>>>>
>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:338
>>>>> Adding API Function 'av_codec'
>>>>>
>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390
>>>>> Adding File Format 'av'
>>>>>
>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390
>>>>> Adding File Format 'rtmp'
>>>>>
>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390
>>>>> Adding File Format 'mp4'
>>>>>
>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390
>>>>> Adding File Format 'mov'
>>>>>
>>>>>
>>>>> It was av:// not avformat:// i was remembering the old version.
>>>>>
>>>>>
>>>>> if av:///tmp/somefile.ext works then it can be added to mod_avformat_load
>>>>> function in avformat.c:1949 ish  to register the extension for convenience.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jun 30, 2015 at 2:07 AM, Stanislav Sinyagin <
>>>>> ssinyagin at gmail.com> wrote:
>>>>>
>>>>>> mod_av doesn't seem to be registering an endpoint, so prefixing
>>>>>> avformat:// or av:// does not help. I'll have a closer look later and
>>>>>> probably open a Jira
>>>>>>
>>>>>> On Tue, Jun 30, 2015 at 1:18 AM, Anthony Minessale <
>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>
>>>>>>> Webm has its own module.  Av and vlc both have broken webm at the
>>>>>>> time of coding.
>>>>>>>
>>>>>>>
>>>>>>> mod_vlc can play other formats but they are not registered in the
>>>>>>> module by file exten however you can use vlc:// syntax.
>>>>>>>
>>>>>>> mp4 is the safest bet because it works in mod_av which is more
>>>>>>> stable than vlc.  More formats can be added to mod_av as well but I don't
>>>>>>> remember if its as easy as avformat://<path>
>>>>>>>
>>>>>>> We don't have any choosing best format etc.  It's not going to be a
>>>>>>> point of focus to squeeze performance out of stuff like that in this stage
>>>>>>> of development.
>>>>>>>
>>>>>>> On Monday, June 29, 2015, Stanislav Sinyagin <ssinyagin at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> by the way, is there a way for playback to select a best matching
>>>>>>>> encoding, like it does with audio sample rates?
>>>>>>>>
>>>>>>>> On Tue, Jun 30, 2015 at 12:56 AM, Giovanni Maruzzelli <
>>>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>>> > h264 I believe is supported...
>>>>>>>> >
>>>>>>>> > On Tue, Jun 30, 2015 at 12:48 AM, Stanislav Sinyagin <
>>>>>>>> ssinyagin at gmail.com>
>>>>>>>> > wrote:
>>>>>>>> >>
>>>>>>>> >> the newest 1.7 freeswitch successfully played an .mp4 file with
>>>>>>>> >> "playback" application, and the picture was sent to an VP8 client
>>>>>>>> >> (linphone on Android).
>>>>>>>> >>
>>>>>>>> >> The playback took about 20% CPU usage on a Xeon core -- probably
>>>>>>>> >> because of resising work. The source file was taken from
>>>>>>>> >> http://www.quirksmode.org/html5/tests/video.html
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >> Question: what other file formats are supported?
>>>>>>>> >>
>>>>>>>> >> I tried .ogv and .webm, but I got "Invalid file format" error.
>>>>>>>> >>
>>>>>>>> >> thanks
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> _________________________________________________________________________
>>>>>>>> >> 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
>>>>>>>> > Cell : +39-347-2665618
>>>>>>>> >
>>>>>>>> >
>>>>>>>> _________________________________________________________________________
>>>>>>>> > 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
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> 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
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> 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
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> 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
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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
>>>>
>>>
>>>
>>>
>>> --
>>> 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
>>>
>>> _________________________________________________________________________
>>> 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
>>
>
>
>
> --
> 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
>
> _________________________________________________________________________
> 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  Wed Jul  1 10:23:31 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 1 Jul 2015 02:23:31 -0400
Subject: [Freeswitch-users] Video playback in 1.7
In-Reply-To: <CACOYK=iwPiRRC6uMDiDm83MusFDLpipogmDnCJ5y6Xbd-u6q-w@mail.gmail.com>
References: <CACOYK=h8effa=OwvVweQT9CC3a4u3M8Ff7mzgaDHQ4di26u9Rw@mail.gmail.com>
	<CALXCt0qf+a1tYQt=Wb0DsdwDUCATwpLoRq7J2MeyHy+aHezwnw@mail.gmail.com>
	<CACOYK=jugw0Z5Z6SXh98P5Xd4gM0carohDJ2-_ftuev=jAAKCA@mail.gmail.com>
	<CAKbxfG8d8XpdRvSCq1ZYut71E5+bWQtPb8QnKXSkt-dHBdNXMw@mail.gmail.com>
	<CACOYK=hE=evF8c+HDjmQP1=2-ZCwSe0GAVkPkcjUCEsrv+vMuA@mail.gmail.com>
	<CAKbxfG_5J6uUmLCYPAUsa47dsSPx2A4JaHycGn-fB56nEFE-SA@mail.gmail.com>
	<CACOYK=hnVpaZNbXXTxaj-tcYBP6q_OL3fVxiggVN7=BUzDDVwg@mail.gmail.com>
	<CAKbxfG8xUg+hEoAp76pdhYuji5YA6WkcUJrfEfNpdby2TWx-Zw@mail.gmail.com>
	<CACOYK=jsR4xDPz-SoZ-YKY-J17BL+_ryCZiCso0xkO+dCvNdpw@mail.gmail.com>
	<1C0C06B1-187F-4D2D-A305-76763F975890@jerris.com>
	<CAKbxfG8028tSL-8wj-=qcNQ3xk08xM7uwCqXg-OSM_f=JFc5GQ@mail.gmail.com>
	<CACOYK=iwPiRRC6uMDiDm83MusFDLpipogmDnCJ5y6Xbd-u6q-w@mail.gmail.com>
Message-ID: <CAEwTsAiG6ob-YmFm++0ODxMNMtQQaUhf2XaGPCrkaJdhZAKmZg@mail.gmail.com>

do you have the libav -extra package (I don't recall the full name)
installed?

On Wednesday, July 1, 2015, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> This was all on a freshly installed Jessie, with your 1.7 debs. I will
> have a closer look and document the issue later, now I only had time for a
> quick check.
> On Jul 1, 2015 3:18 AM, "Anthony Minessale" <anthony.minessale at gmail.com
> <javascript:_e(%7B%7D,'cvml','anthony.minessale at gmail.com');>> wrote:
>
>> Like I said, you need the precise versions we have detailed in our Debian
>> jessie packaging.   It does not work on older versions of libav*
>> If you don't want to use jessie you need to see the versions of
>> everything we use and manually build it all and its full chain of cross
>> depends.
>>
>>
>> On Tue, Jun 30, 2015 at 7:54 PM, Michael Jerris <mike at jerris.com
>> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>> wrote:
>>
>>> That error happens when avresample_open call fails.  This is going to
>>> be some sort of issue with how libav was built as this is known working.
>>>
>>> On Jun 30, 2015, at 7:51 PM, Stanislav Sinyagin <ssinyagin at gmail.com
>>> <javascript:_e(%7B%7D,'cvml','ssinyagin at gmail.com');>> wrote:
>>>
>>> I made a few more tests: the original file has AAC audio, and avconv
>>> needs "-strict experimental" option to process that. Probably that's why
>>> FreeSWITCH crashes. After I converted the video to 320x240 and MP3 audio, I
>>> get a different error:
>>>
>>> 2015-07-01 01:46:22.701171 [ERR] avformat.c:1136 Failed to initialize
>>> the resampling context
>>>
>>> same error if I need to produce 48kHz OPUS or 8kHZ G711.
>>>
>>> VLC plays back both the original and converted videos so far.
>>>
>>> I'll play around with it during the week.
>>>
>>>
>>>
>>>
>>> On Tue, Jun 30, 2015 at 8:45 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com
>>> <javascript:_e(%7B%7D,'cvml','anthony.minessale at gmail.com');>> wrote:
>>>
>>>> Depending on what ext you have and the proper build of mod_av I can't
>>>> comment.
>>>>
>>>> I can confirm with the properly installed debian packages for jessie
>>>> that at least mp4 files work either by reference to .mp4 files or by using
>>>> av:///path/to/file.mp4
>>>>
>>>>
>>>>
>>>> On Tue, Jun 30, 2015 at 12:46 PM, Stanislav Sinyagin <
>>>> ssinyagin at gmail.com
>>>> <javascript:_e(%7B%7D,'cvml','ssinyagin at gmail.com');>> wrote:
>>>>
>>>>> av:///tmp/somefile.ext crashed the daemon, I didn't yet find the time
>>>>> to analyze it and file a jira.
>>>>> On Jun 30, 2015 7:43 PM, "Anthony Minessale" <
>>>>> anthony.minessale at gmail.com
>>>>> <javascript:_e(%7B%7D,'cvml','anthony.minessale at gmail.com');>> wrote:
>>>>>
>>>>>> mod_av is not an endpoint its a codec and file format module.
>>>>>>
>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:213
>>>>>> Adding Codec H264 99 H264 Video 90000hz 0ms (VBR)
>>>>>>
>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:292
>>>>>> Adding Application 'record_av'
>>>>>>
>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:338
>>>>>> Adding API Function 'av_format'
>>>>>>
>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:338
>>>>>> Adding API Function 'av_codec'
>>>>>>
>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390
>>>>>> Adding File Format 'av'
>>>>>>
>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390
>>>>>> Adding File Format 'rtmp'
>>>>>>
>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390
>>>>>> Adding File Format 'mp4'
>>>>>>
>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390
>>>>>> Adding File Format 'mov'
>>>>>>
>>>>>>
>>>>>> It was av:// not avformat:// i was remembering the old version.
>>>>>>
>>>>>>
>>>>>> if av:///tmp/somefile.ext works then it can be added to mod_avformat_load
>>>>>> function in avformat.c:1949 ish  to register the extension for convenience.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Jun 30, 2015 at 2:07 AM, Stanislav Sinyagin <
>>>>>> ssinyagin at gmail.com
>>>>>> <javascript:_e(%7B%7D,'cvml','ssinyagin at gmail.com');>> wrote:
>>>>>>
>>>>>>> mod_av doesn't seem to be registering an endpoint, so prefixing
>>>>>>> avformat:// or av:// does not help. I'll have a closer look later and
>>>>>>> probably open a Jira
>>>>>>>
>>>>>>> On Tue, Jun 30, 2015 at 1:18 AM, Anthony Minessale <
>>>>>>> anthony.minessale at gmail.com
>>>>>>> <javascript:_e(%7B%7D,'cvml','anthony.minessale at gmail.com');>>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Webm has its own module.  Av and vlc both have broken webm at the
>>>>>>>> time of coding.
>>>>>>>>
>>>>>>>>
>>>>>>>> mod_vlc can play other formats but they are not registered in the
>>>>>>>> module by file exten however you can use vlc:// syntax.
>>>>>>>>
>>>>>>>> mp4 is the safest bet because it works in mod_av which is more
>>>>>>>> stable than vlc.  More formats can be added to mod_av as well but I don't
>>>>>>>> remember if its as easy as avformat://<path>
>>>>>>>>
>>>>>>>> We don't have any choosing best format etc.  It's not going to be a
>>>>>>>> point of focus to squeeze performance out of stuff like that in this stage
>>>>>>>> of development.
>>>>>>>>
>>>>>>>> On Monday, June 29, 2015, Stanislav Sinyagin <ssinyagin at gmail.com
>>>>>>>> <javascript:_e(%7B%7D,'cvml','ssinyagin at gmail.com');>> wrote:
>>>>>>>>
>>>>>>>>> by the way, is there a way for playback to select a best matching
>>>>>>>>> encoding, like it does with audio sample rates?
>>>>>>>>>
>>>>>>>>> On Tue, Jun 30, 2015 at 12:56 AM, Giovanni Maruzzelli <
>>>>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>>>> > h264 I believe is supported...
>>>>>>>>> >
>>>>>>>>> > On Tue, Jun 30, 2015 at 12:48 AM, Stanislav Sinyagin <
>>>>>>>>> ssinyagin at gmail.com>
>>>>>>>>> > wrote:
>>>>>>>>> >>
>>>>>>>>> >> the newest 1.7 freeswitch successfully played an .mp4 file with
>>>>>>>>> >> "playback" application, and the picture was sent to an VP8
>>>>>>>>> client
>>>>>>>>> >> (linphone on Android).
>>>>>>>>> >>
>>>>>>>>> >> The playback took about 20% CPU usage on a Xeon core -- probably
>>>>>>>>> >> because of resising work. The source file was taken from
>>>>>>>>> >> http://www.quirksmode.org/html5/tests/video.html
>>>>>>>>> >>
>>>>>>>>> >>
>>>>>>>>> >> Question: what other file formats are supported?
>>>>>>>>> >>
>>>>>>>>> >> I tried .ogv and .webm, but I got "Invalid file format" error.
>>>>>>>>> >>
>>>>>>>>> >> thanks
>>>>>>>>> >>
>>>>>>>>> >>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> >> 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
>>>>>>>>> > Cell : +39-347-2665618
>>>>>>>>> >
>>>>>>>>> >
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> > 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
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> 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
>>>>>>>> <javascript:_e(%7B%7D,'cvml','sip:888 at conference.freeswitch.org');>
>>>>>>>>   ? +19193869900
>>>>>>>>
>>>>>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>>>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> <javascript:_e(%7B%7D,'cvml','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
>>>>>>>> <javascript:_e(%7B%7D,'cvml','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
>>>>>>> <javascript:_e(%7B%7D,'cvml','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
>>>>>>> <javascript:_e(%7B%7D,'cvml','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
>>>>>> <javascript:_e(%7B%7D,'cvml','sip:888 at conference.freeswitch.org');>
>>>>>>  ? +19193869900
>>>>>>
>>>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> <javascript:_e(%7B%7D,'cvml','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
>>>>>> <javascript:_e(%7B%7D,'cvml','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
>>>>> <javascript:_e(%7B%7D,'cvml','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
>>>>> <javascript:_e(%7B%7D,'cvml','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
>>>> <javascript:_e(%7B%7D,'cvml','sip:888 at conference.freeswitch.org');>  ?
>>>> +19193869900
>>>>
>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> <javascript:_e(%7B%7D,'cvml','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
>>>> <javascript:_e(%7B%7D,'cvml','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
>>> <javascript:_e(%7B%7D,'cvml','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
>>> <javascript:_e(%7B%7D,'cvml','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
>>> <javascript:_e(%7B%7D,'cvml','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
>>> <javascript:_e(%7B%7D,'cvml','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
>> <javascript:_e(%7B%7D,'cvml','sip:888 at conference.freeswitch.org');>  ?
>> +19193869900
>>
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','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
>> <javascript:_e(%7B%7D,'cvml','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 jurij.ivo at gmail.com  Wed Jul  1 11:36:01 2015
From: jurij.ivo at gmail.com (Jurijs Ivolga)
Date: Wed, 1 Jul 2015 10:36:01 +0300
Subject: [Freeswitch-users] 2 incoming calls on freeswitch with
	bypass_media
In-Reply-To: <CAHzfiZ5NsBS+qMpkBr2Kyxgvyb8xKyrRGwSSO9OXfBpKsiJObg@mail.gmail.com>
References: <CAGYpU67Yb8u1Cn8p4xtWAFnDyhQEWvQaiH6K-3g5oCoV8wLaRg@mail.gmail.com>
	<CALXCt0qQaDDgd_8Qo4c3dBGQCV5YuOdQExH6Rczm5gsp=C5G9w@mail.gmail.com>
	<CAGYpU65Z6Za+2V7_nX-TuYJaqQnfv0qx6_joy+3EikYZEGHrZg@mail.gmail.com>
	<CALXCt0pBnw1DZEGj_hcsFMVtBJB7LT9tO=s7DVmOVzqVBx2w_w@mail.gmail.com>
	<CAGYpU645G0jV-AAnv0L4WywoF11HO29pjQcyj13-13D3_0CZxg@mail.gmail.com>
	<CAHzfiZ5NsBS+qMpkBr2Kyxgvyb8xKyrRGwSSO9OXfBpKsiJObg@mail.gmail.com>
Message-ID: <CAGYpU65K5NihJtjOrpGigMmiuG-_5WwKNonEVxBm4hFxBtTxJA@mail.gmail.com>

Hi,

Thank you a lot, it works. :)

With kind regards,

Jurijs

2015-06-30 21:27 GMT+03:00 Russell Treleaven <rtreleaven at bunnykick.ca>:

> This should work https://wiki.freeswitch.org/wiki/Mod_commands#uuid_media
>
> On Tue, Jun 30, 2015 at 2:16 PM, Jurijs Ivolga <jurij.ivo at gmail.com>
> wrote:
>
>> Hi,
>>
>> So basically I need this:
>>
>> 1) User A will call freeswitch on number 9001 for example
>> 2) User B will call freeswitch too on number 9002 for example
>> 3) After this we need to bridge this 2 calls in 1 call (I'm doing this
>> with Lua script, using uuid_bridge command)
>> 4) As far as there will be a lot of such calls on Freeswitch I would like
>> to move media away from passing Freeswitch
>>
>> Is it possible to have in such case bypassing media or not?
>>
>> In all examples what I saw, there was always one incoming call and one
>> outgoing. So based on info what I got I started to think that this is
>> contrary with SIP RFC.
>>
>> Please help! :)
>>
>> Thank you!
>>
>> With kind regards,
>>
>> Jurijs
>>
>>
>> 2015-06-30 21:05 GMT+03:00 Giovanni Maruzzelli <gmaruzz at gmail.com>:
>>
>>> If they are not calling each other, then, why they would send media each
>>> other?
>>>
>>> Maybe I have not understood your use case
>>>
>>> -giovanni
>>>
>>> On Tue, Jun 30, 2015 at 7:54 PM, Jurijs Ivolga <jurij.ivo at gmail.com>
>>> wrote:
>>>
>>>> Hi,
>>>>
>>>> Thank you a lot for your input, but I would like to emphasize that I
>>>> need to have bypass media when 2 incoming calls are connected on
>>>> freeswitch. Link what you sent is for regular case, when user A calling to
>>>> user B. But I'm asking about a case when user A calling into freeswith and
>>>> user B calling into freeswitch. Then we will connect them in one call using
>>>> uuid_bridge for example.
>>>>
>>>> Is it possible to have media to bypass Freeswitch or not in such case?
>>>>
>>>> I tried to use  bypass_media_after_bridge, but it didn't worked for me.
>>>> On this point I think that in my case is not possible to bypass media, I
>>>> think it is against SIP RFC.
>>>>
>>>> If you have any ideas how to make it work, please share with me.
>>>>
>>>> Thank you a lot!
>>>>
>>>> With kind regards,
>>>>
>>>> Jurijs
>>>>
>>>> 2015-06-30 19:35 GMT+03:00 Giovanni Maruzzelli <gmaruzz at gmail.com>:
>>>>
>>>>> Yes, you can do it, and no, it's not against RFC
>>>>>
>>>>>
>>>>> https://freeswitch.org/confluence/display/FREESWITCH/Bypass+Media+Overview#BypassMediaOverview-Howtodisable/enableitonthefly
>>>>> ?
>>>>>
>>>>> On Tue, Jun 30, 2015 at 5:22 PM, Jurijs Ivolga <jurij.ivo at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Hi!
>>>>>>
>>>>>> I'm not sure that this is right place for this question, but here it
>>>>>> is:
>>>>>>
>>>>>> Is it possible to connect(using uuid_bridge for example) 2 incoming
>>>>>> calls on freeswitch and later to trigger re-invite messages, in such way
>>>>>> that media will start to go directly between endpoint bypassing freeswitch.
>>>>>>
>>>>>> I think this is impossible and it is contrary with SIP RFC.
>>>>>>
>>>>>> Maybe somebody can confirm that this is impossible or not. :) Thank
>>>>>> you 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
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> Cell : +39-347-2665618
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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
>>> Cell : +39-347-2665618
>>>
>>> _________________________________________________________________________
>>> 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 ssinyagin at gmail.com  Wed Jul  1 11:39:30 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 1 Jul 2015 09:39:30 +0200
Subject: [Freeswitch-users] Video playback in 1.7
In-Reply-To: <CAEwTsAiG6ob-YmFm++0ODxMNMtQQaUhf2XaGPCrkaJdhZAKmZg@mail.gmail.com>
References: <CACOYK=h8effa=OwvVweQT9CC3a4u3M8Ff7mzgaDHQ4di26u9Rw@mail.gmail.com>
	<CALXCt0qf+a1tYQt=Wb0DsdwDUCATwpLoRq7J2MeyHy+aHezwnw@mail.gmail.com>
	<CACOYK=jugw0Z5Z6SXh98P5Xd4gM0carohDJ2-_ftuev=jAAKCA@mail.gmail.com>
	<CAKbxfG8d8XpdRvSCq1ZYut71E5+bWQtPb8QnKXSkt-dHBdNXMw@mail.gmail.com>
	<CACOYK=hE=evF8c+HDjmQP1=2-ZCwSe0GAVkPkcjUCEsrv+vMuA@mail.gmail.com>
	<CAKbxfG_5J6uUmLCYPAUsa47dsSPx2A4JaHycGn-fB56nEFE-SA@mail.gmail.com>
	<CACOYK=hnVpaZNbXXTxaj-tcYBP6q_OL3fVxiggVN7=BUzDDVwg@mail.gmail.com>
	<CAKbxfG8xUg+hEoAp76pdhYuji5YA6WkcUJrfEfNpdby2TWx-Zw@mail.gmail.com>
	<CACOYK=jsR4xDPz-SoZ-YKY-J17BL+_ryCZiCso0xkO+dCvNdpw@mail.gmail.com>
	<1C0C06B1-187F-4D2D-A305-76763F975890@jerris.com>
	<CAKbxfG8028tSL-8wj-=qcNQ3xk08xM7uwCqXg-OSM_f=JFc5GQ@mail.gmail.com>
	<CACOYK=iwPiRRC6uMDiDm83MusFDLpipogmDnCJ5y6Xbd-u6q-w@mail.gmail.com>
	<CAEwTsAiG6ob-YmFm++0ODxMNMtQQaUhf2XaGPCrkaJdhZAKmZg@mail.gmail.com>
Message-ID: <CACOYK=hfnRoz5QWQaFf8cKNjRVeT1CiqF7_Yubn54Pow8uK46A@mail.gmail.com>

no, doesn't seem so:

root at fsdemo:~# aptitude search libav | egrep '^i'
i   libav-tools                     - Multimedia player, encoder and
transcoder
i A libavahi-client3                - Avahi client
library
i A libavahi-common-data            - Avahi common data
files
i A libavahi-common3                - Avahi common
library
i A libavc1394-0                    - control IEEE 1394 audio/video
devices
i A libavcodec56                    - Libav codec
library
i A libavdevice55                   - Libav device handling
library
i A libavfilter5                    - Libav video filtering
library
i A libavformat56                   - Libav file format
library
i A libavresample2                  - Libav audio resampling
library
i A libavutil54                     - Libav utility
library
root at fsdemo:~#




On Wed, Jul 1, 2015 at 8:23 AM, Michael Jerris <mike at jerris.com> wrote:

> do you have the libav -extra package (I don't recall the full name)
> installed?
>
>
> On Wednesday, July 1, 2015, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
>> This was all on a freshly installed Jessie, with your 1.7 debs. I will
>> have a closer look and document the issue later, now I only had time for a
>> quick check.
>> On Jul 1, 2015 3:18 AM, "Anthony Minessale" <anthony.minessale at gmail.com>
>> wrote:
>>
>>> Like I said, you need the precise versions we have detailed in our
>>> Debian jessie packaging.   It does not work on older versions of libav*
>>> If you don't want to use jessie you need to see the versions of
>>> everything we use and manually build it all and its full chain of cross
>>> depends.
>>>
>>>
>>> On Tue, Jun 30, 2015 at 7:54 PM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> That error happens when avresample_open call fails.  This is going to
>>>> be some sort of issue with how libav was built as this is known working.
>>>>
>>>> On Jun 30, 2015, at 7:51 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
>>>> wrote:
>>>>
>>>> I made a few more tests: the original file has AAC audio, and avconv
>>>> needs "-strict experimental" option to process that. Probably that's why
>>>> FreeSWITCH crashes. After I converted the video to 320x240 and MP3 audio, I
>>>> get a different error:
>>>>
>>>> 2015-07-01 01:46:22.701171 [ERR] avformat.c:1136 Failed to initialize
>>>> the resampling context
>>>>
>>>> same error if I need to produce 48kHz OPUS or 8kHZ G711.
>>>>
>>>> VLC plays back both the original and converted videos so far.
>>>>
>>>> I'll play around with it during the week.
>>>>
>>>>
>>>>
>>>>
>>>> On Tue, Jun 30, 2015 at 8:45 PM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> Depending on what ext you have and the proper build of mod_av I can't
>>>>> comment.
>>>>>
>>>>> I can confirm with the properly installed debian packages for jessie
>>>>> that at least mp4 files work either by reference to .mp4 files or by using
>>>>> av:///path/to/file.mp4
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jun 30, 2015 at 12:46 PM, Stanislav Sinyagin <
>>>>> ssinyagin at gmail.com> wrote:
>>>>>
>>>>>> av:///tmp/somefile.ext crashed the daemon, I didn't yet find the
>>>>>> time to analyze it and file a jira.
>>>>>> On Jun 30, 2015 7:43 PM, "Anthony Minessale" <
>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>
>>>>>>> mod_av is not an endpoint its a codec and file format module.
>>>>>>>
>>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:213
>>>>>>> Adding Codec H264 99 H264 Video 90000hz 0ms (VBR)
>>>>>>>
>>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:292
>>>>>>> Adding Application 'record_av'
>>>>>>>
>>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:338
>>>>>>> Adding API Function 'av_format'
>>>>>>>
>>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:338
>>>>>>> Adding API Function 'av_codec'
>>>>>>>
>>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390
>>>>>>> Adding File Format 'av'
>>>>>>>
>>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390
>>>>>>> Adding File Format 'rtmp'
>>>>>>>
>>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390
>>>>>>> Adding File Format 'mp4'
>>>>>>>
>>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390
>>>>>>> Adding File Format 'mov'
>>>>>>>
>>>>>>>
>>>>>>> It was av:// not avformat:// i was remembering the old version.
>>>>>>>
>>>>>>>
>>>>>>> if av:///tmp/somefile.ext works then it can be added to mod_avformat_load
>>>>>>> function in avformat.c:1949 ish  to register the extension for convenience.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Jun 30, 2015 at 2:07 AM, Stanislav Sinyagin <
>>>>>>> ssinyagin at gmail.com> wrote:
>>>>>>>
>>>>>>>> mod_av doesn't seem to be registering an endpoint, so prefixing
>>>>>>>> avformat:// or av:// does not help. I'll have a closer look later and
>>>>>>>> probably open a Jira
>>>>>>>>
>>>>>>>> On Tue, Jun 30, 2015 at 1:18 AM, Anthony Minessale <
>>>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Webm has its own module.  Av and vlc both have broken webm at the
>>>>>>>>> time of coding.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> mod_vlc can play other formats but they are not registered in the
>>>>>>>>> module by file exten however you can use vlc:// syntax.
>>>>>>>>>
>>>>>>>>> mp4 is the safest bet because it works in mod_av which is more
>>>>>>>>> stable than vlc.  More formats can be added to mod_av as well but I don't
>>>>>>>>> remember if its as easy as avformat://<path>
>>>>>>>>>
>>>>>>>>> We don't have any choosing best format etc.  It's not going to be
>>>>>>>>> a point of focus to squeeze performance out of stuff like that in this
>>>>>>>>> stage of development.
>>>>>>>>>
>>>>>>>>> On Monday, June 29, 2015, Stanislav Sinyagin <ssinyagin at gmail.com>
>>>>>>>>> wrote:
>>>>>>>>>
>>>>>>>>>> by the way, is there a way for playback to select a best matching
>>>>>>>>>> encoding, like it does with audio sample rates?
>>>>>>>>>>
>>>>>>>>>> On Tue, Jun 30, 2015 at 12:56 AM, Giovanni Maruzzelli <
>>>>>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>>>>> > h264 I believe is supported...
>>>>>>>>>> >
>>>>>>>>>> > On Tue, Jun 30, 2015 at 12:48 AM, Stanislav Sinyagin <
>>>>>>>>>> ssinyagin at gmail.com>
>>>>>>>>>> > wrote:
>>>>>>>>>> >>
>>>>>>>>>> >> the newest 1.7 freeswitch successfully played an .mp4 file with
>>>>>>>>>> >> "playback" application, and the picture was sent to an VP8
>>>>>>>>>> client
>>>>>>>>>> >> (linphone on Android).
>>>>>>>>>> >>
>>>>>>>>>> >> The playback took about 20% CPU usage on a Xeon core --
>>>>>>>>>> probably
>>>>>>>>>> >> because of resising work. The source file was taken from
>>>>>>>>>> >> http://www.quirksmode.org/html5/tests/video.html
>>>>>>>>>> >>
>>>>>>>>>> >>
>>>>>>>>>> >> Question: what other file formats are supported?
>>>>>>>>>> >>
>>>>>>>>>> >> I tried .ogv and .webm, but I got "Invalid file format" error.
>>>>>>>>>> >>
>>>>>>>>>> >> thanks
>>>>>>>>>> >>
>>>>>>>>>> >>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> >> 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
>>>>>>>>>> > Cell : +39-347-2665618
>>>>>>>>>> >
>>>>>>>>>> >
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> > 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
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> 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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> 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
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> 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
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> 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
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> 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
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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
>>>>
>>>
>>>
>>>
>>> --
>>> 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
>>>
>>> _________________________________________________________________________
>>> 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 ricardas.stoma at gmail.com  Wed Jul  1 11:17:54 2015
From: ricardas.stoma at gmail.com (=?UTF-8?Q?Ri=C4=8Dardas_Stoma?=)
Date: Wed, 1 Jul 2015 10:17:54 +0300
Subject: [Freeswitch-users] Asterisk canreinvite (directmedia) equivalent in
	freeswitch
Message-ID: <CAFX-kAeMb+Yz8h0++qmhmSjOvdcO6FMG_5n=MLvzZto7t+eFYA@mail.gmail.com>

Does freeswitch have anything similar to canreinvite (asterisk sip
setting)? I know there is bypass_media but it does not work in the same way
as canreinvite in asterisk.

It looks like bypass_media always tells freeswitch to send media directly
between originator and terminator even though they do not have common
codecs.

I want to use bypass_media only when leg A and leg B have common codecs. If
they do not have then freeswitch should stay in media path and handle codec
transcoding.

Is there an easy way to achieve this in freeswitch?
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From krice at freeswitch.org  Wed Jul  1 16:34:41 2015
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 01 Jul 2015 07:34:41 -0500
Subject: [Freeswitch-users] Asterisk canreinvite (directmedia)
 equivalent in freeswitch
In-Reply-To: <CAFX-kAfd+MwUKYnDwyEohni1fCuRHfSqXZKdvnKyVKU7F=g-yA@mail.gmail.com>
Message-ID: <D1B94891.A0FD7%krice@freeswitch.org>

Theres a few ways to accomplish this... But you are probably looking for
something like bypass_after_bridge

K 


On 6/30/15, 12:16 PM, "Ri?ardas Stoma" <ricardas.stoma at gmail.com> wrote:

> Does freeswitch have anything similar to canreinvite (asterisk sip setting)? I
> know there is bypass_media but it does not work in the same way as canreinvite
> in asterisk.
> 
> It looks like bypass_media always tells freeswitch to send media directly
> between originator and terminator even though they do not have common codecs.
> 
> I want to use bypass_media only when leg A and leg B have common codecs. If
> they do not have then freeswitch should stay in media path and handle codec
> transcoding.
> 
> Is there an easy way to achieve this in freeswitch??
> 
> 
> _________________________________________________________________________
> 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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From mcazzador at gmail.com  Wed Jul  1 16:35:18 2015
From: mcazzador at gmail.com (Matteo Cazzador)
Date: Wed, 1 Jul 2015 14:35:18 +0200
Subject: [Freeswitch-users] internal number blacklist
Message-ID: <CAEFBLw24khyi3QtmR-MCDD+195RE0-HYZuCC57UfpD=UAdiVTQ@mail.gmail.com>

HI, i'm a novice, i'm trying to make a dynamic blacklist
using mod_blacklist.
My scope is block call from internal number to external .
I need to create dynamic blacklist of internal number, not a blacklist of
external number.

Example i need to block dynamically internal number call to external
payment number (over a billing target)

Someone can help me please.

Thanks a lot.
-- 
Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
******************************************
Ing. Matteo Cazzador
Email: mcazzador at gmail.com
******************************************
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From ricardas.stoma at gmail.com  Wed Jul  1 17:06:34 2015
From: ricardas.stoma at gmail.com (=?UTF-8?Q?Ri=C4=8Dardas_Stoma?=)
Date: Wed, 1 Jul 2015 16:06:34 +0300
Subject: [Freeswitch-users] Asterisk canreinvite (directmedia)
 equivalent in freeswitch
In-Reply-To: <D1B94891.A0FD7%krice@freeswitch.org>
References: <CAFX-kAfd+MwUKYnDwyEohni1fCuRHfSqXZKdvnKyVKU7F=g-yA@mail.gmail.com>
	<D1B94891.A0FD7%krice@freeswitch.org>
Message-ID: <CAFX-kAcTdg2SJQO3Mui7wC84cxpmS0piGSfTV98NKQ=ir_X+8A@mail.gmail.com>

I tried bypass_media_after_bridge but calls still get connected with
different codecs and all i can hear is digital noise.

2015-07-01 15:34 GMT+03:00 Ken Rice <krice at freeswitch.org>:

>  Theres a few ways to accomplish this... But you are probably looking for
> something like bypass_after_bridge
>
> K
>
>
> On 6/30/15, 12:16 PM, "Ri?ardas Stoma" <ricardas.stoma at gmail.com> wrote:
>
> Does freeswitch have anything similar to canreinvite (asterisk sip
> setting)? I know there is bypass_media but it does not work in the same way
> as canreinvite in asterisk.
>
> It looks like bypass_media always tells freeswitch to send media directly
> between originator and terminator even though they do not have common
> codecs.
>
> I want to use bypass_media only when leg A and leg B have common codecs.
> If they do not have then freeswitch should stay in media path and handle
> codec transcoding.
>
> Is there an easy way to achieve this in freeswitch?
>
> ------------------------------
> _________________________________________________________________________
> 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
>
>
> --
> Ken
>
>
>
> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
> Twitter: @FreeSWITCH
>
>
> _________________________________________________________________________
> 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 freeswitch.org  Wed Jul  1 17:10:31 2015
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 01 Jul 2015 08:10:31 -0500
Subject: [Freeswitch-users] Asterisk canreinvite (directmedia)
 equivalent in freeswitch
In-Reply-To: <CAFX-kAcTdg2SJQO3Mui7wC84cxpmS0piGSfTV98NKQ=ir_X+8A@mail.gmail.com>
Message-ID: <D1B950F7.A0FDA%krice@freeswitch.org>

If you are hearing noise then you have a different problem... Post a full
trace of the call...

Sounds more like a broken endpoint.

And don't forget you probably want to do some things with the SDPs to make
sure you have matching codecs... You can do testing against them in the
dialplan and other ways


On 7/1/15, 8:06 AM, "Ri?ardas Stoma" <ricardas.stoma at gmail.com> wrote:

> I tried?bypass_media_after_bridge but calls still get connected with different
> codecs and all i can hear is digital noise.?
> 
> 2015-07-01 15:34 GMT+03:00 Ken Rice <krice at freeswitch.org>:
>> Theres a few ways to accomplish this... But you are probably looking for
>> something like bypass_after_bridge
>> 
>> K 
>> 
>> 
>> On 6/30/15, 12:16 PM, "Ri?ardas Stoma" <ricardas.stoma at gmail.com
>> <http://ricardas.stoma at gmail.com> > wrote:
>> 
>>> Does freeswitch have anything similar to canreinvite (asterisk sip setting)?
>>> I know there is bypass_media but it does not work in the same way as
>>> canreinvite in asterisk.
>>> 
>>> It looks like bypass_media always tells freeswitch to send media directly
>>> between originator and terminator even though they do not have common
>>> codecs.
>>> 
>>> I want to use bypass_media only when leg A and leg B have common codecs. If
>>> they do not have then freeswitch should stay in media path and handle codec
>>> transcoding.
>>> 
>>> Is there an easy way to achieve this in freeswitch??
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <http://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://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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From luis.daniel.lucio at gmail.com  Wed Jul  1 18:13:30 2015
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Wed, 1 Jul 2015 10:13:30 -0400
Subject: [Freeswitch-users] So you wanna setup your own CA for
	WSS/SSL/TLS?
In-Reply-To: <CAHKqMb72n4oOpScq7-JgYV0OAO+b=GuD1GoeYAhKYkAc8+a0sA@mail.gmail.com>
References: <CAEJMVkDH0tfmToSr=QooRf8GKum+Ru5WUAq6Lu1fDEXdpt06dg@mail.gmail.com>
	<53D2A7A7.9040400@quentustech.com>
	<CAEJMVkCdzFZrzfxuMXoGa=P1LVYcAOJJgKqLccZGBTdi2z0=ug@mail.gmail.com>
	<CAHKqMb72n4oOpScq7-JgYV0OAO+b=GuD1GoeYAhKYkAc8+a0sA@mail.gmail.com>
Message-ID: <CAFLo2QxgouBO=6vhyc0arcMhVQXhALd62XpF6OPPTJQT56G09g@mail.gmail.com>

More than a script, it would be better if you post minimum cert
requirements to let the FS work

Luis Daniel Lucio Quiroz
CISSP, CISM, CISA
Linux, VoIP and much more fun
www.okay.com.mx

Need LCR? Check out LCR for FusionPBX with FreeSWITCH
Need Billing? Check out Billing for FusionPBX with FreeSWITCH

2015-06-30 23:52 GMT-04:00 Nandy Dagondon <nandy1925 at gmail.com>:

> Hi Brian,
>
> I used your script to generate the certificates to test mod_verto in an
> intranet setup. Questions on your script:
>
> 1) Is 4096 bits required? Or 2048 bits will work, too?
> 2) Examining certs/wss.pem, there should be a  <chain> certificate at the
> end. But the script, inputs only 2 - *.crt and *.key. What should be the
> 3rd?
>
> Tks,
> /Nandy
>
>
> On Sat, Jul 26, 2014 at 2:59 AM, Brian West <brian at freeswitch.org> wrote:
>
>> I've corrected the how-to and put it in tree:
>>
>>
>> https://stash.freeswitch.org/projects/FS/repos/freeswitch/browse/docs/how_to_make_your_own_ca_correctly.txt?raw
>>
>> Importing the ca.crt into your system keychain for it to be trusted is
>> left to the end user to figure out.  If you can't do that step then you'll
>> kinda be SOL, I know on my Mac I just open ca.crt and it does the import
>> for me... Windows I suspect is similar as for Linux NO CLUE.
>>
>>
>> On Fri, Jul 25, 2014 at 1:53 PM, William King <
>> william.king at quentustech.com> wrote:
>>
>>> One correction inline, and did you have any luck getting chrome to work
>>> with the custom CA?
>>>
>>> William King
>>> Senior Engineer
>>> Quentus Technologies, INC
>>> 1037 NE 65th St Suite 273
>>> Seattle, WA 98115
>>> Main:   (877) 211-9337
>>> Office: (206) 388-4772
>>> Cell:   (253) 686-5518
>>> william.king at quentustech.com
>>>
>>> On 07/25/2014 08:12 AM, Brian West wrote:
>>> > Someone should probably turn this into a nice how-to:
>>> >
>>> > Here is how I did it.
>>> >
>>> > wget http://www.openssl.org/contrib/ssl.ca-0.1.tar.gz
>>> > tar zxfv ssl.ca-0.1.tar.gz
>>> > cd ssl.ca-0.1/
>>> > perl -i -pe 's/md5/sha1/g' *.sh
>>> > perl -i -pe 's/2048/2048/g' *.sh
>>> This is a noop. I assume it was suppose to be /2048/4096/ or /1024/2048/
>>> > ./new-root-ca.sh
>>> > ./new-server-cert.sh self.bkw.org <http://self.bkw.org>
>>> > ./sign-server-cert.sh self.bkw.org <http://self.bkw.org>
>>> > cat self.bkw.org.crt self.bkw.org.key >
>>> /usr/local/freeswitch/certs/wss.pem
>>> >
>>> > Setup Apache:
>>> >
>>> > default-ssl:
>>> >
>>> > SSLCertificateFile    /usr/local/freeswitch/certs/wss.pem
>>> > SSLCertificateKeyFile /usr/local/freeswitch/certs/wss.pem
>>> > SSLCertificateChainFile /usr/local/freeswitch/certs/wss.pem
>>> >
>>> > Setup Sofia TLS:
>>> >
>>> > cat self.bkw.org.crt self.bkw.org.key >
>>> > /usr/local/freeswitch/certs/agent.pem
>>> > cat ca.crt > /usr/local/freeswitch/certs/cafile.pem
>>> >
>>> > vars.xml:
>>> >
>>> > <X-PRE-PROCESScmd="set"data="internal_ssl_enable=true"/>
>>> > <X-PRE-PROCESScmd="set"data="external_ssl_enable=true"/>
>>> >
>>> > Restart FreeSWITCH.
>>> >
>>> > Now make sure your system has ca.crt imported so it will trust your new
>>> > found hotness.
>>> >
>>> > TEST:
>>> >
>>> > openssl s_client -connect self.bkw.org:443 <http://self.bkw.org:443>
>>> > openssl s_client -connect self.bkw.org:8082 <http://self.bkw.org:8082>
>>> >
>>> >
>>> > Depending on what you've setup you'll see:
>>> >
>>> > subject=/C=US/ST=Oklahoma/L=McAlester/O=Tonka Truck/OU=Secure Web
>>> > Server/CN=self.bkw.org/emailAddress=brian at bkw.org
>>> > <http://self.bkw.org/emailAddress=brian at bkw.org>
>>> >
>>> > issuer=/C=US/ST=Oklahoma/L=McAlester/O=Whizzzzzzy Bang
>>> > Bang/OU=Certification Services Division/CN=WBB Root
>>> > CA/emailAddress=brian at bkw.org <mailto:brian at bkw.org>
>>> >
>>> > Or there abouts.
>>> >
>>> > --
>>> >
>>> > */Brian West/*
>>> > brian at freeswitch.org <mailto:brian at freeswitch.org>
>>> >
>>> >
>>> > */Twitter: @FreeSWITCH , @briankwest/*
>>> > http://www.freeswitchbook.com
>>> > http://www.freeswitchcookbook.com
>>> >
>>> > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>> >
>>> >
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>> > http://www.cudatel.com
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>> http://www.cudatel.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>> http://www.cudatel.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> 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 wiebittewas at googlemail.com  Wed Jul  1 18:37:37 2015
From: wiebittewas at googlemail.com (wiebittewas)
Date: Wed, 01 Jul 2015 16:37:37 +0200
Subject: [Freeswitch-users] no display-update after transfer
In-Reply-To: <55935A7D.2010701@gmail.com>
References: <55935A7D.2010701@gmail.com>
Message-ID: <5593FB31.40105@gmail.com>

Am 01.07.2015 05:11 schrieb "wiebittewas" bzgl. "no display-update after transfer":

> [ALERT] sofia.c:1198 sofia_update_callee_id() sofia/internal/u620 at 192.168.1.4:5060 Same Callee ID "Outbound Call" <20>

ok, checked this in detail - this is created only after the invites, not the refer, so it has nothing to do with the refer.



Does no-one here use attended/blind transfer and has either a working display-update or can say, if freeswitch is at least able to send it?


regards

w.



From s.safarov at gmail.com  Wed Jul  1 19:38:17 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 01 Jul 2015 15:38:17 +0000
Subject: [Freeswitch-users] internal number blacklist
In-Reply-To: <CAEFBLw24khyi3QtmR-MCDD+195RE0-HYZuCC57UfpD=UAdiVTQ@mail.gmail.com>
References: <CAEFBLw24khyi3QtmR-MCDD+195RE0-HYZuCC57UfpD=UAdiVTQ@mail.gmail.com>
Message-ID: <CAHtxdDewfEwGA6PZntY150Qp7kRL9E+FydRxr3P+s09joHo+JA@mail.gmail.com>

What is type of external billing system? It is RADIUS server?

On Wed, Jul 1, 2015, 15:36 Matteo Cazzador <mcazzador at gmail.com> wrote:

> HI, i'm a novice, i'm trying to make a dynamic blacklist
> using mod_blacklist.
> My scope is block call from internal number to external .
> I need to create dynamic blacklist of internal number, not a blacklist of
> external number.
>
> Example i need to block dynamically internal number call to external
> payment number (over a billing target)
>
> Someone can help me please.
>
> Thanks a lot.
> --
> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
> ******************************************
> Ing. Matteo Cazzador
> Email: mcazzador at gmail.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
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From mcazzador at gmail.com  Wed Jul  1 19:45:02 2015
From: mcazzador at gmail.com (Matteo Cazzador)
Date: Wed, 1 Jul 2015 17:45:02 +0200
Subject: [Freeswitch-users] internal number blacklist
In-Reply-To: <CAHtxdDewfEwGA6PZntY150Qp7kRL9E+FydRxr3P+s09joHo+JA@mail.gmail.com>
References: <CAEFBLw24khyi3QtmR-MCDD+195RE0-HYZuCC57UfpD=UAdiVTQ@mail.gmail.com>
	<CAHtxdDewfEwGA6PZntY150Qp7kRL9E+FydRxr3P+s09joHo+JA@mail.gmail.com>
Message-ID: <CAEFBLw3AntWJGeMVhRyE+HuTkTn-gxTKTgF6_8X3zsNFjq4x1g@mail.gmail.com>

Hi, i explain better, i've a freeswitch fax machine,
I collect cdrs data with cdrs-sqlite, then i 've an external script (php)
part that read table cdrs and calculate the billing.
If a calling number it's over a threshold i want to disable outbound
calling from specific number.


2015-07-01 17:38 GMT+02:00 Sergey Safarov <s.safarov at gmail.com>:

> What is type of external billing system? It is RADIUS server?
>
> On Wed, Jul 1, 2015, 15:36 Matteo Cazzador <mcazzador at gmail.com> wrote:
>
>> HI, i'm a novice, i'm trying to make a dynamic blacklist
>> using mod_blacklist.
>> My scope is block call from internal number to external .
>> I need to create dynamic blacklist of internal number, not a blacklist of
>> external number.
>>
>> Example i need to block dynamically internal number call to external
>> payment number (over a billing target)
>>
>> Someone can help me please.
>>
>> Thanks a lot.
>> --
>> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
>> ******************************************
>> Ing. Matteo Cazzador
>> Email: mcazzador at gmail.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
>
>
> _________________________________________________________________________
> 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
>



-- 
Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
******************************************
Ing. Matteo Cazzador
Email: mcazzador at gmail.com
******************************************
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From mcazzador at gmail.com  Wed Jul  1 19:49:48 2015
From: mcazzador at gmail.com (Matteo Cazzador)
Date: Wed, 1 Jul 2015 17:49:48 +0200
Subject: [Freeswitch-users] internal number blacklist
In-Reply-To: <CAEFBLw3AntWJGeMVhRyE+HuTkTn-gxTKTgF6_8X3zsNFjq4x1g@mail.gmail.com>
References: <CAEFBLw24khyi3QtmR-MCDD+195RE0-HYZuCC57UfpD=UAdiVTQ@mail.gmail.com>
	<CAHtxdDewfEwGA6PZntY150Qp7kRL9E+FydRxr3P+s09joHo+JA@mail.gmail.com>
	<CAEFBLw3AntWJGeMVhRyE+HuTkTn-gxTKTgF6_8X3zsNFjq4x1g@mail.gmail.com>
Message-ID: <CAEFBLw0=kZ0YqC2D+EsmWwF15OnbcLiK4tvzQv-yN8TH1Ln+QQ@mail.gmail.com>

Hi, the calling number is the number that i want to put in blacklist.
Only for outbound call. Tipically outbound call versus expensive foreign
number.
I want to block it.  Better i want to block cheating calling.


2015-07-01 17:45 GMT+02:00 Matteo Cazzador <mcazzador at gmail.com>:

> Hi, i explain better, i've a freeswitch fax machine,
> I collect cdrs data with cdrs-sqlite, then i 've an external script (php)
> part that read table cdrs and calculate the billing.
> If a calling number it's over a threshold i want to disable outbound
> calling from specific number.
>
>
> 2015-07-01 17:38 GMT+02:00 Sergey Safarov <s.safarov at gmail.com>:
>
>> What is type of external billing system? It is RADIUS server?
>>
>> On Wed, Jul 1, 2015, 15:36 Matteo Cazzador <mcazzador at gmail.com> wrote:
>>
>>> HI, i'm a novice, i'm trying to make a dynamic blacklist
>>> using mod_blacklist.
>>> My scope is block call from internal number to external .
>>> I need to create dynamic blacklist of internal number, not a blacklist
>>> of external number.
>>>
>>> Example i need to block dynamically internal number call to external
>>> payment number (over a billing target)
>>>
>>> Someone can help me please.
>>>
>>> Thanks a lot.
>>> --
>>> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
>>> ******************************************
>>> Ing. Matteo Cazzador
>>> Email: mcazzador at gmail.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
>>
>>
>> _________________________________________________________________________
>> 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
>>
>
>
>
> --
> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
> ******************************************
> Ing. Matteo Cazzador
> Email: mcazzador at gmail.com
> ******************************************
>



-- 
Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
******************************************
Ing. Matteo Cazzador
Email: mcazzador at gmail.com
******************************************
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From rtreleaven at bunnykick.ca  Wed Jul  1 19:55:44 2015
From: rtreleaven at bunnykick.ca (Russell Treleaven)
Date: Wed, 1 Jul 2015 11:55:44 -0400
Subject: [Freeswitch-users] internal number blacklist
In-Reply-To: <CAEFBLw3AntWJGeMVhRyE+HuTkTn-gxTKTgF6_8X3zsNFjq4x1g@mail.gmail.com>
References: <CAEFBLw24khyi3QtmR-MCDD+195RE0-HYZuCC57UfpD=UAdiVTQ@mail.gmail.com>
	<CAHtxdDewfEwGA6PZntY150Qp7kRL9E+FydRxr3P+s09joHo+JA@mail.gmail.com>
	<CAEFBLw3AntWJGeMVhRyE+HuTkTn-gxTKTgF6_8X3zsNFjq4x1g@mail.gmail.com>
Message-ID: <CAHzfiZ4YOVJfFAr-p7hJryO1QGPYarcCWw+GYk+h3KG9hn7rLQ@mail.gmail.com>

Why not just bridge the call or not based on  $rate > threshold?
This can all be done in the dialplan or script called from the dialplan.


On Wed, Jul 1, 2015 at 11:45 AM, Matteo Cazzador <mcazzador at gmail.com>
wrote:

> Hi, i explain better, i've a freeswitch fax machine,
> I collect cdrs data with cdrs-sqlite, then i 've an external script (php)
> part that read table cdrs and calculate the billing.
> If a calling number it's over a threshold i want to disable outbound
> calling from specific number.
>
>
> 2015-07-01 17:38 GMT+02:00 Sergey Safarov <s.safarov at gmail.com>:
>
>> What is type of external billing system? It is RADIUS server?
>>
>> On Wed, Jul 1, 2015, 15:36 Matteo Cazzador <mcazzador at gmail.com> wrote:
>>
>>> HI, i'm a novice, i'm trying to make a dynamic blacklist
>>> using mod_blacklist.
>>> My scope is block call from internal number to external .
>>> I need to create dynamic blacklist of internal number, not a blacklist
>>> of external number.
>>>
>>> Example i need to block dynamically internal number call to external
>>> payment number (over a billing target)
>>>
>>> Someone can help me please.
>>>
>>> Thanks a lot.
>>> --
>>> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
>>> ******************************************
>>> Ing. Matteo Cazzador
>>> Email: mcazzador at gmail.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
>>
>>
>> _________________________________________________________________________
>> 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
>>
>
>
>
> --
> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
> ******************************************
> Ing. Matteo Cazzador
> Email: mcazzador at gmail.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
>
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From luis.daniel.lucio at gmail.com  Wed Jul  1 19:57:56 2015
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Wed, 1 Jul 2015 11:57:56 -0400
Subject: [Freeswitch-users] internal number blacklist
In-Reply-To: <CAEFBLw0=kZ0YqC2D+EsmWwF15OnbcLiK4tvzQv-yN8TH1Ln+QQ@mail.gmail.com>
References: <CAEFBLw24khyi3QtmR-MCDD+195RE0-HYZuCC57UfpD=UAdiVTQ@mail.gmail.com>
	<CAHtxdDewfEwGA6PZntY150Qp7kRL9E+FydRxr3P+s09joHo+JA@mail.gmail.com>
	<CAEFBLw3AntWJGeMVhRyE+HuTkTn-gxTKTgF6_8X3zsNFjq4x1g@mail.gmail.com>
	<CAEFBLw0=kZ0YqC2D+EsmWwF15OnbcLiK4tvzQv-yN8TH1Ln+QQ@mail.gmail.com>
Message-ID: <CAFLo2Qz8YPqT+SKizc=Eau9sx78-zRDfkdCY=v9nyfKr3ftZqA@mail.gmail.com>

I think this is easier with dialpla & lua scripting
On Jul 1, 2015 11:51 AM, "Matteo Cazzador" <mcazzador at gmail.com> wrote:

> Hi, the calling number is the number that i want to put in blacklist.
> Only for outbound call. Tipically outbound call versus expensive foreign
> number.
> I want to block it.  Better i want to block cheating calling.
>
>
> 2015-07-01 17:45 GMT+02:00 Matteo Cazzador <mcazzador at gmail.com>:
>
>> Hi, i explain better, i've a freeswitch fax machine,
>> I collect cdrs data with cdrs-sqlite, then i 've an external script (php)
>> part that read table cdrs and calculate the billing.
>> If a calling number it's over a threshold i want to disable outbound
>> calling from specific number.
>>
>>
>> 2015-07-01 17:38 GMT+02:00 Sergey Safarov <s.safarov at gmail.com>:
>>
>>> What is type of external billing system? It is RADIUS server?
>>>
>>> On Wed, Jul 1, 2015, 15:36 Matteo Cazzador <mcazzador at gmail.com> wrote:
>>>
>>>> HI, i'm a novice, i'm trying to make a dynamic blacklist
>>>> using mod_blacklist.
>>>> My scope is block call from internal number to external .
>>>> I need to create dynamic blacklist of internal number, not a blacklist
>>>> of external number.
>>>>
>>>> Example i need to block dynamically internal number call to external
>>>> payment number (over a billing target)
>>>>
>>>> Someone can help me please.
>>>>
>>>> Thanks a lot.
>>>> --
>>>> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
>>>> ******************************************
>>>> Ing. Matteo Cazzador
>>>> Email: mcazzador at gmail.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
>>>
>>>
>>> _________________________________________________________________________
>>> 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
>>>
>>
>>
>>
>> --
>> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
>> ******************************************
>> Ing. Matteo Cazzador
>> Email: mcazzador at gmail.com
>> ******************************************
>>
>
>
>
> --
> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
> ******************************************
> Ing. Matteo Cazzador
> Email: mcazzador at gmail.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
>
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From lylepratt at gmail.com  Wed Jul  1 19:58:10 2015
From: lylepratt at gmail.com (Lyle Pratt)
Date: Wed, 1 Jul 2015 10:58:10 -0500
Subject: [Freeswitch-users] Presence Issues with Secure Websockets
Message-ID: <CAFUDvVj2qnsxqsf1tp-V5kZbcS56dJAAo=TB-cLbtGm1ti7Byg@mail.gmail.com>

We opened a JIRA issue. Has anyone looked into this?
Presence events over WSS are basically broken.
https://freeswitch.org/jira/browse/FS-7680

Thanks,
Lyle Pratt
BetterVoice.com


Michael Jerris <mike at ...> writes:
>
> If we are not sending the NOTIFY, please open a jira.  Will need to
include full sip
debug with sofia debug and sip trace turned up.
>
> On Jun 10, 2015, at 1:11 PM, Thomas Quintana <thomas-
78QPtSbfA2Jofqt5WZNmog at public.gmane.org> wrote:
> Hey Guys,
>
> We're subscribing to presence events via SIP.js using a secure websocket
(WSS)
and we can see the subscriptions being received by FreeSWITCH with a 202
Accepted, but never receive any events via the notify event. Alternatively,
when we
move to a plain websocket (WS) everything seems to work as expected.
>
>
>
> After digging around we found that a few people were experiencing similar
issues
with FreeSWITCH and additionally with Kamailio http://lists.sip-
router.org/pipermail/sr-users/2014-July/084157.html. The issue was resolved
with
Kamailio by patching the contact header. In an attempt to achieve the same
results
we have experimented with the settings below but no success.
>
>
>
>
> <param name="NDLB-force-rport" value="safe"/>
>
>
> <param name="NDLB-received-in-nat-reg-contact" value="true"/>
>
>
> We also have an open ticket with the folks over at
SIP.js https://github.com/onsip/SIP.js/issues/184. Any help would be
greatly
appreciated.
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at ...
> 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 ...
> 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 mcazzador at gmail.com  Wed Jul  1 20:01:57 2015
From: mcazzador at gmail.com (Matteo Cazzador)
Date: Wed, 1 Jul 2015 18:01:57 +0200
Subject: [Freeswitch-users] internal number blacklist
In-Reply-To: <CAFLo2Qz8YPqT+SKizc=Eau9sx78-zRDfkdCY=v9nyfKr3ftZqA@mail.gmail.com>
References: <CAEFBLw24khyi3QtmR-MCDD+195RE0-HYZuCC57UfpD=UAdiVTQ@mail.gmail.com>
	<CAHtxdDewfEwGA6PZntY150Qp7kRL9E+FydRxr3P+s09joHo+JA@mail.gmail.com>
	<CAEFBLw3AntWJGeMVhRyE+HuTkTn-gxTKTgF6_8X3zsNFjq4x1g@mail.gmail.com>
	<CAEFBLw0=kZ0YqC2D+EsmWwF15OnbcLiK4tvzQv-yN8TH1Ln+QQ@mail.gmail.com>
	<CAFLo2Qz8YPqT+SKizc=Eau9sx78-zRDfkdCY=v9nyfKr3ftZqA@mail.gmail.com>
Message-ID: <CAEFBLw3LavdOn_xq9hE-ortZGzW41+CA7V-O=_VrHkw7u-8uJA@mail.gmail.com>

Hi thanks a lot , there is some example about it?

2015-07-01 17:57 GMT+02:00 Luis Daniel Lucio Quiroz <
luis.daniel.lucio at gmail.com>:

> I think this is easier with dialpla & lua scripting
> On Jul 1, 2015 11:51 AM, "Matteo Cazzador" <mcazzador at gmail.com> wrote:
>
>> Hi, the calling number is the number that i want to put in blacklist.
>> Only for outbound call. Tipically outbound call versus expensive foreign
>> number.
>> I want to block it.  Better i want to block cheating calling.
>>
>>
>> 2015-07-01 17:45 GMT+02:00 Matteo Cazzador <mcazzador at gmail.com>:
>>
>>> Hi, i explain better, i've a freeswitch fax machine,
>>> I collect cdrs data with cdrs-sqlite, then i 've an external script
>>> (php) part that read table cdrs and calculate the billing.
>>> If a calling number it's over a threshold i want to disable outbound
>>> calling from specific number.
>>>
>>>
>>> 2015-07-01 17:38 GMT+02:00 Sergey Safarov <s.safarov at gmail.com>:
>>>
>>>> What is type of external billing system? It is RADIUS server?
>>>>
>>>> On Wed, Jul 1, 2015, 15:36 Matteo Cazzador <mcazzador at gmail.com> wrote:
>>>>
>>>>> HI, i'm a novice, i'm trying to make a dynamic blacklist
>>>>> using mod_blacklist.
>>>>> My scope is block call from internal number to external .
>>>>> I need to create dynamic blacklist of internal number, not a blacklist
>>>>> of external number.
>>>>>
>>>>> Example i need to block dynamically internal number call to external
>>>>> payment number (over a billing target)
>>>>>
>>>>> Someone can help me please.
>>>>>
>>>>> Thanks a lot.
>>>>> --
>>>>> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
>>>>> ******************************************
>>>>> Ing. Matteo Cazzador
>>>>> Email: mcazzador at gmail.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
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>>>>
>>>
>>>
>>>
>>> --
>>> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
>>> ******************************************
>>> Ing. Matteo Cazzador
>>> Email: mcazzador at gmail.com
>>> ******************************************
>>>
>>
>>
>>
>> --
>> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
>> ******************************************
>> Ing. Matteo Cazzador
>> Email: mcazzador at gmail.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
>>
>
> _________________________________________________________________________
> 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
>



-- 
Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
******************************************
Ing. Matteo Cazzador
Email: mcazzador at gmail.com
******************************************
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From s.safarov at gmail.com  Wed Jul  1 20:04:08 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 01 Jul 2015 16:04:08 +0000
Subject: [Freeswitch-users] internal number blacklist
In-Reply-To: <CAEFBLw3AntWJGeMVhRyE+HuTkTn-gxTKTgF6_8X3zsNFjq4x1g@mail.gmail.com>
References: <CAEFBLw24khyi3QtmR-MCDD+195RE0-HYZuCC57UfpD=UAdiVTQ@mail.gmail.com>
	<CAHtxdDewfEwGA6PZntY150Qp7kRL9E+FydRxr3P+s09joHo+JA@mail.gmail.com>
	<CAEFBLw3AntWJGeMVhRyE+HuTkTn-gxTKTgF6_8X3zsNFjq4x1g@mail.gmail.com>
Message-ID: <CAHtxdDfy_YgrfWHDjY-6Pr4eXgAsgC+bNfCm=a2hBdAuWHTe2A@mail.gmail.com>

Try mod_curl
https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/3965033
and analyze curl_response_data or curl_response_code

On Wed, Jul 1, 2015, 18:46 Matteo Cazzador <mcazzador at gmail.com> wrote:

> Hi, i explain better, i've a freeswitch fax machine,
> I collect cdrs data with cdrs-sqlite, then i 've an external script (php)
> part that read table cdrs and calculate the billing.
> If a calling number it's over a threshold i want to disable outbound
> calling from specific number.
>
>
> 2015-07-01 17:38 GMT+02:00 Sergey Safarov <s.safarov at gmail.com>:
>
>> What is type of external billing system? It is RADIUS server?
>>
>> On Wed, Jul 1, 2015, 15:36 Matteo Cazzador <mcazzador at gmail.com> wrote:
>>
>>> HI, i'm a novice, i'm trying to make a dynamic blacklist
>>> using mod_blacklist.
>>> My scope is block call from internal number to external .
>>> I need to create dynamic blacklist of internal number, not a blacklist
>>> of external number.
>>>
>>> Example i need to block dynamically internal number call to external
>>> payment number (over a billing target)
>>>
>>> Someone can help me please.
>>>
>>> Thanks a lot.
>>> --
>>> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
>>> ******************************************
>>> Ing. Matteo Cazzador
>>> Email: mcazzador at gmail.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
>>
>>
>> _________________________________________________________________________
>> 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
>>
>
>
>
> --
> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
> ******************************************
> Ing. Matteo Cazzador
> Email: mcazzador at gmail.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
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From mcazzador at gmail.com  Wed Jul  1 20:05:22 2015
From: mcazzador at gmail.com (Matteo Cazzador)
Date: Wed, 1 Jul 2015 18:05:22 +0200
Subject: [Freeswitch-users] internal number blacklist
In-Reply-To: <CAHtxdDfy_YgrfWHDjY-6Pr4eXgAsgC+bNfCm=a2hBdAuWHTe2A@mail.gmail.com>
References: <CAEFBLw24khyi3QtmR-MCDD+195RE0-HYZuCC57UfpD=UAdiVTQ@mail.gmail.com>
	<CAHtxdDewfEwGA6PZntY150Qp7kRL9E+FydRxr3P+s09joHo+JA@mail.gmail.com>
	<CAEFBLw3AntWJGeMVhRyE+HuTkTn-gxTKTgF6_8X3zsNFjq4x1g@mail.gmail.com>
	<CAHtxdDfy_YgrfWHDjY-6Pr4eXgAsgC+bNfCm=a2hBdAuWHTe2A@mail.gmail.com>
Message-ID: <CAEFBLw2mNJbWMPwKbU0YV3aJsop-x-5veYupWvEQZA5F=eCD_w@mail.gmail.com>

thanks a lot!!!

2015-07-01 18:04 GMT+02:00 Sergey Safarov <s.safarov at gmail.com>:

> Try mod_curl
>
> https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/3965033
> and analyze curl_response_data or curl_response_code
>
> On Wed, Jul 1, 2015, 18:46 Matteo Cazzador <mcazzador at gmail.com> wrote:
>
>> Hi, i explain better, i've a freeswitch fax machine,
>> I collect cdrs data with cdrs-sqlite, then i 've an external script (php)
>> part that read table cdrs and calculate the billing.
>> If a calling number it's over a threshold i want to disable outbound
>> calling from specific number.
>>
>>
>> 2015-07-01 17:38 GMT+02:00 Sergey Safarov <s.safarov at gmail.com>:
>>
>>> What is type of external billing system? It is RADIUS server?
>>>
>>> On Wed, Jul 1, 2015, 15:36 Matteo Cazzador <mcazzador at gmail.com> wrote:
>>>
>>>> HI, i'm a novice, i'm trying to make a dynamic blacklist
>>>> using mod_blacklist.
>>>> My scope is block call from internal number to external .
>>>> I need to create dynamic blacklist of internal number, not a blacklist
>>>> of external number.
>>>>
>>>> Example i need to block dynamically internal number call to external
>>>> payment number (over a billing target)
>>>>
>>>> Someone can help me please.
>>>>
>>>> Thanks a lot.
>>>> --
>>>> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
>>>> ******************************************
>>>> Ing. Matteo Cazzador
>>>> Email: mcazzador at gmail.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
>>>
>>>
>>> _________________________________________________________________________
>>> 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
>>>
>>
>>
>>
>> --
>> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
>> ******************************************
>> Ing. Matteo Cazzador
>> Email: mcazzador at gmail.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
>
>
> _________________________________________________________________________
> 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
>



-- 
Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
******************************************
Ing. Matteo Cazzador
Email: mcazzador at gmail.com
******************************************
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From krice at freeswitch.org  Wed Jul  1 21:09:48 2015
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 01 Jul 2015 17:09:48 +0000
Subject: [Freeswitch-users] The FreeSWITCH 1.4.20 release is here!
Message-ID: <55941edc6f234_d2c17b732087359@resque-worker-high.1.mail>

New Post on freeswitch.org from Kathleen
check it out at http://ift.tt/1dy0Hes
The FreeSWITCH 1.4.20 release is here!
The FreeSWITCH 1.4.20 release is here!

This is a routine maintenance release and the resources are located here:

Tarballs: http://ift.tt/1NwaFd8

Packaging: http://ift.tt/1U8FMQN

Security issues:

FS-7708 Fixed docs on enabling cert CN/SAN validation

New features that were added:

FS-7561 [mod_sofia] Add Perfect Forward Secrecy (DHE PFS)

FS-7564 [mod_rayo] Added new algorithms for offering calls to clients

FS-7623 [mod_amqp] Allow for custom exchange name and type for producers and fixed param name ordering bug caused by exposing these params

FS-7720 Improve play_and_detect_speech to set current_application_response channel variable as follows: ?USAGE ERROR?: bad application arguments?, ?GRAMMAR ERROR?: speech recognizer failed to load grammar, ?ASR INIT ERROR?: speech recognizer failed to allocate a session, and ?ERROR?: any other errors

FS-7743 [mod_skinny] Updated SKINNY on-hook action to hang up all calls on a device, except those in a short list of call states (or perform a blind transfer) and added a hook after completing the hangup operation to start ringing if there is an inbound call active on the device.

Improvements in build system, cross platform support, and packaging:

FS-7610 Fixed a gcc5 compilation issue

FS-7426 Only disable mod_amqp on Debian Squeeze and Wheezy

FS-7297 g729 installer

The following bugs were squashed:

FS-7582 FS-7432 Fixed missing a=setup parameter from answering SDP

FS-7650 [mod_verto] Fixed crash when making a call from a verto user with profile-variables in their user profile

FS-7678 Fixed for fail_on_single_reject not working with | bridge

FS-7612 Fixed invalid json format for callflow key

FS-7621 [mod_shout] Fixed a slow interrupt

FS-7432 Fixed missing a=setup parameter from answering SDP

FS-7573 Fixed 80bit tag support for zrtp

FS-7636 Fixed an issue with transfer_after_bridge and park_after_bridge pre-empting transfers

FS-7654 Fixed an issue with eavesdrop audio not working correctly with a mixture of mono and stereo

FS-7579 [mod_conference] Fixed a bug not allowing suppression of play-file-done

FS-7593 [mod_skinny] Fixed a bug where skinny phones would stomp on each other in database when thundering herd occurs

FS-7597 [mod_codec2] Fixed encoded_data_len for MODE 2400, it should be 6 bytes. Also replaced 2550 bps bitrate (obsoleted operation mode) by 2400

FS-7604 [fs_cli] Fixed fs_cli tab completion concurrency issues on newer libedit

FS-7258 FS-7571 [mod_xml_cdr] Properly encode xml cdr for post to web server

FS-7607 Update URLs to reflect https protocol on freeswitch.org websites and update additional URLs to avoid 301 redirects.

FS-7479 Fixed a crash caused by large RTP/PCMA packets and resampling

FS-7524 [mod_callcenter] Fixing tiers, level and position should default to 1 instead of 0

FS-7622 [mod_amqp] Make sure to close the connections on destroy. Currently the connection is malloc?d from the module pool, so there is nothing to destroy.

FS-7689 [mod_lua] Fixed a bug with lua not loading directory configurations

FS-7489 [mod_unimrcp] Fixed a TTS Audio Queue Overflow

FS-7467 [mod_callcenter] Fixing stuck channels using uuid-standby agents

FS-7429 [mod_curl] Fixed a json formatting error

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From wiebittewas at googlemail.com  Thu Jul  2 00:37:24 2015
From: wiebittewas at googlemail.com (wiebittewas)
Date: Wed, 01 Jul 2015 22:37:24 +0200
Subject: [Freeswitch-users] no display-update after transfer
In-Reply-To: <5593FB31.40105@gmail.com>
References: <55935A7D.2010701@gmail.com> <5593FB31.40105@gmail.com>
Message-ID: <55944F84.6070301@gmail.com>

Am 01.07.2015 16:37 schrieb "wiebittewas" bzgl. "Re: no display-update after transfer":

> Does no-one here use attended/blind transfer and has either a working display-update or can say, if freeswitch is at least able to send it?

ok, found the reasons in code.

in sofia_receive_message() case SWITCH_MESSAGE_INDICATE_DISPLAY is checked which system may be on the other end.

there are entries for freeswitch, snom, polycom, aastra, cisco and yealink, but no default at all, so there won't be any sip-msg put onto line for other
would be nice to have the possibility to select one of these items for local users by setting a config-var.

well, after modifying the code here and adding additional calls to sofia_send_callee_id() during transfer-handling fixed this issue.

after the current project has been finished I'll file a bug for this issue.

regards

w.



From grcamauer at gmail.com  Thu Jul  2 02:53:30 2015
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Wed, 1 Jul 2015 19:53:30 -0300
Subject: [Freeswitch-users] Make current failing on Certificates
Message-ID: <CAPNTnnpFn4O=9ATPyehKKXPizKOoGAW2hJyiikaqE43X64+pXw@mail.gmail.com>

I just tried to "make current" and got this:

error: server certificate verification failed. CAfile:
/etc/ssl/certs/ca-certificates.crt CRLfile: none while accessing
https://stash.freeswitch.org/scm/fs/freeswitch.git/info/refs
fatal: HTTP request failed
make[1]: *** [update] Error 1
make[1]: Leaving directory `/usr/src/freeswitch'
make: *** [current] Error 2


-- 
Guillermo Ruiz Camauer
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From rblock at jupiter.com  Thu Jul  2 03:18:12 2015
From: rblock at jupiter.com (Rick Block)
Date: Wed, 1 Jul 2015 23:18:12 +0000
Subject: [Freeswitch-users] RFC 4579 and FreeSWITCH
Message-ID: <C532722C5C191649AF1D91D7D6CDF4D10114135E@mbx029-w1-ca-10.exch029.domain.local>

A while ago there was a question about using REFER (RFC 4579 section 5.5) to add a new participant to a conference, see

http://lists.freeswitch.org/pipermail/freeswitch-users/2012-December/090370.html

Can someone who is successfully using this post a howto including the relevant config, and exact REFER that works?

We're getting a 202 Accepted in response to an out of dialog REFER, but no INVITE is being sent from FreeSWITCH.

Our config and REFER are as below.

Thanks,
Rick Block

    <extension name="rfc4579_conferences">
      <condition field="destination_number" expression="^(conf4579-.+)">
        <action application="answer" data="is_conference"/>
      <action application="conference" data="$1-${domain_name}@cdquality++flags{rfc-4579}"/>
      </condition>
    </extension>

    To: sip:conf4579-1 at 10.4.0.7;isfocus
    From: sip:phil at 10.4.6.31;tag=329478
    Call-ID: 863823
    CSeq: 28138 REFER
    Refer-To: sip:phil at 10.4.0.213
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Contact: sip:phil at 10.4.6.31
    Via: SIP/2.0/UDP HANB65:5060:branch=z9hG4bk457125;rport
    Content-Length: 0



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From anthony.minessale at gmail.com  Thu Jul  2 03:44:46 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 1 Jul 2015 18:44:46 -0500
Subject: [Freeswitch-users] Presence Issues with Secure Websockets
In-Reply-To: <CAFUDvVj2qnsxqsf1tp-V5kZbcS56dJAAo=TB-cLbtGm1ti7Byg@mail.gmail.com>
References: <CAFUDvVj2qnsxqsf1tp-V5kZbcS56dJAAo=TB-cLbtGm1ti7Byg@mail.gmail.com>
Message-ID: <CAKbxfG--0-JC-QOduYakh1ay2B0_-GiWtfGE+VnnSy6U9t0cRA@mail.gmail.com>

Its against our etiquette guidelines to update the list and Jira about the
same thing.  It adds 2 things to tend to about the same problem.
Once you have filed a JIRA, there is no need to refer to it here.


On Wed, Jul 1, 2015 at 10:58 AM, Lyle Pratt <lylepratt at gmail.com> wrote:

> We opened a JIRA issue. Has anyone looked into this?
> Presence events over WSS are basically broken.
> https://freeswitch.org/jira/browse/FS-7680
>
> Thanks,
> Lyle Pratt
> BetterVoice.com
>
>
> Michael Jerris <mike at ...> writes:
> >
> > If we are not sending the NOTIFY, please open a jira.  Will need to
> include full sip
> debug with sofia debug and sip trace turned up.
> >
> > On Jun 10, 2015, at 1:11 PM, Thomas Quintana <thomas-
> 78QPtSbfA2Jofqt5WZNmog at public.gmane.org> wrote:
> > Hey Guys,
> >
> > We're subscribing to presence events via SIP.js using a secure websocket
> (WSS)
> and we can see the subscriptions being received by FreeSWITCH with a 202
> Accepted, but never receive any events via the notify event.
> Alternatively, when we
> move to a plain websocket (WS) everything seems to work as expected.
> >
> >
> >
> > After digging around we found that a few people were experiencing
> similar issues
> with FreeSWITCH and additionally with Kamailio http://lists.sip-
> router.org/pipermail/sr-users/2014-July/084157.html. The issue was
> resolved with
> Kamailio by patching the contact header. In an attempt to achieve the same
> results
> we have experimented with the settings below but no success.
> >
> >
> >
> >
> > <param name="NDLB-force-rport" value="safe"/>
> >
> >
> > <param name="NDLB-received-in-nat-reg-contact" value="true"/>
> >
> >
> > We also have an open ticket with the folks over at
> SIP.js https://github.com/onsip/SIP.js/issues/184. Any help would be
> greatly
> appreciated.
> >
> >
> >
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at ...
> > 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 ...
> > 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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From andrew.keil at visytel.com  Thu Jul  2 05:15:21 2015
From: andrew.keil at visytel.com (Andrew Keil)
Date: Thu, 2 Jul 2015 01:15:21 +0000
Subject: [Freeswitch-users] The FreeSWITCH 1.4.20 release is here!
In-Reply-To: <55941edc6f234_d2c17b732087359@resque-worker-high.1.mail>
References: <55941edc6f234_d2c17b732087359@resque-worker-high.1.mail>
Message-ID: <SIXPR04MB12602B4AC93CC70B02D5016F3970@SIXPR04MB126.apcprd04.prod.outlook.com>

Ken,

Thanks for letting us know that FreeSWITCH 1.4.20 is here!  Can you update the main freeswitch.org website Current release version to 1.4.20, also the link on the page: https://freeswitch.org/the-freeswitch-1-4-20-release-is-here/ for Tarballs: http://files.freeswitch.org/releases/freeswitch/freeswitch-1.4.20.tar.bz2<http://files.freeswitch.org/releases/freeswitch/freeswitch-1.4.19.tar.bz2> actually goes to the 1.4.19 tarball.

Both minor and no rush.

Thanks.

Andrew

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Thursday, 2 July 2015 3:10 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] The FreeSWITCH 1.4.20 release is here!

New Post on freeswitch.org from Kathleen
check it out at http://ift.tt/1dy0Hes
The FreeSWITCH 1.4.20 release is here!

The FreeSWITCH 1.4.20 release is here!

This is a routine maintenance release and the resources are located here:

  *   Tarballs: http://ift.tt/1NwaFd8<http://ift.tt/1Fj7gsO>
  *   Packaging: http://ift.tt/1U8FMQN

Security issues:

  *   FS-7708<http://ift.tt/1ByUASw> Fixed docs on enabling cert CN/SAN validation

New features that were added:

  *   FS-7561<http://ift.tt/1MDcYLa> [mod_sofia] Add Perfect Forward Secrecy (DHE PFS)
  *   FS-7564<http://ift.tt/1MDd0CS> [mod_rayo] Added new algorithms for offering calls to clients
  *   FS-7623<http://ift.tt/1BiZeE6> [mod_amqp] Allow for custom exchange name and type for producers and fixed param name ordering bug caused by exposing these params
  *   FS-7720<http://ift.tt/1LEZrFh> Improve play_and_detect_speech to set current_application_response channel variable as follows: ?USAGE ERROR?: bad application arguments?, ?GRAMMAR ERROR?: speech recognizer failed to load grammar, ?ASR INIT ERROR?: speech recognizer failed to allocate a session, and ?ERROR?: any other errors
  *   FS-7743<http://ift.tt/1LEZtgn> [mod_skinny] Updated SKINNY on-hook action to hang up all calls on a device, except those in a short list of call states (or perform a blind transfer) and added a hook after completing the hangup operation to start ringing if there is an inbound call active on the device.

Improvements in build system, cross platform support, and packaging:

  *   FS-7610<http://ift.tt/1MDcYLg> Fixed a gcc5 compilation issue
  *   FS-7426<http://ift.tt/1MDd1GE> Only disable mod_amqp on Debian Squeeze and Wheezy
  *   FS-7297<http://ift.tt/1DLZjA7> g729 installer

The following bugs were squashed:

  *   FS-7582<http://ift.tt/1Fj7aBF> FS-7432<http://ift.tt/1BiZfrH> Fixed missing a=setup parameter from answering SDP
  *   FS-7650<http://ift.tt/1ByUCd4> [mod_verto] Fixed crash when making a call from a verto user with profile-variables in their user profile
  *   FS-7678<http://ift.tt/1RtuVwD> Fixed for fail_on_single_reject not working with | bridge
  *   FS-7612<http://ift.tt/1GoRcJ1> Fixed invalid json format for callflow key
  *   FS-7621<http://ift.tt/1GoRaAB> [mod_shout] Fixed a slow interrupt
  *   FS-7432<http://ift.tt/1BiZfrH> Fixed missing a=setup parameter from answering SDP
  *   FS-7573<http://ift.tt/1GoRcZh> Fixed 80bit tag support for zrtp
  *   FS-7636<http://ift.tt/1BiZhzE> Fixed an issue with transfer_after_bridge and park_after_bridge pre-empting transfers
  *   FS-7654<http://ift.tt/1GoRaAF> Fixed an issue with eavesdrop audio not working correctly with a mixture of mono and stereo
  *   FS-7579<http://ift.tt/1B2ellg> [mod_conference] Fixed a bug not allowing suppression of play-file-done
  *   FS-7593<http://ift.tt/1B2eo0a> [mod_skinny] Fixed a bug where skinny phones would stomp on each other in database when thundering herd occurs
  *   FS-7597<http://ift.tt/1MDd0Ta> [mod_codec2] Fixed encoded_data_len for MODE 2400, it should be 6 bytes. Also replaced 2550 bps bitrate (obsoleted operation mode) by 2400
  *   FS-7604<http://ift.tt/1B2elBw> [fs_cli] Fixed fs_cli tab completion concurrency issues on newer libedit
  *   FS-7258<http://ift.tt/1MDd0Te> FS-7571<http://ift.tt/1B2elBy> [mod_xml_cdr] Properly encode xml cdr for post to web server
  *   FS-7607<http://ift.tt/1MDd0Tg> Update URLs to reflect https protocol on freeswitch.org websites and update additional URLs to avoid 301 redirects.
  *   FS-7479<http://ift.tt/1B2eo0e> Fixed a crash caused by large RTP/PCMA packets and resampling
  *   FS-7524<http://ift.tt/1MDd1GI> [mod_callcenter] Fixing tiers, level and position should default to 1 instead of 0
  *   FS-7622<http://ift.tt/1GoRaAD> [mod_amqp] Make sure to close the connections on destroy. Currently the connection is malloc?d from the module pool, so there is nothing to destroy.
  *   FS-7689<http://ift.tt/1RtuXEB> [mod_lua] Fixed a bug with lua not loading directory configurations
  *   FS-7489<http://ift.tt/1JmfRig> [mod_unimrcp] Fixed a TTS Audio Queue Overflow
  *   FS-7467<http://ift.tt/1JmfO69> [mod_callcenter] Fixing stuck channels using uuid-standby agents
  *   FS-7429<http://ift.tt/1U8FNE8> [mod_curl] Fixed a json formatting error

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From lylepratt at gmail.com  Thu Jul  2 05:44:08 2015
From: lylepratt at gmail.com (Lyle Pratt)
Date: Wed, 1 Jul 2015 20:44:08 -0500
Subject: [Freeswitch-users] Presence Issues with Secure Websockets
Message-ID: <CAFUDvViU1YmgMCmf9PpU4Gf4av+9NxTgApPtuX6FQ4MwdDndJA@mail.gmail.com>

My sincere apologies on my breach of etiquette. It will not happen again!

Thank you for your update to my question on Jira. We will continue to work
on the problem and contribute a patch ourselves if possible.

-Lyle

---------- Forwarded message ----------
From: Anthony Minessale <anthony.minessale at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Cc:
Date: Wed, 1 Jul 2015 18:44:46 -0500
Subject: Re: [Freeswitch-users] Presence Issues with Secure Websockets
Its against our etiquette guidelines to update the list and Jira about the
same thing.  It adds 2 things to tend to about the same problem.
Once you have filed a JIRA, there is no need to refer to it here.


On Wed, Jul 1, 2015 at 10:58 AM, Lyle Pratt <lylepratt at gmail.com> wrote:

> We opened a JIRA issue. Has anyone looked into this?
> Presence events over WSS are basically broken.
> https://freeswitch.org/jira/browse/FS-7680
>
> Thanks,
> Lyle Pratt
> BetterVoice.com
>
>
> Michael Jerris <mike at ...> writes:
> >
> > If we are not sending the NOTIFY, please open a jira.  Will need to
> include full sip
> debug with sofia debug and sip trace turned up.
> >
> > On Jun 10, 2015, at 1:11 PM, Thomas Quintana <thomas-
> 78QPtSbfA2Jofqt5WZNmog at public.gmane.org> wrote:
> > Hey Guys,
> >
> > We're subscribing to presence events via SIP.js using a secure websocket
> (WSS)
> and we can see the subscriptions being received by FreeSWITCH with a 202
> Accepted, but never receive any events via the notify event.
> Alternatively, when we
> move to a plain websocket (WS) everything seems to work as expected.
> >
> >
> >
> > After digging around we found that a few people were experiencing
> similar issues
> with FreeSWITCH and additionally with Kamailio http://lists.sip-
> router.org/pipermail/sr-users/2014-July/084157.html. The issue was
> resolved with
> Kamailio by patching the contact header. In an attempt to achieve the same
> results
> we have experimented with the settings below but no success.
> >
> >
> >
> >
> > <param name="NDLB-force-rport" value="safe"/>
> >
> >
> > <param name="NDLB-received-in-nat-reg-contact" value="true"/>
> >
> >
> > We also have an open ticket with the folks over at
> SIP.js https://github.com/onsip/SIP.js/issues/184. Any help would be
> greatly
> appreciated.
> >
> >
> >
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at ...
> > 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 ...
> > 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 kathleen.king at quentustech.com  Thu Jul  2 05:45:40 2015
From: kathleen.king at quentustech.com (Kathleen King)
Date: Wed, 01 Jul 2015 18:45:40 -0700
Subject: [Freeswitch-users] The FreeSWITCH 1.4.20 release is here!
In-Reply-To: <SIXPR04MB12602B4AC93CC70B02D5016F3970@SIXPR04MB126.apcprd04.prod.outlook.com>
References: <55941edc6f234_d2c17b732087359@resque-worker-high.1.mail>
	<SIXPR04MB12602B4AC93CC70B02D5016F3970@SIXPR04MB126.apcprd04.prod.outlook.com>
Message-ID: <958F26A9-6ED4-418B-857C-768A979C4117@quentustech.com>

Thanks for letting us know about the broken link! I've updated it and it should be pointing to the right place now. 

On July 1, 2015 6:15:21 PM PDT, Andrew Keil <andrew.keil at visytel.com> wrote:
>Ken,
>
>Thanks for letting us know that FreeSWITCH 1.4.20 is here!  Can you
>update the main freeswitch.org website Current release version to
>1.4.20, also the link on the page:
>https://freeswitch.org/the-freeswitch-1-4-20-release-is-here/ for
>Tarballs:
>http://files.freeswitch.org/releases/freeswitch/freeswitch-1.4.20.tar.bz2<http://files.freeswitch.org/releases/freeswitch/freeswitch-1.4.19.tar.bz2>
>actually goes to the 1.4.19 tarball.
>
>Both minor and no rush.
>
>Thanks.
>
>Andrew
>
>From: freeswitch-users-bounces at lists.freeswitch.org
>[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
>Rice
>Sent: Thursday, 2 July 2015 3:10 AM
>To: freeswitch-users at lists.freeswitch.org
>Subject: [Freeswitch-users] The FreeSWITCH 1.4.20 release is here!
>
>New Post on freeswitch.org from Kathleen
>check it out at http://ift.tt/1dy0Hes
>The FreeSWITCH 1.4.20 release is here!
>
>The FreeSWITCH 1.4.20 release is here!
>
>This is a routine maintenance release and the resources are located
>here:
>
>  *   Tarballs: http://ift.tt/1NwaFd8<http://ift.tt/1Fj7gsO>
>  *   Packaging: http://ift.tt/1U8FMQN
>
>Security issues:
>
>*   FS-7708<http://ift.tt/1ByUASw> Fixed docs on enabling cert CN/SAN
>validation
>
>New features that were added:
>
>*   FS-7561<http://ift.tt/1MDcYLa> [mod_sofia] Add Perfect Forward
>Secrecy (DHE PFS)
>*   FS-7564<http://ift.tt/1MDd0CS> [mod_rayo] Added new algorithms for
>offering calls to clients
>*   FS-7623<http://ift.tt/1BiZeE6> [mod_amqp] Allow for custom exchange
>name and type for producers and fixed param name ordering bug caused by
>exposing these params
>*   FS-7720<http://ift.tt/1LEZrFh> Improve play_and_detect_speech to
>set current_application_response channel variable as follows: ?USAGE
>ERROR?: bad application arguments?, ?GRAMMAR ERROR?: speech recognizer
>failed to load grammar, ?ASR INIT ERROR?: speech recognizer failed to
>allocate a session, and ?ERROR?: any other errors
>*   FS-7743<http://ift.tt/1LEZtgn> [mod_skinny] Updated SKINNY on-hook
>action to hang up all calls on a device, except those in a short list
>of call states (or perform a blind transfer) and added a hook after
>completing the hangup operation to start ringing if there is an inbound
>call active on the device.
>
>Improvements in build system, cross platform support, and packaging:
>
>  *   FS-7610<http://ift.tt/1MDcYLg> Fixed a gcc5 compilation issue
>*   FS-7426<http://ift.tt/1MDd1GE> Only disable mod_amqp on Debian
>Squeeze and Wheezy
>  *   FS-7297<http://ift.tt/1DLZjA7> g729 installer
>
>The following bugs were squashed:
>
>*   FS-7582<http://ift.tt/1Fj7aBF> FS-7432<http://ift.tt/1BiZfrH> Fixed
>missing a=setup parameter from answering SDP
>*   FS-7650<http://ift.tt/1ByUCd4> [mod_verto] Fixed crash when making
>a call from a verto user with profile-variables in their user profile
>*   FS-7678<http://ift.tt/1RtuVwD> Fixed for fail_on_single_reject not
>working with | bridge
>*   FS-7612<http://ift.tt/1GoRcJ1> Fixed invalid json format for
>callflow key
>  *   FS-7621<http://ift.tt/1GoRaAB> [mod_shout] Fixed a slow interrupt
>*   FS-7432<http://ift.tt/1BiZfrH> Fixed missing a=setup parameter from
>answering SDP
>  *   FS-7573<http://ift.tt/1GoRcZh> Fixed 80bit tag support for zrtp
>*   FS-7636<http://ift.tt/1BiZhzE> Fixed an issue with
>transfer_after_bridge and park_after_bridge pre-empting transfers
>*   FS-7654<http://ift.tt/1GoRaAF> Fixed an issue with eavesdrop audio
>not working correctly with a mixture of mono and stereo
>*   FS-7579<http://ift.tt/1B2ellg> [mod_conference] Fixed a bug not
>allowing suppression of play-file-done
>*   FS-7593<http://ift.tt/1B2eo0a> [mod_skinny] Fixed a bug where
>skinny phones would stomp on each other in database when thundering
>herd occurs
>*   FS-7597<http://ift.tt/1MDd0Ta> [mod_codec2] Fixed encoded_data_len
>for MODE 2400, it should be 6 bytes. Also replaced 2550 bps bitrate
>(obsoleted operation mode) by 2400
>*   FS-7604<http://ift.tt/1B2elBw> [fs_cli] Fixed fs_cli tab completion
>concurrency issues on newer libedit
>*   FS-7258<http://ift.tt/1MDd0Te> FS-7571<http://ift.tt/1B2elBy>
>[mod_xml_cdr] Properly encode xml cdr for post to web server
>*   FS-7607<http://ift.tt/1MDd0Tg> Update URLs to reflect https
>protocol on freeswitch.org websites and update additional URLs to avoid
>301 redirects.
>*   FS-7479<http://ift.tt/1B2eo0e> Fixed a crash caused by large
>RTP/PCMA packets and resampling
>*   FS-7524<http://ift.tt/1MDd1GI> [mod_callcenter] Fixing tiers, level
>and position should default to 1 instead of 0
>*   FS-7622<http://ift.tt/1GoRaAD> [mod_amqp] Make sure to close the
>connections on destroy. Currently the connection is malloc?d from the
>module pool, so there is nothing to destroy.
>*   FS-7689<http://ift.tt/1RtuXEB> [mod_lua] Fixed a bug with lua not
>loading directory configurations
>*   FS-7489<http://ift.tt/1JmfRig> [mod_unimrcp] Fixed a TTS Audio
>Queue Overflow
>*   FS-7467<http://ift.tt/1JmfO69> [mod_callcenter] Fixing stuck
>channels using uuid-standby agents
>*   FS-7429<http://ift.tt/1U8FNE8> [mod_curl] Fixed a json formatting
>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

-- 
Sent from my Android device with K-9 Mail. Please excuse my brevity.
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From nandy1925 at gmail.com  Thu Jul  2 06:47:17 2015
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Thu, 2 Jul 2015 10:47:17 +0800
Subject: [Freeswitch-users] So you wanna setup your own CA for
	WSS/SSL/TLS?
In-Reply-To: <CAFLo2QxgouBO=6vhyc0arcMhVQXhALd62XpF6OPPTJQT56G09g@mail.gmail.com>
References: <CAEJMVkDH0tfmToSr=QooRf8GKum+Ru5WUAq6Lu1fDEXdpt06dg@mail.gmail.com>
	<53D2A7A7.9040400@quentustech.com>
	<CAEJMVkCdzFZrzfxuMXoGa=P1LVYcAOJJgKqLccZGBTdi2z0=ug@mail.gmail.com>
	<CAHKqMb72n4oOpScq7-JgYV0OAO+b=GuD1GoeYAhKYkAc8+a0sA@mail.gmail.com>
	<CAFLo2QxgouBO=6vhyc0arcMhVQXhALd62XpF6OPPTJQT56G09g@mail.gmail.com>
Message-ID: <CAHKqMb7TxcasnEpG_JJyK-vhR1CViW_Ok+Bk3aH-B8M8UvLk-w@mail.gmail.com>

Hi Luis,

I'm aware of that.  I've digged into SSL/TLS certificates (just self-signed
for my intranet setup) and WebRTC - one by one.  Now, all ports are
listening for mod_verto 8081, 8082 and 7443. However, FS log shows this
error messages like:

2014-08-05 16:44:11.831823 [INFO] mod_verto.c:3209 192.168.10.80:41210 Client
Connect.
2014-08-05 16:44:11.831823 [INFO] mod_verto.c:1379 192.168.10.80:41210 Starting
client thread.
2014-08-05 16:44:11.831823 [DEBUG] mod_verto.c:1292 192.168.10.80:41210 WS
SETUP FAILED
2014-08-05 16:44:11.831823 [INFO] mod_verto.c:1405 192.168.10.80:41210 Ending
client thread.
2014-08-05 16:44:11.831823 [INFO] mod_verto.c:1412 192.168.10.80:41210 Thread
ended

Upon testing with openssl s_client,  port 443 returned Verify code: 19
(self signed certificate in certificate chain). But in port 7443, it's
code: 21 (unable to verify the first certificate).  I encountered this same
error code in port 443 before. Solution: added self-signed CA certificate
in my web server configuration.

I think I can zero in the solution - how to add the CA certificate to
certs/wss.pem?

Or ... the secure-chain parameter in verto.conf.xml, should point to the CA
certificate file?
      <param name="secure-combined" value=
"/usr/local/freeswitch/certs/wss.pem"/>
      <param name="secure-chain" value="/usr/local/freeswitch/certs/wss.pem"
/>

Any input?

Tks,
/Nandy
On Wed, Jul 1, 2015 at 10:13 PM, Luis Daniel Lucio Quiroz <
luis.daniel.lucio at gmail.com> wrote:

> More than a script, it would be better if you post minimum cert
> requirements to let the FS work
>
> Luis Daniel Lucio Quiroz
> CISSP, CISM, CISA
> Linux, VoIP and much more fun
> www.okay.com.mx
>
> Need LCR? Check out LCR for FusionPBX with FreeSWITCH
> Need Billing? Check out Billing for FusionPBX with FreeSWITCH
>
> 2015-06-30 23:52 GMT-04:00 Nandy Dagondon <nandy1925 at gmail.com>:
>
>> Hi Brian,
>>
>> I used your script to generate the certificates to test mod_verto in an
>> intranet setup. Questions on your script:
>>
>> 1) Is 4096 bits required? Or 2048 bits will work, too?
>> 2) Examining certs/wss.pem, there should be a  <chain> certificate at the
>> end. But the script, inputs only 2 - *.crt and *.key. What should be the
>> 3rd?
>>
>> Tks,
>> /Nandy
>>
>>
>> On Sat, Jul 26, 2014 at 2:59 AM, Brian West <brian at freeswitch.org> wrote:
>>
>>> I've corrected the how-to and put it in tree:
>>>
>>>
>>> https://stash.freeswitch.org/projects/FS/repos/freeswitch/browse/docs/how_to_make_your_own_ca_correctly.txt?raw
>>>
>>> Importing the ca.crt into your system keychain for it to be trusted is
>>> left to the end user to figure out.  If you can't do that step then you'll
>>> kinda be SOL, I know on my Mac I just open ca.crt and it does the import
>>> for me... Windows I suspect is similar as for Linux NO CLUE.
>>>
>>>
>>> On Fri, Jul 25, 2014 at 1:53 PM, William King <
>>> william.king at quentustech.com> wrote:
>>>
>>>> One correction inline, and did you have any luck getting chrome to work
>>>> with the custom CA?
>>>>
>>>> William King
>>>> Senior Engineer
>>>> Quentus Technologies, INC
>>>> 1037 NE 65th St Suite 273
>>>> Seattle, WA 98115
>>>> Main:   (877) 211-9337
>>>> Office: (206) 388-4772
>>>> Cell:   (253) 686-5518
>>>> william.king at quentustech.com
>>>>
>>>> On 07/25/2014 08:12 AM, Brian West wrote:
>>>> > Someone should probably turn this into a nice how-to:
>>>> >
>>>> > Here is how I did it.
>>>> >
>>>> > wget http://www.openssl.org/contrib/ssl.ca-0.1.tar.gz
>>>> > tar zxfv ssl.ca-0.1.tar.gz
>>>> > cd ssl.ca-0.1/
>>>> > perl -i -pe 's/md5/sha1/g' *.sh
>>>> > perl -i -pe 's/2048/2048/g' *.sh
>>>> This is a noop. I assume it was suppose to be /2048/4096/ or /1024/2048/
>>>> > ./new-root-ca.sh
>>>> > ./new-server-cert.sh self.bkw.org <http://self.bkw.org>
>>>> > ./sign-server-cert.sh self.bkw.org <http://self.bkw.org>
>>>> > cat self.bkw.org.crt self.bkw.org.key >
>>>> /usr/local/freeswitch/certs/wss.pem
>>>> >
>>>> > Setup Apache:
>>>> >
>>>> > default-ssl:
>>>> >
>>>> > SSLCertificateFile    /usr/local/freeswitch/certs/wss.pem
>>>> > SSLCertificateKeyFile /usr/local/freeswitch/certs/wss.pem
>>>> > SSLCertificateChainFile /usr/local/freeswitch/certs/wss.pem
>>>> >
>>>> > Setup Sofia TLS:
>>>> >
>>>> > cat self.bkw.org.crt self.bkw.org.key >
>>>> > /usr/local/freeswitch/certs/agent.pem
>>>> > cat ca.crt > /usr/local/freeswitch/certs/cafile.pem
>>>> >
>>>> > vars.xml:
>>>> >
>>>> > <X-PRE-PROCESScmd="set"data="internal_ssl_enable=true"/>
>>>> > <X-PRE-PROCESScmd="set"data="external_ssl_enable=true"/>
>>>> >
>>>> > Restart FreeSWITCH.
>>>> >
>>>> > Now make sure your system has ca.crt imported so it will trust your
>>>> new
>>>> > found hotness.
>>>> >
>>>> > TEST:
>>>> >
>>>> > openssl s_client -connect self.bkw.org:443 <http://self.bkw.org:443>
>>>> > openssl s_client -connect self.bkw.org:8082 <http://self.bkw.org:8082
>>>> >
>>>> >
>>>> >
>>>> > Depending on what you've setup you'll see:
>>>> >
>>>> > subject=/C=US/ST=Oklahoma/L=McAlester/O=Tonka Truck/OU=Secure Web
>>>> > Server/CN=self.bkw.org/emailAddress=brian at bkw.org
>>>> > <http://self.bkw.org/emailAddress=brian at bkw.org>
>>>> >
>>>> > issuer=/C=US/ST=Oklahoma/L=McAlester/O=Whizzzzzzy Bang
>>>> > Bang/OU=Certification Services Division/CN=WBB Root
>>>> > CA/emailAddress=brian at bkw.org <mailto:brian at bkw.org>
>>>> >
>>>> > Or there abouts.
>>>> >
>>>> > --
>>>> >
>>>> > */Brian West/*
>>>> > brian at freeswitch.org <mailto:brian at freeswitch.org>
>>>> >
>>>> >
>>>> > */Twitter: @FreeSWITCH , @briankwest/*
>>>> > http://www.freeswitchbook.com
>>>> > http://www.freeswitchcookbook.com
>>>> >
>>>> > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>> > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>> >
>>>> >
>>>> >
>>>> >
>>>> _________________________________________________________________________
>>>> > Professional FreeSWITCH Consulting Services:
>>>> > consulting at freeswitch.org
>>>> > http://www.freeswitchsolutions.com
>>>> >
>>>> > FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>> > http://www.cudatel.com
>>>> >
>>>> > Official FreeSWITCH Sites
>>>> > http://www.freeswitch.org
>>>> > http://wiki.freeswitch.org
>>>> > http://www.cluecon.com
>>>> >
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>> http://www.cudatel.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>> http://www.cudatel.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> 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 nandy1925 at gmail.com  Thu Jul  2 06:54:39 2015
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Thu, 2 Jul 2015 10:54:39 +0800
Subject: [Freeswitch-users] So you wanna setup your own CA for
	WSS/SSL/TLS?
In-Reply-To: <CAHKqMb7TxcasnEpG_JJyK-vhR1CViW_Ok+Bk3aH-B8M8UvLk-w@mail.gmail.com>
References: <CAEJMVkDH0tfmToSr=QooRf8GKum+Ru5WUAq6Lu1fDEXdpt06dg@mail.gmail.com>
	<53D2A7A7.9040400@quentustech.com>
	<CAEJMVkCdzFZrzfxuMXoGa=P1LVYcAOJJgKqLccZGBTdi2z0=ug@mail.gmail.com>
	<CAHKqMb72n4oOpScq7-JgYV0OAO+b=GuD1GoeYAhKYkAc8+a0sA@mail.gmail.com>
	<CAFLo2QxgouBO=6vhyc0arcMhVQXhALd62XpF6OPPTJQT56G09g@mail.gmail.com>
	<CAHKqMb7TxcasnEpG_JJyK-vhR1CViW_Ok+Bk3aH-B8M8UvLk-w@mail.gmail.com>
Message-ID: <CAHKqMb6BNnjToN7Gq=SVZWa1VF7H-FmUaALnoMwgU8yPg7L3XA@mail.gmail.com>

Tried to point "secure-chain" parameter to (self-signed) CAcert.pem. The
same error code returned 21 (unable to verify ... ).


On Thu, Jul 2, 2015 at 10:47 AM, Nandy Dagondon <nandy1925 at gmail.com> wrote:

> Hi Luis,
>
> I'm aware of that.  I've digged into SSL/TLS certificates (just
> self-signed for my intranet setup) and WebRTC - one by one.  Now, all ports
> are listening for mod_verto 8081, 8082 and 7443. However, FS log shows this
> error messages like:
>
> 2014-08-05 16:44:11.831823 [INFO] mod_verto.c:3209 192.168.10.80:41210 Client
> Connect.
> 2014-08-05 16:44:11.831823 [INFO] mod_verto.c:1379 192.168.10.80:41210 Starting
> client thread.
> 2014-08-05 16:44:11.831823 [DEBUG] mod_verto.c:1292 192.168.10.80:41210 WS
> SETUP FAILED
> 2014-08-05 16:44:11.831823 [INFO] mod_verto.c:1405 192.168.10.80:41210 Ending
> client thread.
> 2014-08-05 16:44:11.831823 [INFO] mod_verto.c:1412 192.168.10.80:41210 Thread
> ended
>
> Upon testing with openssl s_client,  port 443 returned Verify code: 19
> (self signed certificate in certificate chain). But in port 7443, it's
> code: 21 (unable to verify the first certificate).  I encountered this same
> error code in port 443 before. Solution: added self-signed CA certificate
> in my web server configuration.
>
> I think I can zero in the solution - how to add the CA certificate to
> certs/wss.pem?
>
> Or ... the secure-chain parameter in verto.conf.xml, should point to the
> CA certificate file?
>       <param name="secure-combined" value=
> "/usr/local/freeswitch/certs/wss.pem"/>
>       <param name="secure-chain" value=
> "/usr/local/freeswitch/certs/wss.pem"/>
>
> Any input?
>
> Tks,
> /Nandy
>
> On Wed, Jul 1, 2015 at 10:13 PM, Luis Daniel Lucio Quiroz <
> luis.daniel.lucio at gmail.com> wrote:
>
>> More than a script, it would be better if you post minimum cert
>> requirements to let the FS work
>>
>> Luis Daniel Lucio Quiroz
>> CISSP, CISM, CISA
>> Linux, VoIP and much more fun
>> www.okay.com.mx
>>
>> Need LCR? Check out LCR for FusionPBX with FreeSWITCH
>> Need Billing? Check out Billing for FusionPBX with FreeSWITCH
>>
>> 2015-06-30 23:52 GMT-04:00 Nandy Dagondon <nandy1925 at gmail.com>:
>>
>>> Hi Brian,
>>>
>>> I used your script to generate the certificates to test mod_verto in an
>>> intranet setup. Questions on your script:
>>>
>>> 1) Is 4096 bits required? Or 2048 bits will work, too?
>>> 2) Examining certs/wss.pem, there should be a  <chain> certificate at
>>> the end. But the script, inputs only 2 - *.crt and *.key. What should be
>>> the 3rd?
>>>
>>> Tks,
>>> /Nandy
>>>
>>>
>>> On Sat, Jul 26, 2014 at 2:59 AM, Brian West <brian at freeswitch.org>
>>> wrote:
>>>
>>>> I've corrected the how-to and put it in tree:
>>>>
>>>>
>>>> https://stash.freeswitch.org/projects/FS/repos/freeswitch/browse/docs/how_to_make_your_own_ca_correctly.txt?raw
>>>>
>>>> Importing the ca.crt into your system keychain for it to be trusted is
>>>> left to the end user to figure out.  If you can't do that step then you'll
>>>> kinda be SOL, I know on my Mac I just open ca.crt and it does the import
>>>> for me... Windows I suspect is similar as for Linux NO CLUE.
>>>>
>>>>
>>>> On Fri, Jul 25, 2014 at 1:53 PM, William King <
>>>> william.king at quentustech.com> wrote:
>>>>
>>>>> One correction inline, and did you have any luck getting chrome to work
>>>>> with the custom CA?
>>>>>
>>>>> William King
>>>>> Senior Engineer
>>>>> Quentus Technologies, INC
>>>>> 1037 NE 65th St Suite 273
>>>>> Seattle, WA 98115
>>>>> Main:   (877) 211-9337
>>>>> Office: (206) 388-4772
>>>>> Cell:   (253) 686-5518
>>>>> william.king at quentustech.com
>>>>>
>>>>> On 07/25/2014 08:12 AM, Brian West wrote:
>>>>> > Someone should probably turn this into a nice how-to:
>>>>> >
>>>>> > Here is how I did it.
>>>>> >
>>>>> > wget http://www.openssl.org/contrib/ssl.ca-0.1.tar.gz
>>>>> > tar zxfv ssl.ca-0.1.tar.gz
>>>>> > cd ssl.ca-0.1/
>>>>> > perl -i -pe 's/md5/sha1/g' *.sh
>>>>> > perl -i -pe 's/2048/2048/g' *.sh
>>>>> This is a noop. I assume it was suppose to be /2048/4096/ or
>>>>> /1024/2048/
>>>>> > ./new-root-ca.sh
>>>>> > ./new-server-cert.sh self.bkw.org <http://self.bkw.org>
>>>>> > ./sign-server-cert.sh self.bkw.org <http://self.bkw.org>
>>>>> > cat self.bkw.org.crt self.bkw.org.key >
>>>>> /usr/local/freeswitch/certs/wss.pem
>>>>> >
>>>>> > Setup Apache:
>>>>> >
>>>>> > default-ssl:
>>>>> >
>>>>> > SSLCertificateFile    /usr/local/freeswitch/certs/wss.pem
>>>>> > SSLCertificateKeyFile /usr/local/freeswitch/certs/wss.pem
>>>>> > SSLCertificateChainFile /usr/local/freeswitch/certs/wss.pem
>>>>> >
>>>>> > Setup Sofia TLS:
>>>>> >
>>>>> > cat self.bkw.org.crt self.bkw.org.key >
>>>>> > /usr/local/freeswitch/certs/agent.pem
>>>>> > cat ca.crt > /usr/local/freeswitch/certs/cafile.pem
>>>>> >
>>>>> > vars.xml:
>>>>> >
>>>>> > <X-PRE-PROCESScmd="set"data="internal_ssl_enable=true"/>
>>>>> > <X-PRE-PROCESScmd="set"data="external_ssl_enable=true"/>
>>>>> >
>>>>> > Restart FreeSWITCH.
>>>>> >
>>>>> > Now make sure your system has ca.crt imported so it will trust your
>>>>> new
>>>>> > found hotness.
>>>>> >
>>>>> > TEST:
>>>>> >
>>>>> > openssl s_client -connect self.bkw.org:443 <http://self.bkw.org:443>
>>>>> > openssl s_client -connect self.bkw.org:8082 <
>>>>> http://self.bkw.org:8082>
>>>>> >
>>>>> >
>>>>> > Depending on what you've setup you'll see:
>>>>> >
>>>>> > subject=/C=US/ST=Oklahoma/L=McAlester/O=Tonka Truck/OU=Secure Web
>>>>> > Server/CN=self.bkw.org/emailAddress=brian at bkw.org
>>>>> > <http://self.bkw.org/emailAddress=brian at bkw.org>
>>>>> >
>>>>> > issuer=/C=US/ST=Oklahoma/L=McAlester/O=Whizzzzzzy Bang
>>>>> > Bang/OU=Certification Services Division/CN=WBB Root
>>>>> > CA/emailAddress=brian at bkw.org <mailto:brian at bkw.org>
>>>>> >
>>>>> > Or there abouts.
>>>>> >
>>>>> > --
>>>>> >
>>>>> > */Brian West/*
>>>>> > brian at freeswitch.org <mailto:brian at freeswitch.org>
>>>>> >
>>>>> >
>>>>> > */Twitter: @FreeSWITCH , @briankwest/*
>>>>> > http://www.freeswitchbook.com
>>>>> > http://www.freeswitchcookbook.com
>>>>> >
>>>>> > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>>> > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>> >
>>>>> >
>>>>> >
>>>>> >
>>>>> _________________________________________________________________________
>>>>> > Professional FreeSWITCH Consulting Services:
>>>>> > consulting at freeswitch.org
>>>>> > http://www.freeswitchsolutions.com
>>>>> >
>>>>> > FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>>> > http://www.cudatel.com
>>>>> >
>>>>> > Official FreeSWITCH Sites
>>>>> > http://www.freeswitch.org
>>>>> > http://wiki.freeswitch.org
>>>>> > http://www.cluecon.com
>>>>> >
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>> >
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>>> http://www.cudatel.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> *Brian West*
>>>> brian at freeswitch.org
>>>>
>>>>
>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>> http://www.freeswitchbook.com
>>>> http://www.freeswitchcookbook.com
>>>>
>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>> http://www.cudatel.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> 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 lexxua at gmail.com  Thu Jul  2 10:17:47 2015
From: lexxua at gmail.com (Volodymyr Fedorov)
Date: Thu, 02 Jul 2015 06:17:47 +0000
Subject: [Freeswitch-users] Pass RPID from B-leg to A-leg
In-Reply-To: <CAO_GGAoqV+qa3LR9EZ6Sdd2VDHpE7ugEDOX9o6wBmM3T6CSYoQ@mail.gmail.com>
References: <CAO_GGAoqV+qa3LR9EZ6Sdd2VDHpE7ugEDOX9o6wBmM3T6CSYoQ@mail.gmail.com>
Message-ID: <CAO_GGArkC178oKB8eh0NjknKqwkyPKJBGbBuVC3GVgW3zuAHHw@mail.gmail.com>

Maybe some explanation needed.
Calls goes from A (freeswitch 1.4.20) -> B (freeswitch 1.4.19) -> C
(Asterisk 13.4) .
Asterisk (C) sends RPID : Remote-Party-ID: "fedorov volodymyr" <
sip:44419 at 77.120.8.7>;party=called;privacy=off;screen=no
But Freeswitch (B) Rewrites it to : Remote-Party-ID: "9944419" <
sip:9944419 at 77.120.8.7>;party=calling;privacy=off;screen=no
On a Freeswitch B I also add these to dial-plan:
<action application="set" data="sip_cid_in_1xx=true"/>
<action application="export" data="sip_cid_in_1xx=true"/>

It seems like bug described in https://freeswitch.org/jira/browse/FS-4357 ,
but i`m not quite sure.

Best regards,
Volodymyr

On Thu, Jun 25, 2015 at 10:06 PM Volodymyr Fedorov <lexxua at gmail.com> wrote:

> Hello,Community.
> Are there any ability to keep RPID which I get in 180 Ringing on A-leg ?
> Maybe some kind of variable.
>
> B-leg RPID:
> Remote-Party-ID: "fedorov volodymyr" sip:44419 at 77.120.80.7
> >;party=called;privacy=off;screen=no
>
> What freeswitch pass back to A-leg:
> Remote-Party-ID: "9944419" <sip:9944419 at 77.120.80.7
> >;party=calling;privacy=off;screen=no
>
> Thanks!
> --
> Cheers !
>
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From clive18 at webmail.co.za  Thu Jul  2 13:21:04 2015
From: clive18 at webmail.co.za (clive engelberg)
Date: Thu, 02 Jul 2015 11:21:04 +0200
Subject: [Freeswitch-users] Disk error / database error help
Message-ID: <c357215130c3dee62c7803ab7ef5c42b@www.webmail.co.za>

Hi Guys.

I have had this happen a few times already, where Freeswitch seems to have a
database error. This starts off slowly, and within a short time, it completely
fills the disk drive with log error files.

the log files look like this:
2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR [disk
I/O error]
BEGIN EXCLUSIVE
2015-07-01 15:19:35.470881 [CRIT] switch_core_sqldb.c:1565 ERROR [disk I/O
error]
2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR
[cannot commit - no transaction is active] 

Any ideas what is causing this.
I suspect maybe a hardware disk error.. not sure.

Thanks in advance
Clive



____________________________________________________________
South Africas premier free email service - www.webmail.co.za




From mcazzador at gmail.com  Thu Jul  2 13:36:03 2015
From: mcazzador at gmail.com (Matteo Cazzador)
Date: Thu, 2 Jul 2015 11:36:03 +0200
Subject: [Freeswitch-users] internal number blacklist
In-Reply-To: <CAEFBLw2mNJbWMPwKbU0YV3aJsop-x-5veYupWvEQZA5F=eCD_w@mail.gmail.com>
References: <CAEFBLw24khyi3QtmR-MCDD+195RE0-HYZuCC57UfpD=UAdiVTQ@mail.gmail.com>
	<CAHtxdDewfEwGA6PZntY150Qp7kRL9E+FydRxr3P+s09joHo+JA@mail.gmail.com>
	<CAEFBLw3AntWJGeMVhRyE+HuTkTn-gxTKTgF6_8X3zsNFjq4x1g@mail.gmail.com>
	<CAHtxdDfy_YgrfWHDjY-6Pr4eXgAsgC+bNfCm=a2hBdAuWHTe2A@mail.gmail.com>
	<CAEFBLw2mNJbWMPwKbU0YV3aJsop-x-5veYupWvEQZA5F=eCD_w@mail.gmail.com>
Message-ID: <CAEFBLw07AzKAZw590GVcuvs_HcS6eOvUGS4a_2AxBLO-88qgog@mail.gmail.com>

I read your suggest link and for me it's clear,
but i don't know how disable  outbount call from a specific internal number.
What i need to do?
Thanks

2015-07-01 18:05 GMT+02:00 Matteo Cazzador <mcazzador at gmail.com>:

> thanks a lot!!!
>
> 2015-07-01 18:04 GMT+02:00 Sergey Safarov <s.safarov at gmail.com>:
>
>> Try mod_curl
>>
>> https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/3965033
>> and analyze curl_response_data or curl_response_code
>>
>> On Wed, Jul 1, 2015, 18:46 Matteo Cazzador <mcazzador at gmail.com> wrote:
>>
>>> Hi, i explain better, i've a freeswitch fax machine,
>>> I collect cdrs data with cdrs-sqlite, then i 've an external script
>>> (php) part that read table cdrs and calculate the billing.
>>> If a calling number it's over a threshold i want to disable outbound
>>> calling from specific number.
>>>
>>>
>>> 2015-07-01 17:38 GMT+02:00 Sergey Safarov <s.safarov at gmail.com>:
>>>
>>>> What is type of external billing system? It is RADIUS server?
>>>>
>>>> On Wed, Jul 1, 2015, 15:36 Matteo Cazzador <mcazzador at gmail.com> wrote:
>>>>
>>>>> HI, i'm a novice, i'm trying to make a dynamic blacklist
>>>>> using mod_blacklist.
>>>>> My scope is block call from internal number to external .
>>>>> I need to create dynamic blacklist of internal number, not a blacklist
>>>>> of external number.
>>>>>
>>>>> Example i need to block dynamically internal number call to external
>>>>> payment number (over a billing target)
>>>>>
>>>>> Someone can help me please.
>>>>>
>>>>> Thanks a lot.
>>>>> --
>>>>> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
>>>>> ******************************************
>>>>> Ing. Matteo Cazzador
>>>>> Email: mcazzador at gmail.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
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>>>>
>>>
>>>
>>>
>>> --
>>> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
>>> ******************************************
>>> Ing. Matteo Cazzador
>>> Email: mcazzador at gmail.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
>>
>>
>> _________________________________________________________________________
>> 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
>>
>
>
>
> --
> Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
> ******************************************
> Ing. Matteo Cazzador
> Email: mcazzador at gmail.com
> ******************************************
>



-- 
Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
******************************************
Ing. Matteo Cazzador
Email: mcazzador at gmail.com
******************************************
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From s.safarov at gmail.com  Thu Jul  2 15:38:06 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 2 Jul 2015 14:38:06 +0300
Subject: [Freeswitch-users] internal number blacklist
In-Reply-To: <CAEFBLw07AzKAZw590GVcuvs_HcS6eOvUGS4a_2AxBLO-88qgog@mail.gmail.com>
References: <CAEFBLw24khyi3QtmR-MCDD+195RE0-HYZuCC57UfpD=UAdiVTQ@mail.gmail.com>
	<CAHtxdDewfEwGA6PZntY150Qp7kRL9E+FydRxr3P+s09joHo+JA@mail.gmail.com>
	<CAEFBLw3AntWJGeMVhRyE+HuTkTn-gxTKTgF6_8X3zsNFjq4x1g@mail.gmail.com>
	<CAHtxdDfy_YgrfWHDjY-6Pr4eXgAsgC+bNfCm=a2hBdAuWHTe2A@mail.gmail.com>
	<CAEFBLw2mNJbWMPwKbU0YV3aJsop-x-5veYupWvEQZA5F=eCD_w@mail.gmail.com>
	<CAEFBLw07AzKAZw590GVcuvs_HcS6eOvUGS4a_2AxBLO-88qgog@mail.gmail.com>
Message-ID: <CAHtxdDfgaC-RJLmAN9RZy6TbyPPoqpNdcHNq=H1pG1PJf3gSbw@mail.gmail.com>

Try write condition to hangup call when received code "403 Forbidden" error
code.

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


On Thu, Jul 2, 2015 at 12:36 PM, Matteo Cazzador <mcazzador at gmail.com>
wrote:

> I read your suggest link and for me it's clear,
> but i don't know how disable  outbount call from a specific internal
> number.
> What i need to do?
> Thanks
>
>
>
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From mcazzador at gmail.com  Thu Jul  2 15:40:58 2015
From: mcazzador at gmail.com (Matteo Cazzador)
Date: Thu, 2 Jul 2015 13:40:58 +0200
Subject: [Freeswitch-users] internal number blacklist
In-Reply-To: <CAHtxdDfgaC-RJLmAN9RZy6TbyPPoqpNdcHNq=H1pG1PJf3gSbw@mail.gmail.com>
References: <CAEFBLw24khyi3QtmR-MCDD+195RE0-HYZuCC57UfpD=UAdiVTQ@mail.gmail.com>
	<CAHtxdDewfEwGA6PZntY150Qp7kRL9E+FydRxr3P+s09joHo+JA@mail.gmail.com>
	<CAEFBLw3AntWJGeMVhRyE+HuTkTn-gxTKTgF6_8X3zsNFjq4x1g@mail.gmail.com>
	<CAHtxdDfy_YgrfWHDjY-6Pr4eXgAsgC+bNfCm=a2hBdAuWHTe2A@mail.gmail.com>
	<CAEFBLw2mNJbWMPwKbU0YV3aJsop-x-5veYupWvEQZA5F=eCD_w@mail.gmail.com>
	<CAEFBLw07AzKAZw590GVcuvs_HcS6eOvUGS4a_2AxBLO-88qgog@mail.gmail.com>
	<CAHtxdDfgaC-RJLmAN9RZy6TbyPPoqpNdcHNq=H1pG1PJf3gSbw@mail.gmail.com>
Message-ID: <CAEFBLw3pPeJFZiZaZQRSFbZqGUZd2XkGhv+cXOVLyG8Q=O83SA@mail.gmail.com>

Ok now it's clear thanks a lot.

2015-07-02 13:38 GMT+02:00 Sergey Safarov <s.safarov at gmail.com>:

> Try write condition to hangup call when received code "403 Forbidden"
> error code.
>
> https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan
>
>
> On Thu, Jul 2, 2015 at 12:36 PM, Matteo Cazzador <mcazzador at gmail.com>
> wrote:
>
>> I read your suggest link and for me it's clear,
>> but i don't know how disable  outbount call from a specific internal
>> number.
>> What i need to do?
>> Thanks
>>
>>
>>
> _________________________________________________________________________
> 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
>



-- 
Rispetta l'ambiente: se non ti ? necessario, non stampare questa mail.
******************************************
Ing. Matteo Cazzador
Email: mcazzador at gmail.com
******************************************
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From s.safarov at gmail.com  Thu Jul  2 16:07:49 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 2 Jul 2015 15:07:49 +0300
Subject: [Freeswitch-users] Why ringback is not generated?
Message-ID: <CAHtxdDfhMUSoHjYYW9yLE613+rM=x8f-iWmcKJ9NzbuGE-3tAA@mail.gmail.com>

In dialplan I set ringback variable

      <extension name="setRingback" continue="true">
         <condition>
            <action application="set" data="ringback=${us-ring}"/>
         </condition>
      </extension>
And diaplan terminate call on extension

<extension name="agent_login">
   <condition field="${destination_number}" expression="^agent_login$">
      <action application="set"
              data="ignore=${callcenter_config(agent set status
${caller_id_number}@${sip_to_host} 'Available')}"
              inline="true"/>
      <action application="ring_ready"/>
      <action application="pre_answer"/>
      <action application="sleep" data="2500"/>
      <action application="answer"/>
      <action application="sleep" data="700"/>
   </condition>
</extension>

But ringback is not generaded. Why is it posible?

Sergey
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From s.safarov at gmail.com  Thu Jul  2 16:15:15 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 2 Jul 2015 15:15:15 +0300
Subject: [Freeswitch-users] Why ringback is not generated?
In-Reply-To: <CAHtxdDfhMUSoHjYYW9yLE613+rM=x8f-iWmcKJ9NzbuGE-3tAA@mail.gmail.com>
References: <CAHtxdDfhMUSoHjYYW9yLE613+rM=x8f-iWmcKJ9NzbuGE-3tAA@mail.gmail.com>
Message-ID: <CAHtxdDe=4Wkc2Kw=tsq2MLESmDjrhPSgUvYjHA0kzcQiQgNnbg@mail.gmail.com>

Wireshark help locate root issue FS send 183 message but not start sending
RTP media. FS start send media after sending 200 message.

It is bug?

On Thu, Jul 2, 2015 at 3:07 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> In dialplan I set ringback variable
>
>       <extension name="setRingback" continue="true">
>          <condition>
>             <action application="set" data="ringback=${us-ring}"/>
>          </condition>
>       </extension>
> And diaplan terminate call on extension
>
> <extension name="agent_login">
>    <condition field="${destination_number}" expression="^agent_login$">
>       <action application="set"
>               data="ignore=${callcenter_config(agent set status
> ${caller_id_number}@${sip_to_host} 'Available')}"
>               inline="true"/>
>       <action application="ring_ready"/>
>       <action application="pre_answer"/>
>       <action application="sleep" data="2500"/>
>       <action application="answer"/>
>       <action application="sleep" data="700"/>
>    </condition>
> </extension>
>
> But ringback is not generaded. Why is it posible?
>
> Sergey
>
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From steveayre at gmail.com  Thu Jul  2 16:18:03 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 2 Jul 2015 13:18:03 +0100
Subject: [Freeswitch-users] Disk error / database error help
In-Reply-To: <c357215130c3dee62c7803ab7ef5c42b@www.webmail.co.za>
References: <c357215130c3dee62c7803ab7ef5c42b@www.webmail.co.za>
Message-ID: <CAFiqYukdk08m2EoNVZ75xot+Fr=EJDt8y1mFqo8_i-xuvxXDuA@mail.gmail.com>

Do you see any errors in the output of 'dmesg'? If there are errors on your
hard disk or controller you'd see messages in there.

On 2 July 2015 at 10:21, clive engelberg <clive18 at webmail.co.za> wrote:

> Hi Guys.
>
> I have had this happen a few times already, where Freeswitch seems to have
> a
> database error. This starts off slowly, and within a short time, it
> completely
> fills the disk drive with log error files.
>
> the log files look like this:
> 2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR
> [disk
> I/O error]
> BEGIN EXCLUSIVE
> 2015-07-01 15:19:35.470881 [CRIT] switch_core_sqldb.c:1565 ERROR [disk I/O
> error]
> 2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR
> [cannot commit - no transaction is active]
>
> Any ideas what is causing this.
> I suspect maybe a hardware disk error.. not sure.
>
> Thanks in advance
> Clive
>
>
>
> ____________________________________________________________
> South Africas premier free email service - www.webmail.co.za
>
>
>
> _________________________________________________________________________
> 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 italorossib at gmail.com  Thu Jul  2 18:08:10 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 2 Jul 2015 11:08:10 -0300
Subject: [Freeswitch-users] Why ringback is not generated?
In-Reply-To: <CAHtxdDe=4Wkc2Kw=tsq2MLESmDjrhPSgUvYjHA0kzcQiQgNnbg@mail.gmail.com>
References: <CAHtxdDfhMUSoHjYYW9yLE613+rM=x8f-iWmcKJ9NzbuGE-3tAA@mail.gmail.com>
	<CAHtxdDe=4Wkc2Kw=tsq2MLESmDjrhPSgUvYjHA0kzcQiQgNnbg@mail.gmail.com>
Message-ID: <CAJR7f2GzBmsH1om_iAUUnFLBev8ksCobmKLmMyhv7GmJ=CW8TA@mail.gmail.com>

FreeSWITCH will send media back to the caller once it received a 18X, if
you want to send immediately use instant_ringback=true

https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-instant_ringback

On Thu, Jul 2, 2015 at 9:15 AM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Wireshark help locate root issue FS send 183 message but not start sending
> RTP media. FS start send media after sending 200 message.
>
> It is bug?
>
> On Thu, Jul 2, 2015 at 3:07 PM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> In dialplan I set ringback variable
>>
>>       <extension name="setRingback" continue="true">
>>          <condition>
>>             <action application="set" data="ringback=${us-ring}"/>
>>          </condition>
>>       </extension>
>> And diaplan terminate call on extension
>>
>> <extension name="agent_login">
>>    <condition field="${destination_number}" expression="^agent_login$">
>>       <action application="set"
>>               data="ignore=${callcenter_config(agent set status
>> ${caller_id_number}@${sip_to_host} 'Available')}"
>>               inline="true"/>
>>       <action application="ring_ready"/>
>>       <action application="pre_answer"/>
>>       <action application="sleep" data="2500"/>
>>       <action application="answer"/>
>>       <action application="sleep" data="700"/>
>>    </condition>
>> </extension>
>>
>> But ringback is not generaded. Why is it posible?
>>
>> Sergey
>>
>
>
> _________________________________________________________________________
> 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
>



-- 
?talo Rossi
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From krice at freeswitch.org  Thu Jul  2 18:25:53 2015
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 02 Jul 2015 09:25:53 -0500
Subject: [Freeswitch-users] Why ringback is not generated?
In-Reply-To: <CAHtxdDe=4Wkc2Kw=tsq2MLESmDjrhPSgUvYjHA0kzcQiQgNnbg@mail.gmail.com>
Message-ID: <D1BAB421.A1078%krice@freeswitch.org>

Is this a 2 leg called where Endpoint A -> FreeSWITCH -> Endpoint B and
Endpoint A is calling Endpoint B, if EndPoint B sends FreeSWITCH a 183 with
SDP indicating that it is providing Early Media, then does not actually send
Early Media, the problem is a problem with Endpoint B as in this case
FreeSWITCH Depends on the endpoint to do what it says its going to do


On 7/2/15, 7:15 AM, "Sergey Safarov" <s.safarov at gmail.com> wrote:

> Wireshark help locate root issue FS send 183 message but not start sending RTP
> media. FS start send media after sending 200 message.
> 
> It is bug?
> 
> On Thu, Jul 2, 2015 at 3:07 PM, Sergey Safarov <s.safarov at gmail.com> wrote:
>> In dialplan I set ringback variable
>> 
>> ?? ? ?<extension name="setRingback" continue="true">
>> ? ? ? ? ?<condition>
>> ? ? ? ? ? ? <action application="set" data="ringback=${us-ring}"/>
>> ? ? ? ? ?</condition>
>> ? ? ? </extension>
>> And diaplan terminate call on extension
>> 
>> <extension name="agent_login">
>> ? ?<condition field="${destination_number}" expression="^agent_login$">
>> ? ? ? <action application="set"
>> ? ? ? ? ? ? ? data="ignore=${callcenter_config(agent set status
>> ${caller_id_number}@${sip_to_host} 'Available')}"
>> ? ? ? ? ? ? ? inline="true"/>
>> ? ? ? <action application="ring_ready"/>
>> ? ? ? <action application="pre_answer"/>
>> ? ? ? <action application="sleep" data="2500"/>
>> ? ? ? <action application="answer"/>
>> ? ? ? <action application="sleep" data="700"/>
>> ? ?</condition>
>> </extension>
>> 
>> But ringback is not generaded. Why is it posible?
>> 
>> Sergey
> 
> 
> 
> _________________________________________________________________________
> 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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From clive18 at webmail.co.za  Thu Jul  2 18:50:55 2015
From: clive18 at webmail.co.za (clive engelberg)
Date: Thu, 02 Jul 2015 16:50:55 +0200
Subject: [Freeswitch-users] Disk error / database error help
In-Reply-To: <CAFiqYukdk08m2EoNVZ75xot+Fr=EJDt8y1mFqo8_i-xuvxXDuA@mail.gmail.com>
References: <c357215130c3dee62c7803ab7ef5c42b@www.webmail.co.za>,
	<CAFiqYukdk08m2EoNVZ75xot+Fr=EJDt8y1mFqo8_i-xuvxXDuA@mail.gmail.com>
Message-ID: <ed6289f3b6f594234ea0e9907c73f721@www.webmail.co.za>

Hi

My dmesg seems to be full of IPtables "denied" statements.

I guess I need to reboot to get a clean dmesg.

Thanks and regards
Clive

On Thu, 2 Jul 2015 13:18:03 +0100 Steven Ayre  wrote

  Do you see any errors in the output of 'dmesg'? If there are errors on your
hard disk or controller you'd see messages in there.

 On 2 July 2015 at 10:21, clive engelberg  wrote:
 Hi Guys.

 I have had this happen a few times already, where Freeswitch seems to have a
 database error. This starts off slowly, and within a short time, it
completely
 fills the disk drive with log error files.

 the log files look like this:
 2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR
[disk
 I/O error]
 BEGIN EXCLUSIVE
 2015-07-01 15:19:35.470881 [CRIT] switch_core_sqldb.c:1565 ERROR [disk I/O
 error]
 2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR
 [cannot commit - no transaction is active]

 Any ideas what is causing this.
 I suspect maybe a hardware disk error.. not sure.

 Thanks in advance
 Clive

 ____________________________________________________________
 South Africas premier free email service - www.webmail.co.za

 _________________________________________________________________________
 Professional FreeSWITCH Consulting Services:
 consulting at freeswitch.org [2]
 http://www.freeswitchsolutions.com [3]

 Official FreeSWITCH Sites
 http://www.freeswitch.org [4]
 http://confluence.freeswitch.org [5]
 http://www.cluecon.com [6]

 FreeSWITCH-users mailing list
 FreeSWITCH-users at lists.freeswitch.org [7]
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [8]
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [9]
 http://www.freeswitch.org [10]   

Links:
------
[1] mailto:clive18 at webmail.co.za
[2] mailto:consulting at freeswitch.org
[3] http://www.freeswitchsolutions.com
[4] http://www.freeswitch.org
[5] http://confluence.freeswitch.org
[6] http://www.cluecon.com
[7] mailto:FreeSWITCH-users at lists.freeswitch.org
[8] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[9] http://lists.freeswitch.org/mailman/options/freeswitch-users
[10] http://www.freeswitch.org


____________________________________________________________
South Africas premier free email service - www.webmail.co.za
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From gmaruzz at gmail.com  Thu Jul  2 19:00:51 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 2 Jul 2015 17:00:51 +0200
Subject: [Freeswitch-users] Disk error / database error help
In-Reply-To: <ed6289f3b6f594234ea0e9907c73f721@www.webmail.co.za>
References: <c357215130c3dee62c7803ab7ef5c42b@www.webmail.co.za>
	<CAFiqYukdk08m2EoNVZ75xot+Fr=EJDt8y1mFqo8_i-xuvxXDuA@mail.gmail.com>
	<ed6289f3b6f594234ea0e9907c73f721@www.webmail.co.za>
Message-ID: <CALXCt0ocnA7QJKbJ0gwpsJHkeKHTO9L6KjD+LPxcHLRsiEL5NA@mail.gmail.com>

you probably have a failing disk...

btw, is better to mount freeswitch db in a tmpfs ram dilesystem



On Thu, Jul 2, 2015 at 4:50 PM, clive engelberg <clive18 at webmail.co.za>
wrote:

> Hi
>
> My dmesg seems to be full of IPtables "denied" statements.
>
> I guess I need to reboot to get a clean dmesg.
>
> Thanks and regards
> Clive
>
> On Thu, 2 Jul 2015 13:18:03 +0100 Steven Ayre <steveayre at gmail.com> wrote
>
>  Do you see any errors in the output of 'dmesg'? If there are errors on
> your hard disk or controller you'd see messages in there.
>
> On 2 July 2015 at 10:21, clive engelberg <clive18 at webmail.co.za> wrote:
>
>> Hi Guys.
>>
>> I have had this happen a few times already, where Freeswitch seems to
>> have a
>> database error. This starts off slowly, and within a short time, it
>> completely
>> fills the disk drive with log error files.
>>
>> the log files look like this:
>> 2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR
>> [disk
>> I/O error]
>> BEGIN EXCLUSIVE
>> 2015-07-01 15:19:35.470881 [CRIT] switch_core_sqldb.c:1565 ERROR [disk I/O
>> error]
>> 2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR
>> [cannot commit - no transaction is active]
>>
>> Any ideas what is causing this.
>> I suspect maybe a hardware disk error.. not sure.
>>
>> Thanks in advance
>> Clive
>>
>>
>>
>> ____________________________________________________________
>> South Africas premier fr ee email service - www.webmail.co.za
>>
>>
>>
>> _________________________________________________________________________
>> 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://lists.freeswitch.org/mailman/options/%20freeswitch-users>
>> http://www.freeswitch.org
>
>
> ------------------------------
> South Africa premier free email service - webmail.co.za
> <http://www.webmail.co.za/>
> _________________________________________________________________________
> 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
Cell : +39-347-2665618
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From s.safarov at gmail.com  Thu Jul  2 19:09:55 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 2 Jul 2015 18:09:55 +0300
Subject: [Freeswitch-users] Why ringback is not generated?
In-Reply-To: <CAJR7f2GzBmsH1om_iAUUnFLBev8ksCobmKLmMyhv7GmJ=CW8TA@mail.gmail.com>
References: <CAHtxdDfhMUSoHjYYW9yLE613+rM=x8f-iWmcKJ9NzbuGE-3tAA@mail.gmail.com>
	<CAHtxdDe=4Wkc2Kw=tsq2MLESmDjrhPSgUvYjHA0kzcQiQgNnbg@mail.gmail.com>
	<CAJR7f2GzBmsH1om_iAUUnFLBev8ksCobmKLmMyhv7GmJ=CW8TA@mail.gmail.com>
Message-ID: <CAHtxdDdNfMYTLUwWU7-1qsKTPnB-X3hUUBJa1nM8RYUiA-j7uA@mail.gmail.com>

thank you Italo, Ken for quick response
I set "instant_ringback=true" but it is not help.
Usage schema as described below
endpointA -> FS1 -> FS2

FS2 must generate ringback, play "you a logged in" wav file and hangup

FS1 - user endpoint registrator, managed by kazoo
FS2 - host implemening callcenter queries.

On Thu, Jul 2, 2015 at 5:08 PM, ?talo Rossi <italorossib at gmail.com> wrote:

> FreeSWITCH will send media back to the caller once it received a 18X, if
> you want to send immediately use instant_ringback=true
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-instant_ringback
>
> On Thu, Jul 2, 2015 at 9:15 AM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> Wireshark help locate root issue FS send 183 message but not start
>> sending RTP media. FS start send media after sending 200 message.
>>
>> It is bug?
>>
>> On Thu, Jul 2, 2015 at 3:07 PM, Sergey Safarov <s.safarov at gmail.com>
>> wrote:
>>
>>> In dialplan I set ringback variable
>>>
>>>       <extension name="setRingback" continue="true">
>>>          <condition>
>>>             <action application="set" data="ringback=${us-ring}"/>
>>>          </condition>
>>>       </extension>
>>> And diaplan terminate call on extension
>>>
>>> <extension name="agent_login">
>>>    <condition field="${destination_number}" expression="^agent_login$">
>>>       <action application="set"
>>>               data="ignore=${callcenter_config(agent set status
>>> ${caller_id_number}@${sip_to_host} 'Available')}"
>>>               inline="true"/>
>>>       <action application="ring_ready"/>
>>>       <action application="pre_answer"/>
>>>       <action application="sleep" data="2500"/>
>>>       <action application="answer"/>
>>>       <action application="sleep" data="700"/>
>>>    </condition>
>>> </extension>
>>>
>>> But ringback is not generaded. Why is it posible?
>>>
>>> Sergey
>>>
>>
>>
>> _________________________________________________________________________
>> 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
>>
>
>
>
> --
> ?talo Rossi
>
> _________________________________________________________________________
> 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 freeswitch.org  Thu Jul  2 19:17:53 2015
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 02 Jul 2015 10:17:53 -0500
Subject: [Freeswitch-users] Why ringback is not generated?
In-Reply-To: <CAHtxdDdNfMYTLUwWU7-1qsKTPnB-X3hUUBJa1nM8RYUiA-j7uA@mail.gmail.com>
Message-ID: <D1BAC051.A107F%krice@freeswitch.org>

On FS2 did you pre-answer the call? That will put FS into the state to start
generating early media... And you?ll probably have to sleep or something to
delay the exchange long enough to actually get ringing... Just playing a
file and firing a script to check a pin or something and firing a DB query
happens so fast that 99.99999% of the time its worthless to send ringing


On 7/2/15, 10:09 AM, "Sergey Safarov" <s.safarov at gmail.com> wrote:

> thank you Italo, Ken for quick response
> I set "instant_ringback=true" but it is not help.
> Usage schema as described below
> endpointA -> FS1 -> FS2
> 
> FS2 must generate ringback, play "you a logged in" wav file and hangup
> 
> FS1 - user endpoint registrator, managed by kazoo
> FS2 - host implemening callcenter queries.
> 
> On Thu, Jul 2, 2015 at 5:08 PM, ?talo Rossi <italorossib at gmail.com> wrote:
>> FreeSWITCH will send media back to the caller once it received a 18X, if you
>> want to send immediately use instant_ringback=true
>> 
>> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#Channe
>> lVariables-instant_ringback
>> 
>> On Thu, Jul 2, 2015 at 9:15 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
>>> Wireshark help locate root issue FS send 183 message but not start sending
>>> RTP media. FS start send media after sending 200 message.
>>> 
>>> It is bug?
>>> 
>>> On Thu, Jul 2, 2015 at 3:07 PM, Sergey Safarov <s.safarov at gmail.com> wrote:
>>>> In dialplan I set ringback variable
>>>> 
>>>> ?? ? ?<extension name="setRingback" continue="true">
>>>> ? ? ? ? ?<condition>
>>>> ? ? ? ? ? ? <action application="set" data="ringback=${us-ring}"/>
>>>> ? ? ? ? ?</condition>
>>>> ? ? ? </extension>
>>>> And diaplan terminate call on extension
>>>> 
>>>> <extension name="agent_login">
>>>> ? ?<condition field="${destination_number}" expression="^agent_login$">
>>>> ? ? ? <action application="set"
>>>> ? ? ? ? ? ? ? data="ignore=${callcenter_config(agent set status
>>>> ${caller_id_number}@${sip_to_host} 'Available')}"
>>>> ? ? ? ? ? ? ? inline="true"/>
>>>> ? ? ? <action application="ring_ready"/>
>>>> ? ? ? <action application="pre_answer"/>
>>>> ? ? ? <action application="sleep" data="2500"/>
>>>> ? ? ? <action application="answer"/>
>>>> ? ? ? <action application="sleep" data="700"/>
>>>> ? ?</condition>
>>>> </extension>
>>>> 
>>>> But ringback is not generaded. Why is it posible?
>>>> 
>>>> Sergey
>>> 
>>> 
>>> _________________________________________________________________________
>>> 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
>> 
>> 

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From mike at jerris.com  Thu Jul  2 19:30:48 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 2 Jul 2015 11:30:48 -0400
Subject: [Freeswitch-users] Disk error / database error help
In-Reply-To: <CALXCt0ocnA7QJKbJ0gwpsJHkeKHTO9L6KjD+LPxcHLRsiEL5NA@mail.gmail.com>
References: <c357215130c3dee62c7803ab7ef5c42b@www.webmail.co.za>
	<CAFiqYukdk08m2EoNVZ75xot+Fr=EJDt8y1mFqo8_i-xuvxXDuA@mail.gmail.com>
	<ed6289f3b6f594234ea0e9907c73f721@www.webmail.co.za>
	<CALXCt0ocnA7QJKbJ0gwpsJHkeKHTO9L6KjD+LPxcHLRsiEL5NA@mail.gmail.com>
Message-ID: <A69D283B-46EC-4E75-87F9-B62DB1E49A56@jerris.com>

unless you have any need for persistent data in there, such as voicemail

> On Jul 2, 2015, at 11:00 AM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> 
> you probably have a failing disk...
> 
> btw, is better to mount freeswitch db in a tmpfs ram dilesystem
> 
> 
> 
> On Thu, Jul 2, 2015 at 4:50 PM, clive engelberg <clive18 at webmail.co.za <mailto:clive18 at webmail.co.za>> wrote:
> Hi
> 
> My dmesg seems to be full of IPtables "denied" statements.
> 
> I guess I need to reboot to get a clean dmesg.
> 
> Thanks and regards
> Clive
> 
> On Thu, 2 Jul 2015 13:18:03 +0100 Steven Ayre <steveayre at gmail.com <mailto:steveayre at gmail.com>> wrote
> 
> Do you see any errors in the output of 'dmesg'? If there are errors on your hard disk or controller you'd see messages in there.
> 
> On 2 July 2015 at 10:21, clive engelberg <clive18 at webmail.co.za <mailto:clive18 at webmail.co.za>> wrote:
> Hi Guys.
> 
> I have had this happen a few times already, where Freeswitch seems to have a
> database error. This starts off slowly, and within a short time, it completely
> fills the disk drive with log error files.
> 
> the log files look like this:
> 2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR [disk
> I/O error]
> BEGIN EXCLUSIVE
> 2015-07-01 15:19:35.470881 [CRIT] switch_core_sqldb.c:1565 ERROR [disk I/O
> error]
> 2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR
> [cannot commit - no transaction is active]
> 
> Any ideas what is causing this.
> I suspect maybe a hardware disk error.. not sure.
> 
> Thanks in advance
> Clive
> 
> 
> 
> ____________________________________________________________
> South Africas premier fr ee email service - www.webmail.co.za <>
> 
> 
> 
> _________________________________________________________________________
> 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/%20freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> South Africa premier free email service - webmail.co.za <http://www.webmail.co.za/> 
> _________________________________________________________________________
> 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/>
> 
> 
> 
> -- 
> Sincerely,
> 
> Giovanni Maruzzelli
> Cell : +39-347-2665618
> _________________________________________________________________________
> 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 Jul  2 19:41:45 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 2 Jul 2015 17:41:45 +0200
Subject: [Freeswitch-users] Disk error / database error help
In-Reply-To: <A69D283B-46EC-4E75-87F9-B62DB1E49A56@jerris.com>
References: <c357215130c3dee62c7803ab7ef5c42b@www.webmail.co.za>
	<CAFiqYukdk08m2EoNVZ75xot+Fr=EJDt8y1mFqo8_i-xuvxXDuA@mail.gmail.com>
	<ed6289f3b6f594234ea0e9907c73f721@www.webmail.co.za>
	<CALXCt0ocnA7QJKbJ0gwpsJHkeKHTO9L6KjD+LPxcHLRsiEL5NA@mail.gmail.com>
	<A69D283B-46EC-4E75-87F9-B62DB1E49A56@jerris.com>
Message-ID: <CALXCt0oZLuuU=6mvY0F_LAX4NDqXftprNiZNBdW-1mca38vubg@mail.gmail.com>

On Thu, Jul 2, 2015 at 5:30 PM, Michael Jerris <mike at jerris.com> wrote:

> unless you have any need for persistent data in there, such as voicemail
>
>
oooops, my fault, agreed!
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From ssinyagin at gmail.com  Thu Jul  2 19:46:21 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Thu, 2 Jul 2015 17:46:21 +0200
Subject: [Freeswitch-users] Video playback in 1.7
In-Reply-To: <CACOYK=hfnRoz5QWQaFf8cKNjRVeT1CiqF7_Yubn54Pow8uK46A@mail.gmail.com>
References: <CACOYK=h8effa=OwvVweQT9CC3a4u3M8Ff7mzgaDHQ4di26u9Rw@mail.gmail.com>
	<CALXCt0qf+a1tYQt=Wb0DsdwDUCATwpLoRq7J2MeyHy+aHezwnw@mail.gmail.com>
	<CACOYK=jugw0Z5Z6SXh98P5Xd4gM0carohDJ2-_ftuev=jAAKCA@mail.gmail.com>
	<CAKbxfG8d8XpdRvSCq1ZYut71E5+bWQtPb8QnKXSkt-dHBdNXMw@mail.gmail.com>
	<CACOYK=hE=evF8c+HDjmQP1=2-ZCwSe0GAVkPkcjUCEsrv+vMuA@mail.gmail.com>
	<CAKbxfG_5J6uUmLCYPAUsa47dsSPx2A4JaHycGn-fB56nEFE-SA@mail.gmail.com>
	<CACOYK=hnVpaZNbXXTxaj-tcYBP6q_OL3fVxiggVN7=BUzDDVwg@mail.gmail.com>
	<CAKbxfG8xUg+hEoAp76pdhYuji5YA6WkcUJrfEfNpdby2TWx-Zw@mail.gmail.com>
	<CACOYK=jsR4xDPz-SoZ-YKY-J17BL+_ryCZiCso0xkO+dCvNdpw@mail.gmail.com>
	<1C0C06B1-187F-4D2D-A305-76763F975890@jerris.com>
	<CAKbxfG8028tSL-8wj-=qcNQ3xk08xM7uwCqXg-OSM_f=JFc5GQ@mail.gmail.com>
	<CACOYK=iwPiRRC6uMDiDm83MusFDLpipogmDnCJ5y6Xbd-u6q-w@mail.gmail.com>
	<CAEwTsAiG6ob-YmFm++0ODxMNMtQQaUhf2XaGPCrkaJdhZAKmZg@mail.gmail.com>
	<CACOYK=hfnRoz5QWQaFf8cKNjRVeT1CiqF7_Yubn54Pow8uK46A@mail.gmail.com>
Message-ID: <CACOYK=jX1TBRJ2htCQPypo8vBsAWU+3TiaU_-R6T+XKcQsiEsA@mail.gmail.com>

reported here:  https://freeswitch.org/jira/browse/FS-7762

On Wed, Jul 1, 2015 at 9:39 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> no, doesn't seem so:
>
> root at fsdemo:~# aptitude search libav | egrep '^i'
> i   libav-tools                     - Multimedia player, encoder and
> transcoder
> i A libavahi-client3                - Avahi client
> library
> i A libavahi-common-data            - Avahi common data
> files
> i A libavahi-common3                - Avahi common
> library
> i A libavc1394-0                    - control IEEE 1394 audio/video
> devices
> i A libavcodec56                    - Libav codec
> library
> i A libavdevice55                   - Libav device handling
> library
> i A libavfilter5                    - Libav video filtering
> library
> i A libavformat56                   - Libav file format
> library
> i A libavresample2                  - Libav audio resampling
> library
> i A libavutil54                     - Libav utility
> library
> root at fsdemo:~#
>
>
>
>
> On Wed, Jul 1, 2015 at 8:23 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> do you have the libav -extra package (I don't recall the full name)
>> installed?
>>
>>
>> On Wednesday, July 1, 2015, Stanislav Sinyagin <ssinyagin at gmail.com>
>> wrote:
>>
>>> This was all on a freshly installed Jessie, with your 1.7 debs. I will
>>> have a closer look and document the issue later, now I only had time for a
>>> quick check.
>>> On Jul 1, 2015 3:18 AM, "Anthony Minessale" <anthony.minessale at gmail.com>
>>> wrote:
>>>
>>>> Like I said, you need the precise versions we have detailed in our
>>>> Debian jessie packaging.   It does not work on older versions of libav*
>>>> If you don't want to use jessie you need to see the versions of
>>>> everything we use and manually build it all and its full chain of cross
>>>> depends.
>>>>
>>>>
>>>> On Tue, Jun 30, 2015 at 7:54 PM, Michael Jerris <mike at jerris.com>
>>>> wrote:
>>>>
>>>>> That error happens when avresample_open call fails.  This is going to
>>>>> be some sort of issue with how libav was built as this is known working.
>>>>>
>>>>> On Jun 30, 2015, at 7:51 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
>>>>> wrote:
>>>>>
>>>>> I made a few more tests: the original file has AAC audio, and avconv
>>>>> needs "-strict experimental" option to process that. Probably that's why
>>>>> FreeSWITCH crashes. After I converted the video to 320x240 and MP3 audio, I
>>>>> get a different error:
>>>>>
>>>>> 2015-07-01 01:46:22.701171 [ERR] avformat.c:1136 Failed to initialize
>>>>> the resampling context
>>>>>
>>>>> same error if I need to produce 48kHz OPUS or 8kHZ G711.
>>>>>
>>>>> VLC plays back both the original and converted videos so far.
>>>>>
>>>>> I'll play around with it during the week.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jun 30, 2015 at 8:45 PM, Anthony Minessale <
>>>>> anthony.minessale at gmail.com> wrote:
>>>>>
>>>>>> Depending on what ext you have and the proper build of mod_av I can't
>>>>>> comment.
>>>>>>
>>>>>> I can confirm with the properly installed debian packages for jessie
>>>>>> that at least mp4 files work either by reference to .mp4 files or by using
>>>>>> av:///path/to/file.mp4
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Jun 30, 2015 at 12:46 PM, Stanislav Sinyagin <
>>>>>> ssinyagin at gmail.com> wrote:
>>>>>>
>>>>>>> av:///tmp/somefile.ext crashed the daemon, I didn't yet find the
>>>>>>> time to analyze it and file a jira.
>>>>>>> On Jun 30, 2015 7:43 PM, "Anthony Minessale" <
>>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>>
>>>>>>>> mod_av is not an endpoint its a codec and file format module.
>>>>>>>>
>>>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:213
>>>>>>>> Adding Codec H264 99 H264 Video 90000hz 0ms (VBR)
>>>>>>>>
>>>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:292
>>>>>>>> Adding Application 'record_av'
>>>>>>>>
>>>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:338
>>>>>>>> Adding API Function 'av_format'
>>>>>>>>
>>>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:338
>>>>>>>> Adding API Function 'av_codec'
>>>>>>>>
>>>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390
>>>>>>>> Adding File Format 'av'
>>>>>>>>
>>>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390
>>>>>>>> Adding File Format 'rtmp'
>>>>>>>>
>>>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390
>>>>>>>> Adding File Format 'mp4'
>>>>>>>>
>>>>>>>> 2015-06-30 12:40:40.248642 [NOTICE] switch_loadable_module.c:390
>>>>>>>> Adding File Format 'mov'
>>>>>>>>
>>>>>>>>
>>>>>>>> It was av:// not avformat:// i was remembering the old version.
>>>>>>>>
>>>>>>>>
>>>>>>>> if av:///tmp/somefile.ext works then it can be added to mod_avformat_load
>>>>>>>> function in avformat.c:1949 ish  to register the extension for convenience.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Tue, Jun 30, 2015 at 2:07 AM, Stanislav Sinyagin <
>>>>>>>> ssinyagin at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> mod_av doesn't seem to be registering an endpoint, so prefixing
>>>>>>>>> avformat:// or av:// does not help. I'll have a closer look later and
>>>>>>>>> probably open a Jira
>>>>>>>>>
>>>>>>>>> On Tue, Jun 30, 2015 at 1:18 AM, Anthony Minessale <
>>>>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Webm has its own module.  Av and vlc both have broken webm at the
>>>>>>>>>> time of coding.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> mod_vlc can play other formats but they are not registered in the
>>>>>>>>>> module by file exten however you can use vlc:// syntax.
>>>>>>>>>>
>>>>>>>>>> mp4 is the safest bet because it works in mod_av which is more
>>>>>>>>>> stable than vlc.  More formats can be added to mod_av as well but I don't
>>>>>>>>>> remember if its as easy as avformat://<path>
>>>>>>>>>>
>>>>>>>>>> We don't have any choosing best format etc.  It's not going to be
>>>>>>>>>> a point of focus to squeeze performance out of stuff like that in this
>>>>>>>>>> stage of development.
>>>>>>>>>>
>>>>>>>>>> On Monday, June 29, 2015, Stanislav Sinyagin <ssinyagin at gmail.com>
>>>>>>>>>> wrote:
>>>>>>>>>>
>>>>>>>>>>> by the way, is there a way for playback to select a best matching
>>>>>>>>>>> encoding, like it does with audio sample rates?
>>>>>>>>>>>
>>>>>>>>>>> On Tue, Jun 30, 2015 at 12:56 AM, Giovanni Maruzzelli <
>>>>>>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>>>>>> > h264 I believe is supported...
>>>>>>>>>>> >
>>>>>>>>>>> > On Tue, Jun 30, 2015 at 12:48 AM, Stanislav Sinyagin <
>>>>>>>>>>> ssinyagin at gmail.com>
>>>>>>>>>>> > wrote:
>>>>>>>>>>> >>
>>>>>>>>>>> >> the newest 1.7 freeswitch successfully played an .mp4 file
>>>>>>>>>>> with
>>>>>>>>>>> >> "playback" application, and the picture was sent to an VP8
>>>>>>>>>>> client
>>>>>>>>>>> >> (linphone on Android).
>>>>>>>>>>> >>
>>>>>>>>>>> >> The playback took about 20% CPU usage on a Xeon core --
>>>>>>>>>>> probably
>>>>>>>>>>> >> because of resising work. The source file was taken from
>>>>>>>>>>> >> http://www.quirksmode.org/html5/tests/video.html
>>>>>>>>>>> >>
>>>>>>>>>>> >>
>>>>>>>>>>> >> Question: what other file formats are supported?
>>>>>>>>>>> >>
>>>>>>>>>>> >> I tried .ogv and .webm, but I got "Invalid file format" error.
>>>>>>>>>>> >>
>>>>>>>>>>> >> thanks
>>>>>>>>>>> >>
>>>>>>>>>>> >>
>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>> >> 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
>>>>>>>>>>> > Cell : +39-347-2665618
>>>>>>>>>>> >
>>>>>>>>>>> >
>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>> > 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
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>> 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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> 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
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> 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
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> 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
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> 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
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> 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
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> 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
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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 dragic.dusan at gmail.com  Thu Jul  2 21:00:25 2015
From: dragic.dusan at gmail.com (=?UTF-8?B?RHXFoWFuIERyYWdpxIc=?=)
Date: Thu, 2 Jul 2015 19:00:25 +0200
Subject: [Freeswitch-users] Why ringback is not generated?
In-Reply-To: <CAHtxdDe=4Wkc2Kw=tsq2MLESmDjrhPSgUvYjHA0kzcQiQgNnbg@mail.gmail.com>
References: <CAHtxdDfhMUSoHjYYW9yLE613+rM=x8f-iWmcKJ9NzbuGE-3tAA@mail.gmail.com>
	<CAHtxdDe=4Wkc2Kw=tsq2MLESmDjrhPSgUvYjHA0kzcQiQgNnbg@mail.gmail.com>
Message-ID: <CAPHNFco65d=YXkTPOE75eKwcsQwET-6=HY0D1TRen6BZb9Hb8A@mail.gmail.com>

I think it's working as designed, pre_answer doesn't honour the
ringback variable and doesn't send anything automatically, you have to
send it yourself.

In your case you could probably use something like this on FS2:
<action application="pre_answer"/>
<action application="playback" data="tone_stream://${us-ring};loops=2"/>
<action application="answer"/>
<action application="playback" data="you_are_logged_in.wav"/>
<action application="hangup"/>

also you don't need "ring_ready", it only sends 180 without SDP.

On 2 July 2015 at 14:15, Sergey Safarov <s.safarov at gmail.com> wrote:
> Wireshark help locate root issue FS send 183 message but not start sending
> RTP media. FS start send media after sending 200 message.
>
> It is bug?
>
> On Thu, Jul 2, 2015 at 3:07 PM, Sergey Safarov <s.safarov at gmail.com> wrote:
>>
>> In dialplan I set ringback variable
>>
>>       <extension name="setRingback" continue="true">
>>          <condition>
>>             <action application="set" data="ringback=${us-ring}"/>
>>          </condition>
>>       </extension>
>> And diaplan terminate call on extension
>>
>> <extension name="agent_login">
>>    <condition field="${destination_number}" expression="^agent_login$">
>>       <action application="set"
>>               data="ignore=${callcenter_config(agent set status
>> ${caller_id_number}@${sip_to_host} 'Available')}"
>>               inline="true"/>
>>       <action application="ring_ready"/>
>>       <action application="pre_answer"/>
>>       <action application="sleep" data="2500"/>
>>       <action application="answer"/>
>>       <action application="sleep" data="700"/>
>>    </condition>
>> </extension>
>>
>> But ringback is not generaded. Why is it posible?
>>
>> Sergey
>
>
>
> _________________________________________________________________________
> 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



-- 
Du?an Dragi?


From anthony.minessale at gmail.com  Thu Jul  2 21:18:54 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 2 Jul 2015 12:18:54 -0500
Subject: [Freeswitch-users] Why ringback is not generated?
In-Reply-To: <CAPHNFco65d=YXkTPOE75eKwcsQwET-6=HY0D1TRen6BZb9Hb8A@mail.gmail.com>
References: <CAHtxdDfhMUSoHjYYW9yLE613+rM=x8f-iWmcKJ9NzbuGE-3tAA@mail.gmail.com>
	<CAHtxdDe=4Wkc2Kw=tsq2MLESmDjrhPSgUvYjHA0kzcQiQgNnbg@mail.gmail.com>
	<CAPHNFco65d=YXkTPOE75eKwcsQwET-6=HY0D1TRen6BZb9Hb8A@mail.gmail.com>
Message-ID: <CAKbxfG9OmDNUQrp6X-gF4U-_DJKYB3OZLbNsVrV4syRmziV74w@mail.gmail.com>

Correct,

The last response is exactly right.
The ringback var is only relevant to the bridge app.  pre_answer is just an
indication.



On Thu, Jul 2, 2015 at 12:00 PM, Du?an Dragi? <dragic.dusan at gmail.com>
wrote:

> I think it's working as designed, pre_answer doesn't honour the
> ringback variable and doesn't send anything automatically, you have to
> send it yourself.
>
> In your case you could probably use something like this on FS2:
> <action application="pre_answer"/>
> <action application="playback" data="tone_stream://${us-ring};loops=2"/>
> <action application="answer"/>
> <action application="playback" data="you_are_logged_in.wav"/>
> <action application="hangup"/>
>
> also you don't need "ring_ready", it only sends 180 without SDP.
>
> On 2 July 2015 at 14:15, Sergey Safarov <s.safarov at gmail.com> wrote:
> > Wireshark help locate root issue FS send 183 message but not start
> sending
> > RTP media. FS start send media after sending 200 message.
> >
> > It is bug?
> >
> > On Thu, Jul 2, 2015 at 3:07 PM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
> >>
> >> In dialplan I set ringback variable
> >>
> >>       <extension name="setRingback" continue="true">
> >>          <condition>
> >>             <action application="set" data="ringback=${us-ring}"/>
> >>          </condition>
> >>       </extension>
> >> And diaplan terminate call on extension
> >>
> >> <extension name="agent_login">
> >>    <condition field="${destination_number}" expression="^agent_login$">
> >>       <action application="set"
> >>               data="ignore=${callcenter_config(agent set status
> >> ${caller_id_number}@${sip_to_host} 'Available')}"
> >>               inline="true"/>
> >>       <action application="ring_ready"/>
> >>       <action application="pre_answer"/>
> >>       <action application="sleep" data="2500"/>
> >>       <action application="answer"/>
> >>       <action application="sleep" data="700"/>
> >>    </condition>
> >> </extension>
> >>
> >> But ringback is not generaded. Why is it posible?
> >>
> >> Sergey
> >
> >
> >
> > _________________________________________________________________________
> > 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
>
>
>
> --
> Du?an Dragi?
>
> _________________________________________________________________________
> 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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From william.king at quentustech.com  Thu Jul  2 21:47:16 2015
From: william.king at quentustech.com (William King)
Date: Thu, 2 Jul 2015 10:47:16 -0700
Subject: [Freeswitch-users] Disk error / database error help
In-Reply-To: <A69D283B-46EC-4E75-87F9-B62DB1E49A56@jerris.com>
References: <c357215130c3dee62c7803ab7ef5c42b@www.webmail.co.za>
	<CAFiqYukdk08m2EoNVZ75xot+Fr=EJDt8y1mFqo8_i-xuvxXDuA@mail.gmail.com>
	<ed6289f3b6f594234ea0e9907c73f721@www.webmail.co.za>
	<CALXCt0ocnA7QJKbJ0gwpsJHkeKHTO9L6KjD+LPxcHLRsiEL5NA@mail.gmail.com>
	<A69D283B-46EC-4E75-87F9-B62DB1E49A56@jerris.com>
Message-ID: <55957924.2080407@quentustech.com>

In which case it's probably better to use a dedicated DB rather than
sqlite.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 7/2/15 8:30 AM, Michael Jerris wrote:
> unless you have any need for persistent data in there, such as voicemail
> 
>> On Jul 2, 2015, at 11:00 AM, Giovanni Maruzzelli <gmaruzz at gmail.com
>> <mailto:gmaruzz at gmail.com>> wrote:
>>
>> you probably have a failing disk...
>>
>> btw, is better to mount freeswitch db in a tmpfs ram dilesystem
>>
>>
>>
>> On Thu, Jul 2, 2015 at 4:50 PM, clive engelberg <clive18 at webmail.co.za
>> <mailto:clive18 at webmail.co.za>> wrote:
>>
>>     Hi
>>
>>     My dmesg seems to be full of IPtables "denied" statements.
>>
>>     I guess I need to reboot to get a clean dmesg.
>>
>>     Thanks and regards
>>     Clive
>>
>>     On Thu, 2 Jul 2015 13:18:03 +0100 Steven Ayre <steveayre at gmail.com
>>     <mailto:steveayre at gmail.com>> wrote
>>
>>     Do you see any errors in the output of 'dmesg'? If there are
>>     errors on your hard disk or controller you'd see messages in there.
>>
>>     On 2 July 2015 at 10:21, clive engelberg <clive18 at webmail.co.za
>>     <mailto:clive18 at webmail.co.za>> wrote:
>>
>>         Hi Guys.
>>
>>         I have had this happen a few times already, where Freeswitch
>>         seems to have a
>>         database error. This starts off slowly, and within a short
>>         time, it completely
>>         fills the disk drive with log error files.
>>
>>         the log files look like this:
>>         2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583
>>         NATIVE SQL ERR [disk
>>         I/O error]
>>         BEGIN EXCLUSIVE
>>         2015-07-01 15:19:35.470881 [CRIT] switch_core_sqldb.c:1565
>>         ERROR [disk I/O
>>         error]
>>         2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583
>>         NATIVE SQL ERR
>>         [cannot commit - no transaction is active]
>>
>>         Any ideas what is causing this.
>>         I suspect maybe a hardware disk error.. not sure.
>>
>>         Thanks in advance
>>         Clive
>>
>>
>>
>>         ____________________________________________________________
>>         South Africas premier fr ee email service - www.webmail.co.za
>>
>>
>>
>>         _________________________________________________________________________
>>         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
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         <http://lists.freeswitch.org/mailman/options/%20freeswitch-users>
>>         http://www.freeswitch.org <http://www.freeswitch.org/>
>>
>>
>>     ------------------------------------------------------------------------
>>     South Africa premier free email service - webmail.co.za
>>     <http://www.webmail.co.za/>
>>     _________________________________________________________________________
>>     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
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>
>>
>>
>>
>> -- 
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>> _________________________________________________________________________
>> 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
> 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 sdevoy at bizfocused.com  Thu Jul  2 21:55:43 2015
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 2 Jul 2015 17:55:43 +0000
Subject: [Freeswitch-users] Make current failing on Certificates
In-Reply-To: <CAPNTnnpFn4O=9ATPyehKKXPizKOoGAW2hJyiikaqE43X64+pXw@mail.gmail.com>
References: <CAPNTnnpFn4O=9ATPyehKKXPizKOoGAW2hJyiikaqE43X64+pXw@mail.gmail.com>
Message-ID: <BY2PR0101MB0933722DB275CD39FD2C7B58BB970@BY2PR0101MB0933.prod.exchangelabs.com>

Is this problem in the new ?stable? 1.4.20 build or the current development tree?

Sean

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Guillermo Ruiz Camauer
Sent: Wednesday, July 01, 2015 6:54 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Make current failing on Certificates



I just tried to "make current" and got this:

error: server certificate verification failed. CAfile: /etc/ssl/certs/ca-certificates.crt CRLfile: none while accessing https://stash.freeswitch.org/scm/fs/freeswitch.git/info/refs
fatal: HTTP request failed
make[1]: *** [update] Error 1
make[1]: Leaving directory `/usr/src/freeswitch'
make: *** [current] Error 2


--
Guillermo Ruiz Camauer
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From mike at jerris.com  Thu Jul  2 22:07:18 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 2 Jul 2015 14:07:18 -0400
Subject: [Freeswitch-users] Make current failing on Certificates
In-Reply-To: <CAPNTnnpFn4O=9ATPyehKKXPizKOoGAW2hJyiikaqE43X64+pXw@mail.gmail.com>
References: <CAPNTnnpFn4O=9ATPyehKKXPizKOoGAW2hJyiikaqE43X64+pXw@mail.gmail.com>
Message-ID: <B4D7040C-6087-40D7-8C05-9652224DECB3@jerris.com>

The issue is the url you have checked out..  

git remote set-url origin https://freeswitch.org/stash/scm/fs/freeswitch.git <https://freeswitch.org/stash/scm/fs/freeswitch.git>

more info at:
http://lists.freeswitch.org/pipermail/freeswitch-users/2014-October/108907.html <http://lists.freeswitch.org/pipermail/freeswitch-users/2014-October/108907.html>



> On Jul 1, 2015, at 6:53 PM, Guillermo Ruiz Camauer <grcamauer at gmail.com> wrote:
> 
> 
> 
> I just tried to "make current" and got this:
> 
> error: server certificate verification failed. CAfile: /etc/ssl/certs/ca-certificates.crt CRLfile: none while accessing https://stash.freeswitch.org/scm/fs/freeswitch.git/info/refs <https://stash.freeswitch.org/scm/fs/freeswitch.git/info/refs>
> fatal: HTTP request failed
> make[1]: *** [update] Error 1
> make[1]: Leaving directory `/usr/src/freeswitch'
> make: *** [current] Error 2
> 
> 
> -- 
> Guillermo Ruiz Camauer
> _________________________________________________________________________
> 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 sangdrax8 at gmail.com  Thu Jul  2 22:13:32 2015
From: sangdrax8 at gmail.com (sangdrax8)
Date: Thu, 2 Jul 2015 14:13:32 -0400
Subject: [Freeswitch-users] mod_sms duplicate messages
Message-ID: <CAL8ieHovSzYzJS--wiEx8VPff1_bKBXugy+ZhT=NXfpF8d0NTw@mail.gmail.com>

I am trying to test out mod_sms for the chatplans and it seems to be
sending two messages when I call a send.  Can someone tell me if there
is some configuration I am doing wrong, of if I need to file a Jira
bug?  If I unload the mod_sms module, only one messages i sent as
expected but I was looking for the chatplan functionality.

I am currently testing on Version 1.4.19 git 73f45e3, as this was the
suggested branch for production usage.

my chatplan is very basic:
<include>
  <context name="public">
    <extension name="send">
      <condition>
        <action application="send"/>
      </condition>
    </extension>
  </context>
</include>

when I send a message I get the following in the CLI and two sip
messages are sent.

2015-07-02 18:03:38.950750 [INFO] mod_sms.c:336 Processing text
message 19995550008->19995550001 in context public
Chatplan: 19995550001 parsing [public->send] continue=false
Chatplan: 19995550001 at demo Absolute Condition [send]
Chatplan: 19995550001 at demo Action send()


With siptrace on I can see two send events that are identical but
happen less than a millisecond apart.

send 1173 bytes to tls/[192.168.0.40]:54156 at 18:03:38.965479:
send 1173 bytes to tls/[192.168.0.40]:54156 at 18:03:38.965878:


From steveayre at gmail.com  Thu Jul  2 22:15:20 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 2 Jul 2015 19:15:20 +0100
Subject: [Freeswitch-users] Disk error / database error help
In-Reply-To: <ed6289f3b6f594234ea0e9907c73f721@www.webmail.co.za>
References: <c357215130c3dee62c7803ab7ef5c42b@www.webmail.co.za>
	<CAFiqYukdk08m2EoNVZ75xot+Fr=EJDt8y1mFqo8_i-xuvxXDuA@mail.gmail.com>
	<ed6289f3b6f594234ea0e9907c73f721@www.webmail.co.za>
Message-ID: <CAFiqYuk8boPdBMBk0Z4NbZFfADha+oLOMgXFpk+6BVJLk-kgww@mail.gmail.com>

Try /var/log/messsages.log (and its archives), but yes iptables logging
could knock useful messages out rather quickly

On 2 July 2015 at 15:50, clive engelberg <clive18 at webmail.co.za> wrote:

> Hi
>
> My dmesg seems to be full of IPtables "denied" statements.
>
> I guess I need to reboot to get a clean dmesg.
>
> Thanks and regards
> Clive
>
> On Thu, 2 Jul 2015 13:18:03 +0100 Steven Ayre <steveayre at gmail.com> wrote
>
>  Do you see any errors in the output of 'dmesg'? If there are errors on
> your hard disk or controller you'd see messages in there.
>
> On 2 July 2015 at 10:21, clive engelberg <clive18 at webmail.co.za> wrote:
>
>> Hi Guys.
>>
>> I have had this happen a few times already, where Freeswitch seems to
>> have a
>> database error. This starts off slowly, and within a short time, it
>> completely
>> fills the disk drive with log error files.
>>
>> the log files look like this:
>> 2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR
>> [disk
>> I/O error]
>> BEGIN EXCLUSIVE
>> 2015-07-01 15:19:35.470881 [CRIT] switch_core_sqldb.c:1565 ERROR [disk I/O
>> error]
>> 2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR
>> [cannot commit - no transaction is active]
>>
>> Any ideas what is causing this.
>> I suspect maybe a hardware disk error.. not sure.
>>
>> Thanks in advance
>> Clive
>>
>>
>>
>> ____________________________________________________________
>> South Africas premier fr ee email service - www.webmail.co.za
>>
>>
>>
>> _________________________________________________________________________
>> 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://lists.freeswitch.org/mailman/options/%20freeswitch-users>
>> http://www.freeswitch.org
>
>
> ------------------------------
> South Africa premier free email service - webmail.co.za
> <http://www.webmail.co.za/>
> _________________________________________________________________________
> 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 steveayre at gmail.com  Thu Jul  2 22:24:29 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 2 Jul 2015 19:24:29 +0100
Subject: [Freeswitch-users] Disk error / database error help
In-Reply-To: <CALXCt0ocnA7QJKbJ0gwpsJHkeKHTO9L6KjD+LPxcHLRsiEL5NA@mail.gmail.com>
References: <c357215130c3dee62c7803ab7ef5c42b@www.webmail.co.za>
	<CAFiqYukdk08m2EoNVZ75xot+Fr=EJDt8y1mFqo8_i-xuvxXDuA@mail.gmail.com>
	<ed6289f3b6f594234ea0e9907c73f721@www.webmail.co.za>
	<CALXCt0ocnA7QJKbJ0gwpsJHkeKHTO9L6KjD+LPxcHLRsiEL5NA@mail.gmail.com>
Message-ID: <CAFiqYukVtSCCYndxj5pZwTTLXVQEG0fm_edxbZLjvFjQ4tjn8g@mail.gmail.com>

In some cases that might not help, since tmpfs can swap to disk if it needs
the space.

On 2 July 2015 at 16:00, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> you probably have a failing disk...
>
> btw, is better to mount freeswitch db in a tmpfs ram dilesystem
>
>
>
> On Thu, Jul 2, 2015 at 4:50 PM, clive engelberg <clive18 at webmail.co.za>
> wrote:
>
>> Hi
>>
>> My dmesg seems to be full of IPtables "denied" statements.
>>
>> I guess I need to reboot to get a clean dmesg.
>>
>> Thanks and regards
>> Clive
>>
>> On Thu, 2 Jul 2015 13:18:03 +0100 Steven Ayre <steveayre at gmail.com> wrote
>>
>>  Do you see any errors in the output of 'dmesg'? If there are errors on
>> your hard disk or controller you'd see messages in there.
>>
>> On 2 July 2015 at 10:21, clive engelberg <clive18 at webmail.co.za> wrote:
>>
>>> Hi Guys.
>>>
>>> I have had this happen a few times already, where Freeswitch seems to
>>> have a
>>> database error. This starts off slowly, and within a short time, it
>>> completely
>>> fills the disk drive with log error files.
>>>
>>> the log files look like this:
>>> 2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR
>>> [disk
>>> I/O error]
>>> BEGIN EXCLUSIVE
>>> 2015-07-01 15:19:35.470881 [CRIT] switch_core_sqldb.c:1565 ERROR [disk
>>> I/O
>>> error]
>>> 2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR
>>> [cannot commit - no transaction is active]
>>>
>>> Any ideas what is causing this.
>>> I suspect maybe a hardware disk error.. not sure.
>>>
>>> Thanks in advance
>>> Clive
>>>
>>>
>>>
>>> ____________________________________________________________
>>> South Africas premier fr ee email service - www.webmail.co.za
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> 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://lists.freeswitch.org/mailman/options/%20freeswitch-users>
>>> http://www.freeswitch.org
>>
>>
>> ------------------------------
>> South Africa premier free email service - webmail.co.za
>> <http://www.webmail.co.za/>
>> _________________________________________________________________________
>> 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
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> 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 william.king at quentustech.com  Thu Jul  2 22:34:48 2015
From: william.king at quentustech.com (William King)
Date: Thu, 2 Jul 2015 11:34:48 -0700
Subject: [Freeswitch-users] mod_sms duplicate messages
In-Reply-To: <CAL8ieHovSzYzJS--wiEx8VPff1_bKBXugy+ZhT=NXfpF8d0NTw@mail.gmail.com>
References: <CAL8ieHovSzYzJS--wiEx8VPff1_bKBXugy+ZhT=NXfpF8d0NTw@mail.gmail.com>
Message-ID: <55958448.1050704@quentustech.com>

If you are receiving the message over sip, and it is being delivered
unchanged to another sip endpoint, then you don't need to have an action
'send' in your chatplan. If you want to keep the chatplan action, then
you must set the header on the event that you are manually handling the
delivery of the sms message:
<action application="set" data="final_delivery=true"/>


William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 7/2/15 11:13 AM, sangdrax8 wrote:
> I am trying to test out mod_sms for the chatplans and it seems to be
> sending two messages when I call a send.  Can someone tell me if there
> is some configuration I am doing wrong, of if I need to file a Jira
> bug?  If I unload the mod_sms module, only one messages i sent as
> expected but I was looking for the chatplan functionality.
> 
> I am currently testing on Version 1.4.19 git 73f45e3, as this was the
> suggested branch for production usage.
> 
> my chatplan is very basic:
> <include>
>   <context name="public">
>     <extension name="send">
>       <condition>
>         <action application="send"/>
>       </condition>
>     </extension>
>   </context>
> </include>
> 
> when I send a message I get the following in the CLI and two sip
> messages are sent.
> 
> 2015-07-02 18:03:38.950750 [INFO] mod_sms.c:336 Processing text
> message 19995550008->19995550001 in context public
> Chatplan: 19995550001 parsing [public->send] continue=false
> Chatplan: 19995550001 at demo Absolute Condition [send]
> Chatplan: 19995550001 at demo Action send()
> 
> 
> With siptrace on I can see two send events that are identical but
> happen less than a millisecond apart.
> 
> send 1173 bytes to tls/[192.168.0.40]:54156 at 18:03:38.965479:
> send 1173 bytes to tls/[192.168.0.40]:54156 at 18:03:38.965878:
> 
> _________________________________________________________________________
> 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 sangdrax8 at gmail.com  Thu Jul  2 23:12:14 2015
From: sangdrax8 at gmail.com (sangdrax8)
Date: Thu, 2 Jul 2015 15:12:14 -0400
Subject: [Freeswitch-users] mod_sms duplicate messages
In-Reply-To: <55958448.1050704@quentustech.com>
References: <CAL8ieHovSzYzJS--wiEx8VPff1_bKBXugy+ZhT=NXfpF8d0NTw@mail.gmail.com>
	<55958448.1050704@quentustech.com>
Message-ID: <CAL8ieHoHvNeTjb3UEad8NUre_KEWc8Tng-J8Ka5CWTerkQu89A@mail.gmail.com>

I have set that field, and this solved the duplicate messages.  I knew
there had to be something I was missing but I didn't find any mention
of it on the mod_sms confluence page.

So this variable indicates that the message shouldn't be delivered
through SIP because I am calling the send my self, correct?  If I
don't call send, then I assume the message would get 1 delivery, but
what if I DO edit the message.  If I don't call send, will the message
be delivered with the edits, or do I then have to send it my self and
use this variable to indicate the original should no longer be
delivered?

Thank you!



On Thu, Jul 2, 2015 at 2:35 PM William King
<william.king at quentustech.com> wrote:
>
> If you are receiving the message over sip, and it is being delivered
> unchanged to another sip endpoint, then you don't need to have an action
> 'send' in your chatplan. If you want to keep the chatplan action, then
> you must set the header on the event that you are manually handling the
> delivery of the sms message:
> <action application="set" data="final_delivery=true"/>
>
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
>
> On 7/2/15 11:13 AM, sangdrax8 wrote:
> > I am trying to test out mod_sms for the chatplans and it seems to be
> > sending two messages when I call a send.  Can someone tell me if there
> > is some configuration I am doing wrong, of if I need to file a Jira
> > bug?  If I unload the mod_sms module, only one messages i sent as
> > expected but I was looking for the chatplan functionality.
> >
> > I am currently testing on Version 1.4.19 git 73f45e3, as this was the
> > suggested branch for production usage.
> >
> > my chatplan is very basic:
> > <include>
> >   <context name="public">
> >     <extension name="send">
> >       <condition>
> >         <action application="send"/>
> >       </condition>
> >     </extension>
> >   </context>
> > </include>
> >
> > when I send a message I get the following in the CLI and two sip
> > messages are sent.
> >
> > 2015-07-02 18:03:38.950750 [INFO] mod_sms.c:336 Processing text
> > message 19995550008->19995550001 in context public
> > Chatplan: 19995550001 parsing [public->send] continue=false
> > Chatplan: 19995550001 at demo Absolute Condition [send]
> > Chatplan: 19995550001 at demo Action send()
> >
> >
> > With siptrace on I can see two send events that are identical but
> > happen less than a millisecond apart.
> >
> > send 1173 bytes to tls/[192.168.0.40]:54156 at 18:03:38.965479:
> > send 1173 bytes to tls/[192.168.0.40]:54156 at 18:03:38.965878:
> >
> > _________________________________________________________________________
> > 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


From grcamauer at gmail.com  Fri Jul  3 01:30:15 2015
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Thu, 2 Jul 2015 18:30:15 -0300
Subject: [Freeswitch-users] Make current failing on Certificates
In-Reply-To: <B4D7040C-6087-40D7-8C05-9652224DECB3@jerris.com>
References: <CAPNTnnpFn4O=9ATPyehKKXPizKOoGAW2hJyiikaqE43X64+pXw@mail.gmail.com>
	<B4D7040C-6087-40D7-8C05-9652224DECB3@jerris.com>
Message-ID: <CAPNTnnpbyh301kQE_t7pjVKu5v0BCUfhgHQm8GcJiMOJAUEP=A@mail.gmail.com>

That fixed it, thanks!

Guillermo

On Thu, Jul 2, 2015 at 3:07 PM, Michael Jerris <mike at jerris.com> wrote:

> The issue is the url you have checked out..
>
> git remote set-url origin *https://freeswitch.org/stash/scm/fs/freeswitch.git
> <https://freeswitch.org/stash/scm/fs/freeswitch.git>*
>
> more info at:
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2014-October/108907.html
>
>
>
> On Jul 1, 2015, at 6:53 PM, Guillermo Ruiz Camauer <grcamauer at gmail.com>
> wrote:
>
>
>
> I just tried to "make current" and got this:
>
> error: server certificate verification failed. CAfile:
> /etc/ssl/certs/ca-certificates.crt CRLfile: none while accessing
> https://stash.freeswitch.org/scm/fs/freeswitch.git/info/refs
> fatal: HTTP request failed
> make[1]: *** [update] Error 1
> make[1]: Leaving directory `/usr/src/freeswitch'
> make: *** [current] Error 2
>
>
> --
> Guillermo Ruiz Camauer
>  _________________________________________________________________________
> 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
>



-- 
Guillermo Ruiz Camauer
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From william.king at quentustech.com  Fri Jul  3 04:35:36 2015
From: william.king at quentustech.com (William King)
Date: Thu, 2 Jul 2015 17:35:36 -0700
Subject: [Freeswitch-users] mod_sms duplicate messages
In-Reply-To: <CAL8ieHoHvNeTjb3UEad8NUre_KEWc8Tng-J8Ka5CWTerkQu89A@mail.gmail.com>
References: <CAL8ieHovSzYzJS--wiEx8VPff1_bKBXugy+ZhT=NXfpF8d0NTw@mail.gmail.com>
	<55958448.1050704@quentustech.com>
	<CAL8ieHoHvNeTjb3UEad8NUre_KEWc8Tng-J8Ka5CWTerkQu89A@mail.gmail.com>
Message-ID: <5595D8D8.9070009@quentustech.com>

If you make a change to the message event, it should be delivered with
the change you made. I would suggest testing/exporing, and reporting
back your results. If you find a use case that isn't well handled, I'd
be very interested. I'm already deep into the SMS handling code with the
new mod_smpp module about to be merged into master, so now is a good
time to point out bugs, or feature requests.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 7/2/15 12:12 PM, sangdrax8 wrote:
> I have set that field, and this solved the duplicate messages.  I knew
> there had to be something I was missing but I didn't find any mention
> of it on the mod_sms confluence page.
> 
> So this variable indicates that the message shouldn't be delivered
> through SIP because I am calling the send my self, correct?  If I
> don't call send, then I assume the message would get 1 delivery, but
> what if I DO edit the message.  If I don't call send, will the message
> be delivered with the edits, or do I then have to send it my self and
> use this variable to indicate the original should no longer be
> delivered?
> 
> Thank you!
> 
> 
> 
> On Thu, Jul 2, 2015 at 2:35 PM William King
> <william.king at quentustech.com> wrote:
>>
>> If you are receiving the message over sip, and it is being delivered
>> unchanged to another sip endpoint, then you don't need to have an action
>> 'send' in your chatplan. If you want to keep the chatplan action, then
>> you must set the header on the event that you are manually handling the
>> delivery of the sms message:
>> <action application="set" data="final_delivery=true"/>
>>
>>
>> William King
>> Senior Engineer
>> Quentus Technologies, INC
>> 1037 NE 65th St Suite 273
>> Seattle, WA 98115
>> Main:   (877) 211-9337
>> Office: (206) 388-4772
>> Cell:   (253) 686-5518
>> william.king at quentustech.com
>>
>> On 7/2/15 11:13 AM, sangdrax8 wrote:
>>> I am trying to test out mod_sms for the chatplans and it seems to be
>>> sending two messages when I call a send.  Can someone tell me if there
>>> is some configuration I am doing wrong, of if I need to file a Jira
>>> bug?  If I unload the mod_sms module, only one messages i sent as
>>> expected but I was looking for the chatplan functionality.
>>>
>>> I am currently testing on Version 1.4.19 git 73f45e3, as this was the
>>> suggested branch for production usage.
>>>
>>> my chatplan is very basic:
>>> <include>
>>>   <context name="public">
>>>     <extension name="send">
>>>       <condition>
>>>         <action application="send"/>
>>>       </condition>
>>>     </extension>
>>>   </context>
>>> </include>
>>>
>>> when I send a message I get the following in the CLI and two sip
>>> messages are sent.
>>>
>>> 2015-07-02 18:03:38.950750 [INFO] mod_sms.c:336 Processing text
>>> message 19995550008->19995550001 in context public
>>> Chatplan: 19995550001 parsing [public->send] continue=false
>>> Chatplan: 19995550001 at demo Absolute Condition [send]
>>> Chatplan: 19995550001 at demo Action send()
>>>
>>>
>>> With siptrace on I can see two send events that are identical but
>>> happen less than a millisecond apart.
>>>
>>> send 1173 bytes to tls/[192.168.0.40]:54156 at 18:03:38.965479:
>>> send 1173 bytes to tls/[192.168.0.40]:54156 at 18:03:38.965878:
>>>
>>> _________________________________________________________________________
>>> 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
> 


From nandy1925 at gmail.com  Fri Jul  3 05:47:51 2015
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Fri, 3 Jul 2015 09:47:51 +0800
Subject: [Freeswitch-users] Which for agent.pem? openssl or gentls_cert
Message-ID: <CAHKqMb6A0U+JG4gS2iY88uSu1o5uhY44vuaHaFaOGEwhxK3UxA@mail.gmail.com>

Hi to all,

I'm testing mod_verto on FS v1.5.14 Debian Wheezy 64-bit .  In setting up
the certificates (self-signed), I followed the wiki. I encounter this WS
SETUP FAILED error even on Windows Chrome inspite of  all ports (443,
5061,5081,8081, 8082 and 7443) are listening.  I'm suspecting a certificate
setup problem.

In setting up the WSS certificates, it refers to WebRTC wiki, specifically
bkw's link:

https://freeswitch.org/confluence/display/FREESWITCH/WebRTC#WebRTC-Aquickhowtofrombkw(BrianK.West):

So far openssl is used to generate the certificates.

This portion # Setup Sofia TLS the "agent.pem" is created and led me to ...

https://wiki.freeswitch.org/wiki/SIP_TLS#Configuration

However, gentls_cert is used to create "agent.pem"! Is there a conflict
here? Which one to use?

I appreciate for your guidance since I'm new to SSL/TLS.

/Nandy
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From nandy1925 at gmail.com  Fri Jul  3 05:52:34 2015
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Fri, 3 Jul 2015 09:52:34 +0800
Subject: [Freeswitch-users] Make current failing on Certificates
In-Reply-To: <CAPNTnnpbyh301kQE_t7pjVKu5v0BCUfhgHQm8GcJiMOJAUEP=A@mail.gmail.com>
References: <CAPNTnnpFn4O=9ATPyehKKXPizKOoGAW2hJyiikaqE43X64+pXw@mail.gmail.com>
	<B4D7040C-6087-40D7-8C05-9652224DECB3@jerris.com>
	<CAPNTnnpbyh301kQE_t7pjVKu5v0BCUfhgHQm8GcJiMOJAUEP=A@mail.gmail.com>
Message-ID: <CAHKqMb62d1x2Ff4DLe242cP-EW=nDy5ydZKVN=wYU33S6N_zzQ@mail.gmail.com>

Thanks, too, Mike and Guillermo. :-)



On Fri, Jul 3, 2015 at 5:30 AM, Guillermo Ruiz Camauer <grcamauer at gmail.com>
wrote:

> That fixed it, thanks!
>
> Guillermo
>
> On Thu, Jul 2, 2015 at 3:07 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> The issue is the url you have checked out..
>>
>> git remote set-url origin *https://freeswitch.org/stash/scm/fs/freeswitch.git
>> <https://freeswitch.org/stash/scm/fs/freeswitch.git>*
>>
>> more info at:
>>
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2014-October/108907.html
>>
>>
>>
>> On Jul 1, 2015, at 6:53 PM, Guillermo Ruiz Camauer <grcamauer at gmail.com>
>> wrote:
>>
>>
>>
>> I just tried to "make current" and got this:
>>
>> error: server certificate verification failed. CAfile:
>> /etc/ssl/certs/ca-certificates.crt CRLfile: none while accessing
>> https://stash.freeswitch.org/scm/fs/freeswitch.git/info/refs
>> fatal: HTTP request failed
>> make[1]: *** [update] Error 1
>> make[1]: Leaving directory `/usr/src/freeswitch'
>> make: *** [current] Error 2
>>
>>
>> --
>> Guillermo Ruiz Camauer
>>  _________________________________________________________________________
>> 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
>>
>
>
>
> --
> Guillermo Ruiz Camauer
>
> _________________________________________________________________________
> 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  Fri Jul  3 09:42:41 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 3 Jul 2015 07:42:41 +0200
Subject: [Freeswitch-users] Which for agent.pem? openssl or gentls_cert
In-Reply-To: <CAHKqMb6A0U+JG4gS2iY88uSu1o5uhY44vuaHaFaOGEwhxK3UxA@mail.gmail.com>
References: <CAHKqMb6A0U+JG4gS2iY88uSu1o5uhY44vuaHaFaOGEwhxK3UxA@mail.gmail.com>
Message-ID: <CALXCt0rS0azCjCzjvtGKHJzt_=WyKEJWp3uW=gevbhhjsByOSw@mail.gmail.com>

Follow exactly, step by step, the instruction in confluence page
"freeswitch 1.6".
They work and are up to date.

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On Jul 3, 2015 3:48 AM, "Nandy Dagondon" <nandy1925 at gmail.com> wrote:

> Hi to all,
>
> I'm testing mod_verto on FS v1.5.14 Debian Wheezy 64-bit .  In setting up
> the certificates (self-signed), I followed the wiki. I encounter this WS
> SETUP FAILED error even on Windows Chrome inspite of  all ports (443,
> 5061,5081,8081, 8082 and 7443) are listening.  I'm suspecting a certificate
> setup problem.
>
> In setting up the WSS certificates, it refers to WebRTC wiki, specifically
> bkw's link:
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/WebRTC#WebRTC-Aquickhowtofrombkw(BrianK.West):
>
> So far openssl is used to generate the certificates.
>
> This portion # Setup Sofia TLS the "agent.pem" is created and led me to ...
>
> https://wiki.freeswitch.org/wiki/SIP_TLS#Configuration
>
> However, gentls_cert is used to create "agent.pem"! Is there a conflict
> here? Which one to use?
>
> I appreciate for your guidance since I'm new to SSL/TLS.
>
> /Nandy
>
>
>
> _________________________________________________________________________
> 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 gliu at tecomtech.com  Fri Jul  3 10:39:25 2015
From: gliu at tecomtech.com (Bryan)
Date: Fri, 3 Jul 2015 14:39:25 +0800
Subject: [Freeswitch-users] No Video in Media Proxy mode
Message-ID: <003f01d0b55b$1f25b280$5d711780$@com>

Hi, I'm testing FreeSWITCH with Android Linphone. I found the Video stream
was not forwarded by FS. Audio is OK.

 

Any Ideas? Or this is a bug?

 

The test envrionmet as following:

Android Linphone < -- > Wifi Router < -- > FS < -- >PC Linphone.

 

The FS Version is the Latest Windows MSI package. 

 

2015-07-03 14:31:57.718750 [NOTICE] switch_channel.c:1055 New Channel
sofia/inte

rnal/1009 at 172.16.1.115 [c8a19d44-3084-4539-b817-3276fd7ec43d]

2015-07-03 14:31:57.750000 [INFO] mod_dialplan_xml.c:635 Processing 1009
<1009>-

>1007 in context default

2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *1
execut

e_extension::dx XML features

2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *2
record

_session::C:/Program
Files/FreeSWITCH/recordings/1009.2015-07-03-14-31-57.wav

2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *3
execut

e_extension::cf XML features

2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *4
execut

e_extension::att_xfer XML features

2015-07-03 14:31:57.781250 [NOTICE] switch_channel.c:1055 New Channel
sofia/inte

rnal/sip:1007 at 172.16.1.115:2887 [4cfdf0e6-7010-467b-97b3-20e4dc295422]

2015-07-03 14:31:57.843750 [NOTICE] sofia.c:6716 Ring-Ready
sofia/internal/sip:1

007 at 172.16.1.115:2887!

2015-07-03 14:31:57.843750 [NOTICE] mod_sofia.c:2107 Ring-Ready
sofia/internal/1

009 at 172.16.1.115!

2015-07-03 14:31:57.859375 [NOTICE] switch_ivr_originate.c:527 Ring Ready
sofia/

internal/1009 at 172.16.1.115!

2015-07-03 14:32:00.312500 [NOTICE] sofia.c:7437 Channel
[sofia/internal/sip:100

7 at 172.16.1.115:2887] has been answered

2015-07-03 14:32:00.312500 [NOTICE] switch_core_media.c:4405
sofia/internal/sip:

1007 at 172.16.1.115:2887 Starting Video thread

2015-07-03 14:32:00.343750 [NOTICE] switch_core_media.c:4405
sofia/internal/1009

@172.16.1.115 Starting Video thread

2015-07-03 14:32:00.343750 [NOTICE] switch_ivr_originate.c:3522 Channel
[sofia/i

nternal/1009 at 172.16.1.115] has been answered

2015-07-03 14:32:00.843750 [INFO] switch_rtp.c:5832 Auto Changing port from
61.1

83.139.155:64131 to 172.16.15.111:7078

2015-07-03 14:32:23.750000 [NOTICE] sofia.c:952 Hangup
sofia/internal/sip:1007 at 1

72.16.1.115:2887 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]

2015-07-03 14:32:23.750000 [NOTICE] switch_ivr_bridge.c:1608 Hangup
sofia/intern

al/1009 at 172.16.1.115 [CS_EXECUTE] [NORMAL_CLEARING]

2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1633 Session 4
(sofia/

internal/sip:1007 at 172.16.1.115:2887) Ended

2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1637 Close Channel
sof

ia/internal/sip:1007 at 172.16.1.115:2887 [CS_DESTROY]

2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1633 Session 3
(sofia/

internal/1009 at 172.16.1.115) Ended

2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1637 Close Channel
sof

ia/internal/1009 at 172.16.1.115 [CS_DESTROY]

 

 

 

Tony

 

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From s.safarov at gmail.com  Fri Jul  3 10:52:05 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 3 Jul 2015 09:52:05 +0300
Subject: [Freeswitch-users] Why ringback is not generated?
In-Reply-To: <CAPHNFco65d=YXkTPOE75eKwcsQwET-6=HY0D1TRen6BZb9Hb8A@mail.gmail.com>
References: <CAHtxdDfhMUSoHjYYW9yLE613+rM=x8f-iWmcKJ9NzbuGE-3tAA@mail.gmail.com>
	<CAHtxdDe=4Wkc2Kw=tsq2MLESmDjrhPSgUvYjHA0kzcQiQgNnbg@mail.gmail.com>
	<CAPHNFco65d=YXkTPOE75eKwcsQwET-6=HY0D1TRen6BZb9Hb8A@mail.gmail.com>
Message-ID: <CAHtxdDdqGp4TRCcMBP53GNhg4AkO-dee52Srhn7sFxR1ygGLfw@mail.gmail.com>

Thank you Du?an, Anthony
Playback of tone_stream is help in my case.

On Thu, Jul 2, 2015 at 8:00 PM, Du?an Dragi? <dragic.dusan at gmail.com> wrote:

> I think it's working as designed, pre_answer doesn't honour the
> ringback variable and doesn't send anything automatically, you have to
> send it yourself.
>
> In your case you could probably use something like this on FS2:
> <action application="pre_answer"/>
> <action application="playback" data="tone_stream://${us-ring};loops=2"/>
> <action application="answer"/>
> <action application="playback" data="you_are_logged_in.wav"/>
> <action application="hangup"/>
>
> also you don't need "ring_ready", it only sends 180 without SDP.
>
> On 2 July 2015 at 14:15, Sergey Safarov <s.safarov at gmail.com> wrote:
> > Wireshark help locate root issue FS send 183 message but not start
> sending
> > RTP media. FS start send media after sending 200 message.
> >
> > It is bug?
> >
> > On Thu, Jul 2, 2015 at 3:07 PM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
> >>
> >> In dialplan I set ringback variable
> >>
> >>       <extension name="setRingback" continue="true">
> >>          <condition>
> >>             <action application="set" data="ringback=${us-ring}"/>
> >>          </condition>
> >>       </extension>
> >> And diaplan terminate call on extension
> >>
> >> <extension name="agent_login">
> >>    <condition field="${destination_number}" expression="^agent_login$">
> >>       <action application="set"
> >>               data="ignore=${callcenter_config(agent set status
> >> ${caller_id_number}@${sip_to_host} 'Available')}"
> >>               inline="true"/>
> >>       <action application="ring_ready"/>
> >>       <action application="pre_answer"/>
> >>       <action application="sleep" data="2500"/>
> >>       <action application="answer"/>
> >>       <action application="sleep" data="700"/>
> >>    </condition>
> >> </extension>
> >>
> >> But ringback is not generaded. Why is it posible?
> >>
> >> Sergey
> >
> >
> >
> > _________________________________________________________________________
> > 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
>
>
>
> --
> Du?an Dragi?
>
> _________________________________________________________________________
> 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 ricardas.stoma at gmail.com  Fri Jul  3 13:03:06 2015
From: ricardas.stoma at gmail.com (=?UTF-8?Q?Ri=C4=8Dardas_Stoma?=)
Date: Fri, 3 Jul 2015 12:03:06 +0300
Subject: [Freeswitch-users] Asterisk canreinvite (directmedia)
 equivalent in freeswitch
In-Reply-To: <D1B950F7.A0FDA%krice@freeswitch.org>
References: <CAFX-kAcTdg2SJQO3Mui7wC84cxpmS0piGSfTV98NKQ=ir_X+8A@mail.gmail.com>
	<D1B950F7.A0FDA%krice@freeswitch.org>
Message-ID: <CAFX-kAeFAnhks=hT8ULm86AD_He8LKMFsbOY7ObyY5TGuAapBA@mail.gmail.com>

I think everything is ok with my endpoints. I use two softphones (linphone
and zoiper). On linphone (leg A) i enable only GSM codec and on zoiper (leg
B) i enable only PCMA codec. When bypass_media_after bridge is disabled,
media goes through freeswitch and freeswitch does transcoding. In this case
i can hear voices.

When bypass_media_after_bridge is enabled, media goes from leg A to leg B
directly but the problem is that codecs do not match and i can hear only
digital noise. SIP trace looks like this http://i.imgur.com/iOn6ajJ.png

If i set PCMA codec on linphone (leg A) then everything is ok with direct
media between softphones, codecs matches and i can hear voices. What i want
is IF leg A and leg B codecs matches then bypass_media should take effect
and media should go directly from leg A to leg B. IF codecs do not match,
then media should go through freeswitch and freeswitch should do
transcoding.

Is this possible with freeswitch? Asterisk does this when canreinvite is
enabled ("If the clients use different codecs, Asterisk will not issue a
re-invite")
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From krice at freeswitch.org  Fri Jul  3 18:01:03 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 03 Jul 2015 14:01:03 +0000
Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <5596959fbf64_98b36b33426c8@resque-worker.14.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
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From brian at freeswitch.org  Fri Jul  3 18:05:17 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 3 Jul 2015 09:05:17 -0500
Subject: [Freeswitch-users] Asterisk canreinvite (directmedia)
 equivalent in freeswitch
In-Reply-To: <CAFX-kAeFAnhks=hT8ULm86AD_He8LKMFsbOY7ObyY5TGuAapBA@mail.gmail.com>
References: <CAFX-kAcTdg2SJQO3Mui7wC84cxpmS0piGSfTV98NKQ=ir_X+8A@mail.gmail.com>
	<D1B950F7.A0FDA%krice@freeswitch.org>
	<CAFX-kAeFAnhks=hT8ULm86AD_He8LKMFsbOY7ObyY5TGuAapBA@mail.gmail.com>
Message-ID: <CAEJMVkB0+PwiqbaRTOYEe6PK3y9uYXg4qs04=V8hgEELn7yzEA@mail.gmail.com>

For bypass to work you'll have to have some codecs in common!  Without that
it sounds like your end points don't behave properly in his case.

/b

On Friday, July 3, 2015, Ri?ardas Stoma <ricardas.stoma at gmail.com> wrote:

> I think everything is ok with my endpoints. I use two softphones (linphone
> and zoiper). On linphone (leg A) i enable only GSM codec and on zoiper (leg
> B) i enable only PCMA codec. When bypass_media_after bridge is disabled,
> media goes through freeswitch and freeswitch does transcoding. In this case
> i can hear voices.
>
> When bypass_media_after_bridge is enabled, media goes from leg A to leg B
> directly but the problem is that codecs do not match and i can hear only
> digital noise. SIP trace looks like this http://i.imgur.com/iOn6ajJ.png
>
> If i set PCMA codec on linphone (leg A) then everything is ok with direct
> media between softphones, codecs matches and i can hear voices. What i want
> is IF leg A and leg B codecs matches then bypass_media should take effect
> and media should go directly from leg A to leg B. IF codecs do not match,
> then media should go through freeswitch and freeswitch should do
> transcoding.
>
> Is this possible with freeswitch? Asterisk does this when canreinvite is
> enabled ("If the clients use different codecs, Asterisk will not issue a
> re-invite")
>


-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
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From brian at freeswitch.org  Fri Jul  3 18:05:57 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 3 Jul 2015 09:05:57 -0500
Subject: [Freeswitch-users] No Video in Media Proxy mode
In-Reply-To: <003f01d0b55b$1f25b280$5d711780$@com>
References: <003f01d0b55b$1f25b280$5d711780$@com>
Message-ID: <CAEJMVkBTD7dfj31cuuU6DzO0dGim0FqkgpjwYZmq713-=PXTSw@mail.gmail.com>

What rev of Freeswitch, and why are you using proxy media?

On Friday, July 3, 2015, Bryan <gliu at tecomtech.com> wrote:

>  Hi, I?m testing FreeSWITCH with Android Linphone. I found the Video
> stream was not forwarded by FS. Audio is OK.
>
>
>
> Any Ideas? Or this is a bug?
>
>
>
> The test envrionmet as following:
>
> Android Linphone < -- > Wifi Router < -- > FS < -- >PC Linphone.
>
>
>
> The FS Version is the Latest Windows MSI package.
>
>
>
> 2015-07-03 14:31:57.718750 [NOTICE] switch_channel.c:1055 New Channel
> sofia/inte
>
> rnal/1009 at 172.16.1.115 <javascript:_e(%7B%7D,'cvml','1009 at 172.16.1.115');>
> [c8a19d44-3084-4539-b817-3276fd7ec43d]
>
> 2015-07-03 14:31:57.750000 [INFO] mod_dialplan_xml.c:635 Processing 1009
> <1009>-
>
> >1007 in context default
>
> 2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *1
> execut
>
> e_extension::dx XML features
>
> 2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *2
> record
>
> _session::C:/Program
> Files/FreeSWITCH/recordings/1009.2015-07-03-14-31-57.wav
>
> 2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *3
> execut
>
> e_extension::cf XML features
>
> 2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *4
> execut
>
> e_extension::att_xfer XML features
>
> 2015-07-03 14:31:57.781250 [NOTICE] switch_channel.c:1055 New Channel
> sofia/inte
>
> rnal/sip:1007 at 172.16.1.115:2887 [4cfdf0e6-7010-467b-97b3-20e4dc295422]
>
> 2015-07-03 14:31:57.843750 [NOTICE] sofia.c:6716 Ring-Ready
> sofia/internal/sip:1
>
> 007 at 172.16.1.115:2887!
>
> 2015-07-03 14:31:57.843750 [NOTICE] mod_sofia.c:2107 Ring-Ready
> sofia/internal/1
>
> 009 at 172.16.1.115 <javascript:_e(%7B%7D,'cvml','009 at 172.16.1.115');>!
>
> 2015-07-03 14:31:57.859375 [NOTICE] switch_ivr_originate.c:527 Ring Ready
> sofia/
>
> internal/1009 at 172.16.1.115
> <javascript:_e(%7B%7D,'cvml','1009 at 172.16.1.115');>!
>
> 2015-07-03 14:32:00.312500 [NOTICE] sofia.c:7437 Channel
> [sofia/internal/sip:100
>
> 7 at 172.16.1.115:2887] has been answered
>
> 2015-07-03 14:32:00.312500 [NOTICE] switch_core_media.c:4405
> sofia/internal/sip:
>
> 1007 at 172.16.1.115:2887 Starting Video thread
>
> 2015-07-03 14:32:00.343750 [NOTICE] switch_core_media.c:4405
> sofia/internal/1009
>
> @172.16.1.115 Starting Video thread
>
> 2015-07-03 14:32:00.343750 [NOTICE] switch_ivr_originate.c:3522 Channel
> [sofia/i
>
> nternal/1009 at 172.16.1.115
> <javascript:_e(%7B%7D,'cvml','1009 at 172.16.1.115');>] has been answered
>
> 2015-07-03 14:32:00.843750 [INFO] switch_rtp.c:5832 Auto Changing port
> from 61.1
>
> 83.139.155:64131 to 172.16.15.111:7078
>
> 2015-07-03 14:32:23.750000 [NOTICE] sofia.c:952 Hangup
> sofia/internal/sip:1007 at 1
>
> 72.16.1.115:2887 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>
> 2015-07-03 14:32:23.750000 [NOTICE] switch_ivr_bridge.c:1608 Hangup
> sofia/intern
>
> al/1009 at 172.16.1.115 <javascript:_e(%7B%7D,'cvml','1009 at 172.16.1.115');>
> [CS_EXECUTE] [NORMAL_CLEARING]
>
> 2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1633 Session 4
> (sofia/
>
> internal/sip:1007 at 172.16.1.115:2887) Ended
>
> 2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1637 Close
> Channel sof
>
> ia/internal/sip:1007 at 172.16.1.115:2887 [CS_DESTROY]
>
> 2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1633 Session 3
> (sofia/
>
> internal/1009 at 172.16.1.115
> <javascript:_e(%7B%7D,'cvml','1009 at 172.16.1.115');>) Ended
>
> 2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1637 Close
> Channel sof
>
> ia/internal/1009 at 172.16.1.115
> <javascript:_e(%7B%7D,'cvml','1009 at 172.16.1.115');> [CS_DESTROY]
>
>
>
>
>
>
>
> Tony
>
>
>


-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
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From victor.medina at cibersys.com  Fri Jul  3 18:15:29 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Fri, 3 Jul 2015 09:45:29 -0430
Subject: [Freeswitch-users] Which for agent.pem? openssl or gentls_cert
In-Reply-To: <CALXCt0rS0azCjCzjvtGKHJzt_=WyKEJWp3uW=gevbhhjsByOSw@mail.gmail.com>
References: <CAHKqMb6A0U+JG4gS2iY88uSu1o5uhY44vuaHaFaOGEwhxK3UxA@mail.gmail.com>
	<CALXCt0rS0azCjCzjvtGKHJzt_=WyKEJWp3uW=gevbhhjsByOSw@mail.gmail.com>
Message-ID: <CACXsapMe3iJ8TEGAyWpFFa-rgKVVAN29QO-krarHjFx+-01z5w@mail.gmail.com>

Recent versions of freeswitch will create the certificates for you; on the
fly, but this is only for testing purposes only.

agent.pem(cert and key, in this order!) and cafile.pem are used for sip
wss.pem (one file, cert, key, cacert, in this order) is used form WSS.





2015-07-03 1:12 GMT-04:30 Giovanni Maruzzelli <gmaruzz at gmail.com>:

> Follow exactly, step by step, the instruction in confluence page
> "freeswitch 1.6".
> They work and are up to date.
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On Jul 3, 2015 3:48 AM, "Nandy Dagondon" <nandy1925 at gmail.com> wrote:
>
>> Hi to all,
>>
>> I'm testing mod_verto on FS v1.5.14 Debian Wheezy 64-bit .  In setting up
>> the certificates (self-signed), I followed the wiki. I encounter this WS
>> SETUP FAILED error even on Windows Chrome inspite of  all ports (443,
>> 5061,5081,8081, 8082 and 7443) are listening.  I'm suspecting a certificate
>> setup problem.
>>
>> In setting up the WSS certificates, it refers to WebRTC wiki,
>> specifically bkw's link:
>>
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/WebRTC#WebRTC-Aquickhowtofrombkw(BrianK.West):
>>
>> So far openssl is used to generate the certificates.
>>
>> This portion # Setup Sofia TLS the "agent.pem" is created and led me to
>> ...
>>
>> https://wiki.freeswitch.org/wiki/SIP_TLS#Configuration
>>
>> However, gentls_cert is used to create "agent.pem"! Is there a conflict
>> here? Which one to use?
>>
>> I appreciate for your guidance since I'm new to SSL/TLS.
>>
>> /Nandy
>>
>>
>>
>> _________________________________________________________________________
>> 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
>



-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From ricardas.stoma at gmail.com  Fri Jul  3 18:20:50 2015
From: ricardas.stoma at gmail.com (=?UTF-8?Q?Ri=C4=8Dardas_Stoma?=)
Date: Fri, 3 Jul 2015 17:20:50 +0300
Subject: [Freeswitch-users] Asterisk canreinvite (directmedia)
 equivalent in freeswitch
In-Reply-To: <CAEJMVkB0+PwiqbaRTOYEe6PK3y9uYXg4qs04=V8hgEELn7yzEA@mail.gmail.com>
References: <CAFX-kAcTdg2SJQO3Mui7wC84cxpmS0piGSfTV98NKQ=ir_X+8A@mail.gmail.com>
	<D1B950F7.A0FDA%krice@freeswitch.org>
	<CAFX-kAeFAnhks=hT8ULm86AD_He8LKMFsbOY7ObyY5TGuAapBA@mail.gmail.com>
	<CAEJMVkB0+PwiqbaRTOYEe6PK3y9uYXg4qs04=V8hgEELn7yzEA@mail.gmail.com>
Message-ID: <CAFX-kAfp7GT+Wukuzvu7d8jB8riiGCva42-L5wEqPq6HxPZM2Q@mail.gmail.com>

Yes, i see that bypass_media works only if end points have common codecs.
That's why i'm asking if there is a way to use bypass_media only when end
points have common codecs, otherwise freeswitch should handle transcoding.
Asterisk has single setting called 'canreinvite' which works this way.
Maybe freeswitch has something like this? Or does it require some fancy
dialplan/scripting to achieve this?

2015-07-03 17:05 GMT+03:00 Brian West <brian at freeswitch.org>:

> For bypass to work you'll have to have some codecs in common!  Without
> that it sounds like your end points don't behave properly in his case.
>
> /b
>
>
> On Friday, July 3, 2015, Ri?ardas Stoma <ricardas.stoma at gmail.com> wrote:
>
>> I think everything is ok with my endpoints. I use two softphones
>> (linphone and zoiper). On linphone (leg A) i enable only GSM codec and on
>> zoiper (leg B) i enable only PCMA codec. When bypass_media_after bridge is
>> disabled, media goes through freeswitch and freeswitch does transcoding. In
>> this case i can hear voices.
>>
>> When bypass_media_after_bridge is enabled, media goes from leg A to leg B
>> directly but the problem is that codecs do not match and i can hear only
>> digital noise. SIP trace looks like this http://i.imgur.com/iOn6ajJ.png
>>
>> If i set PCMA codec on linphone (leg A) then everything is ok with direct
>> media between softphones, codecs matches and i can hear voices. What i want
>> is IF leg A and leg B codecs matches then bypass_media should take effect
>> and media should go directly from leg A to leg B. IF codecs do not match,
>> then media should go through freeswitch and freeswitch should do
>> transcoding.
>>
>> Is this possible with freeswitch? Asterisk does this when canreinvite is
>> enabled ("If the clients use different codecs, Asterisk will not issue a
>> re-invite")
>>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>
>
> _________________________________________________________________________
> 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 Prashanth.Devarajappa at enghouse.com  Fri Jul  3 15:39:15 2015
From: Prashanth.Devarajappa at enghouse.com (Prashanth Devarajappa)
Date: Fri, 3 Jul 2015 11:39:15 +0000
Subject: [Freeswitch-users] Preferred Crypto for outbound calls
Message-ID: <159fe211186f415c89ee70143926d02d@UK-MAIL-001.edge.local>

Hello,

How do I set my preferred crypto suite to be used for SRTP outbound call ? I am running FS 1.4.18.
In old version of FS(1.2.3), I could do that using sip_secure_media=AES_CM_128_HMAC_SHA1_80. But this variable doesn't exist anymore, by the looks of it.

Regards
Prashanth
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From mike at jerris.com  Fri Jul  3 18:59:06 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 3 Jul 2015 10:59:06 -0400
Subject: [Freeswitch-users] Preferred Crypto for outbound calls
In-Reply-To: <159fe211186f415c89ee70143926d02d@UK-MAIL-001.edge.local>
References: <159fe211186f415c89ee70143926d02d@UK-MAIL-001.edge.local>
Message-ID: <FA888BBC-DD6F-4131-BBF8-BCC429A50623@jerris.com>

the variable changed in 1.4 to rtp_secure_media.  Also note we released 1.4.20 yesterday, I recommend using it as we have added quite a few bug fixes.

> On Jul 3, 2015, at 7:39 AM, Prashanth Devarajappa <Prashanth.Devarajappa at enghouse.com> wrote:
> 
> Hello,
>  
> How do I set my preferred crypto suite to be used for SRTP outbound call ? I am running FS 1.4.18.
> In old version of FS(1.2.3), I could do that using sip_secure_media=AES_CM_128_HMAC_SHA1_80. But this variable doesn?t exist anymore, by the looks of it.
>  
> Regards
> Prashanth
> _________________________________________________________________________
> 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 agoulis at opensips.org  Fri Jul  3 22:10:24 2015
From: agoulis at opensips.org (Alex Goulis)
Date: Fri, 03 Jul 2015 13:10:24 -0500
Subject: [Freeswitch-users] OpenSIPS Workshop @ClueCon August 3, 2015
Message-ID: <5596D010.30309@opensips.org>

It's almost that time again! The OpenSIPS team is coming to ClueCon 
2015! The OpenSIPS Workshop @ClueCon has been scheduled for *3rd of 
August 2015, at InterContinental Chicago, Chicago, IL, USA*.

The OpenSIPS Workshop is a new format that will focus more on providing 
attendees with more of a technical approach to integrating OpenSIPS 
rather than case studies - the sessions will provide working setups 
(with explanations) for OpenSIPS in different scenarios . The Workshops 
will cover new OpenSIPS release and features, FreeSwitch integration, 
Edge Proxy setup, Asynchronous I/O support and Fraud prevention.

The Official schedule is still under construction:
http://www.opensips.org/Community/Workshop-2015Chicago-Schedule

But we're happy to announce the following workshop leaders:

Bogdan-Andrei Iancu - Founder OpenSIPS project
Dan Christian Bogos - ITsysCOM
Vlad Paiu - OpenSIPS Project
Flavio Goncalves - SIPPulse
Alex Goulis - CID(name)
Pete Kelly - SourceVox

Whether you are a beginner or a seasoned professional, the OpenSIPS 
Workshop @ClueCon has something for you.

For beginners, get answers to questions like:

* how to extend, expand and enhance FreeSwitch with OpenSIPS
* why OpenSIPS is the right choice for your network
* what are the capabilities, features and limits of OpenSIPS

Seasoned professionals can:

* Learn about new modules and features for the 2.x release
* Discuss advanced topics
* Discuss the development roadmap for OpenSIPS


The registration fees (per person) are:

     $199 for those attending only the OpenSIPS Workshop
     $149 for those attending ClueCon as well or having a discount code 
(*CLUECON2015*)  or are registering 3 or more participants

Space is limited, so don't wait too long to register.

More information can be found at:
http://www.opensips.org/Community/Workshop-2015Chicago

We're looking forward to seeing you in Chicago!

- The OpenSIPS Team
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From william.king at quentustech.com  Sat Jul  4 04:42:02 2015
From: william.king at quentustech.com (William King)
Date: Fri, 3 Jul 2015 17:42:02 -0700
Subject: [Freeswitch-users] New module mod_smpp merged into FreeSWITCH
Message-ID: <55972BDA.7020806@quentustech.com>

Just merged it in today into the master branch. It's not in the FS
unstable debian repo yet(have to get libsmpp34 packaged first), but here
is a quick write up of how to use it:
https://quentustech.com/smpp-support-in-freeswitch.html

-- 
William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


From jungleboogie0 at gmail.com  Sat Jul  4 05:05:17 2015
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Fri, 3 Jul 2015 18:05:17 -0700
Subject: [Freeswitch-users] New module mod_smpp merged into FreeSWITCH
In-Reply-To: <55972BDA.7020806@quentustech.com>
References: <55972BDA.7020806@quentustech.com>
Message-ID: <CAKE2PDsRkhkTdWdX1NbQfs2mBuypohS+FnBEtuZbt1V3r66ftw@mail.gmail.com>

Hi William,
On 3 July 2015 at 17:42, William King <william.king at quentustech.com> wrote:
> Just merged it in today into the master branch. It's not in the FS
> unstable debian repo yet(have to get libsmpp34 packaged first), but here
> is a quick write up of how to use it:
> https://quentustech.com/smpp-support-in-freeswitch.html

I think this:
> ./bootstrap

Needs to be:
./bootstrap.sh





-- 
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si


From william.king at quentustech.com  Sat Jul  4 05:15:45 2015
From: william.king at quentustech.com (William King)
Date: Fri, 3 Jul 2015 18:15:45 -0700
Subject: [Freeswitch-users] New module mod_smpp merged into FreeSWITCH
In-Reply-To: <CAKE2PDsRkhkTdWdX1NbQfs2mBuypohS+FnBEtuZbt1V3r66ftw@mail.gmail.com>
References: <55972BDA.7020806@quentustech.com>
	<CAKE2PDsRkhkTdWdX1NbQfs2mBuypohS+FnBEtuZbt1V3r66ftw@mail.gmail.com>
Message-ID: <559733C1.5090104@quentustech.com>

fixed. thanks.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 7/3/15 6:05 PM, jungle Boogie wrote:
> Hi William,
> On 3 July 2015 at 17:42, William King <william.king at quentustech.com> wrote:
>> Just merged it in today into the master branch. It's not in the FS
>> unstable debian repo yet(have to get libsmpp34 packaged first), but here
>> is a quick write up of how to use it:
>> https://quentustech.com/smpp-support-in-freeswitch.html
> 
> I think this:
>> ./bootstrap
> 
> Needs to be:
> ./bootstrap.sh
> 
> 
> 
> 
> 


From hi-tecc at hotmail.com  Sat Jul  4 05:59:17 2015
From: hi-tecc at hotmail.com (DP .)
Date: Fri, 3 Jul 2015 21:59:17 -0400
Subject: [Freeswitch-users] New module mod_smpp merged into FreeSWITCH
In-Reply-To: <559733C1.5090104@quentustech.com>
References: <55972BDA.7020806@quentustech.com>,
	<CAKE2PDsRkhkTdWdX1NbQfs2mBuypohS+FnBEtuZbt1V3r66ftw@mail.gmail.com>,
	<559733C1.5090104@quentustech.com>
Message-ID: <BLU184-W627A7F81C852A4627EF807F2950@phx.gbl>

This is great! Will definitely try this mod out. Thanks

> To: freeswitch-users at lists.freeswitch.org
> From: william.king at quentustech.com
> Date: Fri, 3 Jul 2015 18:15:45 -0700
> Subject: Re: [Freeswitch-users] New module mod_smpp merged into FreeSWITCH
> 
> fixed. thanks.
> 
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
> 
> On 7/3/15 6:05 PM, jungle Boogie wrote:
> > Hi William,
> > On 3 July 2015 at 17:42, William King <william.king at quentustech.com> wrote:
> >> Just merged it in today into the master branch. It's not in the FS
> >> unstable debian repo yet(have to get libsmpp34 packaged first), but here
> >> is a quick write up of how to use it:
> >> https://quentustech.com/smpp-support-in-freeswitch.html
> > 
> > I think this:
> >> ./bootstrap
> > 
> > Needs to be:
> > ./bootstrap.sh
> > 
> > 
> > 
> > 
> > 
> 
> _________________________________________________________________________
> 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 steveu at coppice.org  Sat Jul  4 07:12:50 2015
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 04 Jul 2015 11:12:50 +0800
Subject: [Freeswitch-users] New module mod_smpp merged into FreeSWITCH
In-Reply-To: <55972BDA.7020806@quentustech.com>
References: <55972BDA.7020806@quentustech.com>
Message-ID: <55974F32.8080403@coppice.org>

On 07/04/2015 08:42 AM, William King wrote:
> Just merged it in today into the master branch. It's not in the FS
> unstable debian repo yet(have to get libsmpp34 packaged first), but here
> is a quick write up of how to use it:
> https://quentustech.com/smpp-support-in-freeswitch.html
>
It will be interesting to see if that gets much use. I've been offering 
to provide SMPP support, based on old code of mine, for years. I've 
never had much reaction.

Regards,
Steve



From mitch.capper at gmail.com  Sat Jul  4 08:50:28 2015
From: mitch.capper at gmail.com (Mitch Capper)
Date: Fri, 3 Jul 2015 21:50:28 -0700
Subject: [Freeswitch-users] Quality VOIP provider with SMS support
Message-ID: <CALn233p1xn8Ghu+wOcTw5=AGK2ettDHY-PMssjX87DbS1sF=CQ@mail.gmail.com>

After using several providers from vitelity to flowroute and callwithus, I
figured it was time to move on from vitelity given some very bad SMS
support as of late.

Looked at Nexmo, voxbone, and a few others (noticed nexmo mentioned in the
new mod_snmpp thread).   I figured I would see what people suggest.

As a bonus right now do a lot of faxing with FS and flowroute and getting
some great results for both incoming and outgoing.      While happy to keep
using many providers for different things if able to converge on one all
the better:)

~Mitch
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From saghar.ayyaz at yahoo.com  Sat Jul  4 13:10:12 2015
From: saghar.ayyaz at yahoo.com (Saghar Ayyaz)
Date: Sat, 4 Jul 2015 09:10:12 +0000 (UTC)
Subject: [Freeswitch-users] conference_set_auto_outcall not working with
	proxy server
Message-ID: <366805815.2377706.1436001012471.JavaMail.yahoo@mail.yahoo.com>

conference_set_auto_outcall not working with proxy server. I am calling application from lua with following code. Suppose 167 is sip phone no.session:execute("conference_set_auto_outcall", "sofia/gateway/Kamailio/167")
Gateway is defined in external.xml<gateway name="Kamailio">
         <param name="extension" value="conference"/>
         <param name="expire-seconds" value="125"/>
         <param name="register-transport" value="udp"/>
         <param name="proxy" value="xyz.com"/>
         <param name="register" value="false"/>
         <param name="caller-id-in-from" value="true"/> 
         <param name="username" value="not-used"/>
         <param name="password" value="not-used"/>
    </gateway>
Error:2015-07-03 16:06:50.193203 [NOTICE] switch_core_session.c:1633 Session 35 (sofia/external/167) Ended 2015-07-03 16:06:50.193203 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/167 [CS_DESTROY] 2015-07-03 16:06:50.193203 [DEBUG] switch_core_state_machine.c:626 (sofia/external/167) Running State Change CS_DESTROY 2015-07-03 16:06:50.193203 [DEBUG] switch_core_state_machine.c:636 (sofia/external/167) State DESTROY 2015-07-03 16:06:50.193203 [DEBUG] mod_sofia.c:323 sofia/external/167 SOFIA DESTROY 2015-07-03 16:06:50.193203 [DEBUG] switch_core_state_machine.c:111 sofia/external/167 Standard DESTROY 2015-07-03 16:06:50.193203 [DEBUG] switch_core_state_machine.c:636 (sofia/external/167) State DESTROY going to sleep?2015-07-03 16:06:50.193203 [ERR] mod_conference.c:8469 Cannot create outgoing channel, cause: NORMAL_TEMPORARY_FAILURE?2015-07-03 16:06:50.193203 [DEBUG] switch_ivr_originate.c:2079 Parsing global variables
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From brian at freeswitch.org  Sat Jul  4 19:34:30 2015
From: brian at freeswitch.org (Brian West)
Date: Sat, 4 Jul 2015 10:34:30 -0500
Subject: [Freeswitch-users] New module mod_smpp merged into FreeSWITCH
In-Reply-To: <55974F32.8080403@coppice.org>
References: <55972BDA.7020806@quentustech.com> <55974F32.8080403@coppice.org>
Message-ID: <CAEJMVkB7G6G4Oad75av3pEAMsrUd2q==jG1T4Kkpm7v4oYnDqQ@mail.gmail.com>

Steve, what does your code for this look like?  I suspect yours may be
easier to use.

On Fri, Jul 3, 2015 at 10:12 PM, Steve Underwood <steveu at coppice.org> wrote:

> On 07/04/2015 08:42 AM, William King wrote:
> > Just merged it in today into the master branch. It's not in the FS
> > unstable debian repo yet(have to get libsmpp34 packaged first), but here
> > is a quick write up of how to use it:
> > https://quentustech.com/smpp-support-in-freeswitch.html
> >
> It will be interesting to see if that gets much use. I've been offering
> to provide SMPP support, based on old code of mine, for years. I've
> never had much reaction.
>
> Regards,
> Steve
>
>
> _________________________________________________________________________
> 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
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From gliu at tecomtech.com  Mon Jul  6 05:06:47 2015
From: gliu at tecomtech.com (Bryan)
Date: Mon, 6 Jul 2015 09:06:47 +0800
Subject: [Freeswitch-users] =?utf-8?b?562U5aSNOiAgTm8gVmlkZW8gaW4gTWVkaWEg?=
	=?utf-8?q?Proxy_mode?=
In-Reply-To: <CAEJMVkBTD7dfj31cuuU6DzO0dGim0FqkgpjwYZmq713-=PXTSw@mail.gmail.com>
References: <003f01d0b55b$1f25b280$5d711780$@com>
	<CAEJMVkBTD7dfj31cuuU6DzO0dGim0FqkgpjwYZmq713-=PXTSw@mail.gmail.com>
Message-ID: <007c01d0b788$24441020$6ccc3060$@com>

HI, Actually we tested FS 1.4.18 , FS 1.4.19 and the latest Windows MSI installation package, all have the same issue.

 

The reason why we test Proxy mode is to see the performance of FS than Media relay (with Transcoding).

 

Thanks.

 

 

???: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] ?? Brian West
????: 2015?7?3? 22:06
???: FreeSWITCH Users Help
??: Re: [Freeswitch-users] No Video in Media Proxy mode

 

What rev of Freeswitch, and why are you using proxy media?

On Friday, July 3, 2015, Bryan <gliu at tecomtech.com> wrote:

Hi, I?m testing FreeSWITCH with Android Linphone. I found the Video stream was not forwarded by FS. Audio is OK.

 

Any Ideas? Or this is a bug?

 

The test envrionmet as following:

Android Linphone < -- > Wifi Router < -- > FS < -- >PC Linphone.

 

The FS Version is the Latest Windows MSI package. 

 

2015-07-03 14:31:57.718750 [NOTICE] switch_channel.c:1055 New Channel sofia/inte

rnal/1009 at 172.16.1.115 <javascript:_e(%7B%7D,'cvml','1009 at 172.16.1.115');>  [c8a19d44-3084-4539-b817-3276fd7ec43d]

2015-07-03 14:31:57.750000 [INFO] mod_dialplan_xml.c:635 Processing 1009 <1009>-

>1007 in context default

2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *1 execut

e_extension::dx XML features

2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *2 record

_session::C:/Program Files/FreeSWITCH/recordings/1009.2015-07-03-14-31-57.wav

2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *3 execut

e_extension::cf XML features

2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *4 execut

e_extension::att_xfer XML features

2015-07-03 14:31:57.781250 [NOTICE] switch_channel.c:1055 New Channel sofia/inte

rnal/sip:1007 at 172.16.1.115:2887 [4cfdf0e6-7010-467b-97b3-20e4dc295422]

2015-07-03 14:31:57.843750 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:1

007 at 172.16.1.115:2887!

2015-07-03 14:31:57.843750 [NOTICE] mod_sofia.c:2107 Ring-Ready sofia/internal/1

009 at 172.16.1.115 <javascript:_e(%7B%7D,'cvml','009 at 172.16.1.115');> !

2015-07-03 14:31:57.859375 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/

internal/1009 at 172.16.1.115 <javascript:_e(%7B%7D,'cvml','1009 at 172.16.1.115');> !

2015-07-03 14:32:00.312500 [NOTICE] sofia.c:7437 Channel [sofia/internal/sip:100

7 at 172.16.1.115:2887] has been answered

2015-07-03 14:32:00.312500 [NOTICE] switch_core_media.c:4405 sofia/internal/sip:

1007 at 172.16.1.115:2887 Starting Video thread

2015-07-03 14:32:00.343750 [NOTICE] switch_core_media.c:4405 sofia/internal/1009

@172.16.1.115 Starting Video thread

2015-07-03 14:32:00.343750 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/i

nternal/1009 at 172.16.1.115 <javascript:_e(%7B%7D,'cvml','1009 at 172.16.1.115');> ] has been answered

2015-07-03 14:32:00.843750 [INFO] switch_rtp.c:5832 Auto Changing port from 61.1

83.139.155:64131 to 172.16.15.111:7078

2015-07-03 14:32:23.750000 [NOTICE] sofia.c:952 Hangup sofia/internal/sip:1007 at 1

72.16.1.115:2887 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]

2015-07-03 14:32:23.750000 [NOTICE] switch_ivr_bridge.c:1608 Hangup sofia/intern

al/1009 at 172.16.1.115 <javascript:_e(%7B%7D,'cvml','1009 at 172.16.1.115');>  [CS_EXECUTE] [NORMAL_CLEARING]

2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1633 Session 4 (sofia/

internal/sip:1007 at 172.16.1.115:2887) Ended

2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1637 Close Channel sof

ia/internal/sip:1007 at 172.16.1.115:2887 [CS_DESTROY]

2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1633 Session 3 (sofia/

internal/1009 at 172.16.1.115 <javascript:_e(%7B%7D,'cvml','1009 at 172.16.1.115');> ) Ended

2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1637 Close Channel sof

ia/internal/1009 at 172.16.1.115 <javascript:_e(%7B%7D,'cvml','1009 at 172.16.1.115');>  [CS_DESTROY]

 

 

 

Tony

 



-- 

Brian West
brian at freeswitch.org

  <http://billing.freeswitch.org/templates/default/img/whmcslogo.png> 

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for <https://www.cluecon.com/call-for-speakers/>  Speakers | Register <https://freeswitch.com/cart.php?gid=1>  TODAY! | 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

 

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From paulcuttler at gmail.com  Mon Jul  6 08:44:57 2015
From: paulcuttler at gmail.com (Paul Cuttler)
Date: Mon, 6 Jul 2015 14:44:57 +1000
Subject: [Freeswitch-users] mod_rtmp onStatus messages
Message-ID: <CAH3KVWoHQELDCv5UeuqOgePFd0E2qkn+LRjNDvO7Fsm4RyBZ6w@mail.gmail.com>

I was wondering about the onStatus messages that are sent by mod_rtmp to
Flash Player. The RTMP Spec (see page 38 of
http://wwwimages.adobe.com/content/dam/Adobe/en/devnet/rtmp/pdf/rtmp_specification_1.0.pdf)
says that the transaction ID for onStatus messages is supposed to be 0, but
mod_rtmp uses 1. It seems to work fine, but I wanted to check if anyone
knew if it might lead to problems not by following the spec.

This is in rtmp_sig.c, for example in rtmp_i_play.

regards,

Paul
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From emamirazavi at gmail.com  Mon Jul  6 10:18:25 2015
From: emamirazavi at gmail.com (S.Mohammad Emami Razavi)
Date: Mon, 6 Jul 2015 10:48:25 +0430
Subject: [Freeswitch-users] Fwd: mod_shout and mod_sndfile are not loaded
In-Reply-To: <CAJEstwNYPi-JPkNG4_F12vz0BnLaNvYrX_HYGDnYqSiTGkHaVw@mail.gmail.com>
References: <CAJEstwNYPi-JPkNG4_F12vz0BnLaNvYrX_HYGDnYqSiTGkHaVw@mail.gmail.com>
Message-ID: <CAJEstwPDABeLdx_VKRpBnDUW4mFfuVYzGs-L97AjEESOyidS-A@mail.gmail.com>

I? have problem with loading this two important modules in version
FreeSWITCH Version 1.7.0+git~20150626T195140Z~2a1195e55d~64bit
I have installed them:
[root at voip freeswitch]# make mod_sndfile-install
make[1]: Entering directory `/usr/local/src/freeswitch'
 /usr/bin/mkdir -p '/usr/local/freeswitch/lib'
 /bin/sh /usr/local/src/freeswitch/libtool   --mode=install
/usr/bin/install -c   libfreeswitch.la '/usr/local/freeswitch/lib'
libtool: install: /usr/bin/install -c .libs/libfreeswitch.so.1.0.0
/usr/local/freeswitch/lib/libfreeswitch.so.1.0.0
libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f
libfreeswitch.so.1.0.0 libfreeswitch.so.1 || { rm -f libfreeswitch.so.1 &&
ln -s libfreeswitch.so.1.0.0 libfreeswitch.so.1; }; })
libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f
libfreeswitch.so.1.0.0 libfreeswitch.so || { rm -f libfreeswitch.so && ln
-s libfreeswitch.so.1.0.0 libfreeswitch.so; }; })
libtool: install: /usr/bin/install -c .libs/libfreeswitch.lai
/usr/local/freeswitch/lib/libfreeswitch.la
libtool: install: /usr/bin/install -c .libs/libfreeswitch.a
/usr/local/freeswitch/lib/libfreeswitch.a
libtool: install: chmod 644 /usr/local/freeswitch/lib/libfreeswitch.a
libtool: install: ranlib /usr/local/freeswitch/lib/libfreeswitch.a
libtool: finish:
PATH="/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin:/sbin"
ldconfig -n /usr/local/freeswitch/lib
----------------------------------------------------------------------
Libraries have been installed in:
   /usr/local/freeswitch/lib

If you ever happen to want to link against installed libraries
in a given directory, LIBDIR, you must either use libtool, and
specify the full pathname of the library, or use the `-LLIBDIR'
flag during linking and do at least one of the following:
   - add LIBDIR to the `LD_LIBRARY_PATH' environment variable
     during execution
   - add LIBDIR to the `LD_RUN_PATH' environment variable
     during linking
   - use the `-Wl,-rpath -Wl,LIBDIR' linker flag
   - have your system administrator add LIBDIR to `/etc/ld.so.conf'

See any operating system documentation about shared libraries for
more information, such as the ld(1) and ld.so(8) manual pages.
----------------------------------------------------------------------
make[1]: Leaving directory `/usr/local/src/freeswitch'
make[1]: Entering directory `/usr/local/src/freeswitch/src/mod'

making install mod_sndfile
make[2]: Entering directory
`/usr/local/src/freeswitch/src/mod/formats/mod_sndfile'
make[2]: Leaving directory
`/usr/local/src/freeswitch/src/mod/formats/mod_sndfile'
make[1]: Leaving directory `/usr/local/src/freeswitch/src/mod'




and i configure:
[root at voip freeswitch]# vim
/usr/local/freeswitch/conf/autoload_configs/modules.conf.xml

and i have uncommented <load module="mod_sndfile"/>

but in fs_cli when i call module_exists it prints false or calling load
mod_sndfile results:
freeswitch at internal> load mod_sndfile
+OK Reloading XML
-ERR [module load file routine returned an error]

2015-07-05 17:30:11.737311 [INFO] mod_enum.c:880 ENUM Reloaded
2015-07-05 17:30:11.737311 [CRIT] switch_loadable_module.c:1520 Error
Loading module /usr/local/freeswitch/mod/mod_sndfile.so
**/usr/local/freeswitch/mod/mod_sndfile.so: undefined symbol:
sf_writef_int**
2015-07-05 17:30:11.737311 [INFO] switch_time.c:1411 Timezone reloaded 1781
definitions
freeswitch at internal>


My OS is CentOS Linux release 7.1.1503 (Core)

Any view?
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From vishal.sharma at knowlarity.com  Mon Jul  6 10:40:47 2015
From: vishal.sharma at knowlarity.com (Vishal Sharma)
Date: Mon, 6 Jul 2015 12:10:47 +0530
Subject: [Freeswitch-users] No Bridge Media
Message-ID: <CAEaAqfj3DJ384pRxkc3h82ik-0a23TQA76E75AUOQFNVkKLP+g@mail.gmail.com>

Hi,
When I use following commnd on fs_cli, I can hear ring sound or any message
telecome provider play on destination number

originate freetdm/outgoing/r/09899790092
&bridge(freetdm/outgoing/r/09899488723)

but when I use one of following commands

freeswitch at internal> originate
{bridge_early_media=true}freetdm/outgoing/r/09899790092 &bridge
(freetdm/outgoing/r/09899488723)


freeswitch at internal> originate
{bridge_early_media=true}freetdm/outgoing/r/09899790092
&bridge({bridge_early_media=true}freetdm/outgoing/r/09899488723)

or when I bridge a call which is received on FS, I don't get bridge media.

Is there any perticular config required, or it's bug..


Regards,
Vishal Sharma

-- 
SuperReceptionist is now available on Android mobiles. Track your business 
on the go with call analytics, recordings, insights and more: Download the 
app here 
<https://play.google.com/store/apps/details?id=com.superreceptionist>
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From zvi at lexifone.com  Mon Jul  6 12:03:51 2015
From: zvi at lexifone.com (Zvi Agmon)
Date: Mon, 6 Jul 2015 11:03:51 +0300
Subject: [Freeswitch-users] Check session ready after session transfer
In-Reply-To: <CAKYJ0qYsJDnHyYVtfU_bX52pYFcZLn9RV+vydGZGxW=S=dUzVw@mail.gmail.com>
References: <CAKYJ0qYsJDnHyYVtfU_bX52pYFcZLn9RV+vydGZGxW=S=dUzVw@mail.gmail.com>
Message-ID: <CAKYJ0qZ0O_V=vu6LCHbih4AdKb0EQgvef2+-ZEByZX6hMEVRPQ@mail.gmail.com>

Hi,

Can anyone help with this please?
Didn't get any response - maybe the question is not clear enough...

I'm trying to figure out the correct way to check for session state after
it was transferred to another dial plan - meaning - a lua script is called
from the new dial plan and in it I want to know if the session is in ready
state.

Thanks a lot

Zvi Agmon
Lexifone <http://www.lexifone.com>
zvi at lexifone.com


Best regards

Zvi Agmon
Lexifone <http://www.lexifone.com/>
email: zvi at lexifone.com
Office: +972-4-6817711
Cell: +972-54-4505109

On Tue, Jun 30, 2015 at 2:07 PM, Zvi Agmon <zvi at lexifone.com> wrote:

> Hello,
>
> Need some help regarding session state validation.
>
> In documentation I see this:
>
> session:ready
>
> - checks whether the session is still active (true anytime between call
> starts and hangup)
>
> - also session:ready will return false if the call is being transferred.
> Bottom line is you should always be checking session:ready on any loops and
> periodically throughout your script and exit asap if it returns false.
>
>
> My experience is that after transfer the return value of session:ready()
> is is not consistent - what is the correct way to check session state in
> that case?
>
> Thanks
>
> Zvi Agmon
> Lexifone <http://www.lexifone.com/>
> email: zvi at lexifone.com
> Office: +972-4-6817711
> Cell: +972-54-4505109
>
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From lakindia89 at gmail.com  Mon Jul  6 12:20:08 2015
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 6 Jul 2015 13:50:08 +0530
Subject: [Freeswitch-users] No Bridge Media
In-Reply-To: <CAEaAqfj3DJ384pRxkc3h82ik-0a23TQA76E75AUOQFNVkKLP+g@mail.gmail.com>
References: <CAEaAqfj3DJ384pRxkc3h82ik-0a23TQA76E75AUOQFNVkKLP+g@mail.gmail.com>
Message-ID: <CAMx6C7cajdkOmX7xoe66pq+dZ_ge2TVRUbG2o5Yq=5751dUAtA@mail.gmail.com>

Try using ignore_early_media=false and bridge_early_media=true.

Try the below

freeswitch at internal> originate
{ignore_early_media=true}freetdm/outgoing/r/09899790092 &bridge
({ignore_early_media=false,bridge_early_media=true}freetdm/outgoing/r/09899488723)
On 06-Jul-2015 12:11, "Vishal Sharma" <vishal.sharma at knowlarity.com> wrote:

> Hi,
> When I use following commnd on fs_cli, I can hear ring sound or any
> message telecome provider play on destination number
>
> originate freetdm/outgoing/r/09899790092
> &bridge(freetdm/outgoing/r/09899488723)
>
> but when I use one of following commands
>
> freeswitch at internal> originate
> {bridge_early_media=true}freetdm/outgoing/r/09899790092 &bridge
> (freetdm/outgoing/r/09899488723)
>
>
> freeswitch at internal> originate
> {bridge_early_media=true}freetdm/outgoing/r/09899790092
> &bridge({bridge_early_media=true}freetdm/outgoing/r/09899488723)
>
> or when I bridge a call which is received on FS, I don't get bridge media.
>
> Is there any perticular config required, or it's bug..
>
>
> Regards,
> Vishal Sharma
>
> SuperReceptionist is now available on Android mobiles. Track your business
> on the go with call analytics, recordings, insights and more: Download
> the app here
> <https://play.google.com/store/apps/details?id=com.superreceptionist>
> _________________________________________________________________________
> 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 clive18 at webmail.co.za  Mon Jul  6 15:31:23 2015
From: clive18 at webmail.co.za (clive engelberg)
Date: Mon, 06 Jul 2015 13:31:23 +0200
Subject: [Freeswitch-users] Disk error / database error help
In-Reply-To: <CAFiqYukVtSCCYndxj5pZwTTLXVQEG0fm_edxbZLjvFjQ4tjn8g@mail.gmail.com>
References: <c357215130c3dee62c7803ab7ef5c42b@www.webmail.co.za>
	<CAFiqYukdk08m2EoNVZ75xot+Fr=EJDt8y1mFqo8_i-xuvxXDuA@mail.gmail.com>
	<ed6289f3b6f594234ea0e9907c73f721@www.webmail.co.za>
	<CALXCt0ocnA7QJKbJ0gwpsJHkeKHTO9L6KjD+LPxcHLRsiEL5NA@mail.gmail.com>,
	<CAFiqYukVtSCCYndxj5pZwTTLXVQEG0fm_edxbZLjvFjQ4tjn8g@mail.gmail.com>
Message-ID: <3a66ccd35b014370dc9d2604fbf9547c@www.webmail.co.za>

Thanks for all your help.
I suspect its a failing disk as well.... was hoping it wasnt, but anyway.

Time to go hardware shopping :(

Thanks again.
regards
Clive
On Thu, 2 Jul 2015 19:24:29 +0100 Steven Ayre  wrote

  In some cases that might not help, since tmpfs can swap to disk if it needs
the space.

 On 2 July 2015 at 16:00, Giovanni Maruzzelli  wrote:
    you probably have a failing disk...

 btw, is better to mount freeswitch db in a tmpfs ram dilesystem

 On Thu, Jul 2, 2015 at 4:50 PM, clive engelberg  wrote:
 Hi

My dmesg seems to be full of IPtables "denied" statements.

I guess I need to reboot to get a clean dmesg.

Thanks and regards
Clive

On Thu, 2 Jul 2015 13:18:03 +0100 Steven Ayre  wrote

  Do you see any errors in the output of 'dmesg'? If there are errors on your
hard disk or controller you'd see messages in there.

 On 2 July 2015 at 10:21, clive engelberg  wrote:
  Hi Guys.

 I have had this happen a few times already, where Freeswitch seems to have a
 database error. This starts off slowly, and within a short time, it
completely
 fills the disk drive with log error files.

 the log files look like this:
 2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR
[disk
 I/O error]
 BEGIN EXCLUSIVE
 2015-07-01 15:19:35.470881 [CRIT] switch_core_sqldb.c:1565 ERROR [disk I/O
 error]
 2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR
 [cannot commit - no transaction is active]

 Any ideas what is causing this.
 I suspect maybe a hardware disk error.. not sure.

 Thanks in advance
 Clive

 ____________________________________________________________
 South Africas premier fr ee email service - www.webmail.co.za

 _________________________________________________________________________
 Professional FreeSWITCH Consulting Services:
 consulting at freeswitch.org [5]
 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 [10]
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [11]
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[12]
 http://www.freeswitch.org [13]    

-------------------------
South Africa premier free email service - webmail.co.za [14] 
_________________________________________________________________________
 Professional FreeSWITCH Consulting Services:
 consulting at freeswitch.org [15]
 http://www.freeswitchsolutions.com [16]

 Official FreeSWITCH Sites
 http://www.freeswitch.org [17]
 http://confluence.freeswitch.org [18]
 http://www.cluecon.com [19]

 FreeSWITCH-users mailing list
 FreeSWITCH-users at lists.freeswitch.org [20]
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [21]
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[22]
 http://www.freeswitch.org [23]  

  -- 
 Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618  
_________________________________________________________________________
 Professional FreeSWITCH Consulting Services:
 consulting at freeswitch.org [24]
 http://www.freeswitchsolutions.com [25]

 Official FreeSWITCH Sites
 http://www.freeswitch.org [26]
 http://confluence.freeswitch.org [27]
 http://www.cluecon.com [28]

 FreeSWITCH-users mailing list
 FreeSWITCH-users at lists.freeswitch.org [29]
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [30]
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[31]
 http://www.freeswitch.org [32]   

Links:
------
[1] mailto:gmaruzz at gmail.com
[2] mailto:clive18 at webmail.co.za
[3] mailto:steveayre at gmail.com
[4] mailto:clive18 at webmail.co.za
[5] mailto:consulting at freeswitch.org
[6] http://www.freeswitchsolutions.com
[7] http://www.freeswitch.org
[8] http://confluence.freeswitch.org
[9] http://www.cluecon.com
[10] mailto:FreeSWITCH-users at lists.freeswitch.org
[11] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[12] http://lists.freeswitch.org/mailman/options/%20freeswitch-users
[13] http://www.freeswitch.org
[14] http://www.webmail.co.za/
[15] mailto:consulting at freeswitch.org
[16] http://www.freeswitchsolutions.com
[17] http://www.freeswitch.org
[18] http://confluence.freeswitch.org
[19] http://www.cluecon.com
[20] mailto:FreeSWITCH-users at lists.freeswitch.org
[21] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[22] http://lists.freeswitch.org/mailman/options/freeswitch-users
[23] http://www.freeswitch.org
[24] mailto:consulting at freeswitch.org
[25] http://www.freeswitchsolutions.com
[26] http://www.freeswitch.org
[27] http://confluence.freeswitch.org
[28] http://www.cluecon.com
[29] mailto:FreeSWITCH-users at lists.freeswitch.org
[30] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[31] http://lists.freeswitch.org/mailman/options/freeswitch-users
[32] http://www.freeswitch.org


____________________________________________________________
South Africas premier free email service - www.webmail.co.za
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From aronp at guaranteedplus.com  Mon Jul  6 18:30:45 2015
From: aronp at guaranteedplus.com (Podrigal, Aron)
Date: Mon, 6 Jul 2015 10:30:45 -0400
Subject: [Freeswitch-users] Fwd: mod_shout and mod_sndfile are not loaded
In-Reply-To: <CAJEstwPDABeLdx_VKRpBnDUW4mFfuVYzGs-L97AjEESOyidS-A@mail.gmail.com>
References: <CAJEstwNYPi-JPkNG4_F12vz0BnLaNvYrX_HYGDnYqSiTGkHaVw@mail.gmail.com>
	<CAJEstwPDABeLdx_VKRpBnDUW4mFfuVYzGs-L97AjEESOyidS-A@mail.gmail.com>
Message-ID: <CANJp-yggtQyQCa4daam_2VGPVps=y00XxhELijW8wTxPC1PX5Q@mail.gmail.com>

I guess you get nothing when you run this command

nm -D `locate libsndfile.so | head -n 1` | grep sf_writef_int

Try installing libsndfile from http://www.mega-nerd.com/libsndfile/#Download

On Mon, Jul 6, 2015 at 2:18 AM, S.Mohammad Emami Razavi <
emamirazavi at gmail.com> wrote:

> I? have problem with loading this two important modules in version
> FreeSWITCH Version 1.7.0+git~20150626T195140Z~2a1195e55d~64bit
> I have installed them:
> [root at voip freeswitch]# make mod_sndfile-install
> make[1]: Entering directory `/usr/local/src/freeswitch'
>  /usr/bin/mkdir -p '/usr/local/freeswitch/lib'
>  /bin/sh /usr/local/src/freeswitch/libtool   --mode=install
> /usr/bin/install -c   libfreeswitch.la '/usr/local/freeswitch/lib'
> libtool: install: /usr/bin/install -c .libs/libfreeswitch.so.1.0.0
> /usr/local/freeswitch/lib/libfreeswitch.so.1.0.0
> libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f
> libfreeswitch.so.1.0.0 libfreeswitch.so.1 || { rm -f libfreeswitch.so.1 &&
> ln -s libfreeswitch.so.1.0.0 libfreeswitch.so.1; }; })
> libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f
> libfreeswitch.so.1.0.0 libfreeswitch.so || { rm -f libfreeswitch.so && ln
> -s libfreeswitch.so.1.0.0 libfreeswitch.so; }; })
> libtool: install: /usr/bin/install -c .libs/libfreeswitch.lai
> /usr/local/freeswitch/lib/libfreeswitch.la
> libtool: install: /usr/bin/install -c .libs/libfreeswitch.a
> /usr/local/freeswitch/lib/libfreeswitch.a
> libtool: install: chmod 644 /usr/local/freeswitch/lib/libfreeswitch.a
> libtool: install: ranlib /usr/local/freeswitch/lib/libfreeswitch.a
> libtool: finish:
> PATH="/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin:/sbin"
> ldconfig -n /usr/local/freeswitch/lib
> ----------------------------------------------------------------------
> Libraries have been installed in:
>    /usr/local/freeswitch/lib
>
> If you ever happen to want to link against installed libraries
> in a given directory, LIBDIR, you must either use libtool, and
> specify the full pathname of the library, or use the `-LLIBDIR'
> flag during linking and do at least one of the following:
>    - add LIBDIR to the `LD_LIBRARY_PATH' environment variable
>      during execution
>    - add LIBDIR to the `LD_RUN_PATH' environment variable
>      during linking
>    - use the `-Wl,-rpath -Wl,LIBDIR' linker flag
>    - have your system administrator add LIBDIR to `/etc/ld.so.conf'
>
> See any operating system documentation about shared libraries for
> more information, such as the ld(1) and ld.so(8) manual pages.
> ----------------------------------------------------------------------
> make[1]: Leaving directory `/usr/local/src/freeswitch'
> make[1]: Entering directory `/usr/local/src/freeswitch/src/mod'
>
> making install mod_sndfile
> make[2]: Entering directory
> `/usr/local/src/freeswitch/src/mod/formats/mod_sndfile'
> make[2]: Leaving directory
> `/usr/local/src/freeswitch/src/mod/formats/mod_sndfile'
> make[1]: Leaving directory `/usr/local/src/freeswitch/src/mod'
>
>
>
>
> and i configure:
> [root at voip freeswitch]# vim
> /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml
>
> and i have uncommented <load module="mod_sndfile"/>
>
> but in fs_cli when i call module_exists it prints false or calling load
> mod_sndfile results:
> freeswitch at internal> load mod_sndfile
> +OK Reloading XML
> -ERR [module load file routine returned an error]
>
> 2015-07-05 17:30:11.737311 [INFO] mod_enum.c:880 ENUM Reloaded
> 2015-07-05 17:30:11.737311 [CRIT] switch_loadable_module.c:1520 Error
> Loading module /usr/local/freeswitch/mod/mod_sndfile.so
> **/usr/local/freeswitch/mod/mod_sndfile.so: undefined symbol:
> sf_writef_int**
> 2015-07-05 17:30:11.737311 [INFO] switch_time.c:1411 Timezone reloaded
> 1781 definitions
> freeswitch at internal>
>
>
> My OS is CentOS Linux release 7.1.1503 (Core)
>
> Any view?
>
>
> _________________________________________________________________________
> 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
>



-- 
Aron Podrigal
-
//Be happy :-)
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From nandy1925 at gmail.com  Mon Jul  6 18:54:18 2015
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Mon, 6 Jul 2015 22:54:18 +0800
Subject: [Freeswitch-users] Which for agent.pem? openssl or gentls_cert
In-Reply-To: <CACXsapMe3iJ8TEGAyWpFFa-rgKVVAN29QO-krarHjFx+-01z5w@mail.gmail.com>
References: <CAHKqMb6A0U+JG4gS2iY88uSu1o5uhY44vuaHaFaOGEwhxK3UxA@mail.gmail.com>
	<CALXCt0rS0azCjCzjvtGKHJzt_=WyKEJWp3uW=gevbhhjsByOSw@mail.gmail.com>
	<CACXsapMe3iJ8TEGAyWpFFa-rgKVVAN29QO-krarHjFx+-01z5w@mail.gmail.com>
Message-ID: <CAHKqMb4pavpKOuWH+PjkhoRyKCQeVYXGWfKnMXG2WLyWAcqiHw@mail.gmail.com>

Thanks for your inputs.  Somehow I managed to make it work in my intranet
setup.  There are some modifications in the certificate
generation/installation script.  I'll test it when I'll install for public
access.

/Nandy

On Fri, Jul 3, 2015 at 10:15 PM, Victor Medina <victor.medina at cibersys.com>
wrote:

> Recent versions of freeswitch will create the certificates for you; on the
> fly, but this is only for testing purposes only.
>
> agent.pem(cert and key, in this order!) and cafile.pem are used for sip
> wss.pem (one file, cert, key, cacert, in this order) is used form WSS.
>
>
>
>
>
> 2015-07-03 1:12 GMT-04:30 Giovanni Maruzzelli <gmaruzz at gmail.com>:
>
>> Follow exactly, step by step, the instruction in confluence page
>> "freeswitch 1.6".
>> They work and are up to date.
>>
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> On Jul 3, 2015 3:48 AM, "Nandy Dagondon" <nandy1925 at gmail.com> wrote:
>>
>>> Hi to all,
>>>
>>> I'm testing mod_verto on FS v1.5.14 Debian Wheezy 64-bit .  In setting
>>> up the certificates (self-signed), I followed the wiki. I encounter this WS
>>> SETUP FAILED error even on Windows Chrome inspite of  all ports (443,
>>> 5061,5081,8081, 8082 and 7443) are listening.  I'm suspecting a certificate
>>> setup problem.
>>>
>>> In setting up the WSS certificates, it refers to WebRTC wiki,
>>> specifically bkw's link:
>>>
>>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/WebRTC#WebRTC-Aquickhowtofrombkw(BrianK.West):
>>>
>>> So far openssl is used to generate the certificates.
>>>
>>> This portion # Setup Sofia TLS the "agent.pem" is created and led me to
>>> ...
>>>
>>> https://wiki.freeswitch.org/wiki/SIP_TLS#Configuration
>>>
>>> However, gentls_cert is used to create "agent.pem"! Is there a conflict
>>> here? Which one to use?
>>>
>>> I appreciate for your guidance since I'm new to SSL/TLS.
>>>
>>> /Nandy
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> 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
>>
>
>
>
> --
>
>
>
> V?ctor E. Medina M.
> Platform Architect / Chief Infrastructure
> +58424 291 4561
> BB #79A8AFA2
> @VMCibersys
>
>
> _________________________________________________________________________
> 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 Jul  6 19:38:45 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 6 Jul 2015 11:38:45 -0400
Subject: [Freeswitch-users] Check session ready after session transfer
In-Reply-To: <CAKYJ0qZ0O_V=vu6LCHbih4AdKb0EQgvef2+-ZEByZX6hMEVRPQ@mail.gmail.com>
References: <CAKYJ0qYsJDnHyYVtfU_bX52pYFcZLn9RV+vydGZGxW=S=dUzVw@mail.gmail.com>
	<CAKYJ0qZ0O_V=vu6LCHbih4AdKb0EQgvef2+-ZEByZX6hMEVRPQ@mail.gmail.com>
Message-ID: <CAEwTsAh45HAnxQT6uPeCmO46MPzcns2dRZwLUSjhaS8KtLwcDg@mail.gmail.com>

this is a very vague question.  you are alluding to a problem but never
coming out and saying what problem you have.

On Monday, July 6, 2015, Zvi Agmon <zvi at lexifone.com> wrote:

> Hi,
>
> Can anyone help with this please?
> Didn't get any response - maybe the question is not clear enough...
>
> I'm trying to figure out the correct way to check for session state after
> it was transferred to another dial plan - meaning - a lua script is called
> from the new dial plan and in it I want to know if the session is in ready
> state.
>
> Thanks a lot
>
> Zvi Agmon
> Lexifone <http://www.lexifone.com>
> zvi at lexifone.com <javascript:_e(%7B%7D,'cvml','zvi at lexifone.com');>
>
>
> Best regards
>
> Zvi Agmon
> Lexifone <http://www.lexifone.com/>
> email: zvi at lexifone.com <javascript:_e(%7B%7D,'cvml','zvi at lexifone.com');>
> Office: +972-4-6817711
> Cell: +972-54-4505109
>
> On Tue, Jun 30, 2015 at 2:07 PM, Zvi Agmon <zvi at lexifone.com
> <javascript:_e(%7B%7D,'cvml','zvi at lexifone.com');>> wrote:
>
>> Hello,
>>
>> Need some help regarding session state validation.
>>
>> In documentation I see this:
>>
>> session:ready
>>
>> - checks whether the session is still active (true anytime between call
>> starts and hangup)
>>
>> - also session:ready will return false if the call is being transferred.
>> Bottom line is you should always be checking session:ready on any loops and
>> periodically throughout your script and exit asap if it returns false.
>>
>>
>> My experience is that after transfer the return value of session:ready()
>> is is not consistent - what is the correct way to check session state in
>> that case?
>>
>> Thanks
>>
>> Zvi Agmon
>> Lexifone <http://www.lexifone.com/>
>> email: zvi at lexifone.com
>> <javascript:_e(%7B%7D,'cvml','zvi at lexifone.com');>
>> Office: +972-4-6817711
>> Cell: +972-54-4505109
>>
>
>
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From mike at jerris.com  Mon Jul  6 19:40:42 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 6 Jul 2015 11:40:42 -0400
Subject: [Freeswitch-users] No Bridge Media
In-Reply-To: <CAEaAqfj3DJ384pRxkc3h82ik-0a23TQA76E75AUOQFNVkKLP+g@mail.gmail.com>
References: <CAEaAqfj3DJ384pRxkc3h82ik-0a23TQA76E75AUOQFNVkKLP+g@mail.gmail.com>
Message-ID: <CAEwTsAggC3+rCC3a=HcYqSwVMkA2rNr7-Eq-X5qofNES6NzCDA@mail.gmail.com>

what are you trying to accomplish by this config?

On Monday, July 6, 2015, Vishal Sharma <vishal.sharma at knowlarity.com> wrote:

> Hi,
> When I use following commnd on fs_cli, I can hear ring sound or any
> message telecome provider play on destination number
>
> originate freetdm/outgoing/r/09899790092
> &bridge(freetdm/outgoing/r/09899488723)
>
> but when I use one of following commands
>
> freeswitch at internal> originate
> {bridge_early_media=true}freetdm/outgoing/r/09899790092 &bridge
> (freetdm/outgoing/r/09899488723)
>
>
> freeswitch at internal> originate
> {bridge_early_media=true}freetdm/outgoing/r/09899790092
> &bridge({bridge_early_media=true}freetdm/outgoing/r/09899488723)
>
> or when I bridge a call which is received on FS, I don't get bridge media.
>
> Is there any perticular config required, or it's bug..
>
>
> Regards,
> Vishal Sharma
>
> SuperReceptionist is now available on Android mobiles. Track your business
> on the go with call analytics, recordings, insights and more: Download
> the app here
> <https://play.google.com/store/apps/details?id=com.superreceptionist>
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From mike at jerris.com  Mon Jul  6 20:05:47 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 6 Jul 2015 12:05:47 -0400
Subject: [Freeswitch-users] conference_set_auto_outcall not working with
	proxy server
In-Reply-To: <366805815.2377706.1436001012471.JavaMail.yahoo@mail.yahoo.com>
References: <366805815.2377706.1436001012471.JavaMail.yahoo@mail.yahoo.com>
Message-ID: <C7720B93-2515-4BFA-9D05-BEF7F32CF176@jerris.com>

Its hard to say exactly why with the truncated logs.  The reason why is probably above the portion you posted.  Try reading through those to find the reason.

> On Jul 4, 2015, at 5:10 AM, Saghar Ayyaz <saghar.ayyaz at yahoo.com> wrote:
> 
> conference_set_auto_outcall not working with proxy server. I am calling application from lua with following code. Suppose 167 is sip phone no.
> session:execute("conference_set_auto_outcall", "sofia/gateway/Kamailio/167")
> Gateway is defined in external.xml
> <gateway name="Kamailio">
>          <param name="extension" value="conference"/>
>          <param name="expire-seconds" value="125"/>
>          <param name="register-transport" value="udp"/>
>          <param name="proxy" value="xyz.com"/>
>          <param name="register" value="false"/>
>          <param name="caller-id-in-from" value="true"/> 
>          <param name="username" value="not-used"/>
>          <param name="password" value="not-used"/>
>     </gateway>
> Error:
> 2015-07-03 16:06:50.193203 [NOTICE] switch_core_session.c:1633 Session 35 (sofia/external/167) Ended 2015-07-03 16:06:50.193203 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/167 [CS_DESTROY] 2015-07-03 16:06:50.193203 [DEBUG] switch_core_state_machine.c:626 (sofia/external/167) Running State Change CS_DESTROY 2015-07-03 16:06:50.193203 [DEBUG] switch_core_state_machine.c:636 (sofia/external/167) State DESTROY 2015-07-03 16:06:50.193203 [DEBUG] mod_sofia.c:323 sofia/external/167 SOFIA DESTROY 2015-07-03 16:06:50.193203 [DEBUG] switch_core_state_machine.c:111 sofia/external/167 Standard DESTROY 2015-07-03 16:06:50.193203 [DEBUG] switch_core_state_machine.c:636 (sofia/external/167) State DESTROY going to sleep 2015-07-03 16:06:50.193203 [ERR] mod_conference.c:8469 Cannot create outgoing channel, cause: NORMAL_TEMPORARY_FAILURE 2015-07-03 16:06:50.193203 [DEBUG] switch_ivr_originate.c:2079 Parsing global variables
> _________________________________________________________________________
> 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 zvi at lexifone.com  Mon Jul  6 22:10:06 2015
From: zvi at lexifone.com (Zvi Agmon)
Date: Mon, 6 Jul 2015 21:10:06 +0300
Subject: [Freeswitch-users] Check session ready after session transfer
In-Reply-To: <CAEwTsAh45HAnxQT6uPeCmO46MPzcns2dRZwLUSjhaS8KtLwcDg@mail.gmail.com>
References: <CAKYJ0qYsJDnHyYVtfU_bX52pYFcZLn9RV+vydGZGxW=S=dUzVw@mail.gmail.com>
	<CAKYJ0qZ0O_V=vu6LCHbih4AdKb0EQgvef2+-ZEByZX6hMEVRPQ@mail.gmail.com>
	<CAEwTsAh45HAnxQT6uPeCmO46MPzcns2dRZwLUSjhaS8KtLwcDg@mail.gmail.com>
Message-ID: <CAKYJ0qYfhSnvLdZ7u5h-qrt5TH7Yu+nAOxTWCDmgSUOah1vqOg@mail.gmail.com>

Hi Michael,
Thanks for your response.

The issues I'm facing is in this scenario:
- an inbound call is answered and hit the first dial plan
- after performing application logic the call is transferred to another
dial plan.
- in this dial plan the call is bridged to an out bound leg and a lua
script is run when the outbound call is answered.
- in this lua script I need to do some logic but want to make sure the
session is alive.
  - for that I'm calling the method* session:ready* but in some cases the
method returns* false although the call is alive*. Also the documentation
says that after transfer session:ready should return false.

So my question is - how would you check that the session is in working
state at this stage?

Thanks
Zvi Agmon



Best regards

Zvi Agmon
Lexifone <http://www.lexifone.com/>
email: zvi at lexifone.com
Office: +972-4-6817711
Cell: +972-54-4505109

On Mon, Jul 6, 2015 at 6:38 PM, Michael Jerris <mike at jerris.com> wrote:

> this is a very vague question.  you are alluding to a problem but never
> coming out and saying what problem you have.
>
> On Monday, July 6, 2015, Zvi Agmon <zvi at lexifone.com> wrote:
>
>> Hi,
>>
>> Can anyone help with this please?
>> Didn't get any response - maybe the question is not clear enough...
>>
>> I'm trying to figure out the correct way to check for session state after
>> it was transferred to another dial plan - meaning - a lua script is called
>> from the new dial plan and in it I want to know if the session is in ready
>> state.
>>
>> Thanks a lot
>>
>> Zvi Agmon
>> Lexifone <http://www.lexifone.com>
>> zvi at lexifone.com
>>
>>
>> Best regards
>>
>> Zvi Agmon
>> Lexifone <http://www.lexifone.com/>
>> email: zvi at lexifone.com
>> Office: +972-4-6817711
>> Cell: +972-54-4505109
>>
>> On Tue, Jun 30, 2015 at 2:07 PM, Zvi Agmon <zvi at lexifone.com> wrote:
>>
>>> Hello,
>>>
>>> Need some help regarding session state validation.
>>>
>>> In documentation I see this:
>>>
>>> session:ready
>>>
>>> - checks whether the session is still active (true anytime between call
>>> starts and hangup)
>>>
>>> - also session:ready will return false if the call is being transferred.
>>> Bottom line is you should always be checking session:ready on any loops and
>>> periodically throughout your script and exit asap if it returns false.
>>>
>>>
>>> My experience is that after transfer the return value of session:ready()
>>> is is not consistent - what is the correct way to check session state in
>>> that case?
>>>
>>> Thanks
>>>
>>> Zvi Agmon
>>> Lexifone <http://www.lexifone.com/>
>>> email: zvi at lexifone.com
>>> Office: +972-4-6817711
>>> Cell: +972-54-4505109
>>>
>>
>>
> _________________________________________________________________________
> 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 Jul  6 22:30:43 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 6 Jul 2015 14:30:43 -0400
Subject: [Freeswitch-users] Check session ready after session transfer
In-Reply-To: <CAKYJ0qYfhSnvLdZ7u5h-qrt5TH7Yu+nAOxTWCDmgSUOah1vqOg@mail.gmail.com>
References: <CAKYJ0qYsJDnHyYVtfU_bX52pYFcZLn9RV+vydGZGxW=S=dUzVw@mail.gmail.com>
	<CAKYJ0qZ0O_V=vu6LCHbih4AdKb0EQgvef2+-ZEByZX6hMEVRPQ@mail.gmail.com>
	<CAEwTsAh45HAnxQT6uPeCmO46MPzcns2dRZwLUSjhaS8KtLwcDg@mail.gmail.com>
	<CAKYJ0qYfhSnvLdZ7u5h-qrt5TH7Yu+nAOxTWCDmgSUOah1vqOg@mail.gmail.com>
Message-ID: <4AAFEBD9-6160-42A0-8986-56542FB24A24@jerris.com>

What specifically are you looking for in "alive"

> On Jul 6, 2015, at 2:10 PM, Zvi Agmon <zvi at lexifone.com> wrote:
> 
> Hi Michael,
> Thanks for your response.
> 
> The issues I'm facing is in this scenario:
> - an inbound call is answered and hit the first dial plan
> - after performing application logic the call is transferred to another dial plan.
> - in this dial plan the call is bridged to an out bound leg and a lua script is run when the outbound call is answered.
> - in this lua script I need to do some logic but want to make sure the session is alive. 
>   - for that I'm calling the method session:ready but in some cases the method returns false although the call is alive. Also the documentation says that after transfer session:ready should return false.
> 
> So my question is - how would you check that the session is in working state at this stage?
> 
> Thanks
> Zvi Agmon
> 
> 
> 
> Best regards
> 
> Zvi Agmon
> Lexifone <http://www.lexifone.com/>
> email: zvi at lexifone.com <mailto:zvi at lexifone.com>
> Office: +972-4-6817711
> Cell: +972-54-4505109
> 
> On Mon, Jul 6, 2015 at 6:38 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> this is a very vague question.  you are alluding to a problem but never coming out and saying what problem you have.
> 
> On Monday, July 6, 2015, Zvi Agmon <zvi at lexifone.com <mailto:zvi at lexifone.com>> wrote:
> Hi,
> 
> Can anyone help with this please? 
> Didn't get any response - maybe the question is not clear enough...
> 
> I'm trying to figure out the correct way to check for session state after it was transferred to another dial plan - meaning - a lua script is called from the new dial plan and in it I want to know if the session is in ready state.
> 
> Thanks a lot
> 
> Zvi Agmon
> Lexifone <http://www.lexifone.com/>
> zvi at lexifone.com <>
> 
> 
> Best regards
> 
> Zvi Agmon
> Lexifone <http://www.lexifone.com/>
> email: zvi at lexifone.com <>
> Office: +972-4-6817711 <tel:%2B972-4-6817711>
> Cell: +972-54-4505109 <tel:%2B972-54-4505109>
> On Tue, Jun 30, 2015 at 2:07 PM, Zvi Agmon <zvi at lexifone.com <>> wrote:
> Hello,
> 
> Need some help regarding session state validation.
> 
> In documentation I see this:
> 
> session:ready
> - checks whether the session is still active (true anytime between call starts and hangup)
> - also session:ready will return false if the call is being transferred. Bottom line is you should always be checking session:ready on any loops and periodically throughout your script and exit asap if it returns false.
> 
> My experience is that after transfer the return value of session:ready() is is not consistent - what is the correct way to check session state in that case?
> 
> Thanks
> 
> Zvi Agmon
> Lexifone <http://www.lexifone.com/>
> email: zvi at lexifone.com <>
> Office: +972-4-6817711 <tel:%2B972-4-6817711>
> Cell: +972-54-4505109 <tel:%2B972-54-4505109>
> 
> _________________________________________________________________________
> 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
> 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 govoiper at gmail.com  Tue Jul  7 00:12:42 2015
From: govoiper at gmail.com (SamyGo)
Date: Mon, 6 Jul 2015 16:12:42 -0400
Subject: [Freeswitch-users] att_xfer/blind_xfer not working with B-leg DTMF
Message-ID: <CAJUJwtiWJfOAyq1oWcf1iJvQF2AWdmrUHrVMX0jSGCNeYiPu-g@mail.gmail.com>

Hi list,

Good day, I've been playing around with my FreeSwitch installation, all
good things except I can't seem to get att_xfer or blind transfer working
as expected.Tried several different ways but all lead to same results.

I've a LUA script running for each incoming call and this script makes all
the decisions. I've this line added before originating the B leg.

*session:execute("bind_meta_app", "7 ab s execute_extension::att_xfer XML
features")*

I can have this extension executed perfectly fine from leg A, but Leg B
shows strange behaviour.

Leg B can dial *7 and I get this att_xfer extension loaded which in turn
executes "read" and collect DTMF for the destination number.

*<action application="read" data="3 15 'tone_stream://%(10000,0,350,440)'
digits 30000 #"/>*

Now, I can see FS console showing up the RECV DTMF but I get nothing in
${digits}

I wonder why it stops collecting DTMF for B Leg while FS did detect the *7
code.

Any pointers would be appreciated.

BR,
Sammy
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From schoch+freeswitch.org at xwin32.com  Tue Jul  7 00:46:51 2015
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Mon, 6 Jul 2015 13:46:51 -0700
Subject: [Freeswitch-users] Faxing
Message-ID: <CAOkuKde3ocBmoW4T_6u24vTOk-VMnNczqK4XA9AZHFnynN_C1g@mail.gmail.com>

I haven't worked on it in a while, but I'm using Flowroute, and faxing has
always been trouble. Our you faxing directly from FS, or are you using a
fax machine? And are you using T38 (or force T38) in your receiving
settings?

-- 
Steve

On Fri, Jul 3, 2015 at 9:50 PM, Mitch Capper <mitch.capper at gmail.com> wrote:

>
> As a bonus right now do a lot of faxing with FS and flowroute and getting
> some great results for both incoming and outgoing.      While happy to keep
> using many providers for different things if able to converge on one all
> the better:)
>
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From brian at freeswitch.org  Tue Jul  7 00:50:17 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 6 Jul 2015 15:50:17 -0500
Subject: [Freeswitch-users] Faxing
In-Reply-To: <CAOkuKde3ocBmoW4T_6u24vTOk-VMnNczqK4XA9AZHFnynN_C1g@mail.gmail.com>
References: <CAOkuKde3ocBmoW4T_6u24vTOk-VMnNczqK4XA9AZHFnynN_C1g@mail.gmail.com>
Message-ID: <CAEJMVkD+n9xE5horubS7_Ev9+f5xqOVxnzaQ1LZMf3uYAWswow@mail.gmail.com>

I do 100% of my dev work on faxing against Flowroute, So if you're having
issues with say an ATA its probably a buggy firmware, but I've only had a
few weird cases of failures, I opened tickets with flowroute and they've
been resolved upstream.

On Mon, Jul 6, 2015 at 3:46 PM, Steven Schoch <
schoch+freeswitch.org at xwin32.com> wrote:

> I haven't worked on it in a while, but I'm using Flowroute, and faxing has
> always been trouble. Our you faxing directly from FS, or are you using a
> fax machine? And are you using T38 (or force T38) in your receiving
> settings?
>
> --
> Steve
>
> On Fri, Jul 3, 2015 at 9:50 PM, Mitch Capper <mitch.capper at gmail.com>
> wrote:
>
>>
>> As a bonus right now do a lot of faxing with FS and flowroute and getting
>> some great results for both incoming and outgoing.      While happy to keep
>> using many providers for different things if able to converge on one all
>> the better:)
>>
>
>
> _________________________________________________________________________
> 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
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From anthony.minessale at gmail.com  Tue Jul  7 02:37:27 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 6 Jul 2015 17:37:27 -0500
Subject: [Freeswitch-users] Check session ready after session transfer
In-Reply-To: <4AAFEBD9-6160-42A0-8986-56542FB24A24@jerris.com>
References: <CAKYJ0qYsJDnHyYVtfU_bX52pYFcZLn9RV+vydGZGxW=S=dUzVw@mail.gmail.com>
	<CAKYJ0qZ0O_V=vu6LCHbih4AdKb0EQgvef2+-ZEByZX6hMEVRPQ@mail.gmail.com>
	<CAEwTsAh45HAnxQT6uPeCmO46MPzcns2dRZwLUSjhaS8KtLwcDg@mail.gmail.com>
	<CAKYJ0qYfhSnvLdZ7u5h-qrt5TH7Yu+nAOxTWCDmgSUOah1vqOg@mail.gmail.com>
	<4AAFEBD9-6160-42A0-8986-56542FB24A24@jerris.com>
Message-ID: <CAKbxfG-GLSR9QtjG5tD0JSckJpETesVVdVxzd2Ha3KPO7_bhWg@mail.gmail.com>

As you quoted if session:ready returns false you MUST fall through to the
end of the script and allow it to exit without trying to hang it up or do
anything else.
When the state machine wants to move the session state from CS_EXECUTE
(running your script) to CS_ROUTING (new lookup in dialplan)  session ready
will be false.


On Mon, Jul 6, 2015 at 1:30 PM, Michael Jerris <mike at jerris.com> wrote:

> What specifically are you looking for in "alive"
>
> On Jul 6, 2015, at 2:10 PM, Zvi Agmon <zvi at lexifone.com> wrote:
>
> Hi Michael,
> Thanks for your response.
>
> The issues I'm facing is in this scenario:
> - an inbound call is answered and hit the first dial plan
> - after performing application logic the call is transferred to another
> dial plan.
> - in this dial plan the call is bridged to an out bound leg and a lua
> script is run when the outbound call is answered.
> - in this lua script I need to do some logic but want to make sure the
> session is alive.
>   - for that I'm calling the method* session:ready* but in some cases the
> method returns* false although the call is alive*. Also the documentation
> says that after transfer session:ready should return false.
>
> So my question is - how would you check that the session is in working
> state at this stage?
>
> Thanks
> Zvi Agmon
>
>
>
> Best regards
>
> Zvi Agmon
> Lexifone <http://www.lexifone.com/>
> email: zvi at lexifone.com
> Office: +972-4-6817711
> Cell: +972-54-4505109
>
> On Mon, Jul 6, 2015 at 6:38 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> this is a very vague question.  you are alluding to a problem but never
>> coming out and saying what problem you have.
>>
>> On Monday, July 6, 2015, Zvi Agmon <zvi at lexifone.com> wrote:
>>
>>> Hi,
>>>
>>> Can anyone help with this please?
>>> Didn't get any response - maybe the question is not clear enough...
>>>
>>> I'm trying to figure out the correct way to check for session state
>>> after it was transferred to another dial plan - meaning - a lua script is
>>> called from the new dial plan and in it I want to know if the session is in
>>> ready state.
>>>
>>> Thanks a lot
>>>
>>> Zvi Agmon
>>> Lexifone <http://www.lexifone.com/>
>>> zvi at lexifone.com
>>>
>>>
>>> Best regards
>>>
>>> Zvi Agmon
>>> Lexifone <http://www.lexifone.com/>
>>> email: zvi at lexifone.com
>>> Office: +972-4-6817711
>>> Cell: +972-54-4505109
>>>
>>> On Tue, Jun 30, 2015 at 2:07 PM, Zvi Agmon <zvi at lexifone.com> wrote:
>>>
>>>> Hello,
>>>>
>>>> Need some help regarding session state validation.
>>>>
>>>> In documentation I see this:
>>>>
>>>> session:ready
>>>> - checks whether the session is still active (true anytime between call
>>>> starts and hangup)
>>>> - also session:ready will return false if the call is being
>>>> transferred. Bottom line is you should always be checking session:ready on
>>>> any loops and periodically throughout your script and exit asap if it
>>>> returns false.
>>>>
>>>>
>>>> My experience is that after transfer the return value of
>>>> session:ready() is is not consistent - what is the correct way to check
>>>> session state in that case?
>>>>
>>>> Thanks
>>>>
>>>> Zvi Agmon
>>>> Lexifone <http://www.lexifone.com/>
>>>> email: zvi at lexifone.com
>>>> Office: +972-4-6817711
>>>> Cell: +972-54-4505109
>>>>
>>>
>>>
>> _________________________________________________________________________
>> 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
>



-- 
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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From mitch.capper at gmail.com  Tue Jul  7 06:43:46 2015
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 6 Jul 2015 19:43:46 -0700
Subject: [Freeswitch-users] Faxing
In-Reply-To: <CAEJMVkD+n9xE5horubS7_Ev9+f5xqOVxnzaQ1LZMf3uYAWswow@mail.gmail.com>
References: <CAOkuKde3ocBmoW4T_6u24vTOk-VMnNczqK4XA9AZHFnynN_C1g@mail.gmail.com>
	<CAEJMVkD+n9xE5horubS7_Ev9+f5xqOVxnzaQ1LZMf3uYAWswow@mail.gmail.com>
Message-ID: <CALn233qkFpPpH+qDbBMQGRdL-mqxa2uKCX=KbniyPgWaFjSE5Q@mail.gmail.com>

We fax with flowroute all the time.  If you search in the archives you will
find where I post stats and the methods we use for highly successful faxing.

~Mitch

On Mon, Jul 6, 2015 at 1:50 PM, Brian West <brian at freeswitch.org> wrote:

> I do 100% of my dev work on faxing against Flowroute, So if you're having
> issues with say an ATA its probably a buggy firmware, but I've only had a
> few weird cases of failures, I opened tickets with flowroute and they've
> been resolved upstream.
>
> On Mon, Jul 6, 2015 at 3:46 PM, Steven Schoch <
> schoch+freeswitch.org at xwin32.com> wrote:
>
>> I haven't worked on it in a while, but I'm using Flowroute, and faxing
>> has always been trouble. Our you faxing directly from FS, or are you using
>> a fax machine? And are you using T38 (or force T38) in your receiving
>> settings?
>>
>> --
>> Steve
>>
>> On Fri, Jul 3, 2015 at 9:50 PM, Mitch Capper <mitch.capper at gmail.com>
>> wrote:
>>
>>>
>>> As a bonus right now do a lot of faxing with FS and flowroute and
>>> getting some great results for both incoming and outgoing.      While happy
>>> to keep using many providers for different things if able to converge on
>>> one all the better:)
>>>
>>
>>
>> _________________________________________________________________________
>> 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
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>
> _________________________________________________________________________
> 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 freeswitch.org  Tue Jul  7 06:55:03 2015
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 07 Jul 2015 02:55:03 +0000
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) June
	27th-July 3rd
Message-ID: <559b3f8745bc8_8b6610113186543e@resque-worker.5.mail>

New Post on freeswitch.org from Kathleen King
check it out at http://ift.tt/1KIMIBl
FreeSWITCH Week in Review (Master Branch) June 27th-July 3rd
Hello, again. This passed week in the FreeSWITCH master branch we had 49 commits. This week we had a bunch of new features with most of them being helpful little improvements, but we also had two new modules merged in! The 2600hz guys added mod_kazoo and William King merged in mod_smpp. You can find out more about mod_smpp by going here. And, the 2600hz patches are all slated to be merged in by the 1.6 release.

Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support.

New features that were added:

FS-7732 Continue recording with uuid_transfer

FS-7752 [mod_rayo] Increase maximum number of elements from 30 to 1024 to allow adhearsion to create large grammars to navigate IVR menus.

FS-7750 [mod_commands] Allow for uuid_setvar to handle arrays

FS-7758 [mod_loopback] Emit an event if a loopback bowout occurs

FS-7759 [mod_sofia] Added the channel variable ignore_completed_elsewhere to suppress setting the completed elsewhere cause

FS-7771 Set a channel variable if the recording is terminated due to silence hits

FS-7760 Added xml fetch for channels to externally support nightmare transfer depends on channel-xml-fetch-on-nightmare-transfer profile param (default is disabled)

FS-7730 [mod_smpp] Added mod_smpp as an event handler module
and fixed the default configs to provided sample load option for mod_sms and mod_smpp

FS-7774 Add mod_kazoo

Improvements in build system, cross platform support, and packaging:

OPENZAP-238 [freetdm] Fix some GSM compilation errors and do a bit of code cleanup

OPENZAP-237 [freetdm] Use __func__ instead of __FUNCTION__ to comply with c99 in gcc 5.1

The following bugs were squashed:

FS-7734 [mod_nibblebill] Fixed a deadlock

FS-7726 Fixed a bug with recording a video session on DTMF command

FS-7721 Fixed a segfault caused when using session:recordFile() and session:unsetInputCallback in a lua script

FS-7429 [mod_curl] Fixed to output valid json

FS-7746 [mod_verto] Fixed a device permission error in verto client

FS-7753 [mod_local_stream] Fixed some glitching and freezing video when using hold/unhold

FS-7761 [core] Fix shutdown races running api commands during shutdown

FS-7767 [mod_sofia] Fixed a segfault caused by invalid arguments to sip_dig

FS-7744 [mod_conference] Fixed a bug causing the first user?s video stream to stop when another verto user calls the conference

FS-7486 [mod_sofia] Fixed the handling of queued requests

FS-7775 [mod_conference] Fix threading issue causing stuck worker threads

FS-7777 [mod_imagick] Fixed a regression causing a segfault when playing png &amp; pdf in conference

And, this passed week in the FreeSWITCH 1.4 branch we had 2 commits merged in from master.

FS-7486 [mod_sofia] Fixed the handling of queued requests

FS-7750 [mod_commands] Set uuid_setvar to handle arrays

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From covici at ccs.covici.com  Tue Jul  7 08:04:22 2015
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 07 Jul 2015 00:04:22 -0400
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch)
	June 27th-July 3rd
In-Reply-To: <559b3f8745bc8_8b6610113186543e@resque-worker.5.mail>
References: <559b3f8745bc8_8b6610113186543e@resque-worker.5.mail>
Message-ID: <12443.1436241862@ccs.covici.com>

What the heck is a nightmare transfer?

Ken Rice <krice at freeswitch.org> wrote:

> New Post on freeswitch.org from Kathleen King
> check it out at http://ift.tt/1KIMIBl
> FreeSWITCH Week in Review (Master Branch) June 27th-July 3rd
> Hello, again. This passed week in the FreeSWITCH master branch we had 49 commits. This week we had a bunch of new features with most of them being helpful little improvements, but we also had two new modules merged in! The 2600hz guys added mod_kazoo and William King merged in mod_smpp. You can find out more about mod_smpp by going here. And, the 2600hz patches are all slated to be merged in by the 1.6 release.
> 
> Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support.
> 
> New features that were added:
> 
> FS-7732 Continue recording with uuid_transfer
> 
> FS-7752 [mod_rayo] Increase maximum number of elements from 30 to 1024 to allow adhearsion to create large grammars to navigate IVR menus.
> 
> FS-7750 [mod_commands] Allow for uuid_setvar to handle arrays
> 
> FS-7758 [mod_loopback] Emit an event if a loopback bowout occurs
> 
> FS-7759 [mod_sofia] Added the channel variable ignore_completed_elsewhere to suppress setting the completed elsewhere cause
> 
> FS-7771 Set a channel variable if the recording is terminated due to silence hits
> 
> FS-7760 Added xml fetch for channels to externally support nightmare transfer depends on channel-xml-fetch-on-nightmare-transfer profile param (default is disabled)
> 
> FS-7730 [mod_smpp] Added mod_smpp as an event handler module
> and fixed the default configs to provided sample load option for mod_sms and mod_smpp
> 
> FS-7774 Add mod_kazoo
> 
> Improvements in build system, cross platform support, and packaging:
> 
> OPENZAP-238 [freetdm] Fix some GSM compilation errors and do a bit of code cleanup
> 
> OPENZAP-237 [freetdm] Use __func__ instead of __FUNCTION__ to comply with c99 in gcc 5.1
> 
> The following bugs were squashed:
> 
> FS-7734 [mod_nibblebill] Fixed a deadlock
> 
> FS-7726 Fixed a bug with recording a video session on DTMF command
> 
> FS-7721 Fixed a segfault caused when using session:recordFile() and session:unsetInputCallback in a lua script
> 
> FS-7429 [mod_curl] Fixed to output valid json
> 
> FS-7746 [mod_verto] Fixed a device permission error in verto client
> 
> FS-7753 [mod_local_stream] Fixed some glitching and freezing video when using hold/unhold
> 
> FS-7761 [core] Fix shutdown races running api commands during shutdown
> 
> FS-7767 [mod_sofia] Fixed a segfault caused by invalid arguments to sip_dig
> 
> FS-7744 [mod_conference] Fixed a bug causing the first user?s video stream to stop when another verto user calls the conference
> 
> FS-7486 [mod_sofia] Fixed the handling of queued requests
> 
> FS-7775 [mod_conference] Fix threading issue causing stuck worker threads
> 
> FS-7777 [mod_imagick] Fixed a regression causing a segfault when playing png &amp; pdf in conference
> 
> And, this passed week in the FreeSWITCH 1.4 branch we had 2 commits merged in from master.
> 
> FS-7486 [mod_sofia] Fixed the handling of queued requests
> 
> FS-7750 [mod_commands] Set uuid_setvar to handle arrays
> 
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> 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
-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From zvi at lexifone.com  Tue Jul  7 10:35:00 2015
From: zvi at lexifone.com (Zvi Agmon)
Date: Tue, 7 Jul 2015 09:35:00 +0300
Subject: [Freeswitch-users] Check session ready after session transfer
In-Reply-To: <CAKbxfG-GLSR9QtjG5tD0JSckJpETesVVdVxzd2Ha3KPO7_bhWg@mail.gmail.com>
References: <CAKYJ0qYsJDnHyYVtfU_bX52pYFcZLn9RV+vydGZGxW=S=dUzVw@mail.gmail.com>
	<CAKYJ0qZ0O_V=vu6LCHbih4AdKb0EQgvef2+-ZEByZX6hMEVRPQ@mail.gmail.com>
	<CAEwTsAh45HAnxQT6uPeCmO46MPzcns2dRZwLUSjhaS8KtLwcDg@mail.gmail.com>
	<CAKYJ0qYfhSnvLdZ7u5h-qrt5TH7Yu+nAOxTWCDmgSUOah1vqOg@mail.gmail.com>
	<4AAFEBD9-6160-42A0-8986-56542FB24A24@jerris.com>
	<CAKbxfG-GLSR9QtjG5tD0JSckJpETesVVdVxzd2Ha3KPO7_bhWg@mail.gmail.com>
Message-ID: <CAKYJ0qbzrBnx1tAKKcjwk8F+sTiLCk1ekKYG=7RyNtywx+3Mtw@mail.gmail.com>

Thanks Anthony,

I'm trying to figure out what happens next - after the route to new dial
plan completed. Now a new lua script is executed (as a hookup to answer) -
what is the expected mode at that state - isn't it CS_EXECUTE again?
How should I validate the session is ready in my answer hookup script?

My experience is that in most cases session:ready() returns true at that
stage, but sometime it returns false although  the call continue without
any problem (2 legs are bridged and hears each other).

Thanks
Zvi Agmon
Lexifone


Best regards

Zvi Agmon
Lexifone <http://www.lexifone.com/>
email: zvi at lexifone.com
Office: +972-4-6817711
Cell: +972-54-4505109

On Tue, Jul 7, 2015 at 1:37 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> As you quoted if session:ready returns false you MUST fall through to the
> end of the script and allow it to exit without trying to hang it up or do
> anything else.
> When the state machine wants to move the session state from CS_EXECUTE
> (running your script) to CS_ROUTING (new lookup in dialplan)  session ready
> will be false.
>
>
> On Mon, Jul 6, 2015 at 1:30 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> What specifically are you looking for in "alive"
>>
>> On Jul 6, 2015, at 2:10 PM, Zvi Agmon <zvi at lexifone.com> wrote:
>>
>> Hi Michael,
>> Thanks for your response.
>>
>> The issues I'm facing is in this scenario:
>> - an inbound call is answered and hit the first dial plan
>> - after performing application logic the call is transferred to another
>> dial plan.
>> - in this dial plan the call is bridged to an out bound leg and a lua
>> script is run when the outbound call is answered.
>> - in this lua script I need to do some logic but want to make sure the
>> session is alive.
>>   - for that I'm calling the method* session:ready* but in some cases
>> the method returns* false although the call is alive*. Also the
>> documentation says that after transfer session:ready should return false.
>>
>> So my question is - how would you check that the session is in working
>> state at this stage?
>>
>> Thanks
>> Zvi Agmon
>>
>>
>>
>> Best regards
>>
>> Zvi Agmon
>> Lexifone <http://www.lexifone.com/>
>> email: zvi at lexifone.com
>> Office: +972-4-6817711
>> Cell: +972-54-4505109
>>
>> On Mon, Jul 6, 2015 at 6:38 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> this is a very vague question.  you are alluding to a problem but never
>>> coming out and saying what problem you have.
>>>
>>> On Monday, July 6, 2015, Zvi Agmon <zvi at lexifone.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> Can anyone help with this please?
>>>> Didn't get any response - maybe the question is not clear enough...
>>>>
>>>> I'm trying to figure out the correct way to check for session state
>>>> after it was transferred to another dial plan - meaning - a lua script is
>>>> called from the new dial plan and in it I want to know if the session is in
>>>> ready state.
>>>>
>>>> Thanks a lot
>>>>
>>>> Zvi Agmon
>>>> Lexifone <http://www.lexifone.com/>
>>>> zvi at lexifone.com
>>>>
>>>>
>>>> Best regards
>>>>
>>>> Zvi Agmon
>>>> Lexifone <http://www.lexifone.com/>
>>>> email: zvi at lexifone.com
>>>> Office: +972-4-6817711
>>>> Cell: +972-54-4505109
>>>>
>>>> On Tue, Jun 30, 2015 at 2:07 PM, Zvi Agmon <zvi at lexifone.com> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> Need some help regarding session state validation.
>>>>>
>>>>> In documentation I see this:
>>>>>
>>>>> session:ready
>>>>> - checks whether the session is still active (true anytime between
>>>>> call starts and hangup)
>>>>> - also session:ready will return false if the call is being
>>>>> transferred. Bottom line is you should always be checking session:ready on
>>>>> any loops and periodically throughout your script and exit asap if it
>>>>> returns false.
>>>>>
>>>>>
>>>>> My experience is that after transfer the return value of
>>>>> session:ready() is is not consistent - what is the correct way to check
>>>>> session state in that case?
>>>>>
>>>>> Thanks
>>>>>
>>>>> Zvi Agmon
>>>>> Lexifone <http://www.lexifone.com/>
>>>>> email: zvi at lexifone.com
>>>>> Office: +972-4-6817711
>>>>> Cell: +972-54-4505109
>>>>>
>>>>
>>>>
>>> _________________________________________________________________________
>>> 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
>>
>
>
>
> --
> 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
> [image: ?] sip:888 at conference.freeswitch.org  [image: ?] +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> 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 babak.freeswitch at gmail.com  Tue Jul  7 12:21:55 2015
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Tue, 7 Jul 2015 12:51:55 +0430
Subject: [Freeswitch-users] calling intercept from event socket
Message-ID: <CAEVfcaaCKqme2wKKxaf=ct1eOwQU1Vrtj+eWFYpj0F07vQYKcg@mail.gmail.com>

Hi
I'm using event socket to do something like freeswitch callcenter module.
members(aleg) call and are connected to socket application, this is my
dialplan:
<extension name="socket">
       <condition field="destination_number" expression="^999$">
     <action application="set" data="socket_resume=true"/>
             <action application="socket" data="172.16.90.1:8084 async
full"/>
  <action application="info"/>
       </condition>
     </extension>
these are executed using socket:
connect
myevents
set park_after_bridge=true
answer
playback local_stream://moh

now a new channel is originated to agent(bleg) and on answer I'm calling
intercept on aleg from socket to bridge it to agent channel. if I
set socket_resume=true in dialplan, after intercept both legs are stuck
(dead silent). if I do not set socket_resume=true legs are bridged but
execution continues through dialplan (I placed info to test this).
problem is if  socket_resume=false and agent transfers the call to 999
during the call (same socket app) call hangsup. but if socket_resume=true
intercept is not working as expected but transfer is done. how can I solve
this?
thanks
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From igorolhovskiy at gmail.com  Tue Jul  7 11:50:16 2015
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Tue, 7 Jul 2015 10:50:16 +0300
Subject: [Freeswitch-users] FS 1.7 / [CS_CONSUME_MEDIA]
	[INCOMPATIBLE_DESTINATION]
Message-ID: <CAJTkRNvtZ4VFuRdASQBticwcgRszZGAzhgBqMjT79S3TS5-VcA@mail.gmail.com>

Hi!

Trying to up FS 1.7 (master release from 07.07.2015) with web-based SIP
(jsSIP 0.6) (Yes, I know about mod_verto)
When making an ORIGINATE request (one of legs is Web-softphone) got this:

2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:40
sofia/internal/rlakoni6 at avkit3h1g7jo.invalid Standard INIT
2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:48
(sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) State Change CS_INIT ->
CS_ROUTING
2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:516
(sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) State INIT going to sleep
2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:473
(sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) Running State Change
CS_ROUTING
2015-07-07 07:35:30.354313 [DEBUG] sofia.c:6708 Channel
sofia/internal/rlakoni6 at avkit3h1g7jo.invalid entering state [calling][0]
2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:532
(sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) State ROUTING
2015-07-07 07:35:30.354313 [DEBUG] mod_sofia.c:141
sofia/internal/rlakoni6 at avkit3h1g7jo.invalid SOFIA ROUTING
2015-07-07 07:35:30.354313 [DEBUG] switch_ivr_originate.c:67
(sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:532
(sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) State ROUTING going to sleep
2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:473
(sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) Running State Change
CS_CONSUME_MEDIA
2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:551
(sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) State CONSUME_MEDIA
2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:551
(sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) State CONSUME_MEDIA going to
sleep
2015-07-07 07:35:30.594339 [DEBUG] sofia.c:6708 Channel
sofia/internal/rlakoni6 at avkit3h1g7jo.invalid entering state
[proceeding][180]
2015-07-07 07:35:30.594339 [NOTICE] sofia.c:6810 Ring-Ready
sofia/internal/rlakoni6 at avkit3h1g7jo.invalid!
2015-07-07 07:35:30.594339 [DEBUG] switch_channel.c:3332
(sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) Callstate Change DOWN ->
RINGING
2015-07-07 07:35:32.434365 [DEBUG] sofia.c:6708 Channel
sofia/internal/rlakoni6 at avkit3h1g7jo.invalid entering state
[completing][200]
2015-07-07 07:35:32.434365 [DEBUG] sofia.c:6718 Remote SDP:
v=0
o=- 4629109737787807592 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS nIK9U7F3jq79chtGTGAtUOON7Lbekhy8msI2
m=audio 61483 RTP/SAVPF 0 8 13
c=IN IP4 192.168.88.51
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:708354780 1 udp 2122260223 192.168.88.51 61483 typ host
generation 0
a=candidate:1690000940 1 tcp 1518280447 192.168.88.51 0 typ host tcptype
active generation 0
a=ice-ufrag:Bb2sJM3lTievaC/+
a=ice-pwd:0wnAgFcoY7QnqfqNg11YYkkA
a=fingerprint:sha-256
90:46:2D:15:D0:D6:24:34:F1:8E:8C:8C:20:F9:49:3D:F6:9B:C1:AD:92:16:5F:A9:46:18:30:9C:59:99:41:C0
a=setup:active
a=mid:audio
a=rtcp-mux
a=ssrc:2731247554 cname:d/DW/QebHgyWUgUz
a=ssrc:2731247554 msid:nIK9U7F3jq79chtGTGAtUOON7Lbekhy8msI2
c3732e3c-701f-496c-a0df-92bac3364375
a=ssrc:2731247554 mslabel:nIK9U7F3jq79chtGTGAtUOON7Lbekhy8msI2
a=ssrc:2731247554 label:c3732e3c-701f-496c-a0df-92bac3364375

2015-07-07 07:35:32.434365 [DEBUG] sofia.c:6708 Channel
sofia/internal/rlakoni6 at avkit3h1g7jo.invalid entering state [ready][200]
2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4065 Audio Codec
Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
*2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4120 Audio Codec
Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match*
2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4065 Audio Codec
Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4065 Audio Codec
Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4065 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4065 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
*2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4120 Audio Codec
Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match*
2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4065 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:2851 Set Codec
sofia/internal/rlakoni6 at avkit3h1g7jo.invalid PCMU/8000 20 ms 160 samples
64000 bits 1 channels
2015-07-07 07:35:32.434365 [DEBUG] switch_core_codec.c:111
sofia/internal/rlakoni6 at avkit3h1g7jo.invalid Original read codec set to
PCMU:0
2015-07-07 07:35:32.434365 [WARNING] switch_core_media.c:3187 NO candidate
ACL defined, Defaulting to wan.auto
2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:3216 Save audio
Candidate cid: 1 proto: udp type: host addr: 192.168.88.51:61483
*2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:3256 Searching for
rtp candidate.*
*2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:3256 Searching for
rtcp candidate.*
*2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:3300
sofia/internal/rlakoni6 at avkit3h1g7jo.invalid no suitable candidates found.*
2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4345 No 2833 in
SDP.  Disable 2833 dtmf and switch to INFO
2015-07-07 07:35:32.434365 [NOTICE] sofia.c:7591 Hangup
sofia/internal/rlakoni6 at avkit3h1g7jo.invalid [CS_CONSUME_MEDIA]
[INCOMPATIBLE_DESTINATION]
2015-07-07 07:35:32.434365 [DEBUG] sofia.c:1355 Channel is already hungup.

PCMA and PCMU are in there. But why incompatible dest? I've got ring, but
after accept mic sharing, got hangup.

On FS 1.5.15 same config, all is ok.
-- 
Best regards,
Igor
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From igorolhovskiy at gmail.com  Tue Jul  7 12:27:18 2015
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Tue, 7 Jul 2015 11:27:18 +0300
Subject: [Freeswitch-users] calling intercept from event socket
In-Reply-To: <CAEVfcaaCKqme2wKKxaf=ct1eOwQU1Vrtj+eWFYpj0F07vQYKcg@mail.gmail.com>
References: <CAEVfcaaCKqme2wKKxaf=ct1eOwQU1Vrtj+eWFYpj0F07vQYKcg@mail.gmail.com>
Message-ID: <CAJTkRNtZ7wtCkjS7nSfHmMs+y0Eh_sToqDhqwFVoAK9Vs7n3CQ@mail.gmail.com>

Try to remove async.

2015-07-07 11:21 GMT+03:00 Babak Yakhchali <babak.freeswitch at gmail.com>:

> Hi
> I'm using event socket to do something like freeswitch callcenter module.
> members(aleg) call and are connected to socket application, this is my
> dialplan:
> <extension name="socket">
>        <condition field="destination_number" expression="^999$">
>      <action application="set" data="socket_resume=true"/>
>              <action application="socket" data="172.16.90.1:8084 async
> full"/>
>   <action application="info"/>
>        </condition>
>      </extension>
> these are executed using socket:
> connect
> myevents
> set park_after_bridge=true
> answer
> playback local_stream://moh
>
> now a new channel is originated to agent(bleg) and on answer I'm calling
> intercept on aleg from socket to bridge it to agent channel. if I
> set socket_resume=true in dialplan, after intercept both legs are stuck
> (dead silent). if I do not set socket_resume=true legs are bridged but
> execution continues through dialplan (I placed info to test this).
> problem is if  socket_resume=false and agent transfers the call to 999
> during the call (same socket app) call hangsup. but if socket_resume=true
> intercept is not working as expected but transfer is done. how can I solve
> this?
> thanks
>
>
> _________________________________________________________________________
> 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
>



-- 
Best regards,
Igor
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From babak.freeswitch at gmail.com  Tue Jul  7 12:59:16 2015
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Tue, 7 Jul 2015 13:29:16 +0430
Subject: [Freeswitch-users] calling intercept from event socket
In-Reply-To: <CAJTkRNtZ7wtCkjS7nSfHmMs+y0Eh_sToqDhqwFVoAK9Vs7n3CQ@mail.gmail.com>
References: <CAEVfcaaCKqme2wKKxaf=ct1eOwQU1Vrtj+eWFYpj0F07vQYKcg@mail.gmail.com>
	<CAJTkRNtZ7wtCkjS7nSfHmMs+y0Eh_sToqDhqwFVoAK9Vs7n3CQ@mail.gmail.com>
Message-ID: <CAEVfcaYBqeTWreL5hp+3YsAXrDRk6L6Bejt90LcHSwiS7db0pA@mail.gmail.com>

when I'm playing moh I'm listening for events. I think if async is removed
I can not do something like this? can I?

On Tue, Jul 7, 2015 at 12:57 PM, Igor Olhovskiy <igorolhovskiy at gmail.com>
wrote:

> Try to remove async.
>
> 2015-07-07 11:21 GMT+03:00 Babak Yakhchali <babak.freeswitch at gmail.com>:
>
>> Hi
>> I'm using event socket to do something like freeswitch callcenter module.
>> members(aleg) call and are connected to socket application, this is my
>> dialplan:
>> <extension name="socket">
>>        <condition field="destination_number" expression="^999$">
>>      <action application="set" data="socket_resume=true"/>
>>              <action application="socket" data="172.16.90.1:8084 async
>> full"/>
>>   <action application="info"/>
>>        </condition>
>>      </extension>
>> these are executed using socket:
>> connect
>> myevents
>> set park_after_bridge=true
>> answer
>> playback local_stream://moh
>>
>> now a new channel is originated to agent(bleg) and on answer I'm calling
>> intercept on aleg from socket to bridge it to agent channel. if I
>> set socket_resume=true in dialplan, after intercept both legs are stuck
>> (dead silent). if I do not set socket_resume=true legs are bridged but
>> execution continues through dialplan (I placed info to test this).
>> problem is if  socket_resume=false and agent transfers the call to 999
>> during the call (same socket app) call hangsup. but if socket_resume=true
>> intercept is not working as expected but transfer is done. how can I solve
>> this?
>> thanks
>>
>>
>> _________________________________________________________________________
>> 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
>>
>
>
>
> --
> Best 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 mike at jerris.com  Tue Jul  7 18:37:43 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 7 Jul 2015 10:37:43 -0400
Subject: [Freeswitch-users] FS 1.7 / [CS_CONSUME_MEDIA]
	[INCOMPATIBLE_DESTINATION]
In-Reply-To: <CAJTkRNvtZ4VFuRdASQBticwcgRszZGAzhgBqMjT79S3TS5-VcA@mail.gmail.com>
References: <CAJTkRNvtZ4VFuRdASQBticwcgRszZGAzhgBqMjT79S3TS5-VcA@mail.gmail.com>
Message-ID: <CAEwTsAhbzM1zfz96qtvG9+MUr13w1XJqGX4AgetuN55mZCTsYg@mail.gmail.com>

jssip is known broken in multiple ways, you can review the mailing list
archives for details.  It simply does not work correctly for anyone using it

On Tuesday, July 7, 2015, Igor Olhovskiy <igorolhovskiy at gmail.com> wrote:

> Hi!
>
> Trying to up FS 1.7 (master release from 07.07.2015) with web-based SIP
> (jsSIP 0.6) (Yes, I know about mod_verto)
> When making an ORIGINATE request (one of legs is Web-softphone) got this:
>
> 2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:40
> sofia/internal/rlakoni6 at avkit3h1g7jo.invalid Standard INIT
> 2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:48
> (sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) State Change CS_INIT ->
> CS_ROUTING
> 2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:516
> (sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) State INIT going to sleep
> 2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:473
> (sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) Running State Change
> CS_ROUTING
> 2015-07-07 07:35:30.354313 [DEBUG] sofia.c:6708 Channel
> sofia/internal/rlakoni6 at avkit3h1g7jo.invalid entering state [calling][0]
> 2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:532
> (sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) State ROUTING
> 2015-07-07 07:35:30.354313 [DEBUG] mod_sofia.c:141
> sofia/internal/rlakoni6 at avkit3h1g7jo.invalid SOFIA ROUTING
> 2015-07-07 07:35:30.354313 [DEBUG] switch_ivr_originate.c:67
> (sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:532
> (sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) State ROUTING going to
> sleep
> 2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:473
> (sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) Running State Change
> CS_CONSUME_MEDIA
> 2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:551
> (sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) State CONSUME_MEDIA
> 2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:551
> (sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) State CONSUME_MEDIA going
> to sleep
> 2015-07-07 07:35:30.594339 [DEBUG] sofia.c:6708 Channel
> sofia/internal/rlakoni6 at avkit3h1g7jo.invalid entering state
> [proceeding][180]
> 2015-07-07 07:35:30.594339 [NOTICE] sofia.c:6810 Ring-Ready
> sofia/internal/rlakoni6 at avkit3h1g7jo.invalid!
> 2015-07-07 07:35:30.594339 [DEBUG] switch_channel.c:3332
> (sofia/internal/rlakoni6 at avkit3h1g7jo.invalid) Callstate Change DOWN ->
> RINGING
> 2015-07-07 07:35:32.434365 [DEBUG] sofia.c:6708 Channel
> sofia/internal/rlakoni6 at avkit3h1g7jo.invalid entering state
> [completing][200]
> 2015-07-07 07:35:32.434365 [DEBUG] sofia.c:6718 Remote SDP:
> v=0
> o=- 4629109737787807592 2 IN IP4 127.0.0.1
> s=-
> t=0 0
> a=msid-semantic: WMS nIK9U7F3jq79chtGTGAtUOON7Lbekhy8msI2
> m=audio 61483 RTP/SAVPF 0 8 13
> c=IN IP4 192.168.88.51
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:13 CN/8000
> a=rtcp:9 IN IP4 0.0.0.0
> a=candidate:708354780 1 udp 2122260223 192.168.88.51 61483 typ host
> generation 0
> a=candidate:1690000940 1 tcp 1518280447 192.168.88.51 0 typ host tcptype
> active generation 0
> a=ice-ufrag:Bb2sJM3lTievaC/+
> a=ice-pwd:0wnAgFcoY7QnqfqNg11YYkkA
> a=fingerprint:sha-256
> 90:46:2D:15:D0:D6:24:34:F1:8E:8C:8C:20:F9:49:3D:F6:9B:C1:AD:92:16:5F:A9:46:18:30:9C:59:99:41:C0
> a=setup:active
> a=mid:audio
> a=rtcp-mux
> a=ssrc:2731247554 cname:d/DW/QebHgyWUgUz
> a=ssrc:2731247554 msid:nIK9U7F3jq79chtGTGAtUOON7Lbekhy8msI2
> c3732e3c-701f-496c-a0df-92bac3364375
> a=ssrc:2731247554 mslabel:nIK9U7F3jq79chtGTGAtUOON7Lbekhy8msI2
> a=ssrc:2731247554 label:c3732e3c-701f-496c-a0df-92bac3364375
>
> 2015-07-07 07:35:32.434365 [DEBUG] sofia.c:6708 Channel
> sofia/internal/rlakoni6 at avkit3h1g7jo.invalid entering state [ready][200]
> 2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4065 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
> *2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4120 Audio Codec
> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match*
> 2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4065 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
> 2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4065 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
> 2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4065 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
> 2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4065 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
> *2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4120 Audio Codec
> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match*
> 2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4065 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
> 2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:2851 Set Codec
> sofia/internal/rlakoni6 at avkit3h1g7jo.invalid PCMU/8000 20 ms 160 samples
> 64000 bits 1 channels
> 2015-07-07 07:35:32.434365 [DEBUG] switch_core_codec.c:111
> sofia/internal/rlakoni6 at avkit3h1g7jo.invalid Original read codec set to
> PCMU:0
> 2015-07-07 07:35:32.434365 [WARNING] switch_core_media.c:3187 NO candidate
> ACL defined, Defaulting to wan.auto
> 2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:3216 Save audio
> Candidate cid: 1 proto: udp type: host addr: 192.168.88.51:61483
> *2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:3256 Searching for
> rtp candidate.*
> *2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:3256 Searching for
> rtcp candidate.*
> *2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:3300
> sofia/internal/rlakoni6 at avkit3h1g7jo.invalid no suitable candidates found.*
> 2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4345 No 2833 in
> SDP.  Disable 2833 dtmf and switch to INFO
> 2015-07-07 07:35:32.434365 [NOTICE] sofia.c:7591 Hangup
> sofia/internal/rlakoni6 at avkit3h1g7jo.invalid [CS_CONSUME_MEDIA]
> [INCOMPATIBLE_DESTINATION]
> 2015-07-07 07:35:32.434365 [DEBUG] sofia.c:1355 Channel is already hungup.
>
> PCMA and PCMU are in there. But why incompatible dest? I've got ring, but
> after accept mic sharing, got hangup.
>
> On FS 1.5.15 same config, all is ok.
> --
> Best regards,
> Igor
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> 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 <javascript:;>
> 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.org  Tue Jul  7 18:37:32 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 7 Jul 2015 09:37:32 -0500
Subject: [Freeswitch-users] ClueCon 2015, Hotel Cutoff date is July 15th,
	Register NOW!
Message-ID: <CAEJMVkAWvYCn0OWx8-dKuwyTA-sArv2Y9U0G67tKozKMdwXB_g@mail.gmail.com>

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-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
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http://www.freeswitchcookbook.com

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Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit:
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*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From igorolhovskiy at gmail.com  Tue Jul  7 20:22:21 2015
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Tue, 7 Jul 2015 19:22:21 +0300
Subject: [Freeswitch-users] FS 1.7 / [CS_CONSUME_MEDIA]
	[INCOMPATIBLE_DESTINATION]
In-Reply-To: <CAEwTsAhbzM1zfz96qtvG9+MUr13w1XJqGX4AgetuN55mZCTsYg@mail.gmail.com>
References: <CAJTkRNvtZ4VFuRdASQBticwcgRszZGAzhgBqMjT79S3TS5-VcA@mail.gmail.com>
	<CAEwTsAhbzM1zfz96qtvG9+MUr13w1XJqGX4AgetuN55mZCTsYg@mail.gmail.com>
Message-ID: <CAJTkRNv3PoLdASAZWT6adT5oUAV_YpH9Kiscz0yD-zpH-BfcXw@mail.gmail.com>

Ok.... Will try to tell client to move on verto.... Never had much problems
with it.


-- 
Best regards,
Igor
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From igorolhovskiy at gmail.com  Tue Jul  7 21:18:02 2015
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Tue, 7 Jul 2015 20:18:02 +0300
Subject: [Freeswitch-users] calling intercept from event socket
In-Reply-To: <CAEVfcaYBqeTWreL5hp+3YsAXrDRk6L6Bejt90LcHSwiS7db0pA@mail.gmail.com>
References: <CAEVfcaaCKqme2wKKxaf=ct1eOwQU1Vrtj+eWFYpj0F07vQYKcg@mail.gmail.com>
	<CAJTkRNtZ7wtCkjS7nSfHmMs+y0Eh_sToqDhqwFVoAK9Vs7n3CQ@mail.gmail.com>
	<CAEVfcaYBqeTWreL5hp+3YsAXrDRk6L6Bejt90LcHSwiS7db0pA@mail.gmail.com>
Message-ID: <CAJTkRNte6pJfitXiH_bwh=+-GTrt8h0kKd3wibh5xSzRzjjqtA@mail.gmail.com>

As I've used it with nodejs, bindings works well.
Or may be separate this processes? Means one is listening, second is socket.

2015-07-07 11:59 GMT+03:00 Babak Yakhchali <babak.freeswitch at gmail.com>:

> when I'm playing moh I'm listening for events. I think if async is removed
> I can not do something like this? can I?
>
> On Tue, Jul 7, 2015 at 12:57 PM, Igor Olhovskiy <igorolhovskiy at gmail.com>
> wrote:
>
>> Try to remove async.
>>
>> 2015-07-07 11:21 GMT+03:00 Babak Yakhchali <babak.freeswitch at gmail.com>:
>>
>>> Hi
>>> I'm using event socket to do something like freeswitch callcenter
>>> module. members(aleg) call and are connected to socket application, this is
>>> my dialplan:
>>> <extension name="socket">
>>>        <condition field="destination_number" expression="^999$">
>>>      <action application="set" data="socket_resume=true"/>
>>>              <action application="socket" data="172.16.90.1:8084 async
>>> full"/>
>>>   <action application="info"/>
>>>        </condition>
>>>      </extension>
>>> these are executed using socket:
>>> connect
>>> myevents
>>> set park_after_bridge=true
>>> answer
>>> playback local_stream://moh
>>>
>>> now a new channel is originated to agent(bleg) and on answer I'm calling
>>> intercept on aleg from socket to bridge it to agent channel. if I
>>> set socket_resume=true in dialplan, after intercept both legs are stuck
>>> (dead silent). if I do not set socket_resume=true legs are bridged but
>>> execution continues through dialplan (I placed info to test this).
>>> problem is if  socket_resume=false and agent transfers the call to 999
>>> during the call (same socket app) call hangsup. but if socket_resume=true
>>> intercept is not working as expected but transfer is done. how can I solve
>>> this?
>>> thanks
>>>
>>>
>>> _________________________________________________________________________
>>> 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
>>>
>>
>>
>>
>> --
>> Best 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
>



-- 
Best regards,
Igor
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From victor.medina at cibersys.com  Tue Jul  7 22:28:36 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Tue, 7 Jul 2015 13:58:36 -0430
Subject: [Freeswitch-users] FS 1.7 / [CS_CONSUME_MEDIA]
	[INCOMPATIBLE_DESTINATION]
In-Reply-To: <CAJTkRNv3PoLdASAZWT6adT5oUAV_YpH9Kiscz0yD-zpH-BfcXw@mail.gmail.com>
References: <CAJTkRNvtZ4VFuRdASQBticwcgRszZGAzhgBqMjT79S3TS5-VcA@mail.gmail.com>
	<CAEwTsAhbzM1zfz96qtvG9+MUr13w1XJqGX4AgetuN55mZCTsYg@mail.gmail.com>
	<CAJTkRNv3PoLdASAZWT6adT5oUAV_YpH9Kiscz0yD-zpH-BfcXw@mail.gmail.com>
Message-ID: <CACXsapM1KMQCi+T6svkbH3NoSf30kinBy6yDxJ0fmrZrKVqVRQ@mail.gmail.com>

As Michael noted... it seems to be broken in several levels. You have a few
choices there if you still want to work with it.

.- By pass media entirely, deliver directly to clients, you'll lose all PBX
functionality thou, as explained in, try it in an extra channel. Calls
works.

https://freeswitch.org/confluence/display/FREESWITCH/Bypass+Media+Overview

.- Use SIP.JS seems a better, well maintained option. Its also a fork from
jsSip, has a migration guide that eases switching

2015-07-07 11:52 GMT-04:30 Igor Olhovskiy <igorolhovskiy at gmail.com>:

> Ok.... Will try to tell client to move on verto.... Never had much
> problems with it.
>
>
> --
> Best 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
>



-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From martyn at magiccow.co.uk  Wed Jul  8 01:58:39 2015
From: martyn at magiccow.co.uk (Martyn Davies)
Date: Tue, 7 Jul 2015 22:58:39 +0100
Subject: [Freeswitch-users] DTMF bind digit action not firing
Message-ID: <CA++4AS6oNiLCHHR+q4abM2kQrHkV2htr7fGPCQw05Tbp-rwvrg@mail.gmail.com>

I have an action set up in my conference call controls like so:

<action application="bind_digit_action"
data="confi,#*,exec:javascript,/usr/share/freeswitch/scripts/record-my-name.js"
/>

Which oddly works with some calls but not with others (all the calls are
coming via the same operator network - the SIP/RTP comes from a media
gateway somewhere in their network).

In the case of a good one, we see this in the console:

2015-07-07 14:41:08.597322 [DEBUG] switch_rtp.c:6039 RTP RECV DTMF #:480
2015-07-07 14:41:08.597322 [DEBUG] switch_channel.c:488 RECV DTMF #:480
2015-07-07 14:41:08.937325 [DEBUG] switch_rtp.c:6039 RTP RECV DTMF *:480
2015-07-07 14:41:08.937325 [DEBUG] switch_channel.c:488 RECV DTMF *:480
2015-07-07 14:41:08.937325 [DEBUG] mod_conference.c:4131 Execute app:
javascript, /usr/share/freeswitch/scripts/record-my-name.js

and in the case of a bad one:

2015-07-07 14:43:13.897322 [DEBUG] switch_rtp.c:6039 RTP RECV DTMF #:480
2015-07-07 14:43:13.897322 [DEBUG] switch_channel.c:488 RECV DTMF #:480
2015-07-07 14:43:14.757320 [DEBUG] switch_rtp.c:6039 RTP RECV DTMF *:480
2015-07-07 14:43:14.757320 [DEBUG] switch_channel.c:488 RECV DTMF *:480


In other words the DTMF are seen, but the bind action doesn't fire.  I
grabbed a trace of the RTP, and the two cases look indistinguishable, there
are start/stop events, with sensible volume and duration (in fact these
parameters are the same in the two cases).

Can anyone tell me what further diagnostics are available in Freeswitch to
tell why the DTMF don't fire the bind action?


Regards,

Martyn
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From alexeymelnichuck at gmail.com  Wed Jul  8 13:48:01 2015
From: alexeymelnichuck at gmail.com (Alexey Melnichuk)
Date: Wed, 8 Jul 2015 09:48:01 +0000 (UTC)
Subject: [Freeswitch-users] Fax problem
Message-ID: <loom.20150708T111911-436@post.gmane.org>

FreeSWITCH Version 1.4.20+git~20150703T164215Z~b95362f965~32bit (git b95362f 
2015-07-03 16:42:15Z 32bit)

I think I found bug in spandsp.

`t4_image_put_handler_t` defined as (__cdecl)

typedef int (*t4_image_put_handler_t)(void *user_data, const uint8_t buf[], 
size_t len);

But `tXX_decode_put` functions defined as `__stdcall`.
So when I do

s->image_put_handler = (t4_image_put_handler_t) t4_t6_decode_put;
...
s->image_put_handler(...)

I got AV.

After I declare t4_image_put_handler_t as __stdcall problem solved.
(Alse need update declaretion to `pre_encoded_put` function)

But I start getting heap corruption on
`span_free((char *) s->tiff.file);`

Problem is that it uses `strdup` function with different allocator.
So I just copy function to `alloc.c`

SPAN_DECLARE(char *) span_strdup(const char *str)
{
    char *sdup;
    size_t len = strlen(str) + 1;

    sdup = (char *) span_alloc(len);
    memcpy(sdup, str, len);

    return sdup;
}

And use it as
ds->tiff.file = span_strdup(file);




From jurij.ivo at gmail.com  Wed Jul  8 14:03:42 2015
From: jurij.ivo at gmail.com (Jurijs Ivolga)
Date: Wed, 8 Jul 2015 13:03:42 +0300
Subject: [Freeswitch-users] FreeSWITCH 1.6 Video + streaming
Message-ID: <CAGYpU65JQ9-eO3Ojwk=u7S4O5Zbkwrxtq-MqBw6cfQ2EtxYNiQ@mail.gmail.com>

Hi,

I installed Freeswitch as described here:

https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video

Additionally I complied and loaded mod_vlc.

Now I need to stream video call.

I read wiki documentation for module mod_vlc and I was trying following
config for streaming:

<action application="set" data="record_concat_video=true"/>
<action application="record" data="vlc://#standard{access=http,mux=raw,dst=
198.168.0.100:8080/bla}"/>

With such config call is failing.

Nevertheless I'm able to record video of call with such config:

<action application="set" data="record_concat_video=true"/>
<action application="set" data="execute_on_answer=record_session
$${recordings_dir}/${strftime(%Y%m%d%H%M%S)}_${caller_id_number}.mp4"/>

Maybe somebody know how is possible to stream video call?

Any help will be appreciated!

Thank you!
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From stefano.favaro at edistar.com  Wed Jul  8 14:19:40 2015
From: stefano.favaro at edistar.com (stefano)
Date: Wed, 8 Jul 2015 03:19:40 -0700 (MST)
Subject: [Freeswitch-users] no audio with bypass_media_after_bridge
Message-ID: <1436350780327-7596174.post@n2.nabble.com>

Hello,

I'm trying to use "bypass_media_after_bridge" in my dialplan.
In this scenario I have 
a) softphone client using g729
b) FS version: FreeSWITCH Version 1.5.13b~64bit ( 64bit)
c) Dialogic HMP 3.0 (g729 + g711)

I'd like to use FS in passthrough mode so calls from a) will be rerouted to
c).

If I use the "bypass_media_after_bridge" signalling of calls is ok but i
have no audio on a).

If I use "bypass_media" audio is ok but I don't have the answer 200 OK back
to my softphone a)

This is my dialplan:

<extension name="calls_to_ivr">
      <condition field="destination_number" expression="^9999">
        <action application="export" data="jitterbuffer_msec=60:200:40"/>
        <action application="set" data="bypass_media_after_bridge=true"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="ignore_early_media=true"/>
        <action application="set" data="effective_caller_id_number=1234"/>
        <action application="bridge"
data="{execute_on_answer=start_dtmf_generate}sofia/internal/9000 at 10.1.1.1"/>
        <action application="pre_answer"/>
      </condition>
</extension>

In my sip profile the "inbound-late-negotiation" param is set to true.

Is there any mistake in my dialplan?

Thanks,

Stefano F.




--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/no-audio-with-bypass-media-after-bridge-tp7596174.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From alexeymelnichuck at gmail.com  Wed Jul  8 14:57:08 2015
From: alexeymelnichuck at gmail.com (Alexey Melnichuk)
Date: Wed, 8 Jul 2015 10:57:08 +0000 (UTC)
Subject: [Freeswitch-users] Fax problem
Message-ID: <loom.20150708T124042-729@post.gmane.org>

FreeSWITCH Version 1.4.20+git~20150703T164215Z~b95362f965~32bit (git b95362f 
2015-07-03 16:42:15Z 32bit)

I think I found bug in spandsp.

`t4_image_put_handler_t` defined as (__cdecl)

typedef int (*t4_image_put_handler_t)(void *user_data, const uint8_t buf[], 
size_t len);

But `tXX_decode_put` functions defined as `__stdcall`.
So when I do

s->image_put_handler = (t4_image_put_handler_t) t4_t6_decode_put;
...
s->image_put_handler(...)

I got AV.

After I declare t4_image_put_handler_t as __stdcall problem solved.
(Alse need update declaretion to `pre_encoded_put` function)

But I start getting heap corruption on
`span_free((char *) s->tiff.file);`

Problem is that it uses `strdup` function with different allocator.
So I just copy function to `alloc.c`

SPAN_DECLARE(char *) span_strdup(const char *str)
{
    char *sdup;
    size_t len = strlen(str) + 1;

    sdup = (char *) span_alloc(len);
    memcpy(sdup, str, len);

    return sdup;
}

And use it as
ds->tiff.file = span_strdup(file);




From gmaruzz at gmail.com  Wed Jul  8 16:31:36 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 8 Jul 2015 14:31:36 +0200
Subject: [Freeswitch-users] no audio with bypass_media_after_bridge
In-Reply-To: <1436350780327-7596174.post@n2.nabble.com>
References: <1436350780327-7596174.post@n2.nabble.com>
Message-ID: <CALXCt0r-z1uC_c1yREFWh4OdfPVTmj0YnSSFCSbX+rSAdP=LVA@mail.gmail.com>

Hi Stefano,

just to check the simplest of cases, use an extension like the following,
and get a SIP trace (eg: from fs_cli: sofia global siptrace on)


<extension name="calls_to_ivr">
      <condition field="destination_number" expression="^9999">
        <action application="set" data="bypass_media=true"/>
        <action application="set" data="ignore_early_media=true"/>
        <action application="bridge" data="sofia/internal/9000 at 10.1.1.1"/>
      </condition>
</extension>

On Wed, Jul 8, 2015 at 12:19 PM, stefano <stefano.favaro at edistar.com> wrote:

> Hello,
>
> I'm trying to use "bypass_media_after_bridge" in my dialplan.
> In this scenario I have
> a) softphone client using g729
> b) FS version: FreeSWITCH Version 1.5.13b~64bit ( 64bit)
> c) Dialogic HMP 3.0 (g729 + g711)
>
> I'd like to use FS in passthrough mode so calls from a) will be rerouted to
> c).
>
> If I use the "bypass_media_after_bridge" signalling of calls is ok but i
> have no audio on a).
>
> If I use "bypass_media" audio is ok but I don't have the answer 200 OK back
> to my softphone a)
>
> This is my dialplan:
>
> <extension name="calls_to_ivr">
>       <condition field="destination_number" expression="^9999">
>         <action application="export" data="jitterbuffer_msec=60:200:40"/>
>         <action application="set" data="bypass_media_after_bridge=true"/>
>         <action application="set" data="hangup_after_bridge=true"/>
>         <action application="set" data="ignore_early_media=true"/>
>         <action application="set" data="effective_caller_id_number=1234"/>
>         <action application="bridge"
> data="{execute_on_answer=start_dtmf_generate}sofia/internal/9000 at 10.1.1.1
> "/>
>         <action application="pre_answer"/>
>       </condition>
> </extension>
>
> In my sip profile the "inbound-late-negotiation" param is set to true.
>
> Is there any mistake in my dialplan?
>
> Thanks,
>
> Stefano F.
>
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/no-audio-with-bypass-media-after-bridge-tp7596174.html
> Sent from the freeswitch-users mailing list archive at 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
> 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
Cell : +39-347-2665618
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From vishal.sharma at knowlarity.com  Wed Jul  8 17:11:50 2015
From: vishal.sharma at knowlarity.com (Vishal Sharma)
Date: Wed, 8 Jul 2015 18:41:50 +0530
Subject: [Freeswitch-users] No Bridge Media
In-Reply-To: <CAEwTsAggC3+rCC3a=HcYqSwVMkA2rNr7-Eq-X5qofNES6NzCDA@mail.gmail.com>
References: <CAEaAqfj3DJ384pRxkc3h82ik-0a23TQA76E75AUOQFNVkKLP+g@mail.gmail.com>
	<CAEwTsAggC3+rCC3a=HcYqSwVMkA2rNr7-Eq-X5qofNES6NzCDA@mail.gmail.com>
Message-ID: <CAEaAqfhP8o82PxhaqRsqL6wuSDwxROQTvaNf1rCyyH8-K+KmTA@mail.gmail.com>

@lakindia89 at gmail.com,
 I tried command ... it doesn't work ....
@mike at jerris.com
I am receiving call on FS and control it using ESL, from esl i am bridging
incoming call to another number, I want early media palyed by this number's
telco to reach my incoming call but bridge_early_media=true desn't work 100
% of time.
Earlier I used to think it's a telco issue and telco doesn't provide the
early media but on fs_cli I can see call does get early media, but FS
doesn'tforward it to incoming call (in ss_cli case first call )

Thanks a lot

Vishal Sharma


On Mon, Jul 6, 2015 at 9:10 PM, Michael Jerris <mike at jerris.com> wrote:

> what are you trying to accomplish by this config?
>
> On Monday, July 6, 2015, Vishal Sharma <vishal.sharma at knowlarity.com>
> wrote:
>
>> Hi,
>> When I use following commnd on fs_cli, I can hear ring sound or any
>> message telecome provider play on destination number
>>
>> originate freetdm/outgoing/r/09899790092
>> &bridge(freetdm/outgoing/r/09899488723)
>>
>> but when I use one of following commands
>>
>> freeswitch at internal> originate
>> {bridge_early_media=true}freetdm/outgoing/r/09899790092 &bridge
>> (freetdm/outgoing/r/09899488723)
>>
>>
>> freeswitch at internal> originate
>> {bridge_early_media=true}freetdm/outgoing/r/09899790092
>> &bridge({bridge_early_media=true}freetdm/outgoing/r/09899488723)
>>
>> or when I bridge a call which is received on FS, I don't get bridge media.
>>
>> Is there any perticular config required, or it's bug..
>>
>>
>> Regards,
>> Vishal Sharma
>>
>> SuperReceptionist is now available on Android mobiles. Track your
>> business on the go with call analytics, recordings, insights and more: Download
>> the app here
>> <https://play.google.com/store/apps/details?id=com.superreceptionist>
>
>
> _________________________________________________________________________
> 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
>

-- 
SuperReceptionist is now available on Android mobiles. Track your business 
on the go with call analytics, recordings, insights and more: Download the 
app here 
<https://play.google.com/store/apps/details?id=com.superreceptionist>
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From nboric at yx.cl  Wed Jul  8 17:23:45 2015
From: nboric at yx.cl (Neven Boric)
Date: Wed, 8 Jul 2015 10:23:45 -0300
Subject: [Freeswitch-users] Some problems with bypass-media
Message-ID: <CAHQ=dcK0bfZbZ30zt9nYUH1z-R6BnSZBM1W0kAHommJspLKftg@mail.gmail.com>

Hi,

I have been using FS for a long time on a local server without issues, but
now I want to move to a remote server to support some new usage scenarios.
I'm trying to use inbound-bypass-media=true to keep the audio out of the
server. This mostly works, but I have two different issues, depending on
which FS version I use:

- I first tried with the 1.2 branch, as that was the version I was using
locally. The issue with that version is that when a phone holds and then
unholds a call, I get no audio on the phone that started the hold. I found
option bypass-media-after-hold, but realized that it was not included in
the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93 from
the master branch. Now FS correctly tries to restore direct media between
the endpoints with reINVITEs, but unfortunately includes a 'sendonly' in
the last SDP to the phone that unholds, so I still get no audio. I haven't
found a way to fix this.

- I also tried with the 1.4 branch and hold/unhold works correctly, but now
attended transfer doesn't work, FS after some time ends the call on the
side that was put on hold waiting to be transferred. This time seems to be
inconsistent, and sometimes the transfer actually works, so maybe it is a
timing issue. This seems similar/related to FS-4038.

I can get logs and SIP captures, what would be the preferred way to provide
them (pcap, inline text, pastebin)? Or maybe it's better to go ahead and
file a bug?

Best regards
Neven Boric
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From gmaruzz at gmail.com  Wed Jul  8 18:03:51 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 8 Jul 2015 16:03:51 +0200
Subject: [Freeswitch-users] Some problems with bypass-media
In-Reply-To: <CAHQ=dcK0bfZbZ30zt9nYUH1z-R6BnSZBM1W0kAHommJspLKftg@mail.gmail.com>
References: <CAHQ=dcK0bfZbZ30zt9nYUH1z-R6BnSZBM1W0kAHommJspLKftg@mail.gmail.com>
Message-ID: <CALXCt0p4OYuJtLEYKxSv6=0aoqRdOLzAAquS3uctMoDxX959UA@mail.gmail.com>

use 1.4.x latest (1.4.20), traces in pcap and pastebin


On Wed, Jul 8, 2015 at 3:23 PM, Neven Boric <nboric at yx.cl> wrote:

> Hi,
>
> I have been using FS for a long time on a local server without issues, but
> now I want to move to a remote server to support some new usage scenarios.
> I'm trying to use inbound-bypass-media=true to keep the audio out of the
> server. This mostly works, but I have two different issues, depending on
> which FS version I use:
>
> - I first tried with the 1.2 branch, as that was the version I was using
> locally. The issue with that version is that when a phone holds and then
> unholds a call, I get no audio on the phone that started the hold. I found
> option bypass-media-after-hold, but realized that it was not included in
> the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93 from
> the master branch. Now FS correctly tries to restore direct media between
> the endpoints with reINVITEs, but unfortunately includes a 'sendonly' in
> the last SDP to the phone that unholds, so I still get no audio. I haven't
> found a way to fix this.
>
> - I also tried with the 1.4 branch and hold/unhold works correctly, but
> now attended transfer doesn't work, FS after some time ends the call on the
> side that was put on hold waiting to be transferred. This time seems to be
> inconsistent, and sometimes the transfer actually works, so maybe it is a
> timing issue. This seems similar/related to FS-4038.
>
> I can get logs and SIP captures, what would be the preferred way to
> provide them (pcap, inline text, pastebin)? Or maybe it's better to go
> ahead and file a bug?
>
> Best regards
> Neven Boric
>
> _________________________________________________________________________
> 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
Cell : +39-347-2665618
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From aronp at guaranteedplus.com  Wed Jul  8 20:12:28 2015
From: aronp at guaranteedplus.com (Podrigal, Aron)
Date: Wed, 8 Jul 2015 12:12:28 -0400
Subject: [Freeswitch-users] Fwd: sofia_reg_handle_register_token - exp
	possibly always false
In-Reply-To: <CANJp-yhS4260A26B=Y7cC=MFi4X_m5k1PTh3u3BFFJLC6wsWsg@mail.gmail.com>
References: <CANJp-yhS4260A26B=Y7cC=MFi4X_m5k1PTh3u3BFFJLC6wsWsg@mail.gmail.com>
Message-ID: <CANJp-yiauO6q-bsWEbcPvWZrsD3LHG+0PN9fU7sjCUifCJ2R1A@mail.gmail.com>

---------- Forwarded message ----------
From: Podrigal, Aron <aronp at guaranteedplus.com>
Date: Tue, Jul 7, 2015 at 7:24 PM
Subject: sofia_reg_handle_register_token - exp possibly always false
To: freeswitch-dev <freeswitch-dev at lists.freeswitch.org>


Hi,

According to grep the function sofia_reg_handle_register_token is called
from a single place src/mod/endpoints/mod_sofia/sofia_reg.c:2305
<https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/endpoints/mod_sofia/sofia_reg.c#2305>
where
the argument *v_event* would always be NULL at that time. So I'm wondering
what the line at src/mod/endpoints/mod_sofia/sofia_reg.c:1297
<https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/endpoints/mod_sofia/sofia_reg.c#1297>
  if (v_event && *v_event) pres_on_reg = switch_event_get_header(*v_event,
"send-presence-on-register"); is suppose to do, won't this always evaluate
to false?

-- 
Aron Podrigal
-
//Be happy :-)



-- 
Aron Podrigal
-
//Be happy :-)
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From nboric at yx.cl  Wed Jul  8 20:10:13 2015
From: nboric at yx.cl (Neven Boric)
Date: Wed, 8 Jul 2015 13:10:13 -0300
Subject: [Freeswitch-users] Some problems with bypass-media
In-Reply-To: <CALXCt0p4OYuJtLEYKxSv6=0aoqRdOLzAAquS3uctMoDxX959UA@mail.gmail.com>
References: <CAHQ=dcK0bfZbZ30zt9nYUH1z-R6BnSZBM1W0kAHommJspLKftg@mail.gmail.com>
	<CALXCt0p4OYuJtLEYKxSv6=0aoqRdOLzAAquS3uctMoDxX959UA@mail.gmail.com>
Message-ID: <CAHQ=dcJjMr7fg+WiFc=cFhCuNKXo1KWvm6S0bu9RfrEarZdfqw@mail.gmail.com>

Thanks.

Here is a pcap and the pastebin is at:

https://pastebin.freeswitch.org/24299

I realized that if I wait about 10 seconds with the first call on hold, and
don't even attempt to transfer, FS will hangup both legs of the original
call. Also, there is an error before FS decides to hangup:

2015-07-08 15:54:36.617835 [CRIT] switch_core_io.c:173 sofia/internal/
140 at 10.50.100.16:5060 reading on a session with no media!

Appreciate your help.

Neven

On Wed, Jul 8, 2015 at 11:03 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> use 1.4.x latest (1.4.20), traces in pcap and pastebin
>
>
> On Wed, Jul 8, 2015 at 3:23 PM, Neven Boric <nboric at yx.cl> wrote:
>
>> Hi,
>>
>> I have been using FS for a long time on a local server without issues,
>> but now I want to move to a remote server to support some new usage
>> scenarios. I'm trying to use inbound-bypass-media=true to keep the audio
>> out of the server. This mostly works, but I have two different issues,
>> depending on which FS version I use:
>>
>> - I first tried with the 1.2 branch, as that was the version I was using
>> locally. The issue with that version is that when a phone holds and then
>> unholds a call, I get no audio on the phone that started the hold. I found
>> option bypass-media-after-hold, but realized that it was not included in
>> the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93 from
>> the master branch. Now FS correctly tries to restore direct media between
>> the endpoints with reINVITEs, but unfortunately includes a 'sendonly' in
>> the last SDP to the phone that unholds, so I still get no audio. I haven't
>> found a way to fix this.
>>
>> - I also tried with the 1.4 branch and hold/unhold works correctly, but
>> now attended transfer doesn't work, FS after some time ends the call on the
>> side that was put on hold waiting to be transferred. This time seems to be
>> inconsistent, and sometimes the transfer actually works, so maybe it is a
>> timing issue. This seems similar/related to FS-4038.
>>
>> I can get logs and SIP captures, what would be the preferred way to
>> provide them (pcap, inline text, pastebin)? Or maybe it's better to go
>> ahead and file a bug?
>>
>> Best regards
>> Neven Boric
>>
>> _________________________________________________________________________
>> 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
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> 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 steveayre at gmail.com  Wed Jul  8 21:17:12 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 8 Jul 2015 18:17:12 +0100
Subject: [Freeswitch-users] Some problems with bypass-media
In-Reply-To: <CAHQ=dcK0bfZbZ30zt9nYUH1z-R6BnSZBM1W0kAHommJspLKftg@mail.gmail.com>
References: <CAHQ=dcK0bfZbZ30zt9nYUH1z-R6BnSZBM1W0kAHommJspLKftg@mail.gmail.com>
Message-ID: <CAFiqYukicT1Af87M6K3KUn-EYTPGGq2qtjWLi2TeAB+=kNodmw@mail.gmail.com>

>
> - I first tried with the 1.2 branch, as that was the version I was using
> locally. The issue with that version is that when a phone holds and then
> unholds a call, I get no audio on the phone that started the hold. I found
> option bypass-media-after-hold, but realized that it was not included in
> the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93 from
> the master branch. Now FS correctly tries to restore direct media between
> the endpoints with reINVITEs, but unfortunately includes a 'sendonly' in
> the last SDP to the phone that unholds, so I still get no audio. I haven't
> found a way to fix this.


Which probably means there's other commits required. 1.2 is EOL and no
longer supported, 1.4 is the current stable release. I wouldn't spend any
time getting 1.2 working, and instead work on upgrading.

- I also tried with the 1.4 branch and hold/unhold works correctly, but now
> attended transfer doesn't work, FS after some time ends the call on the
> side that was put on hold waiting to be transferred. This time seems to be
> inconsistent, and sometimes the transfer actually works, so maybe it is a
> timing issue. This seems similar/related to FS-4038.
>

Also see if you can replicate it on master, as that's close to the upcoming
1.6 release and will have a lot of changes over 1.4.


On 8 July 2015 at 14:23, Neven Boric <nboric at yx.cl> wrote:

> Hi,
>
> I have been using FS for a long time on a local server without issues, but
> now I want to move to a remote server to support some new usage scenarios.
> I'm trying to use inbound-bypass-media=true to keep the audio out of the
> server. This mostly works, but I have two different issues, depending on
> which FS version I use:
>
> - I first tried with the 1.2 branch, as that was the version I was using
> locally. The issue with that version is that when a phone holds and then
> unholds a call, I get no audio on the phone that started the hold. I found
> option bypass-media-after-hold, but realized that it was not included in
> the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93 from
> the master branch. Now FS correctly tries to restore direct media between
> the endpoints with reINVITEs, but unfortunately includes a 'sendonly' in
> the last SDP to the phone that unholds, so I still get no audio. I haven't
> found a way to fix this.
>
> - I also tried with the 1.4 branch and hold/unhold works correctly, but
> now attended transfer doesn't work, FS after some time ends the call on the
> side that was put on hold waiting to be transferred. This time seems to be
> inconsistent, and sometimes the transfer actually works, so maybe it is a
> timing issue. This seems similar/related to FS-4038.
>
> I can get logs and SIP captures, what would be the preferred way to
> provide them (pcap, inline text, pastebin)? Or maybe it's better to go
> ahead and file a bug?
>
> Best regards
> Neven Boric
>
> _________________________________________________________________________
> 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 nboric at yx.cl  Wed Jul  8 22:41:01 2015
From: nboric at yx.cl (Neven Boric)
Date: Wed, 8 Jul 2015 15:41:01 -0300
Subject: [Freeswitch-users] Some problems with bypass-media
In-Reply-To: <CAFiqYukicT1Af87M6K3KUn-EYTPGGq2qtjWLi2TeAB+=kNodmw@mail.gmail.com>
References: <CAHQ=dcK0bfZbZ30zt9nYUH1z-R6BnSZBM1W0kAHommJspLKftg@mail.gmail.com>
	<CAFiqYukicT1Af87M6K3KUn-EYTPGGq2qtjWLi2TeAB+=kNodmw@mail.gmail.com>
Message-ID: <CAHQ=dcJ+JNsU720QdKMznpYxSs0AryYif-MQbswFu2H2VrBAgQ@mail.gmail.com>

On Wed, Jul 8, 2015 at 2:17 PM, Steven Ayre <steveayre at gmail.com> wrote:

> - I first tried with the 1.2 branch, as that was the version I was using
>> locally. The issue with that version is that when a phone holds and then
>> unholds a call, I get no audio on the phone that started the hold. I found
>> option bypass-media-after-hold, but realized that it was not included in
>> the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93 from
>> the master branch. Now FS correctly tries to restore direct media between
>> the endpoints with reINVITEs, but unfortunately includes a 'sendonly' in
>> the last SDP to the phone that unholds, so I still get no audio. I haven't
>> found a way to fix this.
>
>
> Which probably means there's other commits required. 1.2 is EOL and no
> longer supported, 1.4 is the current stable release. I wouldn't spend any
> time getting 1.2 working, and instead work on upgrading.
>

Yes, I know it's not currently supported, but maybe somebody had the same
problem and fixed it in a different way. Or maybe somebody could point me
in the right direction to remove that final 'sendonly'.


>
> - I also tried with the 1.4 branch and hold/unhold works correctly, but
>> now attended transfer doesn't work, FS after some time ends the call on the
>> side that was put on hold waiting to be transferred. This time seems to be
>> inconsistent, and sometimes the transfer actually works, so maybe it is a
>> timing issue. This seems similar/related to FS-4038.
>>
>
> Also see if you can replicate it on master, as that's close to the
> upcoming 1.6 release and will have a lot of changes over 1.4.
>

I will try with master and report back, thanks.


>
>
> On 8 July 2015 at 14:23, Neven Boric <nboric at yx.cl> wrote:
>
>> Hi,
>>
>> I have been using FS for a long time on a local server without issues,
>> but now I want to move to a remote server to support some new usage
>> scenarios. I'm trying to use inbound-bypass-media=true to keep the audio
>> out of the server. This mostly works, but I have two different issues,
>> depending on which FS version I use:
>>
>> - I first tried with the 1.2 branch, as that was the version I was using
>> locally. The issue with that version is that when a phone holds and then
>> unholds a call, I get no audio on the phone that started the hold. I found
>> option bypass-media-after-hold, but realized that it was not included in
>> the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93 from
>> the master branch. Now FS correctly tries to restore direct media between
>> the endpoints with reINVITEs, but unfortunately includes a 'sendonly' in
>> the last SDP to the phone that unholds, so I still get no audio. I haven't
>> found a way to fix this.
>>
>> - I also tried with the 1.4 branch and hold/unhold works correctly, but
>> now attended transfer doesn't work, FS after some time ends the call on the
>> side that was put on hold waiting to be transferred. This time seems to be
>> inconsistent, and sometimes the transfer actually works, so maybe it is a
>> timing issue. This seems similar/related to FS-4038.
>>
>> I can get logs and SIP captures, what would be the preferred way to
>> provide them (pcap, inline text, pastebin)? Or maybe it's better to go
>> ahead and file a bug?
>>
>> Best regards
>> Neven Boric
>>
>> _________________________________________________________________________
>> 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 alipey at gmail.com  Wed Jul  8 23:03:59 2015
From: alipey at gmail.com (Ali Pey)
Date: Wed, 8 Jul 2015 15:03:59 -0400
Subject: [Freeswitch-users] park_after_bridge still broken in 1.4.20
Message-ID: <CA+q4kRL39YtE4DsQG87rJPKrbiQyd7kBwQpgOxJeC7t5pCPvKw@mail.gmail.com>

Hello,

A fix went in 1.4.20 for this:

   - FS-7636 <https://freeswitch.org/jira/browse/FS-7636> Fixed an issue
   with transfer_after_bridge and park_after_bridge pre-empting transfers


However, park after bridge still doesn't work in 1.4.20. It only works if I
downgrade to 1.4.15.


Is there a bug open for this? Should I open a new bug or re-open FS-7636?

Regards,
Ali Pey
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From mike at jerris.com  Wed Jul  8 23:23:50 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 8 Jul 2015 15:23:50 -0400
Subject: [Freeswitch-users] park_after_bridge still broken in 1.4.20
In-Reply-To: <CA+q4kRL39YtE4DsQG87rJPKrbiQyd7kBwQpgOxJeC7t5pCPvKw@mail.gmail.com>
References: <CA+q4kRL39YtE4DsQG87rJPKrbiQyd7kBwQpgOxJeC7t5pCPvKw@mail.gmail.com>
Message-ID: <2D46B275-48BA-4775-9D67-AA2E6D7B3B86@jerris.com>

Lets open a new bug for this and troubleshoot further there.

> On Jul 8, 2015, at 3:03 PM, Ali Pey <alipey at gmail.com> wrote:
> 
> Hello,
> 
> A fix went in 1.4.20 for this:
> FS-7636 <https://freeswitch.org/jira/browse/FS-7636> Fixed an issue with transfer_after_bridge and park_after_bridge pre-empting transfers
> 
> However, park after bridge still doesn't work in 1.4.20. It only works if I downgrade to 1.4.15.
> 
> 
> Is there a bug open for this? Should I open a new bug or re-open FS-7636?
> 
> Regards,
> Ali Pey

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From brian at freeswitch.org  Thu Jul  9 01:04:02 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 8 Jul 2015 16:04:02 -0500
Subject: [Freeswitch-users] How to test a Pull Request! aka how to help us
	out!
Message-ID: <CAEJMVkBYRr=F63CbeZfN16Vg=7NAEHoVc7DqDE=3=-5XTKJ7yg@mail.gmail.com>

Checkout freeswitch master first!

git clone https://freeswitch.org/stash/scm/fs/freeswitch.git freeswitch.git

cd freeswitch.git

git config --add remote.origin.fetch
'+refs/pull-requests/*/from:refs/remotes/origin/pr/*'
git config alias.pr '!f() { git checkout pr/$1; git config
branch.pr/$1.merge refs/pull-requests/$1/from;} ; f'

git pr <PR_NUMBER>

build, test, report back.

You can view the list of pull requests here:

https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests

Questions? Comments?

Thanks,


-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
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From mrjoli021 at gmail.com  Thu Jul  9 04:34:27 2015
From: mrjoli021 at gmail.com (Joli Martinez)
Date: Wed, 8 Jul 2015 20:34:27 -0400
Subject: [Freeswitch-users] sip trunk with edgemarc
Message-ID: <0BCDF028-9BD8-481E-8602-C9F20D153953@gmail.com>

Hello,

I have an Edgemarc as an SBC and behind it is a Freeswitch pbx.  I have configured the edgemarc as scenario 1 on the link below.  So only does topology hiding but Freeswitch does the registration.  On the sip-profile external I have my xml file which points to the LAN side of the Edgemarc.

I am unable to register to the provider I get 503 service unvail.

https://edgewaternetworks.com/wp-content/uploads/2014/07/Edgewater_ConfigureEdgeMarcForSIPTrunking.pdf

Please help.

 

From andrew.keil at visytel.com  Thu Jul  9 05:27:04 2015
From: andrew.keil at visytel.com (Andrew Keil)
Date: Thu, 9 Jul 2015 01:27:04 +0000
Subject: [Freeswitch-users] Sangoma's FreeTDM installation and setup on
 CentOS 6.6 with FreeSWITCH 1.4.20
Message-ID: <SIXPR04MB12638E528DA6201014B128DF3900@SIXPR04MB126.apcprd04.prod.outlook.com>

To all,

There are some important bug fixes in Sangoma's FreeTDM and everyone should use the Sangoma updated version.  In order to fast-track the changes I have created a tar.gz file on my FTP site and a PDF document (see attached) to setup the latest production release of FreeSWITCH (version 1.4.20) on CentOS 6.6 with the Sangoma updated FreeTDM 1.9.0.  It may help anyone using FreeSWITCH with Sangoma hardware for the first time.

Sangoma hope to merge FreeTDM (with the fixes) with FreeSWITCH in August/September this year.  I will attempt to keep the PDF and FreeTDM tar.gz file up-to-date alongside future production releases of FreeSWITCH.

FYI: Sangoma support also have a copy of this PDF file and they are thankful for the efforts I have made to fast-track new FreeSWITCH users to use their hardware.

As with any instruction document, Visytel Pty Ltd accepts no liability or loss of earnings from following the attached PDF.

Kind Regards,

Andrew Keil
Visytel Pty Ltd

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From igorolhovskiy at gmail.com  Thu Jul  9 09:03:16 2015
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Thu, 9 Jul 2015 08:03:16 +0300
Subject: [Freeswitch-users] sip trunk with edgemarc
In-Reply-To: <0BCDF028-9BD8-481E-8602-C9F20D153953@gmail.com>
References: <0BCDF028-9BD8-481E-8602-C9F20D153953@gmail.com>
Message-ID: <CAJTkRNtVCscSKEcdAAG4Po-9vXPXX-Z4LNacZyeJ0S2QW+GN6g@mail.gmail.com>

Please point, that EM has a simple Asterisk inside it to make an SBC.
Also, make sure, that you have your provider IP address in SIP server list.
Pointing on LAN side of your EM is useful, when there is no registrations,
otherwise point at your provider directly from FS.
To make a full-hiding-topology scenario, please follow scenario C in your
document.

2015-07-09 3:34 GMT+03:00 Joli Martinez <mrjoli021 at gmail.com>:

> Hello,
>
> I have an Edgemarc as an SBC and behind it is a Freeswitch pbx.  I have
> configured the edgemarc as scenario 1 on the link below.  So only does
> topology hiding but Freeswitch does the registration.  On the sip-profile
> external I have my xml file which points to the LAN side of the Edgemarc.
>
> I am unable to register to the provider I get 503 service unvail.
>
>
> https://edgewaternetworks.com/wp-content/uploads/2014/07/Edgewater_ConfigureEdgeMarcForSIPTrunking.pdf
>
> Please help.
>
>
> _________________________________________________________________________
> 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
>



-- 
Best regards,
Igor
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From lakindia89 at gmail.com  Thu Jul  9 10:14:55 2015
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 9 Jul 2015 11:44:55 +0530
Subject: [Freeswitch-users] No Bridge Media
In-Reply-To: <CAEaAqfhP8o82PxhaqRsqL6wuSDwxROQTvaNf1rCyyH8-K+KmTA@mail.gmail.com>
References: <CAEaAqfj3DJ384pRxkc3h82ik-0a23TQA76E75AUOQFNVkKLP+g@mail.gmail.com>
	<CAEwTsAggC3+rCC3a=HcYqSwVMkA2rNr7-Eq-X5qofNES6NzCDA@mail.gmail.com>
	<CAEaAqfhP8o82PxhaqRsqL6wuSDwxROQTvaNf1rCyyH8-K+KmTA@mail.gmail.com>
Message-ID: <CAMx6C7fMSSM=xXEejquEX8gYqqFk0Lmx6EquQ+X-Fk56kccGLQ@mail.gmail.com>

Hi,

Just to narrow down the issue, dont go for ESL now. When you receive an
incoming call, use "set" and "bridge" application in the dialplan itself
and see what happens.. Post the dialplan and the freeswitch log...

Also which PRI card are you using ??

On Wed, Jul 8, 2015 at 6:41 PM, Vishal Sharma <vishal.sharma at knowlarity.com>
wrote:

> @lakindia89 at gmail.com,
>  I tried command ... it doesn't work ....
> @mike at jerris.com
> I am receiving call on FS and control it using ESL, from esl i am bridging
> incoming call to another number, I want early media palyed by this number's
> telco to reach my incoming call but bridge_early_media=true desn't work 100
> % of time.
> Earlier I used to think it's a telco issue and telco doesn't provide the
> early media but on fs_cli I can see call does get early media, but FS
> doesn'tforward it to incoming call (in ss_cli case first call )
>
> Thanks a lot
>
> Vishal Sharma
>
>
> On Mon, Jul 6, 2015 at 9:10 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> what are you trying to accomplish by this config?
>>
>> On Monday, July 6, 2015, Vishal Sharma <vishal.sharma at knowlarity.com>
>> wrote:
>>
>>> Hi,
>>> When I use following commnd on fs_cli, I can hear ring sound or any
>>> message telecome provider play on destination number
>>>
>>> originate freetdm/outgoing/r/09899790092
>>> &bridge(freetdm/outgoing/r/09899488723)
>>>
>>> but when I use one of following commands
>>>
>>> freeswitch at internal> originate
>>> {bridge_early_media=true}freetdm/outgoing/r/09899790092 &bridge
>>> (freetdm/outgoing/r/09899488723)
>>>
>>>
>>> freeswitch at internal> originate
>>> {bridge_early_media=true}freetdm/outgoing/r/09899790092
>>> &bridge({bridge_early_media=true}freetdm/outgoing/r/09899488723)
>>>
>>> or when I bridge a call which is received on FS, I don't get bridge
>>> media.
>>>
>>> Is there any perticular config required, or it's bug..
>>>
>>>
>>> Regards,
>>> Vishal Sharma
>>>
>>> SuperReceptionist is now available on Android mobiles. Track your
>>> business on the go with call analytics, recordings, insights and more: Download
>>> the app here
>>> <https://play.google.com/store/apps/details?id=com.superreceptionist>
>>
>>
>> _________________________________________________________________________
>> 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
>>
>
>
> SuperReceptionist is now available on Android mobiles. Track your business
> on the go with call analytics, recordings, insights and more: Download
> the app here
> <https://play.google.com/store/apps/details?id=com.superreceptionist>
>
> _________________________________________________________________________
> 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 jurij.ivo at gmail.com  Thu Jul  9 12:49:15 2015
From: jurij.ivo at gmail.com (Jurijs Ivolga)
Date: Thu, 9 Jul 2015 11:49:15 +0300
Subject: [Freeswitch-users] extra Bye message from Freeswitch
Message-ID: <CAGYpU67t+bh-mDAMEBE9W2Gcqa3YVzFRN+YKzTYUZEFF_fJgLQ@mail.gmail.com>

Hi!

I started to run some performance tests against Freeswitch and in 5-10%
tests are failing because of Freeswitch sending extra bye message. I would
like to add that I got such messages on very low load, 1 new call in 5
seconds and first fail was after 9th call.

Here how call looks like in 90-95% of time:

SIPp UAC            Freeswitch
    |(1) INVITE         |
    |------------------>|
    |(2) 100 (optional) |
    |<------------------|
    |(3) 200            |
    |<------------------|
    |(4) ACK            |
    |------------------>|
    |(5) BYE            |
    |------------------>|
    |(6) 200            |
    |<------------------|


Below you can see how call looks like in 5-10% of time.
Somehow Freeswitch replies on bye message from Sipp with extra bye(around
5-10% of all calls) and it creates a mess:

SIPp UAC            Freeswitch
    |(1) INVITE         |
    |------------------>|
    |(2) 100 (optional) |
    |<------------------|
    |(3) 200            |
    |<------------------|
    |(4) ACK            |
    |------------------>|
    |(5) BYE            |
    |------------------>|
    |(6) BYE            |
    |<------------------|
    |(7) 200            |
    |------------------>|
    |(8) 200            |
    |<------------------|

I'm not sure, but I think such behavior is not expected. Maybe somebody can
let me know if it is proper Freeswitch behavior and if it is, why such
behavior only happens with 5-10% of calls. Below you can find full sip
trace, for a call with extra bye message.


U 2015/07/09 11:07:07.269998 sippserver:5060 -> freeswitchserver:5060
INVITE sip:service at freeswitchserver:5060 SIP/2.0.
Via: SIP/2.0/UDP sippserver:5060;branch=z9hG4bK-11560-9-0.
From: sipp <sip:sipp at sippserver:5060>;tag=11560SIPpTag009.
To: service <sip:service at freeswitchserver:5060>.
Call-ID: 9-11560 at sippserver.
CSeq: 1 INVITE.
Contact: sip:sipp at sippserver:5060.
Max-Forwards: 70.
Subject: Performance Test.
Content-Type: application/sdp.
Content-Length:   139.
.
v=0.
o=user1 53655765 2353687637 IN IP4 sippserver.
s=-.
c=IN IP4 sippserver.
t=0 0.
m=audio 6000 RTP/AVP 0.
a=rtpmap:0 PCMU/8000.

#
U 2015/07/09 11:07:07.271005 freeswitchserver:5060 -> sippserver:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP sippserver:5060;branch=z9hG4bK-11560-9-0.
From: sipp <sip:sipp at sippserver:5060>;tag=11560SIPpTag009.
To: service <sip:service at freeswitchserver:5060>.
Call-ID: 9-11560 at sippserver.
CSeq: 1 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.4.18~64bit.
Content-Length: 0.
.

#
U 2015/07/09 11:07:07.276784 freeswitchserver:5060 -> sippserver:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sippserver:5060;branch=z9hG4bK-11560-9-0.
From: sipp <sip:sipp at sippserver:5060>;tag=11560SIPpTag009.
To: service <sip:service at freeswitchserver:5060>;tag=S3Kvy09a3BrtS.
Call-ID: 9-11560 at sippserver.
CSeq: 1 INVITE.
Contact: <sip:service at freeswitchserver:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.4.18~64bit.
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: 168.
Remote-Party-ID: "service" <sip:service at freeswitchserver
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1436408221 1436408222 IN IP4 freeswitchserver.
s=FreeSWITCH.
c=IN IP4 freeswitchserver.
t=0 0.
m=audio 21006 RTP/AVP 0.
a=rtpmap:0 PCMU/8000.
a=ptime:20.

#
U 2015/07/09 11:07:07.277349 sippserver:5060 -> freeswitchserver:5060
ACK sip:service at freeswitchserver:5060 SIP/2.0.
Via: SIP/2.0/UDP sippserver:5060;branch=z9hG4bK-11560-9-5.
From: sipp <sip:sipp at sippserver:5060>;tag=11560SIPpTag009.
To: service <sip:service at freeswitchserver:5060>;tag=S3Kvy09a3BrtS.
Call-ID: 9-11560 at sippserver.
CSeq: 1 ACK.
Contact: sip:sipp at sippserver:5060.
Max-Forwards: 70.
Subject: Performance Test.
Content-Length: 0.
.

#
U 2015/07/09 11:07:07.278861 sippserver:5060 -> freeswitchserver:5060
BYE sip:service at freeswitchserver:5060 SIP/2.0.
Via: SIP/2.0/UDP sippserver:5060;branch=z9hG4bK-11560-9-7.
From: sipp <sip:sipp at sippserver:5060>;tag=11560SIPpTag009.
To: service <sip:service at freeswitchserver:5060>;tag=S3Kvy09a3BrtS.
Call-ID: 9-11560 at sippserver.
CSeq: 2 BYE.
Contact: sip:sipp at sippserver:5060.
Max-Forwards: 70.
Subject: Performance Test.
Content-Length: 0.
.

#
U 2015/07/09 11:07:07.278931 freeswitchserver:5060 -> sippserver:5060
BYE sip:sipp at sippserver:5060 SIP/2.0.
Via: SIP/2.0/UDP freeswitchserver;rport;branch=z9hG4bKF5jjXD7vrF3Qp.
Max-Forwards: 70.
From: service <sip:service at freeswitchserver:5060>;tag=S3Kvy09a3BrtS.
To: sipp <sip:sipp at sippserver:5060>;tag=11560SIPpTag009.
Call-ID: 9-11560 at sippserver.
CSeq: 77878549 BYE.
User-Agent: FreeSWITCH-mod_sofia/1.4.18~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Reason: Q.850;cause=16;text="NORMAL_CLEARING".
Content-Length: 0.
.

#
U 2015/07/09 11:07:07.279599 sippserver:5060 -> freeswitchserver:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP freeswitchserver;rport;branch=z9hG4bKF5jjXD7vrF3Qp.
From: service <sip:service at freeswitchserver:5060>;tag=S3Kvy09a3BrtS.
To: sipp <sip:sipp at sippserver:5060>;tag=11560SIPpTag009.
Call-ID: 9-11560 at sippserver.
CSeq: 77878549 BYE.
Contact: <sip:sippserver:5060;transport=UDP>.
Content-Length: 0.
.

#
U 2015/07/09 11:07:07.280338 freeswitchserver:5060 -> sippserver:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sippserver:5060;branch=z9hG4bK-11560-9-7.
From: sipp <sip:sipp at sippserver:5060>;tag=11560SIPpTag009.
To: service <sip:service at freeswitchserver:5060>;tag=S3Kvy09a3BrtS.
Call-ID: 9-11560 at sippserver.
CSeq: 2 BYE.
Content-Length: 0.
.

I will appreciate any help!

Thank you!
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From jurij.ivo at gmail.com  Thu Jul  9 14:28:09 2015
From: jurij.ivo at gmail.com (Jurijs Ivolga)
Date: Thu, 9 Jul 2015 13:28:09 +0300
Subject: [Freeswitch-users] extra Bye message from Freeswitch
In-Reply-To: <CAGYpU67t+bh-mDAMEBE9W2Gcqa3YVzFRN+YKzTYUZEFF_fJgLQ@mail.gmail.com>
References: <CAGYpU67t+bh-mDAMEBE9W2Gcqa3YVzFRN+YKzTYUZEFF_fJgLQ@mail.gmail.com>
Message-ID: <CAGYpU67Yb-hz-xjyrwxvZciOXbBUZYCppxjCg5c79e8U7b7Waw@mail.gmail.com>

Hi,

Issue is solved. :)

Problem was in my dialplan. Basically there was empty dialplan and when
call hit Freeswitch, Freeswitch immediately hanged up a call, cause there
was nothing to do, that why I got that extra bye message. In other cases
sipp hanged up a call before Freeswitch and this was that 90%-95%.

To fix this I just needed to add sleep dialplan tool to my dialplan.

2015-07-09 11:49 GMT+03:00 Jurijs Ivolga <jurij.ivo at gmail.com>:

> Hi!
>
> I started to run some performance tests against Freeswitch and in 5-10%
> tests are failing because of Freeswitch sending extra bye message. I would
> like to add that I got such messages on very low load, 1 new call in 5
> seconds and first fail was after 9th call.
>
> Here how call looks like in 90-95% of time:
>
> SIPp UAC            Freeswitch
>     |(1) INVITE         |
>     |------------------>|
>     |(2) 100 (optional) |
>     |<------------------|
>     |(3) 200            |
>     |<------------------|
>     |(4) ACK            |
>     |------------------>|
>     |(5) BYE            |
>     |------------------>|
>     |(6) 200            |
>     |<------------------|
>
>
> Below you can see how call looks like in 5-10% of time.
> Somehow Freeswitch replies on bye message from Sipp with extra bye(around
> 5-10% of all calls) and it creates a mess:
>
> SIPp UAC            Freeswitch
>     |(1) INVITE         |
>     |------------------>|
>     |(2) 100 (optional) |
>     |<------------------|
>     |(3) 200            |
>     |<------------------|
>     |(4) ACK            |
>     |------------------>|
>     |(5) BYE            |
>     |------------------>|
>     |(6) BYE            |
>     |<------------------|
>     |(7) 200            |
>     |------------------>|
>     |(8) 200            |
>     |<------------------|
>
> I'm not sure, but I think such behavior is not expected. Maybe somebody
> can let me know if it is proper Freeswitch behavior and if it is, why such
> behavior only happens with 5-10% of calls. Below you can find full sip
> trace, for a call with extra bye message.
>
>
> U 2015/07/09 11:07:07.269998 sippserver:5060 -> freeswitchserver:5060
> INVITE sip:service at freeswitchserver:5060 SIP/2.0.
> Via: SIP/2.0/UDP sippserver:5060;branch=z9hG4bK-11560-9-0.
> From: sipp <sip:sipp at sippserver:5060>;tag=11560SIPpTag009.
> To: service <sip:service at freeswitchserver:5060>.
> Call-ID: 9-11560 at sippserver.
> CSeq: 1 INVITE.
> Contact: sip:sipp at sippserver:5060.
> Max-Forwards: 70.
> Subject: Performance Test.
> Content-Type: application/sdp.
> Content-Length:   139.
> .
> v=0.
> o=user1 53655765 2353687637 IN IP4 sippserver.
> s=-.
> c=IN IP4 sippserver.
> t=0 0.
> m=audio 6000 RTP/AVP 0.
> a=rtpmap:0 PCMU/8000.
>
> #
> U 2015/07/09 11:07:07.271005 freeswitchserver:5060 -> sippserver:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP sippserver:5060;branch=z9hG4bK-11560-9-0.
> From: sipp <sip:sipp at sippserver:5060>;tag=11560SIPpTag009.
> To: service <sip:service at freeswitchserver:5060>.
> Call-ID: 9-11560 at sippserver.
> CSeq: 1 INVITE.
> User-Agent: FreeSWITCH-mod_sofia/1.4.18~64bit.
> Content-Length: 0.
> .
>
> #
> U 2015/07/09 11:07:07.276784 freeswitchserver:5060 -> sippserver:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP sippserver:5060;branch=z9hG4bK-11560-9-0.
> From: sipp <sip:sipp at sippserver:5060>;tag=11560SIPpTag009.
> To: service <sip:service at freeswitchserver:5060>;tag=S3Kvy09a3BrtS.
> Call-ID: 9-11560 at sippserver.
> CSeq: 1 INVITE.
> Contact: <sip:service at freeswitchserver:5060;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.4.18~64bit.
> 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: 168.
> Remote-Party-ID: "service" <sip:service at freeswitchserver
> >;party=calling;privacy=off;screen=no.
> .
> v=0.
> o=FreeSWITCH 1436408221 1436408222 IN IP4 freeswitchserver.
> s=FreeSWITCH.
> c=IN IP4 freeswitchserver.
> t=0 0.
> m=audio 21006 RTP/AVP 0.
> a=rtpmap:0 PCMU/8000.
> a=ptime:20.
>
> #
> U 2015/07/09 11:07:07.277349 sippserver:5060 -> freeswitchserver:5060
> ACK sip:service at freeswitchserver:5060 SIP/2.0.
> Via: SIP/2.0/UDP sippserver:5060;branch=z9hG4bK-11560-9-5.
> From: sipp <sip:sipp at sippserver:5060>;tag=11560SIPpTag009.
> To: service <sip:service at freeswitchserver:5060>;tag=S3Kvy09a3BrtS.
> Call-ID: 9-11560 at sippserver.
> CSeq: 1 ACK.
> Contact: sip:sipp at sippserver:5060.
> Max-Forwards: 70.
> Subject: Performance Test.
> Content-Length: 0.
> .
>
> #
> U 2015/07/09 11:07:07.278861 sippserver:5060 -> freeswitchserver:5060
> BYE sip:service at freeswitchserver:5060 SIP/2.0.
> Via: SIP/2.0/UDP sippserver:5060;branch=z9hG4bK-11560-9-7.
> From: sipp <sip:sipp at sippserver:5060>;tag=11560SIPpTag009.
> To: service <sip:service at freeswitchserver:5060>;tag=S3Kvy09a3BrtS.
> Call-ID: 9-11560 at sippserver.
> CSeq: 2 BYE.
> Contact: sip:sipp at sippserver:5060.
> Max-Forwards: 70.
> Subject: Performance Test.
> Content-Length: 0.
> .
>
> #
> U 2015/07/09 11:07:07.278931 freeswitchserver:5060 -> sippserver:5060
> BYE sip:sipp at sippserver:5060 SIP/2.0.
> Via: SIP/2.0/UDP freeswitchserver;rport;branch=z9hG4bKF5jjXD7vrF3Qp.
> Max-Forwards: 70.
> From: service <sip:service at freeswitchserver:5060>;tag=S3Kvy09a3BrtS.
> To: sipp <sip:sipp at sippserver:5060>;tag=11560SIPpTag009.
> Call-ID: 9-11560 at sippserver.
> CSeq: 77878549 BYE.
> User-Agent: FreeSWITCH-mod_sofia/1.4.18~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
> Content-Length: 0.
> .
>
> #
> U 2015/07/09 11:07:07.279599 sippserver:5060 -> freeswitchserver:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP freeswitchserver;rport;branch=z9hG4bKF5jjXD7vrF3Qp.
> From: service <sip:service at freeswitchserver:5060>;tag=S3Kvy09a3BrtS.
> To: sipp <sip:sipp at sippserver:5060>;tag=11560SIPpTag009.
> Call-ID: 9-11560 at sippserver.
> CSeq: 77878549 BYE.
> Contact: <sip:sippserver:5060;transport=UDP>.
> Content-Length: 0.
> .
>
> #
> U 2015/07/09 11:07:07.280338 freeswitchserver:5060 -> sippserver:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP sippserver:5060;branch=z9hG4bK-11560-9-7.
> From: sipp <sip:sipp at sippserver:5060>;tag=11560SIPpTag009.
> To: service <sip:service at freeswitchserver:5060>;tag=S3Kvy09a3BrtS.
> Call-ID: 9-11560 at sippserver.
> CSeq: 2 BYE.
> Content-Length: 0.
> .
>
> I will appreciate any help!
>
> Thank you!
>
>
>
>
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From igorolhovskiy at gmail.com  Thu Jul  9 15:47:47 2015
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Thu, 9 Jul 2015 14:47:47 +0300
Subject: [Freeswitch-users] Eavesdrop a call.
Message-ID: <CAJTkRNvU3PQRHjFF1MxF1KsCq4L5fo4cZxBtNLFxiSLDvDzdwg@mail.gmail.com>

Hi!
I'm using Freeswitch 1.4.20 (git b95362f 2015-07-03 16:42:15Z 64bit))
(stable one)
and trying to get eavesdropping. But no luck.
Main scheme:
DID - > extension (210). 210 answers
eavesdropping from extension 110.
While giving a command
*eavesdrop <channel-uuid> *
I've got log full of messages
*2015-07-09 13:29:00.325667 [ERR] switch_core_io.c:1531 Write Buffer 0
bytes Failed!*
and silence on 110 extension

While giving a command
*userspy 210@${domain_name}* or *userspy <channel-uuid>*
I've got music-on-hold on 110, no errors on log.

In both cases used <channel-uuid> of extension 210 (answered one)

Is this a bug or I'm doing something wrong?
-- 
Best regards,
Igor
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From Alexander.Haugg at c4b.de  Thu Jul  9 15:57:01 2015
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Thu, 9 Jul 2015 11:57:01 +0000
Subject: [Freeswitch-users] Update To and From Header of the legs after
 an uuid_bridge
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC8517C1222@vc4bex2.corp.c4b.de>

I had test it again and it doesn?t work.
I added the property ?<param name="send-display-update" value="true"/>? to the sip profile with the same result.

I try it in the dialplan with ?<action application="send_display" data="servus|123" /> ? without success
Is there a possibility to force a send SIP Update as API or dialplan APP command?
What can I do?

Here is the uuid_bridge CLI output.

freeswitch at WIN-OHIC1AKL8UU> uuid_bridge 302eba5831664c8a e627aa5a996b07ca

+OK e627aa5a996b07ca

2015-07-09 13:35:31.922297 [DEBUG] switch_ivr_bridge.c:1873 (sofia/H3KSip/+4989840798170 at 172.16.1.26) State Change CS_EXECUTE -> CS_HIBERNATE
2015-07-09 13:35:31.922297 [DEBUG] switch_core_session.c:1388 Send signal sofia/H3KSip/+4989840798170 at 172.16.1.26 [BREAK]
2015-07-09 13:35:31.922297 [DEBUG] switch_ivr_bridge.c:1875 (sofia/H3KSip/+4985047079741 at 172.16.1.26) State Change CS_EXECUTE -> CS_HIBERNATE
2015-07-09 13:35:31.922297 [DEBUG] switch_core_session.c:1388 Send signal sofia/H3KSip/+4985047079741 at 172.16.1.26 [BREAK]
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:535 (sofia/H3KSip/+4989840798170 at 172.16.1.26) State EXECUTE going to sleep
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:472 (sofia/H3KSip/+4989840798170 at 172.16.1.26) Running State Change CS_HIBERNATE
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:550 (sofia/H3KSip/+4989840798170 at 172.16.1.26) State HIBERNATE
2015-07-09 13:35:31.942298 [DEBUG] mod_sofia.c:160 sofia/H3KSip/+4989840798170 at 172.16.1.26 SOFIA HIBERNATE
2015-07-09 13:35:31.942298 [DEBUG] switch_ivr_bridge.c:835 (sofia/H3KSip/+4989840798170 at 172.16.1.26) State Change CS_HIBERNATE -> CS_RESET
2015-07-09 13:35:31.942298 [DEBUG] switch_core_session.c:1388 Send signal sofia/H3KSip/+4989840798170 at 172.16.1.26 [BREAK]
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:550 (sofia/H3KSip/+4989840798170 at 172.16.1.26) State HIBERNATE going to sleep
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:472 (sofia/H3KSip/+4989840798170 at 172.16.1.26) Running State Change CS_RESET
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:531 (sofia/H3KSip/+4989840798170 at 172.16.1.26) State RESET
2015-07-09 13:35:31.942298 [DEBUG] mod_sofia.c:141 sofia/H3KSip/+4989840798170 at 172.16.1.26 SOFIA RESET
2015-07-09 13:35:31.942298 [DEBUG] switch_ivr_bridge.c:820 sofia/H3KSip/+4989840798170 at 172.16.1.26 CUSTOM RESET
2015-07-09 13:35:31.942298 [DEBUG] switch_ivr_bridge.c:827 (sofia/H3KSip/+4989840798170 at 172.16.1.26) State Change CS_RESET -> CS_SOFT_EXECUTE
2015-07-09 13:35:31.942298 [DEBUG] switch_core_session.c:1388 Send signal sofia/H3KSip/+4989840798170 at 172.16.1.26 [BREAK]
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:531 (sofia/H3KSip/+4989840798170 at 172.16.1.26) State RESET going to sleep
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:472 (sofia/H3KSip/+4989840798170 at 172.16.1.26) Running State Change CS_SOFT_EXECUTE
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:541 (sofia/H3KSip/+4989840798170 at 172.16.1.26) State SOFT_EXECUTE
2015-07-09 13:35:31.942298 [DEBUG] mod_sofia.c:600 SOFIA SOFT_EXECUTE
2015-07-09 13:35:31.942298 [DEBUG] switch_ivr_bridge.c:845 sofia/H3KSip/+4989840798170 at 172.16.1.26 CUSTOM SOFT_EXECUTE
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:535 (sofia/H3KSip/+4985047079741 at 172.16.1.26) State EXECUTE going to sleep
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:472 (sofia/H3KSip/+4985047079741 at 172.16.1.26) Running State Change CS_HIBERNATE
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:550 (sofia/H3KSip/+4985047079741 at 172.16.1.26) State HIBERNATE
2015-07-09 13:35:31.942298 [DEBUG] mod_sofia.c:160 sofia/H3KSip/+4985047079741 at 172.16.1.26 SOFIA HIBERNATE
2015-07-09 13:35:31.942298 [DEBUG] switch_ivr_bridge.c:835 (sofia/H3KSip/+4985047079741 at 172.16.1.26) State Change CS_HIBERNATE -> CS_RESET
2015-07-09 13:35:31.942298 [DEBUG] switch_core_session.c:1388 Send signal sofia/H3KSip/+4985047079741 at 172.16.1.26 [BREAK]
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:550 (sofia/H3KSip/+4985047079741 at 172.16.1.26) State HIBERNATE going to sleep
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:472 (sofia/H3KSip/+4985047079741 at 172.16.1.26) Running State Change CS_RESET
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:531 (sofia/H3KSip/+4985047079741 at 172.16.1.26) State RESET
2015-07-09 13:35:31.942298 [DEBUG] mod_sofia.c:141 sofia/H3KSip/+4985047079741 at 172.16.1.26 SOFIA RESET
2015-07-09 13:35:31.942298 [DEBUG] switch_ivr_bridge.c:820 sofia/H3KSip/+4985047079741 at 172.16.1.26 CUSTOM RESET
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:118 sofia/H3KSip/+4985047079741 at 172.16.1.26 Standard RESET
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:531 (sofia/H3KSip/+4985047079741 at 172.16.1.26) State RESET going to sleep
2015-07-09 13:35:31.962299 [DEBUG] switch_ivr_bridge.c:877 (sofia/H3KSip/+4985047079741 at 172.16.1.26) State Change CS_RESET -> CS_SOFT_EXECUTE
2015-07-09 13:35:31.962299 [DEBUG] switch_core_session.c:1388 Send signal sofia/H3KSip/+4985047079741 at 172.16.1.26 [BREAK]
2015-07-09 13:35:31.962299 [DEBUG] switch_core_state_machine.c:472 (sofia/H3KSip/+4985047079741 at 172.16.1.26) Running State Change CS_SOFT_EXECUTE
2015-07-09 13:35:31.962299 [DEBUG] switch_core_state_machine.c:541 (sofia/H3KSip/+4985047079741 at 172.16.1.26) State SOFT_EXECUTE
2015-07-09 13:35:31.962299 [DEBUG] mod_sofia.c:600 SOFIA SOFT_EXECUTE
2015-07-09 13:35:31.962299 [DEBUG] switch_ivr_bridge.c:845 sofia/H3KSip/+4985047079741 at 172.16.1.26 CUSTOM SOFT_EXECUTE
2015-07-09 13:35:31.962299 [DEBUG] switch_core_state_machine.c:330 sofia/H3KSip/+4985047079741 at 172.16.1.26 Standard SOFT_EXECUTE
2015-07-09 13:35:31.962299 [DEBUG] switch_core_state_machine.c:541 (sofia/H3KSip/+4985047079741 at 172.16.1.26) State SOFT_EXECUTE going to sleep
2015-07-09 13:35:31.982300 [DEBUG] switch_ivr_bridge.c:1360 (sofia/H3KSip/+4985047079741 at 172.16.1.26) State Change CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
2015-07-09 13:35:31.982300 [DEBUG] switch_core_session.c:1388 Send signal sofia/H3KSip/+4985047079741 at 172.16.1.26 [BREAK]
2015-07-09 13:35:31.982300 [DEBUG] switch_core_session.c:908 Send signal sofia/H3KSip/+4985047079741 at 172.16.1.26 [BREAK]
2015-07-09 13:35:31.982300 [DEBUG] switch_core_session.c:908 Send signal sofia/H3KSip/+4989840798170 at 172.16.1.26 [BREAK]
2015-07-09 13:35:31.982300 [DEBUG] switch_ivr_bridge.c:1465 (sofia/H3KSip/+4985047079741 at 172.16.1.26) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2015-07-09 13:35:31.982300 [DEBUG] switch_core_session.c:1388 Send signal sofia/H3KSip/+4985047079741 at 172.16.1.26 [BREAK]
2015-07-09 13:35:31.982300 [DEBUG] switch_core_state_machine.c:472 (sofia/H3KSip/+4985047079741 at 172.16.1.26) Running State Change CS_EXCHANGE_MEDIA
2015-07-09 13:35:31.982300 [DEBUG] switch_core_state_machine.c:538 (sofia/H3KSip/+4985047079741 at 172.16.1.26) State EXCHANGE_MEDIA
2015-07-09 13:35:31.982300 [DEBUG] mod_sofia.c:594 SOFIA EXCHANGE_MEDIA
2015-07-09 13:35:31.982300 [DEBUG] switch_core_session.c:970 Send signal sofia/H3KSip/+4985047079741 at 172.16.1.26 [BREAK]
2015-07-09 13:35:31.982300 [DEBUG] switch_core_session.c:970 Send signal sofia/H3KSip/+4989840798170 at 172.16.1.26 [BREAK]

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Jerris
Gesendet: Donnerstag, 25. Juni 2015 14:05
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Update To and From Header of the legs after an uuid_bridge

This happens automatically using a sip update packet unless you have disabled display updates

On Thursday, June 25, 2015, Alexander Haugg <Alexander.Haugg at c4b.de<mailto:Alexander.Haugg at c4b.de>> wrote:
Hi,

have anyone an answer?

More scenario information?s?

LegA connected with LegB
Now LegA bridged to (with uuid_bridge) LegC (LegB will terminated)

It is very important for me that LegA have the Partner Number of LegC and vice versa!
I need to transport this information?s back to a PBX via SIP Trunk!
What can I do for this?

Thanks a lot

Von: freeswitch-users-bounces at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');> [mailto:freeswitch-users-bounces at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>] Im Auftrag von Alexander Haugg
Gesendet: Donnerstag, 18. Juni 2015 11:16
An: freeswitch-users at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users at lists.freeswitch.org');>
Betreff: [Freeswitch-users] Update To and From Header of the legs after an uuid_bridge

Hi All,

What can i do to change (fix) the from- and the to- header on SIP side after an uuid_bridge (SIP UPDATE or re-INVITE)?

Thanks a lot!
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From brian at freeswitch.org  Thu Jul  9 16:52:38 2015
From: brian at freeswitch.org (Brian West)
Date: Thu, 9 Jul 2015 07:52:38 -0500
Subject: [Freeswitch-users] Fax problem
In-Reply-To: <loom.20150708T111911-436@post.gmane.org>
References: <loom.20150708T111911-436@post.gmane.org>
Message-ID: <CAEJMVkBrP3Uwhud0iNwEUhu87xhEY2iPem5V33vR9FpiW-p4eA@mail.gmail.com>

Bug reports go in JIRA, FreeSWITCH.org/jira

On Wednesday, July 8, 2015, Alexey Melnichuk <alexeymelnichuck at gmail.com>
wrote:

> FreeSWITCH Version 1.4.20+git~20150703T164215Z~b95362f965~32bit (git
> b95362f
> 2015-07-03 16:42:15Z 32bit)
>
> I think I found bug in spandsp.
>
> `t4_image_put_handler_t` defined as (__cdecl)
>
> typedef int (*t4_image_put_handler_t)(void *user_data, const uint8_t buf[],
> size_t len);
>
> But `tXX_decode_put` functions defined as `__stdcall`.
> So when I do
>
> s->image_put_handler = (t4_image_put_handler_t) t4_t6_decode_put;
> ...
> s->image_put_handler(...)
>
> I got AV.
>
> After I declare t4_image_put_handler_t as __stdcall problem solved.
> (Alse need update declaretion to `pre_encoded_put` function)
>
> But I start getting heap corruption on
> `span_free((char *) s->tiff.file);`
>
> Problem is that it uses `strdup` function with different allocator.
> So I just copy function to `alloc.c`
>
> SPAN_DECLARE(char *) span_strdup(const char *str)
> {
>     char *sdup;
>     size_t len = strlen(str) + 1;
>
>     sdup = (char *) span_alloc(len);
>     memcpy(sdup, str, len);
>
>     return sdup;
> }
>
> And use it as
> ds->tiff.file = span_strdup(file);
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> 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 <javascript:;>
> 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
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From brian at freeswitch.org  Thu Jul  9 16:52:57 2015
From: brian at freeswitch.org (Brian West)
Date: Thu, 9 Jul 2015 07:52:57 -0500
Subject: [Freeswitch-users] Fwd: sofia_reg_handle_register_token - exp
 possibly always false
In-Reply-To: <CANJp-yiauO6q-bsWEbcPvWZrsD3LHG+0PN9fU7sjCUifCJ2R1A@mail.gmail.com>
References: <CANJp-yhS4260A26B=Y7cC=MFi4X_m5k1PTh3u3BFFJLC6wsWsg@mail.gmail.com>
	<CANJp-yiauO6q-bsWEbcPvWZrsD3LHG+0PN9fU7sjCUifCJ2R1A@mail.gmail.com>
Message-ID: <CAEJMVkC9FhAACbB1piDqp5D9J16reQqDShfX3cysGbN3D0=b8w@mail.gmail.com>

Please file bugs in jira

On Wednesday, July 8, 2015, Podrigal, Aron <aronp at guaranteedplus.com> wrote:

>
> ---------- Forwarded message ----------
> From: Podrigal, Aron <aronp at guaranteedplus.com
> <javascript:_e(%7B%7D,'cvml','aronp at guaranteedplus.com');>>
> Date: Tue, Jul 7, 2015 at 7:24 PM
> Subject: sofia_reg_handle_register_token - exp possibly always false
> To: freeswitch-dev <freeswitch-dev at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-dev at lists.freeswitch.org');>>
>
>
> Hi,
>
> According to grep the function sofia_reg_handle_register_token is called
> from a single place src/mod/endpoints/mod_sofia/sofia_reg.c:2305
> <https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/endpoints/mod_sofia/sofia_reg.c#2305> where
> the argument *v_event* would always be NULL at that time. So I'm
> wondering what the line at src/mod/endpoints/mod_sofia/sofia_reg.c:1297
> <https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/endpoints/mod_sofia/sofia_reg.c#1297>
>   if (v_event && *v_event) pres_on_reg =
> switch_event_get_header(*v_event, "send-presence-on-register"); is
> suppose to do, won't this always evaluate to false?
>
> --
> Aron Podrigal
> -
> //Be happy :-)
>
>
>
> --
> Aron Podrigal
> -
> //Be happy :-)
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> 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 <javascript:;>
> 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit:
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*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
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From brian at freeswitch.org  Thu Jul  9 16:58:45 2015
From: brian at freeswitch.org (Brian West)
Date: Thu, 9 Jul 2015 07:58:45 -0500
Subject: [Freeswitch-users] mod_rtmp onStatus messages
In-Reply-To: <CAH3KVWoHQELDCv5UeuqOgePFd0E2qkn+LRjNDvO7Fsm4RyBZ6w@mail.gmail.com>
References: <CAH3KVWoHQELDCv5UeuqOgePFd0E2qkn+LRjNDvO7Fsm4RyBZ6w@mail.gmail.com>
Message-ID: <CAEJMVkCBLyz=ho631aoy1YbRvVzdSh+goY5KaP0-Rq=u=3Oc6Q@mail.gmail.com>

Maybe file a jira with the details for discussion and tracking

On Sunday, July 5, 2015, Paul Cuttler <paulcuttler at gmail.com> wrote:

> I was wondering about the onStatus messages that are sent by mod_rtmp to
> Flash Player. The RTMP Spec (see page 38 of
> http://wwwimages.adobe.com/content/dam/Adobe/en/devnet/rtmp/pdf/rtmp_specification_1.0.pdf)
> says that the transaction ID for onStatus messages is supposed to be 0, but
> mod_rtmp uses 1. It seems to work fine, but I wanted to check if anyone
> knew if it might lead to problems not by following the spec.
>
> This is in rtmp_sig.c, for example in rtmp_i_play.
>
> regards,
>
> Paul
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> 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 <javascript:;>
> 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
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From brian at freeswitch.org  Thu Jul  9 17:00:56 2015
From: brian at freeswitch.org (Brian West)
Date: Thu, 9 Jul 2015 08:00:56 -0500
Subject: [Freeswitch-users] No Video in Media Proxy mode
In-Reply-To: <007c01d0b788$24441020$6ccc3060$@com>
References: <003f01d0b55b$1f25b280$5d711780$@com>
	<CAEJMVkBTD7dfj31cuuU6DzO0dGim0FqkgpjwYZmq713-=PXTSw@mail.gmail.com>
	<007c01d0b788$24441020$6ccc3060$@com>
Message-ID: <CAEJMVkBdgncQUREb6ZtttTECQm7v6+7xsPX88QONE22Bo3yVsQ@mail.gmail.com>

FreeSWITCH will already avoid transcoding if both legs have common codecs,
there is zero gain from using proxy media in most cases.  It was designed
for codecs FreeSWITCH doesn't know about.  Any performance gains are
probably purely mythical at best.

On Sunday, July 5, 2015, Bryan <gliu at tecomtech.com> wrote:

>  HI, Actually we tested FS 1.4.18 , FS 1.4.19 and the latest Windows MSI
> installation package, all have the same issue.
>
>
>
> The reason why we test Proxy mode is to see the performance of FS than
> Media relay (with Transcoding).
>
>
>
> Thanks.
>
>
>
>
>
> *???:* freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>]
> *?? *Brian West
> *????:* 2015?7?3? 22:06
> *???:* FreeSWITCH Users Help
> *??:* Re: [Freeswitch-users] No Video in Media Proxy mode
>
>
>
> What rev of Freeswitch, and why are you using proxy media?
>
> On Friday, July 3, 2015, Bryan <gliu at tecomtech.com
> <javascript:_e(%7B%7D,'cvml','gliu at tecomtech.com');>> wrote:
>
> Hi, I?m testing FreeSWITCH with Android Linphone. I found the Video stream
> was not forwarded by FS. Audio is OK.
>
>
>
> Any Ideas? Or this is a bug?
>
>
>
> The test envrionmet as following:
>
> Android Linphone < -- > Wifi Router < -- > FS < -- >PC Linphone.
>
>
>
> The FS Version is the Latest Windows MSI package.
>
>
>
> 2015-07-03 14:31:57.718750 [NOTICE] switch_channel.c:1055 New Channel
> sofia/inte
>
> rnal/1009 at 172.16.1.115 [c8a19d44-3084-4539-b817-3276fd7ec43d]
>
> 2015-07-03 14:31:57.750000 [INFO] mod_dialplan_xml.c:635 Processing 1009
> <1009>-
>
> >1007 in context default
>
> 2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *1
> execut
>
> e_extension::dx XML features
>
> 2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *2
> record
>
> _session::C:/Program
> Files/FreeSWITCH/recordings/1009.2015-07-03-14-31-57.wav
>
> 2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *3
> execut
>
> e_extension::cf XML features
>
> 2015-07-03 14:31:57.781250 [INFO] switch_ivr_async.c:3822 Bound B-Leg: *4
> execut
>
> e_extension::att_xfer XML features
>
> 2015-07-03 14:31:57.781250 [NOTICE] switch_channel.c:1055 New Channel
> sofia/inte
>
> rnal/sip:1007 at 172.16.1.115:2887 [4cfdf0e6-7010-467b-97b3-20e4dc295422]
>
> 2015-07-03 14:31:57.843750 [NOTICE] sofia.c:6716 Ring-Ready
> sofia/internal/sip:1
>
> 007 at 172.16.1.115:2887!
>
> 2015-07-03 14:31:57.843750 [NOTICE] mod_sofia.c:2107 Ring-Ready
> sofia/internal/1
>
> 009 at 172.16.1.115!
>
> 2015-07-03 14:31:57.859375 [NOTICE] switch_ivr_originate.c:527 Ring Ready
> sofia/
>
> internal/1009 at 172.16.1.115!
>
> 2015-07-03 14:32:00.312500 [NOTICE] sofia.c:7437 Channel
> [sofia/internal/sip:100
>
> 7 at 172.16.1.115:2887] has been answered
>
> 2015-07-03 14:32:00.312500 [NOTICE] switch_core_media.c:4405
> sofia/internal/sip:
>
> 1007 at 172.16.1.115:2887 Starting Video thread
>
> 2015-07-03 14:32:00.343750 [NOTICE] switch_core_media.c:4405
> sofia/internal/1009
>
> @172.16.1.115 Starting Video thread
>
> 2015-07-03 14:32:00.343750 [NOTICE] switch_ivr_originate.c:3522 Channel
> [sofia/i
>
> nternal/1009 at 172.16.1.115] has been answered
>
> 2015-07-03 14:32:00.843750 [INFO] switch_rtp.c:5832 Auto Changing port
> from 61.1
>
> 83.139.155:64131 to 172.16.15.111:7078
>
> 2015-07-03 14:32:23.750000 [NOTICE] sofia.c:952 Hangup
> sofia/internal/sip:1007 at 1
>
> 72.16.1.115:2887 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>
> 2015-07-03 14:32:23.750000 [NOTICE] switch_ivr_bridge.c:1608 Hangup
> sofia/intern
>
> al/1009 at 172.16.1.115 [CS_EXECUTE] [NORMAL_CLEARING]
>
> 2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1633 Session 4
> (sofia/
>
> internal/sip:1007 at 172.16.1.115:2887) Ended
>
> 2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1637 Close
> Channel sof
>
> ia/internal/sip:1007 at 172.16.1.115:2887 [CS_DESTROY]
>
> 2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1633 Session 3
> (sofia/
>
> internal/1009 at 172.16.1.115) Ended
>
> 2015-07-03 14:32:23.750000 [NOTICE] switch_core_session.c:1637 Close
> Channel sof
>
> ia/internal/1009 at 172.16.1.115 [CS_DESTROY]
>
>
>
>
>
>
>
> Tony
>
>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','brian at freeswitch.org');>
>
>  *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> 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 <javascript:;>
> 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
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From brian at freeswitch.org  Thu Jul  9 17:02:38 2015
From: brian at freeswitch.org (Brian West)
Date: Thu, 9 Jul 2015 08:02:38 -0500
Subject: [Freeswitch-users] Preferred Crypto for outbound calls
In-Reply-To: <FA888BBC-DD6F-4131-BBF8-BCC429A50623@jerris.com>
References: <159fe211186f415c89ee70143926d02d@UK-MAIL-001.edge.local>
	<FA888BBC-DD6F-4131-BBF8-BCC429A50623@jerris.com>
Message-ID: <CAEJMVkBf3LTLgsUwEaVvmekgdBU2u7Ym8q-HPYHg=EQQE5TinQ@mail.gmail.com>

Also read the very extensive docs on this topic in 1.4's vars.xml

On Friday, July 3, 2015, Michael Jerris <mike at jerris.com> wrote:

> the variable changed in 1.4 to rtp_secure_media.  Also note we released
> 1.4.20 yesterday, I recommend using it as we have added quite a few bug
> fixes.
>
> On Jul 3, 2015, at 7:39 AM, Prashanth Devarajappa <
> Prashanth.Devarajappa at enghouse.com
> <javascript:_e(%7B%7D,'cvml','Prashanth.Devarajappa at enghouse.com');>>
> wrote:
>
> Hello,
>
> How do I set my preferred crypto suite to be used for SRTP outbound call ?
> I am running FS 1.4.18.
> In old version of FS(1.2.3), I could do that using
> sip_secure_media=AES_CM_128_HMAC_SHA1_80. But this variable doesn?t exist
> anymore, by the looks of it.
>
> Regards
> Prashanth
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','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
> <javascript:_e(%7B%7D,'cvml','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 <javascript:;>
> 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 <javascript:;>
> 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
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From steven.szeto at mitel.com  Thu Jul  9 18:33:29 2015
From: steven.szeto at mitel.com (Steven Szeto)
Date: Thu, 9 Jul 2015 14:33:29 +0000
Subject: [Freeswitch-users] Eavesdrop a call.
In-Reply-To: <CAJTkRNvU3PQRHjFF1MxF1KsCq4L5fo4cZxBtNLFxiSLDvDzdwg@mail.gmail.com>
References: <CAJTkRNvU3PQRHjFF1MxF1KsCq4L5fo4cZxBtNLFxiSLDvDzdwg@mail.gmail.com>
Message-ID: <BY2PR12MB0242B06E87CF2434520281BB8A900@BY2PR12MB0242.namprd12.prod.outlook.com>

Igor,

You need a call leg to perform the eavesdrop. The <channel-uuid> is the channel of an active call that is to be eavesdropped upon.

These examples work from the fs_cli command window. You will have to modify the commands with extensions and the correct uuid:

Step 1: establish a 2 party call:

two party call:
originate sofia/internal/5401 at 10.47.41.109 &bridge(sofia/internal/5901 at 10.47.41.109)

Step 2: invoke eavesdrop with the desired behavior:

whisper/coach:
originate sofia/internal/5902 at 10.47.41.109 'queue_dtmf:w2 at 500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861' inline

originate sofia/internal/4901 at 10.47.32.159 'queue_dtmf:w2 at 500,eavesdrop:35b70871-78d4-4587-a186-5c0457482515' inline

silent monitor:
originate sofia/internal/5401 at 10.47.41.109 'eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861' inline
originate sofia/internal/5002 at 10.47.32.159 'queue_dtmf:w0 at 500,eavesdrop:abefa174-12dc-4ccb-956e-67c51475c414' inline


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Igor Olhovskiy
Sent: Thursday, July 09, 2015 7:48 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Eavesdrop a call.

Hi!
I'm using Freeswitch 1.4.20 (git b95362f 2015-07-03 16:42:15Z 64bit)) (stable one)
and trying to get eavesdropping. But no luck.
Main scheme:
DID - > extension (210). 210 answers
eavesdropping from extension 110.
While giving a command
eavesdrop <channel-uuid>
I've got log full of messages
2015-07-09 13:29:00.325667 [ERR] switch_core_io.c:1531 Write Buffer 0 bytes Failed!
and silence on 110 extension

While giving a command
userspy 210@${domain_name} or userspy <channel-uuid>
I've got music-on-hold on 110, no errors on log.

In both cases used <channel-uuid> of extension 210 (answered one)

Is this a bug or I'm doing something wrong?
--
Best regards,
Igor

________________________________
NOTE: This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message.
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From nboric at yx.cl  Thu Jul  9 18:42:12 2015
From: nboric at yx.cl (Neven Boric)
Date: Thu, 9 Jul 2015 11:42:12 -0300
Subject: [Freeswitch-users] Some problems with bypass-media
In-Reply-To: <CAHQ=dcJ+JNsU720QdKMznpYxSs0AryYif-MQbswFu2H2VrBAgQ@mail.gmail.com>
References: <CAHQ=dcK0bfZbZ30zt9nYUH1z-R6BnSZBM1W0kAHommJspLKftg@mail.gmail.com>
	<CAFiqYukicT1Af87M6K3KUn-EYTPGGq2qtjWLi2TeAB+=kNodmw@mail.gmail.com>
	<CAHQ=dcJ+JNsU720QdKMznpYxSs0AryYif-MQbswFu2H2VrBAgQ@mail.gmail.com>
Message-ID: <CAHQ=dc+_XbYXvTAYymG7AGypod1aWq87M9By_WEJU4NUrS74mQ@mail.gmail.com>

Ok, I tried with master, and something strange happened. I'm pretty sure it
worked last night, but now I'm testing again and I'm getting the same
behavior I described with 1.4. I don't even have to get a third phone
involved. All I do is call from A to B, put B on hold and then wait, and
about ten seconds later, FS will hangup both calls, as if some timeout was
triggered. I get the same error on the log:

2015-07-09 14:36:43.506725 [CRIT] switch_core_io.c:93 sofia/internal/
140 at 10.50.100.16:5060 reading on a session with no media!

I also get no MOH on the phone that was put on hold.

On Wed, Jul 8, 2015 at 3:41 PM, Neven Boric <nboric at yx.cl> wrote:

>
>
> On Wed, Jul 8, 2015 at 2:17 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> - I first tried with the 1.2 branch, as that was the version I was using
>>> locally. The issue with that version is that when a phone holds and then
>>> unholds a call, I get no audio on the phone that started the hold. I found
>>> option bypass-media-after-hold, but realized that it was not included in
>>> the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93 from
>>> the master branch. Now FS correctly tries to restore direct media between
>>> the endpoints with reINVITEs, but unfortunately includes a 'sendonly' in
>>> the last SDP to the phone that unholds, so I still get no audio. I haven't
>>> found a way to fix this.
>>
>>
>> Which probably means there's other commits required. 1.2 is EOL and no
>> longer supported, 1.4 is the current stable release. I wouldn't spend any
>> time getting 1.2 working, and instead work on upgrading.
>>
>
> Yes, I know it's not currently supported, but maybe somebody had the same
> problem and fixed it in a different way. Or maybe somebody could point me
> in the right direction to remove that final 'sendonly'.
>
>
>>
>> - I also tried with the 1.4 branch and hold/unhold works correctly, but
>>> now attended transfer doesn't work, FS after some time ends the call on the
>>> side that was put on hold waiting to be transferred. This time seems to be
>>> inconsistent, and sometimes the transfer actually works, so maybe it is a
>>> timing issue. This seems similar/related to FS-4038.
>>>
>>
>> Also see if you can replicate it on master, as that's close to the
>> upcoming 1.6 release and will have a lot of changes over 1.4.
>>
>
> I will try with master and report back, thanks.
>
>
>>
>>
>> On 8 July 2015 at 14:23, Neven Boric <nboric at yx.cl> wrote:
>>
>>> Hi,
>>>
>>> I have been using FS for a long time on a local server without issues,
>>> but now I want to move to a remote server to support some new usage
>>> scenarios. I'm trying to use inbound-bypass-media=true to keep the audio
>>> out of the server. This mostly works, but I have two different issues,
>>> depending on which FS version I use:
>>>
>>> - I first tried with the 1.2 branch, as that was the version I was using
>>> locally. The issue with that version is that when a phone holds and then
>>> unholds a call, I get no audio on the phone that started the hold. I found
>>> option bypass-media-after-hold, but realized that it was not included in
>>> the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93 from
>>> the master branch. Now FS correctly tries to restore direct media between
>>> the endpoints with reINVITEs, but unfortunately includes a 'sendonly' in
>>> the last SDP to the phone that unholds, so I still get no audio. I haven't
>>> found a way to fix this.
>>>
>>> - I also tried with the 1.4 branch and hold/unhold works correctly, but
>>> now attended transfer doesn't work, FS after some time ends the call on the
>>> side that was put on hold waiting to be transferred. This time seems to be
>>> inconsistent, and sometimes the transfer actually works, so maybe it is a
>>> timing issue. This seems similar/related to FS-4038.
>>>
>>> I can get logs and SIP captures, what would be the preferred way to
>>> provide them (pcap, inline text, pastebin)? Or maybe it's better to go
>>> ahead and file a bug?
>>>
>>> Best regards
>>> Neven Boric
>>>
>>> _________________________________________________________________________
>>> 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 igorolhovskiy at gmail.com  Thu Jul  9 20:17:48 2015
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Thu, 9 Jul 2015 19:17:48 +0300
Subject: [Freeswitch-users] Eavesdrop a call.
In-Reply-To: <BY2PR12MB0242B06E87CF2434520281BB8A900@BY2PR12MB0242.namprd12.prod.outlook.com>
References: <CAJTkRNvU3PQRHjFF1MxF1KsCq4L5fo4cZxBtNLFxiSLDvDzdwg@mail.gmail.com>
	<BY2PR12MB0242B06E87CF2434520281BB8A900@BY2PR12MB0242.namprd12.prod.outlook.com>
Message-ID: <CAJTkRNv_3T9=LQRRCGTsw7BCb+EHMXaNBuyA3wo=8dZ90pJ1yg@mail.gmail.com>

Thanks for an answer, but while calling eavesdrop I'm giving as a parameter
existing call-leg ID (answered one)
But I'll try with originate one command, thanks.

???????, 9 ???? 2015 ?. ???????????? Steven Szeto ???????:

>  Igor,
>
>
>
> You need a call leg to perform the eavesdrop. The <channel-uuid> is the
> channel of an active call that is to be eavesdropped upon.
>
>
>
> These examples work from the fs_cli command window. You will have to
> modify the commands with extensions and the correct uuid:
>
>
>
> Step 1: establish a 2 party call:
>
>
>
> two party call:
>
> originate sofia/internal/5401 at 10.47.41.109
> <javascript:_e(%7B%7D,'cvml','5401 at 10.47.41.109');>
> &bridge(sofia/internal/5901 at 10.47.41.109
> <javascript:_e(%7B%7D,'cvml','5901 at 10.47.41.109');>)
>
>
>
> Step 2: invoke eavesdrop with the desired behavior:
>
>
>
> whisper/coach:
>
> originate sofia/internal/5902 at 10.47.41.109
> <javascript:_e(%7B%7D,'cvml','5902 at 10.47.41.109');> 'queue_dtmf:w2 at 500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861'
> inline
>
>
>
> originate sofia/internal/4901 at 10.47.32.159
> <javascript:_e(%7B%7D,'cvml','4901 at 10.47.32.159');> 'queue_dtmf:w2 at 500,eavesdrop:35b70871-78d4-4587-a186-5c0457482515'
> inline
>
>
>
> silent monitor:
>
> originate sofia/internal/5401 at 10.47.41.109
> <javascript:_e(%7B%7D,'cvml','5401 at 10.47.41.109');>
> 'eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861' inline
>
> originate sofia/internal/5002 at 10.47.32.159
> <javascript:_e(%7B%7D,'cvml','5002 at 10.47.32.159');> 'queue_dtmf:w0 at 500,eavesdrop:abefa174-12dc-4ccb-956e-67c51475c414'
> inline
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>]
> *On Behalf Of *Igor Olhovskiy
> *Sent:* Thursday, July 09, 2015 7:48 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Eavesdrop a call.
>
>
>
> Hi!
>
> I'm using Freeswitch 1.4.20 (git b95362f 2015-07-03 16:42:15Z 64bit))
> (stable one)
>
> and trying to get eavesdropping. But no luck.
>
> Main scheme:
>
> DID - > extension (210). 210 answers
>
> eavesdropping from extension 110.
>
> While giving a command
>
> *eavesdrop <channel-uuid> *
>
> I've got log full of messages
>
> *2015-07-09 13:29:00.325667 [ERR] switch_core_io.c:1531 Write Buffer 0
> bytes Failed!*
>
> and silence on 110 extension
>
>
>
> While giving a command
>
> *userspy 210@${domain_name}* or *userspy <channel-uuid>*
>
> I've got music-on-hold on 110, no errors on log.
>
>
>
> In both cases used <channel-uuid> of extension 210 (answered one)
>
>
>
> Is this a bug or I'm doing something wrong?
>
> --
>
> Best regards,
>
> Igor
>
> ------------------------------
> NOTE: This e-mail (including any attachments) is for the sole use of the
> intended recipient(s) and may contain information that is confidential
> and/or protected by legal privilege. Any unauthorized review, use, copy,
> disclosure or distribution of this e-mail is strictly prohibited. If you
> are not the intended recipient, please notify Mitel immediately and destroy
> all copies of this e-mail. Mitel does not accept any liability for breach
> of security, error or virus that may result from the transmission of this
> message.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> 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 <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


-- 
Best regards,
Igor
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From brian at freeswitch.org  Fri Jul 10 00:11:07 2015
From: brian at freeswitch.org (Brian West)
Date: Thu, 9 Jul 2015 15:11:07 -0500
Subject: [Freeswitch-users] Eavesdrop a call.
In-Reply-To: <CAJTkRNv_3T9=LQRRCGTsw7BCb+EHMXaNBuyA3wo=8dZ90pJ1yg@mail.gmail.com>
References: <CAJTkRNvU3PQRHjFF1MxF1KsCq4L5fo4cZxBtNLFxiSLDvDzdwg@mail.gmail.com>
	<BY2PR12MB0242B06E87CF2434520281BB8A900@BY2PR12MB0242.namprd12.prod.outlook.com>
	<CAJTkRNv_3T9=LQRRCGTsw7BCb+EHMXaNBuyA3wo=8dZ90pJ1yg@mail.gmail.com>
Message-ID: <CAEJMVkDBKs_cUUkCK3wYQk=sk9MLHM1xygJXo3ZFdEd7MZJ4+g@mail.gmail.com>

You can also just say 'all', and cycle thru them all see the confluence
page over this its fairly detailed.

On Thu, Jul 9, 2015 at 11:17 AM, Igor Olhovskiy <igorolhovskiy at gmail.com>
wrote:

> Thanks for an answer, but while calling eavesdrop I'm giving as a
> parameter existing call-leg ID (answered one)
> But I'll try with originate one command, thanks.
>
> ???????, 9 ???? 2015 ?. ???????????? Steven Szeto ???????:
>
>>  Igor,
>>
>>
>>
>> You need a call leg to perform the eavesdrop. The <channel-uuid> is the
>> channel of an active call that is to be eavesdropped upon.
>>
>>
>>
>> These examples work from the fs_cli command window. You will have to
>> modify the commands with extensions and the correct uuid:
>>
>>
>>
>> Step 1: establish a 2 party call:
>>
>>
>>
>> two party call:
>>
>> originate sofia/internal/5401 at 10.47.41.109 &bridge(sofia/internal/
>> 5901 at 10.47.41.109)
>>
>>
>>
>> Step 2: invoke eavesdrop with the desired behavior:
>>
>>
>>
>> whisper/coach:
>>
>> originate sofia/internal/5902 at 10.47.41.109 'queue_dtmf:w2 at 500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861'
>> inline
>>
>>
>>
>> originate sofia/internal/4901 at 10.47.32.159 'queue_dtmf:w2 at 500,eavesdrop:35b70871-78d4-4587-a186-5c0457482515'
>> inline
>>
>>
>>
>> silent monitor:
>>
>> originate sofia/internal/5401 at 10.47.41.109
>> 'eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861' inline
>>
>> originate sofia/internal/5002 at 10.47.32.159 'queue_dtmf:w0 at 500,eavesdrop:abefa174-12dc-4ccb-956e-67c51475c414'
>> inline
>>
>>
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Igor
>> Olhovskiy
>> *Sent:* Thursday, July 09, 2015 7:48 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* [Freeswitch-users] Eavesdrop a call.
>>
>>
>>
>> Hi!
>>
>> I'm using Freeswitch 1.4.20 (git b95362f 2015-07-03 16:42:15Z 64bit))
>> (stable one)
>>
>> and trying to get eavesdropping. But no luck.
>>
>> Main scheme:
>>
>> DID - > extension (210). 210 answers
>>
>> eavesdropping from extension 110.
>>
>> While giving a command
>>
>> *eavesdrop <channel-uuid> *
>>
>> I've got log full of messages
>>
>> *2015-07-09 13:29:00.325667 [ERR] switch_core_io.c:1531 Write Buffer 0
>> bytes Failed!*
>>
>> and silence on 110 extension
>>
>>
>>
>> While giving a command
>>
>> *userspy 210@${domain_name}* or *userspy <channel-uuid>*
>>
>> I've got music-on-hold on 110, no errors on log.
>>
>>
>>
>> In both cases used <channel-uuid> of extension 210 (answered one)
>>
>>
>>
>> Is this a bug or I'm doing something wrong?
>>
>> --
>>
>> Best regards,
>>
>> Igor
>>
>> ------------------------------
>> NOTE: This e-mail (including any attachments) is for the sole use of the
>> intended recipient(s) and may contain information that is confidential
>> and/or protected by legal privilege. Any unauthorized review, use, copy,
>> disclosure or distribution of this e-mail is strictly prohibited. If you
>> are not the intended recipient, please notify Mitel immediately and destroy
>> all copies of this e-mail. Mitel does not accept any liability for breach
>> of security, error or virus that may result from the transmission of this
>> message.
>>
>>
>> _________________________________________________________________________
>> 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
>>
>
>
> --
> Best 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
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
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From jackson at forwardit.com.au  Fri Jul 10 05:11:28 2015
From: jackson at forwardit.com.au (Jackson Fisher - Forward.IT)
Date: Fri, 10 Jul 2015 01:11:28 +0000
Subject: [Freeswitch-users] BLF on Cisco spa 252g with Attendant console
Message-ID: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit>

Hi All,

I am trying to get BLF working on a cisco spa 525g attendant console.
Can anyone point me in the right direction for the string I need to put into each button.


Thanks, Jackson Fisher




[Email-signature.jpg]

Jackson Fisher   |  Network Engineer

A : Level 1 - Unit 8,  55-57 Lathlain Street,  Belconnen ACT 2617

P : (02) 6162 0070

E : Jackson at forwardit.com.au<mailto:Jackson at forwardit.com.au>
W : http://www.forwardit.com.au<http://www.forwardit.com.au/>




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From yadenis at seznam.cz  Fri Jul 10 09:01:02 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Fri, 10 Jul 2015 07:01:02 +0200
Subject: [Freeswitch-users] how to improve the performance of Web RTC?
In-Reply-To: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit>
References: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit>
Message-ID: <69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>

Hi All !

I have a question. I use several different libraries (jssip, sip.js) for RTC Web communications. In both variants I observed during the freezing of the video communication. Sometimes it works great and without freezing. Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've tried various settings Dialplan and internal.xml. But the result is almost the same. 
Maybe dear colleagues will share tips on how to make the connection stable? 

PS: I use FreeSwitch 1.7 on Debian 8. 



-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

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From igorolhovskiy at gmail.com  Fri Jul 10 13:05:07 2015
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Fri, 10 Jul 2015 12:05:07 +0300
Subject: [Freeswitch-users] how to improve the performance of Web RTC?
In-Reply-To: <69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>
References: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit>
	<69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>
Message-ID: <CAJTkRNt_u2QjGKy14zZhVG=dE6y7J0hzJJM0uP-LeGRS4k2TDw@mail.gmail.com>

My strongest suggestion - switch to mod_verto. All sip implementation for
JavaScript is buggy in many different ways. And with every new release of
the browser you will get new surprises.

???????, 10 ???? 2015 ?. ???????????? Denis Jakovlev ???????:

> Hi All !
>
> I have a question. I use several different libraries (jssip, sip.js) for
> RTC Web communications. In both variants I observed during the freezing of
> the video communication. Sometimes it works great and without freezing.
> Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've
> tried various settings Dialplan and internal.xml. But the result is almost
> the same.
> Maybe dear colleagues will share tips on how to make the connection
> stable?
>
> PS: I use FreeSwitch 1.7 on Debian 8.
>
>
>
> *-- *
> *S pozdravem,*
> *Ing.Denis Jakovlev                           *
> *mob.tel <http://mob.tel>. 775-415-382*
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> 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 <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


-- 
Best regards,
Igor
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From ricardas.stoma at gmail.com  Fri Jul 10 13:18:33 2015
From: ricardas.stoma at gmail.com (=?UTF-8?Q?Ri=C4=8Dardas_Stoma?=)
Date: Fri, 10 Jul 2015 12:18:33 +0300
Subject: [Freeswitch-users] Bypass media prevents bridge from blocking LUA
	script execution
Message-ID: <CAFX-kAc+=EXyFxipnRgWv7_i2cRxkM9b9F3EU3P7qkmKb_UByw@mail.gmail.com>

I use LUA script to bridge calls but now i have problem with bypass media.
Let's say i have this LUA script:

session:execute("set", "bypass_media=true");
session:execute("bridge", "sofia/external/123 at 192.168.0.100:5061");
freeswitch.consoleLog("info", "This is my message\n");

When bypass_media is enabled, log message "This is my message" appears once
legB answers call. If bypass_media is disabled then log message appears
when call ends. In other words bypass_media prevents "bridge" from blocking
script execution.

Is there a way to block script execution until call ends when bypass_media
is enabled?
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From yadenis at seznam.cz  Fri Jul 10 13:40:58 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Fri, 10 Jul 2015 11:40:58 +0200
Subject: [Freeswitch-users] how to improve the performance of Web RTC?
In-Reply-To: <CAJTkRNt_u2QjGKy14zZhVG=dE6y7J0hzJJM0uP-LeGRS4k2TDw@mail.gmail.com>
References: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit> 
	<69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>
	<CAJTkRNt_u2QjGKy14zZhVG=dE6y7J0hzJJM0uP-LeGRS4k2TDw@mail.gmail.com>
Message-ID: <588319577.20150710114058@seznam.cz>

Dobr? den,

About a month ago, I asked here about connection problems with the video. And to me strongly suggest it sipjs. I can not change the library every month.
The site freeswitch recommend these 2 libraries (sipjs, jssip). Why can not I use them? 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

p?tek 10. ?ervence 2015, 11:05:07, napsal jste:


My strongest suggestion - switch to mod_verto. All sip implementation for JavaScript is buggy in many different ways. And with every new release of the browser you will get new surprises. 

???????, 10 ???? 2015 ?. ???????????? Denis Jakovlev ???????:
Hi All !

I have a question. I use several different libraries (jssip, sip.js) for RTC Web communications. In both variants I observed during the freezing of the video communication. Sometimes it works great and without freezing. Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've tried various settings Dialplan and internal.xml. But the result is almost the same. 
Maybe dear colleagues will share tips on how to make the connection stable? 

PS: I use FreeSwitch 1.7 on Debian 8. 



-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

_________________________________________________________________________
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


-- 
Best regards,
Igor
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From igorolhovskiy at gmail.com  Fri Jul 10 15:28:27 2015
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Fri, 10 Jul 2015 14:28:27 +0300
Subject: [Freeswitch-users] how to improve the performance of Web RTC?
In-Reply-To: <588319577.20150710114058@seznam.cz>
References: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit>
	<69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>
	<CAJTkRNt_u2QjGKy14zZhVG=dE6y7J0hzJJM0uP-LeGRS4k2TDw@mail.gmail.com>
	<588319577.20150710114058@seznam.cz>
Message-ID: <CAJTkRNvN-V7YzarmrrD-EctGoERcv-0Mddy5fRL4E68xGTemHA@mail.gmail.com>

You can.
But this library add a support of a really not light protocol (SIP) in your
browser. And I'm not sure about quality of these libraries. With Verto you
got all from "one hands", as front-part (verto.js) and back part (mod_verto)

2015-07-10 12:40 GMT+03:00 Denis Jakovlev <yadenis at seznam.cz>:

>  Dobr? den,
>
> About a month ago, I asked here about connection problems with the video.
> And to me strongly suggest it sipjs. I can not change the library every
> month.
> The site freeswitch recommend these 2 libraries (sipjs, jssip). Why can
> not I use them?
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 p?tek 10. ?ervence 2015, 11:05:07, napsal
> jste: *
>
>  My strongest suggestion - switch to mod_verto. All sip implementation
> for JavaScript is buggy in many different ways. And with every new release
> of the browser you will get new surprises.
>
> ???????, 10 ???? 2015 ?. ???????????? Denis Jakovlev ???????:
> Hi All !
>
> I have a question. I use several different libraries (jssip, sip.js) for
> RTC Web communications. In both variants I observed during the freezing of
> the video communication. Sometimes it works great and without freezing.
> Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've
> tried various settings Dialplan and internal.xml. But the result is almost
> the same.
> Maybe dear colleagues will share tips on how to make the connection
> stable?
>
> PS: I use FreeSwitch 1.7 on Debian 8.
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
> *. 775-415-382 *
> _________________________________________________________________________
> 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
>
>
> --
> Best 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
>



-- 
Best regards,
Igor
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From yadenis at seznam.cz  Fri Jul 10 15:43:18 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Fri, 10 Jul 2015 13:43:18 +0200
Subject: [Freeswitch-users] how to improve the performance of Web RTC?
In-Reply-To: <CAJTkRNvN-V7YzarmrrD-EctGoERcv-0Mddy5fRL4E68xGTemHA@mail.gmail.com>
References: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit> 
	<69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>
	<CAJTkRNt_u2QjGKy14zZhVG=dE6y7J0hzJJM0uP-LeGRS4k2TDw@mail.gmail.com>
	<588319577.20150710114058@seznam.cz>
	<CAJTkRNvN-V7YzarmrrD-EctGoERcv-0Mddy5fRL4E68xGTemHA@mail.gmail.com>
Message-ID: <1192410958.20150710134318@seznam.cz>

Dobr? den,

Ok. And there is some documentation for verto.js?
-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

p?tek 10. ?ervence 2015, 13:28:27, napsal jste:


You can.
But this library add a support of a really not light protocol (SIP) in your browser. And I'm not sure about quality of these libraries. With Verto you got all from "one hands", as front-part (verto.js) and back part (mod_verto)

2015-07-10 12:40 GMT+03:00 Denis Jakovlev <yadenis at seznam.cz>:
Dobr? den,

About a month ago, I asked here about connection problems with the video. And to me strongly suggest it sipjs. I can not change the library every month.
The site freeswitch recommend these 2 libraries (sipjs, jssip). Why can not I use them? 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

p?tek 10. ?ervence 2015, 11:05:07, napsal jste:


My strongest suggestion - switch to mod_verto. All sip implementation for JavaScript is buggy in many different ways. And with every new release of the browser you will get new surprises. 

???????, 10 ???? 2015 ?. ???????????? Denis Jakovlev ???????:
Hi All !

I have a question. I use several different libraries (jssip, sip.js) for RTC Web communications. In both variants I observed during the freezing of the video communication. Sometimes it works great and without freezing. Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've tried various settings Dialplan and internal.xml. But the result is almost the same. 
Maybe dear colleagues will share tips on how to make the connection stable? 

PS: I use FreeSwitch 1.7 on Debian 8. 



-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

_________________________________________________________________________
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


-- 
Best 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



-- 
Best regards,
Igor
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From igorolhovskiy at gmail.com  Fri Jul 10 15:57:03 2015
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Fri, 10 Jul 2015 14:57:03 +0300
Subject: [Freeswitch-users] how to improve the performance of Web RTC?
In-Reply-To: <1192410958.20150710134318@seznam.cz>
References: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit>
	<69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>
	<CAJTkRNt_u2QjGKy14zZhVG=dE6y7J0hzJJM0uP-LeGRS4k2TDw@mail.gmail.com>
	<588319577.20150710114058@seznam.cz>
	<CAJTkRNvN-V7YzarmrrD-EctGoERcv-0Mddy5fRL4E68xGTemHA@mail.gmail.com>
	<1192410958.20150710134318@seznam.cz>
Message-ID: <CAJTkRNtXQwzLsAtkF-dEsZgU=DysOWrwAbUbJCnOuYwZW50T8Q@mail.gmail.com>

https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/html5/verto
For now - only this one, as I know.

2015-07-10 14:43 GMT+03:00 Denis Jakovlev <yadenis at seznam.cz>:

>  Dobr? den,
>
> Ok. And there is some documentation for verto.js?
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 p?tek 10. ?ervence 2015, 13:28:27, napsal
> jste: *
>
>  You can.
> But this library add a support of a really not light protocol (SIP) in
> your browser. And I'm not sure about quality of these libraries. With Verto
> you got all from "one hands", as front-part (verto.js) and back part
> (mod_verto)
>
> 2015-07-10 12:40 GMT+03:00 Denis Jakovlev <yadenis at seznam.cz>:
> Dobr? den,
>
> About a month ago, I asked here about connection problems with the video.
> And to me strongly suggest it sipjs. I can not change the library every
> month.
> The site freeswitch recommend these 2 libraries (sipjs, jssip). Why can
> not I use them?
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
>
>
> *. 775-415-382 p?tek 10. ?ervence 2015, 11:05:07, napsal jste: *
>  My strongest suggestion - switch to mod_verto. All sip implementation
> for JavaScript is buggy in many different ways. And with every new release
> of the browser you will get new surprises.
>
> ???????, 10 ???? 2015 ?. ???????????? Denis Jakovlev ???????:
> Hi All !
>
> I have a question. I use several different libraries (jssip, sip.js) for
> RTC Web communications. In both variants I observed during the freezing of
> the video communication. Sometimes it works great and without freezing.
> Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've
> tried various settings Dialplan and internal.xml. But the result is almost
> the same.
> Maybe dear colleagues will share tips on how to make the connection
> stable?
>
> PS: I use FreeSwitch 1.7 on Debian 8.
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
> *. 775-415-382 *
> _________________________________________________________________________
> 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
>
>
> --
> Best 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
>
>
>
> --
> Best 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
>



-- 
Best regards,
Igor
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From mbodbg at gmx.net  Fri Jul 10 16:26:51 2015
From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=)
Date: Fri, 10 Jul 2015 14:26:51 +0200
Subject: [Freeswitch-users] Preset channel variables
Message-ID: <9CA1AD50-924E-4589-9488-1867FE3E7CB1@gmx.net>

I?m setting a channel variable in the dial plan, with the set application. In rare cases where the originator cancels the call, it happens that the set command is not executed and the value of the variable is null. To preset the variable I tried set it in vars.xml, but it seems I can only use the variable in the dial plan, but it?s not set in the channel variables posted by mod_xml_cdr. Is there a proper way to preset channel variables?

Thanks and Regards  

Markus

From manish.talwar at nexxuspg.com  Fri Jul 10 17:28:41 2015
From: manish.talwar at nexxuspg.com (Manish Talwar)
Date: Fri, 10 Jul 2015 13:28:41 +0000
Subject: [Freeswitch-users] Send a alert by email when freeswith server is
	down
Message-ID: <DM2PR07MB416DCE1CCE63AAC4E5996588C9F0@DM2PR07MB416.namprd07.prod.outlook.com>

Hi,


Is there any way to send an alert by email or SMS to someone whenever FreeSwitch is down (either by manually shutdown or server crashed/restart).


Please let me know the steps for its configuration if possible.


Thanks,


Regards,

Manish Talwar
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From mike at jerris.com  Fri Jul 10 17:31:32 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 10 Jul 2015 09:31:32 -0400
Subject: [Freeswitch-users] how to improve the performance of Web RTC?
In-Reply-To: <588319577.20150710114058@seznam.cz>
References: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit>
	<69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>
	<CAJTkRNt_u2QjGKy14zZhVG=dE6y7J0hzJJM0uP-LeGRS4k2TDw@mail.gmail.com>
	<588319577.20150710114058@seznam.cz>
Message-ID: <CAEwTsAgfdw+S672chqSikW10souR06ash_tacSOj4LnQXYNPvA@mail.gmail.com>

where on the site is that?  that should be removed.

On Friday, July 10, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Dobr? den,
>
> About a month ago, I asked here about connection problems with the video.
> And to me strongly suggest it sipjs. I can not change the library every
> month.
> The site freeswitch recommend these 2 libraries (sipjs, jssip). Why can
> not I use them?
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 p?tek 10. ?ervence 2015, 11:05:07, napsal
> jste: *
>  My strongest suggestion - switch to mod_verto. All sip implementation
> for JavaScript is buggy in many different ways. And with every new release
> of the browser you will get new surprises.
>
> ???????, 10 ???? 2015 ?. ???????????? Denis Jakovlev ???????:
> Hi All !
>
> I have a question. I use several different libraries (jssip, sip.js) for
> RTC Web communications. In both variants I observed during the freezing of
> the video communication. Sometimes it works great and without freezing.
> Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've
> tried various settings Dialplan and internal.xml. But the result is almost
> the same.
> Maybe dear colleagues will share tips on how to make the connection
> stable?
>
> PS: I use FreeSwitch 1.7 on Debian 8.
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
> *. 775-415-382 *
> _________________________________________________________________________
> 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
>
>
> --
> Best regards,
> Igor
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> 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 <javascript:;>
> 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 ben at langfeld.co.uk  Fri Jul 10 17:38:37 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Fri, 10 Jul 2015 10:38:37 -0300
Subject: [Freeswitch-users] Send a alert by email when freeswith server
	is down
In-Reply-To: <DM2PR07MB416DCE1CCE63AAC4E5996588C9F0@DM2PR07MB416.namprd07.prod.outlook.com>
References: <DM2PR07MB416DCE1CCE63AAC4E5996588C9F0@DM2PR07MB416.namprd07.prod.outlook.com>
Message-ID: <CAAyX+KEDZKAEka5ozGCE3gwFdHMPjR_Z1xtzYw520CCasLSr4g@mail.gmail.com>

This would be functionality of a monitoring service and not of FreeSWITCH.
Look at Nagios, PagerDuty, DotComMonitor, among others.

On 10 July 2015 at 10:28, Manish Talwar <manish.talwar at nexxuspg.com> wrote:

>  Hi,
>
>
>  Is there any way to send an alert by email or SMS to someone whenever
> FreeSwitch is down (either by manually shutdown or server crashed/restart).
>
>
>  Please let me know the steps for its configuration if possible.
>
>
>  Thanks,
>
>
>  Regards,
>
> Manish Talwar
>
> _________________________________________________________________________
> 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 ben at langfeld.co.uk  Fri Jul 10 17:44:28 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Fri, 10 Jul 2015 10:44:28 -0300
Subject: [Freeswitch-users] how to improve the performance of Web RTC?
In-Reply-To: <588319577.20150710114058@seznam.cz>
References: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit>
	<69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>
	<CAJTkRNt_u2QjGKy14zZhVG=dE6y7J0hzJJM0uP-LeGRS4k2TDw@mail.gmail.com>
	<588319577.20150710114058@seznam.cz>
Message-ID: <CAAyX+KFO5FcefkB86vJj8m=8UQuPTZWm4L90MPpX1QQPZHnSNw@mail.gmail.com>

You absolutely can use SIP.js, and infact I do. There are bugs, of course,
but all software has bugs. It seems like the scare-mongering about how
awful SIP in the browser is is more of a marketing effort for Verto than
reality.

Verto may be great, and given the FreeSWITCH core team's track record of
quality I don't doubt it, but it's not the great saviour that will fix all
of your problems while making you a cup of tea, nor is SIP.js the devil
that should be avoided at all costs.

There are tradeoffs to be made on both sides, and I've used SIP.js on
several occasions and been very happy with my choice.

More specifically, it's very unlikely that your choice of signalling
protocol will impact the stability of media flow. The suggestion to switch
to Verto in this case seems premature.

On 10 July 2015 at 06:40, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Dobr? den,
>
> About a month ago, I asked here about connection problems with the video.
> And to me strongly suggest it sipjs. I can not change the library every
> month.
> The site freeswitch recommend these 2 libraries (sipjs, jssip). Why can
> not I use them?
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 p?tek 10. ?ervence 2015, 11:05:07, napsal
> jste: *
>
>  My strongest suggestion - switch to mod_verto. All sip implementation
> for JavaScript is buggy in many different ways. And with every new release
> of the browser you will get new surprises.
>
> ???????, 10 ???? 2015 ?. ???????????? Denis Jakovlev ???????:
> Hi All !
>
> I have a question. I use several different libraries (jssip, sip.js) for
> RTC Web communications. In both variants I observed during the freezing of
> the video communication. Sometimes it works great and without freezing.
> Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've
> tried various settings Dialplan and internal.xml. But the result is almost
> the same.
> Maybe dear colleagues will share tips on how to make the connection
> stable?
>
> PS: I use FreeSwitch 1.7 on Debian 8.
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
> *. 775-415-382 *
> _________________________________________________________________________
> 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
>
>
> --
> Best 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 krice at freeswitch.org  Fri Jul 10 18:01:03 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 10 Jul 2015 14:01:03 +0000
Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <559fd01f3614f_8ab37a7330709bc@resque-worker.6.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
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From yadenis at seznam.cz  Fri Jul 10 18:01:12 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Fri, 10 Jul 2015 16:01:12 +0200
Subject: [Freeswitch-users] how to improve the performance of Web RTC?
In-Reply-To: <CAAyX+KFO5FcefkB86vJj8m=8UQuPTZWm4L90MPpX1QQPZHnSNw@mail.gmail.com>
References: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit> 
	<69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>
	<CAJTkRNt_u2QjGKy14zZhVG=dE6y7J0hzJJM0uP-LeGRS4k2TDw@mail.gmail.com>
	<588319577.20150710114058@seznam.cz>
	<CAAyX+KFO5FcefkB86vJj8m=8UQuPTZWm4L90MPpX1QQPZHnSNw@mail.gmail.com>
Message-ID: <1631200356.20150710160112@seznam.cz>

Dobr? den,

Thanks. I also think that perhaps Verto is a good choice, but why all break if working. The only thing I want is a little tips that I can do to improve.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

p?tek 10. ?ervence 2015, 15:44:28, napsal jste:


You absolutely can use SIP.js, and infact I do. There are bugs, of course, but all software has bugs. It seems like the scare-mongering about how awful SIP in the browser is is more of a marketing effort for Verto than reality.

Verto may be great, and given the FreeSWITCH core team's track record of quality I don't doubt it, but it's not the great saviour that will fix all of your problems while making you a cup of tea, nor is SIP.js the devil that should be avoided at all costs.

There are tradeoffs to be made on both sides, and I've used SIP.js on several occasions and been very happy with my choice.

More specifically, it's very unlikely that your choice of signalling protocol will impact the stability of media flow. The suggestion to switch to Verto in this case seems premature.

On 10 July 2015 at 06:40, Denis Jakovlev <yadenis at seznam.cz> wrote:
Dobr? den,

About a month ago, I asked here about connection problems with the video. And to me strongly suggest it sipjs. I can not change the library every month.
The site freeswitch recommend these 2 libraries (sipjs, jssip). Why can not I use them? 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

p?tek 10. ?ervence 2015, 11:05:07, napsal jste:


My strongest suggestion - switch to mod_verto. All sip implementation for JavaScript is buggy in many different ways. And with every new release of the browser you will get new surprises. 

???????, 10 ???? 2015 ?. ???????????? Denis Jakovlev ???????:
Hi All !

I have a question. I use several different libraries (jssip, sip.js) for RTC Web communications. In both variants I observed during the freezing of the video communication. Sometimes it works great and without freezing. Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've tried various settings Dialplan and internal.xml. But the result is almost the same. 
Maybe dear colleagues will share tips on how to make the connection stable? 

PS: I use FreeSwitch 1.7 on Debian 8. 



-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

_________________________________________________________________________
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


-- 
Best 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 brian at freeswitch.org  Fri Jul 10 18:06:15 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 10 Jul 2015 09:06:15 -0500
Subject: [Freeswitch-users] how to improve the performance of Web RTC?
In-Reply-To: <CAAyX+KFO5FcefkB86vJj8m=8UQuPTZWm4L90MPpX1QQPZHnSNw@mail.gmail.com>
References: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit>
	<69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>
	<CAJTkRNt_u2QjGKy14zZhVG=dE6y7J0hzJJM0uP-LeGRS4k2TDw@mail.gmail.com>
	<588319577.20150710114058@seznam.cz>
	<CAAyX+KFO5FcefkB86vJj8m=8UQuPTZWm4L90MPpX1QQPZHnSNw@mail.gmail.com>
Message-ID: <CAEJMVkDiWkNeM2M4ydjOGfd3G4q72ayRhUHrdohbUmFpg7zpug@mail.gmail.com>

I need to have a POW WOW with the community and devise an automated testing
approach that'll help us and the community.

I think the community would benefit from a collaborative testing frame
work, that allows us to present a set of machines public facing and some
automation to help us test things on a continual basis.

Who's up for the challenge of starting with a clean slate and building a
testing infrastructure from the ground up?



On Fri, Jul 10, 2015 at 8:44 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:

> You absolutely can use SIP.js, and infact I do. There are bugs, of course,
> but all software has bugs. It seems like the scare-mongering about how
> awful SIP in the browser is is more of a marketing effort for Verto than
> reality.
>
> Verto may be great, and given the FreeSWITCH core team's track record of
> quality I don't doubt it, but it's not the great saviour that will fix all
> of your problems while making you a cup of tea, nor is SIP.js the devil
> that should be avoided at all costs.
>
> There are tradeoffs to be made on both sides, and I've used SIP.js on
> several occasions and been very happy with my choice.
>
> More specifically, it's very unlikely that your choice of signalling
> protocol will impact the stability of media flow. The suggestion to switch
> to Verto in this case seems premature.
>
> On 10 July 2015 at 06:40, Denis Jakovlev <yadenis at seznam.cz> wrote:
>
>>  Dobr? den,
>>
>> About a month ago, I asked here about connection problems with the video.
>> And to me strongly suggest it sipjs. I can not change the library every
>> month.
>> The site freeswitch recommend these 2 libraries (sipjs, jssip). Why can
>> not I use them?
>>
>>
>>
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>> <http://mob.tel>. 775-415-382 p?tek 10. ?ervence 2015, 11:05:07, napsal
>> jste: *
>>
>>  My strongest suggestion - switch to mod_verto. All sip implementation
>> for JavaScript is buggy in many different ways. And with every new release
>> of the browser you will get new surprises.
>>
>> ???????, 10 ???? 2015 ?. ???????????? Denis Jakovlev ???????:
>> Hi All !
>>
>> I have a question. I use several different libraries (jssip, sip.js) for
>> RTC Web communications. In both variants I observed during the freezing of
>> the video communication. Sometimes it works great and without freezing.
>> Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've
>> tried various settings Dialplan and internal.xml. But the result is almost
>> the same.
>> Maybe dear colleagues will share tips on how to make the connection
>> stable?
>>
>> PS: I use FreeSwitch 1.7 on Debian 8.
>>
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>>
>> *. 775-415-382 *
>> _________________________________________________________________________
>> 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
>>
>>
>> --
>> Best 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
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
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From bpriddy at bryantschools.org  Fri Jul 10 18:10:29 2015
From: bpriddy at bryantschools.org (Blake Priddy)
Date: Fri, 10 Jul 2015 09:10:29 -0500
Subject: [Freeswitch-users] how to improve the performance of Web RTC?
In-Reply-To: <CAEJMVkDiWkNeM2M4ydjOGfd3G4q72ayRhUHrdohbUmFpg7zpug@mail.gmail.com>
References: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit>
	<69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>
	<CAJTkRNt_u2QjGKy14zZhVG=dE6y7J0hzJJM0uP-LeGRS4k2TDw@mail.gmail.com>
	<588319577.20150710114058@seznam.cz>
	<CAAyX+KFO5FcefkB86vJj8m=8UQuPTZWm4L90MPpX1QQPZHnSNw@mail.gmail.com>
	<CAEJMVkDiWkNeM2M4ydjOGfd3G4q72ayRhUHrdohbUmFpg7zpug@mail.gmail.com>
Message-ID: <CADK-BjttMxqbJzGrSfsXbw01rJwXe3Tvu+oxqqh3cdbGOzctRg@mail.gmail.com>

Brian I would love to be a part of that. I would also like to record it and
post to YouTube such as a video "somewhat tutorial/best practices" about
how this works. I think a freeswitch YouTube channel (which already exists)
that had video about implementation of mods and how all these things work
together would be great for the community! :)
On Jul 10, 2015 9:07 AM, "Brian West" <brian at freeswitch.org> wrote:

> I need to have a POW WOW with the community and devise an automated
> testing approach that'll help us and the community.
>
> I think the community would benefit from a collaborative testing frame
> work, that allows us to present a set of machines public facing and some
> automation to help us test things on a continual basis.
>
> Who's up for the challenge of starting with a clean slate and building a
> testing infrastructure from the ground up?
>
>
>
> On Fri, Jul 10, 2015 at 8:44 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>
>> You absolutely can use SIP.js, and infact I do. There are bugs, of
>> course, but all software has bugs. It seems like the scare-mongering about
>> how awful SIP in the browser is is more of a marketing effort for Verto
>> than reality.
>>
>> Verto may be great, and given the FreeSWITCH core team's track record of
>> quality I don't doubt it, but it's not the great saviour that will fix all
>> of your problems while making you a cup of tea, nor is SIP.js the devil
>> that should be avoided at all costs.
>>
>> There are tradeoffs to be made on both sides, and I've used SIP.js on
>> several occasions and been very happy with my choice.
>>
>> More specifically, it's very unlikely that your choice of signalling
>> protocol will impact the stability of media flow. The suggestion to switch
>> to Verto in this case seems premature.
>>
>> On 10 July 2015 at 06:40, Denis Jakovlev <yadenis at seznam.cz> wrote:
>>
>>>  Dobr? den,
>>>
>>> About a month ago, I asked here about connection problems with the
>>> video. And to me strongly suggest it sipjs. I can not change the library
>>> every month.
>>> The site freeswitch recommend these 2 libraries (sipjs, jssip). Why can
>>> not I use them?
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>>> <http://mob.tel>. 775-415-382 p?tek 10. ?ervence 2015, 11:05:07, napsal
>>> jste: *
>>>
>>>  My strongest suggestion - switch to mod_verto. All sip implementation
>>> for JavaScript is buggy in many different ways. And with every new release
>>> of the browser you will get new surprises.
>>>
>>> ???????, 10 ???? 2015 ?. ???????????? Denis Jakovlev ???????:
>>> Hi All !
>>>
>>> I have a question. I use several different libraries (jssip, sip.js) for
>>> RTC Web communications. In both variants I observed during the freezing of
>>> the video communication. Sometimes it works great and without freezing.
>>> Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've
>>> tried various settings Dialplan and internal.xml. But the result is almost
>>> the same.
>>> Maybe dear colleagues will share tips on how to make the connection
>>> stable?
>>>
>>> PS: I use FreeSwitch 1.7 on Debian 8.
>>>
>>>
>>>
>>>
>>>
>>>
>>> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>>>
>>> *. 775-415-382 *
>>> _________________________________________________________________________
>>> 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
>>>
>>>
>>> --
>>> Best 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
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>
> _________________________________________________________________________
> 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.org  Fri Jul 10 18:13:42 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 10 Jul 2015 09:13:42 -0500
Subject: [Freeswitch-users] how to improve the performance of Web RTC?
In-Reply-To: <CADK-BjttMxqbJzGrSfsXbw01rJwXe3Tvu+oxqqh3cdbGOzctRg@mail.gmail.com>
References: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit>
	<69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>
	<CAJTkRNt_u2QjGKy14zZhVG=dE6y7J0hzJJM0uP-LeGRS4k2TDw@mail.gmail.com>
	<588319577.20150710114058@seznam.cz>
	<CAAyX+KFO5FcefkB86vJj8m=8UQuPTZWm4L90MPpX1QQPZHnSNw@mail.gmail.com>
	<CAEJMVkDiWkNeM2M4ydjOGfd3G4q72ayRhUHrdohbUmFpg7zpug@mail.gmail.com>
	<CADK-BjttMxqbJzGrSfsXbw01rJwXe3Tvu+oxqqh3cdbGOzctRg@mail.gmail.com>
Message-ID: <CAEJMVkBycnGYcMrWGtM8v4Ns9FwnOgvQAZoV=fyNN2EaA9EgRQ@mail.gmail.com>

Maybe I can plan a meeting time next week where we can all get together and
plan out an approach for automated testing that can be done with little to
no human intervention!

Who's up for that? Say next Tuesday?

On Fri, Jul 10, 2015 at 9:10 AM, Blake Priddy <bpriddy at bryantschools.org>
wrote:

> Brian I would love to be a part of that. I would also like to record it
> and post to YouTube such as a video "somewhat tutorial/best practices"
> about how this works. I think a freeswitch YouTube channel (which already
> exists) that had video about implementation of mods and how all these
> things work together would be great for the community! :)
> On Jul 10, 2015 9:07 AM, "Brian West" <brian at freeswitch.org> wrote:
>
>> I need to have a POW WOW with the community and devise an automated
>> testing approach that'll help us and the community.
>>
>> I think the community would benefit from a collaborative testing frame
>> work, that allows us to present a set of machines public facing and some
>> automation to help us test things on a continual basis.
>>
>> Who's up for the challenge of starting with a clean slate and building a
>> testing infrastructure from the ground up?
>>
>>
>>
>> On Fri, Jul 10, 2015 at 8:44 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>>
>>> You absolutely can use SIP.js, and infact I do. There are bugs, of
>>> course, but all software has bugs. It seems like the scare-mongering about
>>> how awful SIP in the browser is is more of a marketing effort for Verto
>>> than reality.
>>>
>>> Verto may be great, and given the FreeSWITCH core team's track record of
>>> quality I don't doubt it, but it's not the great saviour that will fix all
>>> of your problems while making you a cup of tea, nor is SIP.js the devil
>>> that should be avoided at all costs.
>>>
>>> There are tradeoffs to be made on both sides, and I've used SIP.js on
>>> several occasions and been very happy with my choice.
>>>
>>> More specifically, it's very unlikely that your choice of signalling
>>> protocol will impact the stability of media flow. The suggestion to switch
>>> to Verto in this case seems premature.
>>>
>>> On 10 July 2015 at 06:40, Denis Jakovlev <yadenis at seznam.cz> wrote:
>>>
>>>>  Dobr? den,
>>>>
>>>> About a month ago, I asked here about connection problems with the
>>>> video. And to me strongly suggest it sipjs. I can not change the library
>>>> every month.
>>>> The site freeswitch recommend these 2 libraries (sipjs, jssip). Why can
>>>> not I use them?
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>>>> <http://mob.tel>. 775-415-382 p?tek 10. ?ervence 2015, 11:05:07, napsal
>>>> jste: *
>>>>
>>>>  My strongest suggestion - switch to mod_verto. All sip implementation
>>>> for JavaScript is buggy in many different ways. And with every new release
>>>> of the browser you will get new surprises.
>>>>
>>>> ???????, 10 ???? 2015 ?. ???????????? Denis Jakovlev ???????:
>>>> Hi All !
>>>>
>>>> I have a question. I use several different libraries (jssip, sip.js)
>>>> for RTC Web communications. In both variants I observed during the freezing
>>>> of the video communication. Sometimes it works great and without freezing.
>>>> Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've
>>>> tried various settings Dialplan and internal.xml. But the result is almost
>>>> the same.
>>>> Maybe dear colleagues will share tips on how to make the connection
>>>> stable?
>>>>
>>>> PS: I use FreeSwitch 1.7 on Debian 8.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>>>>
>>>> *. 775-415-382 *
>>>> _________________________________________________________________________
>>>> 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
>>>>
>>>>
>>>> --
>>>> Best 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
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> ClueCon 2015 Call for Speakers
>> <https://www.cluecon.com/call-for-speakers/> | Register
>> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>>
>> _________________________________________________________________________
>> 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*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
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From ben at langfeld.co.uk  Fri Jul 10 19:00:09 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Fri, 10 Jul 2015 12:00:09 -0300
Subject: [Freeswitch-users] how to improve the performance of Web RTC?
In-Reply-To: <CAEJMVkDiWkNeM2M4ydjOGfd3G4q72ayRhUHrdohbUmFpg7zpug@mail.gmail.com>
References: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit>
	<69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>
	<CAJTkRNt_u2QjGKy14zZhVG=dE6y7J0hzJJM0uP-LeGRS4k2TDw@mail.gmail.com>
	<588319577.20150710114058@seznam.cz>
	<CAAyX+KFO5FcefkB86vJj8m=8UQuPTZWm4L90MPpX1QQPZHnSNw@mail.gmail.com>
	<CAEJMVkDiWkNeM2M4ydjOGfd3G4q72ayRhUHrdohbUmFpg7zpug@mail.gmail.com>
Message-ID: <CAAyX+KEnJRVwWtg1J4c3st9TJAvv+kVZ18xDC48XVgmEgN-QPQ@mail.gmail.com>

You might want to take a look at http://testrtc.com/ which Tsahi announced
this week. I'll be discussing it with him and his partners in detail on
Tuesday and I'll provide feedback here about how it might be relevant to
FreeSWITCH.

On 10 July 2015 at 11:06, Brian West <brian at freeswitch.org> wrote:

> I need to have a POW WOW with the community and devise an automated
> testing approach that'll help us and the community.
>
> I think the community would benefit from a collaborative testing frame
> work, that allows us to present a set of machines public facing and some
> automation to help us test things on a continual basis.
>
> Who's up for the challenge of starting with a clean slate and building a
> testing infrastructure from the ground up?
>
>
>
> On Fri, Jul 10, 2015 at 8:44 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>
>> You absolutely can use SIP.js, and infact I do. There are bugs, of
>> course, but all software has bugs. It seems like the scare-mongering about
>> how awful SIP in the browser is is more of a marketing effort for Verto
>> than reality.
>>
>> Verto may be great, and given the FreeSWITCH core team's track record of
>> quality I don't doubt it, but it's not the great saviour that will fix all
>> of your problems while making you a cup of tea, nor is SIP.js the devil
>> that should be avoided at all costs.
>>
>> There are tradeoffs to be made on both sides, and I've used SIP.js on
>> several occasions and been very happy with my choice.
>>
>> More specifically, it's very unlikely that your choice of signalling
>> protocol will impact the stability of media flow. The suggestion to switch
>> to Verto in this case seems premature.
>>
>> On 10 July 2015 at 06:40, Denis Jakovlev <yadenis at seznam.cz> wrote:
>>
>>>  Dobr? den,
>>>
>>> About a month ago, I asked here about connection problems with the
>>> video. And to me strongly suggest it sipjs. I can not change the library
>>> every month.
>>> The site freeswitch recommend these 2 libraries (sipjs, jssip). Why can
>>> not I use them?
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>>> <http://mob.tel>. 775-415-382 p?tek 10. ?ervence 2015, 11:05:07, napsal
>>> jste: *
>>>
>>>  My strongest suggestion - switch to mod_verto. All sip implementation
>>> for JavaScript is buggy in many different ways. And with every new release
>>> of the browser you will get new surprises.
>>>
>>> ???????, 10 ???? 2015 ?. ???????????? Denis Jakovlev ???????:
>>> Hi All !
>>>
>>> I have a question. I use several different libraries (jssip, sip.js) for
>>> RTC Web communications. In both variants I observed during the freezing of
>>> the video communication. Sometimes it works great and without freezing.
>>> Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've
>>> tried various settings Dialplan and internal.xml. But the result is almost
>>> the same.
>>> Maybe dear colleagues will share tips on how to make the connection
>>> stable?
>>>
>>> PS: I use FreeSwitch 1.7 on Debian 8.
>>>
>>>
>>>
>>>
>>>
>>>
>>> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>>>
>>> *. 775-415-382 *
>>> _________________________________________________________________________
>>> 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
>>>
>>>
>>> --
>>> Best 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
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>
> _________________________________________________________________________
> 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  Fri Jul 10 19:19:50 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 10 Jul 2015 17:19:50 +0200
Subject: [Freeswitch-users] how to improve the performance of Web RTC?
In-Reply-To: <CAEJMVkBycnGYcMrWGtM8v4Ns9FwnOgvQAZoV=fyNN2EaA9EgRQ@mail.gmail.com>
References: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit>
	<69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>
	<CAJTkRNt_u2QjGKy14zZhVG=dE6y7J0hzJJM0uP-LeGRS4k2TDw@mail.gmail.com>
	<588319577.20150710114058@seznam.cz>
	<CAAyX+KFO5FcefkB86vJj8m=8UQuPTZWm4L90MPpX1QQPZHnSNw@mail.gmail.com>
	<CAEJMVkDiWkNeM2M4ydjOGfd3G4q72ayRhUHrdohbUmFpg7zpug@mail.gmail.com>
	<CADK-BjttMxqbJzGrSfsXbw01rJwXe3Tvu+oxqqh3cdbGOzctRg@mail.gmail.com>
	<CAEJMVkBycnGYcMrWGtM8v4Ns9FwnOgvQAZoV=fyNN2EaA9EgRQ@mail.gmail.com>
Message-ID: <CALXCt0qYS7zAQ8XsOxBRbjyqaeR9VuvNf8w7LFsvireL3Nz_bg@mail.gmail.com>

Brian,
Dunno what help I can be, but I'd be interested in attending the meeting.
Let us know when it will be scheduled.
-giovanni

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On Jul 10, 2015 4:15 PM, "Brian West" <brian at freeswitch.org> wrote:

> Maybe I can plan a meeting time next week where we can all get together
> and plan out an approach for automated testing that can be done with little
> to no human intervention!
>
> Who's up for that? Say next Tuesday?
>
> On Fri, Jul 10, 2015 at 9:10 AM, Blake Priddy <bpriddy at bryantschools.org>
> wrote:
>
>> Brian I would love to be a part of that. I would also like to record it
>> and post to YouTube such as a video "somewhat tutorial/best practices"
>> about how this works. I think a freeswitch YouTube channel (which already
>> exists) that had video about implementation of mods and how all these
>> things work together would be great for the community! :)
>> On Jul 10, 2015 9:07 AM, "Brian West" <brian at freeswitch.org> wrote:
>>
>>> I need to have a POW WOW with the community and devise an automated
>>> testing approach that'll help us and the community.
>>>
>>> I think the community would benefit from a collaborative testing frame
>>> work, that allows us to present a set of machines public facing and some
>>> automation to help us test things on a continual basis.
>>>
>>> Who's up for the challenge of starting with a clean slate and building a
>>> testing infrastructure from the ground up?
>>>
>>>
>>>
>>> On Fri, Jul 10, 2015 at 8:44 AM, Ben Langfeld <ben at langfeld.co.uk>
>>> wrote:
>>>
>>>> You absolutely can use SIP.js, and infact I do. There are bugs, of
>>>> course, but all software has bugs. It seems like the scare-mongering about
>>>> how awful SIP in the browser is is more of a marketing effort for Verto
>>>> than reality.
>>>>
>>>> Verto may be great, and given the FreeSWITCH core team's track record
>>>> of quality I don't doubt it, but it's not the great saviour that will fix
>>>> all of your problems while making you a cup of tea, nor is SIP.js the devil
>>>> that should be avoided at all costs.
>>>>
>>>> There are tradeoffs to be made on both sides, and I've used SIP.js on
>>>> several occasions and been very happy with my choice.
>>>>
>>>> More specifically, it's very unlikely that your choice of signalling
>>>> protocol will impact the stability of media flow. The suggestion to switch
>>>> to Verto in this case seems premature.
>>>>
>>>> On 10 July 2015 at 06:40, Denis Jakovlev <yadenis at seznam.cz> wrote:
>>>>
>>>>>  Dobr? den,
>>>>>
>>>>> About a month ago, I asked here about connection problems with the
>>>>> video. And to me strongly suggest it sipjs. I can not change the library
>>>>> every month.
>>>>> The site freeswitch recommend these 2 libraries (sipjs, jssip). Why
>>>>> can not I use them?
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>>>>> <http://mob.tel>. 775-415-382 p?tek 10. ?ervence 2015, 11:05:07, napsal
>>>>> jste: *
>>>>>
>>>>>  My strongest suggestion - switch to mod_verto. All sip
>>>>> implementation for JavaScript is buggy in many different ways. And with
>>>>> every new release of the browser you will get new surprises.
>>>>>
>>>>> ???????, 10 ???? 2015 ?. ???????????? Denis Jakovlev ???????:
>>>>> Hi All !
>>>>>
>>>>> I have a question. I use several different libraries (jssip, sip.js)
>>>>> for RTC Web communications. In both variants I observed during the freezing
>>>>> of the video communication. Sometimes it works great and without freezing.
>>>>> Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've
>>>>> tried various settings Dialplan and internal.xml. But the result is almost
>>>>> the same.
>>>>> Maybe dear colleagues will share tips on how to make the connection
>>>>> stable?
>>>>>
>>>>> PS: I use FreeSwitch 1.7 on Debian 8.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>>>>>
>>>>> *. 775-415-382 *
>>>>> _________________________________________________________________________
>>>>> 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
>>>>>
>>>>>
>>>>> --
>>>>> Best 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
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> ClueCon 2015 Call for Speakers
>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>>>
>>> _________________________________________________________________________
>>> 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*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>
> _________________________________________________________________________
> 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  Fri Jul 10 19:22:27 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 10 Jul 2015 17:22:27 +0200
Subject: [Freeswitch-users] Send a alert by email when freeswith server
	is down
In-Reply-To: <DM2PR07MB416DCE1CCE63AAC4E5996588C9F0@DM2PR07MB416.namprd07.prod.outlook.com>
References: <DM2PR07MB416DCE1CCE63AAC4E5996588C9F0@DM2PR07MB416.namprd07.prod.outlook.com>
Message-ID: <CALXCt0qNULiy_4CGZcK+J=JL21OrEYJctP2Gi8RWf1BUn8LKvQ@mail.gmail.com>

You can have a script in a crontab that checks freeswitch status, and act
accordingly.

Eg:
fs_cli -x "status"

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On Jul 10, 2015 3:29 PM, "Manish Talwar" <manish.talwar at nexxuspg.com> wrote:

>  Hi,
>
>
>  Is there any way to send an alert by email or SMS to someone whenever
> FreeSwitch is down (either by manually shutdown or server crashed/restart).
>
>
>  Please let me know the steps for its configuration if possible.
>
>
>  Thanks,
>
>
>  Regards,
>
> Manish Talwar
>
> _________________________________________________________________________
> 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 mrjoli021 at gmail.com  Fri Jul 10 19:26:44 2015
From: mrjoli021 at gmail.com (Joli Martinez)
Date: Fri, 10 Jul 2015 11:26:44 -0400
Subject: [Freeswitch-users] freeswitch and fusionpbx
Message-ID: <29DF6A9F-BC9C-409C-B2D0-B10203217F66@gmail.com>

I am running CentOS 6.5 with freeswitch installed on /usr/local/freeswitch.  This was working for a while, I decided to install fusionPBX to get a gui interface.  I ran through the install and I am able to login to FusionPBX but it seems it is not tied to the Freeswitch install.   It does not see my current extensions and when I add extensions on Fusion they do not show up in Freeswitch.  Freeswitch is installed with MySql.

Where do I go to add my current freeswitch install to fusionpbx?

Thanks

From nboric at yx.cl  Fri Jul 10 19:34:36 2015
From: nboric at yx.cl (Neven Boric)
Date: Fri, 10 Jul 2015 12:34:36 -0300
Subject: [Freeswitch-users] Some problems with bypass-media
In-Reply-To: <CAHQ=dc+_XbYXvTAYymG7AGypod1aWq87M9By_WEJU4NUrS74mQ@mail.gmail.com>
References: <CAHQ=dcK0bfZbZ30zt9nYUH1z-R6BnSZBM1W0kAHommJspLKftg@mail.gmail.com>
	<CAFiqYukicT1Af87M6K3KUn-EYTPGGq2qtjWLi2TeAB+=kNodmw@mail.gmail.com>
	<CAHQ=dcJ+JNsU720QdKMznpYxSs0AryYif-MQbswFu2H2VrBAgQ@mail.gmail.com>
	<CAHQ=dc+_XbYXvTAYymG7AGypod1aWq87M9By_WEJU4NUrS74mQ@mail.gmail.com>
Message-ID: <CAHQ=dcK0KJh6hsnfPiUuWEzzWt+6Q8YRSJD+L7gxECwNocLa3A@mail.gmail.com>

I compared the logs between 1.2 and 1.4, and I'm pretty sure in 1.4 one of
the threads is getting stuck until it times out.

Here is the output for 1.4 right after I try to put the call on hold:

2015-07-09 20:08:19.952352 [DEBUG] switch_core_session.c:1061 Send signal
sofia/internal/171 at yx.cl [BREAK]
2015-07-09 20:08:19.952352 [DEBUG] switch_core_session.c:1061 Send signal
sofia/internal/171 at yx.cl [BREAK]
2015-07-09 20:08:19.952352 [DEBUG] sofia.c:6627 Channel sofia/internal/
171 at yx.cl entering state [ready][200]
2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:3680 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:3735 Audio Codec
Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:3596 Set
telephone-event payload to 101
2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:2507 Set Codec
sofia/internal/171 at yx.cl PCMA/8000 20 ms 160 samples 64000 bits 1 channels
2015-07-09 20:08:19.952352 [DEBUG] switch_core_codec.c:111 sofia/internal/
171 at yx.cl Original read codec set to PCMA:8
2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:3943 Set 2833 dtmf
send/recv payload to 101
2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:5179 AUDIO RTP
[sofia/internal/171 at yx.cl] 10.176.0.1 port 21846 -> 10.176.4.121 port 50140
codec: 8 ms: 20
2015-07-09 20:08:19.952352 [DEBUG] switch_rtp.c:3569 Starting timer [soft]
160 bytes per 20ms
2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:5477 Set 2833 dtmf
send payload to 101
2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:5483 Set 2833 dtmf
receive payload to 101
2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:5505 sofia/internal/
171 at yx.cl Set rtp dtmf delay to 40
2015-07-09 20:08:19.972367 [DEBUG] switch_core_session.c:912 Send signal
sofia/internal/140 at 10.50.100.16:5060 [BREAK]
2015-07-09 20:08:30.732483 [CRIT] switch_core_io.c:173 sofia/internal/
140 at 10.50.100.16:5060 reading on a session with no media!

Nothing happens for 10 seconds between 2015-07-09 20:08:19.972367
and 2015-07-09 20:08:30.732483, FS sends no MOH and then decides to hangup
both calls.

And here is the output for 1.2 in a similar situation. It's a different
server with different extensions, but I tried to capture the exact same
moment.


2015-07-09 20:02:28.214025 [DEBUG] switch_core_session.c:1016 Send signal
sofia/internal/sip:3315 at 172.17.100.86:5062 [BREAK]
2015-07-09 20:02:28.214025 [DEBUG] switch_core_session.c:1016 Send signal
sofia/internal/sip:3315 at 172.17.100.86:5062 [BREAK]
2015-07-09 20:02:28.214025 [DEBUG] sofia.c:5815 Channel sofia/internal/
sip:3315 at 172.17.100.86:5062 entering state [ready][200]
2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:5282 Audio Codec Compare
[PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3190 Set Codec
sofia/internal/sip:3315 at 172.17.100.86:5062 PCMA/8000 20 ms 160 samples
64000 bits
2015-07-09 20:02:28.214025 [DEBUG] switch_core_codec.c:111 sofia/internal/
sip:3315 at 172.17.100.86:5062 Original read codec set to PCMA:8
2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:5442 Set 2833 dtmf send
payload to 101
2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3449 AUDIO RTP
[sofia/internal/sip:3315 at 172.17.100.86:5062] 172.30.0.93 port 23690 ->
10.10.1.173 port 50692 codec: 8 ms: 20
2015-07-09 20:02:28.214025 [DEBUG] switch_rtp.c:2040 Starting timer [soft]
160 bytes per 20ms
2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3716 Set 2833 dtmf send
payload to 101
2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3722 Set 2833 dtmf receive
payload to 101
2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3749 sofia/internal/
sip:3315 at 172.17.100.86:5062 Set rtp dtmf delay to 40
2015-07-09 20:02:28.223984 [DEBUG] switch_ivr_bridge.c:1852
(sofia/external/8781 at 10.20.1.125.6:5075) State Change CS_HIBERNATE ->
CS_CONSUME_MEDIA
2015-07-09 20:02:28.223984 [DEBUG] switch_core_state_machine.c:415
(sofia/external/8781 at 10.20.1.125.6:5075) Running State Change
CS_CONSUME_MEDIA
2015-07-09 20:02:28.223984 [DEBUG] switch_core_session.c:933 Send signal
sofia/external/8781 at 10.20.1.125.6:5075 [BREAK]
2015-07-09 20:02:28.223984 [DEBUG] switch_core_session.c:1351 Send signal
sofia/external/8781 at 10.20.1.125.6:5075 [BREAK]
2015-07-09 20:02:28.223984 [DEBUG] switch_ivr_bridge.c:1854 (sofia/internal/
sip:3315 at 172.17.100.86:5062) State Change CS_HIBERNATE -> CS_CONSUME_MEDIA
2015-07-09 20:02:28.223984 [DEBUG] switch_core_session.c:933 Send signal
sofia/internal/sip:3315 at 172.17.100.86:5062 [BREAK]

... lots of other things and then FS plays MOH correctly.

As I mentioned, the same thing happens in master.

Does anyone have an idea of what could be happening? I can make all the
tests and provide any logs necessary.

Thanks

On Thu, Jul 9, 2015 at 11:42 AM, Neven Boric <nboric at yx.cl> wrote:

> Ok, I tried with master, and something strange happened. I'm pretty sure
> it worked last night, but now I'm testing again and I'm getting the same
> behavior I described with 1.4. I don't even have to get a third phone
> involved. All I do is call from A to B, put B on hold and then wait, and
> about ten seconds later, FS will hangup both calls, as if some timeout was
> triggered. I get the same error on the log:
>
> 2015-07-09 14:36:43.506725 [CRIT] switch_core_io.c:93 sofia/internal/
> 140 at 10.50.100.16:5060 reading on a session with no media!
>
> I also get no MOH on the phone that was put on hold.
>
> On Wed, Jul 8, 2015 at 3:41 PM, Neven Boric <nboric at yx.cl> wrote:
>
>>
>>
>> On Wed, Jul 8, 2015 at 2:17 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> - I first tried with the 1.2 branch, as that was the version I was using
>>>> locally. The issue with that version is that when a phone holds and then
>>>> unholds a call, I get no audio on the phone that started the hold. I found
>>>> option bypass-media-after-hold, but realized that it was not included in
>>>> the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93 from
>>>> the master branch. Now FS correctly tries to restore direct media between
>>>> the endpoints with reINVITEs, but unfortunately includes a 'sendonly' in
>>>> the last SDP to the phone that unholds, so I still get no audio. I haven't
>>>> found a way to fix this.
>>>
>>>
>>> Which probably means there's other commits required. 1.2 is EOL and no
>>> longer supported, 1.4 is the current stable release. I wouldn't spend any
>>> time getting 1.2 working, and instead work on upgrading.
>>>
>>
>> Yes, I know it's not currently supported, but maybe somebody had the same
>> problem and fixed it in a different way. Or maybe somebody could point me
>> in the right direction to remove that final 'sendonly'.
>>
>>
>>>
>>> - I also tried with the 1.4 branch and hold/unhold works correctly, but
>>>> now attended transfer doesn't work, FS after some time ends the call on the
>>>> side that was put on hold waiting to be transferred. This time seems to be
>>>> inconsistent, and sometimes the transfer actually works, so maybe it is a
>>>> timing issue. This seems similar/related to FS-4038.
>>>>
>>>
>>> Also see if you can replicate it on master, as that's close to the
>>> upcoming 1.6 release and will have a lot of changes over 1.4.
>>>
>>
>> I will try with master and report back, thanks.
>>
>>
>>>
>>>
>>> On 8 July 2015 at 14:23, Neven Boric <nboric at yx.cl> wrote:
>>>
>>>> Hi,
>>>>
>>>> I have been using FS for a long time on a local server without issues,
>>>> but now I want to move to a remote server to support some new usage
>>>> scenarios. I'm trying to use inbound-bypass-media=true to keep the audio
>>>> out of the server. This mostly works, but I have two different issues,
>>>> depending on which FS version I use:
>>>>
>>>> - I first tried with the 1.2 branch, as that was the version I was
>>>> using locally. The issue with that version is that when a phone holds and
>>>> then unholds a call, I get no audio on the phone that started the hold. I
>>>> found option bypass-media-after-hold, but realized that it was not included
>>>> in the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93
>>>> from the master branch. Now FS correctly tries to restore direct media
>>>> between the endpoints with reINVITEs, but unfortunately includes a
>>>> 'sendonly' in the last SDP to the phone that unholds, so I still get no
>>>> audio. I haven't found a way to fix this.
>>>>
>>>> - I also tried with the 1.4 branch and hold/unhold works correctly, but
>>>> now attended transfer doesn't work, FS after some time ends the call on the
>>>> side that was put on hold waiting to be transferred. This time seems to be
>>>> inconsistent, and sometimes the transfer actually works, so maybe it is a
>>>> timing issue. This seems similar/related to FS-4038.
>>>>
>>>> I can get logs and SIP captures, what would be the preferred way to
>>>> provide them (pcap, inline text, pastebin)? Or maybe it's better to go
>>>> ahead and file a bug?
>>>>
>>>> Best regards
>>>> Neven Boric
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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 gmaruzz at gmail.com  Fri Jul 10 19:36:45 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 10 Jul 2015 17:36:45 +0200
Subject: [Freeswitch-users] freeswitch and fusionpbx
In-Reply-To: <29DF6A9F-BC9C-409C-B2D0-B10203217F66@gmail.com>
References: <29DF6A9F-BC9C-409C-B2D0-B10203217F66@gmail.com>
Message-ID: <CALXCt0pr+Lj8v5WcdrcNj1hag4nDVX0EL-OcUWeg0pifqwPc0w@mail.gmail.com>

Erase your install of both, and use the fusionpbx installer on a fresh
machine, from scratch.

It will install and configure all os needed.

Also, this is not a list for  fusionpbx.

-giovanni

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On Jul 10, 2015 5:27 PM, "Joli Martinez" <mrjoli021 at gmail.com> wrote:

> I am running CentOS 6.5 with freeswitch installed on
> /usr/local/freeswitch.  This was working for a while, I decided to install
> fusionPBX to get a gui interface.  I ran through the install and I am able
> to login to FusionPBX but it seems it is not tied to the Freeswitch
> install.   It does not see my current extensions and when I add extensions
> on Fusion they do not show up in Freeswitch.  Freeswitch is installed with
> MySql.
>
> Where do I go to add my current freeswitch install to fusionpbx?
>
> Thanks
> _________________________________________________________________________
> 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  Fri Jul 10 20:02:44 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 10 Jul 2015 12:02:44 -0400
Subject: [Freeswitch-users] how to improve the performance of Web RTC?
In-Reply-To: <CAAyX+KFO5FcefkB86vJj8m=8UQuPTZWm4L90MPpX1QQPZHnSNw@mail.gmail.com>
References: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit>
	<69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>
	<CAJTkRNt_u2QjGKy14zZhVG=dE6y7J0hzJJM0uP-LeGRS4k2TDw@mail.gmail.com>
	<588319577.20150710114058@seznam.cz>
	<CAAyX+KFO5FcefkB86vJj8m=8UQuPTZWm4L90MPpX1QQPZHnSNw@mail.gmail.com>
Message-ID: <CAEwTsAiTUEKjd5hqxudTQYYRQps8h4K+-rQGGvLuEyAiwNEX-Q@mail.gmail.com>

certainly sip.js is not the devil, but it has some distinct issues we
address in verto.  It's a hack with little value.  Jssip however IS the
devil, everyone using it has problems due to bugs that are fixed in sip.js
that have remained open for many months in jssip with no action.  We should
not be recommending either sip js stack, but I would expect sip.js to
generally be functional for most thingsz

On Friday, July 10, 2015, Ben Langfeld <ben at langfeld.co.uk> wrote:

> You absolutely can use SIP.js, and infact I do. There are bugs, of course,
> but all software has bugs. It seems like the scare-mongering about how
> awful SIP in the browser is is more of a marketing effort for Verto than
> reality.
>
> Verto may be great, and given the FreeSWITCH core team's track record of
> quality I don't doubt it, but it's not the great saviour that will fix all
> of your problems while making you a cup of tea, nor is SIP.js the devil
> that should be avoided at all costs.
>
> There are tradeoffs to be made on both sides, and I've used SIP.js on
> several occasions and been very happy with my choice.
>
> More specifically, it's very unlikely that your choice of signalling
> protocol will impact the stability of media flow. The suggestion to switch
> to Verto in this case seems premature.
>
> On 10 July 2015 at 06:40, Denis Jakovlev <yadenis at seznam.cz
> <javascript:_e(%7B%7D,'cvml','yadenis at seznam.cz');>> wrote:
>
>>  Dobr? den,
>>
>> About a month ago, I asked here about connection problems with the video.
>> And to me strongly suggest it sipjs. I can not change the library every
>> month.
>> The site freeswitch recommend these 2 libraries (sipjs, jssip). Why can
>> not I use them?
>>
>>
>>
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>> <http://mob.tel>. 775-415-382 p?tek 10. ?ervence 2015, 11:05:07, napsal
>> jste: *
>>
>>  My strongest suggestion - switch to mod_verto. All sip implementation
>> for JavaScript is buggy in many different ways. And with every new release
>> of the browser you will get new surprises.
>>
>> ???????, 10 ???? 2015 ?. ???????????? Denis Jakovlev ???????:
>> Hi All !
>>
>> I have a question. I use several different libraries (jssip, sip.js) for
>> RTC Web communications. In both variants I observed during the freezing of
>> the video communication. Sometimes it works great and without freezing.
>> Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've
>> tried various settings Dialplan and internal.xml. But the result is almost
>> the same.
>> Maybe dear colleagues will share tips on how to make the connection
>> stable?
>>
>> PS: I use FreeSwitch 1.7 on Debian 8.
>>
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>>
>> *. 775-415-382 *
>> _________________________________________________________________________
>> 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
>>
>>
>> --
>> Best regards,
>> Igor
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','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
>> <javascript:_e(%7B%7D,'cvml','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 <javascript:;>
> 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 <javascript:;>
> 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  Fri Jul 10 20:03:51 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 10 Jul 2015 12:03:51 -0400
Subject: [Freeswitch-users] how to improve the performance of Web RTC?
In-Reply-To: <1631200356.20150710160112@seznam.cz>
References: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit>
	<69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>
	<CAJTkRNt_u2QjGKy14zZhVG=dE6y7J0hzJJM0uP-LeGRS4k2TDw@mail.gmail.com>
	<588319577.20150710114058@seznam.cz>
	<CAAyX+KFO5FcefkB86vJj8m=8UQuPTZWm4L90MPpX1QQPZHnSNw@mail.gmail.com>
	<1631200356.20150710160112@seznam.cz>
Message-ID: <CAEwTsAh1FO5sANTg=xrigZ09dW9utMYeA-E24uZQXHZtCDT9zQ@mail.gmail.com>

once you get media stream established with webrtc, if you have issues after
that with the media, it likely has nothing at all to do with the signaling
method

On Friday, July 10, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Dobr? den,
>
> Thanks. I also think that perhaps Verto is a good choice, but why all
> break if working. The only thing I want is a little tips that I can do to
> improve.
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 p?tek 10. ?ervence 2015, 15:44:28, napsal
> jste: *
>  You absolutely can use SIP.js, and infact I do. There are bugs, of
> course, but all software has bugs. It seems like the scare-mongering about
> how awful SIP in the browser is is more of a marketing effort for Verto
> than reality.
>
> Verto may be great, and given the FreeSWITCH core team's track record of
> quality I don't doubt it, but it's not the great saviour that will fix all
> of your problems while making you a cup of tea, nor is SIP.js the devil
> that should be avoided at all costs.
>
> There are tradeoffs to be made on both sides, and I've used SIP.js on
> several occasions and been very happy with my choice.
>
> More specifically, it's very unlikely that your choice of signalling
> protocol will impact the stability of media flow. The suggestion to switch
> to Verto in this case seems premature.
>
> On 10 July 2015 at 06:40, Denis Jakovlev <yadenis at seznam.cz
> <javascript:_e(%7B%7D,'cvml','yadenis at seznam.cz');>> wrote:
> Dobr? den,
>
> About a month ago, I asked here about connection problems with the video.
> And to me strongly suggest it sipjs. I can not change the library every
> month.
> The site freeswitch recommend these 2 libraries (sipjs, jssip). Why can
> not I use them?
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
>
>
> *. 775-415-382 p?tek 10. ?ervence 2015, 11:05:07, napsal jste: *
>  My strongest suggestion - switch to mod_verto. All sip implementation
> for JavaScript is buggy in many different ways. And with every new release
> of the browser you will get new surprises.
>
> ???????, 10 ???? 2015 ?. ???????????? Denis Jakovlev ???????:
> Hi All !
>
> I have a question. I use several different libraries (jssip, sip.js) for
> RTC Web communications. In both variants I observed during the freezing of
> the video communication. Sometimes it works great and without freezing.
> Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've
> tried various settings Dialplan and internal.xml. But the result is almost
> the same.
> Maybe dear colleagues will share tips on how to make the connection
> stable?
>
> PS: I use FreeSwitch 1.7 on Debian 8.
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
> *. 775-415-382 *
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','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
> <javascript:_e(%7B%7D,'cvml','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
>
>
> --
> Best regards,
> Igor
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','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
> <javascript:_e(%7B%7D,'cvml','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 <javascript:;>
> 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 <javascript:;>
> 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 mrjoli021 at gmail.com  Fri Jul 10 20:05:37 2015
From: mrjoli021 at gmail.com (Joli Martinez)
Date: Fri, 10 Jul 2015 12:05:37 -0400
Subject: [Freeswitch-users] freeswitch and fusionpbx
In-Reply-To: <CALXCt0pr+Lj8v5WcdrcNj1hag4nDVX0EL-OcUWeg0pifqwPc0w@mail.gmail.com>
References: <29DF6A9F-BC9C-409C-B2D0-B10203217F66@gmail.com>
	<CALXCt0pr+Lj8v5WcdrcNj1hag4nDVX0EL-OcUWeg0pifqwPc0w@mail.gmail.com>
Message-ID: <CANeteie8bo2pTD6smzGzYBCeeXojnoNPTtBXQEgA8ch2VfxPpw@mail.gmail.com>

I have other things running on that box, so adding a new box is out of the
question.  I have fusionpbx working and freeswitch working.  They are just
not seeing each other.  I would like to know what files I need to edit to
have fusionpbx look at the /usr/local/freeswitch directory.

On Fri, Jul 10, 2015 at 11:36 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Erase your install of both, and use the fusionpbx installer on a fresh
> machine, from scratch.
>
> It will install and configure all os needed.
>
> Also, this is not a list for  fusionpbx.
>
> -giovanni
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On Jul 10, 2015 5:27 PM, "Joli Martinez" <mrjoli021 at gmail.com> wrote:
>
>> I am running CentOS 6.5 with freeswitch installed on
>> /usr/local/freeswitch.  This was working for a while, I decided to install
>> fusionPBX to get a gui interface.  I ran through the install and I am able
>> to login to FusionPBX but it seems it is not tied to the Freeswitch
>> install.   It does not see my current extensions and when I add extensions
>> on Fusion they do not show up in Freeswitch.  Freeswitch is installed with
>> MySql.
>>
>> Where do I go to add my current freeswitch install to fusionpbx?
>>
>> Thanks
>> _________________________________________________________________________
>> 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.org  Fri Jul 10 20:16:11 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 10 Jul 2015 11:16:11 -0500
Subject: [Freeswitch-users] how to improve the performance of Web RTC?
In-Reply-To: <CALXCt0qYS7zAQ8XsOxBRbjyqaeR9VuvNf8w7LFsvireL3Nz_bg@mail.gmail.com>
References: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit>
	<69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>
	<CAJTkRNt_u2QjGKy14zZhVG=dE6y7J0hzJJM0uP-LeGRS4k2TDw@mail.gmail.com>
	<588319577.20150710114058@seznam.cz>
	<CAAyX+KFO5FcefkB86vJj8m=8UQuPTZWm4L90MPpX1QQPZHnSNw@mail.gmail.com>
	<CAEJMVkDiWkNeM2M4ydjOGfd3G4q72ayRhUHrdohbUmFpg7zpug@mail.gmail.com>
	<CADK-BjttMxqbJzGrSfsXbw01rJwXe3Tvu+oxqqh3cdbGOzctRg@mail.gmail.com>
	<CAEJMVkBycnGYcMrWGtM8v4Ns9FwnOgvQAZoV=fyNN2EaA9EgRQ@mail.gmail.com>
	<CALXCt0qYS7zAQ8XsOxBRbjyqaeR9VuvNf8w7LFsvireL3Nz_bg@mail.gmail.com>
Message-ID: <CAEJMVkCyHfFe2L8UBLtQzxxdXRSA6QQh-jwyObB3snAsP5QQSQ@mail.gmail.com>

We'll schedule something next week!

On Fri, Jul 10, 2015 at 10:19 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Brian,
> Dunno what help I can be, but I'd be interested in attending the meeting.
> Let us know when it will be scheduled.
> -giovanni
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On Jul 10, 2015 4:15 PM, "Brian West" <brian at freeswitch.org> wrote:
>
>> Maybe I can plan a meeting time next week where we can all get together
>> and plan out an approach for automated testing that can be done with little
>> to no human intervention!
>>
>> Who's up for that? Say next Tuesday?
>>
>> On Fri, Jul 10, 2015 at 9:10 AM, Blake Priddy <bpriddy at bryantschools.org>
>> wrote:
>>
>>> Brian I would love to be a part of that. I would also like to record it
>>> and post to YouTube such as a video "somewhat tutorial/best practices"
>>> about how this works. I think a freeswitch YouTube channel (which already
>>> exists) that had video about implementation of mods and how all these
>>> things work together would be great for the community! :)
>>> On Jul 10, 2015 9:07 AM, "Brian West" <brian at freeswitch.org> wrote:
>>>
>>>> I need to have a POW WOW with the community and devise an automated
>>>> testing approach that'll help us and the community.
>>>>
>>>> I think the community would benefit from a collaborative testing frame
>>>> work, that allows us to present a set of machines public facing and some
>>>> automation to help us test things on a continual basis.
>>>>
>>>> Who's up for the challenge of starting with a clean slate and building
>>>> a testing infrastructure from the ground up?
>>>>
>>>>
>>>>
>>>> On Fri, Jul 10, 2015 at 8:44 AM, Ben Langfeld <ben at langfeld.co.uk>
>>>> wrote:
>>>>
>>>>> You absolutely can use SIP.js, and infact I do. There are bugs, of
>>>>> course, but all software has bugs. It seems like the scare-mongering about
>>>>> how awful SIP in the browser is is more of a marketing effort for Verto
>>>>> than reality.
>>>>>
>>>>> Verto may be great, and given the FreeSWITCH core team's track record
>>>>> of quality I don't doubt it, but it's not the great saviour that will fix
>>>>> all of your problems while making you a cup of tea, nor is SIP.js the devil
>>>>> that should be avoided at all costs.
>>>>>
>>>>> There are tradeoffs to be made on both sides, and I've used SIP.js on
>>>>> several occasions and been very happy with my choice.
>>>>>
>>>>> More specifically, it's very unlikely that your choice of signalling
>>>>> protocol will impact the stability of media flow. The suggestion to switch
>>>>> to Verto in this case seems premature.
>>>>>
>>>>> On 10 July 2015 at 06:40, Denis Jakovlev <yadenis at seznam.cz> wrote:
>>>>>
>>>>>>  Dobr? den,
>>>>>>
>>>>>> About a month ago, I asked here about connection problems with the
>>>>>> video. And to me strongly suggest it sipjs. I can not change the library
>>>>>> every month.
>>>>>> The site freeswitch recommend these 2 libraries (sipjs, jssip). Why
>>>>>> can not I use them?
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>>>>>> <http://mob.tel>. 775-415-382 p?tek 10. ?ervence 2015, 11:05:07, napsal
>>>>>> jste: *
>>>>>>
>>>>>>  My strongest suggestion - switch to mod_verto. All sip
>>>>>> implementation for JavaScript is buggy in many different ways. And with
>>>>>> every new release of the browser you will get new surprises.
>>>>>>
>>>>>> ???????, 10 ???? 2015 ?. ???????????? Denis Jakovlev ???????:
>>>>>> Hi All !
>>>>>>
>>>>>> I have a question. I use several different libraries (jssip, sip.js)
>>>>>> for RTC Web communications. In both variants I observed during the freezing
>>>>>> of the video communication. Sometimes it works great and without freezing.
>>>>>> Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've
>>>>>> tried various settings Dialplan and internal.xml. But the result is almost
>>>>>> the same.
>>>>>> Maybe dear colleagues will share tips on how to make the connection
>>>>>> stable?
>>>>>>
>>>>>> PS: I use FreeSwitch 1.7 on Debian 8.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> *-- S pozdravem, Ing.Denis Jakovlev                           *
>>>>>> mob.tel
>>>>>>
>>>>>> *. 775-415-382 *
>>>>>> _________________________________________________________________________
>>>>>> 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
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Best 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
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> *Brian West*
>>>> brian at freeswitch.org
>>>>
>>>>
>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>> http://www.freeswitchbook.com
>>>> http://www.freeswitchcookbook.com
>>>>
>>>> ClueCon 2015 Call for Speakers
>>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> ClueCon 2015 Call for Speakers
>> <https://www.cluecon.com/call-for-speakers/> | Register
>> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>>
>> _________________________________________________________________________
>> 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*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
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From covici at ccs.covici.com  Fri Jul 10 20:16:51 2015
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 10 Jul 2015 12:16:51 -0400
Subject: [Freeswitch-users] freeswitch and fusionpbx
In-Reply-To: <29DF6A9F-BC9C-409C-B2D0-B10203217F66@gmail.com>
References: <29DF6A9F-BC9C-409C-B2D0-B10203217F66@gmail.com>
Message-ID: <11939.1436545011@ccs.covici.com>

You have to add your extensions over again in fusionpbx -- your previous
install is gone.

Joli Martinez <mrjoli021 at gmail.com> wrote:

> I am running CentOS 6.5 with freeswitch installed on /usr/local/freeswitch.  This was working for a while, I decided to install fusionPBX to get a gui interface.  I ran through the install and I am able to login to FusionPBX but it seems it is not tied to the Freeswitch install.   It does not see my current extensions and when I add extensions on Fusion they do not show up in Freeswitch.  Freeswitch is installed with MySql.
> 
> Where do I go to add my current freeswitch install to fusionpbx?
> 
> Thanks
> _________________________________________________________________________
> 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

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

         John Covici
         covici at ccs.covici.com


From luis.daniel.lucio at gmail.com  Fri Jul 10 20:42:35 2015
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Fri, 10 Jul 2015 12:42:35 -0400
Subject: [Freeswitch-users] freeswitch and fusionpbx
In-Reply-To: <11939.1436545011@ccs.covici.com>
References: <29DF6A9F-BC9C-409C-B2D0-B10203217F66@gmail.com>
	<11939.1436545011@ccs.covici.com>
Message-ID: <CAFLo2QzV6Sc-J139iYHaLSJsGn7EabRqiZFZmUgEzJ8WCXKQYg@mail.gmail.com>

You can try my RPM's
they both install FusionPBX and Freeswitch ready to go
https://okay.com.mx/en/entrepreneurs/install-fusionpbx-in-a-moment.html


Luis Daniel Lucio Quiroz
CISSP, CISM, CISA
Linux, VoIP and much more fun
www.okay.com.mx

Need LCR? Check out LCR for FusionPBX with FreeSWITCH
Need Billing? Check out Billing for FusionPBX with FreeSWITCH

2015-07-10 12:16 GMT-04:00 <covici at ccs.covici.com>:

> You have to add your extensions over again in fusionpbx -- your previous
> install is gone.
>
> Joli Martinez <mrjoli021 at gmail.com> wrote:
>
> > I am running CentOS 6.5 with freeswitch installed on
> /usr/local/freeswitch.  This was working for a while, I decided to install
> fusionPBX to get a gui interface.  I ran through the install and I am able
> to login to FusionPBX but it seems it is not tied to the Freeswitch
> install.   It does not see my current extensions and when I add extensions
> on Fusion they do not show up in Freeswitch.  Freeswitch is installed with
> MySql.
> >
> > Where do I go to add my current freeswitch install to fusionpbx?
> >
> > Thanks
> > _________________________________________________________________________
> > 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
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>          John Covici
>          covici at ccs.covici.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
>
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From krice at freeswitch.org  Fri Jul 10 20:59:26 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 10 Jul 2015 11:59:26 -0500
Subject: [Freeswitch-users] freeswitch and fusionpbx
In-Reply-To: <CAFLo2QzV6Sc-J139iYHaLSJsGn7EabRqiZFZmUgEzJ8WCXKQYg@mail.gmail.com>
Message-ID: <D1C5641E.A1E11%krice@freeswitch.org>

Please DO NOT post unofficial repositories to this list. The community has
not way verify these packages as they do with the official FreeSWITCH
Packages that we release that are signed with the official FreeSWITCH
Package/Repo signing keys.

This is done so that users can be assured they are not getting altered
packages.

Also, for FusionPBX end users should be directed to them as Mark and his
team are the one official source for FusionPBX.

We don?t want any possible SourceForging of packages floating around out
there.

Also the proper place for signatures is that the bottom of the email not
inline/near the top


On 7/10/15, 11:42 AM, "Luis Daniel Lucio Quiroz"
<luis.daniel.lucio at gmail.com> wrote:

> You can try my RPM's
> they both install FusionPBX and Freeswitch ready to go
> https://okay.com.mx/en/entrepreneurs/install-fusionpbx-in-a-moment.html
> 
> 
> 
> 2015-07-10 12:16 GMT-04:00  <covici at ccs.covici.com>:
>> You have to add your extensions over again in fusionpbx -- your previous
>> install is gone.
>> 
>> Joli Martinez <mrjoli021 at gmail.com> wrote:
>> 
>>> > I am running CentOS 6.5 with freeswitch installed on
>>> /usr/local/freeswitch.? This was working for a while, I decided to install
>>> fusionPBX to get a gui interface.? I ran through the install and I am able
>>> to login to FusionPBX but it seems it is not tied to the Freeswitch
>>> install.? ?It does not see my current extensions and when I add extensions
>>> on Fusion they do not show up in Freeswitch.? Freeswitch is installed with
>>> MySql.
>>> >
>>> > Where do I go to add my current freeswitch install to fusionpbx?
>>> >
>>> > Thanks
>>> > _________________________________________________________________________
>>> > 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
>> 
>> --
>> Your life is like a penny.? You're going to lose it.? The question is:
>> How do
>> you spend it?
>> 
>> ? ? ? ? ?John Covici
>> ? ? ? ? ?covici at ccs.covici.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
> 
> 
> 
> _________________________________________________________________________
> 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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From brian at freeswitch.org  Fri Jul 10 21:00:51 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 10 Jul 2015 12:00:51 -0500
Subject: [Freeswitch-users] Testing/Automation Meeting Tuesday July 14th 2PM
	Central
Message-ID: <CAEJMVkCkn0xNAwkop8KXqDkxGLA-4t1ubwWXO_XpbirWvmkcag@mail.gmail.com>

Where: https://cantina.freeswitch.org/conf/

Topic: Discussion about Automated QA and Unit testing framework.

Reply to this thread and we can work on our agenda prior to the call.

Thanks,

-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
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From anthony.minessale at gmail.com  Fri Jul 10 21:14:33 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 10 Jul 2015 12:14:33 -0500
Subject: [Freeswitch-users] how to improve the performance of Web RTC?
In-Reply-To: <CAEJMVkCyHfFe2L8UBLtQzxxdXRSA6QQh-jwyObB3snAsP5QQSQ@mail.gmail.com>
References: <8ff876e8d4a94296909832da06b05581@FIT-EX-04.local.forwardit>
	<69CB68B7-85B2-4199-9AFA-7AAA5575C7CF@seznam.cz>
	<CAJTkRNt_u2QjGKy14zZhVG=dE6y7J0hzJJM0uP-LeGRS4k2TDw@mail.gmail.com>
	<588319577.20150710114058@seznam.cz>
	<CAAyX+KFO5FcefkB86vJj8m=8UQuPTZWm4L90MPpX1QQPZHnSNw@mail.gmail.com>
	<CAEJMVkDiWkNeM2M4ydjOGfd3G4q72ayRhUHrdohbUmFpg7zpug@mail.gmail.com>
	<CADK-BjttMxqbJzGrSfsXbw01rJwXe3Tvu+oxqqh3cdbGOzctRg@mail.gmail.com>
	<CAEJMVkBycnGYcMrWGtM8v4Ns9FwnOgvQAZoV=fyNN2EaA9EgRQ@mail.gmail.com>
	<CALXCt0qYS7zAQ8XsOxBRbjyqaeR9VuvNf8w7LFsvireL3Nz_bg@mail.gmail.com>
	<CAEJMVkCyHfFe2L8UBLtQzxxdXRSA6QQh-jwyObB3snAsP5QQSQ@mail.gmail.com>
Message-ID: <CAKbxfG9pmFv1nWfLEE3iQOdYUvWbbn_eTNY-WzHo1iaz7biVUA@mail.gmail.com>

Try setting this in vars.xml

<X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth=500kb"/>

or 1mb etc.. If you don't set it to something Browser will attempt to use
4mb.  All this does is add a constraint to the sdp to lower the bw.

In Chrome, while you are in a call you can visit chrome://webrtc-internals
and look at the send recv graph.

You can also start your chrome in debug mode from the command line and get
some hints if something is wrong.

Also make sure you update every single day.  We are busy developing and it
will change daily.





On Fri, Jul 10, 2015 at 11:16 AM, Brian West <brian at freeswitch.org> wrote:

> We'll schedule something next week!
>
> On Fri, Jul 10, 2015 at 10:19 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> Brian,
>> Dunno what help I can be, but I'd be interested in attending the meeting.
>> Let us know when it will be scheduled.
>> -giovanni
>>
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> On Jul 10, 2015 4:15 PM, "Brian West" <brian at freeswitch.org> wrote:
>>
>>> Maybe I can plan a meeting time next week where we can all get together
>>> and plan out an approach for automated testing that can be done with little
>>> to no human intervention!
>>>
>>> Who's up for that? Say next Tuesday?
>>>
>>> On Fri, Jul 10, 2015 at 9:10 AM, Blake Priddy <bpriddy at bryantschools.org
>>> > wrote:
>>>
>>>> Brian I would love to be a part of that. I would also like to record it
>>>> and post to YouTube such as a video "somewhat tutorial/best practices"
>>>> about how this works. I think a freeswitch YouTube channel (which already
>>>> exists) that had video about implementation of mods and how all these
>>>> things work together would be great for the community! :)
>>>> On Jul 10, 2015 9:07 AM, "Brian West" <brian at freeswitch.org> wrote:
>>>>
>>>>> I need to have a POW WOW with the community and devise an automated
>>>>> testing approach that'll help us and the community.
>>>>>
>>>>> I think the community would benefit from a collaborative testing frame
>>>>> work, that allows us to present a set of machines public facing and some
>>>>> automation to help us test things on a continual basis.
>>>>>
>>>>> Who's up for the challenge of starting with a clean slate and building
>>>>> a testing infrastructure from the ground up?
>>>>>
>>>>>
>>>>>
>>>>> On Fri, Jul 10, 2015 at 8:44 AM, Ben Langfeld <ben at langfeld.co.uk>
>>>>> wrote:
>>>>>
>>>>>> You absolutely can use SIP.js, and infact I do. There are bugs, of
>>>>>> course, but all software has bugs. It seems like the scare-mongering about
>>>>>> how awful SIP in the browser is is more of a marketing effort for Verto
>>>>>> than reality.
>>>>>>
>>>>>> Verto may be great, and given the FreeSWITCH core team's track record
>>>>>> of quality I don't doubt it, but it's not the great saviour that will fix
>>>>>> all of your problems while making you a cup of tea, nor is SIP.js the devil
>>>>>> that should be avoided at all costs.
>>>>>>
>>>>>> There are tradeoffs to be made on both sides, and I've used SIP.js on
>>>>>> several occasions and been very happy with my choice.
>>>>>>
>>>>>> More specifically, it's very unlikely that your choice of signalling
>>>>>> protocol will impact the stability of media flow. The suggestion to switch
>>>>>> to Verto in this case seems premature.
>>>>>>
>>>>>> On 10 July 2015 at 06:40, Denis Jakovlev <yadenis at seznam.cz> wrote:
>>>>>>
>>>>>>>  Dobr? den,
>>>>>>>
>>>>>>> About a month ago, I asked here about connection problems with the
>>>>>>> video. And to me strongly suggest it sipjs. I can not change the library
>>>>>>> every month.
>>>>>>> The site freeswitch recommend these 2 libraries (sipjs, jssip). Why
>>>>>>> can not I use them?
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> *-- S pozdravem, Ing.Denis Jakovlev
>>>>>>> mob.tel <http://mob.tel>. 775-415-382 p?tek 10. ?ervence 2015, 11:05:07,
>>>>>>> napsal jste: *
>>>>>>>
>>>>>>>  My strongest suggestion - switch to mod_verto. All sip
>>>>>>> implementation for JavaScript is buggy in many different ways. And with
>>>>>>> every new release of the browser you will get new surprises.
>>>>>>>
>>>>>>> ???????, 10 ???? 2015 ?. ???????????? Denis Jakovlev ???????:
>>>>>>> Hi All !
>>>>>>>
>>>>>>> I have a question. I use several different libraries (jssip, sip.js)
>>>>>>> for RTC Web communications. In both variants I observed during the freezing
>>>>>>> of the video communication. Sometimes it works great and without freezing.
>>>>>>> Sometimes it can be frozen for 10-15 seconds. Most often with lags. I've
>>>>>>> tried various settings Dialplan and internal.xml. But the result is almost
>>>>>>> the same.
>>>>>>> Maybe dear colleagues will share tips on how to make the connection
>>>>>>> stable?
>>>>>>>
>>>>>>> PS: I use FreeSwitch 1.7 on Debian 8.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> *-- S pozdravem, Ing.Denis Jakovlev                           *
>>>>>>> mob.tel
>>>>>>>
>>>>>>> *. 775-415-382 *
>>>>>>> _________________________________________________________________________
>>>>>>> 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
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Best 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
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>> *Brian West*
>>>>> brian at freeswitch.org
>>>>>
>>>>>
>>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>> http://www.freeswitchbook.com
>>>>> http://www.freeswitchcookbook.com
>>>>>
>>>>> ClueCon 2015 Call for Speakers
>>>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> ClueCon 2015 Call for Speakers
>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>>>
>>> _________________________________________________________________________
>>> 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*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>
> _________________________________________________________________________
> 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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From s.safarov at gmail.com  Fri Jul 10 21:17:32 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 10 Jul 2015 17:17:32 +0000
Subject: [Freeswitch-users] Preset channel variables
In-Reply-To: <9CA1AD50-924E-4589-9488-1867FE3E7CB1@gmx.net>
References: <9CA1AD50-924E-4589-9488-1867FE3E7CB1@gmx.net>
Message-ID: <CAHtxdDc1sf7ozmroRxFuw_zUKOCfMmScQEwPu_Xi_zL-t+edyA@mail.gmail.com>

Please provide example of dialplan extension where var initialized.

On Fri, Jul 10, 2015, 15:27 Markus B?nke <mbodbg at gmx.net> wrote:

> I?m setting a channel variable in the dial plan, with the set application.
> In rare cases where the originator cancels the call, it happens that the
> set command is not executed and the value of the variable is null. To
> preset the variable I tried set it in vars.xml, but it seems I can only use
> the variable in the dial plan, but it?s not set in the channel variables
> posted by mod_xml_cdr. Is there a proper way to preset channel variables?
>
> 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
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From brian at freeswitch.org  Fri Jul 10 21:34:19 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 10 Jul 2015 12:34:19 -0500
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Friday
	FreeForAll Reminder!
In-Reply-To: <559fd01f3614f_8ab37a7330709bc@resque-worker.6.mail>
References: <559fd01f3614f_8ab37a7330709bc@resque-worker.6.mail>
Message-ID: <CAEJMVkB=4=r7HeQpRz_nTKk5Vo3Y31G3Vg9ppJyHDsMR_nJigA@mail.gmail.com>

Stop by anyone wishing to help on Tuesdays call, I want to get an outline
prepared for testing / automation call.

https://docs.google.com/document/d/1C460sc0SkKvrKrqOXcnmO0NAVhzjoUDdUR7KL9PSotY/edit?usp=sharing

I've started to jot down some things please join me.

/b


On Fri, Jul 10, 2015 at 9:01 AM, Ken Rice <krice at freeswitch.org> wrote:

> FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH
> Friday FreeFor All
> Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled
> Browser and headset, Call sip:888 at conference.freeswitch.org or see
> http://ift.tt/1prwIZL for access info!
> -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch
> Twitter: @FreeSWITCH @ClueCon
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
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From govoiper at gmail.com  Fri Jul 10 22:32:25 2015
From: govoiper at gmail.com (SamyGo)
Date: Fri, 10 Jul 2015 14:32:25 -0400
Subject: [Freeswitch-users] Parking lot with Presence [mod_valet_parking]
Message-ID: <CAJUJwtjmUgbKgLNCSEOAZokYwd6sgdDi-XGhTAe8aWuDHik_eg@mail.gmail.com>

Hi All,
Is it possible to have multi-domain parking lot in auto mode with presence ?

For Example:
 <action application="valet_park" data="${lot_name}@{$domain_name} auto in
${min} ${max}"/>

Thanks,
Sammy
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From jungleboogie0 at gmail.com  Fri Jul 10 22:38:06 2015
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Fri, 10 Jul 2015 11:38:06 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Friday
	FreeForAll Reminder!
In-Reply-To: <CAEJMVkB=4=r7HeQpRz_nTKk5Vo3Y31G3Vg9ppJyHDsMR_nJigA@mail.gmail.com>
References: <559fd01f3614f_8ab37a7330709bc@resque-worker.6.mail>
	<CAEJMVkB=4=r7HeQpRz_nTKk5Vo3Y31G3Vg9ppJyHDsMR_nJigA@mail.gmail.com>
Message-ID: <CAKE2PDv_G1mvz_nT4bjmgmS+Q5J5P9wVLmddnBZd2pmWcyzU_Q@mail.gmail.com>

On 10 July 2015 at 10:34, Brian West <brian at freeswitch.org> wrote:
>
> Stop by anyone wishing to help on Tuesdays call, I want to get an outline prepared for testing / automation call.
>
> https://docs.google.com/document/d/1C460sc0SkKvrKrqOXcnmO0NAVhzjoUDdUR7KL9PSotY/edit?usp=sharing
>
> I've started to jot down some things please join me.
>
> /b


Looks like this is read/view only.

Does freeswitch use some kind of automated build process like jenkins
to test builds?

The freebsd group uses jenkins: https://jenkins.freebsd.org/

If a build fails, an email is sent out to the list with a link to the
log so it can be corrected, and then another email reporting the build
is back to normal.


-- 
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si


From mike at jerris.com  Fri Jul 10 22:45:52 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 10 Jul 2015 14:45:52 -0400
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Friday
	FreeForAll Reminder!
In-Reply-To: <CAKE2PDv_G1mvz_nT4bjmgmS+Q5J5P9wVLmddnBZd2pmWcyzU_Q@mail.gmail.com>
References: <559fd01f3614f_8ab37a7330709bc@resque-worker.6.mail>
	<CAEJMVkB=4=r7HeQpRz_nTKk5Vo3Y31G3Vg9ppJyHDsMR_nJigA@mail.gmail.com>
	<CAKE2PDv_G1mvz_nT4bjmgmS+Q5J5P9wVLmddnBZd2pmWcyzU_Q@mail.gmail.com>
Message-ID: <6F490867-1C04-48AF-9EE3-911825F2930D@jerris.com>


> On Jul 10, 2015, at 2:38 PM, jungle Boogie <jungleboogie0 at gmail.com> wrote:
> 
> On 10 July 2015 at 10:34, Brian West <brian at freeswitch.org> wrote:
>> 
>> Stop by anyone wishing to help on Tuesdays call, I want to get an outline prepared for testing / automation call.
>> 
>> https://docs.google.com/document/d/1C460sc0SkKvrKrqOXcnmO0NAVhzjoUDdUR7KL9PSotY/edit?usp=sharing
>> 
>> I've started to jot down some things please join me.
>> 
>> /b
> 
> 
> Looks like this is read/view only.
> 
> Does freeswitch use some kind of automated build process like jenkins
> to test builds?
> 
> The freebsd group uses jenkins: https://jenkins.freebsd.org/
> 
> If a build fails, an email is sent out to the list with a link to the
> log so it can be corrected, and then another email reporting the build
> is back to normal.
> 
> 

We use a product from Atlassian called bamboo.  Our instance is at https://freeswitch.org/bamboo .  This is the automated system we use to build all of our packages and tarballs currently and will be the system we use to run any automated tests (as well as being able to run them manually).

Mike




From brian at freeswitch.org  Fri Jul 10 22:48:43 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 10 Jul 2015 13:48:43 -0500
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Friday
	FreeForAll Reminder!
In-Reply-To: <CAKE2PDv_G1mvz_nT4bjmgmS+Q5J5P9wVLmddnBZd2pmWcyzU_Q@mail.gmail.com>
References: <559fd01f3614f_8ab37a7330709bc@resque-worker.6.mail>
	<CAEJMVkB=4=r7HeQpRz_nTKk5Vo3Y31G3Vg9ppJyHDsMR_nJigA@mail.gmail.com>
	<CAKE2PDv_G1mvz_nT4bjmgmS+Q5J5P9wVLmddnBZd2pmWcyzU_Q@mail.gmail.com>
Message-ID: <CAEJMVkChtnd+Ud05Lf2Nka8KHa2dq9HJncH+8s9zU1V_wVWFBw@mail.gmail.com>

Try now

https://docs.google.com/document/d/1C460sc0SkKvrKrqOXcnmO0NAVhzjoUDdUR7KL9PSotY/edit?usp=sharing


On Fri, Jul 10, 2015 at 1:38 PM, jungle Boogie <jungleboogie0 at gmail.com>
wrote:

> On 10 July 2015 at 10:34, Brian West <brian at freeswitch.org> wrote:
> >
> > Stop by anyone wishing to help on Tuesdays call, I want to get an
> outline prepared for testing / automation call.
> >
> >
> https://docs.google.com/document/d/1C460sc0SkKvrKrqOXcnmO0NAVhzjoUDdUR7KL9PSotY/edit?usp=sharing
> >
> > I've started to jot down some things please join me.
> >
> > /b
>
>
> Looks like this is read/view only.
>
> Does freeswitch use some kind of automated build process like jenkins
> to test builds?
>
> The freebsd group uses jenkins: https://jenkins.freebsd.org/
>
> If a build fails, an email is sent out to the list with a link to the
> log so it can be corrected, and then another email reporting the build
> is back to normal.
>
>
> --
> -------
> inum: 883510009027723
> sip: jungleboogie at sip2sip.info
> xmpp: jungle-boogie at jit.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
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
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From vipkilla at gmail.com  Fri Jul 10 23:43:07 2015
From: vipkilla at gmail.com (Vik Killa)
Date: Fri, 10 Jul 2015 15:43:07 -0400
Subject: [Freeswitch-users] Parking lot with Presence [mod_valet_parking]
In-Reply-To: <CAJUJwtjmUgbKgLNCSEOAZokYwd6sgdDi-XGhTAe8aWuDHik_eg@mail.gmail.com>
References: <CAJUJwtjmUgbKgLNCSEOAZokYwd6sgdDi-XGhTAe8aWuDHik_eg@mail.gmail.com>
Message-ID: <CAC-LwPNg22ExLs5XiVwzapxnrwUs_5N213YMy2mRceOVxB+ovQ@mail.gmail.com>

yes it is

On Fri, Jul 10, 2015 at 2:32 PM, SamyGo <govoiper at gmail.com> wrote:

> Hi All,
> Is it possible to have multi-domain parking lot in auto mode with presence
> ?
>
> For Example:
>  <action application="valet_park" data="${lot_name}@{$domain_name} auto
> in ${min} ${max}"/>
>
> Thanks,
> Sammy
>
>
>
>
> _________________________________________________________________________
> 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 ssinyagin at gmail.com  Sat Jul 11 03:00:01 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sat, 11 Jul 2015 01:00:01 +0200
Subject: [Freeswitch-users] 1.7 under systemd
Message-ID: <CACOYK=iASWbP121GQynf9HT3Jb6kxQF3uJ3A7OOromMLT=GheA@mail.gmail.com>

If you install the current 1.7 debs on Jessie, the FreeSWITCH process
is automatically restarted by systemd in case of a crash.

Is there a way to tell systemd not to restart the daemon automatically?

I'm completely new to systemd, still need to get used to it.

thanks


From covici at ccs.covici.com  Sat Jul 11 03:15:09 2015
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 10 Jul 2015 19:15:09 -0400
Subject: [Freeswitch-users] 1.7 under systemd
In-Reply-To: <CACOYK=iASWbP121GQynf9HT3Jb6kxQF3uJ3A7OOromMLT=GheA@mail.gmail.com>
References: <CACOYK=iASWbP121GQynf9HT3Jb6kxQF3uJ3A7OOromMLT=GheA@mail.gmail.com>
Message-ID: <27188.1436570109@ccs.covici.com>

You can use systemctl stop freeswitch to stop it at any time,  so you
may want to keep the unit the same, so it will restart in case of a
segfault or something.  Otherwise, copy the unit to /etc/systemd/system
and delete  the Restart line in the service section.

Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> If you install the current 1.7 debs on Jessie, the FreeSWITCH process
> is automatically restarted by systemd in case of a crash.
> 
> Is there a way to tell systemd not to restart the daemon automatically?
> 
> I'm completely new to systemd, still need to get used to it.
> 
> thanks
> 
> _________________________________________________________________________
> 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

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

         John Covici
         covici at ccs.covici.com


From babak.freeswitch at gmail.com  Sat Jul 11 10:16:23 2015
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Sat, 11 Jul 2015 10:46:23 +0430
Subject: [Freeswitch-users] socket application execution
Message-ID: <CAEVfcaZyGrehA1O1-vkKNrtRL-dhu4NbPYV-Jrtd8amkdnAL1Q@mail.gmail.com>

Hi
On what conditions socket application execution ends in dialplan and the
next application execution starts? for example in this dialplan when info
is executed?
<extension name="socket_test">
       <condition field="destination_number" expression="^999$">
           <action application="socket" data="172.16.90.1:8084 async
full"/>
   <action application="info"/>
       </condition>
</extension>
thanks
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From babak.freeswitch at gmail.com  Sat Jul 11 10:48:20 2015
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Sat, 11 Jul 2015 11:18:20 +0430
Subject: [Freeswitch-users] calling intercept from event socket
In-Reply-To: <CAJTkRNte6pJfitXiH_bwh=+-GTrt8h0kKd3wibh5xSzRzjjqtA@mail.gmail.com>
References: <CAEVfcaaCKqme2wKKxaf=ct1eOwQU1Vrtj+eWFYpj0F07vQYKcg@mail.gmail.com>
	<CAJTkRNtZ7wtCkjS7nSfHmMs+y0Eh_sToqDhqwFVoAK9Vs7n3CQ@mail.gmail.com>
	<CAEVfcaYBqeTWreL5hp+3YsAXrDRk6L6Bejt90LcHSwiS7db0pA@mail.gmail.com>
	<CAJTkRNte6pJfitXiH_bwh=+-GTrt8h0kKd3wibh5xSzRzjjqtA@mail.gmail.com>
Message-ID: <CAEVfcaZKeZiJschvVRfpOz4Exx99nZe-CYcN2wMPz=jsTFRFJg@mail.gmail.com>

bindings are working well for me too. the only problem I have here is when
the call is transferred to the same socket app the new socket terminates
with the previous one if I dont set socket_resume=true. and if I set it to
true intercept is not working.

On Tue, Jul 7, 2015 at 9:48 PM, Igor Olhovskiy <igorolhovskiy at gmail.com>
wrote:

> As I've used it with nodejs, bindings works well.
> Or may be separate this processes? Means one is listening, second is
> socket.
>
> 2015-07-07 11:59 GMT+03:00 Babak Yakhchali <babak.freeswitch at gmail.com>:
>
>> when I'm playing moh I'm listening for events. I think if async is
>> removed I can not do something like this? can I?
>>
>> On Tue, Jul 7, 2015 at 12:57 PM, Igor Olhovskiy <igorolhovskiy at gmail.com>
>> wrote:
>>
>>> Try to remove async.
>>>
>>> 2015-07-07 11:21 GMT+03:00 Babak Yakhchali <babak.freeswitch at gmail.com>:
>>>
>>>> Hi
>>>> I'm using event socket to do something like freeswitch callcenter
>>>> module. members(aleg) call and are connected to socket application, this is
>>>> my dialplan:
>>>> <extension name="socket">
>>>>        <condition field="destination_number" expression="^999$">
>>>>      <action application="set" data="socket_resume=true"/>
>>>>              <action application="socket" data="172.16.90.1:8084 async
>>>> full"/>
>>>>   <action application="info"/>
>>>>        </condition>
>>>>      </extension>
>>>> these are executed using socket:
>>>> connect
>>>> myevents
>>>> set park_after_bridge=true
>>>> answer
>>>> playback local_stream://moh
>>>>
>>>> now a new channel is originated to agent(bleg) and on answer I'm
>>>> calling intercept on aleg from socket to bridge it to agent channel. if I
>>>> set socket_resume=true in dialplan, after intercept both legs are stuck
>>>> (dead silent). if I do not set socket_resume=true legs are bridged but
>>>> execution continues through dialplan (I placed info to test this).
>>>> problem is if  socket_resume=false and agent transfers the call to 999
>>>> during the call (same socket app) call hangsup. but if socket_resume=true
>>>> intercept is not working as expected but transfer is done. how can I solve
>>>> this?
>>>> thanks
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>>>>
>>>
>>>
>>>
>>> --
>>> Best 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
>>
>
>
>
> --
> Best 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 godson.g at gmail.com  Sat Jul 11 15:48:07 2015
From: godson.g at gmail.com (Godson Gera)
Date: Sat, 11 Jul 2015 06:48:07 -0500
Subject: [Freeswitch-users] socket application execution
In-Reply-To: <CAEVfcaZyGrehA1O1-vkKNrtRL-dhu4NbPYV-Jrtd8amkdnAL1Q@mail.gmail.com>
References: <CAEVfcaZyGrehA1O1-vkKNrtRL-dhu4NbPYV-Jrtd8amkdnAL1Q@mail.gmail.com>
Message-ID: <CAPTXd0qBVErRuXXPcLaSLHbVgu0HEUkP2AY4ux4Og=E12sAnzg@mail.gmail.com>

When socket program closes connection. FS will move on to next instruction
in dialplan unless you executed some commands in sockets program that
breaks the dialplan flow

On Sat, Jul 11, 2015 at 1:16 AM, Babak Yakhchali <babak.freeswitch at gmail.com
> wrote:

> Hi
> On what conditions socket application execution ends in dialplan and the
> next application execution starts? for example in this dialplan when info
> is executed?
> <extension name="socket_test">
>        <condition field="destination_number" expression="^999$">
>            <action application="socket" data="172.16.90.1:8084 async
> full"/>
>    <action application="info"/>
>        </condition>
> </extension>
> thanks
>
> _________________________________________________________________________
> 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
>



-- 
Thanks & Regards,
Godson Gera
FreeSWTICH Consultant <http://godson.in>
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From mrjoli021 at gmail.com  Sat Jul 11 23:35:39 2015
From: mrjoli021 at gmail.com (Joli Martinez)
Date: Sat, 11 Jul 2015 15:35:39 -0400
Subject: [Freeswitch-users] LAN being blocked
Message-ID: <88633954-656C-4CDC-BBA1-5D84C1BE604B@gmail.com>

Hello,

I have installed Freeswitch and fusionPBX.  My LAN network is 192.168.21.0/24.  PBX is 250.  I am trying to register a phone to Freeswitch.  When I do tcpdump on the LAN interface on port 5060, I see traffic hitting the box from the phone, but on the Freeswitch CLI I see nothing.  I am assuming an ACL is blocking the traffic.  Where can I allow the LAN side to register?

I have tried the acl.xml and added the following and it still doesn't work.  After I did this I reloaded the ACL and also restarted FS.

<list name="lan" default="allow">
      <!-- <node type="deny" cidr="192.168.21.0/24"/> -->
      <node type="allow" cidr="192.168.21.0/24"/>
    </list>

    <!--
        This will traverse the directory adding all users 
        with the cidr= tag to this ACL, when this ACL matches
        the users variables and params apply as if they 
        digest authenticated.
    -->
    <list name="domains" default="deny">
      <!-- domain= is special it scans the domain from the directory to build the ACL -->
      <node type="allow" domain="$${domain}"/>
      <node type="allow" cidr="192.168.21.1/32"/>
      <!-- use cidr= if you wish to allow ip ranges to this domains acl. -->
      <node type="allow" cidr="192.168.21.0/24"/>
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From ssinyagin at gmail.com  Sat Jul 11 23:52:41 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sat, 11 Jul 2015 21:52:41 +0200
Subject: [Freeswitch-users] LAN being blocked
In-Reply-To: <88633954-656C-4CDC-BBA1-5D84C1BE604B@gmail.com>
References: <88633954-656C-4CDC-BBA1-5D84C1BE604B@gmail.com>
Message-ID: <CACOYK=i4f9L_2BrGq86x=0K9UJMA=hvJppQ+OOiumDUP=-FNGA@mail.gmail.com>

What do you see in "sofia status" output? It should list the IP addresses
and ports where it binds.

Also "netstat -an" is useful for troubleshooting.
On Jul 11, 2015 9:37 PM, "Joli Martinez" <mrjoli021 at gmail.com> wrote:

> Hello,
>
> I have installed Freeswitch and fusionPBX.  My LAN network is
> 192.168.21.0/24.  PBX is 250.  I am trying to register a phone to
> Freeswitch.  When I do tcpdump on the LAN interface on port 5060, I see
> traffic hitting the box from the phone, but on the Freeswitch CLI I see
> nothing.  I am assuming an ACL is blocking the traffic.  Where can I allow
> the LAN side to register?
>
> I have tried the acl.xml and added the following and it still doesn't
> work.  After I did this I reloaded the ACL and also restarted FS.
>
> <list name="lan" default="allow">
>       <!-- <node type="deny" cidr="192.168.21.0/24"/> -->
>       <node type="allow" cidr="192.168.21.0/24"/>
>     </list>
>
>     <!--
>         This will traverse the directory adding all users
>         with the cidr= tag to this ACL, when this ACL matches
>         the users variables and params apply as if they
>         digest authenticated.
>     -->
>     <list name="domains" default="deny">
>       <!-- domain= is special it scans the domain from the directory to
> build the ACL -->
>       <node type="allow" domain="$${domain}"/>
>       <node type="allow" cidr="192.168.21.1/32"/>
>       <!-- use cidr= if you wish to allow ip ranges to this domains acl.
> -->
>       <node type="allow" cidr="192.168.21.0/24"/>
>
> _________________________________________________________________________
> 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 mrjoli021 at gmail.com  Sun Jul 12 01:06:57 2015
From: mrjoli021 at gmail.com (Joli Martinez)
Date: Sat, 11 Jul 2015 17:06:57 -0400
Subject: [Freeswitch-users] LAN being blocked
In-Reply-To: <CACOYK=i4f9L_2BrGq86x=0K9UJMA=hvJppQ+OOiumDUP=-FNGA@mail.gmail.com>
References: <88633954-656C-4CDC-BBA1-5D84C1BE604B@gmail.com>
	<CACOYK=i4f9L_2BrGq86x=0K9UJMA=hvJppQ+OOiumDUP=-FNGA@mail.gmail.com>
Message-ID: <CANeteifovS=3D_azmRa5ZvF_EG8QXbCNO2ASTWdv8aR1YUOKSw@mail.gmail.com>

freeswitch at internal> sofia status

                     Name   Type                                       Data
State

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

            192.168.1.250   alias                                   internal
ALIASED

           192.168.21.250   alias                                   internal
ALIASED

                 external profile
sip:mod_sofia at 99.58.100.184:5080 RUNNING
(0)

            internal-ipv6 profile
sip:mod_sofia at 99.58.100.184:6693 RUNNING
(0)

                 internal profile
sip:mod_sofia at 99.58.100.184:6693 RUNNING
(0)

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

3 profiles 2 aliases


freeswitch at internal>


192.168.21.250 is the interface for FS.  192.168.1.250 is Lan2.  LAN2
should not be connected to FS.

How can I fix this?



On Sat, Jul 11, 2015 at 3:52 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> What do you see in "sofia status" output? It should list the IP addresses
> and ports where it binds.
>
> Also "netstat -an" is useful for troubleshooting.
> On Jul 11, 2015 9:37 PM, "Joli Martinez" <mrjoli021 at gmail.com> wrote:
>
>> Hello,
>>
>> I have installed Freeswitch and fusionPBX.  My LAN network is
>> 192.168.21.0/24.  PBX is 250.  I am trying to register a phone to
>> Freeswitch.  When I do tcpdump on the LAN interface on port 5060, I see
>> traffic hitting the box from the phone, but on the Freeswitch CLI I see
>> nothing.  I am assuming an ACL is blocking the traffic.  Where can I allow
>> the LAN side to register?
>>
>> I have tried the acl.xml and added the following and it still doesn't
>> work.  After I did this I reloaded the ACL and also restarted FS.
>>
>> <list name="lan" default="allow">
>>       <!-- <node type="deny" cidr="192.168.21.0/24"/> -->
>>       <node type="allow" cidr="192.168.21.0/24"/>
>>     </list>
>>
>>     <!--
>>         This will traverse the directory adding all users
>>         with the cidr= tag to this ACL, when this ACL matches
>>         the users variables and params apply as if they
>>         digest authenticated.
>>     -->
>>     <list name="domains" default="deny">
>>       <!-- domain= is special it scans the domain from the directory to
>> build the ACL -->
>>       <node type="allow" domain="$${domain}"/>
>>       <node type="allow" cidr="192.168.21.1/32"/>
>>       <!-- use cidr= if you wish to allow ip ranges to this domains acl.
>> -->
>>       <node type="allow" cidr="192.168.21.0/24"/>
>>
>> _________________________________________________________________________
>> 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  Sun Jul 12 01:18:30 2015
From: mike at jerris.com (Michael Jerris)
Date: Sat, 11 Jul 2015 17:18:30 -0400
Subject: [Freeswitch-users] LAN being blocked
In-Reply-To: <CANeteifovS=3D_azmRa5ZvF_EG8QXbCNO2ASTWdv8aR1YUOKSw@mail.gmail.com>
References: <88633954-656C-4CDC-BBA1-5D84C1BE604B@gmail.com>
	<CACOYK=i4f9L_2BrGq86x=0K9UJMA=hvJppQ+OOiumDUP=-FNGA@mail.gmail.com>
	<CANeteifovS=3D_azmRa5ZvF_EG8QXbCNO2ASTWdv8aR1YUOKSw@mail.gmail.com>
Message-ID: <CAEwTsAhhPVA4-FLo71F+aBYrpmtW-QqfEKH_t6T93WgUQr7-aA@mail.gmail.com>

by default we figure out the IPv4 address which would route to the Internet
and we set that to a variable.  you can override this, check out vars.xml,
it should be documented in there

On Saturday, July 11, 2015, Joli Martinez <mrjoli021 at gmail.com> wrote:

> freeswitch at internal> sofia status
>
>                      Name   Type
> Data State
>
>
> =================================================================================================
>
>             192.168.1.250   alias
> internal ALIASED
>
>            192.168.21.250   alias
> internal ALIASED
>
>                  external profile
> sip:mod_sofia at 99.58.100.184:5080 RUNNING (0)
>
>             internal-ipv6 profile
> sip:mod_sofia at 99.58.100.184:6693 RUNNING (0)
>
>                  internal profile
> sip:mod_sofia at 99.58.100.184:6693 RUNNING (0)
>
>
> =================================================================================================
>
> 3 profiles 2 aliases
>
>
> freeswitch at internal>
>
>
> 192.168.21.250 is the interface for FS.  192.168.1.250 is Lan2.  LAN2
> should not be connected to FS.
>
> How can I fix this?
>
>
>
> On Sat, Jul 11, 2015 at 3:52 PM, Stanislav Sinyagin <ssinyagin at gmail.com
> <javascript:_e(%7B%7D,'cvml','ssinyagin at gmail.com');>> wrote:
>
>> What do you see in "sofia status" output? It should list the IP addresses
>> and ports where it binds.
>>
>> Also "netstat -an" is useful for troubleshooting.
>> On Jul 11, 2015 9:37 PM, "Joli Martinez" <mrjoli021 at gmail.com
>> <javascript:_e(%7B%7D,'cvml','mrjoli021 at gmail.com');>> wrote:
>>
>>> Hello,
>>>
>>> I have installed Freeswitch and fusionPBX.  My LAN network is
>>> 192.168.21.0/24.  PBX is 250.  I am trying to register a phone to
>>> Freeswitch.  When I do tcpdump on the LAN interface on port 5060, I see
>>> traffic hitting the box from the phone, but on the Freeswitch CLI I see
>>> nothing.  I am assuming an ACL is blocking the traffic.  Where can I allow
>>> the LAN side to register?
>>>
>>> I have tried the acl.xml and added the following and it still doesn't
>>> work.  After I did this I reloaded the ACL and also restarted FS.
>>>
>>> <list name="lan" default="allow">
>>>       <!-- <node type="deny" cidr="192.168.21.0/24"/> -->
>>>       <node type="allow" cidr="192.168.21.0/24"/>
>>>     </list>
>>>
>>>     <!--
>>>         This will traverse the directory adding all users
>>>         with the cidr= tag to this ACL, when this ACL matches
>>>         the users variables and params apply as if they
>>>         digest authenticated.
>>>     -->
>>>     <list name="domains" default="deny">
>>>       <!-- domain= is special it scans the domain from the directory to
>>> build the ACL -->
>>>       <node type="allow" domain="$${domain}"/>
>>>       <node type="allow" cidr="192.168.21.1/32"/>
>>>       <!-- use cidr= if you wish to allow ip ranges to this domains
>>> acl. -->
>>>       <node type="allow" cidr="192.168.21.0/24"/>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','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
>>> <javascript:_e(%7B%7D,'cvml','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
>> <javascript:_e(%7B%7D,'cvml','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
>> <javascript:_e(%7B%7D,'cvml','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 mrjoli021 at gmail.com  Sun Jul 12 01:35:04 2015
From: mrjoli021 at gmail.com (Joli Martinez)
Date: Sat, 11 Jul 2015 17:35:04 -0400
Subject: [Freeswitch-users] LAN being blocked
In-Reply-To: <CAEwTsAhhPVA4-FLo71F+aBYrpmtW-QqfEKH_t6T93WgUQr7-aA@mail.gmail.com>
References: <88633954-656C-4CDC-BBA1-5D84C1BE604B@gmail.com>
	<CACOYK=i4f9L_2BrGq86x=0K9UJMA=hvJppQ+OOiumDUP=-FNGA@mail.gmail.com>
	<CANeteifovS=3D_azmRa5ZvF_EG8QXbCNO2ASTWdv8aR1YUOKSw@mail.gmail.com>
	<CAEwTsAhhPVA4-FLo71F+aBYrpmtW-QqfEKH_t6T93WgUQr7-aA@mail.gmail.com>
Message-ID: <CANeteic_TuFYM3Qo_Nhoox=JzmfWeWs5WuT9u5WCzLFDjuQQYA@mail.gmail.com>

I changed the vars.xml and the internal.xml restarted Freeswitch and still
shows both IP's

vars.xml

 <!-- <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/> -->

  <X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.21.250"/>

On Sat, Jul 11, 2015 at 5:18 PM, Michael Jerris <mike at jerris.com> wrote:

> by default we figure out the IPv4 address which would route to the
> Internet and we set that to a variable.  you can override this, check out
> vars.xml, it should be documented in there
>
> On Saturday, July 11, 2015, Joli Martinez <mrjoli021 at gmail.com> wrote:
>
>> freeswitch at internal> sofia status
>>
>>                      Name   Type
>> Data State
>>
>>
>> =================================================================================================
>>
>>             192.168.1.250   alias
>> internal ALIASED
>>
>>            192.168.21.250   alias
>> internal ALIASED
>>
>>                  external profile
>> sip:mod_sofia at 99.58.100.184:5080 RUNNING (0)
>>
>>             internal-ipv6 profile
>> sip:mod_sofia at 99.58.100.184:6693 RUNNING (0)
>>
>>                  internal profile
>> sip:mod_sofia at 99.58.100.184:6693 RUNNING (0)
>>
>>
>> =================================================================================================
>>
>> 3 profiles 2 aliases
>>
>>
>> freeswitch at internal>
>>
>>
>> 192.168.21.250 is the interface for FS.  192.168.1.250 is Lan2.  LAN2
>> should not be connected to FS.
>>
>> How can I fix this?
>>
>>
>>
>> On Sat, Jul 11, 2015 at 3:52 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
>> wrote:
>>
>>> What do you see in "sofia status" output? It should list the IP
>>> addresses and ports where it binds.
>>>
>>> Also "netstat -an" is useful for troubleshooting.
>>> On Jul 11, 2015 9:37 PM, "Joli Martinez" <mrjoli021 at gmail.com> wrote:
>>>
>>>> Hello,
>>>>
>>>> I have installed Freeswitch and fusionPBX.  My LAN network is
>>>> 192.168.21.0/24.  PBX is 250.  I am trying to register a phone to
>>>> Freeswitch.  When I do tcpdump on the LAN interface on port 5060, I see
>>>> traffic hitting the box from the phone, but on the Freeswitch CLI I see
>>>> nothing.  I am assuming an ACL is blocking the traffic.  Where can I allow
>>>> the LAN side to register?
>>>>
>>>> I have tried the acl.xml and added the following and it still doesn't
>>>> work.  After I did this I reloaded the ACL and also restarted FS.
>>>>
>>>> <list name="lan" default="allow">
>>>>       <!-- <node type="deny" cidr="192.168.21.0/24"/> -->
>>>>       <node type="allow" cidr="192.168.21.0/24"/>
>>>>     </list>
>>>>
>>>>     <!--
>>>>         This will traverse the directory adding all users
>>>>         with the cidr= tag to this ACL, when this ACL matches
>>>>         the users variables and params apply as if they
>>>>         digest authenticated.
>>>>     -->
>>>>     <list name="domains" default="deny">
>>>>       <!-- domain= is special it scans the domain from the directory
>>>> to build the ACL -->
>>>>       <node type="allow" domain="$${domain}"/>
>>>>       <node type="allow" cidr="192.168.21.1/32"/>
>>>>       <!-- use cidr= if you wish to allow ip ranges to this domains
>>>> acl. -->
>>>>       <node type="allow" cidr="192.168.21.0/24"/>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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 mrjoli021 at gmail.com  Sun Jul 12 04:54:16 2015
From: mrjoli021 at gmail.com (Joli Martinez)
Date: Sat, 11 Jul 2015 20:54:16 -0400
Subject: [Freeswitch-users] freeswitch not listening on 5060
Message-ID: <A5A508B7-3767-4ABE-9D87-562F337E59A9@gmail.com>

Hello,

I installed Freeswitch with fusionPBX.  I am able to get my public SIP trunk to register on port 5080 and when I do a netstat I see it listening on port 5080.  I do not see it listening on port 5060 therefore I cant get any internal extensions to resister.
I have checked the internal.xml and it is set to the internal sip port which in the vars.xml is set to 5060.  What am I missing to get Freeswitch to listen on both 5080 for external and 5060 for internal.  


The box has two interfaces, but only one is supposed to be used for freeswitch.  Also There is an SBC infront of Freeswich.

Thanks,

From luis.daniel.lucio at gmail.com  Sun Jul 12 07:59:25 2015
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Sat, 11 Jul 2015 23:59:25 -0400
Subject: [Freeswitch-users] freeswitch not listening on 5060
In-Reply-To: <A5A508B7-3767-4ABE-9D87-562F337E59A9@gmail.com>
References: <A5A508B7-3767-4ABE-9D87-562F337E59A9@gmail.com>
Message-ID: <CAFLo2Qxno60izsWmN5SuG6atNsx9jjL4NWaZY9TmLMVTcpaeXg@mail.gmail.com>

Please open you google hangouts chat, I sent you a text there

Luis Daniel Lucio Quiroz
CISSP, CISM, CISA
Linux, VoIP and much more fun
www.okay.com.mx

Need LCR? Check out LCR for FusionPBX with FreeSWITCH
Need Billing? Check out Billing for FusionPBX with FreeSWITCH

2015-07-11 20:54 GMT-04:00 Joli Martinez <mrjoli021 at gmail.com>:

> Hello,
>
> I installed Freeswitch with fusionPBX.  I am able to get my public SIP
> trunk to register on port 5080 and when I do a netstat I see it listening
> on port 5080.  I do not see it listening on port 5060 therefore I cant get
> any internal extensions to resister.
> I have checked the internal.xml and it is set to the internal sip port
> which in the vars.xml is set to 5060.  What am I missing to get Freeswitch
> to listen on both 5080 for external and 5060 for internal.
>
>
> The box has two interfaces, but only one is supposed to be used for
> freeswitch.  Also There is an SBC infront of Freeswich.
>
> Thanks,
> _________________________________________________________________________
> 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 ssinyagin at gmail.com  Sun Jul 12 09:50:24 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sun, 12 Jul 2015 07:50:24 +0200
Subject: [Freeswitch-users] LAN being blocked
In-Reply-To: <CANeteic_TuFYM3Qo_Nhoox=JzmfWeWs5WuT9u5WCzLFDjuQQYA@mail.gmail.com>
References: <88633954-656C-4CDC-BBA1-5D84C1BE604B@gmail.com>
	<CACOYK=i4f9L_2BrGq86x=0K9UJMA=hvJppQ+OOiumDUP=-FNGA@mail.gmail.com>
	<CANeteifovS=3D_azmRa5ZvF_EG8QXbCNO2ASTWdv8aR1YUOKSw@mail.gmail.com>
	<CAEwTsAhhPVA4-FLo71F+aBYrpmtW-QqfEKH_t6T93WgUQr7-aA@mail.gmail.com>
	<CANeteic_TuFYM3Qo_Nhoox=JzmfWeWs5WuT9u5WCzLFDjuQQYA@mail.gmail.com>
Message-ID: <CACOYK=giNbyb3++fhTR0iK_Q=D2m=H_yTL59DpM6xgLNMME1cw@mail.gmail.com>

it's the wrong parameter, see the description above it in vars.xml.
The right ones are external_rtp_ip and external_sip_ip


On Sat, Jul 11, 2015 at 11:35 PM, Joli Martinez <mrjoli021 at gmail.com> wrote:
> I changed the vars.xml and the internal.xml restarted Freeswitch and still
> shows both IP's
>
> vars.xml
>
>  <!-- <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/> -->
>
>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.21.250"/>
>
>
> On Sat, Jul 11, 2015 at 5:18 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>> by default we figure out the IPv4 address which would route to the
>> Internet and we set that to a variable.  you can override this, check out
>> vars.xml, it should be documented in there
>>
>> On Saturday, July 11, 2015, Joli Martinez <mrjoli021 at gmail.com> wrote:
>>>
>>> freeswitch at internal> sofia status
>>>
>>>                      Name   Type
>>> Data State
>>>
>>>
>>> =================================================================================================
>>>
>>>             192.168.1.250   alias
>>> internal ALIASED
>>>
>>>            192.168.21.250   alias
>>> internal ALIASED
>>>
>>>                  external profile
>>> sip:mod_sofia at 99.58.100.184:5080 RUNNING (0)
>>>
>>>             internal-ipv6 profile
>>> sip:mod_sofia at 99.58.100.184:6693 RUNNING (0)
>>>
>>>                  internal profile
>>> sip:mod_sofia at 99.58.100.184:6693 RUNNING (0)
>>>
>>>
>>> =================================================================================================
>>>
>>> 3 profiles 2 aliases
>>>
>>>
>>> freeswitch at internal>
>>>
>>>
>>>
>>> 192.168.21.250 is the interface for FS.  192.168.1.250 is Lan2.  LAN2
>>> should not be connected to FS.
>>>
>>> How can I fix this?
>>>
>>>
>>>
>>> On Sat, Jul 11, 2015 at 3:52 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
>>> wrote:
>>>>
>>>> What do you see in "sofia status" output? It should list the IP
>>>> addresses and ports where it binds.
>>>>
>>>> Also "netstat -an" is useful for troubleshooting.
>>>>
>>>> On Jul 11, 2015 9:37 PM, "Joli Martinez" <mrjoli021 at gmail.com> wrote:
>>>>>
>>>>> Hello,
>>>>>
>>>>> I have installed Freeswitch and fusionPBX.  My LAN network is
>>>>> 192.168.21.0/24.  PBX is 250.  I am trying to register a phone to
>>>>> Freeswitch.  When I do tcpdump on the LAN interface on port 5060, I see
>>>>> traffic hitting the box from the phone, but on the Freeswitch CLI I see
>>>>> nothing.  I am assuming an ACL is blocking the traffic.  Where can I allow
>>>>> the LAN side to register?
>>>>>
>>>>> I have tried the acl.xml and added the following and it still doesn't
>>>>> work.  After I did this I reloaded the ACL and also restarted FS.
>>>>>
>>>>> <list name="lan" default="allow">
>>>>>       <!-- <node type="deny" cidr="192.168.21.0/24"/> -->
>>>>>       <node type="allow" cidr="192.168.21.0/24"/>
>>>>>     </list>
>>>>>
>>>>>     <!--
>>>>>         This will traverse the directory adding all users
>>>>>         with the cidr= tag to this ACL, when this ACL matches
>>>>>         the users variables and params apply as if they
>>>>>         digest authenticated.
>>>>>     -->
>>>>>     <list name="domains" default="deny">
>>>>>       <!-- domain= is special it scans the domain from the directory to
>>>>> build the ACL -->
>>>>>       <node type="allow" domain="$${domain}"/>
>>>>>       <node type="allow" cidr="192.168.21.1/32"/>
>>>>>       <!-- use cidr= if you wish to allow ip ranges to this domains
>>>>> acl. -->
>>>>>       <node type="allow" cidr="192.168.21.0/24"/>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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


From mrjoli021 at gmail.com  Mon Jul 13 02:54:30 2015
From: mrjoli021 at gmail.com (Joli Martinez)
Date: Sun, 12 Jul 2015 18:54:30 -0400
Subject: [Freeswitch-users] LAN being blocked
In-Reply-To: <CACOYK=giNbyb3++fhTR0iK_Q=D2m=H_yTL59DpM6xgLNMME1cw@mail.gmail.com>
References: <88633954-656C-4CDC-BBA1-5D84C1BE604B@gmail.com>
	<CACOYK=i4f9L_2BrGq86x=0K9UJMA=hvJppQ+OOiumDUP=-FNGA@mail.gmail.com>
	<CANeteifovS=3D_azmRa5ZvF_EG8QXbCNO2ASTWdv8aR1YUOKSw@mail.gmail.com>
	<CAEwTsAhhPVA4-FLo71F+aBYrpmtW-QqfEKH_t6T93WgUQr7-aA@mail.gmail.com>
	<CANeteic_TuFYM3Qo_Nhoox=JzmfWeWs5WuT9u5WCzLFDjuQQYA@mail.gmail.com>
	<CACOYK=giNbyb3++fhTR0iK_Q=D2m=H_yTL59DpM6xgLNMME1cw@mail.gmail.com>
Message-ID: <CANeteidudboggAr8Hijy9zUYAq-dL92UoCwD+OHdz3UqKPpH1g@mail.gmail.com>

I want my external profile to pick up the public IP.  which it is doing
correctly, but my internal profile to use IP 192.168.21.250 on port 5060.
I have changed the internal.xml to point to 192.168.21.250 and for some
reason the 192.168.1.250 keeps showing up.  Also I need the profile to be
on 5060.  I have also changed the the vars.xml to show 5060 and nothing.  I
have restarted Freeswitch after each change and verified that nothing else
is running on 5060.  The server does have two interfaces and the default gw
is set to 192.168.21.50.  The 192.168.1.250 is only for out of band
management and will not need access to FS.

Sofia status

freeswitch at internal> sofia status

                     Name   Type                                       Data
State

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

            192.168.1.250   alias                                   internal
ALIASED

           192.168.21.250   alias                                   internal
ALIASED

                 external profile           sip:mod_sofia at 1.1.1.1:5080 RUNNING
(0)

            internal-ipv6 profile           sip:mod_sofia at 1.1.1.1:5040 RUNNING
(0)

                 internal profile           sip:mod_sofia at 1.1.1.1:5041 RUNNING
(0)

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

3 profiles 2 aliases


netstat -nat

[root at Switch01 sip_profiles]# netstat -nat | grep 192.168.21.250

tcp        0      0 192.168.21.250:5080         0.0.0.0:*
LISTEN

internal.xml

<!-- port to bind to for sip traffic -->

                <param name="sip-port" value="$${internal_sip_port}"/>

                <!-- <param name="sip-port" value="5060"/> -->

                <param name="dialplan" value="XML"/>

                <param name="dtmf-duration" value="2000"/>

                <param name="inbound-codec-prefs" value=
"$${global_codec_prefs}"/>

                <param name="outbound-codec-prefs" value=
"$${global_codec_prefs}"/>

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

                <!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY
IP ADDRESSES -->

                <!-- <param name="rtp-ip" value="$${local_ip_v4}"/> -->

                <param name="rtp-ip" value="192.168.21.250"/>

                <!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP
ADDRESSES -->

                <!-- <param name="sip-ip" value="$${local_ip_v4}"/> -->

                <param name="sip-ip" value="192.168.21.250"/>


vars.xml

<!-- Internal SIP Profile -->

  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>

  <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>

  <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>

  <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>

  <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>


  <!-- External SIP Profile -->

  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>

  <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>

  <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>

  <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>

  <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/>

On Sun, Jul 12, 2015 at 1:50 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> it's the wrong parameter, see the description above it in vars.xml.
> The right ones are external_rtp_ip and external_sip_ip
>
>
> On Sat, Jul 11, 2015 at 11:35 PM, Joli Martinez <mrjoli021 at gmail.com>
> wrote:
> > I changed the vars.xml and the internal.xml restarted Freeswitch and
> still
> > shows both IP's
> >
> > vars.xml
> >
> >  <!-- <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/> -->
> >
> >   <X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.21.250"/>
> >
> >
> > On Sat, Jul 11, 2015 at 5:18 PM, Michael Jerris <mike at jerris.com> wrote:
> >>
> >> by default we figure out the IPv4 address which would route to the
> >> Internet and we set that to a variable.  you can override this, check
> out
> >> vars.xml, it should be documented in there
> >>
> >> On Saturday, July 11, 2015, Joli Martinez <mrjoli021 at gmail.com> wrote:
> >>>
> >>> freeswitch at internal> sofia status
> >>>
> >>>                      Name   Type
> >>> Data State
> >>>
> >>>
> >>>
> =================================================================================================
> >>>
> >>>             192.168.1.250   alias
> >>> internal ALIASED
> >>>
> >>>            192.168.21.250   alias
> >>> internal ALIASED
> >>>
> >>>                  external profile
> >>> sip:mod_sofia at 99.58.100.184:5080 RUNNING (0)
> >>>
> >>>             internal-ipv6 profile
> >>> sip:mod_sofia at 99.58.100.184:6693 RUNNING (0)
> >>>
> >>>                  internal profile
> >>> sip:mod_sofia at 99.58.100.184:6693 RUNNING (0)
> >>>
> >>>
> >>>
> =================================================================================================
> >>>
> >>> 3 profiles 2 aliases
> >>>
> >>>
> >>> freeswitch at internal>
> >>>
> >>>
> >>>
> >>> 192.168.21.250 is the interface for FS.  192.168.1.250 is Lan2.  LAN2
> >>> should not be connected to FS.
> >>>
> >>> How can I fix this?
> >>>
> >>>
> >>>
> >>> On Sat, Jul 11, 2015 at 3:52 PM, Stanislav Sinyagin <
> ssinyagin at gmail.com>
> >>> wrote:
> >>>>
> >>>> What do you see in "sofia status" output? It should list the IP
> >>>> addresses and ports where it binds.
> >>>>
> >>>> Also "netstat -an" is useful for troubleshooting.
> >>>>
> >>>> On Jul 11, 2015 9:37 PM, "Joli Martinez" <mrjoli021 at gmail.com> wrote:
> >>>>>
> >>>>> Hello,
> >>>>>
> >>>>> I have installed Freeswitch and fusionPBX.  My LAN network is
> >>>>> 192.168.21.0/24.  PBX is 250.  I am trying to register a phone to
> >>>>> Freeswitch.  When I do tcpdump on the LAN interface on port 5060, I
> see
> >>>>> traffic hitting the box from the phone, but on the Freeswitch CLI I
> see
> >>>>> nothing.  I am assuming an ACL is blocking the traffic.  Where can I
> allow
> >>>>> the LAN side to register?
> >>>>>
> >>>>> I have tried the acl.xml and added the following and it still doesn't
> >>>>> work.  After I did this I reloaded the ACL and also restarted FS.
> >>>>>
> >>>>> <list name="lan" default="allow">
> >>>>>       <!-- <node type="deny" cidr="192.168.21.0/24"/> -->
> >>>>>       <node type="allow" cidr="192.168.21.0/24"/>
> >>>>>     </list>
> >>>>>
> >>>>>     <!--
> >>>>>         This will traverse the directory adding all users
> >>>>>         with the cidr= tag to this ACL, when this ACL matches
> >>>>>         the users variables and params apply as if they
> >>>>>         digest authenticated.
> >>>>>     -->
> >>>>>     <list name="domains" default="deny">
> >>>>>       <!-- domain= is special it scans the domain from the directory
> to
> >>>>> build the ACL -->
> >>>>>       <node type="allow" domain="$${domain}"/>
> >>>>>       <node type="allow" cidr="192.168.21.1/32"/>
> >>>>>       <!-- use cidr= if you wish to allow ip ranges to this domains
> >>>>> acl. -->
> >>>>>       <node type="allow" cidr="192.168.21.0/24"/>
> >>>>>
> >>>>>
> >>>>>
> _________________________________________________________________________
> >>>>> 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
>
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From engineerzuhairraza at gmail.com  Mon Jul 13 04:30:40 2015
From: engineerzuhairraza at gmail.com (Zohair Raza)
Date: Mon, 13 Jul 2015 04:30:40 +0400
Subject: [Freeswitch-users] Send a alert by email when freeswith server
	is down
In-Reply-To: <CALXCt0qNULiy_4CGZcK+J=JL21OrEYJctP2Gi8RWf1BUn8LKvQ@mail.gmail.com>
References: <DM2PR07MB416DCE1CCE63AAC4E5996588C9F0@DM2PR07MB416.namprd07.prod.outlook.com>
	<CALXCt0qNULiy_4CGZcK+J=JL21OrEYJctP2Gi8RWf1BUn8LKvQ@mail.gmail.com>
Message-ID: <CAF_PTevqsc2wKfZpRFLAq3HCyJgO13AYUHUkob1-kVdjP5v6Rw@mail.gmail.com>

I would suggest monit, it also tries to bring the service back together
with alert
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From igorolhovskiy at gmail.com  Mon Jul 13 14:40:21 2015
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Mon, 13 Jul 2015 13:40:21 +0300
Subject: [Freeswitch-users] Eavesdrop a call.
In-Reply-To: <CAEJMVkDBKs_cUUkCK3wYQk=sk9MLHM1xygJXo3ZFdEd7MZJ4+g@mail.gmail.com>
References: <CAJTkRNvU3PQRHjFF1MxF1KsCq4L5fo4cZxBtNLFxiSLDvDzdwg@mail.gmail.com>
	<BY2PR12MB0242B06E87CF2434520281BB8A900@BY2PR12MB0242.namprd12.prod.outlook.com>
	<CAJTkRNv_3T9=LQRRCGTsw7BCb+EHMXaNBuyA3wo=8dZ90pJ1yg@mail.gmail.com>
	<CAEJMVkDBKs_cUUkCK3wYQk=sk9MLHM1xygJXo3ZFdEd7MZJ4+g@mail.gmail.com>
Message-ID: <CAJTkRNuyG2hFJGfObVcw4bgmpTv=eA+EWZ7XbYhG8ZwMu11b+g@mail.gmail.com>

Tested again....
With
*eavesdrop all*
I got same error:
*2015-07-13 12:38:45.125634 [ERR] switch_core_io.c:1531 Write Buffer 0
bytes Failed!*


2015-07-09 23:11 GMT+03:00 Brian West <brian at freeswitch.org>:

> You can also just say 'all', and cycle thru them all see the confluence
> page over this its fairly detailed.
>
> On Thu, Jul 9, 2015 at 11:17 AM, Igor Olhovskiy <igorolhovskiy at gmail.com>
> wrote:
>
>> Thanks for an answer, but while calling eavesdrop I'm giving as a
>> parameter existing call-leg ID (answered one)
>> But I'll try with originate one command, thanks.
>>
>> ???????, 9 ???? 2015 ?. ???????????? Steven Szeto ???????:
>>
>>>  Igor,
>>>
>>>
>>>
>>> You need a call leg to perform the eavesdrop. The <channel-uuid> is the
>>> channel of an active call that is to be eavesdropped upon.
>>>
>>>
>>>
>>> These examples work from the fs_cli command window. You will have to
>>> modify the commands with extensions and the correct uuid:
>>>
>>>
>>>
>>> Step 1: establish a 2 party call:
>>>
>>>
>>>
>>> two party call:
>>>
>>> originate sofia/internal/5401 at 10.47.41.109 &bridge(sofia/internal/
>>> 5901 at 10.47.41.109)
>>>
>>>
>>>
>>> Step 2: invoke eavesdrop with the desired behavior:
>>>
>>>
>>>
>>> whisper/coach:
>>>
>>> originate sofia/internal/5902 at 10.47.41.109 'queue_dtmf:w2 at 500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861'
>>> inline
>>>
>>>
>>>
>>> originate sofia/internal/4901 at 10.47.32.159 'queue_dtmf:w2 at 500,eavesdrop:35b70871-78d4-4587-a186-5c0457482515'
>>> inline
>>>
>>>
>>>
>>> silent monitor:
>>>
>>> originate sofia/internal/5401 at 10.47.41.109
>>> 'eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861' inline
>>>
>>> originate sofia/internal/5002 at 10.47.32.159 'queue_dtmf:w0 at 500,eavesdrop:abefa174-12dc-4ccb-956e-67c51475c414'
>>> inline
>>>
>>>
>>>
>>>
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Igor
>>> Olhovskiy
>>> *Sent:* Thursday, July 09, 2015 7:48 AM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* [Freeswitch-users] Eavesdrop a call.
>>>
>>>
>>>
>>> Hi!
>>>
>>> I'm using Freeswitch 1.4.20 (git b95362f 2015-07-03 16:42:15Z 64bit))
>>> (stable one)
>>>
>>> and trying to get eavesdropping. But no luck.
>>>
>>> Main scheme:
>>>
>>> DID - > extension (210). 210 answers
>>>
>>> eavesdropping from extension 110.
>>>
>>> While giving a command
>>>
>>> *eavesdrop <channel-uuid> *
>>>
>>> I've got log full of messages
>>>
>>> *2015-07-09 13:29:00.325667 [ERR] switch_core_io.c:1531 Write Buffer 0
>>> bytes Failed!*
>>>
>>> and silence on 110 extension
>>>
>>>
>>>
>>> While giving a command
>>>
>>> *userspy 210@${domain_name}* or *userspy <channel-uuid>*
>>>
>>> I've got music-on-hold on 110, no errors on log.
>>>
>>>
>>>
>>> In both cases used <channel-uuid> of extension 210 (answered one)
>>>
>>>
>>>
>>> Is this a bug or I'm doing something wrong?
>>>
>>> --
>>>
>>> Best regards,
>>>
>>> Igor
>>>
>>> ------------------------------
>>> NOTE: This e-mail (including any attachments) is for the sole use of the
>>> intended recipient(s) and may contain information that is confidential
>>> and/or protected by legal privilege. Any unauthorized review, use, copy,
>>> disclosure or distribution of this e-mail is strictly prohibited. If you
>>> are not the intended recipient, please notify Mitel immediately and destroy
>>> all copies of this e-mail. Mitel does not accept any liability for breach
>>> of security, error or virus that may result from the transmission of this
>>> message.
>>>
>>>
>>> _________________________________________________________________________
>>> 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
>>>
>>
>>
>> --
>> Best 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
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>
> _________________________________________________________________________
> 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
>



-- 
Best regards,
Igor
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From achinthau at gmail.com  Mon Jul 13 16:38:39 2015
From: achinthau at gmail.com (Achintha)
Date: Mon, 13 Jul 2015 18:08:39 +0530
Subject: [Freeswitch-users] profile context is not overridden
Message-ID: <CA+JwzxtSkHASHOeteOxRozH8fQwBCk2eYO_G36cmAb+43LwFGA@mail.gmail.com>

hi all,


We are having freeswitch testing setup with Internal and external profiles
with xml_curl for Directory management. In this setup profile context is
not overridden with user_context property.

Kindly help me to solve this issue.



Thanking you.


Achintha
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From steveayre at gmail.com  Mon Jul 13 18:33:24 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 13 Jul 2015 15:33:24 +0100
Subject: [Freeswitch-users] profile context is not overridden
In-Reply-To: <CA+JwzxtSkHASHOeteOxRozH8fQwBCk2eYO_G36cmAb+43LwFGA@mail.gmail.com>
References: <CA+JwzxtSkHASHOeteOxRozH8fQwBCk2eYO_G36cmAb+43LwFGA@mail.gmail.com>
Message-ID: <CAFiqYunofjieNLW0NLfHAku+Pfx=3eqj-SWvDYF1oVS2exVW0w@mail.gmail.com>

First off verify the call is getting authenticated as the user. If you
examine the XML CDR or the 'info' output do you see the user_context
variable set?

On 13 July 2015 at 13:38, Achintha <achinthau at gmail.com> wrote:

> hi all,
>
>
> We are having freeswitch testing setup with Internal and external profiles
> with xml_curl for Directory management. In this setup profile context is
> not overridden with user_context property.
>
> Kindly help me to solve this issue.
>
>
>
> Thanking you.
>
>
> Achintha
>
> _________________________________________________________________________
> 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 yadenis at seznam.cz  Mon Jul 13 21:19:45 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Mon, 13 Jul 2015 19:19:45 +0200
Subject: [Freeswitch-users] Some automation for uuid_video_refresh?
In-Reply-To: <CAFiqYunofjieNLW0NLfHAku+Pfx=3eqj-SWvDYF1oVS2exVW0w@mail.gmail.com>
References: <CA+JwzxtSkHASHOeteOxRozH8fQwBCk2eYO_G36cmAb+43LwFGA@mail.gmail.com>
	<CAFiqYunofjieNLW0NLfHAku+Pfx=3eqj-SWvDYF1oVS2exVW0w@mail.gmail.com>
Message-ID: <5D371568-3A98-4F04-9D36-6D7B85A12150@seznam.cz>

Hi all!

Order to avoid lags quite not bad helps ?uuid_video_refresh?. Is there a possibility to automate this process? For example the command is run when one of the parties to freeze the video (how?). Or let's say a timer every 5 seconds (how?). Or maybe there are some system functions or commands to do uuid_video_refresh some other way?


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

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From kathleen.king at quentustech.com  Mon Jul 13 21:46:10 2015
From: kathleen.king at quentustech.com (Kathleen King)
Date: Mon, 13 Jul 2015 10:46:10 -0700
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master
	Branch)	June 27th-July 3rd
In-Reply-To: <12443.1436241862@ccs.covici.com>
References: <559b3f8745bc8_8b6610113186543e@resque-worker.5.mail>
	<12443.1436241862@ccs.covici.com>
Message-ID: <A07853D3-C728-4FB5-AC1E-CACD90CAB6A0@quentustech.com>

A nightmare transfer is where the call leg you are trying to transfer isn't on the box. 

On July 6, 2015 9:04:22 PM PDT, covici at ccs.covici.com wrote:
>What the heck is a nightmare transfer?
>
>Ken Rice <krice at freeswitch.org> wrote:
>
>> New Post on freeswitch.org from Kathleen King
>> check it out at http://ift.tt/1KIMIBl
>> FreeSWITCH Week in Review (Master Branch) June 27th-July 3rd
>> Hello, again. This passed week in the FreeSWITCH master branch we had
>49 commits. This week we had a bunch of new features with most of them
>being helpful little improvements, but we also had two new modules
>merged in! The 2600hz guys added mod_kazoo and William King merged in
>mod_smpp. You can find out more about mod_smpp by going here. And, the
>2600hz patches are all slated to be merged in by the 1.6 release.
>> 
>> Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And
>head over to freeswitch.com to learn more about FreeSWITCH support.
>> 
>> New features that were added:
>> 
>> FS-7732 Continue recording with uuid_transfer
>> 
>> FS-7752 [mod_rayo] Increase maximum number of elements from 30 to
>1024 to allow adhearsion to create large grammars to navigate IVR
>menus.
>> 
>> FS-7750 [mod_commands] Allow for uuid_setvar to handle arrays
>> 
>> FS-7758 [mod_loopback] Emit an event if a loopback bowout occurs
>> 
>> FS-7759 [mod_sofia] Added the channel variable
>ignore_completed_elsewhere to suppress setting the completed elsewhere
>cause
>> 
>> FS-7771 Set a channel variable if the recording is terminated due to
>silence hits
>> 
>> FS-7760 Added xml fetch for channels to externally support nightmare
>transfer depends on channel-xml-fetch-on-nightmare-transfer profile
>param (default is disabled)
>> 
>> FS-7730 [mod_smpp] Added mod_smpp as an event handler module
>> and fixed the default configs to provided sample load option for
>mod_sms and mod_smpp
>> 
>> FS-7774 Add mod_kazoo
>> 
>> Improvements in build system, cross platform support, and packaging:
>> 
>> OPENZAP-238 [freetdm] Fix some GSM compilation errors and do a bit of
>code cleanup
>> 
>> OPENZAP-237 [freetdm] Use __func__ instead of __FUNCTION__ to comply
>with c99 in gcc 5.1
>> 
>> The following bugs were squashed:
>> 
>> FS-7734 [mod_nibblebill] Fixed a deadlock
>> 
>> FS-7726 Fixed a bug with recording a video session on DTMF command
>> 
>> FS-7721 Fixed a segfault caused when using session:recordFile() and
>session:unsetInputCallback in a lua script
>> 
>> FS-7429 [mod_curl] Fixed to output valid json
>> 
>> FS-7746 [mod_verto] Fixed a device permission error in verto client
>> 
>> FS-7753 [mod_local_stream] Fixed some glitching and freezing video
>when using hold/unhold
>> 
>> FS-7761 [core] Fix shutdown races running api commands during
>shutdown
>> 
>> FS-7767 [mod_sofia] Fixed a segfault caused by invalid arguments to
>sip_dig
>> 
>> FS-7744 [mod_conference] Fixed a bug causing the first user?s video
>stream to stop when another verto user calls the conference
>> 
>> FS-7486 [mod_sofia] Fixed the handling of queued requests
>> 
>> FS-7775 [mod_conference] Fix threading issue causing stuck worker
>threads
>> 
>> FS-7777 [mod_imagick] Fixed a regression causing a segfault when
>playing png &amp; pdf in conference
>> 
>> And, this passed week in the FreeSWITCH 1.4 branch we had 2 commits
>merged in from master.
>> 
>> FS-7486 [mod_sofia] Fixed the handling of queued requests
>> 
>> FS-7750 [mod_commands] Set uuid_setvar to handle arrays
>> 
>> 
>> ----------------------------------------------------
>> Alternatives:
>> 
>> ----------------------------------------------------
>>
>_________________________________________________________________________
>> 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
>-- 
>Your life is like a penny.  You're going to lose it.  The question is:
>How do
>you spend it?
>
>         John Covici
>         covici at ccs.covici.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

-- 
Sent from my Android device with K-9 Mail. Please excuse my brevity.
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From vipkilla at gmail.com  Mon Jul 13 22:04:08 2015
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 13 Jul 2015 14:04:08 -0400
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch)
 June 27th-July 3rd
In-Reply-To: <A07853D3-C728-4FB5-AC1E-CACD90CAB6A0@quentustech.com>
References: <559b3f8745bc8_8b6610113186543e@resque-worker.5.mail>
	<12443.1436241862@ccs.covici.com>
	<A07853D3-C728-4FB5-AC1E-CACD90CAB6A0@quentustech.com>
Message-ID: <CAC-LwPPFBq5RqCUa_kzQbntFvnmF3Jh0WEPBnSjNvpwxjmMymg@mail.gmail.com>

Is there any documentation on how to use the new nightmare transfer binding?
I added it to my xml_curl binding and setup a test nightmare xfer but my
xml_curl server was never hit with a request from FS.
I'm assuming the XML returned is supposed to tell FS where to find the real
channel.

On Mon, Jul 13, 2015 at 1:46 PM, Kathleen King <
kathleen.king at quentustech.com> wrote:

> A nightmare transfer is where the call leg you are trying to transfer
> isn't on the box.
>
> On July 6, 2015 9:04:22 PM PDT, covici at ccs.covici.com wrote:
>
>> What the heck is a nightmare transfer?
>>
>> Ken Rice <krice at freeswitch.org> wrote:
>>
>>  New Post on freeswitch.org from Kathleen King
>>>  check it out at http://ift.tt/1KIMIBl
>>>  FreeSWITCH Week in Review (Master Branch) June 27th-July 3rd
>>>  Hello, again. This passed week in the FreeSWITCH master branch we had 49 commits. This week we had a bunch of new features with most of them being helpful little improvements, but we also had two new modules merged in! The 2600hz guys added mod_kazoo and William King merged in mod_smpp. You can find out more about mod_smpp by going here. And, the 2600hz patches are all slated to be merged in by the 1.6 release.
>>>
>>>  Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support.
>>>
>>>  New features that were added:
>>>
>>>  FS-7732 Continue recording with uuid_transfer
>>>
>>>  FS-7752 [mod_rayo] Increase maximum number of elements from 30 to 1024 to allow adhearsion to create large grammars to navigate IVR menus.
>>>
>>>  FS-7750 [mod_commands] Allow for uuid_setvar to handle arrays
>>>
>>>  FS-7758 [mod_loopback] Emit an event if a loopback bowout occurs
>>>
>>>  FS-7759 [mod_sofia] Added the channel variable ignore_completed_elsewhere to suppress setting the completed elsewhere cause
>>>
>>>  FS-7771 Set a channel variable if the recording is terminated due to silence hits
>>>
>>>  FS-7760 Added xml fetch for channels to externally support nightmare transfer depends on channel-xml-fetch-on-nightmare-transfer profile param (default is disabled)
>>>
>>>  FS-7730 [mod_smpp] Added mod_smpp as an event handler module
>>>  and fixed
>>> the default configs to provided sample load option for mod_sms and mod_smpp
>>>
>>>  FS-7774 Add mod_kazoo
>>>
>>>  Improvements in build system, cross platform support, and packaging:
>>>
>>>  OPENZAP-238 [freetdm] Fix some GSM compilation errors and do a bit of code cleanup
>>>
>>>  OPENZAP-237 [freetdm] Use __func__ instead of __FUNCTION__ to comply with c99 in gcc 5.1
>>>
>>>  The following bugs were squashed:
>>>
>>>  FS-7734 [mod_nibblebill] Fixed a deadlock
>>>
>>>  FS-7726 Fixed a bug with recording a video session on DTMF command
>>>
>>>  FS-7721 Fixed a segfault caused when using session:recordFile() and session:unsetInputCallback in a lua script
>>>
>>>  FS-7429 [mod_curl] Fixed to output valid json
>>>
>>>  FS-7746 [mod_verto] Fixed a device permission error in verto client
>>>
>>>  FS-7753 [mod_local_stream] Fixed some glitching and freezing video when using hold/unhold
>>>
>>>  FS-7761 [core] Fix shutdown races running
>>> api commands during shutdown
>>>
>>>  FS-7767 [mod_sofia] Fixed a segfault caused by invalid arguments to sip_dig
>>>
>>>  FS-7744 [mod_conference] Fixed a bug causing the first user?s video stream to stop when another verto user calls the conference
>>>
>>>  FS-7486 [mod_sofia] Fixed the handling of queued requests
>>>
>>>  FS-7775 [mod_conference] Fix threading issue causing stuck worker threads
>>>
>>>  FS-7777 [mod_imagick] Fixed a regression causing a segfault when playing png &amp; pdf in conference
>>>
>>>  And, this passed week in the FreeSWITCH 1.4 branch we had 2 commits merged in from master.
>>>
>>>  FS-7486 [mod_sofia] Fixed the handling of queued requests
>>>
>>>  FS-7750 [mod_commands] Set uuid_setvar to handle arrays
>>>
>>>
>>> ------------------------------
>>>
>>>  Alternatives:
>>>
>>> ------------------------------
>>>
>>> ------------------------------
>>>
>>>  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
>>>
>> --
>> Your life is like a penny.  You're going to lose it.  The question is:
>> How do
>> you spend it?
>>
>>          John Covici
>>         covici at ccs.covici.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
>>
>>
> --
> Sent from my Android device with K-9 Mail. Please excuse my brevity.
>
> _________________________________________________________________________
> 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
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From nboric at yx.cl  Tue Jul 14 00:47:48 2015
From: nboric at yx.cl (Neven Boric)
Date: Mon, 13 Jul 2015 17:47:48 -0300
Subject: [Freeswitch-users] Some problems with bypass-media
In-Reply-To: <CAHQ=dcK0KJh6hsnfPiUuWEzzWt+6Q8YRSJD+L7gxECwNocLa3A@mail.gmail.com>
References: <CAHQ=dcK0bfZbZ30zt9nYUH1z-R6BnSZBM1W0kAHommJspLKftg@mail.gmail.com>
	<CAFiqYukicT1Af87M6K3KUn-EYTPGGq2qtjWLi2TeAB+=kNodmw@mail.gmail.com>
	<CAHQ=dcJ+JNsU720QdKMznpYxSs0AryYif-MQbswFu2H2VrBAgQ@mail.gmail.com>
	<CAHQ=dc+_XbYXvTAYymG7AGypod1aWq87M9By_WEJU4NUrS74mQ@mail.gmail.com>
	<CAHQ=dcK0KJh6hsnfPiUuWEzzWt+6Q8YRSJD+L7gxECwNocLa3A@mail.gmail.com>
Message-ID: <CAHQ=dc+1Qz0uLTjV-N1bB8nOKiZgkQW3=fnaOZ1vvFszcaQJ8A@mail.gmail.com>

Anyone, please? It seems very strange that I'm the only one having this
issue. The bypass-media option doesn't seem to be so unpopular.

On Fri, Jul 10, 2015 at 12:34 PM, Neven Boric <nboric at yx.cl> wrote:

> I compared the logs between 1.2 and 1.4, and I'm pretty sure in 1.4 one of
> the threads is getting stuck until it times out.
>
> Here is the output for 1.4 right after I try to put the call on hold:
>
> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/171 at yx.cl [BREAK]
> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/171 at yx.cl [BREAK]
> 2015-07-09 20:08:19.952352 [DEBUG] sofia.c:6627 Channel sofia/internal/
> 171 at yx.cl entering state [ready][200]
> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:3680 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:3735 Audio Codec
> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:3596 Set
> telephone-event payload to 101
> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:2507 Set Codec
> sofia/internal/171 at yx.cl PCMA/8000 20 ms 160 samples 64000 bits 1 channels
> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_codec.c:111 sofia/internal/
> 171 at yx.cl Original read codec set to PCMA:8
> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:3943 Set 2833 dtmf
> send/recv payload to 101
> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:5179 AUDIO RTP
> [sofia/internal/171 at yx.cl] 10.176.0.1 port 21846 -> 10.176.4.121 port
> 50140 codec: 8 ms: 20
> 2015-07-09 20:08:19.952352 [DEBUG] switch_rtp.c:3569 Starting timer [soft]
> 160 bytes per 20ms
> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:5477 Set 2833 dtmf
> send payload to 101
> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:5483 Set 2833 dtmf
> receive payload to 101
> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:5505 sofia/internal/
> 171 at yx.cl Set rtp dtmf delay to 40
> 2015-07-09 20:08:19.972367 [DEBUG] switch_core_session.c:912 Send signal
> sofia/internal/140 at 10.50.100.16:5060 [BREAK]
> 2015-07-09 20:08:30.732483 [CRIT] switch_core_io.c:173 sofia/internal/
> 140 at 10.50.100.16:5060 reading on a session with no media!
>
> Nothing happens for 10 seconds between 2015-07-09 20:08:19.972367
> and 2015-07-09 20:08:30.732483, FS sends no MOH and then decides to hangup
> both calls.
>
> And here is the output for 1.2 in a similar situation. It's a different
> server with different extensions, but I tried to capture the exact same
> moment.
>
>
> 2015-07-09 20:02:28.214025 [DEBUG] switch_core_session.c:1016 Send signal
> sofia/internal/sip:3315 at 172.17.100.86:5062 [BREAK]
> 2015-07-09 20:02:28.214025 [DEBUG] switch_core_session.c:1016 Send signal
> sofia/internal/sip:3315 at 172.17.100.86:5062 [BREAK]
> 2015-07-09 20:02:28.214025 [DEBUG] sofia.c:5815 Channel sofia/internal/
> sip:3315 at 172.17.100.86:5062 entering state [ready][200]
> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:5282 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3190 Set Codec
> sofia/internal/sip:3315 at 172.17.100.86:5062 PCMA/8000 20 ms 160 samples
> 64000 bits
> 2015-07-09 20:02:28.214025 [DEBUG] switch_core_codec.c:111 sofia/internal/
> sip:3315 at 172.17.100.86:5062 Original read codec set to PCMA:8
> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:5442 Set 2833 dtmf send
> payload to 101
> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3449 AUDIO RTP
> [sofia/internal/sip:3315 at 172.17.100.86:5062] 172.30.0.93 port 23690 ->
> 10.10.1.173 port 50692 codec: 8 ms: 20
> 2015-07-09 20:02:28.214025 [DEBUG] switch_rtp.c:2040 Starting timer [soft]
> 160 bytes per 20ms
> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3716 Set 2833 dtmf send
> payload to 101
> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3722 Set 2833 dtmf receive
> payload to 101
> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3749 sofia/internal/
> sip:3315 at 172.17.100.86:5062 Set rtp dtmf delay to 40
> 2015-07-09 20:02:28.223984 [DEBUG] switch_ivr_bridge.c:1852
> (sofia/external/8781 at 10.20.1.125.6:5075) State Change CS_HIBERNATE ->
> CS_CONSUME_MEDIA
> 2015-07-09 20:02:28.223984 [DEBUG] switch_core_state_machine.c:415
> (sofia/external/8781 at 10.20.1.125.6:5075) Running State Change
> CS_CONSUME_MEDIA
> 2015-07-09 20:02:28.223984 [DEBUG] switch_core_session.c:933 Send signal
> sofia/external/8781 at 10.20.1.125.6:5075 [BREAK]
> 2015-07-09 20:02:28.223984 [DEBUG] switch_core_session.c:1351 Send signal
> sofia/external/8781 at 10.20.1.125.6:5075 [BREAK]
> 2015-07-09 20:02:28.223984 [DEBUG] switch_ivr_bridge.c:1854
> (sofia/internal/sip:3315 at 172.17.100.86:5062) State Change CS_HIBERNATE ->
> CS_CONSUME_MEDIA
> 2015-07-09 20:02:28.223984 [DEBUG] switch_core_session.c:933 Send signal
> sofia/internal/sip:3315 at 172.17.100.86:5062 [BREAK]
>
> ... lots of other things and then FS plays MOH correctly.
>
> As I mentioned, the same thing happens in master.
>
> Does anyone have an idea of what could be happening? I can make all the
> tests and provide any logs necessary.
>
> Thanks
>
> On Thu, Jul 9, 2015 at 11:42 AM, Neven Boric <nboric at yx.cl> wrote:
>
>> Ok, I tried with master, and something strange happened. I'm pretty sure
>> it worked last night, but now I'm testing again and I'm getting the same
>> behavior I described with 1.4. I don't even have to get a third phone
>> involved. All I do is call from A to B, put B on hold and then wait, and
>> about ten seconds later, FS will hangup both calls, as if some timeout was
>> triggered. I get the same error on the log:
>>
>> 2015-07-09 14:36:43.506725 [CRIT] switch_core_io.c:93 sofia/internal/
>> 140 at 10.50.100.16:5060 reading on a session with no media!
>>
>> I also get no MOH on the phone that was put on hold.
>>
>> On Wed, Jul 8, 2015 at 3:41 PM, Neven Boric <nboric at yx.cl> wrote:
>>
>>>
>>>
>>> On Wed, Jul 8, 2015 at 2:17 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>>> - I first tried with the 1.2 branch, as that was the version I was
>>>>> using locally. The issue with that version is that when a phone holds and
>>>>> then unholds a call, I get no audio on the phone that started the hold. I
>>>>> found option bypass-media-after-hold, but realized that it was not included
>>>>> in the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93
>>>>> from the master branch. Now FS correctly tries to restore direct media
>>>>> between the endpoints with reINVITEs, but unfortunately includes a
>>>>> 'sendonly' in the last SDP to the phone that unholds, so I still get no
>>>>> audio. I haven't found a way to fix this.
>>>>
>>>>
>>>> Which probably means there's other commits required. 1.2 is EOL and no
>>>> longer supported, 1.4 is the current stable release. I wouldn't spend any
>>>> time getting 1.2 working, and instead work on upgrading.
>>>>
>>>
>>> Yes, I know it's not currently supported, but maybe somebody had the
>>> same problem and fixed it in a different way. Or maybe somebody could point
>>> me in the right direction to remove that final 'sendonly'.
>>>
>>>
>>>>
>>>> - I also tried with the 1.4 branch and hold/unhold works correctly, but
>>>>> now attended transfer doesn't work, FS after some time ends the call on the
>>>>> side that was put on hold waiting to be transferred. This time seems to be
>>>>> inconsistent, and sometimes the transfer actually works, so maybe it is a
>>>>> timing issue. This seems similar/related to FS-4038.
>>>>>
>>>>
>>>> Also see if you can replicate it on master, as that's close to the
>>>> upcoming 1.6 release and will have a lot of changes over 1.4.
>>>>
>>>
>>> I will try with master and report back, thanks.
>>>
>>>
>>>>
>>>>
>>>> On 8 July 2015 at 14:23, Neven Boric <nboric at yx.cl> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> I have been using FS for a long time on a local server without issues,
>>>>> but now I want to move to a remote server to support some new usage
>>>>> scenarios. I'm trying to use inbound-bypass-media=true to keep the audio
>>>>> out of the server. This mostly works, but I have two different issues,
>>>>> depending on which FS version I use:
>>>>>
>>>>> - I first tried with the 1.2 branch, as that was the version I was
>>>>> using locally. The issue with that version is that when a phone holds and
>>>>> then unholds a call, I get no audio on the phone that started the hold. I
>>>>> found option bypass-media-after-hold, but realized that it was not included
>>>>> in the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93
>>>>> from the master branch. Now FS correctly tries to restore direct media
>>>>> between the endpoints with reINVITEs, but unfortunately includes a
>>>>> 'sendonly' in the last SDP to the phone that unholds, so I still get no
>>>>> audio. I haven't found a way to fix this.
>>>>>
>>>>> - I also tried with the 1.4 branch and hold/unhold works correctly,
>>>>> but now attended transfer doesn't work, FS after some time ends the call on
>>>>> the side that was put on hold waiting to be transferred. This time seems to
>>>>> be inconsistent, and sometimes the transfer actually works, so maybe it is
>>>>> a timing issue. This seems similar/related to FS-4038.
>>>>>
>>>>> I can get logs and SIP captures, what would be the preferred way to
>>>>> provide them (pcap, inline text, pastebin)? Or maybe it's better to go
>>>>> ahead and file a bug?
>>>>>
>>>>> Best regards
>>>>> Neven Boric
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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 anthony.minessale at gmail.com  Tue Jul 14 03:00:09 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 13 Jul 2015 18:00:09 -0500
Subject: [Freeswitch-users] Some problems with bypass-media
In-Reply-To: <CAHQ=dc+1Qz0uLTjV-N1bB8nOKiZgkQW3=fnaOZ1vvFszcaQJ8A@mail.gmail.com>
References: <CAHQ=dcK0bfZbZ30zt9nYUH1z-R6BnSZBM1W0kAHommJspLKftg@mail.gmail.com>
	<CAFiqYukicT1Af87M6K3KUn-EYTPGGq2qtjWLi2TeAB+=kNodmw@mail.gmail.com>
	<CAHQ=dcJ+JNsU720QdKMznpYxSs0AryYif-MQbswFu2H2VrBAgQ@mail.gmail.com>
	<CAHQ=dc+_XbYXvTAYymG7AGypod1aWq87M9By_WEJU4NUrS74mQ@mail.gmail.com>
	<CAHQ=dcK0KJh6hsnfPiUuWEzzWt+6Q8YRSJD+L7gxECwNocLa3A@mail.gmail.com>
	<CAHQ=dc+1Qz0uLTjV-N1bB8nOKiZgkQW3=fnaOZ1vvFszcaQJ8A@mail.gmail.com>
Message-ID: <CAKbxfG_=exe3V-hM+MkLVWm3E1mrpyKDdsc9EbH8tSW7a1115A@mail.gmail.com>

Your first mistake is filing bugs on the mailing list.  Its impossible to
keep track of random mailing threads.
https://freeswitch.org/jira



On Mon, Jul 13, 2015 at 3:47 PM, Neven Boric <nboric at yx.cl> wrote:

> Anyone, please? It seems very strange that I'm the only one having this
> issue. The bypass-media option doesn't seem to be so unpopular.
>
> On Fri, Jul 10, 2015 at 12:34 PM, Neven Boric <nboric at yx.cl> wrote:
>
>> I compared the logs between 1.2 and 1.4, and I'm pretty sure in 1.4 one
>> of the threads is getting stuck until it times out.
>>
>> Here is the output for 1.4 right after I try to put the call on hold:
>>
>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_session.c:1061 Send signal
>> sofia/internal/171 at yx.cl [BREAK]
>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_session.c:1061 Send signal
>> sofia/internal/171 at yx.cl [BREAK]
>> 2015-07-09 20:08:19.952352 [DEBUG] sofia.c:6627 Channel sofia/internal/
>> 171 at yx.cl entering state [ready][200]
>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:3680 Audio Codec
>> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:3735 Audio Codec
>> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:3596 Set
>> telephone-event payload to 101
>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:2507 Set Codec
>> sofia/internal/171 at yx.cl PCMA/8000 20 ms 160 samples 64000 bits 1
>> channels
>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_codec.c:111 sofia/internal/
>> 171 at yx.cl Original read codec set to PCMA:8
>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:3943 Set 2833 dtmf
>> send/recv payload to 101
>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:5179 AUDIO RTP
>> [sofia/internal/171 at yx.cl] 10.176.0.1 port 21846 -> 10.176.4.121 port
>> 50140 codec: 8 ms: 20
>> 2015-07-09 20:08:19.952352 [DEBUG] switch_rtp.c:3569 Starting timer
>> [soft] 160 bytes per 20ms
>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:5477 Set 2833 dtmf
>> send payload to 101
>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:5483 Set 2833 dtmf
>> receive payload to 101
>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:5505
>> sofia/internal/171 at yx.cl Set rtp dtmf delay to 40
>> 2015-07-09 20:08:19.972367 [DEBUG] switch_core_session.c:912 Send signal
>> sofia/internal/140 at 10.50.100.16:5060 [BREAK]
>> 2015-07-09 20:08:30.732483 [CRIT] switch_core_io.c:173 sofia/internal/
>> 140 at 10.50.100.16:5060 reading on a session with no media!
>>
>> Nothing happens for 10 seconds between 2015-07-09 20:08:19.972367
>> and 2015-07-09 20:08:30.732483, FS sends no MOH and then decides to hangup
>> both calls.
>>
>> And here is the output for 1.2 in a similar situation. It's a different
>> server with different extensions, but I tried to capture the exact same
>> moment.
>>
>>
>> 2015-07-09 20:02:28.214025 [DEBUG] switch_core_session.c:1016 Send signal
>> sofia/internal/sip:3315 at 172.17.100.86:5062 [BREAK]
>> 2015-07-09 20:02:28.214025 [DEBUG] switch_core_session.c:1016 Send signal
>> sofia/internal/sip:3315 at 172.17.100.86:5062 [BREAK]
>> 2015-07-09 20:02:28.214025 [DEBUG] sofia.c:5815 Channel sofia/internal/
>> sip:3315 at 172.17.100.86:5062 entering state [ready][200]
>> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:5282 Audio Codec Compare
>> [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
>> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3190 Set Codec
>> sofia/internal/sip:3315 at 172.17.100.86:5062 PCMA/8000 20 ms 160 samples
>> 64000 bits
>> 2015-07-09 20:02:28.214025 [DEBUG] switch_core_codec.c:111 sofia/internal/
>> sip:3315 at 172.17.100.86:5062 Original read codec set to PCMA:8
>> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:5442 Set 2833 dtmf send
>> payload to 101
>> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3449 AUDIO RTP
>> [sofia/internal/sip:3315 at 172.17.100.86:5062] 172.30.0.93 port 23690 ->
>> 10.10.1.173 port 50692 codec: 8 ms: 20
>> 2015-07-09 20:02:28.214025 [DEBUG] switch_rtp.c:2040 Starting timer
>> [soft] 160 bytes per 20ms
>> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3716 Set 2833 dtmf send
>> payload to 101
>> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3722 Set 2833 dtmf
>> receive payload to 101
>> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3749 sofia/internal/
>> sip:3315 at 172.17.100.86:5062 Set rtp dtmf delay to 40
>> 2015-07-09 20:02:28.223984 [DEBUG] switch_ivr_bridge.c:1852
>> (sofia/external/8781 at 10.20.1.125.6:5075) State Change CS_HIBERNATE ->
>> CS_CONSUME_MEDIA
>> 2015-07-09 20:02:28.223984 [DEBUG] switch_core_state_machine.c:415
>> (sofia/external/8781 at 10.20.1.125.6:5075) Running State Change
>> CS_CONSUME_MEDIA
>> 2015-07-09 20:02:28.223984 [DEBUG] switch_core_session.c:933 Send signal
>> sofia/external/8781 at 10.20.1.125.6:5075 [BREAK]
>> 2015-07-09 20:02:28.223984 [DEBUG] switch_core_session.c:1351 Send signal
>> sofia/external/8781 at 10.20.1.125.6:5075 [BREAK]
>> 2015-07-09 20:02:28.223984 [DEBUG] switch_ivr_bridge.c:1854
>> (sofia/internal/sip:3315 at 172.17.100.86:5062) State Change CS_HIBERNATE
>> -> CS_CONSUME_MEDIA
>> 2015-07-09 20:02:28.223984 [DEBUG] switch_core_session.c:933 Send signal
>> sofia/internal/sip:3315 at 172.17.100.86:5062 [BREAK]
>>
>> ... lots of other things and then FS plays MOH correctly.
>>
>> As I mentioned, the same thing happens in master.
>>
>> Does anyone have an idea of what could be happening? I can make all the
>> tests and provide any logs necessary.
>>
>> Thanks
>>
>> On Thu, Jul 9, 2015 at 11:42 AM, Neven Boric <nboric at yx.cl> wrote:
>>
>>> Ok, I tried with master, and something strange happened. I'm pretty sure
>>> it worked last night, but now I'm testing again and I'm getting the same
>>> behavior I described with 1.4. I don't even have to get a third phone
>>> involved. All I do is call from A to B, put B on hold and then wait, and
>>> about ten seconds later, FS will hangup both calls, as if some timeout was
>>> triggered. I get the same error on the log:
>>>
>>> 2015-07-09 14:36:43.506725 [CRIT] switch_core_io.c:93 sofia/internal/
>>> 140 at 10.50.100.16:5060 reading on a session with no media!
>>>
>>> I also get no MOH on the phone that was put on hold.
>>>
>>> On Wed, Jul 8, 2015 at 3:41 PM, Neven Boric <nboric at yx.cl> wrote:
>>>
>>>>
>>>>
>>>> On Wed, Jul 8, 2015 at 2:17 PM, Steven Ayre <steveayre at gmail.com>
>>>> wrote:
>>>>
>>>>> - I first tried with the 1.2 branch, as that was the version I was
>>>>>> using locally. The issue with that version is that when a phone holds and
>>>>>> then unholds a call, I get no audio on the phone that started the hold. I
>>>>>> found option bypass-media-after-hold, but realized that it was not included
>>>>>> in the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93
>>>>>> from the master branch. Now FS correctly tries to restore direct media
>>>>>> between the endpoints with reINVITEs, but unfortunately includes a
>>>>>> 'sendonly' in the last SDP to the phone that unholds, so I still get no
>>>>>> audio. I haven't found a way to fix this.
>>>>>
>>>>>
>>>>> Which probably means there's other commits required. 1.2 is EOL and no
>>>>> longer supported, 1.4 is the current stable release. I wouldn't spend any
>>>>> time getting 1.2 working, and instead work on upgrading.
>>>>>
>>>>
>>>> Yes, I know it's not currently supported, but maybe somebody had the
>>>> same problem and fixed it in a different way. Or maybe somebody could point
>>>> me in the right direction to remove that final 'sendonly'.
>>>>
>>>>
>>>>>
>>>>> - I also tried with the 1.4 branch and hold/unhold works correctly,
>>>>>> but now attended transfer doesn't work, FS after some time ends the call on
>>>>>> the side that was put on hold waiting to be transferred. This time seems to
>>>>>> be inconsistent, and sometimes the transfer actually works, so maybe it is
>>>>>> a timing issue. This seems similar/related to FS-4038.
>>>>>>
>>>>>
>>>>> Also see if you can replicate it on master, as that's close to the
>>>>> upcoming 1.6 release and will have a lot of changes over 1.4.
>>>>>
>>>>
>>>> I will try with master and report back, thanks.
>>>>
>>>>
>>>>>
>>>>>
>>>>> On 8 July 2015 at 14:23, Neven Boric <nboric at yx.cl> wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I have been using FS for a long time on a local server without
>>>>>> issues, but now I want to move to a remote server to support some new usage
>>>>>> scenarios. I'm trying to use inbound-bypass-media=true to keep the audio
>>>>>> out of the server. This mostly works, but I have two different issues,
>>>>>> depending on which FS version I use:
>>>>>>
>>>>>> - I first tried with the 1.2 branch, as that was the version I was
>>>>>> using locally. The issue with that version is that when a phone holds and
>>>>>> then unholds a call, I get no audio on the phone that started the hold. I
>>>>>> found option bypass-media-after-hold, but realized that it was not included
>>>>>> in the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93
>>>>>> from the master branch. Now FS correctly tries to restore direct media
>>>>>> between the endpoints with reINVITEs, but unfortunately includes a
>>>>>> 'sendonly' in the last SDP to the phone that unholds, so I still get no
>>>>>> audio. I haven't found a way to fix this.
>>>>>>
>>>>>> - I also tried with the 1.4 branch and hold/unhold works correctly,
>>>>>> but now attended transfer doesn't work, FS after some time ends the call on
>>>>>> the side that was put on hold waiting to be transferred. This time seems to
>>>>>> be inconsistent, and sometimes the transfer actually works, so maybe it is
>>>>>> a timing issue. This seems similar/related to FS-4038.
>>>>>>
>>>>>> I can get logs and SIP captures, what would be the preferred way to
>>>>>> provide them (pcap, inline text, pastebin)? Or maybe it's better to go
>>>>>> ahead and file a bug?
>>>>>>
>>>>>> Best regards
>>>>>> Neven Boric
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> 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
>



-- 
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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From krice at freeswitch.org  Tue Jul 14 03:25:08 2015
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 13 Jul 2015 23:25:08 +0000
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) July
	4th-July 11th
Message-ID: <55a448d44e7b_c6e069b32478210@resque-worker-high.0.mail>

New Post on freeswitch.org from Kathleen King
check it out at http://ift.tt/1O2b2vR
FreeSWITCH Week in Review (Master Branch) July 4th-July 11th
Hello, again. This passed week in the FreeSWITCH master branch we had 54 commits. We had a bunch of new commits for features this week! Those pull requests are really helping to get things reviewed and accepted. Keep it up community!

Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support.

New features that were added:

FS-7780 Add new channel variable max_session_transfers. If set, this variable is used to count the number of session transfers allowed instead of the max_forwards variable. If not set, the existing behavior is preserved.

FS-7783 Add channel variable for capturing DTMF input when using play_and_get_digits when the response does not match

FS-7772 [mod_opus] Add functionality to keep FEC enabled on the encoder by modifying the bitrate if packet loss changes (Opus codec specific behaviour).

FS-7799 [mod_png] Add API command uuid_write_png

FS-7801 [mod_opus] Added support to set CBR mode

FS-7685 [mod_say_nl] Fix Dutch numbers pronunciation

FS-7198 Add coma separated values and reverse ranges for time-of-day and day-of-week matches

FS-7809 [mod_opus] Added 60 ms ptime for Opus at 8 khz ( opus at 8000h@60i )

FS-7405 [mod_dialplan_xml] Fix condition regex=?all? to work with time conditions

FS-7819 [mod_opus] Restore bitrate (if there?s no more packet loss) and added step for 60 ms

FS-7773 [mod_sofia] Adding additional transfer events when the fire-transfer-events=true profile parameter is set

FS-7820 FreeSWITCH automated unit test and micro benchmark framework

Improvements in build system, cross platform support, and packaging:

FS-7628 [mod_erlang_event] Removed unused variables causing a compilation error

FS-7776 Add mod_kazoo to packaging

The following bugs were squashed:

FS-7778 [mod_sofia] Fixed a bug causing a SQL statement to fail because of a double quote instead of a single quote

FS-7754 [freetdm] Fixed a bug relating to single digit dial-regex with analog devices

FS-7785 [mod_opus] Fix for invalid ptime 30 ms for opus at 8000h . Replaced 30 ms with 40 ms.

FS-7762 [mod_av] Handle buffer allocation failures of large buffers

And, this passed week in the FreeSWITCH 1.4 branch we had no new commits merged in from master.

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From nzaytsevc at gmail.com  Sat Jul 11 11:33:56 2015
From: nzaytsevc at gmail.com (Nikolay Zaytsev)
Date: Sat, 11 Jul 2015 11:33:56 +0400
Subject: [Freeswitch-users] E1 cards
Message-ID: <CADeoHurK9F3hU4DMSCpHtLzPDcBWM-uw9KECribTM=bm2NuPmg@mail.gmail.com>

Hi everyone)
Could you advise me please which card is the best choice for using with
Freeswitch?
I need the card with 2 E1 ports.
So far I have been thinking about Sangoma A102DE, but maybe you will advise
me something else)
Thank you for your time,
Best regards,
Nikolay
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From stefano.favaro at edistar.com  Mon Jul 13 18:41:13 2015
From: stefano.favaro at edistar.com (stefano)
Date: Mon, 13 Jul 2015 07:41:13 -0700 (MST)
Subject: [Freeswitch-users] Force codec on A leg
Message-ID: <1436798473130-7596175.post@n2.nabble.com>

Hello,

I have this scenario:
A = user
B = my FreeSwitch
C = Dialogic IVR HMP

A leg place a call with codec in sdp in this order
g729
PCMU
PCMA

B reroute the call to C 

C answer with codec PCMU, PCMA

I'd like to remove codec PCMU,PCMA from originator (A leg) and use only g729
offered and let the FS make the transcoding vs C.

I have used 
on the sip_profile:




on my dialplan:

 <action application="export"
data="nolocal:absolute_codec_string=PCMA,PCMU"/>
 before outdialing to the IVR

How can I discard the use of PCMU,PCMA from the A leg?

Thanks. 







--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Force-codec-on-A-leg-tp7596175.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at microcomaustralia.com.au  Tue Jul 14 03:34:48 2015
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 13 Jul 2015 23:34:48 +0000
Subject: [Freeswitch-users] voice dropping out
Message-ID: <CAA0ZO6BioOfhStV1hvk6XjzfzuAZAzx+-y5JS2jubNZEO6COtw@mail.gmail.com>

Hello,

I have had reports when talking from a sflphone SIP phone to a Gigaset C610
phone, my voice keeps dropping out, As it the other end can't here complete
words I say.

If however I test with the freeswitch echo service, it works fine, Similar
if I tell sflphone to record, it records everything fine. So it appears to
be a transport problem, as opposed to a microphone pick up issue.

Similarly, the Gigaset C610 is known to be good, and works fine with other
sources.

freeswitch shows PCMU codec is being used for both legs of the call.

(I intended to use PCMA, as from memory that is more "standard" in
Australia but I am doubtful that is the problem)

How do I debug?

Thanks
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From jcabezas at inovax.com.br  Tue Jul 14 03:58:59 2015
From: jcabezas at inovax.com.br (Julio Cesar Esteves Cabezas)
Date: Mon, 13 Jul 2015 20:58:59 -0300
Subject: [Freeswitch-users] Transparency in SUBSCRIBE and NOTIFY
Message-ID: <09f801d0bdc7$e2c30e80$a8492b80$@inovax.com.br>

Hi,          

                My  SIP User-Agents need to establish between themselves a
mesh of SIP subscriptions (using SUBSCRIBE and NOTIFY).

                The event-packages of those subscriptions are two, their
names are:  "dialog" and "p-np-session".

 

                But FreeSWITCH replies 489 Bad Event to SUBSCRIBEs  to
"p-np-session" and 200 OK to SUBSCRIBEs to "dialog" (because FreeSWITCH
implements this package but that's not what I need)

                So, is there some way to make FreeSWITCH transparent to
those subscriptions (like a vanilla SIP proxy would be)  ?    

Any suggestions ?

 

Many Thanks.

 

 

 

 

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From brian at freeswitch.org  Tue Jul 14 04:02:57 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 13 Jul 2015 19:02:57 -0500
Subject: [Freeswitch-users] Transparency in SUBSCRIBE and NOTIFY
In-Reply-To: <09f801d0bdc7$e2c30e80$a8492b80$@inovax.com.br>
References: <09f801d0bdc7$e2c30e80$a8492b80$@inovax.com.br>
Message-ID: <CAEJMVkAzWrOUdxn1t=26kOYR+d_6GETbYJUr6A1pw2=m-Nuwvw@mail.gmail.com>

Use a proxy! :). Like kamailio or opensips

On Monday, July 13, 2015, Julio Cesar Esteves Cabezas <
jcabezas at inovax.com.br> wrote:

> Hi,
>
>                 My  SIP User-Agents need to establish between themselves a
> mesh of SIP subscriptions (using SUBSCRIBE and NOTIFY).
>
>                 The event-packages of those subscriptions are two, their
> names are:  ?dialog? and ?p-np-session?.
>
>
>
>                 But FreeSWITCH replies 489 Bad Event to SUBSCRIBEs  to
> ?p-np-session? and 200 OK to SUBSCRIBEs to ?dialog? (because FreeSWITCH
> implements this package but that?s not what I need)
>
>                 So, is there some way to make FreeSWITCH transparent to
> those subscriptions (like a vanilla SIP proxy would be)  ?
>
> Any suggestions ?
>
>
>
> Many Thanks.
>
>
>
>
>
>
>
>
>


-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
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From Alexander.Haugg at c4b.de  Tue Jul 14 10:47:12 2015
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Tue, 14 Jul 2015 06:47:12 +0000
Subject: [Freeswitch-users] Update To and From Header of the legs after
 an uuid_bridge
In-Reply-To: <B60DA91F5EA0E24A96E73CDE43CC9EC8517C1222@vc4bex2.corp.c4b.de>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC8517C1222@vc4bex2.corp.c4b.de>
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC8517C1752@vc4bex2.corp.c4b.de>

Hi All,

I can see (source code), that the parameter "send-display-update" is not set, it is true by default.
For the dialplan app ?send_display?, the Freeswitch Event on the event socket have the new value.

But on SIP side in direction to the PBX trunk the freeswitch don?t send a SIP Update.
See my last post.

Waht can I do?

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Alexander Haugg
Gesendet: Donnerstag, 9. Juli 2015 13:57
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Update To and From Header of the legs after an uuid_bridge

I had test it again and it doesn?t work.
I added the property ?<param name="send-display-update" value="true"/>? to the sip profile with the same result.

I try it in the dialplan with ?<action application="send_display" data="servus|123" /> ? without success
Is there a possibility to force a send SIP Update as API or dialplan APP command?
What can I do?

Here is the uuid_bridge CLI output.

freeswitch at WIN-OHIC1AKL8UU> uuid_bridge 302eba5831664c8a e627aa5a996b07ca

+OK e627aa5a996b07ca

2015-07-09 13:35:31.922297 [DEBUG] switch_ivr_bridge.c:1873 (sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26>) State Change CS_EXECUTE -> CS_HIBERNATE
2015-07-09 13:35:31.922297 [DEBUG] switch_core_session.c:1388 Send signal sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26> [BREAK]
2015-07-09 13:35:31.922297 [DEBUG] switch_ivr_bridge.c:1875 (sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26>) State Change CS_EXECUTE -> CS_HIBERNATE
2015-07-09 13:35:31.922297 [DEBUG] switch_core_session.c:1388 Send signal sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26> [BREAK]
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:535 (sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26>) State EXECUTE going to sleep
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:472 (sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26>) Running State Change CS_HIBERNATE
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:550 (sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26>) State HIBERNATE
2015-07-09 13:35:31.942298 [DEBUG] mod_sofia.c:160 sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26> SOFIA HIBERNATE
2015-07-09 13:35:31.942298 [DEBUG] switch_ivr_bridge.c:835 (sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26>) State Change CS_HIBERNATE -> CS_RESET
2015-07-09 13:35:31.942298 [DEBUG] switch_core_session.c:1388 Send signal sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26> [BREAK]
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:550 (sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26>) State HIBERNATE going to sleep
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:472 (sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26>) Running State Change CS_RESET
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:531 (sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26>) State RESET
2015-07-09 13:35:31.942298 [DEBUG] mod_sofia.c:141 sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26> SOFIA RESET
2015-07-09 13:35:31.942298 [DEBUG] switch_ivr_bridge.c:820 sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26> CUSTOM RESET
2015-07-09 13:35:31.942298 [DEBUG] switch_ivr_bridge.c:827 (sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26>) State Change CS_RESET -> CS_SOFT_EXECUTE
2015-07-09 13:35:31.942298 [DEBUG] switch_core_session.c:1388 Send signal sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26> [BREAK]
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:531 (sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26>) State RESET going to sleep
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:472 (sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26>) Running State Change CS_SOFT_EXECUTE
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:541 (sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26>) State SOFT_EXECUTE
2015-07-09 13:35:31.942298 [DEBUG] mod_sofia.c:600 SOFIA SOFT_EXECUTE
2015-07-09 13:35:31.942298 [DEBUG] switch_ivr_bridge.c:845 sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26> CUSTOM SOFT_EXECUTE
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:535 (sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26>) State EXECUTE going to sleep
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:472 (sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26>) Running State Change CS_HIBERNATE
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:550 (sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26>) State HIBERNATE
2015-07-09 13:35:31.942298 [DEBUG] mod_sofia.c:160 sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26> SOFIA HIBERNATE
2015-07-09 13:35:31.942298 [DEBUG] switch_ivr_bridge.c:835 (sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26>) State Change CS_HIBERNATE -> CS_RESET
2015-07-09 13:35:31.942298 [DEBUG] switch_core_session.c:1388 Send signal sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26> [BREAK]
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:550 (sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26>) State HIBERNATE going to sleep
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:472 (sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26>) Running State Change CS_RESET
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:531 (sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26>) State RESET
2015-07-09 13:35:31.942298 [DEBUG] mod_sofia.c:141 sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26> SOFIA RESET
2015-07-09 13:35:31.942298 [DEBUG] switch_ivr_bridge.c:820 sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26> CUSTOM RESET
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:118 sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26> Standard RESET
2015-07-09 13:35:31.942298 [DEBUG] switch_core_state_machine.c:531 (sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26>) State RESET going to sleep
2015-07-09 13:35:31.962299 [DEBUG] switch_ivr_bridge.c:877 (sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26>) State Change CS_RESET -> CS_SOFT_EXECUTE
2015-07-09 13:35:31.962299 [DEBUG] switch_core_session.c:1388 Send signal sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26> [BREAK]
2015-07-09 13:35:31.962299 [DEBUG] switch_core_state_machine.c:472 (sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26>) Running State Change CS_SOFT_EXECUTE
2015-07-09 13:35:31.962299 [DEBUG] switch_core_state_machine.c:541 (sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26>) State SOFT_EXECUTE
2015-07-09 13:35:31.962299 [DEBUG] mod_sofia.c:600 SOFIA SOFT_EXECUTE
2015-07-09 13:35:31.962299 [DEBUG] switch_ivr_bridge.c:845 sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26> CUSTOM SOFT_EXECUTE
2015-07-09 13:35:31.962299 [DEBUG] switch_core_state_machine.c:330 sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26> Standard SOFT_EXECUTE
2015-07-09 13:35:31.962299 [DEBUG] switch_core_state_machine.c:541 (sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26>) State SOFT_EXECUTE going to sleep
2015-07-09 13:35:31.982300 [DEBUG] switch_ivr_bridge.c:1360 (sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26>) State Change CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
2015-07-09 13:35:31.982300 [DEBUG] switch_core_session.c:1388 Send signal sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26> [BREAK]
2015-07-09 13:35:31.982300 [DEBUG] switch_core_session.c:908 Send signal sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26> [BREAK]
2015-07-09 13:35:31.982300 [DEBUG] switch_core_session.c:908 Send signal sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26> [BREAK]
2015-07-09 13:35:31.982300 [DEBUG] switch_ivr_bridge.c:1465 (sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26>) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2015-07-09 13:35:31.982300 [DEBUG] switch_core_session.c:1388 Send signal sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26> [BREAK]
2015-07-09 13:35:31.982300 [DEBUG] switch_core_state_machine.c:472 (sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26>) Running State Change CS_EXCHANGE_MEDIA
2015-07-09 13:35:31.982300 [DEBUG] switch_core_state_machine.c:538 (sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26>) State EXCHANGE_MEDIA
2015-07-09 13:35:31.982300 [DEBUG] mod_sofia.c:594 SOFIA EXCHANGE_MEDIA
2015-07-09 13:35:31.982300 [DEBUG] switch_core_session.c:970 Send signal sofia/H3KSip/+4985047079741 at 172.16.1.26<mailto:sofia/H3KSip/+4985047079741 at 172.16.1.26> [BREAK]
2015-07-09 13:35:31.982300 [DEBUG] switch_core_session.c:970 Send signal sofia/H3KSip/+4989840798170 at 172.16.1.26<mailto:sofia/H3KSip/+4989840798170 at 172.16.1.26> [BREAK]

Von: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Jerris
Gesendet: Donnerstag, 25. Juni 2015 14:05
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Update To and From Header of the legs after an uuid_bridge

This happens automatically using a sip update packet unless you have disabled display updates

On Thursday, June 25, 2015, Alexander Haugg <Alexander.Haugg at c4b.de<mailto:Alexander.Haugg at c4b.de>> wrote:
Hi,

have anyone an answer?

More scenario information?s?

LegA connected with LegB
Now LegA bridged to (with uuid_bridge) LegC (LegB will terminated)

It is very important for me that LegA have the Partner Number of LegC and vice versa!
I need to transport this information?s back to a PBX via SIP Trunk!
What can I do for this?

Thanks a lot

Von: freeswitch-users-bounces at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');> [mailto:freeswitch-users-bounces at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>] Im Auftrag von Alexander Haugg
Gesendet: Donnerstag, 18. Juni 2015 11:16
An: freeswitch-users at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users at lists.freeswitch.org');>
Betreff: [Freeswitch-users] Update To and From Header of the legs after an uuid_bridge

Hi All,

What can i do to change (fix) the from- and the to- header on SIP side after an uuid_bridge (SIP UPDATE or re-INVITE)?

Thanks a lot!
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From achinthau at gmail.com  Tue Jul 14 14:09:20 2015
From: achinthau at gmail.com (Achintha)
Date: Tue, 14 Jul 2015 15:39:20 +0530
Subject: [Freeswitch-users] profile context is not overridden
In-Reply-To: <CAFiqYunofjieNLW0NLfHAku+Pfx=3eqj-SWvDYF1oVS2exVW0w@mail.gmail.com>
References: <CA+JwzxtSkHASHOeteOxRozH8fQwBCk2eYO_G36cmAb+43LwFGA@mail.gmail.com>
	<CAFiqYunofjieNLW0NLfHAku+Pfx=3eqj-SWvDYF1oVS2exVW0w@mail.gmail.com>
Message-ID: <CA+Jwzxsp_sbP4ModfcuQ7qnVeq9d4UJ09zwofJdJBfFVVse1aA@mail.gmail.com>

hi steven,

i'm unable to find user_context in CDR file.

On Mon, Jul 13, 2015 at 8:03 PM, Steven Ayre <steveayre at gmail.com> wrote:

> First off verify the call is getting authenticated as the user. If you
> examine the XML CDR or the 'info' output do you see the user_context
> variable set?
>
> On 13 July 2015 at 13:38, Achintha <achinthau at gmail.com> wrote:
>
>> hi all,
>>
>>
>> We are having freeswitch testing setup with Internal and external
>> profiles with xml_curl for Directory management. In this setup profile
>> context is not overridden with user_context property.
>>
>> Kindly help me to solve this issue.
>>
>>
>>
>> Thanking you.
>>
>>
>> Achintha
>>
>> _________________________________________________________________________
>> 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
>



-- 

Achintha
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From steveayre at gmail.com  Tue Jul 14 14:25:39 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 14 Jul 2015 11:25:39 +0100
Subject: [Freeswitch-users] profile context is not overridden
In-Reply-To: <CA+Jwzxsp_sbP4ModfcuQ7qnVeq9d4UJ09zwofJdJBfFVVse1aA@mail.gmail.com>
References: <CA+JwzxtSkHASHOeteOxRozH8fQwBCk2eYO_G36cmAb+43LwFGA@mail.gmail.com>
	<CAFiqYunofjieNLW0NLfHAku+Pfx=3eqj-SWvDYF1oVS2exVW0w@mail.gmail.com>
	<CA+Jwzxsp_sbP4ModfcuQ7qnVeq9d4UJ09zwofJdJBfFVVse1aA@mail.gmail.com>
Message-ID: <CAFiqYu=QoM-2XuZfEiD4OJ8G_EVKjo6WoPy_RrtRs3m57Vsexg@mail.gmail.com>

Are you sure the call is getting authenticating as the user? Perhaps it's
being processed as an unauthenticated user?

On 14 July 2015 at 11:09, Achintha <achinthau at gmail.com> wrote:

> hi steven,
>
> i'm unable to find user_context in CDR file.
>
> On Mon, Jul 13, 2015 at 8:03 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> First off verify the call is getting authenticated as the user. If you
>> examine the XML CDR or the 'info' output do you see the user_context
>> variable set?
>>
>> On 13 July 2015 at 13:38, Achintha <achinthau at gmail.com> wrote:
>>
>>> hi all,
>>>
>>>
>>> We are having freeswitch testing setup with Internal and external
>>> profiles with xml_curl for Directory management. In this setup profile
>>> context is not overridden with user_context property.
>>>
>>> Kindly help me to solve this issue.
>>>
>>>
>>>
>>> Thanking you.
>>>
>>>
>>> Achintha
>>>
>>> _________________________________________________________________________
>>> 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
>>
>
>
>
> --
>
> Achintha
>
> _________________________________________________________________________
> 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 bhavikpatel14388 at gmail.com  Tue Jul 14 16:08:11 2015
From: bhavikpatel14388 at gmail.com (bhavik patel)
Date: Tue, 14 Jul 2015 17:38:11 +0530
Subject: [Freeswitch-users] BLF Implementation
Message-ID: <CAG6TCM9xynte8HzQxFXQ0VaZKmz4Njh+qd0TtZ6WP+KNgkxHTw@mail.gmail.com>

Hi,

I am trying to implement BLF in Grandstream (GXP1400) with freeswitch
1.4.20.

For that , i used "<param name="send-presence-on-register" value="true"/>"
in sip profile.

But Not getting any NOTIFY request to sip phone,and BLF is not working.

Can any one suggest me how to enable this function in Freeswitch ?

Any help would be much appreciated.

-- 
Thanks,
Bhavik Patel
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From yadenis at seznam.cz  Tue Jul 14 16:36:46 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Tue, 14 Jul 2015 14:36:46 +0200
Subject: [Freeswitch-users] How do I know uuid both leg before bridge?
In-Reply-To: <CAG6TCM9xynte8HzQxFXQ0VaZKmz4Njh+qd0TtZ6WP+KNgkxHTw@mail.gmail.com>
References: <CAG6TCM9xynte8HzQxFXQ0VaZKmz4Njh+qd0TtZ6WP+KNgkxHTw@mail.gmail.com>
Message-ID: <1971438456.20150714143646@seznam.cz>

Hi All,

For my tasks need to know uuid both feet before bridge. How can i do this?

If I do in the dialplan
<action application="set" data="my_uuid=${create_uuid foo}" inline="true"/> 
and then 
<action application="bridge" data="{origination_uuid=${uuid}}user/1003"/> 

I get an error 
2015-07-14 14:25:32.350565 [CRIT] switch_core_session.c:2342 Duplicate UUID!
2015-07-14 14:25:32.350565 [CRIT] mod_sofia.c:4376 Error Creating Session 

How correctly before bridge I assign or generate a uuid and use later?

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From jcabezas at inovax.com.br  Tue Jul 14 16:41:06 2015
From: jcabezas at inovax.com.br (Julio Cesar Esteves Cabezas)
Date: Tue, 14 Jul 2015 09:41:06 -0300
Subject: [Freeswitch-users] Transparency in SUBSCRIBE and NOTIFY
In-Reply-To: <CAEJMVkAzWrOUdxn1t=26kOYR+d_6GETbYJUr6A1pw2=m-Nuwvw@mail.gmail.com>
References: <09f801d0bdc7$e2c30e80$a8492b80$@inovax.com.br>
	<CAEJMVkAzWrOUdxn1t=26kOYR+d_6GETbYJUr6A1pw2=m-Nuwvw@mail.gmail.com>
Message-ID: <0a9b01d0be32$5a93c3e0$0fbb4ba0$@inovax.com.br>

 

    Are you suggesting to combine FreeSWITCH and a SIP Proxy together ?   

    Can you ellaborate about that topology ?   I want to retain the unique features of FreeSWITCH in my system.

 

Thanks.

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Monday, July 13, 2015 9:03 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Transparency in SUBSCRIBE and NOTIFY

 

Use a proxy! :). Like kamailio or opensips 

On Monday, July 13, 2015, Julio Cesar Esteves Cabezas <jcabezas at inovax.com.br> wrote:

Hi,          

                My  SIP User-Agents need to establish between themselves a mesh of SIP subscriptions (using SUBSCRIBE and NOTIFY).

                The event-packages of those subscriptions are two, their names are:  ?dialog? and ?p-np-session?.

 

                But FreeSWITCH replies 489 Bad Event to SUBSCRIBEs  to ?p-np-session? and 200 OK to SUBSCRIBEs to ?dialog? (because FreeSWITCH implements this package but that?s not what I need)

                So, is there some way to make FreeSWITCH transparent to those subscriptions (like a vanilla SIP proxy would be)  ?    

Any suggestions ?

 

Many Thanks.

 

 

 

 



-- 

Brian West
brian at freeswitch.org

  <http://billing.freeswitch.org/templates/default/img/whmcslogo.png> 

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/>  | Register <https://freeswitch.com/cart.php?gid=1>  TODAY! | 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

 

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From steveayre at gmail.com  Tue Jul 14 16:51:28 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 14 Jul 2015 13:51:28 +0100
Subject: [Freeswitch-users] How do I know uuid both leg before bridge?
In-Reply-To: <1971438456.20150714143646@seznam.cz>
References: <CAG6TCM9xynte8HzQxFXQ0VaZKmz4Njh+qd0TtZ6WP+KNgkxHTw@mail.gmail.com>
	<1971438456.20150714143646@seznam.cz>
Message-ID: <CAFiqYun34_2EqXjE=MFzuDWCoyAN-AfTLAwDFPAwK4TntPtQDQ@mail.gmail.com>

Use origination_uuid=${my_uuid} not ${uuid}

${uuid} is the existing a-leg, origination_uuid will be the new b-leg

Currently you're trying to create a b-leg with the same uuid as the a-leg,
hence the error.

I also suggest you put origination_uuid in [] not {} - if you ever add
extra legs you'll have the duplicate error return as all the b-legs would
get the same uuid.

You also definitely *don't* need inline=true, I would avoid using it as
it'll execute much earlier than you need it, and if you use set the same
variable inline later in your dialplan then it may not create the bridge
with the uuid you expect.

Steve



On 14 July 2015 at 13:36, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Hi All,
>
> For my tasks need to know uuid both feet before bridge. How can i do this?
>
> If I do in the dialplan
> <action application="set" data="my_uuid=${create_uuid foo}" inline="true"/>
>
> and then
> <action application="bridge" data="{origination_uuid=${uuid}}user/1003"/>
>
> I get an error
> 2015-07-14 14:25:32.350565 [CRIT] switch_core_session.c:2342 Duplicate
> UUID!
> 2015-07-14 14:25:32.350565 [CRIT] mod_sofia.c:4376 Error Creating Session
>
> How correctly before bridge I assign or generate a uuid and use later?
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 *
>
> _________________________________________________________________________
> 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 yadenis at seznam.cz  Tue Jul 14 17:35:25 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Tue, 14 Jul 2015 15:35:25 +0200
Subject: [Freeswitch-users] How do I know uuid both leg before bridge?
In-Reply-To: <CAFiqYun34_2EqXjE=MFzuDWCoyAN-AfTLAwDFPAwK4TntPtQDQ@mail.gmail.com>
References: <CAG6TCM9xynte8HzQxFXQ0VaZKmz4Njh+qd0TtZ6WP+KNgkxHTw@mail.gmail.com>
	<1971438456.20150714143646@seznam.cz>
	<CAFiqYun34_2EqXjE=MFzuDWCoyAN-AfTLAwDFPAwK4TntPtQDQ@mail.gmail.com>
Message-ID: <1349206034.20150714153525@seznam.cz>

Dobr? den,

Great! Thanks a lot for the help. Now it is working as it should. 

I think it is worth adding these things in questions and answers, so that people know about these brackets [ ] and how it works inline=true 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

?ter? 14. ?ervence 2015, 14:51:28, napsal jste:


Use origination_uuid=${my_uuid} not ${uuid}

${uuid} is the existing a-leg, origination_uuid will be the new b-leg

Currently you're trying to create a b-leg with the same uuid as the a-leg, hence the error.

I also suggest you put origination_uuid in [] not {} - if you ever add extra legs you'll have the duplicate error return as all the b-legs would get the same uuid.

You also definitely don't need inline=true, I would avoid using it as it'll execute much earlier than you need it, and if you use set the same variable inline later in your dialplan then it may not create the bridge with the uuid you expect.

Steve



On 14 July 2015 at 13:36, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi All,

For my tasks need to know uuid both feet before bridge. How can i do this?

If I do in the dialplan
<action application="set" data="my_uuid=${create_uuid foo}" inline="true"/> 
and then 
<action application="bridge" data="{origination_uuid=${uuid}}user/1003"/> 

I get an error 
2015-07-14 14:25:32.350565 [CRIT] switch_core_session.c:2342 Duplicate UUID!
2015-07-14 14:25:32.350565 [CRIT] mod_sofia.c:4376 Error Creating Session 

How correctly before bridge I assign or generate a uuid and use later?

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

_________________________________________________________________________
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From techfaltu at gmail.com  Tue Jul 14 13:15:59 2015
From: techfaltu at gmail.com (Niraj Roy)
Date: Tue, 14 Jul 2015 14:45:59 +0530
Subject: [Freeswitch-users] Issues with simultaneous call hangup
Message-ID: <CAO2xAXM3KW7u4t=c4gK6Gt8OXywe7ZkCk+svbQ63K=xS-my2=A@mail.gmail.com>

Hello Gurus,
    We are using FreeSWITCH Version 1.4.18~64bit. We observed when both
caller and caller hangs up the call simultaneously exec_after_bridge_app
function does not execute for one of the legs.
    Here is the segment of executing the xml file.
                <action application="set"
data="exec_after_bridge_app=transfer"/>
                <action application="set"
data="exec_after_bridge_arg=endcall-handler XML CALLENDING"/>
                <action application="set_zombie_exec"/>
                <action application="set" data="continue_on_fail=true"/>
                <action inline="true" application="set"
data="hangup_after_bridge=false"/>

For normal or sequential call clearing everything works fine, but somehow
this creates a race condition and state mismatch in the freeswitch which I
don't know.
Can anybody shed some light on this?

Thanks,
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From brian at freeswitch.org  Tue Jul 14 17:40:27 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 14 Jul 2015 08:40:27 -0500
Subject: [Freeswitch-users] ClueCon 2015,
	Hotel Cutoff date is Tomorrow July 15th, Register TODAY!
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From brian at freeswitch.org  Tue Jul 14 18:11:23 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 14 Jul 2015 09:11:23 -0500
Subject: [Freeswitch-users] Call @ 2PM Central , Testing/QA/Unit Tests
Message-ID: <CAEJMVkA4bk6fs8_zdyhZyb_6GQd93r6-jPNAc7=NjUJ=mY8WzQ@mail.gmail.com>

Everyone interested in helping evaluate, review, discuss approaches to
increase our usage of automation and unit tests in FreeSWITCH!

https://cantina.freeswitch.org/conf/

Join in!  Everyone is welcome!

-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit:
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*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
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From nboric at yx.cl  Tue Jul 14 19:11:43 2015
From: nboric at yx.cl (Neven Boric)
Date: Tue, 14 Jul 2015 12:11:43 -0300
Subject: [Freeswitch-users] Some problems with bypass-media
In-Reply-To: <CAKbxfG_=exe3V-hM+MkLVWm3E1mrpyKDdsc9EbH8tSW7a1115A@mail.gmail.com>
References: <CAHQ=dcK0bfZbZ30zt9nYUH1z-R6BnSZBM1W0kAHommJspLKftg@mail.gmail.com>
	<CAFiqYukicT1Af87M6K3KUn-EYTPGGq2qtjWLi2TeAB+=kNodmw@mail.gmail.com>
	<CAHQ=dcJ+JNsU720QdKMznpYxSs0AryYif-MQbswFu2H2VrBAgQ@mail.gmail.com>
	<CAHQ=dc+_XbYXvTAYymG7AGypod1aWq87M9By_WEJU4NUrS74mQ@mail.gmail.com>
	<CAHQ=dcK0KJh6hsnfPiUuWEzzWt+6Q8YRSJD+L7gxECwNocLa3A@mail.gmail.com>
	<CAHQ=dc+1Qz0uLTjV-N1bB8nOKiZgkQW3=fnaOZ1vvFszcaQJ8A@mail.gmail.com>
	<CAKbxfG_=exe3V-hM+MkLVWm3E1mrpyKDdsc9EbH8tSW7a1115A@mail.gmail.com>
Message-ID: <CAHQ=dcKfN+_Ei-hFQuaXku_r0jWMW0mDA_NF-dB2TpddaXSjVA@mail.gmail.com>

Filed bug FS-7834

On Mon, Jul 13, 2015 at 8:00 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Your first mistake is filing bugs on the mailing list.  Its impossible to
> keep track of random mailing threads.
> https://freeswitch.org/jira
>
>
>
> On Mon, Jul 13, 2015 at 3:47 PM, Neven Boric <nboric at yx.cl> wrote:
>
>> Anyone, please? It seems very strange that I'm the only one having this
>> issue. The bypass-media option doesn't seem to be so unpopular.
>>
>> On Fri, Jul 10, 2015 at 12:34 PM, Neven Boric <nboric at yx.cl> wrote:
>>
>>> I compared the logs between 1.2 and 1.4, and I'm pretty sure in 1.4 one
>>> of the threads is getting stuck until it times out.
>>>
>>> Here is the output for 1.4 right after I try to put the call on hold:
>>>
>>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_session.c:1061 Send
>>> signal sofia/internal/171 at yx.cl [BREAK]
>>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_session.c:1061 Send
>>> signal sofia/internal/171 at yx.cl [BREAK]
>>> 2015-07-09 20:08:19.952352 [DEBUG] sofia.c:6627 Channel sofia/internal/
>>> 171 at yx.cl entering state [ready][200]
>>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:3680 Audio Codec
>>> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:3735 Audio Codec
>>> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:3596 Set
>>> telephone-event payload to 101
>>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:2507 Set Codec
>>> sofia/internal/171 at yx.cl PCMA/8000 20 ms 160 samples 64000 bits 1
>>> channels
>>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_codec.c:111
>>> sofia/internal/171 at yx.cl Original read codec set to PCMA:8
>>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:3943 Set 2833
>>> dtmf send/recv payload to 101
>>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:5179 AUDIO RTP
>>> [sofia/internal/171 at yx.cl] 10.176.0.1 port 21846 -> 10.176.4.121 port
>>> 50140 codec: 8 ms: 20
>>> 2015-07-09 20:08:19.952352 [DEBUG] switch_rtp.c:3569 Starting timer
>>> [soft] 160 bytes per 20ms
>>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:5477 Set 2833
>>> dtmf send payload to 101
>>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:5483 Set 2833
>>> dtmf receive payload to 101
>>> 2015-07-09 20:08:19.952352 [DEBUG] switch_core_media.c:5505
>>> sofia/internal/171 at yx.cl Set rtp dtmf delay to 40
>>> 2015-07-09 20:08:19.972367 [DEBUG] switch_core_session.c:912 Send signal
>>> sofia/internal/140 at 10.50.100.16:5060 [BREAK]
>>> 2015-07-09 20:08:30.732483 [CRIT] switch_core_io.c:173 sofia/internal/
>>> 140 at 10.50.100.16:5060 reading on a session with no media!
>>>
>>> Nothing happens for 10 seconds between 2015-07-09 20:08:19.972367
>>> and 2015-07-09 20:08:30.732483, FS sends no MOH and then decides to hangup
>>> both calls.
>>>
>>> And here is the output for 1.2 in a similar situation. It's a different
>>> server with different extensions, but I tried to capture the exact same
>>> moment.
>>>
>>>
>>> 2015-07-09 20:02:28.214025 [DEBUG] switch_core_session.c:1016 Send
>>> signal sofia/internal/sip:3315 at 172.17.100.86:5062 [BREAK]
>>> 2015-07-09 20:02:28.214025 [DEBUG] switch_core_session.c:1016 Send
>>> signal sofia/internal/sip:3315 at 172.17.100.86:5062 [BREAK]
>>> 2015-07-09 20:02:28.214025 [DEBUG] sofia.c:5815 Channel sofia/internal/
>>> sip:3315 at 172.17.100.86:5062 entering state [ready][200]
>>> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:5282 Audio Codec Compare
>>> [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
>>> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3190 Set Codec
>>> sofia/internal/sip:3315 at 172.17.100.86:5062 PCMA/8000 20 ms 160 samples
>>> 64000 bits
>>> 2015-07-09 20:02:28.214025 [DEBUG] switch_core_codec.c:111
>>> sofia/internal/sip:3315 at 172.17.100.86:5062 Original read codec set to
>>> PCMA:8
>>> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:5442 Set 2833 dtmf send
>>> payload to 101
>>> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3449 AUDIO RTP
>>> [sofia/internal/sip:3315 at 172.17.100.86:5062] 172.30.0.93 port 23690 ->
>>> 10.10.1.173 port 50692 codec: 8 ms: 20
>>> 2015-07-09 20:02:28.214025 [DEBUG] switch_rtp.c:2040 Starting timer
>>> [soft] 160 bytes per 20ms
>>> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3716 Set 2833 dtmf send
>>> payload to 101
>>> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3722 Set 2833 dtmf
>>> receive payload to 101
>>> 2015-07-09 20:02:28.214025 [DEBUG] sofia_glue.c:3749 sofia/internal/
>>> sip:3315 at 172.17.100.86:5062 Set rtp dtmf delay to 40
>>> 2015-07-09 20:02:28.223984 [DEBUG] switch_ivr_bridge.c:1852
>>> (sofia/external/8781 at 10.20.1.125.6:5075) State Change CS_HIBERNATE ->
>>> CS_CONSUME_MEDIA
>>> 2015-07-09 20:02:28.223984 [DEBUG] switch_core_state_machine.c:415
>>> (sofia/external/8781 at 10.20.1.125.6:5075) Running State Change
>>> CS_CONSUME_MEDIA
>>> 2015-07-09 20:02:28.223984 [DEBUG] switch_core_session.c:933 Send signal
>>> sofia/external/8781 at 10.20.1.125.6:5075 [BREAK]
>>> 2015-07-09 20:02:28.223984 [DEBUG] switch_core_session.c:1351 Send
>>> signal sofia/external/8781 at 10.20.1.125.6:5075 [BREAK]
>>> 2015-07-09 20:02:28.223984 [DEBUG] switch_ivr_bridge.c:1854
>>> (sofia/internal/sip:3315 at 172.17.100.86:5062) State Change CS_HIBERNATE
>>> -> CS_CONSUME_MEDIA
>>> 2015-07-09 20:02:28.223984 [DEBUG] switch_core_session.c:933 Send signal
>>> sofia/internal/sip:3315 at 172.17.100.86:5062 [BREAK]
>>>
>>> ... lots of other things and then FS plays MOH correctly.
>>>
>>> As I mentioned, the same thing happens in master.
>>>
>>> Does anyone have an idea of what could be happening? I can make all the
>>> tests and provide any logs necessary.
>>>
>>> Thanks
>>>
>>> On Thu, Jul 9, 2015 at 11:42 AM, Neven Boric <nboric at yx.cl> wrote:
>>>
>>>> Ok, I tried with master, and something strange happened. I'm pretty
>>>> sure it worked last night, but now I'm testing again and I'm getting the
>>>> same behavior I described with 1.4. I don't even have to get a third phone
>>>> involved. All I do is call from A to B, put B on hold and then wait, and
>>>> about ten seconds later, FS will hangup both calls, as if some timeout was
>>>> triggered. I get the same error on the log:
>>>>
>>>> 2015-07-09 14:36:43.506725 [CRIT] switch_core_io.c:93 sofia/internal/
>>>> 140 at 10.50.100.16:5060 reading on a session with no media!
>>>>
>>>> I also get no MOH on the phone that was put on hold.
>>>>
>>>> On Wed, Jul 8, 2015 at 3:41 PM, Neven Boric <nboric at yx.cl> wrote:
>>>>
>>>>>
>>>>>
>>>>> On Wed, Jul 8, 2015 at 2:17 PM, Steven Ayre <steveayre at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> - I first tried with the 1.2 branch, as that was the version I was
>>>>>>> using locally. The issue with that version is that when a phone holds and
>>>>>>> then unholds a call, I get no audio on the phone that started the hold. I
>>>>>>> found option bypass-media-after-hold, but realized that it was not included
>>>>>>> in the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93
>>>>>>> from the master branch. Now FS correctly tries to restore direct media
>>>>>>> between the endpoints with reINVITEs, but unfortunately includes a
>>>>>>> 'sendonly' in the last SDP to the phone that unholds, so I still get no
>>>>>>> audio. I haven't found a way to fix this.
>>>>>>
>>>>>>
>>>>>> Which probably means there's other commits required. 1.2 is EOL and
>>>>>> no longer supported, 1.4 is the current stable release. I wouldn't spend
>>>>>> any time getting 1.2 working, and instead work on upgrading.
>>>>>>
>>>>>
>>>>> Yes, I know it's not currently supported, but maybe somebody had the
>>>>> same problem and fixed it in a different way. Or maybe somebody could point
>>>>> me in the right direction to remove that final 'sendonly'.
>>>>>
>>>>>
>>>>>>
>>>>>> - I also tried with the 1.4 branch and hold/unhold works correctly,
>>>>>>> but now attended transfer doesn't work, FS after some time ends the call on
>>>>>>> the side that was put on hold waiting to be transferred. This time seems to
>>>>>>> be inconsistent, and sometimes the transfer actually works, so maybe it is
>>>>>>> a timing issue. This seems similar/related to FS-4038.
>>>>>>>
>>>>>>
>>>>>> Also see if you can replicate it on master, as that's close to the
>>>>>> upcoming 1.6 release and will have a lot of changes over 1.4.
>>>>>>
>>>>>
>>>>> I will try with master and report back, thanks.
>>>>>
>>>>>
>>>>>>
>>>>>>
>>>>>> On 8 July 2015 at 14:23, Neven Boric <nboric at yx.cl> wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> I have been using FS for a long time on a local server without
>>>>>>> issues, but now I want to move to a remote server to support some new usage
>>>>>>> scenarios. I'm trying to use inbound-bypass-media=true to keep the audio
>>>>>>> out of the server. This mostly works, but I have two different issues,
>>>>>>> depending on which FS version I use:
>>>>>>>
>>>>>>> - I first tried with the 1.2 branch, as that was the version I was
>>>>>>> using locally. The issue with that version is that when a phone holds and
>>>>>>> then unholds a call, I get no audio on the phone that started the hold. I
>>>>>>> found option bypass-media-after-hold, but realized that it was not included
>>>>>>> in the 1.2 branch, so I "ported" commits 3ecb504, 8fa385b and e4e9b1b9f93
>>>>>>> from the master branch. Now FS correctly tries to restore direct media
>>>>>>> between the endpoints with reINVITEs, but unfortunately includes a
>>>>>>> 'sendonly' in the last SDP to the phone that unholds, so I still get no
>>>>>>> audio. I haven't found a way to fix this.
>>>>>>>
>>>>>>> - I also tried with the 1.4 branch and hold/unhold works correctly,
>>>>>>> but now attended transfer doesn't work, FS after some time ends the call on
>>>>>>> the side that was put on hold waiting to be transferred. This time seems to
>>>>>>> be inconsistent, and sometimes the transfer actually works, so maybe it is
>>>>>>> a timing issue. This seems similar/related to FS-4038.
>>>>>>>
>>>>>>> I can get logs and SIP captures, what would be the preferred way to
>>>>>>> provide them (pcap, inline text, pastebin)? Or maybe it's better to go
>>>>>>> ahead and file a bug?
>>>>>>>
>>>>>>> Best regards
>>>>>>> Neven Boric
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> 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
>>
>
>
>
> --
> 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
>
> _________________________________________________________________________
> 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 Jul 14 19:42:06 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 14 Jul 2015 11:42:06 -0400
Subject: [Freeswitch-users] Issues with simultaneous call hangup
In-Reply-To: <CAO2xAXM3KW7u4t=c4gK6Gt8OXywe7ZkCk+svbQ63K=xS-my2=A@mail.gmail.com>
References: <CAO2xAXM3KW7u4t=c4gK6Gt8OXywe7ZkCk+svbQ63K=xS-my2=A@mail.gmail.com>
Message-ID: <CAEwTsAjP46mxk-FouXSikQKv+dWdd1iah7=-Vwy=ebDyXS9ZwA@mail.gmail.com>

Does this happen in the latest 1.4.20 release

On Tuesday, July 14, 2015, Niraj Roy <techfaltu at gmail.com> wrote:

> Hello Gurus,
>     We are using FreeSWITCH Version 1.4.18~64bit. We observed when both
> caller and caller hangs up the call simultaneously exec_after_bridge_app
> function does not execute for one of the legs.
>     Here is the segment of executing the xml file.
>                 <action application="set"
> data="exec_after_bridge_app=transfer"/>
>                 <action application="set"
> data="exec_after_bridge_arg=endcall-handler XML CALLENDING"/>
>                 <action application="set_zombie_exec"/>
>                 <action application="set" data="continue_on_fail=true"/>
>                 <action inline="true" application="set"
> data="hangup_after_bridge=false"/>
>
> For normal or sequential call clearing everything works fine, but somehow
> this creates a race condition and state mismatch in the freeswitch which I
> don't know.
> Can anybody shed some light on this?
>
> Thanks,
>
>
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From steveayre at gmail.com  Tue Jul 14 19:45:04 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 14 Jul 2015 16:45:04 +0100
Subject: [Freeswitch-users] How do I know uuid both leg before bridge?
In-Reply-To: <1349206034.20150714153525@seznam.cz>
References: <CAG6TCM9xynte8HzQxFXQ0VaZKmz4Njh+qd0TtZ6WP+KNgkxHTw@mail.gmail.com>
	<1971438456.20150714143646@seznam.cz>
	<CAFiqYun34_2EqXjE=MFzuDWCoyAN-AfTLAwDFPAwK4TntPtQDQ@mail.gmail.com>
	<1349206034.20150714153525@seznam.cz>
Message-ID: <CAFiqYumnNqQXVSpxrq2c1Hj3XH18_tx2FFBGE-6TqmkLsDuh5A@mail.gmail.com>

>
> I think it is worth adding these things in questions and answers, so that
> people know about these brackets [ ] and how it works inline=true


They're documented on Confluence / the wiki, but to include them in this
thread and for your info

Variables in {} are set on all the legs
Variables in [] are set only on the leg they appear before

{a=123}[b=456]user/1000|[b=789]user/1001
The call to 1000 will have a=123 & b=456
The call to 1001 will have a=123 & b=789

Dialplan parsing happens in to stages - hunting and execution. In hunting
it goes through the extensions, looks for all matching conditions, and
builds up a list of all actions to execute. It then executes those actions
in order. When an action has inline=true it is executed immediately during
the hunt instead of being queued for the execution stage. That means the
variable value is available for other conditions to check during the
hunting stage. However the actions don't expand the value until they're
executed so ${my_uuid} in your bridge isn't expanded until execution and
therefore the variable doesn't need to be set inline.

On 14 July 2015 at 14:35, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Dobr? den,
>
> Great! Thanks a lot for the help. Now it is working as it should.
>
> I think it is worth adding these things in questions and answers, so that
> people know about these brackets [ ] and how it works inline=true
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 ?ter? 14. ?ervence 2015, 14:51:28, napsal
> jste: *
>
>  Use origination_uuid=${my_uuid} not ${uuid}
>
> ${uuid} is the existing a-leg, origination_uuid will be the new b-leg
>
> Currently you're trying to create a b-leg with the same uuid as the a-leg,
> hence the error.
>
> I also suggest you put origination_uuid in [] not {} - if you ever add
> extra legs you'll have the duplicate error return as all the b-legs would
> get the same uuid.
>
> You also definitely *don't* need inline=true, I would avoid using it as
> it'll execute much earlier than you need it, and if you use set the same
> variable inline later in your dialplan then it may not create the bridge
> with the uuid you expect.
>
> Steve
>
>
>
> On 14 July 2015 at 13:36, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Hi All,
>
> For my tasks need to know uuid both feet before bridge. How can i do this?
>
> If I do in the dialplan
> <action application="set" data="my_uuid=${create_uuid foo}" inline="true"/>
>
> and then
> <action application="bridge" data="{origination_uuid=${uuid}}user/1003"/>
>
> I get an error
> 2015-07-14 14:25:32.350565 [CRIT] switch_core_session.c:2342 Duplicate
> UUID!
> 2015-07-14 14:25:32.350565 [CRIT] mod_sofia.c:4376 Error Creating Session
>
> How correctly before bridge I assign or generate a uuid and use later?
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
> *. 775-415-382 *
> _________________________________________________________________________
> 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.org  Tue Jul 14 19:48:42 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 14 Jul 2015 10:48:42 -0500
Subject: [Freeswitch-users] Transparency in SUBSCRIBE and NOTIFY
In-Reply-To: <0a9b01d0be32$5a93c3e0$0fbb4ba0$@inovax.com.br>
References: <09f801d0bdc7$e2c30e80$a8492b80$@inovax.com.br>
	<CAEJMVkAzWrOUdxn1t=26kOYR+d_6GETbYJUr6A1pw2=m-Nuwvw@mail.gmail.com>
	<0a9b01d0be32$5a93c3e0$0fbb4ba0$@inovax.com.br>
Message-ID: <CAEJMVkCWreG-bgYYV30Y6X58uzN7bn8DGwrt7rnVnTcem87vGw@mail.gmail.com>

FreeSWITCH is not a proxy, so you'll need to add support for what you need
or front end it with a proxy.

On Tue, Jul 14, 2015 at 7:41 AM, Julio Cesar Esteves Cabezas <
jcabezas at inovax.com.br> wrote:

>
>
>     Are you suggesting to combine FreeSWITCH and a SIP Proxy together ?
>
>     Can you ellaborate about that topology ?   I want to retain the unique
> features of FreeSWITCH in my system.
>
>
>
> Thanks.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West
> *Sent:* Monday, July 13, 2015 9:03 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Transparency in SUBSCRIBE and NOTIFY
>
>
>
> Use a proxy! :). Like kamailio or opensips
>
> On Monday, July 13, 2015, Julio Cesar Esteves Cabezas <
> jcabezas at inovax.com.br> wrote:
>
> Hi,
>
>                 My  SIP User-Agents need to establish between themselves a
> mesh of SIP subscriptions (using SUBSCRIBE and NOTIFY).
>
>                 The event-packages of those subscriptions are two, their
> names are:  ?dialog? and ?p-np-session?.
>
>
>
>                 But FreeSWITCH replies 489 Bad Event to SUBSCRIBEs  to
> ?p-np-session? and 200 OK to SUBSCRIBEs to ?dialog? (because FreeSWITCH
> implements this package but that?s not what I need)
>
>                 So, is there some way to make FreeSWITCH transparent to
> those subscriptions (like a vanilla SIP proxy would be)  ?
>
> Any suggestions ?
>
>
>
> Many Thanks.
>
>
>
>
>
>
>
>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>
>
>
> _________________________________________________________________________
> 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
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From alhakeem at gmail.com  Tue Jul 14 20:03:04 2015
From: alhakeem at gmail.com (Abdul Hakeem)
Date: Tue, 14 Jul 2015 17:03:04 +0100
Subject: [Freeswitch-users] File transfer via SIP
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAMTcGfKCujBGiIhKXnPs2YQBAAAAAA==@gmail.com>

Hello,

Has anyone successfully implemented File Transfer in Freeswitch ?
I read this old response, I am hoping there's a follow through:
http://lists.freeswitch.org/pipermail/freeswitch-users/2009-September/047553.htm
l

Best regards,
Abdul Hakeem




From brian at freeswitch.org  Tue Jul 14 20:05:43 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 14 Jul 2015 11:05:43 -0500
Subject: [Freeswitch-users] File transfer via SIP
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAMTcGfKCujBGiIhKXnPs2YQBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAMTcGfKCujBGiIhKXnPs2YQBAAAAAA==@gmail.com>
Message-ID: <CAEJMVkDovNDQM2okFTBvx2dr622X3oR7EfEj0ww1LOxvZKVLmA@mail.gmail.com>

Which protocol? MRSP?

On Tue, Jul 14, 2015 at 11:03 AM, Abdul Hakeem <alhakeem at gmail.com> wrote:

> Hello,
>
> Has anyone successfully implemented File Transfer in Freeswitch ?
> I read this old response, I am hoping there's a follow through:
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-September/047553.htm
> l
>
> Best regards,
> Abdul Hakeem
>
>
>
> _________________________________________________________________________
> 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
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From mike at jerris.com  Tue Jul 14 20:07:17 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 14 Jul 2015 12:07:17 -0400
Subject: [Freeswitch-users] File transfer via SIP
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAMTcGfKCujBGiIhKXnPs2YQBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAMTcGfKCujBGiIhKXnPs2YQBAAAAAA==@gmail.com>
Message-ID: <CAEwTsAg72q1b5KXD=TvRwkJqtpX_9CpY2A8cbsQ5f0rT8_KXLQ@mail.gmail.com>

No, it has not been implemented.

On Tuesday, July 14, 2015, Abdul Hakeem <alhakeem at gmail.com> wrote:

> Hello,
>
> Has anyone successfully implemented File Transfer in Freeswitch ?
> I read this old response, I am hoping there's a follow through:
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-September/047553.htm
> l
>
> Best regards,
> Abdul Hakeem
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> 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 <javascript:;>
> 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 govoiper at gmail.com  Tue Jul 14 20:49:14 2015
From: govoiper at gmail.com (SamyGo)
Date: Tue, 14 Jul 2015 12:49:14 -0400
Subject: [Freeswitch-users] Execute dialplan after valet_park
Message-ID: <CAJUJwtipc+yhpck+w4M4mYYPk0ZRa5NqWhytRChbsi4c6_nwdA@mail.gmail.com>

Hi All,

I need some help in finding  a way to execute some dialplan code after the
parked user Hangups from the parking lot.

I've a scenario where Parked party decides to hangup while listening to MOH
while parked. The Dialplan just Hangsup right away and I have no control.

I've tried using: *session_in_hangup_hook*, and
*api_hangup_hook=myscript.lua *but they get executed for the parker just
after they hear the Parked slot number.

Then I've tried *nolocal:export* these two variables for the other leg but
nothing happens.

Kindly suggest what other option are there.

Best Regards.
Sammy
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From mrjoli021 at gmail.com  Tue Jul 14 21:58:10 2015
From: mrjoli021 at gmail.com (Joli Martinez)
Date: Tue, 14 Jul 2015 13:58:10 -0400
Subject: [Freeswitch-users] internal 5060 not working.
Message-ID: <139A6536-A81C-4C84-B114-1969B869DE5F@gmail.com>



I have changed the internal.xml to point to 192.168.21.250 and for some reason the 192.168.1.250 keeps showing up.  Also I need the profile to be on 5060.  I have also changed the vars.xml to show 5060 and nothing.  I have restarted Freeswitch after each change and verified that nothing else is running on 5060.  The server does have two interfaces and the default gw is set to 192.168.21.50.  The 192.168.1.250 is only for out of band management and will not need access to FS.  I have selinux disabled and the iptables stopped as well.  The external is running on 5080 but not the internal.


Sofia status
freeswitch at internal> sofia status 
                     Name	   Type	                                      Data	State
=================================================================================================
            192.168.1.250	  alias	                                  internal	ALIASED
           192.168.21.250	  alias	                                  internal	ALIASED
                 external	profile	          sip:mod_sofia at 1.1.1.1:5080 <http://sip:mod_sofia at 1.1.1.1:5080/>	RUNNING (0)
            internal-ipv6	profile	          sip:mod_sofia at 1.1.1.1:5040 <http://sip:mod_sofia at 1.1.1.1:5040/>	RUNNING (0)
                 internal	profile	          sip:mod_sofia at 1.1.1.1:5041 <http://sip:mod_sofia at 1.1.1.1:5041/>	RUNNING (0)
=================================================================================================
3 profiles 2 aliases


netstat -nat
[root at Switch01 sip_profiles]# netstat -nat | grep 192.168.21.250
tcp        0      0 192.168.21.250:5080 <http://192.168.21.250:5080/>         0.0.0.0:*                   LISTEN   

internal.xml
<!-- port to bind to for sip traffic -->
                <param name="sip-port" value="$${internal_sip_port}"/>
                <!-- <param name="sip-port" value="5060"/> -->
                <param name="dialplan" value="XML"/>
                <param name="dtmf-duration" value="2000"/>
                <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
                <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
                <param name="rtp-timer-name" value="soft"/>
                <!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
                <!-- <param name="rtp-ip" value="$${local_ip_v4}"/> -->
                <param name="rtp-ip" value="192.168.21.250"/>
                <!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
                <!-- <param name="sip-ip" value="$${local_ip_v4}"/> -->
                <param name="sip-ip" value="192.168.21.250"/>


vars.xml
<!-- Internal SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
  <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
  <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
  <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>
  <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>

  <!-- External SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
  <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
  <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>
  <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/>

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From manpower13.cse at gmail.com  Wed Jul 15 01:07:08 2015
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Wed, 15 Jul 2015 02:37:08 +0530
Subject: [Freeswitch-users] Mod_Verto login problem
Message-ID: <CAMRuXGgTs0+B=17w0KCsFBJN_MNqzz+4ULSTk4CfRjUtn0W26Q@mail.gmail.com>

Hi,

  I successfully setup mod_verto ,But when i try to login from chrome
browser  it show "error":{"code":-32001,"message:'Login Incorrect"}

if i use the same username and password any sip client it work
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From brian at freeswitch.org  Wed Jul 15 01:54:42 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 14 Jul 2015 16:54:42 -0500
Subject: [Freeswitch-users] Mod_Verto login problem
In-Reply-To: <CAMRuXGgTs0+B=17w0KCsFBJN_MNqzz+4ULSTk4CfRjUtn0W26Q@mail.gmail.com>
References: <CAMRuXGgTs0+B=17w0KCsFBJN_MNqzz+4ULSTk4CfRjUtn0W26Q@mail.gmail.com>
Message-ID: <CAEJMVkDqBi6ZHrhpYnZQmWOwfdhPsC99RJ49ktDH66fQObZ0HQ@mail.gmail.com>

Make sure you add this as a param on your user in the directory:

<param name="jsonrpc-allowed-methods" value="verto"/>

:)



On Tue, Jul 14, 2015 at 4:07 PM, Murugan Pandian <manpower13.cse at gmail.com>
wrote:

> Hi,
>
>   I successfully setup mod_verto ,But when i try to login from chrome
> browser  it show "error":{"code":-32001,"message:'Login Incorrect"}
>
> if i use the same username and password any sip client it work
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
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From anthony.minessale at gmail.com  Wed Jul 15 02:16:24 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 14 Jul 2015 17:16:24 -0500
Subject: [Freeswitch-users] PLEASE DO NOT REPORT ISSUES HERE USE
	freeswitch.org/jira
Message-ID: <CAKbxfG_DfSJjTETCJ-CNZeP34y+LZgcAzg-r_AYXBbVRoo4RzA@mail.gmail.com>

When I look at my email I see countless emails with "ISSUE" "PROBLEM" "NOT
WORKING"
We have a bug tracker for such things.  We mostly will lose track of your
email threads that are random issues.  Its impossible to field them from an
email client and keeping track of them mentally.

If you want to learn more about how to make the most of your community
experience; Join the ClueCon weekly online calls to get more info, read the
docs in the WIKI or come see us in person at ClueCon this summer.
http://cluecon.com




-- 
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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From blasterjr at gmail.com  Wed Jul 15 03:57:33 2015
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Tue, 14 Jul 2015 17:57:33 -0600
Subject: [Freeswitch-users] Execute dialplan after valet_park
In-Reply-To: <CAJUJwtipc+yhpck+w4M4mYYPk0ZRa5NqWhytRChbsi4c6_nwdA@mail.gmail.com>
References: <CAJUJwtipc+yhpck+w4M4mYYPk0ZRa5NqWhytRChbsi4c6_nwdA@mail.gmail.com>
Message-ID: <CAO3hyzE0joUxdFeQM4oJ9K64qdLcG_4uock8v3AqHDWc6pSdaw@mail.gmail.com>

You might be able to use an event hook with lua, you can catch the CUSTOM
with a subclass of valet_parking::info, then you can check for the "action"
and it'll be "exit" when they leave (any kind of leave) then can likely
check for a disposition.  I'm not currently using this method, but its
someting we looked into for monitoring when people hangup from the parking
lot.

On Tue, Jul 14, 2015 at 10:49 AM, SamyGo <govoiper at gmail.com> wrote:

> Hi All,
>
> I need some help in finding  a way to execute some dialplan code after the
> parked user Hangups from the parking lot.
>
> I've a scenario where Parked party decides to hangup while listening to
> MOH while parked. The Dialplan just Hangsup right away and I have no
> control.
>
> I've tried using: *session_in_hangup_hook*, and
> *api_hangup_hook=myscript.lua *but they get executed for the parker just
> after they hear the Parked slot number.
>
> Then I've tried *nolocal:export* these two variables for the other leg
> but nothing happens.
>
> Kindly suggest what other option are there.
>
> Best Regards.
> Sammy
>
>
> _________________________________________________________________________
> 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 lexxua at gmail.com  Wed Jul 15 04:52:58 2015
From: lexxua at gmail.com (Volodymyr Fedorov)
Date: Wed, 15 Jul 2015 03:52:58 +0300
Subject: [Freeswitch-users] Mod_Verto login problem
In-Reply-To: <CAEJMVkDqBi6ZHrhpYnZQmWOwfdhPsC99RJ49ktDH66fQObZ0HQ@mail.gmail.com>
References: <CAMRuXGgTs0+B=17w0KCsFBJN_MNqzz+4ULSTk4CfRjUtn0W26Q@mail.gmail.com>
	<CAEJMVkDqBi6ZHrhpYnZQmWOwfdhPsC99RJ49ktDH66fQObZ0HQ@mail.gmail.com>
Message-ID: <CAO_GGArsmFoSKz8YRdfR3S7znVkXE+qxxEtZzMxASdKM+VA-4A@mail.gmail.com>

Hello, can mod_verto use param name="a1-hash" or it uses only plaintext
password ?
For me it doesn't work I get "Login Incorrect".

On Wed, Jul 15, 2015 at 12:54 AM, Brian West <brian at freeswitch.org> wrote:

> Make sure you add this as a param on your user in the directory:
>
> <param name="jsonrpc-allowed-methods" value="verto"/>
>
> :)
>
>
>
> On Tue, Jul 14, 2015 at 4:07 PM, Murugan Pandian <manpower13.cse at gmail.com
> > wrote:
>
>> Hi,
>>
>>   I successfully setup mod_verto ,But when i try to login from chrome
>> browser  it show "error":{"code":-32001,"message:'Login Incorrect"}
>>
>> if i use the same username and password any sip client it work
>>
>>
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> 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
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>
> _________________________________________________________________________
> 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
>



-- 
Cheers !
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From brian at freeswitch.org  Wed Jul 15 05:02:32 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 14 Jul 2015 20:02:32 -0500
Subject: [Freeswitch-users] Mod_Verto login problem
In-Reply-To: <CAO_GGArsmFoSKz8YRdfR3S7znVkXE+qxxEtZzMxASdKM+VA-4A@mail.gmail.com>
References: <CAMRuXGgTs0+B=17w0KCsFBJN_MNqzz+4ULSTk4CfRjUtn0W26Q@mail.gmail.com>
	<CAEJMVkDqBi6ZHrhpYnZQmWOwfdhPsC99RJ49ktDH66fQObZ0HQ@mail.gmail.com>
	<CAO_GGArsmFoSKz8YRdfR3S7znVkXE+qxxEtZzMxASdKM+VA-4A@mail.gmail.com>
Message-ID: <CAEJMVkD9edc_mm+nJaUr7hYBqPBLZQttE406Zpg3Qxb0FCHa0A@mail.gmail.com>

Only plain text.

On Tue, Jul 14, 2015 at 7:52 PM, Volodymyr Fedorov <lexxua at gmail.com> wrote:

> Hello, can mod_verto use param name="a1-hash" or it uses only plaintext
> password ?
> For me it doesn't work I get "Login Incorrect".
>
> On Wed, Jul 15, 2015 at 12:54 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Make sure you add this as a param on your user in the directory:
>>
>> <param name="jsonrpc-allowed-methods" value="verto"/>
>>
>> :)
>>
>>
>>
>> On Tue, Jul 14, 2015 at 4:07 PM, Murugan Pandian <
>> manpower13.cse at gmail.com> wrote:
>>
>>> Hi,
>>>
>>>   I successfully setup mod_verto ,But when i try to login from chrome
>>> browser  it show "error":{"code":-32001,"message:'Login Incorrect"}
>>>
>>> if i use the same username and password any sip client it work
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> 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
>>
>> ClueCon 2015 Call for Speakers
>> <https://www.cluecon.com/call-for-speakers/> | Register
>> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>>
>> _________________________________________________________________________
>> 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
>>
>
>
>
> --
> Cheers !
>
> _________________________________________________________________________
> 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
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From nandy1925 at gmail.com  Wed Jul 15 05:53:22 2015
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Wed, 15 Jul 2015 09:53:22 +0800
Subject: [Freeswitch-users] verto.conf.xml parameter for public access
Message-ID: <CAHKqMb5s9NGtaiEHO5Z5uWDi=gCdeTEmsPBqU6dLRwHc4r=-LQ@mail.gmail.com>

I succesfully installed mod_verto in my intranet.  FS is behind NAT. Next
step, I'm setting up where remote web clients can login.  I setup my router
DMZ to FS.

This parameter is commented out  "ext-rtp-ip" in the Wiki. Is this
parameter necessary for accepting remote clients?
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From manpower13.cse at gmail.com  Wed Jul 15 09:27:04 2015
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Wed, 15 Jul 2015 10:57:04 +0530
Subject: [Freeswitch-users] Mod_Verto login problem
In-Reply-To: <CAEJMVkD9edc_mm+nJaUr7hYBqPBLZQttE406Zpg3Qxb0FCHa0A@mail.gmail.com>
References: <CAMRuXGgTs0+B=17w0KCsFBJN_MNqzz+4ULSTk4CfRjUtn0W26Q@mail.gmail.com>
	<CAEJMVkDqBi6ZHrhpYnZQmWOwfdhPsC99RJ49ktDH66fQObZ0HQ@mail.gmail.com>
	<CAO_GGArsmFoSKz8YRdfR3S7znVkXE+qxxEtZzMxASdKM+VA-4A@mail.gmail.com>
	<CAEJMVkD9edc_mm+nJaUr7hYBqPBLZQttE406Zpg3Qxb0FCHa0A@mail.gmail.com>
Message-ID: <CAMRuXGgnAoX9jtMQnzJ1Ea2r0Ki_spA_8ZV2CE=w-ZcNuQWpOQ@mail.gmail.com>

Hi Brain,

    I don't think any issue in my user directory

     <params>

      <param name="password" value="$${default_password}"/>

      <param name="vm-password" value="1000"/>

       <param name="verto-context" value="public"/>

      <param name="verto-dialplan" value="XML"/>

      <param name="jsonrpc-allowed-methods" value="verto"/>

    <param name="jsonrpc-allowed-event-channels"
value="demo,conference,presenc$

    </params>


  When i try to login getting following message in my console



  2015-07-15 05:25:26.522080 [ALERT] mod_verto.c:1357 READ
192.168.59.3:50097 [{

"jsonrpc": "2.0",

"method": "login",

"params": {

 "login": "1000 at xxxxxxx",

 "passwd": "12345",

 "sessid": "24a57443-715f-7d7a-3a86-c8326db9aa29"

},

"id": 6

}]

2015-07-15 05:25:26.522080 [ALERT] mod_verto.c:604 WRITE 192.168.59.3:50097
[{

"jsonrpc": "2.0",

"id": 6,

"error": {

 "code": -32001,

 "message": "Login Incorrect"

}

}]





On Wed, Jul 15, 2015 at 6:32 AM, Brian West <brian at freeswitch.org> wrote:

> Only plain text.
>
> On Tue, Jul 14, 2015 at 7:52 PM, Volodymyr Fedorov <lexxua at gmail.com>
> wrote:
>
>> Hello, can mod_verto use param name="a1-hash" or it uses only plaintext
>> password ?
>> For me it doesn't work I get "Login Incorrect".
>>
>> On Wed, Jul 15, 2015 at 12:54 AM, Brian West <brian at freeswitch.org>
>> wrote:
>>
>>> Make sure you add this as a param on your user in the directory:
>>>
>>> <param name="jsonrpc-allowed-methods" value="verto"/>
>>>
>>> :)
>>>
>>>
>>>
>>> On Tue, Jul 14, 2015 at 4:07 PM, Murugan Pandian <
>>> manpower13.cse at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>>   I successfully setup mod_verto ,But when i try to login from chrome
>>>> browser  it show "error":{"code":-32001,"message:'Login Incorrect"}
>>>>
>>>> if i use the same username and password any sip client it work
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>>>
>>> ClueCon 2015 Call for Speakers
>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>>>
>>> _________________________________________________________________________
>>> 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
>>>
>>
>>
>>
>> --
>> Cheers !
>>
>> _________________________________________________________________________
>> 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
>
> _________________________________________________________________________
> 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 dm at dwide.com  Wed Jul 15 10:24:14 2015
From: dm at dwide.com (Dmitry Mordovin)
Date: Wed, 15 Jul 2015 10:24:14 +0400
Subject: [Freeswitch-users] Call thru external gate released before INVITE
	(NORMAL_UNSPECIFIED)
Message-ID: <55A5FC8E.1070901@dwide.com>

Hello!

Very strange, some time ago this works, but now not... and I don't know 
where I was made mistake...

 From FS CLI do command

*originate {ignore_early_media=true}sofia/internal/00294539100% 
&bridge(sofia/gateway/voip_provider/919262417024 at voip_provider)*

Call to user 00294539100 established, but second call to external is 
dropped before send INVITE... (local cause NORMAL_UNSPECIFIED)

Could someone give me idea where is a problem?




Trace here


2015-07-15 10:09:19.375204 [DEBUG] switch_ivr_originate.c:2100 Parsing 
global variables
2015-07-15 10:09:19.375204 [DEBUG] switch_event.c:1698 Parsing variable 
[ignore_early_media]=[true]
2015-07-15 10:09:19.375204 [NOTICE] switch_channel.c:1075 New Channel 
sofia/internal/00294539100 [680855ef-9973-43c4-bb9d-7ee1dd27d392]
2015-07-15 10:09:19.375204 [DEBUG] mod_sofia.c:4701 
(sofia/internal/00294539100) State Change CS_NEW -> CS_INIT
2015-07-15 10:09:19.375204 [DEBUG] switch_core_session.c:1396 Send 
signal sofia/internal/00294539100 [BREAK]
2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:472 
(sofia/internal/00294539100) Running State Change CS_INIT
2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:512 
(sofia/internal/00294539100) State INIT
2015-07-15 10:09:19.375204 [DEBUG] mod_sofia.c:87 
sofia/internal/00294539100 SOFIA INIT
freeswitch at internal> 2015-07-15 10:09:19.375204 [DEBUG] 
sofia_glue.c:1236 sofia/internal/00294539100 sending invite version: 
1.5.15b git acdb1ca 2015-05-17 18:45:52Z 32bit
Local SDP:
v=0
o=FreeSWITCH 1436918327 1436918328 IN IP4 194.87.7.17
s=FreeSWITCH
c=IN IP4 194.87.7.17
t=0 0
m=audio 22232 RTP/AVP 102 9 0 8 3 101 13
a=rtpmap:102 opus/48000/2
a=fmtp:102 useinbandfec=1; usedtx=1; maxaveragebitrate=30000; 
maxplaybackrate=48000; ptime=20; minptime=20; maxptime=20; samplerate=48000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 29716 RTP/AVP 103
a=rtpmap:103 VP8/90000

2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:40 
sofia/internal/00294539100 Standard INIT
2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:48 
(sofia/internal/00294539100) State Change CS_INIT -> CS_ROUTING
2015-07-15 10:09:19.375204 [DEBUG] switch_core_session.c:1396 Send 
signal sofia/internal/00294539100 [BREAK]
2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:512 
(sofia/internal/00294539100) State INIT going to sleep
2015-07-15 10:09:19.375204 [DEBUG] switch_core_session.c:1061 Send 
signal sofia/internal/00294539100 [BREAK]
2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:472 
(sofia/internal/00294539100) Running State Change CS_ROUTING
2015-07-15 10:09:19.375204 [DEBUG] sofia.c:6627 Channel 
sofia/internal/00294539100 entering state [calling][0]
2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:528 
(sofia/internal/00294539100) State ROUTING
2015-07-15 10:09:19.375204 [DEBUG] mod_sofia.c:123 
sofia/internal/00294539100 SOFIA ROUTING
2015-07-15 10:09:19.375204 [DEBUG] switch_ivr_originate.c:67 
(sofia/internal/00294539100) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2015-07-15 10:09:19.375204 [DEBUG] switch_core_session.c:1396 Send 
signal sofia/internal/00294539100 [BREAK]
2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:528 
(sofia/internal/00294539100) State ROUTING going to sleep
2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:472 
(sofia/internal/00294539100) Running State Change CS_CONSUME_MEDIA
2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:547 
(sofia/internal/00294539100) State CONSUME_MEDIA
2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:547 
(sofia/internal/00294539100) State CONSUME_MEDIA going to sleep
2015-07-15 10:09:19.455217 [DEBUG] switch_core_session.c:1061 Send 
signal sofia/internal/00294539100 [BREAK]
2015-07-15 10:09:19.455217 [DEBUG] switch_core_session.c:1061 Send 
signal sofia/internal/00294539100 [BREAK]
2015-07-15 10:09:19.455217 [DEBUG] sofia.c:6627 Channel 
sofia/internal/00294539100 entering state [proceeding][180]
2015-07-15 10:09:19.455217 [NOTICE] sofia.c:6729 Ring-Ready 
sofia/internal/00294539100!
2015-07-15 10:09:19.455217 [DEBUG] switch_channel.c:3297 
(sofia/internal/00294539100) Callstate Change DOWN -> RINGING
2015-07-15 10:09:22.075203 [DEBUG] switch_core_session.c:1061 Send 
signal sofia/internal/00294539100 [BREAK]
2015-07-15 10:09:22.075203 [DEBUG] switch_core_session.c:1061 Send 
signal sofia/internal/00294539100 [BREAK]
2015-07-15 10:09:22.075203 [DEBUG] sofia.c:6627 Channel 
sofia/internal/00294539100 entering state [completing][200]
2015-07-15 10:09:22.075203 [DEBUG] sofia.c:6637 Remote SDP:
v=0
o=floxent 3645929363 1 IN IP4 192.168.223.220
s=sflphone
c=IN IP4 192.168.223.220
t=0 0
m=audio 28878 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:28879 IN IP4 192.168.223.220
m=video 0 RTP/AVP 103

2015-07-15 10:09:22.075203 [DEBUG] switch_core_session.c:1061 Send 
signal sofia/internal/00294539100 [BREAK]
2015-07-15 10:09:22.075203 [DEBUG] switch_core_session.c:1061 Send 
signal sofia/internal/00294539100 [BREAK]
2015-07-15 10:09:22.075203 [DEBUG] sofia.c:6627 Channel 
sofia/internal/00294539100 entering state [ready][200]
2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:3672 Audio Codec 
Compare [G722:9:8000:20:64000:1]/[opus:116:48000:20:0:1]
2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:3672 Audio Codec 
Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:3727 Audio Codec 
Compare [G722:9:8000:20:64000:1] ++++ is saved as a match
2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:3672 Audio Codec 
Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:3672 Audio Codec 
Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:3672 Audio Codec 
Compare [G722:9:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:3588 Set 
telephone-event payload to 101
2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:2507 Set Codec 
sofia/internal/00294539100 G722/8000 20 ms 160 samples 64000 bits 1 channels
2015-07-15 10:09:22.075203 [DEBUG] switch_core_codec.c:111 
sofia/internal/00294539100 Original read codec set to G722:9
2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:3926 Set 2833 
dtmf send payload to 101
2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:5171 AUDIO RTP 
[sofia/internal/00294539100] 194.87.7.17 port 22232 -> 192.168.223.220 
port 28878 codec: 9 ms: 20
2015-07-15 10:09:22.075203 [DEBUG] switch_rtp.c:3569 Starting timer 
[soft] 160 bytes per 20ms
2015-07-15 10:09:22.075203 [INFO] switch_core_media.c:5388 Activating 
RTCP PORT 28879
2015-07-15 10:09:22.075203 [DEBUG] switch_rtp.c:3919 RTCP send rate is: 
5000 and packet rate is: 20000 Remote Port: 28879
2015-07-15 10:09:22.075203 [DEBUG] switch_rtp.c:2349 Setting RTCP remote 
addr to 192.168.223.220:28879
2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:5469 Set 2833 
dtmf send payload to 101
2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:5475 Set 2833 
dtmf receive payload to 101
2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:5497 
sofia/internal/00294539100 Set rtp dtmf delay to 40
*2015-07-15 10:09:22.075203 [NOTICE] sofia.c:7488 Channel 
[sofia/internal/00294539100] has been answered*
2015-07-15 10:09:22.075203 [DEBUG] switch_channel.c:3711 
(sofia/internal/00294539100) Callstate Change RINGING -> ACTIVE
2015-07-15 10:09:22.095205 [DEBUG] switch_ivr_originate.c:3577 Originate 
Resulted in Success: [sofia/internal/00294539100]
2015-07-15 10:09:22.095205 [INFO] switch_channel.c:3082 
sofia/internal/00294539100 Flipping CID from "" <0000000000> to 
"Outbound Call" <00294539100%>
2015-07-15 10:09:22.095205 [DEBUG] mod_commands.c:4469 
(sofia/internal/00294539100) State Change CS_CONSUME_MEDIA -> CS_EXECUTE
2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1396 Send 
signal sofia/internal/00294539100 [BREAK]
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:472 
(sofia/internal/00294539100) Running State Change CS_EXECUTE
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:535 
(sofia/internal/00294539100) State EXECUTE
2015-07-15 10:09:22.095205 [DEBUG] mod_sofia.c:178 
sofia/internal/00294539100 SOFIA EXECUTE
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:258 
sofia/internal/00294539100 Standard EXECUTE
EXECUTE sofia/internal/00294539100 
bridge(sofia/gateway/voip_provider/919262417024 at voip_provider)
2015-07-15 10:09:22.095205 [DEBUG] switch_channel.c:1786 
(sofia/internal/00294539100) Callstate Change ACTIVE -> RING_WAIT
2015-07-15 10:09:22.095205 [DEBUG] switch_ivr_originate.c:2100 Parsing 
global variables
2015-07-15 10:09:22.095205 [NOTICE] switch_channel.c:1075 New Channel 
sofia/external/919262417024 at voip_provider 
[32799f05-3018-4a48-bea4-1d3cc3700835]
2015-07-15 10:09:22.095205 [DEBUG] mod_sofia.c:4701 
(sofia/external/919262417024 at voip_provider) State Change CS_NEW -> CS_INIT
2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1396 Send 
signal sofia/external/919262417024 at voip_provider [BREAK]
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:472 
(sofia/external/919262417024 at voip_provider) Running State Change CS_INIT
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:512 
(sofia/external/919262417024 at voip_provider) State INIT
2015-07-15 10:09:22.095205 [DEBUG] mod_sofia.c:87 
sofia/external/919262417024 at voip_provider SOFIA INIT
2015-07-15 10:09:22.095205 [DEBUG] sofia_glue.c:1236 
sofia/external/919262417024 at voip_provider sending invite version: 
1.5.15b git acdb1ca 2015-05-17 18:45:52Z 32bit
Local SDP:
v=0
o=FreeSWITCH 1436921310 1436921311 IN IP4 194.87.7.17
s=FreeSWITCH
c=IN IP4 194.87.7.17
t=0 0
m=audio 19252 RTP/AVP 9 0 8 3 101 13
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 31754 RTP/AVP 103
a=rtpmap:103 VP8/90000

2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:40 
sofia/external/919262417024 at voip_provider Standard INIT
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:48 
(sofia/external/919262417024 at voip_provider) State Change CS_INIT -> 
CS_ROUTING
2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1396 Send 
signal sofia/external/919262417024 at voip_provider [BREAK]
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:512 
(sofia/external/919262417024 at voip_provider) State INIT going to sleep
2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1061 Send 
signal sofia/external/919262417024 at voip_provider [BREAK]
2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1061 Send 
signal sofia/external/919262417024 at voip_provider [BREAK]
2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1061 Send 
signal sofia/external/919262417024 at voip_provider [BREAK]
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:472 
(sofia/external/919262417024 at voip_provider) Running State Change CS_ROUTING
2015-07-15 10:09:22.095205 [DEBUG] sofia.c:6627 Channel 
sofia/external/919262417024 at voip_provider entering state [terminated][900]
*2015-07-15 10:09:22.095205 [NOTICE] sofia.c:7543 Hangup 
sofia/external/919262417024 at voip_provider [CS_ROUTING] [NORMAL_UNSPECIFIED]*
2015-07-15 10:09:22.095205 [DEBUG] switch_ivr_originate.c:3720 Originate 
Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED]
2015-07-15 10:09:22.095205 [DEBUG] switch_channel.c:3242 Send signal 
sofia/external/919262417024 at voip_provider [KILL]
2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1396 Send 
signal sofia/external/919262417024 at voip_provider [BREAK]
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:528 
(sofia/external/919262417024 at voip_provider) State ROUTING
2015-07-15 10:09:22.095205 [DEBUG] mod_sofia.c:123 
sofia/external/919262417024 at voip_provider SOFIA ROUTING
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:528 
(sofia/external/919262417024 at voip_provider) State ROUTING going to sleep
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:472 
(sofia/external/919262417024 at voip_provider) Running State Change CS_HANGUP
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:735 
(sofia/external/919262417024 at voip_provider) Callstate Change DOWN -> HANGUP
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:737 
(sofia/external/919262417024 at voip_provider) State HANGUP
2015-07-15 10:09:22.095205 [DEBUG] mod_sofia.c:413 Channel 
sofia/external/919262417024 at voip_provider hanging up, cause: 
NORMAL_UNSPECIFIED
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:60 
sofia/external/919262417024 at voip_provider Standard HANGUP, cause: 
NORMAL_UNSPECIFIED
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:737 
(sofia/external/919262417024 at voip_provider) State HANGUP going to sleep
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:504 
(sofia/external/919262417024 at voip_provider) State Change CS_HANGUP -> 
CS_REPORTING
2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1396 Send 
signal sofia/external/919262417024 at voip_provider [BREAK]
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:472 
(sofia/external/919262417024 at voip_provider) Running State Change 
CS_REPORTING
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:823 
(sofia/external/919262417024 at voip_provider) State REPORTING
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:104 
sofia/external/919262417024 at voip_provider Standard REPORTING, cause: 
NORMAL_UNSPECIFIED
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:823 
(sofia/external/919262417024 at voip_provider) State REPORTING going to sleep
2015-07-15 10:09:22.095205 [DEBUG] switch_channel.c:1999 
(sofia/internal/00294539100) Callstate Change RING_WAIT -> ACTIVE
2015-07-15 10:09:22.095205 [INFO] mod_dptools.c:3268 Originate Failed.  
Cause: NORMAL_UNSPECIFIED
2015-07-15 10:09:22.095205 [NOTICE] switch_channel.c:4747 Hangup 
sofia/internal/00294539100 [CS_EXECUTE] [NORMAL_UNSPECIFIED]
2015-07-15 10:09:22.095205 [DEBUG] switch_channel.c:3242 Send signal 
sofia/internal/00294539100 [KILL]
2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1396 Send 
signal sofia/internal/00294539100 [BREAK]
2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:498 
(sofia/external/919262417024 at voip_provider) State Change CS_REPORTING -> 
CS_DESTROY
2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1396 Send 
signal sofia/external/919262417024 at voip_provider [BREAK]
2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1623 Session 
202 (sofia/external/919262417024 at voip_provider) Locked, Waiting on 
external entities
2015-07-15 10:09:22.095205 [NOTICE] switch_core_session.c:1641 Session 
202 (sofia/external/919262417024 at voip_provider) Ended
2015-07-15 10:09:22.095205 [NOTICE] switch_core_session.c:1645 Close 
Channel sofia/external/919262417024 at voip_provider [CS_DESTROY]

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From s.safarov at gmail.com  Wed Jul 15 11:14:54 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 15 Jul 2015 10:14:54 +0300
Subject: [Freeswitch-users] Call thru external gate released before
	INVITE (NORMAL_UNSPECIFIED)
In-Reply-To: <55A5FC8E.1070901@dwide.com>
References: <55A5FC8E.1070901@dwide.com>
Message-ID: <CAHtxdDeBST6kmJeEn=VN_TzDdCv1rppT9g5qxX4bcyEq-Hcn+g@mail.gmail.com>

What is "voip_provider"?
Please enable
1) sofia global siptrace on
2) sofia loglevel all 9
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From lists at telefaks.de  Wed Jul 15 11:16:29 2015
From: lists at telefaks.de (Peter Steinbach)
Date: Wed, 15 Jul 2015 09:16:29 +0200
Subject: [Freeswitch-users] Call thru external gate released before
 INVITE (NORMAL_UNSPECIFIED)
In-Reply-To: <55A5FC8E.1070901@dwide.com>
References: <55A5FC8E.1070901@dwide.com>
Message-ID: <55A608CD.5020604@telefaks.de>

Hello Dmitry,

just some thoughts: Is your gateway voip_provider definded properly? Can
you dial this number voip_to provider manually with originate &park()?
Is the DNS lookup to voip_provider successfull? It may be usefull o make
a pcap file and investigate with Wireshark.


Best regards
Peter

On 07/15/15 08:24, Dmitry Mordovin wrote:
> Hello!
>
> Very strange, some time ago this works, but now not... and I don't
> know where I was made mistake...
>
> From FS CLI do command
>
> *originate {ignore_early_media=true}sofia/internal/00294539100%
> &bridge(sofia/gateway/voip_provider/919262417024 at voip_provider)*
>
> Call to user 00294539100 established, but second call to external is
> dropped before send INVITE... (local cause NORMAL_UNSPECIFIED)
>
> Could someone give me idea where is a problem?
>
>
>
>
> Trace here
>
>
> 2015-07-15 10:09:19.375204 [DEBUG] switch_ivr_originate.c:2100 Parsing
> global variables
> 2015-07-15 10:09:19.375204 [DEBUG] switch_event.c:1698 Parsing
> variable [ignore_early_media]=[true]
> 2015-07-15 10:09:19.375204 [NOTICE] switch_channel.c:1075 New Channel
> sofia/internal/00294539100 [680855ef-9973-43c4-bb9d-7ee1dd27d392]
> 2015-07-15 10:09:19.375204 [DEBUG] mod_sofia.c:4701
> (sofia/internal/00294539100) State Change CS_NEW -> CS_INIT
> 2015-07-15 10:09:19.375204 [DEBUG] switch_core_session.c:1396 Send
> signal sofia/internal/00294539100 [BREAK]
> 2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:472
> (sofia/internal/00294539100) Running State Change CS_INIT
> 2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:512
> (sofia/internal/00294539100) State INIT
> 2015-07-15 10:09:19.375204 [DEBUG] mod_sofia.c:87
> sofia/internal/00294539100 SOFIA INIT
> freeswitch at internal> 2015-07-15 10:09:19.375204 [DEBUG]
> sofia_glue.c:1236 sofia/internal/00294539100 sending invite version:
> 1.5.15b git acdb1ca 2015-05-17 18:45:52Z 32bit
> Local SDP:
> v=0
> o=FreeSWITCH 1436918327 1436918328 IN IP4 194.87.7.17
> s=FreeSWITCH
> c=IN IP4 194.87.7.17
> t=0 0
> m=audio 22232 RTP/AVP 102 9 0 8 3 101 13
> a=rtpmap:102 opus/48000/2
> a=fmtp:102 useinbandfec=1; usedtx=1; maxaveragebitrate=30000;
> maxplaybackrate=48000; ptime=20; minptime=20; maxptime=20;
> samplerate=48000
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
> m=video 29716 RTP/AVP 103
> a=rtpmap:103 VP8/90000
>
> 2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:40
> sofia/internal/00294539100 Standard INIT
> 2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:48
> (sofia/internal/00294539100) State Change CS_INIT -> CS_ROUTING
> 2015-07-15 10:09:19.375204 [DEBUG] switch_core_session.c:1396 Send
> signal sofia/internal/00294539100 [BREAK]
> 2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:512
> (sofia/internal/00294539100) State INIT going to sleep
> 2015-07-15 10:09:19.375204 [DEBUG] switch_core_session.c:1061 Send
> signal sofia/internal/00294539100 [BREAK]
> 2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:472
> (sofia/internal/00294539100) Running State Change CS_ROUTING
> 2015-07-15 10:09:19.375204 [DEBUG] sofia.c:6627 Channel
> sofia/internal/00294539100 entering state [calling][0]
> 2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:528
> (sofia/internal/00294539100) State ROUTING
> 2015-07-15 10:09:19.375204 [DEBUG] mod_sofia.c:123
> sofia/internal/00294539100 SOFIA ROUTING
> 2015-07-15 10:09:19.375204 [DEBUG] switch_ivr_originate.c:67
> (sofia/internal/00294539100) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2015-07-15 10:09:19.375204 [DEBUG] switch_core_session.c:1396 Send
> signal sofia/internal/00294539100 [BREAK]
> 2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:528
> (sofia/internal/00294539100) State ROUTING going to sleep
> 2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:472
> (sofia/internal/00294539100) Running State Change CS_CONSUME_MEDIA
> 2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:547
> (sofia/internal/00294539100) State CONSUME_MEDIA
> 2015-07-15 10:09:19.375204 [DEBUG] switch_core_state_machine.c:547
> (sofia/internal/00294539100) State CONSUME_MEDIA going to sleep
> 2015-07-15 10:09:19.455217 [DEBUG] switch_core_session.c:1061 Send
> signal sofia/internal/00294539100 [BREAK]
> 2015-07-15 10:09:19.455217 [DEBUG] switch_core_session.c:1061 Send
> signal sofia/internal/00294539100 [BREAK]
> 2015-07-15 10:09:19.455217 [DEBUG] sofia.c:6627 Channel
> sofia/internal/00294539100 entering state [proceeding][180]
> 2015-07-15 10:09:19.455217 [NOTICE] sofia.c:6729 Ring-Ready
> sofia/internal/00294539100!
> 2015-07-15 10:09:19.455217 [DEBUG] switch_channel.c:3297
> (sofia/internal/00294539100) Callstate Change DOWN -> RINGING
> 2015-07-15 10:09:22.075203 [DEBUG] switch_core_session.c:1061 Send
> signal sofia/internal/00294539100 [BREAK]
> 2015-07-15 10:09:22.075203 [DEBUG] switch_core_session.c:1061 Send
> signal sofia/internal/00294539100 [BREAK]
> 2015-07-15 10:09:22.075203 [DEBUG] sofia.c:6627 Channel
> sofia/internal/00294539100 entering state [completing][200]
> 2015-07-15 10:09:22.075203 [DEBUG] sofia.c:6637 Remote SDP:
> v=0
> o=floxent 3645929363 1 IN IP4 192.168.223.220
> s=sflphone
> c=IN IP4 192.168.223.220
> t=0 0
> m=audio 28878 RTP/AVP 9 101
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=rtcp:28879 IN IP4 192.168.223.220
> m=video 0 RTP/AVP 103
>
> 2015-07-15 10:09:22.075203 [DEBUG] switch_core_session.c:1061 Send
> signal sofia/internal/00294539100 [BREAK]
> 2015-07-15 10:09:22.075203 [DEBUG] switch_core_session.c:1061 Send
> signal sofia/internal/00294539100 [BREAK]
> 2015-07-15 10:09:22.075203 [DEBUG] sofia.c:6627 Channel
> sofia/internal/00294539100 entering state [ready][200]
> 2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:3672 Audio
> Codec Compare [G722:9:8000:20:64000:1]/[opus:116:48000:20:0:1]
> 2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:3672 Audio
> Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1]
> 2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:3727 Audio
> Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match
> 2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:3672 Audio
> Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
> 2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:3672 Audio
> Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
> 2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:3672 Audio
> Codec Compare [G722:9:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
> 2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:3588 Set
> telephone-event payload to 101
> 2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:2507 Set Codec
> sofia/internal/00294539100 G722/8000 20 ms 160 samples 64000 bits 1
> channels
> 2015-07-15 10:09:22.075203 [DEBUG] switch_core_codec.c:111
> sofia/internal/00294539100 Original read codec set to G722:9
> 2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:3926 Set 2833
> dtmf send payload to 101
> 2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:5171 AUDIO RTP
> [sofia/internal/00294539100] 194.87.7.17 port 22232 -> 192.168.223.220
> port 28878 codec: 9 ms: 20
> 2015-07-15 10:09:22.075203 [DEBUG] switch_rtp.c:3569 Starting timer
> [soft] 160 bytes per 20ms
> 2015-07-15 10:09:22.075203 [INFO] switch_core_media.c:5388 Activating
> RTCP PORT 28879
> 2015-07-15 10:09:22.075203 [DEBUG] switch_rtp.c:3919 RTCP send rate
> is: 5000 and packet rate is: 20000 Remote Port: 28879
> 2015-07-15 10:09:22.075203 [DEBUG] switch_rtp.c:2349 Setting RTCP
> remote addr to 192.168.223.220:28879
> 2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:5469 Set 2833
> dtmf send payload to 101
> 2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:5475 Set 2833
> dtmf receive payload to 101
> 2015-07-15 10:09:22.075203 [DEBUG] switch_core_media.c:5497
> sofia/internal/00294539100 Set rtp dtmf delay to 40
> *2015-07-15 10:09:22.075203 [NOTICE] sofia.c:7488 Channel
> [sofia/internal/00294539100] has been answered*
> 2015-07-15 10:09:22.075203 [DEBUG] switch_channel.c:3711
> (sofia/internal/00294539100) Callstate Change RINGING -> ACTIVE
> 2015-07-15 10:09:22.095205 [DEBUG] switch_ivr_originate.c:3577
> Originate Resulted in Success: [sofia/internal/00294539100]
> 2015-07-15 10:09:22.095205 [INFO] switch_channel.c:3082
> sofia/internal/00294539100 Flipping CID from "" <0000000000> to
> "Outbound Call" <00294539100%>
> 2015-07-15 10:09:22.095205 [DEBUG] mod_commands.c:4469
> (sofia/internal/00294539100) State Change CS_CONSUME_MEDIA -> CS_EXECUTE
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1396 Send
> signal sofia/internal/00294539100 [BREAK]
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:472
> (sofia/internal/00294539100) Running State Change CS_EXECUTE
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:535
> (sofia/internal/00294539100) State EXECUTE
> 2015-07-15 10:09:22.095205 [DEBUG] mod_sofia.c:178
> sofia/internal/00294539100 SOFIA EXECUTE
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:258
> sofia/internal/00294539100 Standard EXECUTE
> EXECUTE sofia/internal/00294539100
> bridge(sofia/gateway/voip_provider/919262417024 at voip_provider)
> 2015-07-15 10:09:22.095205 [DEBUG] switch_channel.c:1786
> (sofia/internal/00294539100) Callstate Change ACTIVE -> RING_WAIT
> 2015-07-15 10:09:22.095205 [DEBUG] switch_ivr_originate.c:2100 Parsing
> global variables
> 2015-07-15 10:09:22.095205 [NOTICE] switch_channel.c:1075 New Channel
> sofia/external/919262417024 at voip_provider
> [32799f05-3018-4a48-bea4-1d3cc3700835]
> 2015-07-15 10:09:22.095205 [DEBUG] mod_sofia.c:4701
> (sofia/external/919262417024 at voip_provider) State Change CS_NEW -> CS_INIT
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1396 Send
> signal sofia/external/919262417024 at voip_provider [BREAK]
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:472
> (sofia/external/919262417024 at voip_provider) Running State Change CS_INIT
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:512
> (sofia/external/919262417024 at voip_provider) State INIT
> 2015-07-15 10:09:22.095205 [DEBUG] mod_sofia.c:87
> sofia/external/919262417024 at voip_provider SOFIA INIT
> 2015-07-15 10:09:22.095205 [DEBUG] sofia_glue.c:1236
> sofia/external/919262417024 at voip_provider sending invite version:
> 1.5.15b git acdb1ca 2015-05-17 18:45:52Z 32bit
> Local SDP:
> v=0
> o=FreeSWITCH 1436921310 1436921311 IN IP4 194.87.7.17
> s=FreeSWITCH
> c=IN IP4 194.87.7.17
> t=0 0
> m=audio 19252 RTP/AVP 9 0 8 3 101 13
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
> m=video 31754 RTP/AVP 103
> a=rtpmap:103 VP8/90000
>
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:40
> sofia/external/919262417024 at voip_provider Standard INIT
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:48
> (sofia/external/919262417024 at voip_provider) State Change CS_INIT ->
> CS_ROUTING
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1396 Send
> signal sofia/external/919262417024 at voip_provider [BREAK]
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:512
> (sofia/external/919262417024 at voip_provider) State INIT going to sleep
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1061 Send
> signal sofia/external/919262417024 at voip_provider [BREAK]
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1061 Send
> signal sofia/external/919262417024 at voip_provider [BREAK]
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1061 Send
> signal sofia/external/919262417024 at voip_provider [BREAK]
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:472
> (sofia/external/919262417024 at voip_provider) Running State Change
> CS_ROUTING
> 2015-07-15 10:09:22.095205 [DEBUG] sofia.c:6627 Channel
> sofia/external/919262417024 at voip_provider entering state [terminated][900]
> *2015-07-15 10:09:22.095205 [NOTICE] sofia.c:7543 Hangup
> sofia/external/919262417024 at voip_provider [CS_ROUTING]
> [NORMAL_UNSPECIFIED]*
> 2015-07-15 10:09:22.095205 [DEBUG] switch_ivr_originate.c:3720
> Originate Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED]
> 2015-07-15 10:09:22.095205 [DEBUG] switch_channel.c:3242 Send signal
> sofia/external/919262417024 at voip_provider [KILL]
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1396 Send
> signal sofia/external/919262417024 at voip_provider [BREAK]
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:528
> (sofia/external/919262417024 at voip_provider) State ROUTING
> 2015-07-15 10:09:22.095205 [DEBUG] mod_sofia.c:123
> sofia/external/919262417024 at voip_provider SOFIA ROUTING
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:528
> (sofia/external/919262417024 at voip_provider) State ROUTING going to sleep
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:472
> (sofia/external/919262417024 at voip_provider) Running State Change CS_HANGUP
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:735
> (sofia/external/919262417024 at voip_provider) Callstate Change DOWN ->
> HANGUP
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:737
> (sofia/external/919262417024 at voip_provider) State HANGUP
> 2015-07-15 10:09:22.095205 [DEBUG] mod_sofia.c:413 Channel
> sofia/external/919262417024 at voip_provider hanging up, cause:
> NORMAL_UNSPECIFIED
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:60
> sofia/external/919262417024 at voip_provider Standard HANGUP, cause:
> NORMAL_UNSPECIFIED
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:737
> (sofia/external/919262417024 at voip_provider) State HANGUP going to sleep
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:504
> (sofia/external/919262417024 at voip_provider) State Change CS_HANGUP ->
> CS_REPORTING
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1396 Send
> signal sofia/external/919262417024 at voip_provider [BREAK]
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:472
> (sofia/external/919262417024 at voip_provider) Running State Change
> CS_REPORTING
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:823
> (sofia/external/919262417024 at voip_provider) State REPORTING
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:104
> sofia/external/919262417024 at voip_provider Standard REPORTING, cause:
> NORMAL_UNSPECIFIED
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:823
> (sofia/external/919262417024 at voip_provider) State REPORTING going to sleep
> 2015-07-15 10:09:22.095205 [DEBUG] switch_channel.c:1999
> (sofia/internal/00294539100) Callstate Change RING_WAIT -> ACTIVE
> 2015-07-15 10:09:22.095205 [INFO] mod_dptools.c:3268 Originate
> Failed.  Cause: NORMAL_UNSPECIFIED
> 2015-07-15 10:09:22.095205 [NOTICE] switch_channel.c:4747 Hangup
> sofia/internal/00294539100 [CS_EXECUTE] [NORMAL_UNSPECIFIED]
> 2015-07-15 10:09:22.095205 [DEBUG] switch_channel.c:3242 Send signal
> sofia/internal/00294539100 [KILL]
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1396 Send
> signal sofia/internal/00294539100 [BREAK]
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_state_machine.c:498
> (sofia/external/919262417024 at voip_provider) State Change CS_REPORTING
> -> CS_DESTROY
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1396 Send
> signal sofia/external/919262417024 at voip_provider [BREAK]
> 2015-07-15 10:09:22.095205 [DEBUG] switch_core_session.c:1623 Session
> 202 (sofia/external/919262417024 at voip_provider) Locked, Waiting on
> external entities
> 2015-07-15 10:09:22.095205 [NOTICE] switch_core_session.c:1641 Session
> 202 (sofia/external/919262417024 at voip_provider) Ended
> 2015-07-15 10:09:22.095205 [NOTICE] switch_core_session.c:1645 Close
> Channel sofia/external/919262417024 at voip_provider [CS_DESTROY]
>
>
>
> _________________________________________________________________________
> 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


-- 
With kind regards
Peter Steinbach 

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

-------------- next part --------------
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From dm at dwide.com  Wed Jul 15 11:27:47 2015
From: dm at dwide.com (Dmitry Mordovin)
Date: Wed, 15 Jul 2015 11:27:47 +0400
Subject: [Freeswitch-users] Call thru external gate released before
 INVITE (NORMAL_UNSPECIFIED)
In-Reply-To: <CAHtxdDeBST6kmJeEn=VN_TzDdCv1rppT9g5qxX4bcyEq-Hcn+g@mail.gmail.com>
References: <55A5FC8E.1070901@dwide.com>
	<CAHtxdDeBST6kmJeEn=VN_TzDdCv1rppT9g5qxX4bcyEq-Hcn+g@mail.gmail.com>
Message-ID: <55A60B73.7070506@dwide.com>

Sergey, Peter,

*originate 
{ignore_early_media=true}sofia/gateway/sipnet.ru/**919262417024 at sipnet.ru &bridge(sofia/internal/00294539100%)*

If i change like this - call established fine, but fail if first call to 
internal 00294539100

New trace


2015-07-15 11:17:57.115208 [NOTICE] switch_channel.c:1075 New Channel 
sofia/external/919262417024 at sipnet.ru [4c3ee575-4f0f-4ebd-b4ec-0e5357469690]
2015-07-15 11:17:57.115208 [DEBUG] mod_sofia.c:4701 
(sofia/external/919262417024 at sipnet.ru) State Change CS_NEW -> CS_INIT
2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1396 Send 
signal sofia/external/919262417024 at sipnet.ru [BREAK]
2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:472 
(sofia/external/919262417024 at sipnet.ru) Running State Change CS_INIT
2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:512 
(sofia/external/919262417024 at sipnet.ru) State INIT
2015-07-15 11:17:57.115208 [DEBUG] mod_sofia.c:87 
sofia/external/919262417024 at sipnet.ru SOFIA INIT
nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
2015-07-15 11:17:57.115208 [DEBUG] sofia_glue.c:1236 
sofia/external/919262417024 at sipnet.ru sending invite version: 1.5.15b 
git acdb1ca 2015-05-17 18:45:52Z 32bit
Local SDP:
v=0
o=FreeSWITCH 1436923599 1436923600 IN IP4 194.87.7.17
s=FreeSWITCH
c=IN IP4 194.87.7.17
t=0 0
m=audio 21078 RTP/AVP 9 0 8 3 101 13
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 20298 RTP/AVP 103
a=rtpmap:103 VP8/90000

nua.c:633 nua_invite() nua: nua_invite: entering
nua_stack.c:529 nua_signal() nua(0xb6d54b10): sent signal r_invite
2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:40 
sofia/external/919262417024 at sipnet.ru Standard INIT
nua_stack.c:569 nua_stack_signal() nua(0xb6d54b10): recv signal r_invite
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:48 
(sofia/external/919262417024 at sipnet.ru) State Change CS_INIT -> CS_ROUTING
soa.c:280 soa_clone() soa_clone(static::0x8a45980, 0x8a547e8, 
0xb6d54b10) called
2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1396 Send 
signal sofia/external/919262417024 at sipnet.ru [BREAK]
soa.c:403 soa_set_params() soa_set_params(static::0x8a4cbb0, ...) called
soa.c:403 soa_set_params() soa_set_params(static::0x8a4cbb0, ...) called
2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:512 
(sofia/external/919262417024 at sipnet.ru) State INIT going to sleep
soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x8a4cbb0, (nil), 
0xb6d831a4, -1) called
soa.c:890 soa_set_capability_sdp() 
soa_set_capability_sdp(static::0x8a4cbb0, (nil), 0xb6d831a4, -1) called
nua_stack.c:271 nua_stack_event() nua(0xb6d54b10): event r_invite 900 
Internal error at nua_client.c:552
nua_session.c:4139 signal_call_state_change() nua(0xb6d54b10): call 
state changed: init -> terminated
nua_stack.c:271 nua_stack_event() nua(0xb6d54b10): event i_state 900 
Internal error at nua_client.c:552
nua_stack.c:271 nua_stack_event() nua(0xb6d54b10): event i_terminated 
900 Internal error at nua_client.c:552
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1061 Send 
signal sofia/external/919262417024 at sipnet.ru [BREAK]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1061 Send 
signal sofia/external/919262417024 at sipnet.ru [BREAK]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:472 
(sofia/external/919262417024 at sipnet.ru) Running State Change CS_ROUTING
2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1061 Send 
signal sofia/external/919262417024 at sipnet.ru [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-07-15 11:17:57.115208 [DEBUG] sofia.c:6627 Channel 
sofia/external/919262417024 at sipnet.ru entering state [terminated][900]
2015-07-15 11:17:57.115208 [NOTICE] sofia.c:7543 Hangup 
sofia/external/919262417024 at sipnet.ru [CS_ROUTING] [NORMAL_UNSPECIFIED]




On 07/15/2015 11:14 AM, Sergey Safarov wrote:
> What is "voip_provider"?
> Please enable
> 1) sofia global siptrace on
> 2) sofia loglevel all 9
>
>
>
> _________________________________________________________________________
> 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 omortimer at gmail.com  Wed Jul 15 12:49:57 2015
From: omortimer at gmail.com (Oz Mortimer)
Date: Wed, 15 Jul 2015 09:49:57 +0100
Subject: [Freeswitch-users] Call thru external gate released before
	INVITE (NORMAL_UNSPECIFIED)
In-Reply-To: <55A60B73.7070506@dwide.com>
References: <55A5FC8E.1070901@dwide.com>
	<CAHtxdDeBST6kmJeEn=VN_TzDdCv1rppT9g5qxX4bcyEq-Hcn+g@mail.gmail.com>
	<55A60B73.7070506@dwide.com>
Message-ID: <1B5A6695-D5C9-49F3-A5A4-0384289586A2@gmail.com>

Should be 

originate {ignore_early_media=true}sofia/gateway/sipnet.ru/919262417024 <mailto:919262417024 at sipnet.ru> &bridge(sofia/gateway/sipnet.ru/00294539100)

You seem to be doubling up the signet.ru by having as a gateway then  again @sipnet.ru <http://sipnet.ru/> - either is valid but both at the same time is wrong.
After the destination number you had % - not sure why, but wouldn?t have thought it should be there.
Additionally you are using 2 different number formats (with and without 00 prefix) - I would check with provider what format you should be sending.



> On 15 Jul 2015, at 08:27, Dmitry Mordovin <dm at dwide.com> wrote:
> 
> Sergey, Peter,
> 
> originate {ignore_early_media=true}sofia/gateway/sipnet.ru/919262417024 at sipnet.ru <mailto:919262417024 at sipnet.ru> &bridge(sofia/internal/00294539100%)
> 
> If i change like this - call established fine, but fail if first call to internal 00294539100
> 
> New trace
> 
> 
> 2015-07-15 11:17:57.115208 [NOTICE] switch_channel.c:1075 New Channel sofia/external/919262417024 at sipnet.ru <mailto:sofia/external/919262417024 at sipnet.ru> [4c3ee575-4f0f-4ebd-b4ec-0e5357469690]
> 2015-07-15 11:17:57.115208 [DEBUG] mod_sofia.c:4701 (sofia/external/919262417024 at sipnet.ru <mailto:sofia/external/919262417024 at sipnet.ru>) State Change CS_NEW -> CS_INIT
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1396 Send signal sofia/external/919262417024 at sipnet.ru <mailto:sofia/external/919262417024 at sipnet.ru> [BREAK]
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:472 (sofia/external/919262417024 at sipnet.ru <mailto:sofia/external/919262417024 at sipnet.ru>) Running State Change CS_INIT
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:512 (sofia/external/919262417024 at sipnet.ru <mailto:sofia/external/919262417024 at sipnet.ru>) State INIT
> 2015-07-15 11:17:57.115208 [DEBUG] mod_sofia.c:87 sofia/external/919262417024 at sipnet.ru <mailto:sofia/external/919262417024 at sipnet.ru> SOFIA INIT
> nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
> 2015-07-15 11:17:57.115208 [DEBUG] sofia_glue.c:1236 sofia/external/919262417024 at sipnet.ru <mailto:sofia/external/919262417024 at sipnet.ru> sending invite version: 1.5.15b git acdb1ca 2015-05-17 18:45:52Z 32bit
> Local SDP:
> v=0
> o=FreeSWITCH 1436923599 1436923600 IN IP4 194.87.7.17
> s=FreeSWITCH
> c=IN IP4 194.87.7.17
> t=0 0
> m=audio 21078 RTP/AVP 9 0 8 3 101 13
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
> m=video 20298 RTP/AVP 103
> a=rtpmap:103 VP8/90000
> 
> nua.c:633 nua_invite() nua: nua_invite: entering
> nua_stack.c:529 nua_signal() nua(0xb6d54b10): sent signal r_invite
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:40 sofia/external/919262417024 at sipnet.ru <mailto:sofia/external/919262417024 at sipnet.ru> Standard INIT
> nua_stack.c:569 nua_stack_signal() nua(0xb6d54b10): recv signal r_invite
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:48 (sofia/external/919262417024 at sipnet.ru <mailto:sofia/external/919262417024 at sipnet.ru>) State Change CS_INIT -> CS_ROUTING
> soa.c:280 soa_clone() soa_clone(static::0x8a45980, 0x8a547e8, 0xb6d54b10) called
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1396 Send signal sofia/external/919262417024 at sipnet.ru <mailto:sofia/external/919262417024 at sipnet.ru> [BREAK]
> soa.c:403 soa_set_params() soa_set_params(static::0x8a4cbb0, ...) called
> soa.c:403 soa_set_params() soa_set_params(static::0x8a4cbb0, ...) called
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:512 (sofia/external/919262417024 at sipnet.ru <mailto:sofia/external/919262417024 at sipnet.ru>) State INIT going to sleep
> soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x8a4cbb0, (nil), 0xb6d831a4, -1) called
> soa.c:890 soa_set_capability_sdp() soa_set_capability_sdp(static::0x8a4cbb0, (nil), 0xb6d831a4, -1) called
> nua_stack.c:271 nua_stack_event() nua(0xb6d54b10): event r_invite 900 Internal error at nua_client.c:552
> nua_session.c:4139 signal_call_state_change() nua(0xb6d54b10): call state changed: init -> terminated
> nua_stack.c:271 nua_stack_event() nua(0xb6d54b10): event i_state 900 Internal error at nua_client.c:552
> nua_stack.c:271 nua_stack_event() nua(0xb6d54b10): event i_terminated 900 Internal error at nua_client.c:552
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1061 Send signal sofia/external/919262417024 at sipnet.ru <mailto:sofia/external/919262417024 at sipnet.ru> [BREAK]
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1061 Send signal sofia/external/919262417024 at sipnet.ru <mailto:sofia/external/919262417024 at sipnet.ru> [BREAK]
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:472 (sofia/external/919262417024 at sipnet.ru <mailto:sofia/external/919262417024 at sipnet.ru>) Running State Change CS_ROUTING
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1061 Send signal sofia/external/919262417024 at sipnet.ru <mailto:sofia/external/919262417024 at sipnet.ru> [BREAK]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-07-15 11:17:57.115208 [DEBUG] sofia.c:6627 Channel sofia/external/919262417024 at sipnet.ru <mailto:sofia/external/919262417024 at sipnet.ru> entering state [terminated][900]
> 2015-07-15 11:17:57.115208 [NOTICE] sofia.c:7543 Hangup sofia/external/919262417024 at sipnet.ru <mailto:sofia/external/919262417024 at sipnet.ru> [CS_ROUTING] [NORMAL_UNSPECIFIED]
> 
> 
> 
> 
> On 07/15/2015 11:14 AM, Sergey Safarov wrote:
>> What is "voip_provider"?
>> Please enable
>> 1) sofia global siptrace on
>> 2) sofia loglevel all 9
>> 
>> 
>> 
>> _________________________________________________________________________
>> 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
> 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 yadenis at seznam.cz  Wed Jul 15 13:21:13 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Wed, 15 Jul 2015 11:21:13 +0200
Subject: [Freeswitch-users] Run the command after hanging up
In-Reply-To: <CAEJMVkCWreG-bgYYV30Y6X58uzN7bn8DGwrt7rnVnTcem87vGw@mail.gmail.com>
References: <09f801d0bdc7$e2c30e80$a8492b80$@inovax.com.br> 
	<CAEJMVkAzWrOUdxn1t=26kOYR+d_6GETbYJUr6A1pw2=m-Nuwvw@mail.gmail.com>
	<0a9b01d0be32$5a93c3e0$0fbb4ba0$@inovax.com.br>
	<CAEJMVkCWreG-bgYYV30Y6X58uzN7bn8DGwrt7rnVnTcem87vGw@mail.gmail.com>
Message-ID: <1096950257.20150715112113@seznam.cz>

Hi All,

I need after hanging up, to execute a command. Something like "exec_after_bridge" or "exec_after_answer" but after hanging up. I know that there "api_hangup_hook", but for scripts. And I do not need.

So the question is, how do I run a command after nangup in dialplan?


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From alan.scales74 at gmail.com  Wed Jul 15 13:30:59 2015
From: alan.scales74 at gmail.com (Alan Scales)
Date: Wed, 15 Jul 2015 12:30:59 +0300
Subject: [Freeswitch-users] FreeSWITCH and Sangoma D100 / D150 / D500
Message-ID: <55A62853.20100@gmail.com>

Hi all,

Could anyone with some mod_sangoma_codec experience share some general 
knowledge on the requirements of the different pieces of hardware 
mentioned in the title?
In other words, are there any differences in the FreeSWITCH 
configuration required when dealing with each of them? (e.g. D100 and 
D500 have no RJ45 Ethernet interfaces, how does this impact usage?)

Thank you in advance for any hints or tips!

Yours truly,

---
Alan Scales


From steveayre at gmail.com  Wed Jul 15 13:34:02 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 15 Jul 2015 10:34:02 +0100
Subject: [Freeswitch-users] Run the command after hanging up
In-Reply-To: <1096950257.20150715112113@seznam.cz>
References: <09f801d0bdc7$e2c30e80$a8492b80$@inovax.com.br>
	<CAEJMVkAzWrOUdxn1t=26kOYR+d_6GETbYJUr6A1pw2=m-Nuwvw@mail.gmail.com>
	<0a9b01d0be32$5a93c3e0$0fbb4ba0$@inovax.com.br>
	<CAEJMVkCWreG-bgYYV30Y6X58uzN7bn8DGwrt7rnVnTcem87vGw@mail.gmail.com>
	<1096950257.20150715112113@seznam.cz>
Message-ID: <CAFiqYukiMHMM=0Oy81YFR18bOF03JTG5VkKduouycbheoebhPg@mail.gmail.com>

The api_hangup_hook variable can be used to run any api command, not just
scripts. However you can't execute a dialplan application as the session
has hung up.

On 15 July 2015 at 10:21, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Hi All,
>
> I need after hanging up, to execute a command. Something like
> "exec_after_bridge" or "exec_after_answer" but after hanging up. I know
> that there "api_hangup_hook", but for scripts. And I do not need.
>
> So the question is, how do I run a command after nangup in dialplan?
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 *
>
> _________________________________________________________________________
> 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 dm at dwide.com  Wed Jul 15 13:47:03 2015
From: dm at dwide.com (Dmitry Mordovin)
Date: Wed, 15 Jul 2015 13:47:03 +0400
Subject: [Freeswitch-users] Call thru external gate released before
 INVITE (NORMAL_UNSPECIFIED)
In-Reply-To: <1B5A6695-D5C9-49F3-A5A4-0384289586A2@gmail.com>
References: <55A5FC8E.1070901@dwide.com>	<CAHtxdDeBST6kmJeEn=VN_TzDdCv1rppT9g5qxX4bcyEq-Hcn+g@mail.gmail.com>	<55A60B73.7070506@dwide.com>
	<1B5A6695-D5C9-49F3-A5A4-0384289586A2@gmail.com>
Message-ID: <55A62C17.9020300@dwide.com>

00294539100 - internal user, % used when registered user dialed

Second sipnet.ru crop

Same problem....
If I dial both external numbers, or first external (second internal 
user) - they bridged fine.

Problem occured when internal user dialed first, and bridge to external 
number.


On 07/15/2015 12:49 PM, Oz Mortimer wrote:
> Should be
> *
> *
> *originate {ignore_early_media=true}sofia/gateway/sipnet.ru/ 
> <http://sipnet.ru/>**919262417024 &bridge(sofia/gateway**/sipnet.ru/ 
> <http://sipnet.ru/>**00294539100)*
>
> You seem to be doubling up the signet.ru <http://signet.ru> by having 
> as a gateway then  again @*sipnet.ru <http://sipnet.ru> - *either is 
> valid but both at the same time is wrong.
> After the destination number you had % - not sure why, but wouldn?t 
> have thought it should be there.
> Additionally you are using 2 different number formats (with and 
> without 00 prefix) - I would check with provider what format you 
> should be sending.
>
>
>
>> On 15 Jul 2015, at 08:27, Dmitry Mordovin <dm at dwide.com 
>> <mailto:dm at dwide.com>> wrote:
>>
>> Sergey, Peter,
>>
>> *originate {ignore_early_media=true}sofia/gateway/sipnet.ru/ 
>> <http://sipnet.ru/>**919262417024 at sipnet.ru 
>> &bridge(sofia/internal/00294539100%)*
>>
>> If i change like this - call established fine, but fail if first call 
>> to internal 00294539100
>>
>> New trace
>>
>>
>> 2015-07-15 11:17:57.115208 [NOTICE] switch_channel.c:1075 New Channel 
>> sofia/external/919262417024 at sipnet.ru 
>> [4c3ee575-4f0f-4ebd-b4ec-0e5357469690]
>> 2015-07-15 11:17:57.115208 [DEBUG] mod_sofia.c:4701 
>> (sofia/external/919262417024 at sipnet.ru) State Change CS_NEW -> CS_INIT
>> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1396 Send 
>> signal sofia/external/919262417024 at sipnet.ru [BREAK]
>> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:472 
>> (sofia/external/919262417024 at sipnet.ru) Running State Change CS_INIT
>> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:512 
>> (sofia/external/919262417024 at sipnet.ru) State INIT
>> 2015-07-15 11:17:57.115208 [DEBUG] mod_sofia.c:87 
>> sofia/external/919262417024 at sipnet.ru SOFIA INIT
>> nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
>> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
>> 2015-07-15 11:17:57.115208 [DEBUG] sofia_glue.c:1236 
>> sofia/external/919262417024 at sipnet.ru sending invite version: 1.5.15b 
>> git acdb1ca 2015-05-17 18:45:52Z 32bit
>> Local SDP:
>> v=0
>> o=FreeSWITCH 1436923599 1436923600 IN IP4 194.87.7.17
>> s=FreeSWITCH
>> c=IN IP4 194.87.7.17
>> t=0 0
>> m=audio 21078 RTP/AVP 9 0 8 3 101 13
>> a=rtpmap:9 G722/8000
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> a=sendrecv
>> m=video 20298 RTP/AVP 103
>> a=rtpmap:103 VP8/90000
>>
>> nua.c:633 nua_invite() nua: nua_invite: entering
>> nua_stack.c:529 nua_signal() nua(0xb6d54b10): sent signal r_invite
>> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:40 
>> sofia/external/919262417024 at sipnet.ru Standard INIT
>> nua_stack.c:569 nua_stack_signal() nua(0xb6d54b10): recv signal r_invite
>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: 
>> entering
>> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:48 
>> (sofia/external/919262417024 at sipnet.ru) State Change CS_INIT -> 
>> CS_ROUTING
>> soa.c:280 soa_clone() soa_clone(static::0x8a45980, 0x8a547e8, 
>> 0xb6d54b10) called
>> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1396 Send 
>> signal sofia/external/919262417024 at sipnet.ru [BREAK]
>> soa.c:403 soa_set_params() soa_set_params(static::0x8a4cbb0, ...) called
>> soa.c:403 soa_set_params() soa_set_params(static::0x8a4cbb0, ...) called
>> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:512 
>> (sofia/external/919262417024 at sipnet.ru) State INIT going to sleep
>> soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x8a4cbb0, 
>> (nil), 0xb6d831a4, -1) called
>> soa.c:890 soa_set_capability_sdp() 
>> soa_set_capability_sdp(static::0x8a4cbb0, (nil), 0xb6d831a4, -1) called
>> nua_stack.c:271 nua_stack_event() nua(0xb6d54b10): event r_invite 900 
>> Internal error at nua_client.c:552
>> nua_session.c:4139 signal_call_state_change() nua(0xb6d54b10): call 
>> state changed: init -> terminated
>> nua_stack.c:271 nua_stack_event() nua(0xb6d54b10): event i_state 900 
>> Internal error at nua_client.c:552
>> nua_stack.c:271 nua_stack_event() nua(0xb6d54b10): event i_terminated 
>> 900 Internal error at nua_client.c:552
>> nua_stack.c:359 nua_application_event() nua: nua_application_event: 
>> entering
>> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1061 Send 
>> signal sofia/external/919262417024 at sipnet.ru [BREAK]
>> nua_stack.c:359 nua_application_event() nua: nua_application_event: 
>> entering
>> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1061 Send 
>> signal sofia/external/919262417024 at sipnet.ru [BREAK]
>> nua_stack.c:359 nua_application_event() nua: nua_application_event: 
>> entering
>> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:472 
>> (sofia/external/919262417024 at sipnet.ru) Running State Change CS_ROUTING
>> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1061 Send 
>> signal sofia/external/919262417024 at sipnet.ru [BREAK]
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> 2015-07-15 11:17:57.115208 [DEBUG] sofia.c:6627 Channel 
>> sofia/external/919262417024 at sipnet.ru entering state [terminated][900]
>> 2015-07-15 11:17:57.115208 [NOTICE] sofia.c:7543 Hangup 
>> sofia/external/919262417024 at sipnet.ru [CS_ROUTING] [NORMAL_UNSPECIFIED]
>>
>>
>>
>>
>> On 07/15/2015 11:14 AM, Sergey Safarov wrote:
>>> What is "voip_provider"?
>>> Please enable
>>> 1) sofia global siptrace on
>>> 2) sofia loglevel all 9
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> 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 <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
> 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 yadenis at seznam.cz  Wed Jul 15 13:59:43 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Wed, 15 Jul 2015 11:59:43 +0200
Subject: [Freeswitch-users] Run the command after hanging up
In-Reply-To: <CAFiqYukiMHMM=0Oy81YFR18bOF03JTG5VkKduouycbheoebhPg@mail.gmail.com>
References: <09f801d0bdc7$e2c30e80$a8492b80$@inovax.com.br> 
	<CAEJMVkAzWrOUdxn1t=26kOYR+d_6GETbYJUr6A1pw2=m-Nuwvw@mail.gmail.com>
	<0a9b01d0be32$5a93c3e0$0fbb4ba0$@inovax.com.br>
	<CAEJMVkCWreG-bgYYV30Y6X58uzN7bn8DGwrt7rnVnTcem87vGw@mail.gmail.com>
	<1096950257.20150715112113@seznam.cz>
	<CAFiqYukiMHMM=0Oy81YFR18bOF03JTG5VkKduouycbheoebhPg@mail.gmail.com>
Message-ID: <521212797.20150715115943@seznam.cz>

Dobr? den,

Yeah. Clear. I did not know that "api_hangup_hook" It can do something else besides scripts.

In https://wiki.freeswitch.org/wiki/Variable_api_hangup_hook  an example of the script.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

st?eda 15. ?ervence 2015, 11:34:02, napsal jste:


The api_hangup_hook variable can be used to run any api command, not just scripts. However you can't execute a dialplan application as the session has hung up.

On 15 July 2015 at 10:21, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi All,

I need after hanging up, to execute a command. Something like "exec_after_bridge" or "exec_after_answer" but after hanging up. I know that there "api_hangup_hook", but for scripts. And I do not need.

So the question is, how do I run a command after nangup in dialplan?


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

_________________________________________________________________________
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
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From victor.medina at cibersys.com  Wed Jul 15 16:39:23 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Wed, 15 Jul 2015 08:09:23 -0430
Subject: [Freeswitch-users] About the test framework
Message-ID: <CACXsapNjQKUFPaaX=KLMGS7Vt=7jheX1r8=fe4RUym4QmDHxjw@mail.gmail.com>

Hi guys!

I did my best yesterday to be on the conference but it was impossible due
to my connection..

My two cents.

Tests should include full calls between the differents endpoints. Sip/Sip,
Sip/WebRTC, WebRTC/Sip, WebRTC/WebRTC, udo, tcp, tls, tls/srtp(in case of
sip)

Sample config for the test should be included to inspect and study

PjSip has a command line UA, that could be used to make calls, play some
sort of pre recorded audio using some scripting.


-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From achinthau at gmail.com  Wed Jul 15 16:30:00 2015
From: achinthau at gmail.com (Achintha)
Date: Wed, 15 Jul 2015 18:00:00 +0530
Subject: [Freeswitch-users] profile context is not overridden
In-Reply-To: <CAFiqYu=QoM-2XuZfEiD4OJ8G_EVKjo6WoPy_RrtRs3m57Vsexg@mail.gmail.com>
References: <CA+JwzxtSkHASHOeteOxRozH8fQwBCk2eYO_G36cmAb+43LwFGA@mail.gmail.com>
	<CAFiqYunofjieNLW0NLfHAku+Pfx=3eqj-SWvDYF1oVS2exVW0w@mail.gmail.com>
	<CA+Jwzxsp_sbP4ModfcuQ7qnVeq9d4UJ09zwofJdJBfFVVse1aA@mail.gmail.com>
	<CAFiqYu=QoM-2XuZfEiD4OJ8G_EVKjo6WoPy_RrtRs3m57Vsexg@mail.gmail.com>
Message-ID: <CA+JwzxtQL7_c6egL3FTpyoW8J7ih0A-CBvcuE-wUFS6eSkLfdQ@mail.gmail.com>

hi steven,

yes i was enable <param name="auth-calls" value="true"/> in my internal
profile
my internal profile as follows

<profile name="internal">
  <aliases>
  </aliases>
  <gateways>
    <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
  </gateways>
  <domains>
    <domain name="all" alias="true" parse="false"/>
  </domains>
  <settings>
    <param name="debug" value="0"/>
    <param name="sip-trace" value="no"/>
    <param name="sip-capture" value="no"/>
    <param name="watchdog-enabled" value="no"/>
    <param name="watchdog-step-timeout" value="30000"/>
    <param name="watchdog-event-timeout" value="30000"/>
    <param name="log-auth-failures" value="false"/>
    <param name="forward-unsolicited-mwi-notify" value="false"/>
    <param name="context" value="public"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="$${internal_sip_port}"/>
    <param name="dialplan" value="XML"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="rtp-timer-name" value="soft"/>
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="apply-nat-acl" value="nat.auto"/>
    <param name="apply-inbound-acl" value="domainx"/>
    <param name="local-network-acl" value="localnet.auto"/>
    <param name="record-path" value="$${recordings_dir}"/>
    <param name="record-template"
value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
    <param name="manage-presence" value="true"/>
    <param name="presence-hosts" value="$${domain},$${local_ip_v4}"/>
    <param name="presence-privacy" value="$${presence_privacy}"/>
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="tls" value="$${internal_ssl_enable}"/>
    <param name="tls-only" value="false"/>
    <param name="tls-bind-params" value="transport=tls"/>
    <param name="tls-sip-port" value="$${internal_tls_port}"/>
    <param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
    <param name="tls-passphrase" value=""/>
    <param name="tls-verify-date" value="true"/>
    <param name="tls-verify-policy" value="none"/>
    <param name="tls-verify-depth" value="2"/>
    <param name="tls-verify-in-subjects" value=""/>
    <param name="tls-version" value="$${sip_tls_version}"/>
    <param name="tls-ciphers" value="$${sip_tls_ciphers}"/>
    <param name="inbound-late-negotiation" value="true"/>
    <param name="inbound-zrtp-passthru" value="true"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="true"/>
    <param name="inbound-reg-force-matching-username" value="true"/>
    <param name="auth-all-packets" value="false"/>
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
    <param name="force-register-domain" value="$${domain}"/>
    <param name="force-subscription-domain" value="$${domain}"/>
    <param name="force-register-db-domain" value="$${domain}"/>
  </settings>
</profile>

On Tue, Jul 14, 2015 at 3:55 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Are you sure the call is getting authenticating as the user? Perhaps it's
> being processed as an unauthenticated user?
>
> On 14 July 2015 at 11:09, Achintha <achinthau at gmail.com> wrote:
>
>> hi steven,
>>
>> i'm unable to find user_context in CDR file.
>>
>> On Mon, Jul 13, 2015 at 8:03 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> First off verify the call is getting authenticated as the user. If you
>>> examine the XML CDR or the 'info' output do you see the user_context
>>> variable set?
>>>
>>> On 13 July 2015 at 13:38, Achintha <achinthau at gmail.com> wrote:
>>>
>>>> hi all,
>>>>
>>>>
>>>> We are having freeswitch testing setup with Internal and external
>>>> profiles with xml_curl for Directory management. In this setup profile
>>>> context is not overridden with user_context property.
>>>>
>>>> Kindly help me to solve this issue.
>>>>
>>>>
>>>>
>>>> Thanking you.
>>>>
>>>>
>>>> Achintha
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>>>
>>
>>
>>
>> --
>>
>> Achintha
>>
>> _________________________________________________________________________
>> 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
>



-- 
Regards..
Achintha
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From yadenis at seznam.cz  Wed Jul 15 16:43:07 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Wed, 15 Jul 2015 14:43:07 +0200
Subject: [Freeswitch-users] Reduce connection time when video calling
In-Reply-To: <55A62C17.9020300@dwide.com>
References: <55A5FC8E.1070901@dwide.com> 
	<CAHtxdDeBST6kmJeEn=VN_TzDdCv1rppT9g5qxX4bcyEq-Hcn+g@mail.gmail.com>
	<55A60B73.7070506@dwide.com>
	<1B5A6695-D5C9-49F3-A5A4-0384289586A2@gmail.com>
	<55A62C17.9020300@dwide.com>
Message-ID: <85386622.20150715144307@seznam.cz>

Hi all,

The question in the following. How it is possible to reduce the connection time with the customer video calls? If you do directly with "bypass_media = true", then both callers begin to see each other almost instantaneously. Through FreeSwitch this connection can take from 5 to 10 seconds. It can somehow reduce?

Thanks

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From govoiper at gmail.com  Wed Jul 15 18:22:27 2015
From: govoiper at gmail.com (SamyGo)
Date: Wed, 15 Jul 2015 10:22:27 -0400
Subject: [Freeswitch-users] Execute dialplan after valet_park
In-Reply-To: <CAO3hyzE0joUxdFeQM4oJ9K64qdLcG_4uock8v3AqHDWc6pSdaw@mail.gmail.com>
References: <CAJUJwtipc+yhpck+w4M4mYYPk0ZRa5NqWhytRChbsi4c6_nwdA@mail.gmail.com>
	<CAO3hyzE0joUxdFeQM4oJ9K64qdLcG_4uock8v3AqHDWc6pSdaw@mail.gmail.com>
Message-ID: <CAJUJwtjdBerXshWVJhQ0KJ-M_chiAaNiXgGaEU4hs4_nN3b8+Q@mail.gmail.com>

Thanks Chris,

That was my finding too, I just am not sure if I can monitor this event
from the main LUA script or need an independent lua script only for the
park "hangup" event monitoring. Any further help on this would be highly
appreciated.

I wonder why the api_hangup hook won't trigger on the parked channel,
would've been much easier. Piece of cake only if we could execute further
dialplan application after valet_parking.

BR,
Sammy
You might be able to use an event hook with lua, you can catch the CUSTOM
with a subclass of valet_parking::info, then you can check for the "action"
and it'll be "exit" when they leave (any kind of leave) then can likely
check for a disposition.  I'm not currently using this method, but its
someting we looked into for monitoring when people hangup from the parking
lot.

On Tue, Jul 14, 2015 at 10:49 AM, SamyGo <govoiper at gmail.com> wrote:

> Hi All,
>
> I need some help in finding  a way to execute some dialplan code after the
> parked user Hangups from the parking lot.
>
> I've a scenario where Parked party decides to hangup while listening to
> MOH while parked. The Dialplan just Hangsup right away and I have no
> control.
>
> I've tried using: *session_in_hangup_hook*, and
> *api_hangup_hook=myscript.lua *but they get executed for the parker just
> after they hear the Parked slot number.
>
> Then I've tried *nolocal:export* these two variables for the other leg
> but nothing happens.
>
> Kindly suggest what other option are there.
>
> Best Regards.
> Sammy
>
>
> _________________________________________________________________________
> 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  Wed Jul 15 18:26:47 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 15 Jul 2015 10:26:47 -0400
Subject: [Freeswitch-users] Reduce connection time when video calling
In-Reply-To: <85386622.20150715144307@seznam.cz>
References: <55A5FC8E.1070901@dwide.com>
	<CAHtxdDeBST6kmJeEn=VN_TzDdCv1rppT9g5qxX4bcyEq-Hcn+g@mail.gmail.com>
	<55A60B73.7070506@dwide.com>
	<1B5A6695-D5C9-49F3-A5A4-0384289586A2@gmail.com>
	<55A62C17.9020300@dwide.com> <85386622.20150715144307@seznam.cz>
Message-ID: <CAEwTsAiqUgAiy+bW7vgOAHFa84rtOFbMbKotf05_FOCgaD4YOA@mail.gmail.com>

sounds like answer delay is configured

On Wednesday, July 15, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Hi all,
>
> The question in the following. How it is possible to reduce the connection
> time with the customer video calls? If you do directly with "bypass_media =
> true", then both callers begin to see each other almost instantaneously.
> Through FreeSwitch this connection can take from 5 to 10 seconds. It can
> somehow reduce?
>
> Thanks
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 * <http://www.freeswitch.org>
>
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From igorolhovskiy at gmail.com  Wed Jul 15 14:55:27 2015
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Wed, 15 Jul 2015 13:55:27 +0300
Subject: [Freeswitch-users] Firefox 39, webrtc
Message-ID: <CAJTkRNsXxSV_WYRk6dY9WN=kPuv_h2Y+By+j7KUmCBUx8k8zyw@mail.gmail.com>

Hi!

I'm using sip.js with FS (tried FS 1.5.15 and 1.7). Upgrading to Firefox 39
is seems to be bad idea, cause call through it fails due to

2015-07-15 10:48:39.168250 [ERR] switch_rtp.c:2907 audio Handshake failure 1
2015-07-15 10:48:39.168250 [INFO] switch_rtp.c:2908 Changing audio DTLS
state from HANDSHAKE to FAIL

On FF 38 and Chrome all is ok.
Am I only one with this issue of it's really something wrong with FF?

-- 
Best regards,
Igor
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From yadenis at seznam.cz  Wed Jul 15 18:42:02 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Wed, 15 Jul 2015 16:42:02 +0200
Subject: [Freeswitch-users] Reduce connection time when video calling
In-Reply-To: <CAEwTsAiqUgAiy+bW7vgOAHFa84rtOFbMbKotf05_FOCgaD4YOA@mail.gmail.com>
References: <55A5FC8E.1070901@dwide.com> 
	<CAHtxdDeBST6kmJeEn=VN_TzDdCv1rppT9g5qxX4bcyEq-Hcn+g@mail.gmail.com>
	<55A60B73.7070506@dwide.com>
	<1B5A6695-D5C9-49F3-A5A4-0384289586A2@gmail.com>
	<55A62C17.9020300@dwide.com> <85386622.20150715144307@seznam.cz>
	<CAEwTsAiqUgAiy+bW7vgOAHFa84rtOFbMbKotf05_FOCgaD4YOA@mail.gmail.com>
Message-ID: <1404718420.20150715164202@seznam.cz>

Dobr? den,

How can I check it? I do not have in dialplan delay-time 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

st?eda 15. ?ervence 2015, 16:26:47, napsal jste:


sounds like answer delay is configured

On Wednesday, July 15, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi all,

The question in the following. How it is possible to reduce the connection time with the customer video calls? If you do directly with "bypass_media = true", then both callers begin to see each other almost instantaneously. Through FreeSwitch this connection can take from 5 to 10 seconds. It can somehow reduce?

Thanks

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From victor.medina at cibersys.com  Wed Jul 15 17:05:21 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Wed, 15 Jul 2015 08:35:21 -0430
Subject: [Freeswitch-users] Jira and Stash seems to be down
Message-ID: <CACXsapNuaBAtmE6BciNzwftdiqovKzaDMsZHgG3tGQxKLEfgBg@mail.gmail.com>

Guys.

Stash and Jira seems to be down, created the mail on the list because Jira
is also down.

 A fatal error has occurred

The following problem occurred which prevents Atlassian Stash from starting
correctly:

   - Unable to create and acquire exclusive lock file
   '/opt/stash-home/.lock' for Stash home directory '/opt/stash-home'.

   Please ensure that the user running Stash has permission to write to
   this directory.

   If this is already the case, please check the logs for more information.



-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From s.safarov at gmail.com  Wed Jul 15 14:58:06 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 15 Jul 2015 13:58:06 +0300
Subject: [Freeswitch-users] Call thru external gate released before
	INVITE (NORMAL_UNSPECIFIED)
In-Reply-To: <55A62C17.9020300@dwide.com>
References: <55A5FC8E.1070901@dwide.com>
	<CAHtxdDeBST6kmJeEn=VN_TzDdCv1rppT9g5qxX4bcyEq-Hcn+g@mail.gmail.com>
	<55A60B73.7070506@dwide.com>
	<1B5A6695-D5C9-49F3-A5A4-0384289586A2@gmail.com>
	<55A62C17.9020300@dwide.com>
Message-ID: <CAHtxdDcznQXpg47EVTAqq382Esi6=fqsp6aUoVoF0__hrn+0CA@mail.gmail.com>

Where are you read "% used when registered user dialed"?
Try call FS users via
originate {ignore_early_media=true}user/00294539100

On Wed, Jul 15, 2015 at 12:47 PM, Dmitry Mordovin <dm at dwide.com> wrote:

>  00294539100 - internal user, % used when registered user dialed
>
> Second sipnet.ru crop
>
> Same problem....
> If I dial both external numbers, or first external (second internal user)
> - they bridged fine.
>
> Problem occured when internal user dialed first, and bridge to external
> number.
>
>
>
> On 07/15/2015 12:49 PM, Oz Mortimer wrote:
>
> Should be
>
>  *originate {ignore_early_media=true}sofia/gateway/sipnet.ru/
> <http://sipnet.ru/>**919262417024
> <919262417024 at sipnet.ru> &bridge(sofia/gateway**/sipnet.ru/
> <http://sipnet.ru/>**00294539100)*
>
>  You seem to be doubling up the signet.ru by having as a gateway then
>  again @*sipnet.ru <http://sipnet.ru> - *either is valid but both at the
> same time is wrong.
> After the destination number you had % - not sure why, but wouldn?t have
> thought it should be there.
> Additionally you are using 2 different number formats (with and without 00
> prefix) - I would check with provider what format you should be sending.
>
>
>
>  On 15 Jul 2015, at 08:27, Dmitry Mordovin <dm at dwide.com> wrote:
>
>  Sergey, Peter,
>
> *originate {ignore_early_media=true}sofia/gateway/sipnet.ru/
> <http://sipnet.ru/>**919262417024 at sipnet.ru <919262417024 at sipnet.ru>
> &bridge(sofia/internal/00294539100%)*
>
> If i change like this - call established fine, but fail if first call to
> internal 00294539100
>
> New trace
>
>
> 2015-07-15 11:17:57.115208 [NOTICE] switch_channel.c:1075 New Channel
> sofia/external/919262417024 at sipnet.ru
> [4c3ee575-4f0f-4ebd-b4ec-0e5357469690]
> 2015-07-15 11:17:57.115208 [DEBUG] mod_sofia.c:4701 (
> sofia/external/919262417024 at sipnet.ru) State Change CS_NEW -> CS_INIT
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/external/919262417024 at sipnet.ru [BREAK]
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:472 (
> sofia/external/919262417024 at sipnet.ru) Running State Change CS_INIT
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:512 (
> sofia/external/919262417024 at sipnet.ru) State INIT
> 2015-07-15 11:17:57.115208 [DEBUG] mod_sofia.c:87
> sofia/external/919262417024 at sipnet.ru SOFIA INIT
> nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
> 2015-07-15 11:17:57.115208 [DEBUG] sofia_glue.c:1236
> sofia/external/919262417024 at sipnet.ru sending invite version: 1.5.15b git
> acdb1ca 2015-05-17 18:45:52Z 32bit
> Local SDP:
> v=0
> o=FreeSWITCH 1436923599 1436923600 IN IP4 194.87.7.17
> s=FreeSWITCH
> c=IN IP4 194.87.7.17
> t=0 0
> m=audio 21078 RTP/AVP 9 0 8 3 101 13
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
> m=video 20298 RTP/AVP 103
> a=rtpmap:103 VP8/90000
>
> nua.c:633 nua_invite() nua: nua_invite: entering
> nua_stack.c:529 nua_signal() nua(0xb6d54b10): sent signal r_invite
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:40
> sofia/external/919262417024 at sipnet.ru Standard INIT
> nua_stack.c:569 nua_stack_signal() nua(0xb6d54b10): recv signal r_invite
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:48 (
> sofia/external/919262417024 at sipnet.ru) State Change CS_INIT -> CS_ROUTING
> soa.c:280 soa_clone() soa_clone(static::0x8a45980, 0x8a547e8, 0xb6d54b10)
> called
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/external/919262417024 at sipnet.ru [BREAK]
> soa.c:403 soa_set_params() soa_set_params(static::0x8a4cbb0, ...) called
> soa.c:403 soa_set_params() soa_set_params(static::0x8a4cbb0, ...) called
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:512 (
> sofia/external/919262417024 at sipnet.ru) State INIT going to sleep
> soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x8a4cbb0, (nil),
> 0xb6d831a4, -1) called
> soa.c:890 soa_set_capability_sdp()
> soa_set_capability_sdp(static::0x8a4cbb0, (nil), 0xb6d831a4, -1) called
> nua_stack.c:271 nua_stack_event() nua(0xb6d54b10): event r_invite 900
> Internal error at nua_client.c:552
> nua_session.c:4139 signal_call_state_change() nua(0xb6d54b10): call state
> changed: init -> terminated
> nua_stack.c:271 nua_stack_event() nua(0xb6d54b10): event i_state 900
> Internal error at nua_client.c:552
> nua_stack.c:271 nua_stack_event() nua(0xb6d54b10): event i_terminated 900
> Internal error at nua_client.c:552
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/external/919262417024 at sipnet.ru [BREAK]
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/external/919262417024 at sipnet.ru [BREAK]
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_state_machine.c:472 (
> sofia/external/919262417024 at sipnet.ru) Running State Change CS_ROUTING
> 2015-07-15 11:17:57.115208 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/external/919262417024 at sipnet.ru [BREAK]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-07-15 11:17:57.115208 [DEBUG] sofia.c:6627 Channel
> sofia/external/919262417024 at sipnet.ru entering state [terminated][900]
> 2015-07-15 11:17:57.115208 [NOTICE] sofia.c:7543 Hangup
> sofia/external/919262417024 at sipnet.ru [CS_ROUTING] [NORMAL_UNSPECIFIED]
>
>
>
>
> On 07/15/2015 11:14 AM, Sergey Safarov wrote:
>
>  What is "voip_provider"?
> Please enable
> 1) sofia global siptrace on
> 2) sofia loglevel all 9
>
>
>
> _________________________________________________________________________
> 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.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 brian at freeswitch.org  Wed Jul 15 19:06:06 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 15 Jul 2015 10:06:06 -0500
Subject: [Freeswitch-users] Jira and Stash seems to be down
In-Reply-To: <CACXsapNuaBAtmE6BciNzwftdiqovKzaDMsZHgG3tGQxKLEfgBg@mail.gmail.com>
References: <CACXsapNuaBAtmE6BciNzwftdiqovKzaDMsZHgG3tGQxKLEfgBg@mail.gmail.com>
Message-ID: <CAEJMVkCB5A-P8DjYkuFr_=xh=SMcR5wxt3Gcp4WiWcR_6FasjA@mail.gmail.com>

This is a good example of WHY everyone should follow us on twitter and
facebook, I posted a status update on facebook about this very topic.

Mr. Rice and I have been up since a little before 5am working to restore
everything, There was a major power issue in Dallas that impacted the data
center our equipment is hosted in, multiple failures of various backup
systems happened, We are flushing out all the remaining bugs related to
having everything rebooted.

Thanks,



On Wed, Jul 15, 2015 at 8:05 AM, Victor Medina <victor.medina at cibersys.com>
wrote:

> Guys.
>
> Stash and Jira seems to be down, created the mail on the list because Jira
> is also down.
>
>  A fatal error has occurred
>
> The following problem occurred which prevents Atlassian Stash from
> starting correctly:
>
>    - Unable to create and acquire exclusive lock file
>    '/opt/stash-home/.lock' for Stash home directory '/opt/stash-home'.
>
>    Please ensure that the user running Stash has permission to write to
>    this directory.
>
>    If this is already the case, please check the logs for more
>    information.
>
>
>
> --
>
>
>
> V?ctor E. Medina M.
> Platform Architect / Chief Infrastructure
> +58424 291 4561
> BB #79A8AFA2
> @VMCibersys
>
>
> _________________________________________________________________________
> 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
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From brian at freeswitch.org  Wed Jul 15 19:08:49 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 15 Jul 2015 10:08:49 -0500
Subject: [Freeswitch-users] Firefox 39, webrtc
In-Reply-To: <CAJTkRNsXxSV_WYRk6dY9WN=kPuv_h2Y+By+j7KUmCBUx8k8zyw@mail.gmail.com>
References: <CAJTkRNsXxSV_WYRk6dY9WN=kPuv_h2Y+By+j7KUmCBUx8k8zyw@mail.gmail.com>
Message-ID: <CAEJMVkCqc8y-ne0SOAuZNhxYe159HjNGi0acxMnZQm-+FQtEaQ@mail.gmail.com>

I would like to know what we can do to get the point across about filing
bugs on JIRA?

Even if you file something and it ends up not being a bug, we can at the
very least close it as 'not a bug'.  What we can't do is mentally track
every last issue thats reported on the list.

https://www.youtube.com/watch?v=lxB57TtX27I

Thanks,



On Wed, Jul 15, 2015 at 5:55 AM, Igor Olhovskiy <igorolhovskiy at gmail.com>
wrote:

> Hi!
>
> I'm using sip.js with FS (tried FS 1.5.15 and 1.7). Upgrading to Firefox
> 39 is seems to be bad idea, cause call through it fails due to
>
> 2015-07-15 10:48:39.168250 [ERR] switch_rtp.c:2907 audio Handshake failure
> 1
> 2015-07-15 10:48:39.168250 [INFO] switch_rtp.c:2908 Changing audio DTLS
> state from HANDSHAKE to FAIL
>
> On FF 38 and Chrome all is ok.
> Am I only one with this issue of it's really something wrong with FF?
>
> --
> Best 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
>



-- 

*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
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From igorolhovskiy at gmail.com  Wed Jul 15 15:00:35 2015
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Wed, 15 Jul 2015 14:00:35 +0300
Subject: [Freeswitch-users] Firefox 39, webrtc
In-Reply-To: <CAJTkRNsXxSV_WYRk6dY9WN=kPuv_h2Y+By+j7KUmCBUx8k8zyw@mail.gmail.com>
References: <CAJTkRNsXxSV_WYRk6dY9WN=kPuv_h2Y+By+j7KUmCBUx8k8zyw@mail.gmail.com>
Message-ID: <CAJTkRNuAJznORP0UGreS+i_fR_e2BfNv=kx3h8kVhLCJR2V7Uw@mail.gmail.com>

Ubuntu 14.04, Amazon EC2

2015-07-15 13:55 GMT+03:00 Igor Olhovskiy <igorolhovskiy at gmail.com>:

> Hi!
>
> I'm using sip.js with FS (tried FS 1.5.15 and 1.7). Upgrading to Firefox
> 39 is seems to be bad idea, cause call through it fails due to
>
> 2015-07-15 10:48:39.168250 [ERR] switch_rtp.c:2907 audio Handshake failure
> 1
> 2015-07-15 10:48:39.168250 [INFO] switch_rtp.c:2908 Changing audio DTLS
> state from HANDSHAKE to FAIL
>
> On FF 38 and Chrome all is ok.
> Am I only one with this issue of it's really something wrong with FF?
>
> --
> Best regards,
> Igor
>



-- 
Best regards,
Igor
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From krice at freeswitch.org  Wed Jul 15 19:34:24 2015
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 15 Jul 2015 10:34:24 -0500
Subject: [Freeswitch-users] Join us for ClueCon Weekly Today at 1PM Eastern
 as we visit with Randall Schwartz of Perl and TWiT.tv Fame!
Message-ID: <D1CBE7B0.A25C3%krice@freeswitch.org>

Greetings!

Today on ClueCon Weekly, Randall Schwartz host of FLOSS Weekly and author of
many PERL books will be dropping in for a visit.

Join us live at https://cantina.freeswitch.org/verto and call 888 (have your
WebCam and Headset Ready!)

Today July 15, 1PM EST/10AM PST


Don?t forget to follow us on Twitter @FreeSWITCH and @ClueCon for the latest
updates!

K



-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From igorolhovskiy at gmail.com  Wed Jul 15 19:47:20 2015
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Wed, 15 Jul 2015 18:47:20 +0300
Subject: [Freeswitch-users] Firefox 39, webrtc
In-Reply-To: <CAEJMVkCqc8y-ne0SOAuZNhxYe159HjNGi0acxMnZQm-+FQtEaQ@mail.gmail.com>
References: <CAJTkRNsXxSV_WYRk6dY9WN=kPuv_h2Y+By+j7KUmCBUx8k8zyw@mail.gmail.com>
	<CAEJMVkCqc8y-ne0SOAuZNhxYe159HjNGi0acxMnZQm-+FQtEaQ@mail.gmail.com>
Message-ID: <CAJTkRNugkoY6c_kod0E+bZsKGDN23c9bXhvfh+C8bw7QOoCCow@mail.gmail.com>

I'm trying to figure out if it is a bug of FS :)
So asking if someone is having the same )

2015-07-15 18:08 GMT+03:00 Brian West <brian at freeswitch.org>:

> I would like to know what we can do to get the point across about filing
> bugs on JIRA?
>
> Even if you file something and it ends up not being a bug, we can at the
> very least close it as 'not a bug'.  What we can't do is mentally track
> every last issue thats reported on the list.
>
> https://www.youtube.com/watch?v=lxB57TtX27I
>
> Thanks,
>
>
>
> On Wed, Jul 15, 2015 at 5:55 AM, Igor Olhovskiy <igorolhovskiy at gmail.com>
> wrote:
>
>> Hi!
>>
>> I'm using sip.js with FS (tried FS 1.5.15 and 1.7). Upgrading to Firefox
>> 39 is seems to be bad idea, cause call through it fails due to
>>
>> 2015-07-15 10:48:39.168250 [ERR] switch_rtp.c:2907 audio Handshake
>> failure 1
>> 2015-07-15 10:48:39.168250 [INFO] switch_rtp.c:2908 Changing audio DTLS
>> state from HANDSHAKE to FAIL
>>
>> On FF 38 and Chrome all is ok.
>> Am I only one with this issue of it's really something wrong with FF?
>>
>> --
>> Best 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
>>
>
>
>
> --
>
> *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
>
> _________________________________________________________________________
> 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
>



-- 
Best regards,
Igor
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From carlos.ruizdiaz at gmail.com  Wed Jul 15 20:01:05 2015
From: carlos.ruizdiaz at gmail.com (=?UTF-8?Q?Carlos_Ruiz_D=C3=ADaz?=)
Date: Wed, 15 Jul 2015 11:01:05 -0500
Subject: [Freeswitch-users] Firefox 39, webrtc
In-Reply-To: <CAJTkRNugkoY6c_kod0E+bZsKGDN23c9bXhvfh+C8bw7QOoCCow@mail.gmail.com>
References: <CAJTkRNsXxSV_WYRk6dY9WN=kPuv_h2Y+By+j7KUmCBUx8k8zyw@mail.gmail.com>
	<CAEJMVkCqc8y-ne0SOAuZNhxYe159HjNGi0acxMnZQm-+FQtEaQ@mail.gmail.com>
	<CAJTkRNugkoY6c_kod0E+bZsKGDN23c9bXhvfh+C8bw7QOoCCow@mail.gmail.com>
Message-ID: <CA+3gBcnCz3tgcyiDgXdnY7dddbnbY9s+kbG8TrDR9BJHExN-Gw@mail.gmail.com>

There's a high change this being a bug in FS, caused by FF removing lots of
cipher suites it used to use for the handshake.

I experienced this myself with rtpengine.

This is the commit that fixed the issue on that project, in case it may be
useful for the maintainer(s) in charge of webRTC in FS. [1]

[1]
https://github.com/sipwise/rtpengine/commit/21e1fb680762f421e05fab036f8138b2276f5037

On Wed, Jul 15, 2015 at 10:47 AM, Igor Olhovskiy <igorolhovskiy at gmail.com>
wrote:

> I'm trying to figure out if it is a bug of FS :)
> So asking if someone is having the same )
>
> 2015-07-15 18:08 GMT+03:00 Brian West <brian at freeswitch.org>:
>
>> I would like to know what we can do to get the point across about filing
>> bugs on JIRA?
>>
>> Even if you file something and it ends up not being a bug, we can at the
>> very least close it as 'not a bug'.  What we can't do is mentally track
>> every last issue thats reported on the list.
>>
>> https://www.youtube.com/watch?v=lxB57TtX27I
>>
>> Thanks,
>>
>>
>>
>> On Wed, Jul 15, 2015 at 5:55 AM, Igor Olhovskiy <igorolhovskiy at gmail.com>
>> wrote:
>>
>>> Hi!
>>>
>>> I'm using sip.js with FS (tried FS 1.5.15 and 1.7). Upgrading to Firefox
>>> 39 is seems to be bad idea, cause call through it fails due to
>>>
>>> 2015-07-15 10:48:39.168250 [ERR] switch_rtp.c:2907 audio Handshake
>>> failure 1
>>> 2015-07-15 10:48:39.168250 [INFO] switch_rtp.c:2908 Changing audio DTLS
>>> state from HANDSHAKE to FAIL
>>>
>>> On FF 38 and Chrome all is ok.
>>> Am I only one with this issue of it's really something wrong with FF?
>>>
>>> --
>>> Best 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
>>>
>>
>>
>>
>> --
>>
>> *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
>>
>> _________________________________________________________________________
>> 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
>>
>
>
>
> --
> Best 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
>



-- 
Carlos
http://caruizdiaz.com
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From mike at jerris.com  Wed Jul 15 20:19:14 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 15 Jul 2015 12:19:14 -0400
Subject: [Freeswitch-users] Reduce connection time when video calling
In-Reply-To: <1404718420.20150715164202@seznam.cz>
References: <55A5FC8E.1070901@dwide.com>
	<CAHtxdDeBST6kmJeEn=VN_TzDdCv1rppT9g5qxX4bcyEq-Hcn+g@mail.gmail.com>
	<55A60B73.7070506@dwide.com>
	<1B5A6695-D5C9-49F3-A5A4-0384289586A2@gmail.com>
	<55A62C17.9020300@dwide.com> <85386622.20150715144307@seznam.cz>
	<CAEwTsAiqUgAiy+bW7vgOAHFa84rtOFbMbKotf05_FOCgaD4YOA@mail.gmail.com>
	<1404718420.20150715164202@seznam.cz>
Message-ID: <22A9EB90-4505-4382-878E-ABA9AFD4072A@jerris.com>

do you have anything like 

<X-PRE-PROCESS cmd="set" data="answer_delay=4000"/>

in your vars.xml

> On Jul 15, 2015, at 10:42 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
> 
> Dobr? den,
> 
> How can I check it? I do not have in dialplan delay-time 
> 
> -- 
> S pozdravem,
> Ing.Denis Jakovlev                           
> mob.tel. 775-415-382
> 
> st?eda 15. ?ervence 2015, 16:26:47, napsal jste:
> 
> 
> sounds like answer delay is configured
> 
> On Wednesday, July 15, 2015, Denis Jakovlev <yadenis at seznam.cz <mailto:yadenis at seznam.cz>> wrote:
> Hi all,
> 
> The question in the following. How it is possible to reduce the connection time with the customer video calls? If you do directly with "bypass_media = true", then both callers begin to see each other almost instantaneously. Through FreeSwitch this connection can take from 5 to 10 seconds. It can somehow reduce?
> 
> Thanks
> 
> -- 
> S pozdravem,
> Ing.Denis Jakovlev                           
> mob.tel <http://mob.tel/>. 775-415-382
> _________________________________________________________________________
> 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 blasterjr at gmail.com  Wed Jul 15 20:28:21 2015
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Wed, 15 Jul 2015 10:28:21 -0600
Subject: [Freeswitch-users] Execute dialplan after valet_park
In-Reply-To: <CAJUJwtjdBerXshWVJhQ0KJ-M_chiAaNiXgGaEU4hs4_nN3b8+Q@mail.gmail.com>
References: <CAJUJwtipc+yhpck+w4M4mYYPk0ZRa5NqWhytRChbsi4c6_nwdA@mail.gmail.com>
	<CAO3hyzE0joUxdFeQM4oJ9K64qdLcG_4uock8v3AqHDWc6pSdaw@mail.gmail.com>
	<CAJUJwtjdBerXshWVJhQ0KJ-M_chiAaNiXgGaEU4hs4_nN3b8+Q@mail.gmail.com>
Message-ID: <CAO3hyzF0dsxCEopEOeONnNUneTpyRiwXnxSJVLiTnrOEKRVYHA@mail.gmail.com>

When you use a hook, whenever that event (CUSTOM) is called the lua script
would be called, so its a seperate script, you then have to do conditions
to check that the type is valet_parking::info and execute logic inside that
condition

On Wed, Jul 15, 2015 at 8:22 AM, SamyGo <govoiper at gmail.com> wrote:

> Thanks Chris,
>
> That was my finding too, I just am not sure if I can monitor this event
> from the main LUA script or need an independent lua script only for the
> park "hangup" event monitoring. Any further help on this would be highly
> appreciated.
>
> I wonder why the api_hangup hook won't trigger on the parked channel,
> would've been much easier. Piece of cake only if we could execute further
> dialplan application after valet_parking.
>
> BR,
> Sammy
> You might be able to use an event hook with lua, you can catch the CUSTOM
> with a subclass of valet_parking::info, then you can check for the "action"
> and it'll be "exit" when they leave (any kind of leave) then can likely
> check for a disposition.  I'm not currently using this method, but its
> someting we looked into for monitoring when people hangup from the parking
> lot.
>
> On Tue, Jul 14, 2015 at 10:49 AM, SamyGo <govoiper at gmail.com> wrote:
>
>> Hi All,
>>
>> I need some help in finding  a way to execute some dialplan code after
>> the parked user Hangups from the parking lot.
>>
>> I've a scenario where Parked party decides to hangup while listening to
>> MOH while parked. The Dialplan just Hangsup right away and I have no
>> control.
>>
>> I've tried using: *session_in_hangup_hook*, and
>> *api_hangup_hook=myscript.lua *but they get executed for the parker just
>> after they hear the Parked slot number.
>>
>> Then I've tried *nolocal:export* these two variables for the other leg
>> but nothing happens.
>>
>> Kindly suggest what other option are there.
>>
>> Best Regards.
>> Sammy
>>
>>
>> _________________________________________________________________________
>> 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 brian at freeswitch.org  Wed Jul 15 20:59:16 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 15 Jul 2015 11:59:16 -0500
Subject: [Freeswitch-users] ClueCon 2015,
	Discount Ticket Price extended to July 17th
Message-ID: <CAEJMVkAco93rgf3or3ROz56fZAPjw+tACHFQZhtyHFhHfvG0OQ@mail.gmail.com>

Dear FreeSWITCHers,

In honor of Prime Day, We've decided to extend the current pricing for
ClueCon tickets till the 17th, Hurry and REGISTER before the prices go up.

Hotel cutoff date is still today, so get  your reservations in ASAP.

Thanks,

-- 

*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
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From ssinyagin at gmail.com  Wed Jul 15 22:04:45 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 15 Jul 2015 20:04:45 +0200
Subject: [Freeswitch-users] Jira and Stash seems to be down
In-Reply-To: <CAEJMVkCB5A-P8DjYkuFr_=xh=SMcR5wxt3Gcp4WiWcR_6FasjA@mail.gmail.com>
References: <CACXsapNuaBAtmE6BciNzwftdiqovKzaDMsZHgG3tGQxKLEfgBg@mail.gmail.com>
	<CAEJMVkCB5A-P8DjYkuFr_=xh=SMcR5wxt3Gcp4WiWcR_6FasjA@mail.gmail.com>
Message-ID: <CACOYK=i=FhbYOQj9=3ojgE2=B_cd-gdEsKbP8ccTE0z+AZKS4w@mail.gmail.com>

A simple website like status.freeswitch.org would be much more helpful. It
just needs to be hosted independently :)
On Jul 15, 2015 5:07 PM, "Brian West" <brian at freeswitch.org> wrote:

> This is a good example of WHY everyone should follow us on twitter and
> facebook, I posted a status update on facebook about this very topic.
>
> Mr. Rice and I have been up since a little before 5am working to restore
> everything, There was a major power issue in Dallas that impacted the data
> center our equipment is hosted in, multiple failures of various backup
> systems happened, We are flushing out all the remaining bugs related to
> having everything rebooted.
>
> Thanks,
>
>
>
> On Wed, Jul 15, 2015 at 8:05 AM, Victor Medina <victor.medina at cibersys.com
> > wrote:
>
>> Guys.
>>
>> Stash and Jira seems to be down, created the mail on the list because
>> Jira is also down.
>>
>>  A fatal error has occurred
>>
>> The following problem occurred which prevents Atlassian Stash from
>> starting correctly:
>>
>>    - Unable to create and acquire exclusive lock file
>>    '/opt/stash-home/.lock' for Stash home directory '/opt/stash-home'.
>>
>>    Please ensure that the user running Stash has permission to write to
>>    this directory.
>>
>>    If this is already the case, please check the logs for more
>>    information.
>>
>>
>>
>> --
>>
>>
>>
>> V?ctor E. Medina M.
>> Platform Architect / Chief Infrastructure
>> +58424 291 4561
>> BB #79A8AFA2
>> @VMCibersys
>>
>>
>> _________________________________________________________________________
>> 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
>
> _________________________________________________________________________
> 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 freeswitch.org  Wed Jul 15 22:08:06 2015
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 15 Jul 2015 13:08:06 -0500
Subject: [Freeswitch-users] Jira and Stash seems to be down
In-Reply-To: <CACOYK=i=FhbYOQj9=3ojgE2=B_cd-gdEsKbP8ccTE0z+AZKS4w@mail.gmail.com>
Message-ID: <D1CC0BB6.A2606%krice@freeswitch.org>

We?re looking at several options for things like this up to and including a
way to just move everything to a different site with the click of a mouse
via some BGP tricks


On 7/15/15, 1:04 PM, "Stanislav Sinyagin" <ssinyagin at gmail.com> wrote:

> A simple website like status.freeswitch.org <http://status.freeswitch.org>
> would be much more helpful. It just needs to be hosted independently :)
> 
> On Jul 15, 2015 5:07 PM, "Brian West" <brian at freeswitch.org> wrote:
>> This is a good example of WHY everyone should follow us on twitter and
>> facebook, I posted a status update on facebook about this very topic.
>> 
>> Mr. Rice and I have been up since a little before 5am working to restore
>> everything, There was a major power issue in Dallas that impacted the data
>> center our equipment is hosted in, multiple failures of various backup
>> systems happened, We are flushing out all the remaining bugs related to
>> having everything rebooted.
>> 
>> Thanks,
>> 
>> 
>> 
>> On Wed, Jul 15, 2015 at 8:05 AM, Victor Medina <victor.medina at cibersys.com>
>> wrote:
>>> Guys.
>>> 
>>> Stash and Jira seems to be down, created the mail on the list because Jira
>>> is also down.
>>> 
>>>  
>>>  
>>> A fatal error has occurred
>>>  
>>>  
>>> 
>>> The following problem occurred which prevents Atlassian Stash from starting
>>> correctly:
>>>  
>>> * Unable to create and acquire exclusive lock file '/opt/stash-home/.lock'
>>> for Stash home directory '/opt/stash-home'.
>>> * 
>>> * Please ensure that the user running Stash has permission to write to this
>>> directory. 
>>> * 
>>> * If this is already the case, please check the logs for more information.
>>> 

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From victor.medina at cibersys.com  Thu Jul 16 00:05:48 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Wed, 15 Jul 2015 15:35:48 -0430
Subject: [Freeswitch-users] Jira and Stash seems to be down
In-Reply-To: <D1CC0BB6.A2606%krice@freeswitch.org>
References: <CACOYK=i=FhbYOQj9=3ojgE2=B_cd-gdEsKbP8ccTE0z+AZKS4w@mail.gmail.com>
	<D1CC0BB6.A2606%krice@freeswitch.org>
Message-ID: <CACXsapN=r_UX=jYSEdeoKspeBoy+To5oefbn7LjZoJc1+C3MPw@mail.gmail.com>

Ill follow you on twitter! =)
El 15/07/2015 13:39, "Ken Rice" <krice at freeswitch.org> escribi?:

>  We?re looking at several options for things like this up to and
> including a way to just move everything to a different site with the click
> of a mouse via some BGP tricks
>
>
> On 7/15/15, 1:04 PM, "Stanislav Sinyagin" <ssinyagin at gmail.com> wrote:
>
> A simple website like status.freeswitch.org <http://status.freeswitch.org>
>  would be much more helpful. It just needs to be hosted independently :)
>
> On Jul 15, 2015 5:07 PM, "Brian West" <brian at freeswitch.org> wrote:
>
> This is a good example of WHY everyone should follow us on twitter and
> facebook, I posted a status update on facebook about this very topic.
>
> Mr. Rice and I have been up since a little before 5am working to restore
> everything, There was a major power issue in Dallas that impacted the data
> center our equipment is hosted in, multiple failures of various backup
> systems happened, We are flushing out all the remaining bugs related to
> having everything rebooted.
>
> Thanks,
>
>
>
> On Wed, Jul 15, 2015 at 8:05 AM, Victor Medina <victor.medina at cibersys.com>
> wrote:
>
> Guys.
>
> Stash and Jira seems to be down, created the mail on the list because Jira
> is also down.
>
>
>
>
> *A fatal error has occurred *
>
>
> The following problem occurred which prevents Atlassian Stash from
> starting correctly:
>
>
>    - Unable to create and acquire exclusive lock file
>    '/opt/stash-home/.lock' for Stash home directory '/opt/stash-home'.
>    -
>    - Please ensure that the user running Stash has permission to write to
>    this directory.
>    -
>    - If this is already the case, please check the logs for more
>    information.
>
>
>
> --
> Ken
>
>
>
> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
> Twitter: @FreeSWITCH
>
>
> _________________________________________________________________________
> 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 andrew at cassidywebservices.co.uk  Thu Jul 16 00:21:14 2015
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Wed, 15 Jul 2015 21:21:14 +0100
Subject: [Freeswitch-users] Jira and Stash seems to be down
In-Reply-To: <CACXsapN=r_UX=jYSEdeoKspeBoy+To5oefbn7LjZoJc1+C3MPw@mail.gmail.com>
References: <CACOYK=i=FhbYOQj9=3ojgE2=B_cd-gdEsKbP8ccTE0z+AZKS4w@mail.gmail.com>
	<D1CC0BB6.A2606%krice@freeswitch.org>
	<CACXsapN=r_UX=jYSEdeoKspeBoy+To5oefbn7LjZoJc1+C3MPw@mail.gmail.com>
Message-ID: <CAMbW9oLOS=X9AYio1y-kriPbF+3Vfd6W5yZ3KbzkseMCXkjFqQ@mail.gmail.com>

I use neither facebook or twitter! :D

On 15 July 2015 at 21:05, Victor Medina <victor.medina at cibersys.com> wrote:

> Ill follow you on twitter! =)
> El 15/07/2015 13:39, "Ken Rice" <krice at freeswitch.org> escribi?:
>
>>  We?re looking at several options for things like this up to and
>> including a way to just move everything to a different site with the click
>> of a mouse via some BGP tricks
>>
>>
>> On 7/15/15, 1:04 PM, "Stanislav Sinyagin" <ssinyagin at gmail.com> wrote:
>>
>> A simple website like status.freeswitch.org <http://status.freeswitch.org>
>>  would be much more helpful. It just needs to be hosted independently :)
>>
>> On Jul 15, 2015 5:07 PM, "Brian West" <brian at freeswitch.org> wrote:
>>
>> This is a good example of WHY everyone should follow us on twitter and
>> facebook, I posted a status update on facebook about this very topic.
>>
>> Mr. Rice and I have been up since a little before 5am working to restore
>> everything, There was a major power issue in Dallas that impacted the data
>> center our equipment is hosted in, multiple failures of various backup
>> systems happened, We are flushing out all the remaining bugs related to
>> having everything rebooted.
>>
>> Thanks,
>>
>>
>>
>> On Wed, Jul 15, 2015 at 8:05 AM, Victor Medina <
>> victor.medina at cibersys.com> wrote:
>>
>> Guys.
>>
>> Stash and Jira seems to be down, created the mail on the list because
>> Jira is also down.
>>
>>
>>
>>
>> *A fatal error has occurred *
>>
>>
>> The following problem occurred which prevents Atlassian Stash from
>> starting correctly:
>>
>>
>>    - Unable to create and acquire exclusive lock file
>>    '/opt/stash-home/.lock' for Stash home directory '/opt/stash-home'.
>>    -
>>    - Please ensure that the user running Stash has permission to write
>>    to this directory.
>>    -
>>    - If this is already the case, please check the logs for more
>>    information.
>>
>>
>>
>> --
>> Ken
>>
>>
>>
>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>> Twitter: @FreeSWITCH
>>
>>
>> _________________________________________________________________________
>> 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
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960  *F
<info at cassidywebservices.co.uk> *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From brian at freeswitch.org  Thu Jul 16 00:25:10 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 15 Jul 2015 15:25:10 -0500
Subject: [Freeswitch-users] Jira and Stash seems to be down
In-Reply-To: <CAMbW9oLOS=X9AYio1y-kriPbF+3Vfd6W5yZ3KbzkseMCXkjFqQ@mail.gmail.com>
References: <CACOYK=i=FhbYOQj9=3ojgE2=B_cd-gdEsKbP8ccTE0z+AZKS4w@mail.gmail.com>
	<D1CC0BB6.A2606%krice@freeswitch.org>
	<CACXsapN=r_UX=jYSEdeoKspeBoy+To5oefbn7LjZoJc1+C3MPw@mail.gmail.com>
	<CAMbW9oLOS=X9AYio1y-kriPbF+3Vfd6W5yZ3KbzkseMCXkjFqQ@mail.gmail.com>
Message-ID: <CAEJMVkCvUaRyvbMV3fnNk-eX_mxVwJGeTXiv_X_iXc5Lkq1M=A@mail.gmail.com>

Well in that case!
[image: Inline image 1]

On Wed, Jul 15, 2015 at 3:21 PM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> I use neither facebook or twitter! :D
>
> On 15 July 2015 at 21:05, Victor Medina <victor.medina at cibersys.com>
> wrote:
>
>> Ill follow you on twitter! =)
>> El 15/07/2015 13:39, "Ken Rice" <krice at freeswitch.org> escribi?:
>>
>>>  We?re looking at several options for things like this up to and
>>> including a way to just move everything to a different site with the click
>>> of a mouse via some BGP tricks
>>>
>>>
>>> On 7/15/15, 1:04 PM, "Stanislav Sinyagin" <ssinyagin at gmail.com> wrote:
>>>
>>> A simple website like status.freeswitch.org <
>>> http://status.freeswitch.org>  would be much more helpful. It just
>>> needs to be hosted independently :)
>>>
>>> On Jul 15, 2015 5:07 PM, "Brian West" <brian at freeswitch.org> wrote:
>>>
>>> This is a good example of WHY everyone should follow us on twitter and
>>> facebook, I posted a status update on facebook about this very topic.
>>>
>>> Mr. Rice and I have been up since a little before 5am working to restore
>>> everything, There was a major power issue in Dallas that impacted the data
>>> center our equipment is hosted in, multiple failures of various backup
>>> systems happened, We are flushing out all the remaining bugs related to
>>> having everything rebooted.
>>>
>>> Thanks,
>>>
>>>
>>>
>>> On Wed, Jul 15, 2015 at 8:05 AM, Victor Medina <
>>> victor.medina at cibersys.com> wrote:
>>>
>>> Guys.
>>>
>>> Stash and Jira seems to be down, created the mail on the list because
>>> Jira is also down.
>>>
>>>
>>>
>>>
>>> *A fatal error has occurred *
>>>
>>>
>>> The following problem occurred which prevents Atlassian Stash from
>>> starting correctly:
>>>
>>>
>>>    - Unable to create and acquire exclusive lock file
>>>    '/opt/stash-home/.lock' for Stash home directory '/opt/stash-home'.
>>>    -
>>>    - Please ensure that the user running Stash has permission to write
>>>    to this directory.
>>>    -
>>>    - If this is already the case, please check the logs for more
>>>    information.
>>>
>>>
>>>
>>> --
>>> Ken
>>>
>>>
>>>
>>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>> Twitter: @FreeSWITCH
>>>
>>>
>>> _________________________________________________________________________
>>> 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
>>
>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F
> <info at cassidywebservices.co.uk> *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
> _________________________________________________________________________
> 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
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From jungleboogie0 at gmail.com  Thu Jul 16 00:33:59 2015
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Wed, 15 Jul 2015 13:33:59 -0700
Subject: [Freeswitch-users] Jira and Stash seems to be down
In-Reply-To: <CACOYK=i=FhbYOQj9=3ojgE2=B_cd-gdEsKbP8ccTE0z+AZKS4w@mail.gmail.com>
References: <CACXsapNuaBAtmE6BciNzwftdiqovKzaDMsZHgG3tGQxKLEfgBg@mail.gmail.com>
	<CAEJMVkCB5A-P8DjYkuFr_=xh=SMcR5wxt3Gcp4WiWcR_6FasjA@mail.gmail.com>
	<CACOYK=i=FhbYOQj9=3ojgE2=B_cd-gdEsKbP8ccTE0z+AZKS4w@mail.gmail.com>
Message-ID: <CAKE2PDtf9qtWtdPsaXh=oLwyrZH6gVV7hyEoCYSgCcMOooTVMw@mail.gmail.com>

On 15 July 2015 at 11:04, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>
> A simple website like status.freeswitch.org would be much more helpful. It just needs to be hosted independently :)

For something less 'sophisticated' : https://uptimerobot.com/

Have uptimerobot email out the mailing list when/if there's an outage
and report when it's back up. I think it can also send twitter
messages, but I don't remember if it's the direct message thing or
posts to your twitter page.



-- 
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si


From davegreeko at yahoo.com  Thu Jul 16 02:50:19 2015
From: davegreeko at yahoo.com (Dave Greeko)
Date: Wed, 15 Jul 2015 15:50:19 -0700
Subject: [Freeswitch-users] T.38 Fax
Message-ID: <1437000619.82293.YahooMailBasic@web160303.mail.bf1.yahoo.com>

Hello All,
I'm trying to just relay t.38 fax calls along the voice calls between two endpoints with FS in the middle.
In the dial-plan I set the proxy media mode to true which works extremely well with my setup considering the codec negotiation headache for my voice calls. My question is do I have to explicitly set any Sofila parameters in the SIP profile or any variables related to T.38 in the Bridge dial string to make the fax go from Caller to Callee through FS?

My Freeswitch version is 1.4.19 and mod_spansdp is already loaded

Regards,
Dave


From david.witham at netsip.com.au  Thu Jul 16 04:50:29 2015
From: david.witham at netsip.com.au (David Witham)
Date: Thu, 16 Jul 2015 00:50:29 +0000
Subject: [Freeswitch-users] T.38 Fax
In-Reply-To: <1437000619.82293.YahooMailBasic@web160303.mail.bf1.yahoo.com>
References: <1437000619.82293.YahooMailBasic@web160303.mail.bf1.yahoo.com>
Message-ID: <1437007829472.35178@netsip.com.au>

Hi Dave,

You need to set t38_passthru=true in your dialplan to allow the t38 reINVITE to traverse your FS.

regards,
David
________________________________________
From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Dave Greeko <davegreeko at yahoo.com>
Sent: Thursday, 16 July 2015 08:50
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] T.38 Fax

Hello All,
I'm trying to just relay t.38 fax calls along the voice calls between two endpoints with FS in the middle.
In the dial-plan I set the proxy media mode to true which works extremely well with my setup considering the codec negotiation headache for my voice calls. My question is do I have to explicitly set any Sofila parameters in the SIP profile or any variables related to T.38 in the Bridge dial string to make the fax go from Caller to Callee through FS?

My Freeswitch version is 1.4.19 and mod_spansdp is already loaded

Regards,
Dave

_________________________________________________________________________
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 davegreeko at yahoo.com  Thu Jul 16 05:45:04 2015
From: davegreeko at yahoo.com (Dave Greeko)
Date: Wed, 15 Jul 2015 18:45:04 -0700
Subject: [Freeswitch-users] T.38 Fax
In-Reply-To: <1437007829472.35178@netsip.com.au>
Message-ID: <1437011104.12980.YahooMailBasic@web160303.mail.bf1.yahoo.com>

Many Thanks David. Should I set this once in Sofia Profile (<param name="t38-passthru" value="true"/>) or per dial-plan extension? lastly, if I can do it in the sip profile without worrying about the dial plan will that have any performance cost in terms of system resources - just a question?
P.S: I know for a fact FreeSwitch awesome community may not like these questions because they want us to try it and make it work first than ask....  I'm sorry in advanced


--------------------------------------------
On Wed, 7/15/15, David Witham <david.witham at netsip.com.au> wrote:

 Subject: Re: [Freeswitch-users] T.38 Fax
 To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
 Date: Wednesday, July 15, 2015, 5:50 PM
 
 Hi Dave,
 
 You need to set
 t38_passthru=true in your dialplan to allow the t38 reINVITE
 to traverse your FS.
 
 regards,
 David
 ________________________________________
 From:
 freeswitch-users-bounces at lists.freeswitch.org
 <freeswitch-users-bounces at lists.freeswitch.org>
 on behalf of Dave Greeko <davegreeko at yahoo.com>
 Sent: Thursday, 16 July 2015 08:50
 To: freeswitch-users at lists.freeswitch.org
 Subject: [Freeswitch-users] T.38 Fax
 
 Hello All,
 I'm trying to just relay t.38 fax calls
 along the voice calls between two endpoints with FS in the
 middle.
 In the dial-plan I set the proxy
 media mode to true which works extremely well with my setup
 considering the codec negotiation headache for my voice
 calls. My question is do I have to explicitly set any Sofila
 parameters in the SIP profile or any variables related to
 T.38 in the Bridge dial string to make the fax go from
 Caller to Callee through FS?
 
 My Freeswitch version is 1.4.19 and mod_spansdp
 is already loaded
 
 Regards,
 Dave
 
 _________________________________________________________________________
 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
 


From bhavikpatel14388 at gmail.com  Thu Jul 16 07:39:46 2015
From: bhavikpatel14388 at gmail.com (bhavik patel)
Date: Thu, 16 Jul 2015 09:09:46 +0530
Subject: [Freeswitch-users] BLF Implementation
In-Reply-To: <CAG6TCM9xynte8HzQxFXQ0VaZKmz4Njh+qd0TtZ6WP+KNgkxHTw@mail.gmail.com>
References: <CAG6TCM9xynte8HzQxFXQ0VaZKmz4Njh+qd0TtZ6WP+KNgkxHTw@mail.gmail.com>
Message-ID: <CAG6TCM8v4OjsQ87gxZYCYVz4aHZz-fo-uWJ_CEkA82FMypCfaw@mail.gmail.com>

Any feedback please.

On Tue, Jul 14, 2015 at 5:38 PM, bhavik patel <bhavikpatel14388 at gmail.com>
wrote:

> Hi,
>
> I am trying to implement BLF in Grandstream (GXP1400) with freeswitch
> 1.4.20.
>
> For that , i used "<param name="send-presence-on-register" value="true"/>"
> in sip profile.
>
> But Not getting any NOTIFY request to sip phone,and BLF is not working.
>
> Can any one suggest me how to enable this function in Freeswitch ?
>
> Any help would be much appreciated.
>
> --
> Thanks,
> Bhavik Patel
>
>
>
>


-- 
Thanks,
Bhavik Patel
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From mike at jerris.com  Thu Jul 16 07:53:56 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 15 Jul 2015 23:53:56 -0400
Subject: [Freeswitch-users] BLF Implementation
In-Reply-To: <CAG6TCM8v4OjsQ87gxZYCYVz4aHZz-fo-uWJ_CEkA82FMypCfaw@mail.gmail.com>
References: <CAG6TCM9xynte8HzQxFXQ0VaZKmz4Njh+qd0TtZ6WP+KNgkxHTw@mail.gmail.com>
	<CAG6TCM8v4OjsQ87gxZYCYVz4aHZz-fo-uWJ_CEkA82FMypCfaw@mail.gmail.com>
Message-ID: <CAEwTsAiA+3Jo_6hjxxt7CVeHFwdd63UsU1G-4-DWve3DfXr2GQ@mail.gmail.com>

manage-presence needs to be enabled

On Wednesday, July 15, 2015, bhavik patel <bhavikpatel14388 at gmail.com>
wrote:

> Any feedback please.
>
> On Tue, Jul 14, 2015 at 5:38 PM, bhavik patel <bhavikpatel14388 at gmail.com
> <javascript:_e(%7B%7D,'cvml','bhavikpatel14388 at gmail.com');>> wrote:
>
>> Hi,
>>
>> I am trying to implement BLF in Grandstream (GXP1400) with freeswitch
>> 1.4.20.
>>
>> For that , i used "<param name="send-presence-on-register"
>> value="true"/>" in sip profile.
>>
>> But Not getting any NOTIFY request to sip phone,and BLF is not working.
>>
>> Can any one suggest me how to enable this function in Freeswitch ?
>>
>> Any help would be much appreciated.
>>
>> --
>> Thanks,
>> Bhavik Patel
>>
>>
>>
>>
>
>
> --
> Thanks,
> Bhavik Patel
>
>
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From bhavikpatel14388 at gmail.com  Thu Jul 16 08:08:43 2015
From: bhavikpatel14388 at gmail.com (bhavik patel)
Date: Thu, 16 Jul 2015 09:38:43 +0530
Subject: [Freeswitch-users] BLF Implementation
In-Reply-To: <CAEwTsAiA+3Jo_6hjxxt7CVeHFwdd63UsU1G-4-DWve3DfXr2GQ@mail.gmail.com>
References: <CAG6TCM9xynte8HzQxFXQ0VaZKmz4Njh+qd0TtZ6WP+KNgkxHTw@mail.gmail.com>
	<CAG6TCM8v4OjsQ87gxZYCYVz4aHZz-fo-uWJ_CEkA82FMypCfaw@mail.gmail.com>
	<CAEwTsAiA+3Jo_6hjxxt7CVeHFwdd63UsU1G-4-DWve3DfXr2GQ@mail.gmail.com>
Message-ID: <CAG6TCM_mRoPM-p2Yi1iuLRk8y=wCRTvq4g9Xie3MHWNPjdKjhQ@mail.gmail.com>

I did try "
http://www.microweb10.com/index.php/en/tutorials/enable-presence-fs" but
not getting success.

Please suggest me.

On Thu, Jul 16, 2015 at 9:23 AM, Michael Jerris <mike at jerris.com> wrote:

> manage-presence needs to be enabled
>
>
> On Wednesday, July 15, 2015, bhavik patel <bhavikpatel14388 at gmail.com>
> wrote:
>
>> Any feedback please.
>>
>> On Tue, Jul 14, 2015 at 5:38 PM, bhavik patel <bhavikpatel14388 at gmail.com
>> > wrote:
>>
>>> Hi,
>>>
>>> I am trying to implement BLF in Grandstream (GXP1400) with freeswitch
>>> 1.4.20.
>>>
>>> For that , i used "<param name="send-presence-on-register"
>>> value="true"/>" in sip profile.
>>>
>>> But Not getting any NOTIFY request to sip phone,and BLF is not working.
>>>
>>> Can any one suggest me how to enable this function in Freeswitch ?
>>>
>>> Any help would be much appreciated.
>>>
>>> --
>>> Thanks,
>>> Bhavik Patel
>>>
>>>
>>>
>>>
>>
>>
>> --
>> Thanks,
>> Bhavik Patel
>>
>>
> _________________________________________________________________________
> 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
>



-- 
Thanks,
Bhavik Patel
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From david.witham at netsip.com.au  Thu Jul 16 09:32:45 2015
From: david.witham at netsip.com.au (David Witham)
Date: Thu, 16 Jul 2015 05:32:45 +0000
Subject: [Freeswitch-users] T.38 Fax
In-Reply-To: <1437011104.12980.YahooMailBasic@web160303.mail.bf1.yahoo.com>
References: <1437007829472.35178@netsip.com.au>,
	<1437011104.12980.YahooMailBasic@web160303.mail.bf1.yahoo.com>
Message-ID: <1437024765752.22353@netsip.com.au>

Hi Dave,

It is not a sofia param it is a channel variable so you should set it in the dialplan. Note that it is t38_passthru not t38-passthru. There should be no performance penalty using this variable.

regards,
David
________________________________________
From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Dave Greeko <davegreeko at yahoo.com>
Sent: Thursday, 16 July 2015 11:45
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] T.38 Fax

Many Thanks David. Should I set this once in Sofia Profile (<param name="t38-passthru" value="true"/>) or per dial-plan extension? lastly, if I can do it in the sip profile without worrying about the dial plan will that have any performance cost in terms of system resources - just a question?
P.S: I know for a fact FreeSwitch awesome community may not like these questions because they want us to try it and make it work first than ask....  I'm sorry in advanced


--------------------------------------------
On Wed, 7/15/15, David Witham <david.witham at netsip.com.au> wrote:

 Subject: Re: [Freeswitch-users] T.38 Fax
 To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
 Date: Wednesday, July 15, 2015, 5:50 PM

 Hi Dave,

 You need to set
 t38_passthru=true in your dialplan to allow the t38 reINVITE
 to traverse your FS.

 regards,
 David
 ________________________________________
 From:
 freeswitch-users-bounces at lists.freeswitch.org
 <freeswitch-users-bounces at lists.freeswitch.org>
 on behalf of Dave Greeko <davegreeko at yahoo.com>
 Sent: Thursday, 16 July 2015 08:50
 To: freeswitch-users at lists.freeswitch.org
 Subject: [Freeswitch-users] T.38 Fax

 Hello All,
 I'm trying to just relay t.38 fax calls
 along the voice calls between two endpoints with FS in the
 middle.
 In the dial-plan I set the proxy
 media mode to true which works extremely well with my setup
 considering the codec negotiation headache for my voice
 calls. My question is do I have to explicitly set any Sofila
 parameters in the SIP profile or any variables related to
 T.38 in the Bridge dial string to make the fax go from
 Caller to Callee through FS?

 My Freeswitch version is 1.4.19 and mod_spansdp
 is already loaded

 Regards,
 Dave

 _________________________________________________________________________
 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


From nandy1925 at gmail.com  Thu Jul 16 10:19:31 2015
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Thu, 16 Jul 2015 14:19:31 +0800
Subject: [Freeswitch-users] verto.conf.xml parameter for public access
In-Reply-To: <CAHKqMb5s9NGtaiEHO5Z5uWDi=gCdeTEmsPBqU6dLRwHc4r=-LQ@mail.gmail.com>
References: <CAHKqMb5s9NGtaiEHO5Z5uWDi=gCdeTEmsPBqU6dLRwHc4r=-LQ@mail.gmail.com>
Message-ID: <CAHKqMb6=Hn4ua8FJL-nvD4vUvHEnY8GaGtt_PkhFcUD1_uYguA@mail.gmail.com>

Hi to everyone,

More related question.

I see this line when I execute on fs_cli:

WSS-BIND-URL      sips:mod_sofia at 192.168.0.5:7443;transport=wss

It's my local IP address.  Shouldn't it be my public WAN IP address?

Please shed some light.  Thanks.

/Nandy

On Wed, Jul 15, 2015 at 9:53 AM, Nandy Dagondon <nandy1925 at gmail.com> wrote:

> I succesfully installed mod_verto in my intranet.  FS is behind NAT. Next
> step, I'm setting up where remote web clients can login.  I setup my router
> DMZ to FS.
>
> This parameter is commented out  "ext-rtp-ip" in the Wiki. Is this
> parameter necessary for accepting remote clients?
>
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From dm at dwide.com  Thu Jul 16 11:45:59 2015
From: dm at dwide.com (Dmitry Mordovin)
Date: Thu, 16 Jul 2015 11:45:59 +0400
Subject: [Freeswitch-users] GIT is down
Message-ID: <55A76137.1010304@dwide.com>

Hello!

GIT is down

git clone git://git.freeswitch.org/freeswitch.git
Cloning into 'freeswitch'...
fatal: unable to connect to git.freeswitch.org:
git.freeswitch.org[0: 209.105.235.6]: errno=Connection refused




From gmaruzz at gmail.com  Thu Jul 16 11:47:59 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 16 Jul 2015 09:47:59 +0200
Subject: [Freeswitch-users] GIT is down
In-Reply-To: <55A76137.1010304@dwide.com>
References: <55A76137.1010304@dwide.com>
Message-ID: <CALXCt0prKL=kedT1uEj+r_nr9=JWvyRk6H=SAhi0YENXyusJcw@mail.gmail.com>

You have an old url for git

Go to confluence page for new url

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On Jul 16, 2015 9:46 AM, "Dmitry Mordovin" <dm at dwide.com> wrote:

> Hello!
>
> GIT is down
>
> git clone git://git.freeswitch.org/freeswitch.git
> Cloning into 'freeswitch'...
> fatal: unable to connect to git.freeswitch.org:
> git.freeswitch.org[0: 209.105.235.6]: errno=Connection refused
>
>
>
> _________________________________________________________________________
> 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 yadenis at seznam.cz  Thu Jul 16 13:22:08 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Thu, 16 Jul 2015 11:22:08 +0200
Subject: [Freeswitch-users] Reduce connection time when video calling
In-Reply-To: <22A9EB90-4505-4382-878E-ABA9AFD4072A@jerris.com>
References: <55A5FC8E.1070901@dwide.com> 
	<CAHtxdDeBST6kmJeEn=VN_TzDdCv1rppT9g5qxX4bcyEq-Hcn+g@mail.gmail.com>
	<55A60B73.7070506@dwide.com>
	<1B5A6695-D5C9-49F3-A5A4-0384289586A2@gmail.com>
	<55A62C17.9020300@dwide.com> <85386622.20150715144307@seznam.cz>
	<CAEwTsAiqUgAiy+bW7vgOAHFa84rtOFbMbKotf05_FOCgaD4YOA@mail.gmail.com>
	<1404718420.20150715164202@seznam.cz>
	<22A9EB90-4505-4382-878E-ABA9AFD4072A@jerris.com>
Message-ID: <871415788.20150716112208@seznam.cz>

Dobr? den,

Yes exactly! I did not notice. Reduction of this setting helped me a lot. Thanks.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

st?eda 15. ?ervence 2015, 18:19:14, napsal jste:


do you have anything like 

<X-PRE-PROCESS cmd="set" data="answer_delay=4000"/>

in your vars.xml

On Jul 15, 2015, at 10:42 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:

Re: [Freeswitch-users] Reduce connection time when video calling 
Dobr? den,

How can I check it? I do not have in dialplan delay-time 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

st?eda 15. ?ervence 2015, 16:26:47, napsal jste:


sounds like answer delay is configured

On Wednesday, July 15, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi all,

The question in the following. How it is possible to reduce the connection time with the customer video calls? If you do directly with "bypass_media = true", then both callers begin to see each other almost instantaneously. Through FreeSwitch this connection can take from 5 to 10 seconds. It can somehow reduce?

Thanks

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
_________________________________________________________________________
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 andrew.keil at visytel.com  Thu Jul 16 15:25:44 2015
From: andrew.keil at visytel.com (Andrew Keil)
Date: Thu, 16 Jul 2015 11:25:44 +0000
Subject: [Freeswitch-users] Re- Connecting into British Telecom (BTs) SIP
	sandbox test platform
Message-ID: <SIXPR04MB1267941CA3860C8DE563226F3990@SIXPR04MB126.apcprd04.prod.outlook.com>

To FreeSWITCH users,

Just a quick question that I hope someone can help me with.  I am sure I can solve this myself when BT sort out their firewall, however while I wait 1-2 days I thought I would post this question here.

Currently BT have provided me with two SIP IPs and MEDIA IPs for their Sandbox:

BT SBC1 IP: Signalling xxx.xxx.xxx.26 (Media xxx.xxx.xxx.25)
BT SBC2 IP: Signalling xxx.xxx.xxx.23 (Media xxx.xxx.xxx.22)

Plus a SIP Trunk Number Range: 0551xxxx100 - 0551xxxx109
Model PSTN numbers: 0207xxxx713 and 0207xxxx714

First local CentOS 6.6 FreeSWITCH server (version 1.4.20) has two NICs (eth0: 192.168.2.10 {with public IP address mapped to this IP (xxx.xxx.xxx.67)} ; eth1: 192.168.2.11 (used internally only)}
Second local CentOS 6.6 FreeSWITCH server (version 1.4.20) has two NICs (eth0: 192.168.2.12 {with public IP address mapped to this IP (xxx.xxx.xxx.68)} ; eth1: 192.168.2.13 (used internally only)}

They have stated to me that they do not require and registration or authentication.

Unfortunately when I looked at all the samples on the FreeSWITCH Confluence they all seem to mention registering and sip proxy etc....

Can someone provide a simple .xml file(s) example for the above (even if it is to only one SBC) to fast-track my setup.  Any other hints or tips (or links to freeswitch.org Docs) I would appreciate.

One more thing to note, I am connecting FreeSWITCH via a local Firewall to BT's SBC's listed above, there is no SBC on my side since currently it is not needed based on my client's current requirements.  Hopefully this makes everything a little simpler, however I do understand that this configuration maybe less flexible (especially when commissioning more FreeSWITCH servers).

Thanks in advance,

Andrew Keil
Visytel Pty Ltd

PS. Since I am currently undertaking BT's testing on behalf of my client in the UK, I will provide back to the FreeSWITCH community the completed  BT Interoperability SIP test cases with the associated FreeSWITCH dialplans used and any comments.  Hopefully this will help any other company to pass BT's SIP testing requirements in a fast manner :)

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From mbsip at gazeta.pl  Thu Jul 16 15:30:32 2015
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Thu, 16 Jul 2015 13:30:32 +0200
Subject: [Freeswitch-users] how to record SIP+RTP sessions
Message-ID: <CACynCY4ufxgQ=xFJ+KcC9WNnZn5Qq_TqbrpuqtLQsmZmrede2g@mail.gmail.com>

Hi,

I need some help in finding tool to sniff SIP+RTP packets and separate each
session.
Following the wiki page  https://wiki.freeswitch.org/wiki/Packet_Capture
i've installed pcapsipdump 0.2 rel.
Unfortunately it utilizes just one core to maximum and all .pcap files are
quality affected.

As i assume this is the limitation of this app and there is nothing i can
do to overcome this.
If possitive, could you please point me to some similiar free app.

Thanks
Maciej.
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From mashudi72 at gmail.com  Thu Jul 16 15:40:56 2015
From: mashudi72 at gmail.com (mashudi72 -)
Date: Thu, 16 Jul 2015 18:40:56 +0700
Subject: [Freeswitch-users] Re- Connecting into British Telecom (BTs)
 SIP sandbox test platform
In-Reply-To: <SIXPR04MB1267941CA3860C8DE563226F3990@SIXPR04MB126.apcprd04.prod.outlook.com>
References: <SIXPR04MB1267941CA3860C8DE563226F3990@SIXPR04MB126.apcprd04.prod.outlook.com>
Message-ID: <CAGpNWUkHBv1HaYCUJGRevgqRt6gTbO+PjevVrNAEApELA5yv-A@mail.gmail.com>

Hi Andrew Keil,

Have a look to sip trunk configuration example, this will help.

Best regards.

Mashudi
 Pada 16 Jul 2015 18:29, "Andrew Keil" <andrew.keil at visytel.com> menulis:

>  To FreeSWITCH users,
>
>
>
> Just a quick question that I hope someone can help me with.  I am sure I
> can solve this myself when BT sort out their firewall, however while I wait
> 1-2 days I thought I would post this question here.
>
>
>
> *Currently BT have provided me with two SIP IPs and MEDIA IPs for their
> Sandbox:*
>
>
>
> BT SBC1 IP: Signalling xxx.xxx.xxx.26 (Media xxx.xxx.xxx.25)
>
> BT SBC2 IP: Signalling xxx.xxx.xxx.23 (Media xxx.xxx.xxx.22)
>
>
>
> Plus a SIP Trunk Number Range: 0551xxxx100 ? 0551xxxx109
>
> Model PSTN numbers: 0207xxxx713 and 0207xxxx714
>
>
>
> First local CentOS 6.6 FreeSWITCH server (version 1.4.20) has two NICs
> (eth0: 192.168.2.10 {with public IP address mapped to this IP
> (xxx.xxx.xxx.67)} ; eth1: 192.168.2.11 (used internally only)}
>
> Second local CentOS 6.6 FreeSWITCH server (version 1.4.20) has two NICs
> (eth0: 192.168.2.12 {with public IP address mapped to this IP
> (xxx.xxx.xxx.68)} ; eth1: 192.168.2.13 (used internally only)}
>
>
>
> They have stated to me that they do not require and registration or
> authentication.
>
>
>
> Unfortunately when I looked at all the samples on the FreeSWITCH
> Confluence they all seem to mention registering and sip proxy etc?.
>
>
>
> Can someone provide a simple .xml file(s) example for the above (even if
> it is to only one SBC) to fast-track my setup.  Any other hints or tips (or
> links to freeswitch.org Docs) I would appreciate.
>
>
>
> One more thing to note, I am connecting FreeSWITCH via a local Firewall to
> BT?s SBC?s listed above, there is no SBC on my side since currently it is
> not needed based on my client?s current requirements.  Hopefully this makes
> everything a little simpler, however I do understand that this
> configuration maybe less flexible (especially when commissioning more
> FreeSWITCH servers).
>
>
>
> Thanks in advance,
>
>
>
> Andrew Keil
>
> *Visytel Pty Ltd*
>
>
>
> PS. Since I am currently undertaking BT?s testing on behalf of my client
> in the UK, I will provide back to the FreeSWITCH community the completed
>  BT Interoperability SIP test cases with the associated FreeSWITCH
> dialplans used and any comments.  Hopefully this will help any other
> company to pass BT?s SIP testing requirements in a fast manner J
>
>
>
> _________________________________________________________________________
> 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 s.safarov at gmail.com  Thu Jul 16 15:43:42 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 16 Jul 2015 11:43:42 +0000
Subject: [Freeswitch-users] how to record SIP+RTP sessions
In-Reply-To: <CACynCY4ufxgQ=xFJ+KcC9WNnZn5Qq_TqbrpuqtLQsmZmrede2g@mail.gmail.com>
References: <CACynCY4ufxgQ=xFJ+KcC9WNnZn5Qq_TqbrpuqtLQsmZmrede2g@mail.gmail.com>
Message-ID: <CAHtxdDc1BeWad9cTeX69cXrfK+1xTWJid8PzYc1SpwwzaMA3Zg@mail.gmail.com>

I use wireshark, to analyze sip messages and RTP media

On Thu, Jul 16, 2015, 14:31 Maciej Bylica <mbsip at gazeta.pl> wrote:

> Hi,
>
> I need some help in finding tool to sniff SIP+RTP packets and separate
> each session.
> Following the wiki page  https://wiki.freeswitch.org/wiki/Packet_Capture
> i've installed pcapsipdump 0.2 rel.
> Unfortunately it utilizes just one core to maximum and all .pcap files are
> quality affected.
>
> As i assume this is the limitation of this app and there is nothing i can
> do to overcome this.
> If possitive, could you please point me to some similiar free app.
>
> Thanks
> Maciej.
>
>
>
> _________________________________________________________________________
> 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 italorossib at gmail.com  Thu Jul 16 16:28:02 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 16 Jul 2015 09:28:02 -0300
Subject: [Freeswitch-users] how to record SIP+RTP sessions
In-Reply-To: <CAHtxdDc1BeWad9cTeX69cXrfK+1xTWJid8PzYc1SpwwzaMA3Zg@mail.gmail.com>
References: <CACynCY4ufxgQ=xFJ+KcC9WNnZn5Qq_TqbrpuqtLQsmZmrede2g@mail.gmail.com>
	<CAHtxdDc1BeWad9cTeX69cXrfK+1xTWJid8PzYc1SpwwzaMA3Zg@mail.gmail.com>
Message-ID: <CAJR7f2HuLMY5ZtH8eHUADzaT-nq=w7LnZLEHUuct7MZHkhuq4A@mail.gmail.com>

http://www.sipcapture.org/

On Thu, Jul 16, 2015 at 8:43 AM, Sergey Safarov <s.safarov at gmail.com> wrote:

> I use wireshark, to analyze sip messages and RTP media
>
> On Thu, Jul 16, 2015, 14:31 Maciej Bylica <mbsip at gazeta.pl> wrote:
>
>> Hi,
>>
>> I need some help in finding tool to sniff SIP+RTP packets and separate
>> each session.
>> Following the wiki page  https://wiki.freeswitch.org/wiki/Packet_Capture
>> i've installed pcapsipdump 0.2 rel.
>> Unfortunately it utilizes just one core to maximum and all .pcap files
>> are quality affected.
>>
>> As i assume this is the limitation of this app and there is nothing i can
>> do to overcome this.
>> If possitive, could you please point me to some similiar free app.
>>
>> Thanks
>> Maciej.
>>
>>
>>
>> _________________________________________________________________________
>> 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
>



-- 
?talo Rossi
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From italorossib at gmail.com  Thu Jul 16 16:32:12 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 16 Jul 2015 09:32:12 -0300
Subject: [Freeswitch-users] Reduce connection time when video calling
In-Reply-To: <871415788.20150716112208@seznam.cz>
References: <55A5FC8E.1070901@dwide.com>
	<CAHtxdDeBST6kmJeEn=VN_TzDdCv1rppT9g5qxX4bcyEq-Hcn+g@mail.gmail.com>
	<55A60B73.7070506@dwide.com>
	<1B5A6695-D5C9-49F3-A5A4-0384289586A2@gmail.com>
	<55A62C17.9020300@dwide.com> <85386622.20150715144307@seznam.cz>
	<CAEwTsAiqUgAiy+bW7vgOAHFa84rtOFbMbKotf05_FOCgaD4YOA@mail.gmail.com>
	<1404718420.20150715164202@seznam.cz>
	<22A9EB90-4505-4382-878E-ABA9AFD4072A@jerris.com>
	<871415788.20150716112208@seznam.cz>
Message-ID: <CAJR7f2FQj5Na6iHUnNiEno_41TCzG34ra-2FmufiyhbVwBhxEA@mail.gmail.com>

This is in the vanilla config, not sure if it's better to reduce the value
or just remove it. Thoughts?

On Thu, Jul 16, 2015 at 6:22 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Dobr? den,
>
> Yes exactly! I did not notice. Reduction of this setting helped me a lot.
> Thanks.
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 st?eda 15. ?ervence 2015, 18:19:14, napsal
> jste: *
>  do you have anything like
>
> <X-PRE-PROCESS cmd="set" data="answer_delay=4000"/>
>
> in your vars.xml
>
> On Jul 15, 2015, at 10:42 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
>
> Re: [Freeswitch-users] Reduce connection time when video calling
> Dobr? den,
>
> How can I check it? I do not have in dialplan *delay*-
>
> *time *
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 st?eda 15. ?ervence 2015, 16:26:47, napsal
> jste: *
>
>  sounds like answer delay is configured
>
> On Wednesday, July 15, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Hi all,
>
> The question in the following. How it is possible to reduce the connection
> time with the customer video calls? If you do directly with "bypass_media =
> true", then both callers begin to see each other almost instantaneously.
> Through FreeSwitch this connection can take from 5 to 10 seconds. It can
> somehow reduce?
>
> Thanks
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel*.
> 775-415-382*
> _________________________________________________________________________
> 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
>



-- 
?talo Rossi
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From yadenis at seznam.cz  Thu Jul 16 16:43:07 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Thu, 16 Jul 2015 14:43:07 +0200
Subject: [Freeswitch-users] Reduce connection time when video calling
In-Reply-To: <CAJR7f2FQj5Na6iHUnNiEno_41TCzG34ra-2FmufiyhbVwBhxEA@mail.gmail.com>
References: <55A5FC8E.1070901@dwide.com> 
	<CAHtxdDeBST6kmJeEn=VN_TzDdCv1rppT9g5qxX4bcyEq-Hcn+g@mail.gmail.com>
	<55A60B73.7070506@dwide.com>
	<1B5A6695-D5C9-49F3-A5A4-0384289586A2@gmail.com>
	<55A62C17.9020300@dwide.com> <85386622.20150715144307@seznam.cz>
	<CAEwTsAiqUgAiy+bW7vgOAHFa84rtOFbMbKotf05_FOCgaD4YOA@mail.gmail.com>
	<1404718420.20150715164202@seznam.cz>
	<22A9EB90-4505-4382-878E-ABA9AFD4072A@jerris.com>
	<871415788.20150716112208@seznam.cz>
	<CAJR7f2FQj5Na6iHUnNiEno_41TCzG34ra-2FmufiyhbVwBhxEA@mail.gmail.com>
Message-ID: <1251683864.20150716144307@seznam.cz>

Dobr? den,

When I removed this value at all, nothing has changed. 

By this I reduced to 500 and the connection speed is comparable with the included "bypass_media = true". 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

?tvrtek 16. ?ervence 2015, 14:32:12, napsal jste:


This is in the vanilla config, not sure if it's better to reduce the value or just remove it. Thoughts?

On Thu, Jul 16, 2015 at 6:22 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
Dobr? den,

Yes exactly! I did not notice. Reduction of this setting helped me a lot. Thanks.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

st?eda 15. ?ervence 2015, 18:19:14, napsal jste:


do you have anything like 

<X-PRE-PROCESS cmd="set" data="answer_delay=4000"/>

in your vars.xml

On Jul 15, 2015, at 10:42 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:

Re: [Freeswitch-users] Reduce connection time when video calling 
Dobr? den,

How can I check it? I do not have in dialplan delay-time 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

st?eda 15. ?ervence 2015, 16:26:47, napsal jste:


sounds like answer delay is configured

On Wednesday, July 15, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi all,

The question in the following. How it is possible to reduce the connection time with the customer video calls? If you do directly with "bypass_media = true", then both callers begin to see each other almost instantaneously. Through FreeSwitch this connection can take from 5 to 10 seconds. It can somehow reduce?

Thanks

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
_________________________________________________________________________
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



-- 
?talo Rossi
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From krice at freeswitch.org  Thu Jul 16 18:02:36 2015
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 16 Jul 2015 09:02:36 -0500
Subject: [Freeswitch-users] GIT is down
In-Reply-To: <55A76137.1010304@dwide.com>
Message-ID: <D1CD23AC.A26E4%krice@freeswitch.org>

The git protocol was abandoned a couple of years ago for FreeSWITCH.

Where did you get this URL?


On 7/16/15, 2:45 AM, "Dmitry Mordovin" <dm at dwide.com> wrote:

> Hello!
> 
> GIT is down
> 
> git clone git://git.freeswitch.org/freeswitch.git
> Cloning into 'freeswitch'...
> fatal: unable to connect to git.freeswitch.org:
> git.freeswitch.org[0: 209.105.235.6]: errno=Connection refused
> 
> 
> 
> _________________________________________________________________________
> 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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH





From mike at jerris.com  Thu Jul 16 18:43:00 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 16 Jul 2015 10:43:00 -0400
Subject: [Freeswitch-users] verto.conf.xml parameter for public access
In-Reply-To: <CAHKqMb6=Hn4ua8FJL-nvD4vUvHEnY8GaGtt_PkhFcUD1_uYguA@mail.gmail.com>
References: <CAHKqMb5s9NGtaiEHO5Z5uWDi=gCdeTEmsPBqU6dLRwHc4r=-LQ@mail.gmail.com>
	<CAHKqMb6=Hn4ua8FJL-nvD4vUvHEnY8GaGtt_PkhFcUD1_uYguA@mail.gmail.com>
Message-ID: <CAEwTsAiiesrOWC8Z8RhGwswwwkYJ0yHmYL+sy_hx0BGqehFvyQ@mail.gmail.com>

Is the wan IP address bound to your interface or is nat mapping it to your
internal address.  The bind url is the actual interface address on your
system

On Thursday, July 16, 2015, Nandy Dagondon <nandy1925 at gmail.com> wrote:

> Hi to everyone,
>
> More related question.
>
> I see this line when I execute on fs_cli:
>
> WSS-BIND-URL      sips:mod_sofia at 192.168.0.5:7443;transport=wss
>
> It's my local IP address.  Shouldn't it be my public WAN IP address?
>
> Please shed some light.  Thanks.
>
> /Nandy
>
> On Wed, Jul 15, 2015 at 9:53 AM, Nandy Dagondon <nandy1925 at gmail.com
> <javascript:_e(%7B%7D,'cvml','nandy1925 at gmail.com');>> wrote:
>
>> I succesfully installed mod_verto in my intranet.  FS is behind NAT. Next
>> step, I'm setting up where remote web clients can login.  I setup my router
>> DMZ to FS.
>>
>> This parameter is commented out  "ext-rtp-ip" in the Wiki. Is this
>> parameter necessary for accepting remote clients?
>>
>
>
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From mike at jerris.com  Thu Jul 16 18:46:43 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 16 Jul 2015 10:46:43 -0400
Subject: [Freeswitch-users] Reduce connection time when video calling
In-Reply-To: <CAJR7f2FQj5Na6iHUnNiEno_41TCzG34ra-2FmufiyhbVwBhxEA@mail.gmail.com>
References: <55A5FC8E.1070901@dwide.com>
	<CAHtxdDeBST6kmJeEn=VN_TzDdCv1rppT9g5qxX4bcyEq-Hcn+g@mail.gmail.com>
	<55A60B73.7070506@dwide.com>
	<1B5A6695-D5C9-49F3-A5A4-0384289586A2@gmail.com>
	<55A62C17.9020300@dwide.com> <85386622.20150715144307@seznam.cz>
	<CAEwTsAiqUgAiy+bW7vgOAHFa84rtOFbMbKotf05_FOCgaD4YOA@mail.gmail.com>
	<1404718420.20150715164202@seznam.cz>
	<22A9EB90-4505-4382-878E-ABA9AFD4072A@jerris.com>
	<871415788.20150716112208@seznam.cz>
	<CAJR7f2FQj5Na6iHUnNiEno_41TCzG34ra-2FmufiyhbVwBhxEA@mail.gmail.com>
Message-ID: <CAEwTsAjtKKabRS=7QxrDnW+-Yf888xEki_Lo7OR9HB2pCgR+RA@mail.gmail.com>

I think we removed it from default config

On Thursday, July 16, 2015, ?talo Rossi <italorossib at gmail.com> wrote:

> This is in the vanilla config, not sure if it's better to reduce the value
> or just remove it. Thoughts?
>
> On Thu, Jul 16, 2015 at 6:22 AM, Denis Jakovlev <yadenis at seznam.cz
> <javascript:_e(%7B%7D,'cvml','yadenis at seznam.cz');>> wrote:
>
>>  Dobr? den,
>>
>> Yes exactly! I did not notice. Reduction of this setting helped me a lot.
>> Thanks.
>>
>>
>>
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>> <http://mob.tel>. 775-415-382 st?eda 15. ?ervence 2015, 18:19:14, napsal
>> jste: *
>>  do you have anything like
>>
>> <X-PRE-PROCESS cmd="set" data="answer_delay=4000"/>
>>
>> in your vars.xml
>>
>> On Jul 15, 2015, at 10:42 AM, Denis Jakovlev <yadenis at seznam.cz
>> <javascript:_e(%7B%7D,'cvml','yadenis at seznam.cz');>> wrote:
>>
>> Re: [Freeswitch-users] Reduce connection time when video calling
>> Dobr? den,
>>
>> How can I check it? I do not have in dialplan *delay*-
>>
>> *time *
>>
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>> <http://mob.tel>. 775-415-382 st?eda 15. ?ervence 2015, 16:26:47, napsal
>> jste: *
>>
>>  sounds like answer delay is configured
>>
>> On Wednesday, July 15, 2015, Denis Jakovlev <yadenis at seznam.cz
>> <javascript:_e(%7B%7D,'cvml','yadenis at seznam.cz');>> wrote:
>> Hi all,
>>
>> The question in the following. How it is possible to reduce the
>> connection time with the customer video calls? If you do directly with
>> "bypass_media = true", then both callers begin to see each other almost
>> instantaneously. Through FreeSwitch this connection can take from 5 to 10
>> seconds. It can somehow reduce?
>>
>> Thanks
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel*.
>> 775-415-382*
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','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
>> <javascript:_e(%7B%7D,'cvml','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
>> <javascript:_e(%7B%7D,'cvml','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
>> <javascript:_e(%7B%7D,'cvml','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
>>
>
>
>
> --
> ?talo Rossi
>
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From techfaltu at gmail.com  Thu Jul 16 18:52:42 2015
From: techfaltu at gmail.com (Niraj Roy)
Date: Thu, 16 Jul 2015 20:22:42 +0530
Subject: [Freeswitch-users] Issues with simultaneous call hangup
In-Reply-To: <CAEwTsAjP46mxk-FouXSikQKv+dWdd1iah7=-Vwy=ebDyXS9ZwA@mail.gmail.com>
References: <CAO2xAXM3KW7u4t=c4gK6Gt8OXywe7ZkCk+svbQ63K=xS-my2=A@mail.gmail.com>
	<CAEwTsAjP46mxk-FouXSikQKv+dWdd1iah7=-Vwy=ebDyXS9ZwA@mail.gmail.com>
Message-ID: <CAO2xAXMiicEbs0C0K9u01u=ho26TDYZvspK6qLNN64jbMEVUMA@mail.gmail.com>

Hi Michel,
   Thanks for your reply. I have tested with 4.1.20 and found the same
result.
After digging up some more I found that on end call handler if we perform
any lightweight activity like printing a log, it works fine. But if we
perform a bit heavy activity like calling a curl for CDR, it gets exhausted
and not able to execute saying 'channel is hung up already'.
  I think this is a very common phenomena and should have occur any and
everyone in the said case. I am not an expert in FS hoping of missing
something which somebody can rightly point out here.

Thanks,
Niraj


On Tue, Jul 14, 2015 at 9:12 PM, Michael Jerris <mike at jerris.com> wrote:

> Does this happen in the latest 1.4.20 release
>
>
> On Tuesday, July 14, 2015, Niraj Roy <techfaltu at gmail.com> wrote:
>
>> Hello Gurus,
>>     We are using FreeSWITCH Version 1.4.18~64bit. We observed when both
>> caller and caller hangs up the call simultaneously exec_after_bridge_app
>> function does not execute for one of the legs.
>>     Here is the segment of executing the xml file.
>>                 <action application="set"
>> data="exec_after_bridge_app=transfer"/>
>>                 <action application="set"
>> data="exec_after_bridge_arg=endcall-handler XML CALLENDING"/>
>>                 <action application="set_zombie_exec"/>
>>                 <action application="set" data="continue_on_fail=true"/>
>>                 <action inline="true" application="set"
>> data="hangup_after_bridge=false"/>
>>
>> For normal or sequential call clearing everything works fine, but somehow
>> this creates a race condition and state mismatch in the freeswitch which I
>> don't know.
>> Can anybody shed some light on this?
>>
>> Thanks,
>>
>>
> _________________________________________________________________________
> 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 dm at dwide.com  Thu Jul 16 19:19:58 2015
From: dm at dwide.com (Dmitry Mordovin)
Date: Thu, 16 Jul 2015 19:19:58 +0400
Subject: [Freeswitch-users] GIT is down
In-Reply-To: <D1CD23AC.A26E4%krice@freeswitch.org>
References: <D1CD23AC.A26E4%krice@freeswitch.org>
Message-ID: <55A7CB9E.1020103@dwide.com>

 From http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide

My mistake - I not read "don't use instructions"

And on https://freeswitch.org/confluence/display/FREESWITCH/Installation

I find GIT instructions just on 3-4 visit, they under Debian 7 make 
script link
https://freeswitch.org/confluence/display/FREESWITCH/Debian+7



On 07/16/2015 06:02 PM, Ken Rice wrote:
> The git protocol was abandoned a couple of years ago for FreeSWITCH.
>
> Where did you get this URL?
>
>
> On 7/16/15, 2:45 AM, "Dmitry Mordovin" <dm at dwide.com> wrote:
>
>> Hello!
>>
>> GIT is down
>>
>> git clone git://git.freeswitch.org/freeswitch.git
>> Cloning into 'freeswitch'...
>> fatal: unable to connect to git.freeswitch.org:
>> git.freeswitch.org[0: 209.105.235.6]: errno=Connection refused
>>
>>
>>
>> _________________________________________________________________________
>> 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 mbsip at gazeta.pl  Thu Jul 16 19:49:34 2015
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Thu, 16 Jul 2015 17:49:34 +0200
Subject: [Freeswitch-users] how to record SIP+RTP sessions
In-Reply-To: <CAJR7f2HuLMY5ZtH8eHUADzaT-nq=w7LnZLEHUuct7MZHkhuq4A@mail.gmail.com>
References: <CACynCY4ufxgQ=xFJ+KcC9WNnZn5Qq_TqbrpuqtLQsmZmrede2g@mail.gmail.com>
	<CAHtxdDc1BeWad9cTeX69cXrfK+1xTWJid8PzYc1SpwwzaMA3Zg@mail.gmail.com>
	<CAJR7f2HuLMY5ZtH8eHUADzaT-nq=w7LnZLEHUuct7MZHkhuq4A@mail.gmail.com>
Message-ID: <CACynCY4PBkgEicoTrnYEZL=3m5PkuetcJVYZXa=Ys_BApt_w8g@mail.gmail.com>

Thanks for your replies..

Yes i have Homer already installed but is does not store RTP packets.
Regarding wireshark... yes but in case of heavy traffic it takes two mins
to fill in 100Meg file and quite often it is not enough to store all the
conversation.
Thats why i am looking for sth that could store SIP+RTP data as well as
create separate files for each session.

Thanks.
Maciej

2015-07-16 14:28 GMT+02:00 ?talo Rossi <italorossib at gmail.com>:

> http://www.sipcapture.org/
>
> On Thu, Jul 16, 2015 at 8:43 AM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> I use wireshark, to analyze sip messages and RTP media
>>
>> On Thu, Jul 16, 2015, 14:31 Maciej Bylica <mbsip at gazeta.pl> wrote:
>>
>>> Hi,
>>>
>>> I need some help in finding tool to sniff SIP+RTP packets and separate
>>> each session.
>>> Following the wiki page  https://wiki.freeswitch.org/wiki/Packet_Capture
>>> i've installed pcapsipdump 0.2 rel.
>>> Unfortunately it utilizes just one core to maximum and all .pcap files
>>> are quality affected.
>>>
>>> As i assume this is the limitation of this app and there is nothing i
>>> can do to overcome this.
>>> If possitive, could you please point me to some similiar free app.
>>>
>>> Thanks
>>> Maciej.
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> 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
>>
>
>
>
> --
> ?talo Rossi
>
> _________________________________________________________________________
> 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 nneul at mst.edu  Thu Jul 16 19:53:51 2015
From: nneul at mst.edu (Nathan Neulinger)
Date: Thu, 16 Jul 2015 10:53:51 -0500
Subject: [Freeswitch-users] how to record SIP+RTP sessions
In-Reply-To: <CACynCY4PBkgEicoTrnYEZL=3m5PkuetcJVYZXa=Ys_BApt_w8g@mail.gmail.com>
References: <CACynCY4ufxgQ=xFJ+KcC9WNnZn5Qq_TqbrpuqtLQsmZmrede2g@mail.gmail.com>	<CAHtxdDc1BeWad9cTeX69cXrfK+1xTWJid8PzYc1SpwwzaMA3Zg@mail.gmail.com>	<CAJR7f2HuLMY5ZtH8eHUADzaT-nq=w7LnZLEHUuct7MZHkhuq4A@mail.gmail.com>
	<CACynCY4PBkgEicoTrnYEZL=3m5PkuetcJVYZXa=Ys_BApt_w8g@mail.gmail.com>
Message-ID: <55A7D38F.40703@mst.edu>

voipmonitor capture engine will do this quite nicely. Search list archives for some previous discussions on it.

-- Nathan

On 07/16/2015 10:49 AM, Maciej Bylica wrote:
> Thanks for your replies..
>
> Yes i have Homer already installed but is does not store RTP packets.
> Regarding wireshark... yes but in case of heavy traffic it takes two mins to fill in 100Meg file and quite often it is
> not enough to store all the conversation.
> Thats why i am looking for sth that could store SIP+RTP data as well as create separate files for each session.
>
> Thanks.
> Maciej
>
> 2015-07-16 14:28 GMT+02:00 ?talo Rossi <italorossib at gmail.com <mailto:italorossib at gmail.com>>:
>
>     http://www.sipcapture.org/
>
>     On Thu, Jul 16, 2015 at 8:43 AM, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
>
>         I use wireshark, to analyze sip messages and RTP media
>
>
>         On Thu, Jul 16, 2015, 14:31 Maciej Bylica <mbsip at gazeta.pl <mailto:mbsip at gazeta.pl>> wrote:
>
>             Hi,
>
>             I need some help in finding tool to sniff SIP+RTP packets and separate each session.
>             Following the wiki page https://wiki.freeswitch.org/wiki/Packet_Capture i've installed pcapsipdump 0.2 rel.
>             Unfortunately it utilizes just one core to maximum and all .pcap files are quality affected.
>
>             As i assume this is the limitation of this app and there is nothing i can do to overcome this.
>             If possitive, could you please point me to some similiar free app.
>
>             Thanks
>             Maciej.
>
>
>
>             _________________________________________________________________________
>             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
>
>
>
>
>     --
>     ?talo Rossi
>
>     _________________________________________________________________________
>     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
> 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 danny.gershman at gmail.com  Fri Jul 17 02:49:11 2015
From: danny.gershman at gmail.com (Danny Gershman)
Date: Thu, 16 Jul 2015 22:49:11 +0000
Subject: [Freeswitch-users] Audio lost in recording
Message-ID: <CAEorZykyBuaZjj8KgBK8f7reUJEJ5H8qV26_W=sOD_vMp6nDaA@mail.gmail.com>

In a recent conference recording we lost a lot of data.  My current
jitterbuffer setting is 60:600:60.

https://www.dropbox.com/s/rcw7fyr5pp73lgt/512d1b2e-1a51-48d6-b135-af36fca04603.pcap.zip?dl=0

I want to test how FreeSWITCH is handling the packets and try adjusting the
jitterbuffer accordingly.  Does anyone know how to replay a PCAP file to
FreeSWITCH easily?

Danny
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From david.witham at netsip.com.au  Fri Jul 17 02:50:30 2015
From: david.witham at netsip.com.au (David Witham)
Date: Thu, 16 Jul 2015 22:50:30 +0000
Subject: [Freeswitch-users] how to record SIP+RTP sessions
In-Reply-To: <55A7D38F.40703@mst.edu>
References: <CACynCY4ufxgQ=xFJ+KcC9WNnZn5Qq_TqbrpuqtLQsmZmrede2g@mail.gmail.com>
	<CAHtxdDc1BeWad9cTeX69cXrfK+1xTWJid8PzYc1SpwwzaMA3Zg@mail.gmail.com>
	<CAJR7f2HuLMY5ZtH8eHUADzaT-nq=w7LnZLEHUuct7MZHkhuq4A@mail.gmail.com>
	<CACynCY4PBkgEicoTrnYEZL=3m5PkuetcJVYZXa=Ys_BApt_w8g@mail.gmail.com>,
	<55A7D38F.40703@mst.edu>
Message-ID: <1437087030418.38416@netsip.com.au>

+1 for voipmonitor. Its invaluable for monitoring and diagnosing problems on our voice network.
________________________________________
From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Nathan Neulinger <nneul at mst.edu>
Sent: Friday, 17 July 2015 01:53
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] how to record SIP+RTP sessions

voipmonitor capture engine will do this quite nicely. Search list archives for some previous discussions on it.

-- Nathan

On 07/16/2015 10:49 AM, Maciej Bylica wrote:
> Thanks for your replies..
>
> Yes i have Homer already installed but is does not store RTP packets.
> Regarding wireshark... yes but in case of heavy traffic it takes two mins to fill in 100Meg file and quite often it is
> not enough to store all the conversation.
> Thats why i am looking for sth that could store SIP+RTP data as well as create separate files for each session.
>
> Thanks.
> Maciej
>
> 2015-07-16 14:28 GMT+02:00 ?talo Rossi <italorossib at gmail.com <mailto:italorossib at gmail.com>>:
>
>     http://www.sipcapture.org/
>
>     On Thu, Jul 16, 2015 at 8:43 AM, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
>
>         I use wireshark, to analyze sip messages and RTP media
>
>
>         On Thu, Jul 16, 2015, 14:31 Maciej Bylica <mbsip at gazeta.pl <mailto:mbsip at gazeta.pl>> wrote:
>
>             Hi,
>
>             I need some help in finding tool to sniff SIP+RTP packets and separate each session.
>             Following the wiki page https://wiki.freeswitch.org/wiki/Packet_Capture i've installed pcapsipdump 0.2 rel.
>             Unfortunately it utilizes just one core to maximum and all .pcap files are quality affected.
>
>             As i assume this is the limitation of this app and there is nothing i can do to overcome this.
>             If possitive, could you please point me to some similiar free app.
>
>             Thanks
>             Maciej.
>
>
>
>             _________________________________________________________________________
>             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
>
>
>
>
>     --
>     ?talo Rossi
>
>     _________________________________________________________________________
>     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
> 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

_________________________________________________________________________
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 govoiper at gmail.com  Fri Jul 17 02:57:51 2015
From: govoiper at gmail.com (SamyGo)
Date: Thu, 16 Jul 2015 18:57:51 -0400
Subject: [Freeswitch-users] Execute dialplan after valet_park
In-Reply-To: <CAO3hyzF0dsxCEopEOeONnNUneTpyRiwXnxSJVLiTnrOEKRVYHA@mail.gmail.com>
References: <CAJUJwtipc+yhpck+w4M4mYYPk0ZRa5NqWhytRChbsi4c6_nwdA@mail.gmail.com>
	<CAO3hyzE0joUxdFeQM4oJ9K64qdLcG_4uock8v3AqHDWc6pSdaw@mail.gmail.com>
	<CAJUJwtjdBerXshWVJhQ0KJ-M_chiAaNiXgGaEU4hs4_nN3b8+Q@mail.gmail.com>
	<CAO3hyzF0dsxCEopEOeONnNUneTpyRiwXnxSJVLiTnrOEKRVYHA@mail.gmail.com>
Message-ID: <CAJUJwti-xnSaKZf_W+9NTqCaaCU2W3f1R5zsWmVDFYQkpfhVDA@mail.gmail.com>

+1 Got it all working thanks again Chris.

For anyone else in future here is what I did.

Added this line in *conf/autoload_configs/lua.conf.xml*
<hook event="CUSTOM" subclass="valet_parking::info"
script="unparking_event.lua"/>

Restarted FS, and got this script executed on each of valet_parking sub
events "hold/exit/bridge"

In *unparking_event.lua* collect the event Headers:

local action = event:getHeader("Action")
local park_lot = event:getHeader("Valet-Extension")
...
if action == "exit" then
--  what I needed to do
end




On Wed, Jul 15, 2015 at 12:28 PM, Chris Tunbridge <blasterjr at gmail.com>
wrote:

> When you use a hook, whenever that event (CUSTOM) is called the lua script
> would be called, so its a seperate script, you then have to do conditions
> to check that the type is valet_parking::info and execute logic inside that
> condition
>
> On Wed, Jul 15, 2015 at 8:22 AM, SamyGo <govoiper at gmail.com> wrote:
>
>> Thanks Chris,
>>
>> That was my finding too, I just am not sure if I can monitor this event
>> from the main LUA script or need an independent lua script only for the
>> park "hangup" event monitoring. Any further help on this would be highly
>> appreciated.
>>
>> I wonder why the api_hangup hook won't trigger on the parked channel,
>> would've been much easier. Piece of cake only if we could execute further
>> dialplan application after valet_parking.
>>
>> BR,
>> Sammy
>> You might be able to use an event hook with lua, you can catch the CUSTOM
>> with a subclass of valet_parking::info, then you can check for the "action"
>> and it'll be "exit" when they leave (any kind of leave) then can likely
>> check for a disposition.  I'm not currently using this method, but its
>> someting we looked into for monitoring when people hangup from the parking
>> lot.
>>
>> On Tue, Jul 14, 2015 at 10:49 AM, SamyGo <govoiper at gmail.com> wrote:
>>
>>> Hi All,
>>>
>>> I need some help in finding  a way to execute some dialplan code after
>>> the parked user Hangups from the parking lot.
>>>
>>> I've a scenario where Parked party decides to hangup while listening to
>>> MOH while parked. The Dialplan just Hangsup right away and I have no
>>> control.
>>>
>>> I've tried using: *session_in_hangup_hook*, and
>>> *api_hangup_hook=myscript.lua *but they get executed for the parker
>>> just after they hear the Parked slot number.
>>>
>>> Then I've tried *nolocal:export* these two variables for the other leg
>>> but nothing happens.
>>>
>>> Kindly suggest what other option are there.
>>>
>>> Best Regards.
>>> Sammy
>>>
>>>
>>> _________________________________________________________________________
>>> 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 mike at jerris.com  Fri Jul 17 03:21:28 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 16 Jul 2015 19:21:28 -0400
Subject: [Freeswitch-users] Audio lost in recording
In-Reply-To: <CAEorZykyBuaZjj8KgBK8f7reUJEJ5H8qV26_W=sOD_vMp6nDaA@mail.gmail.com>
References: <CAEorZykyBuaZjj8KgBK8f7reUJEJ5H8qV26_W=sOD_vMp6nDaA@mail.gmail.com>
Message-ID: <9157B140-4914-4D07-86FF-AFF86962A3F1@jerris.com>

sipp has a feature to play a pcap of an rtp stream.  Thats what we have used in the past.  You need to build a sipp scenario that goes with the call and the right args to play back the pcap.


> On Jul 16, 2015, at 6:49 PM, Danny Gershman <danny.gershman at gmail.com> wrote:
> 
> In a recent conference recording we lost a lot of data.  My current jitterbuffer setting is 60:600:60.  
> 
> https://www.dropbox.com/s/rcw7fyr5pp73lgt/512d1b2e-1a51-48d6-b135-af36fca04603.pcap.zip?dl=0 <https://www.dropbox.com/s/rcw7fyr5pp73lgt/512d1b2e-1a51-48d6-b135-af36fca04603.pcap.zip?dl=0>
> 
> I want to test how FreeSWITCH is handling the packets and try adjusting the jitterbuffer accordingly.  Does anyone know how to replay a PCAP file to FreeSWITCH easily?
> 
> Danny

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From nandy1925 at gmail.com  Fri Jul 17 04:08:34 2015
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Fri, 17 Jul 2015 08:08:34 +0800
Subject: [Freeswitch-users] verto.conf.xml parameter for public access
In-Reply-To: <CAEwTsAiiesrOWC8Z8RhGwswwwkYJ0yHmYL+sy_hx0BGqehFvyQ@mail.gmail.com>
References: <CAHKqMb5s9NGtaiEHO5Z5uWDi=gCdeTEmsPBqU6dLRwHc4r=-LQ@mail.gmail.com>
	<CAHKqMb6=Hn4ua8FJL-nvD4vUvHEnY8GaGtt_PkhFcUD1_uYguA@mail.gmail.com>
	<CAEwTsAiiesrOWC8Z8RhGwswwwkYJ0yHmYL+sy_hx0BGqehFvyQ@mail.gmail.com>
Message-ID: <CAHKqMb4F+CWYtfsGHSycowmPQ9Fsz6FnYiKPzFqqpDtQA7Pisg@mail.gmail.com>

NAT map.  The router's DMZ point to this interface. Tks.  /Nandy


On Thu, Jul 16, 2015 at 10:43 PM, Michael Jerris <mike at jerris.com> wrote:

> Is the wan IP address bound to your interface or is nat mapping it to your
> internal address.  The bind url is the actual interface address on your
> system
>
> On Thursday, July 16, 2015, Nandy Dagondon <nandy1925 at gmail.com> wrote:
>
>> Hi to everyone,
>>
>> More related question.
>>
>> I see this line when I execute on fs_cli:
>>
>> WSS-BIND-URL      sips:mod_sofia at 192.168.0.5:7443;transport=wss
>>
>> It's my local IP address.  Shouldn't it be my public WAN IP address?
>>
>> Please shed some light.  Thanks.
>>
>> /Nandy
>>
>> On Wed, Jul 15, 2015 at 9:53 AM, Nandy Dagondon <nandy1925 at gmail.com>
>> wrote:
>>
>>> I succesfully installed mod_verto in my intranet.  FS is behind NAT.
>>> Next step, I'm setting up where remote web clients can login.  I setup my
>>> router DMZ to FS.
>>>
>>> This parameter is commented out  "ext-rtp-ip" in the Wiki. Is this
>>> parameter necessary for accepting remote clients?
>>>
>>
>>
> _________________________________________________________________________
> 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 stefan.kainz at 1012.at  Fri Jul 17 04:13:39 2015
From: stefan.kainz at 1012.at (Stefan Kainz)
Date: Fri, 17 Jul 2015 02:13:39 +0200
Subject: [Freeswitch-users] how to record SIP+RTP sessions
In-Reply-To: <1437087030418.38416@netsip.com.au>
References: <CACynCY4ufxgQ=xFJ+KcC9WNnZn5Qq_TqbrpuqtLQsmZmrede2g@mail.gmail.com>
	<CAHtxdDc1BeWad9cTeX69cXrfK+1xTWJid8PzYc1SpwwzaMA3Zg@mail.gmail.com>
	<CAJR7f2HuLMY5ZtH8eHUADzaT-nq=w7LnZLEHUuct7MZHkhuq4A@mail.gmail.com>
	<CACynCY4PBkgEicoTrnYEZL=3m5PkuetcJVYZXa=Ys_BApt_w8g@mail.gmail.com>
	<55A7D38F.40703@mst.edu> <1437087030418.38416@netsip.com.au>
Message-ID: <DA0FFF64-FF25-4ADA-92CB-E34AFA439DBD@1012.at>

+1 Voipmonitor.

Sent from my iPhone

> On 17 Jul 2015, at 00:50, David Witham <david.witham at netsip.com.au> wrote:
> 
> +1 for voipmonitor. Its invaluable for monitoring and diagnosing problems on our voice network.
> ________________________________________
> From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Nathan Neulinger <nneul at mst.edu>
> Sent: Friday, 17 July 2015 01:53
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] how to record SIP+RTP sessions
> 
> voipmonitor capture engine will do this quite nicely. Search list archives for some previous discussions on it.
> 
> -- Nathan
> 
>> On 07/16/2015 10:49 AM, Maciej Bylica wrote:
>> Thanks for your replies..
>> 
>> Yes i have Homer already installed but is does not store RTP packets.
>> Regarding wireshark... yes but in case of heavy traffic it takes two mins to fill in 100Meg file and quite often it is
>> not enough to store all the conversation.
>> Thats why i am looking for sth that could store SIP+RTP data as well as create separate files for each session.
>> 
>> Thanks.
>> Maciej
>> 
>> 2015-07-16 14:28 GMT+02:00 ?talo Rossi <italorossib at gmail.com <mailto:italorossib at gmail.com>>:
>> 
>>    http://www.sipcapture.org/
>> 
>>    On Thu, Jul 16, 2015 at 8:43 AM, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
>> 
>>        I use wireshark, to analyze sip messages and RTP media
>> 
>> 
>>        On Thu, Jul 16, 2015, 14:31 Maciej Bylica <mbsip at gazeta.pl <mailto:mbsip at gazeta.pl>> wrote:
>> 
>>            Hi,
>> 
>>            I need some help in finding tool to sniff SIP+RTP packets and separate each session.
>>            Following the wiki page https://wiki.freeswitch.org/wiki/Packet_Capture i've installed pcapsipdump 0.2 rel.
>>            Unfortunately it utilizes just one core to maximum and all .pcap files are quality affected.
>> 
>>            As i assume this is the limitation of this app and there is nothing i can do to overcome this.
>>            If possitive, could you please point me to some similiar free app.
>> 
>>            Thanks
>>            Maciej.
>> 
>> 
>> 
>>            _________________________________________________________________________
>>            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
>> 
>> 
>> 
>> 
>>    --
>>    ?talo Rossi
>> 
>>    _________________________________________________________________________
>>    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
>> 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
> 
> _________________________________________________________________________
> 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


From mike at jerris.com  Fri Jul 17 04:53:11 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 16 Jul 2015 20:53:11 -0400
Subject: [Freeswitch-users] verto.conf.xml parameter for public access
In-Reply-To: <CAHKqMb4F+CWYtfsGHSycowmPQ9Fsz6FnYiKPzFqqpDtQA7Pisg@mail.gmail.com>
References: <CAHKqMb5s9NGtaiEHO5Z5uWDi=gCdeTEmsPBqU6dLRwHc4r=-LQ@mail.gmail.com>
	<CAHKqMb6=Hn4ua8FJL-nvD4vUvHEnY8GaGtt_PkhFcUD1_uYguA@mail.gmail.com>
	<CAEwTsAiiesrOWC8Z8RhGwswwwkYJ0yHmYL+sy_hx0BGqehFvyQ@mail.gmail.com>
	<CAHKqMb4F+CWYtfsGHSycowmPQ9Fsz6FnYiKPzFqqpDtQA7Pisg@mail.gmail.com>
Message-ID: <CAEwTsAgKv59OnMAPD1c04_BHBQ7jpmsODXUq5-znmit_J-=m8w@mail.gmail.com>

then the bind url is right

On Thursday, July 16, 2015, Nandy Dagondon <nandy1925 at gmail.com> wrote:

> NAT map.  The router's DMZ point to this interface. Tks.  /Nandy
>
>
> On Thu, Jul 16, 2015 at 10:43 PM, Michael Jerris <mike at jerris.com
> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>> wrote:
>
>> Is the wan IP address bound to your interface or is nat mapping it to
>> your internal address.  The bind url is the actual interface address on
>> your system
>>
>> On Thursday, July 16, 2015, Nandy Dagondon <nandy1925 at gmail.com
>> <javascript:_e(%7B%7D,'cvml','nandy1925 at gmail.com');>> wrote:
>>
>>> Hi to everyone,
>>>
>>> More related question.
>>>
>>> I see this line when I execute on fs_cli:
>>>
>>> WSS-BIND-URL      sips:mod_sofia at 192.168.0.5:7443;transport=wss
>>>
>>> It's my local IP address.  Shouldn't it be my public WAN IP address?
>>>
>>> Please shed some light.  Thanks.
>>>
>>> /Nandy
>>>
>>> On Wed, Jul 15, 2015 at 9:53 AM, Nandy Dagondon <nandy1925 at gmail.com>
>>> wrote:
>>>
>>>> I succesfully installed mod_verto in my intranet.  FS is behind NAT.
>>>> Next step, I'm setting up where remote web clients can login.  I setup my
>>>> router DMZ to FS.
>>>>
>>>> This parameter is commented out  "ext-rtp-ip" in the Wiki. Is this
>>>> parameter necessary for accepting remote clients?
>>>>
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','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
>> <javascript:_e(%7B%7D,'cvml','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 ssinyagin at gmail.com  Fri Jul 17 04:57:52 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 17 Jul 2015 02:57:52 +0200
Subject: [Freeswitch-users] Audio lost in recording
In-Reply-To: <CAEorZykyBuaZjj8KgBK8f7reUJEJ5H8qV26_W=sOD_vMp6nDaA@mail.gmail.com>
References: <CAEorZykyBuaZjj8KgBK8f7reUJEJ5H8qV26_W=sOD_vMp6nDaA@mail.gmail.com>
Message-ID: <CACOYK=iFp9KjUC2i8Tk_WUfn6w98KaZ_o1uLpmYuRKRz68Cn1Q@mail.gmail.com>

Are you running an old release? There were some bugfixrs related to
recording on the past few months.

Wireshark can save a call payload from an rtp stream, but it loads the
whole capture in memory and may crash on big files.
 On Jul 17, 2015 12:50 AM, "Danny Gershman" <danny.gershman at gmail.com>
wrote:

> In a recent conference recording we lost a lot of data.  My current
> jitterbuffer setting is 60:600:60.
>
>
> https://www.dropbox.com/s/rcw7fyr5pp73lgt/512d1b2e-1a51-48d6-b135-af36fca04603.pcap.zip?dl=0
>
> I want to test how FreeSWITCH is handling the packets and try adjusting
> the jitterbuffer accordingly.  Does anyone know how to replay a PCAP file
> to FreeSWITCH easily?
>
> Danny
>
> _________________________________________________________________________
> 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 danny.gershman at gmail.com  Fri Jul 17 06:23:10 2015
From: danny.gershman at gmail.com (Danny Gershman)
Date: Fri, 17 Jul 2015 02:23:10 +0000
Subject: [Freeswitch-users] Audio lost in recording
In-Reply-To: <CACOYK=iFp9KjUC2i8Tk_WUfn6w98KaZ_o1uLpmYuRKRz68Cn1Q@mail.gmail.com>
References: <CAEorZykyBuaZjj8KgBK8f7reUJEJ5H8qV26_W=sOD_vMp6nDaA@mail.gmail.com>
	<CACOYK=iFp9KjUC2i8Tk_WUfn6w98KaZ_o1uLpmYuRKRz68Cn1Q@mail.gmail.com>
Message-ID: <CAEorZymW3aBTv2UUc9T+sxbNwEXWo=eOTWK=r31B0bqT036C=w@mail.gmail.com>

This was running on 1.4.19 release.  I'm familiar with stripped out the RTP
with Wireshark.  I've used SIPP before, I'll try that again.

On Thu, Jul 16, 2015 at 8:58 PM Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> Are you running an old release? There were some bugfixrs related to
> recording on the past few months.
>
> Wireshark can save a call payload from an rtp stream, but it loads the
> whole capture in memory and may crash on big files.
>  On Jul 17, 2015 12:50 AM, "Danny Gershman" <danny.gershman at gmail.com>
> wrote:
>
>> In a recent conference recording we lost a lot of data.  My current
>> jitterbuffer setting is 60:600:60.
>>
>>
>> https://www.dropbox.com/s/rcw7fyr5pp73lgt/512d1b2e-1a51-48d6-b135-af36fca04603.pcap.zip?dl=0
>>
>> I want to test how FreeSWITCH is handling the packets and try adjusting
>> the jitterbuffer accordingly.  Does anyone know how to replay a PCAP file
>> to FreeSWITCH easily?
>>
>> Danny
>>
>> _________________________________________________________________________
>> 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 anthony.minessale at gmail.com  Fri Jul 17 08:19:44 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 16 Jul 2015 23:19:44 -0500
Subject: [Freeswitch-users] Issues with simultaneous call hangup
In-Reply-To: <CAO2xAXMiicEbs0C0K9u01u=ho26TDYZvspK6qLNN64jbMEVUMA@mail.gmail.com>
References: <CAO2xAXM3KW7u4t=c4gK6Gt8OXywe7ZkCk+svbQ63K=xS-my2=A@mail.gmail.com>
	<CAEwTsAjP46mxk-FouXSikQKv+dWdd1iah7=-Vwy=ebDyXS9ZwA@mail.gmail.com>
	<CAO2xAXMiicEbs0C0K9u01u=ho26TDYZvspK6qLNN64jbMEVUMA@mail.gmail.com>
Message-ID: <CAKbxfG-eGdVXcVK_iOFrwPefjL+MqRek696ZLZhed1c7CSDUjA@mail.gmail.com>

What you are doing is illogical and by design the wrong approach.
exec_after_bridge does not work on hungup channels, plus zombie_exec only
works on a handful of apps.
When the leg is hungup it is dead.  You should not try to do things on a
leg after it hangs up.
There are countless events and cdr collecting modules and code hooks to
properly capture a hangup record.
This is not a bug or a race condition, its abuse of the way the sofftware
is designed.  Its like saying you want your hard drive to sync its data
whenever you unplug the power cord.



On Thu, Jul 16, 2015 at 9:52 AM, Niraj Roy <techfaltu at gmail.com> wrote:

> Hi Michel,
>    Thanks for your reply. I have tested with 4.1.20 and found the same
> result.
> After digging up some more I found that on end call handler if we perform
> any lightweight activity like printing a log, it works fine. But if we
> perform a bit heavy activity like calling a curl for CDR, it gets exhausted
> and not able to execute saying 'channel is hung up already'.
>   I think this is a very common phenomena and should have occur any and
> everyone in the said case. I am not an expert in FS hoping of missing
> something which somebody can rightly point out here.
>
> Thanks,
> Niraj
>
>
> On Tue, Jul 14, 2015 at 9:12 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> Does this happen in the latest 1.4.20 release
>>
>>
>> On Tuesday, July 14, 2015, Niraj Roy <techfaltu at gmail.com> wrote:
>>
>>> Hello Gurus,
>>>     We are using FreeSWITCH Version 1.4.18~64bit. We observed when both
>>> caller and caller hangs up the call simultaneously exec_after_bridge_app
>>> function does not execute for one of the legs.
>>>     Here is the segment of executing the xml file.
>>>                 <action application="set"
>>> data="exec_after_bridge_app=transfer"/>
>>>                 <action application="set"
>>> data="exec_after_bridge_arg=endcall-handler XML CALLENDING"/>
>>>                 <action application="set_zombie_exec"/>
>>>                 <action application="set" data="continue_on_fail=true"/>
>>>                 <action inline="true" application="set"
>>> data="hangup_after_bridge=false"/>
>>>
>>> For normal or sequential call clearing everything works fine, but
>>> somehow this creates a race condition and state mismatch in the freeswitch
>>> which I don't know.
>>> Can anybody shed some light on this?
>>>
>>> Thanks,
>>>
>>>
>> _________________________________________________________________________
>> 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
>



-- 
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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From nandy1925 at gmail.com  Fri Jul 17 09:09:17 2015
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Fri, 17 Jul 2015 13:09:17 +0800
Subject: [Freeswitch-users] verto.conf.xml parameter for public access
In-Reply-To: <CAEwTsAgKv59OnMAPD1c04_BHBQ7jpmsODXUq5-znmit_J-=m8w@mail.gmail.com>
References: <CAHKqMb5s9NGtaiEHO5Z5uWDi=gCdeTEmsPBqU6dLRwHc4r=-LQ@mail.gmail.com>
	<CAHKqMb6=Hn4ua8FJL-nvD4vUvHEnY8GaGtt_PkhFcUD1_uYguA@mail.gmail.com>
	<CAEwTsAiiesrOWC8Z8RhGwswwwkYJ0yHmYL+sy_hx0BGqehFvyQ@mail.gmail.com>
	<CAHKqMb4F+CWYtfsGHSycowmPQ9Fsz6FnYiKPzFqqpDtQA7Pisg@mail.gmail.com>
	<CAEwTsAgKv59OnMAPD1c04_BHBQ7jpmsODXUq5-znmit_J-=m8w@mail.gmail.com>
Message-ID: <CAHKqMb6XPd8nLcN2zMOR1MGpRrxxyDhdZGqYLLHi5my1d70_HQ@mail.gmail.com>

Hi Michael,

Will this setup (NAT map) work to accept public clients?
Back to my first question, is "ext-rtp-ip" needed for this NAT map setup?

Finally, I saw this error message in fs_cli when a public client (Chrome on
Android) connects:
WS SETUP FAILED [ ]

Can you give a hint where the problem lies?

Tks,
Nandy

On Fri, Jul 17, 2015 at 8:53 AM, Michael Jerris <mike at jerris.com> wrote:

> then the bind url is right
>
> On Thursday, July 16, 2015, Nandy Dagondon <nandy1925 at gmail.com> wrote:
>
>> NAT map.  The router's DMZ point to this interface. Tks.  /Nandy
>>
>>
>> On Thu, Jul 16, 2015 at 10:43 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> Is the wan IP address bound to your interface or is nat mapping it to
>>> your internal address.  The bind url is the actual interface address on
>>> your system
>>>
>>> On Thursday, July 16, 2015, Nandy Dagondon <nandy1925 at gmail.com> wrote:
>>>
>>>> Hi to everyone,
>>>>
>>>> More related question.
>>>>
>>>> I see this line when I execute on fs_cli:
>>>>
>>>> WSS-BIND-URL      sips:mod_sofia at 192.168.0.5:7443;transport=wss
>>>>
>>>> It's my local IP address.  Shouldn't it be my public WAN IP address?
>>>>
>>>> Please shed some light.  Thanks.
>>>>
>>>> /Nandy
>>>>
>>>> On Wed, Jul 15, 2015 at 9:53 AM, Nandy Dagondon <nandy1925 at gmail.com>
>>>> wrote:
>>>>
>>>>> I succesfully installed mod_verto in my intranet.  FS is behind NAT.
>>>>> Next step, I'm setting up where remote web clients can login.  I setup my
>>>>> router DMZ to FS.
>>>>>
>>>>> This parameter is commented out  "ext-rtp-ip" in the Wiki. Is this
>>>>> parameter necessary for accepting remote clients?
>>>>>
>>>>
>>>>
>>> _________________________________________________________________________
>>> 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 techfaltu at gmail.com  Fri Jul 17 09:44:41 2015
From: techfaltu at gmail.com (Niraj Roy)
Date: Fri, 17 Jul 2015 11:14:41 +0530
Subject: [Freeswitch-users] Issues with simultaneous call hangup
In-Reply-To: <CAKbxfG-eGdVXcVK_iOFrwPefjL+MqRek696ZLZhed1c7CSDUjA@mail.gmail.com>
References: <CAO2xAXM3KW7u4t=c4gK6Gt8OXywe7ZkCk+svbQ63K=xS-my2=A@mail.gmail.com>
	<CAEwTsAjP46mxk-FouXSikQKv+dWdd1iah7=-Vwy=ebDyXS9ZwA@mail.gmail.com>
	<CAO2xAXMiicEbs0C0K9u01u=ho26TDYZvspK6qLNN64jbMEVUMA@mail.gmail.com>
	<CAKbxfG-eGdVXcVK_iOFrwPefjL+MqRek696ZLZhed1c7CSDUjA@mail.gmail.com>
Message-ID: <CAO2xAXP=OxXnj0S-sN6aTHb=rYPJBVSYU=OYmhq-ECxLvUrEpg@mail.gmail.com>

Thank you Anthony !!
I appreciate your reply that has cleared my doubt and provided me the
needed direction.




On Fri, Jul 17, 2015 at 9:49 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> What you are doing is illogical and by design the wrong approach.
> exec_after_bridge does not work on hungup channels, plus zombie_exec only
> works on a handful of apps.
> When the leg is hungup it is dead.  You should not try to do things on a
> leg after it hangs up.
> There are countless events and cdr collecting modules and code hooks to
> properly capture a hangup record.
> This is not a bug or a race condition, its abuse of the way the sofftware
> is designed.  Its like saying you want your hard drive to sync its data
> whenever you unplug the power cord.
>
>
>
> On Thu, Jul 16, 2015 at 9:52 AM, Niraj Roy <techfaltu at gmail.com> wrote:
>
>> Hi Michel,
>>    Thanks for your reply. I have tested with 4.1.20 and found the same
>> result.
>> After digging up some more I found that on end call handler if we perform
>> any lightweight activity like printing a log, it works fine. But if we
>> perform a bit heavy activity like calling a curl for CDR, it gets exhausted
>> and not able to execute saying 'channel is hung up already'.
>>   I think this is a very common phenomena and should have occur any and
>> everyone in the said case. I am not an expert in FS hoping of missing
>> something which somebody can rightly point out here.
>>
>> Thanks,
>> Niraj
>>
>>
>> On Tue, Jul 14, 2015 at 9:12 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> Does this happen in the latest 1.4.20 release
>>>
>>>
>>> On Tuesday, July 14, 2015, Niraj Roy <techfaltu at gmail.com> wrote:
>>>
>>>> Hello Gurus,
>>>>     We are using FreeSWITCH Version 1.4.18~64bit. We observed when both
>>>> caller and caller hangs up the call simultaneously exec_after_bridge_app
>>>> function does not execute for one of the legs.
>>>>     Here is the segment of executing the xml file.
>>>>                 <action application="set"
>>>> data="exec_after_bridge_app=transfer"/>
>>>>                 <action application="set"
>>>> data="exec_after_bridge_arg=endcall-handler XML CALLENDING"/>
>>>>                 <action application="set_zombie_exec"/>
>>>>                 <action application="set" data="continue_on_fail=true"/>
>>>>                 <action inline="true" application="set"
>>>> data="hangup_after_bridge=false"/>
>>>>
>>>> For normal or sequential call clearing everything works fine, but
>>>> somehow this creates a race condition and state mismatch in the freeswitch
>>>> which I don't know.
>>>> Can anybody shed some light on this?
>>>>
>>>> Thanks,
>>>>
>>>>
>>> _________________________________________________________________________
>>> 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
>>
>
>
>
> --
> 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
>
> _________________________________________________________________________
> 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  Fri Jul 17 12:18:36 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 17 Jul 2015 10:18:36 +0200
Subject: [Freeswitch-users] verto.conf.xml parameter for public access
In-Reply-To: <CAHKqMb6XPd8nLcN2zMOR1MGpRrxxyDhdZGqYLLHi5my1d70_HQ@mail.gmail.com>
References: <CAHKqMb5s9NGtaiEHO5Z5uWDi=gCdeTEmsPBqU6dLRwHc4r=-LQ@mail.gmail.com>
	<CAHKqMb6=Hn4ua8FJL-nvD4vUvHEnY8GaGtt_PkhFcUD1_uYguA@mail.gmail.com>
	<CAEwTsAiiesrOWC8Z8RhGwswwwkYJ0yHmYL+sy_hx0BGqehFvyQ@mail.gmail.com>
	<CAHKqMb4F+CWYtfsGHSycowmPQ9Fsz6FnYiKPzFqqpDtQA7Pisg@mail.gmail.com>
	<CAEwTsAgKv59OnMAPD1c04_BHBQ7jpmsODXUq5-znmit_J-=m8w@mail.gmail.com>
	<CAHKqMb6XPd8nLcN2zMOR1MGpRrxxyDhdZGqYLLHi5my1d70_HQ@mail.gmail.com>
Message-ID: <CALXCt0qiWTA-d4Ho4THd-LYoKgPjMfkb=K3JGhB1JjZ1UphNuQ@mail.gmail.com>

Nandy, if you got WS setup failed, it reached you, so NAT is ok. Probably
you have a certificate problem.


On Fri, Jul 17, 2015 at 7:09 AM, Nandy Dagondon <nandy1925 at gmail.com> wrote:

> Hi Michael,
>
> Will this setup (NAT map) work to accept public clients?
> Back to my first question, is "ext-rtp-ip" needed for this NAT map setup?
>
> Finally, I saw this error message in fs_cli when a public client (Chrome
> on Android) connects:
> WS SETUP FAILED [ ]
>
> Can you give a hint where the problem lies?
>
> Tks,
> Nandy
>
> On Fri, Jul 17, 2015 at 8:53 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> then the bind url is right
>>
>> On Thursday, July 16, 2015, Nandy Dagondon <nandy1925 at gmail.com> wrote:
>>
>>> NAT map.  The router's DMZ point to this interface. Tks.  /Nandy
>>>
>>>
>>> On Thu, Jul 16, 2015 at 10:43 PM, Michael Jerris <mike at jerris.com>
>>> wrote:
>>>
>>>> Is the wan IP address bound to your interface or is nat mapping it to
>>>> your internal address.  The bind url is the actual interface address on
>>>> your system
>>>>
>>>> On Thursday, July 16, 2015, Nandy Dagondon <nandy1925 at gmail.com> wrote:
>>>>
>>>>> Hi to everyone,
>>>>>
>>>>> More related question.
>>>>>
>>>>> I see this line when I execute on fs_cli:
>>>>>
>>>>> WSS-BIND-URL      sips:mod_sofia at 192.168.0.5:7443;transport=wss
>>>>>
>>>>> It's my local IP address.  Shouldn't it be my public WAN IP address?
>>>>>
>>>>> Please shed some light.  Thanks.
>>>>>
>>>>> /Nandy
>>>>>
>>>>> On Wed, Jul 15, 2015 at 9:53 AM, Nandy Dagondon <nandy1925 at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> I succesfully installed mod_verto in my intranet.  FS is behind NAT.
>>>>>> Next step, I'm setting up where remote web clients can login.  I setup my
>>>>>> router DMZ to FS.
>>>>>>
>>>>>> This parameter is commented out  "ext-rtp-ip" in the Wiki. Is this
>>>>>> parameter necessary for accepting remote clients?
>>>>>>
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From mbsip at gazeta.pl  Fri Jul 17 12:48:09 2015
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Fri, 17 Jul 2015 10:48:09 +0200
Subject: [Freeswitch-users] how to record SIP+RTP sessions
In-Reply-To: <DA0FFF64-FF25-4ADA-92CB-E34AFA439DBD@1012.at>
References: <CACynCY4ufxgQ=xFJ+KcC9WNnZn5Qq_TqbrpuqtLQsmZmrede2g@mail.gmail.com>
	<CAHtxdDc1BeWad9cTeX69cXrfK+1xTWJid8PzYc1SpwwzaMA3Zg@mail.gmail.com>
	<CAJR7f2HuLMY5ZtH8eHUADzaT-nq=w7LnZLEHUuct7MZHkhuq4A@mail.gmail.com>
	<CACynCY4PBkgEicoTrnYEZL=3m5PkuetcJVYZXa=Ys_BApt_w8g@mail.gmail.com>
	<55A7D38F.40703@mst.edu> <1437087030418.38416@netsip.com.au>
	<DA0FFF64-FF25-4ADA-92CB-E34AFA439DBD@1012.at>
Message-ID: <CACynCY5dO4ZuaYKKN54zrKPRRzzTaMgfoBnFa5-uZfAyYFmhvw@mail.gmail.com>

Great i will take a look on this.

Btw: what about pcapsipdump, it is really so limited in this matter?

Thanks
Maciej.

2015-07-17 2:13 GMT+02:00 Stefan Kainz <stefan.kainz at 1012.at>:

> +1 Voipmonitor.
>
> Sent from my iPhone
>
> > On 17 Jul 2015, at 00:50, David Witham <david.witham at netsip.com.au>
> wrote:
> >
> > +1 for voipmonitor. Its invaluable for monitoring and diagnosing
> problems on our voice network.
> > ________________________________________
> > From: freeswitch-users-bounces at lists.freeswitch.org <
> freeswitch-users-bounces at lists.freeswitch.org> on behalf of Nathan
> Neulinger <nneul at mst.edu>
> > Sent: Friday, 17 July 2015 01:53
> > To: FreeSWITCH Users Help
> > Subject: Re: [Freeswitch-users] how to record SIP+RTP sessions
> >
> > voipmonitor capture engine will do this quite nicely. Search list
> archives for some previous discussions on it.
> >
> > -- Nathan
> >
> >> On 07/16/2015 10:49 AM, Maciej Bylica wrote:
> >> Thanks for your replies..
> >>
> >> Yes i have Homer already installed but is does not store RTP packets.
> >> Regarding wireshark... yes but in case of heavy traffic it takes two
> mins to fill in 100Meg file and quite often it is
> >> not enough to store all the conversation.
> >> Thats why i am looking for sth that could store SIP+RTP data as well as
> create separate files for each session.
> >>
> >> Thanks.
> >> Maciej
> >>
> >> 2015-07-16 14:28 GMT+02:00 ?talo Rossi <italorossib at gmail.com <mailto:
> italorossib at gmail.com>>:
> >>
> >>    http://www.sipcapture.org/
> >>
> >>    On Thu, Jul 16, 2015 at 8:43 AM, Sergey Safarov <s.safarov at gmail.com
> <mailto:s.safarov at gmail.com>> wrote:
> >>
> >>        I use wireshark, to analyze sip messages and RTP media
> >>
> >>
> >>        On Thu, Jul 16, 2015, 14:31 Maciej Bylica <mbsip at gazeta.pl
> <mailto:mbsip at gazeta.pl>> wrote:
> >>
> >>            Hi,
> >>
> >>            I need some help in finding tool to sniff SIP+RTP packets
> and separate each session.
> >>            Following the wiki page
> https://wiki.freeswitch.org/wiki/Packet_Capture i've installed
> pcapsipdump 0.2 rel.
> >>            Unfortunately it utilizes just one core to maximum and all
> .pcap files are quality affected.
> >>
> >>            As i assume this is the limitation of this app and there is
> nothing i can do to overcome this.
> >>            If possitive, could you please point me to some similiar
> free app.
> >>
> >>            Thanks
> >>            Maciej.
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >>            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
> >>
> >>
> >>
> >>
> >>    --
> >>    ?talo Rossi
> >>
> >>
> _________________________________________________________________________
> >>    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
> >> 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
> >
> > _________________________________________________________________________
> > 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 nandy1925 at gmail.com  Fri Jul 17 13:22:38 2015
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Fri, 17 Jul 2015 17:22:38 +0800
Subject: [Freeswitch-users] verto.conf.xml parameter for public access
In-Reply-To: <CALXCt0qiWTA-d4Ho4THd-LYoKgPjMfkb=K3JGhB1JjZ1UphNuQ@mail.gmail.com>
References: <CAHKqMb5s9NGtaiEHO5Z5uWDi=gCdeTEmsPBqU6dLRwHc4r=-LQ@mail.gmail.com>
	<CAHKqMb6=Hn4ua8FJL-nvD4vUvHEnY8GaGtt_PkhFcUD1_uYguA@mail.gmail.com>
	<CAEwTsAiiesrOWC8Z8RhGwswwwkYJ0yHmYL+sy_hx0BGqehFvyQ@mail.gmail.com>
	<CAHKqMb4F+CWYtfsGHSycowmPQ9Fsz6FnYiKPzFqqpDtQA7Pisg@mail.gmail.com>
	<CAEwTsAgKv59OnMAPD1c04_BHBQ7jpmsODXUq5-znmit_J-=m8w@mail.gmail.com>
	<CAHKqMb6XPd8nLcN2zMOR1MGpRrxxyDhdZGqYLLHi5my1d70_HQ@mail.gmail.com>
	<CALXCt0qiWTA-d4Ho4THd-LYoKgPjMfkb=K3JGhB1JjZ1UphNuQ@mail.gmail.com>
Message-ID: <CAHKqMb7DtuNv+ZrRet_BaPi0v+EofZ98HttSBxkerysDordXzw@mail.gmail.com>

I'm using self-signed certificates. It has to be the wss.pem.  I'll explore
further on this area. Thanks for the hint Giovanni.  Rgds.

/Nandy
On Fri, Jul 17, 2015 at 4:18 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Nandy, if you got WS setup failed, it reached you, so NAT is ok. Probably
> you have a certificate problem.
>
>
> On Fri, Jul 17, 2015 at 7:09 AM, Nandy Dagondon <nandy1925 at gmail.com>
> wrote:
>
>> Hi Michael,
>>
>> Will this setup (NAT map) work to accept public clients?
>> Back to my first question, is "ext-rtp-ip" needed for this NAT map setup?
>>
>> Finally, I saw this error message in fs_cli when a public client (Chrome
>> on Android) connects:
>> WS SETUP FAILED [ ]
>>
>> Can you give a hint where the problem lies?
>>
>> Tks,
>> Nandy
>>
>> On Fri, Jul 17, 2015 at 8:53 AM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> then the bind url is right
>>>
>>> On Thursday, July 16, 2015, Nandy Dagondon <nandy1925 at gmail.com> wrote:
>>>
>>>> NAT map.  The router's DMZ point to this interface. Tks.  /Nandy
>>>>
>>>>
>>>> On Thu, Jul 16, 2015 at 10:43 PM, Michael Jerris <mike at jerris.com>
>>>> wrote:
>>>>
>>>>> Is the wan IP address bound to your interface or is nat mapping it to
>>>>> your internal address.  The bind url is the actual interface address on
>>>>> your system
>>>>>
>>>>> On Thursday, July 16, 2015, Nandy Dagondon <nandy1925 at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Hi to everyone,
>>>>>>
>>>>>> More related question.
>>>>>>
>>>>>> I see this line when I execute on fs_cli:
>>>>>>
>>>>>> WSS-BIND-URL      sips:mod_sofia at 192.168.0.5:7443;transport=wss
>>>>>>
>>>>>> It's my local IP address.  Shouldn't it be my public WAN IP address?
>>>>>>
>>>>>> Please shed some light.  Thanks.
>>>>>>
>>>>>> /Nandy
>>>>>>
>>>>>> On Wed, Jul 15, 2015 at 9:53 AM, Nandy Dagondon <nandy1925 at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> I succesfully installed mod_verto in my intranet.  FS is behind NAT.
>>>>>>> Next step, I'm setting up where remote web clients can login.  I setup my
>>>>>>> router DMZ to FS.
>>>>>>>
>>>>>>> This parameter is commented out  "ext-rtp-ip" in the Wiki. Is this
>>>>>>> parameter necessary for accepting remote clients?
>>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> 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  Fri Jul 17 13:28:08 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 17 Jul 2015 11:28:08 +0200
Subject: [Freeswitch-users] verto.conf.xml parameter for public access
In-Reply-To: <CAHKqMb7DtuNv+ZrRet_BaPi0v+EofZ98HttSBxkerysDordXzw@mail.gmail.com>
References: <CAHKqMb5s9NGtaiEHO5Z5uWDi=gCdeTEmsPBqU6dLRwHc4r=-LQ@mail.gmail.com>
	<CAHKqMb6=Hn4ua8FJL-nvD4vUvHEnY8GaGtt_PkhFcUD1_uYguA@mail.gmail.com>
	<CAEwTsAiiesrOWC8Z8RhGwswwwkYJ0yHmYL+sy_hx0BGqehFvyQ@mail.gmail.com>
	<CAHKqMb4F+CWYtfsGHSycowmPQ9Fsz6FnYiKPzFqqpDtQA7Pisg@mail.gmail.com>
	<CAEwTsAgKv59OnMAPD1c04_BHBQ7jpmsODXUq5-znmit_J-=m8w@mail.gmail.com>
	<CAHKqMb6XPd8nLcN2zMOR1MGpRrxxyDhdZGqYLLHi5my1d70_HQ@mail.gmail.com>
	<CALXCt0qiWTA-d4Ho4THd-LYoKgPjMfkb=K3JGhB1JjZ1UphNuQ@mail.gmail.com>
	<CAHKqMb7DtuNv+ZrRet_BaPi0v+EofZ98HttSBxkerysDordXzw@mail.gmail.com>
Message-ID: <CALXCt0qEY3DUKo0F7QqbsbcqgOKyY1YaSJMN4rLt3F=_KN5Nxg@mail.gmail.com>

you can first test with plain ws (eg: not wss) on port 8081, just change
the ws url in verto client.
So you can check network is ok.
Then double check that wss.pem contains cert, key, and chain in the order
specified in FreeSWITCH 1.6 page (eg: CERT, KEY AND CHAIN)

On Fri, Jul 17, 2015 at 11:22 AM, Nandy Dagondon <nandy1925 at gmail.com>
wrote:

>
> I'm using self-signed certificates. It has to be the wss.pem.  I'll
> explore further on this area. Thanks for the hint Giovanni.  Rgds.
>
> /Nandy
>
> On Fri, Jul 17, 2015 at 4:18 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> Nandy, if you got WS setup failed, it reached you, so NAT is ok. Probably
>> you have a certificate problem.
>>
>>
>> On Fri, Jul 17, 2015 at 7:09 AM, Nandy Dagondon <nandy1925 at gmail.com>
>> wrote:
>>
>>> Hi Michael,
>>>
>>> Will this setup (NAT map) work to accept public clients?
>>> Back to my first question, is "ext-rtp-ip" needed for this NAT map setup?
>>>
>>> Finally, I saw this error message in fs_cli when a public client (Chrome
>>> on Android) connects:
>>> WS SETUP FAILED [ ]
>>>
>>> Can you give a hint where the problem lies?
>>>
>>> Tks,
>>> Nandy
>>>
>>> On Fri, Jul 17, 2015 at 8:53 AM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> then the bind url is right
>>>>
>>>> On Thursday, July 16, 2015, Nandy Dagondon <nandy1925 at gmail.com> wrote:
>>>>
>>>>> NAT map.  The router's DMZ point to this interface. Tks.  /Nandy
>>>>>
>>>>>
>>>>> On Thu, Jul 16, 2015 at 10:43 PM, Michael Jerris <mike at jerris.com>
>>>>> wrote:
>>>>>
>>>>>> Is the wan IP address bound to your interface or is nat mapping it to
>>>>>> your internal address.  The bind url is the actual interface address on
>>>>>> your system
>>>>>>
>>>>>> On Thursday, July 16, 2015, Nandy Dagondon <nandy1925 at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Hi to everyone,
>>>>>>>
>>>>>>> More related question.
>>>>>>>
>>>>>>> I see this line when I execute on fs_cli:
>>>>>>>
>>>>>>> WSS-BIND-URL      sips:mod_sofia at 192.168.0.5:7443;transport=wss
>>>>>>>
>>>>>>> It's my local IP address.  Shouldn't it be my public WAN IP address?
>>>>>>>
>>>>>>> Please shed some light.  Thanks.
>>>>>>>
>>>>>>> /Nandy
>>>>>>>
>>>>>>> On Wed, Jul 15, 2015 at 9:53 AM, Nandy Dagondon <nandy1925 at gmail.com
>>>>>>> > wrote:
>>>>>>>
>>>>>>>> I succesfully installed mod_verto in my intranet.  FS is behind
>>>>>>>> NAT. Next step, I'm setting up where remote web clients can login.  I setup
>>>>>>>> my router DMZ to FS.
>>>>>>>>
>>>>>>>> This parameter is commented out  "ext-rtp-ip" in the Wiki. Is this
>>>>>>>> parameter necessary for accepting remote clients?
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> 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
>>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _________________________________________________________________________
>> 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
Cell : +39-347-2665618
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From jason.holden at start.ca  Fri Jul 17 17:41:57 2015
From: jason.holden at start.ca (Jason Holden)
Date: Fri, 17 Jul 2015 09:41:57 -0400
Subject: [Freeswitch-users] shared line or other suggestion
Message-ID: <B0AD74FC10B94C6DBFCDB0F34F2A9112@bob>

I am currently trying to have the ability to place a call on hold and be
able to pick the call up from a displayed list on a group of  Polycom
phones.

I am currently using shared line but when the main phone places a call on
hold the shared lines to not show the held call with the remote user CID but
the primary phonesCID information.

Does anyone have a recommendation that'll work better.

I want to be able to see the remote CID of the caller who is on hold.

 

 

 

 

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From krice at freeswitch.org  Fri Jul 17 18:01:15 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 17 Jul 2015 14:01:15 +0000
Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <55a90aab51ddc_9c5e4d332097760@resque-worker.11.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
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From manpower13.cse at gmail.com  Fri Jul 17 18:19:16 2015
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Fri, 17 Jul 2015 19:49:16 +0530
Subject: [Freeswitch-users] Freeswitch WS+SIPJS+sip2sip
Message-ID: <CAMRuXGgG+7ps9WEs1UEMwzKC73HMF-6C_JX+ZXkC85h_VzeC_Q@mail.gmail.com>

Hi,

  I try to register sip2sip.info  account from browser using ws
signalling,but i am  getting  SIP/2.0 403 Forbidden error  in my console.

 The same sip2sip.info account work in all soft client
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From mike at jerris.com  Fri Jul 17 19:41:48 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 17 Jul 2015 11:41:48 -0400
Subject: [Freeswitch-users] Freeswitch WS+SIPJS+sip2sip
In-Reply-To: <CAMRuXGgG+7ps9WEs1UEMwzKC73HMF-6C_JX+ZXkC85h_VzeC_Q@mail.gmail.com>
References: <CAMRuXGgG+7ps9WEs1UEMwzKC73HMF-6C_JX+ZXkC85h_VzeC_Q@mail.gmail.com>
Message-ID: <CAEwTsAhzY6spK8VFqCWiqBfTZ-Humutxm7aV8sb7u3o94rFmiw@mail.gmail.com>

are you saying you are attempting to register to FreeSWITCH with
credentials for some other service that has nothing to do with your
FreeSWITCH box and its denying you?

On Friday, July 17, 2015, Murugan Pandian <manpower13.cse at gmail.com> wrote:

> Hi,
>
>   I try to register sip2sip.info  account from browser using ws
> signalling,but i am  getting  SIP/2.0 403 Forbidden error  in my console.
>
>  The same sip2sip.info account work in all soft client
>
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From blasterjr at gmail.com  Fri Jul 17 19:54:43 2015
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Fri, 17 Jul 2015 09:54:43 -0600
Subject: [Freeswitch-users] Execute dialplan after valet_park
In-Reply-To: <CAJUJwti-xnSaKZf_W+9NTqCaaCU2W3f1R5zsWmVDFYQkpfhVDA@mail.gmail.com>
References: <CAJUJwtipc+yhpck+w4M4mYYPk0ZRa5NqWhytRChbsi4c6_nwdA@mail.gmail.com>
	<CAO3hyzE0joUxdFeQM4oJ9K64qdLcG_4uock8v3AqHDWc6pSdaw@mail.gmail.com>
	<CAJUJwtjdBerXshWVJhQ0KJ-M_chiAaNiXgGaEU4hs4_nN3b8+Q@mail.gmail.com>
	<CAO3hyzF0dsxCEopEOeONnNUneTpyRiwXnxSJVLiTnrOEKRVYHA@mail.gmail.com>
	<CAJUJwti-xnSaKZf_W+9NTqCaaCU2W3f1R5zsWmVDFYQkpfhVDA@mail.gmail.com>
Message-ID: <CAO3hyzH-UO4SgRqfBH5Vcgsr7p4xmaj97B5OJPUxyGztk72Wdg@mail.gmail.com>

Fantastic, thanks for sharing the end result. You just taught me about
subclass working on the hook, this is great information for me to have.

On Thu, Jul 16, 2015 at 4:57 PM, SamyGo <govoiper at gmail.com> wrote:

> +1 Got it all working thanks again Chris.
>
> For anyone else in future here is what I did.
>
> Added this line in *conf/autoload_configs/lua.conf.xml*
> <hook event="CUSTOM" subclass="valet_parking::info"
> script="unparking_event.lua"/>
>
> Restarted FS, and got this script executed on each of valet_parking sub
> events "hold/exit/bridge"
>
> In *unparking_event.lua* collect the event Headers:
>
> local action = event:getHeader("Action")
> local park_lot = event:getHeader("Valet-Extension")
> ...
> if action == "exit" then
> --  what I needed to do
> end
>
>
>
>
> On Wed, Jul 15, 2015 at 12:28 PM, Chris Tunbridge <blasterjr at gmail.com>
> wrote:
>
>> When you use a hook, whenever that event (CUSTOM) is called the lua
>> script would be called, so its a seperate script, you then have to do
>> conditions to check that the type is valet_parking::info and execute logic
>> inside that condition
>>
>> On Wed, Jul 15, 2015 at 8:22 AM, SamyGo <govoiper at gmail.com> wrote:
>>
>>> Thanks Chris,
>>>
>>> That was my finding too, I just am not sure if I can monitor this event
>>> from the main LUA script or need an independent lua script only for the
>>> park "hangup" event monitoring. Any further help on this would be highly
>>> appreciated.
>>>
>>> I wonder why the api_hangup hook won't trigger on the parked channel,
>>> would've been much easier. Piece of cake only if we could execute further
>>> dialplan application after valet_parking.
>>>
>>> BR,
>>> Sammy
>>> You might be able to use an event hook with lua, you can catch the
>>> CUSTOM with a subclass of valet_parking::info, then you can check for the
>>> "action" and it'll be "exit" when they leave (any kind of leave) then can
>>> likely check for a disposition.  I'm not currently using this method, but
>>> its someting we looked into for monitoring when people hangup from the
>>> parking lot.
>>>
>>> On Tue, Jul 14, 2015 at 10:49 AM, SamyGo <govoiper at gmail.com> wrote:
>>>
>>>> Hi All,
>>>>
>>>> I need some help in finding  a way to execute some dialplan code after
>>>> the parked user Hangups from the parking lot.
>>>>
>>>> I've a scenario where Parked party decides to hangup while listening to
>>>> MOH while parked. The Dialplan just Hangsup right away and I have no
>>>> control.
>>>>
>>>> I've tried using: *session_in_hangup_hook*, and
>>>> *api_hangup_hook=myscript.lua *but they get executed for the parker
>>>> just after they hear the Parked slot number.
>>>>
>>>> Then I've tried *nolocal:export* these two variables for the other leg
>>>> but nothing happens.
>>>>
>>>> Kindly suggest what other option are there.
>>>>
>>>> Best Regards.
>>>> Sammy
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>
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From manpower13.cse at gmail.com  Fri Jul 17 20:00:00 2015
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Fri, 17 Jul 2015 21:30:00 +0530
Subject: [Freeswitch-users] Freeswitch WS+SIPJS+sip2sip
In-Reply-To: <CAEwTsAhzY6spK8VFqCWiqBfTZ-Humutxm7aV8sb7u3o94rFmiw@mail.gmail.com>
References: <CAMRuXGgG+7ps9WEs1UEMwzKC73HMF-6C_JX+ZXkC85h_VzeC_Q@mail.gmail.com>
	<CAEwTsAhzY6spK8VFqCWiqBfTZ-Humutxm7aV8sb7u3o94rFmiw@mail.gmail.com>
Message-ID: <CAMRuXGj+SfLzxeshtX22tmX3LYto837845T+WnRbDi2GgvaqOw@mail.gmail.com>

HI Michael,

     Thanks for you answer,Actually i using sip service account to register
my browser ,here i am using freeswitch my signalling server (means json to
sip because i am using browser) when try to register my sip service account
i am getting SIP/2.0 403 Forbidden error

On Fri, Jul 17, 2015 at 9:11 PM, Michael Jerris <mike at jerris.com> wrote:

> are you saying you are attempting to register to FreeSWITCH with
> credentials for some other service that has nothing to do with your
> FreeSWITCH box and its denying you?
>
> On Friday, July 17, 2015, Murugan Pandian <manpower13.cse at gmail.com>
> wrote:
>
>> Hi,
>>
>>   I try to register sip2sip.info  account from browser using ws
>> signalling,but i am  getting  SIP/2.0 403 Forbidden error  in my console.
>>
>>  The same sip2sip.info account work in all soft 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 ben at langfeld.co.uk  Fri Jul 17 21:21:11 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Fri, 17 Jul 2015 14:21:11 -0300
Subject: [Freeswitch-users] Freeswitch WS+SIPJS+sip2sip
In-Reply-To: <CAMRuXGj+SfLzxeshtX22tmX3LYto837845T+WnRbDi2GgvaqOw@mail.gmail.com>
References: <CAMRuXGgG+7ps9WEs1UEMwzKC73HMF-6C_JX+ZXkC85h_VzeC_Q@mail.gmail.com>
	<CAEwTsAhzY6spK8VFqCWiqBfTZ-Humutxm7aV8sb7u3o94rFmiw@mail.gmail.com>
	<CAMRuXGj+SfLzxeshtX22tmX3LYto837845T+WnRbDi2GgvaqOw@mail.gmail.com>
Message-ID: <CAAyX+KELEg6St9Ag_CvMbp4Ri-buyyLhM7A-yAPzH5oSEFO=_Q@mail.gmail.com>

What you actually want is a SIP-WS to SIP-UDP proxy like Kamailio. What you
have setup is not what you think it is, nor is the terminology (JSON) you
are using correct.

On 17 July 2015 at 13:00, Murugan Pandian <manpower13.cse at gmail.com> wrote:

> HI Michael,
>
>      Thanks for you answer,Actually i using sip service account to
> register my browser ,here i am using freeswitch my signalling server (means
> json to sip because i am using browser) when try to register my sip service
> account i am getting SIP/2.0 403 Forbidden error
>
> On Fri, Jul 17, 2015 at 9:11 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> are you saying you are attempting to register to FreeSWITCH with
>> credentials for some other service that has nothing to do with your
>> FreeSWITCH box and its denying you?
>>
>> On Friday, July 17, 2015, Murugan Pandian <manpower13.cse at gmail.com>
>> wrote:
>>
>>> Hi,
>>>
>>>   I try to register sip2sip.info  account from browser using ws
>>> signalling,but i am  getting  SIP/2.0 403 Forbidden error  in my console.
>>>
>>>  The same sip2sip.info account work in all soft 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
>
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From steveayre at gmail.com  Fri Jul 17 23:44:32 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 17 Jul 2015 20:44:32 +0100
Subject: [Freeswitch-users] Re- Connecting into British Telecom (BTs)
 SIP sandbox test platform
In-Reply-To: <SIXPR04MB1267941CA3860C8DE563226F3990@SIXPR04MB126.apcprd04.prod.outlook.com>
References: <SIXPR04MB1267941CA3860C8DE563226F3990@SIXPR04MB126.apcprd04.prod.outlook.com>
Message-ID: <CAFiqYu=Zhua4WK-qriAJoNtfGQFUmBKUZ8o4f0ZTpKvy2cSHjw@mail.gmail.com>

You might find this useful
http://blog.aeriandi.com/2012/10/08/bt-interoperability-testing-a-guide-to-jumping-the-hoops/

You can configure a pair of sofia gateway to dial them through. Set the
register param to false. You'll need to provide a password, but just give a
dummy value (1234) as it won't be used since they won't challenge for it.

For incoming calls you'll need to set auth-calls to false on the sofia
profile, so that it accepts calls without authenticating them (they
wouldn't have a password if you challenged them). You'd need to verify
within the dialplan that it's a call from BT using ${network_addr} (unless
you can ensure the profile can only receive traffic from BT). Alternatively
you could create a user directory entry that authenticated any calls from
their IP as a 'BT' user (the cidr attribute). The incoming calls won't
authenticate based on the gateway as the magic that allows that does so by
putting a magic value in the Contact header during registration, which
won't be used in this case.

Steve




On 16 July 2015 at 12:25, Andrew Keil <andrew.keil at visytel.com> wrote:

>  To FreeSWITCH users,
>
>
>
> Just a quick question that I hope someone can help me with.  I am sure I
> can solve this myself when BT sort out their firewall, however while I wait
> 1-2 days I thought I would post this question here.
>
>
>
> *Currently BT have provided me with two SIP IPs and MEDIA IPs for their
> Sandbox:*
>
>
>
> BT SBC1 IP: Signalling xxx.xxx.xxx.26 (Media xxx.xxx.xxx.25)
>
> BT SBC2 IP: Signalling xxx.xxx.xxx.23 (Media xxx.xxx.xxx.22)
>
>
>
> Plus a SIP Trunk Number Range: 0551xxxx100 ? 0551xxxx109
>
> Model PSTN numbers: 0207xxxx713 and 0207xxxx714
>
>
>
> First local CentOS 6.6 FreeSWITCH server (version 1.4.20) has two NICs
> (eth0: 192.168.2.10 {with public IP address mapped to this IP
> (xxx.xxx.xxx.67)} ; eth1: 192.168.2.11 (used internally only)}
>
> Second local CentOS 6.6 FreeSWITCH server (version 1.4.20) has two NICs
> (eth0: 192.168.2.12 {with public IP address mapped to this IP
> (xxx.xxx.xxx.68)} ; eth1: 192.168.2.13 (used internally only)}
>
>
>
> They have stated to me that they do not require and registration or
> authentication.
>
>
>
> Unfortunately when I looked at all the samples on the FreeSWITCH
> Confluence they all seem to mention registering and sip proxy etc?.
>
>
>
> Can someone provide a simple .xml file(s) example for the above (even if
> it is to only one SBC) to fast-track my setup.  Any other hints or tips (or
> links to freeswitch.org Docs) I would appreciate.
>
>
>
> One more thing to note, I am connecting FreeSWITCH via a local Firewall to
> BT?s SBC?s listed above, there is no SBC on my side since currently it is
> not needed based on my client?s current requirements.  Hopefully this makes
> everything a little simpler, however I do understand that this
> configuration maybe less flexible (especially when commissioning more
> FreeSWITCH servers).
>
>
>
> Thanks in advance,
>
>
>
> Andrew Keil
>
> *Visytel Pty Ltd*
>
>
>
> PS. Since I am currently undertaking BT?s testing on behalf of my client
> in the UK, I will provide back to the FreeSWITCH community the completed
>  BT Interoperability SIP test cases with the associated FreeSWITCH
> dialplans used and any comments.  Hopefully this will help any other
> company to pass BT?s SIP testing requirements in a fast manner J
>
>
>
> _________________________________________________________________________
> 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 manpower13.cse at gmail.com  Fri Jul 17 23:52:26 2015
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Sat, 18 Jul 2015 01:22:26 +0530
Subject: [Freeswitch-users] Mod_verto vs WebRTC(ws/wss)
Message-ID: <CAMRuXGjVxqiuT75WhKE9nWtSXewRYtRxh-LCu+g9Tac7EvQXWg@mail.gmail.com>

Hi,

   Any one can help me what is difference between mod_verto and
 WebRTC(ws/wss) mod.
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From steven.szeto at mitel.com  Sat Jul 18 00:01:02 2015
From: steven.szeto at mitel.com (Steven Szeto)
Date: Fri, 17 Jul 2015 20:01:02 +0000
Subject: [Freeswitch-users] Is there a way to temporarily disable DTMF
 controls during an eavesdrop (or control an eavesdrop session with non-DTMF
 stimulus)?
Message-ID: <DM2PR12MB025234812017F4ECF076AD758A980@DM2PR12MB0252.namprd12.prod.outlook.com>

I can successfully create a whisper/coach eavesdrop session with the following commands issued from fs_cli:

originate sofia/internal/5401 at 10.47.41.109 &bridge(sofia/internal/5901 at 10.47.41.109)

whisper/coach:
originate sofia/internal/5902 at 10.47.41.109 'queue_dtmf:w2 at 500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861' inline

I would like to temporarily disable DTMF so that the supervisor can not change the eavesdrop mode via DTMF, but this does not work:

originate sofia/internal/5902 at 10.47.41.109 'queue_dtmf:w2 at 500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861, set:eavesdrop_enable_dtmf=false' inline

Moreover, I have a GUI app that presents the supervisor with the ability to change the eavesdrop mode. When he presses the appropriate icon in the GUI, I would then like to reenable DTMF and change the eavesdrop mode on the session:

Uuid_setvar a28739d0-00f0-4a59-8c82-7a5a74ab6861 eavesdrop_enable_dtmf true;
Uuid_recv_dtmf a28739d0-00f0-4a59-8c82-7a5a74ab6861 w3 at 500

Is there another way to change the mode of an eavesdrop session?



________________________________
NOTE: This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message.
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From mike at jerris.com  Sat Jul 18 00:10:08 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 17 Jul 2015 16:10:08 -0400
Subject: [Freeswitch-users] Is there a way to temporarily disable DTMF
	controls during an eavesdrop (or control an eavesdrop session
	with non-DTMF stimulus)?
In-Reply-To: <DM2PR12MB025234812017F4ECF076AD758A980@DM2PR12MB0252.namprd12.prod.outlook.com>
References: <DM2PR12MB025234812017F4ECF076AD758A980@DM2PR12MB0252.namprd12.prod.outlook.com>
Message-ID: <F9A9C658-D1FB-43A5-B94F-BEF4DE0406E2@jerris.com>

does it work if you do the set before the eavesdrop?  That set is not running until after the eavesdrop ends.

> On Jul 17, 2015, at 4:01 PM, Steven Szeto <steven.szeto at mitel.com> wrote:
> 
> I can successfully create a whisper/coach eavesdrop session with the following commands issued from fs_cli:
>  
> originate sofia/internal/5401 at 10.47.41.109 <mailto:sofia/internal/5401 at 10.47.41.109> &bridge(sofia/internal/5901 at 10.47.41.109 <mailto:sofia/internal/5901 at 10.47.41.109>)
>  
> whisper/coach:
> originate sofia/internal/5902 at 10.47.41.109 <mailto:sofia/internal/5902 at 10.47.41.109> 'queue_dtmf:w2 at 500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861' inline
>  
> I would like to temporarily disable DTMF so that the supervisor can not change the eavesdrop mode via DTMF, but this does not work:
>  
> originate sofia/internal/5902 at 10.47.41.109 <mailto:sofia/internal/5902 at 10.47.41.109> 'queue_dtmf:w2 at 500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861, set:eavesdrop_enable_dtmf=false' inline
>  
> Moreover, I have a GUI app that presents the supervisor with the ability to change the eavesdrop mode. When he presses the appropriate icon in the GUI, I would then like to reenable DTMF and change the eavesdrop mode on the session:
>  
> Uuid_setvar a28739d0-00f0-4a59-8c82-7a5a74ab6861 eavesdrop_enable_dtmf true;
> Uuid_recv_dtmf a28739d0-00f0-4a59-8c82-7a5a74ab6861 w3 at 500
>  
> Is there another way to change the mode of an eavesdrop session?
>  
>  
> 
> NOTE: This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message.
> 
> _________________________________________________________________________
> 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 ben at langfeld.co.uk  Sat Jul 18 00:12:37 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Fri, 17 Jul 2015 17:12:37 -0300
Subject: [Freeswitch-users] Mod_verto vs WebRTC(ws/wss)
In-Reply-To: <CAMRuXGjVxqiuT75WhKE9nWtSXewRYtRxh-LCu+g9Tac7EvQXWg@mail.gmail.com>
References: <CAMRuXGjVxqiuT75WhKE9nWtSXewRYtRxh-LCu+g9Tac7EvQXWg@mail.gmail.com>
Message-ID: <CAAyX+KFJCswqb1k7WN86udhAxuBza9RTLjYat695aG6cqAR-dQ@mail.gmail.com>

Both mod_verto and SIP-WS are signalling mechanisms for WebRTC. The former
is a FreeSWITCH-proprietary protocol, the latter is a standards-based
approach. The former is probably easier to understand if you've never done
telephony work, the latter requires you to know more stuff.

On 17 July 2015 at 16:52, Murugan Pandian <manpower13.cse at gmail.com> wrote:

> Hi,
>
>    Any one can help me what is difference between mod_verto and
>  WebRTC(ws/wss) mod.
>
>
>
>
> _________________________________________________________________________
> 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 manpower13.cse at gmail.com  Sat Jul 18 00:24:41 2015
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Sat, 18 Jul 2015 01:54:41 +0530
Subject: [Freeswitch-users] Mod_verto vs WebRTC(ws/wss)
In-Reply-To: <CAAyX+KFJCswqb1k7WN86udhAxuBza9RTLjYat695aG6cqAR-dQ@mail.gmail.com>
References: <CAMRuXGjVxqiuT75WhKE9nWtSXewRYtRxh-LCu+g9Tac7EvQXWg@mail.gmail.com>
	<CAAyX+KFJCswqb1k7WN86udhAxuBza9RTLjYat695aG6cqAR-dQ@mail.gmail.com>
Message-ID: <CAMRuXGj5Z8LtuRd8_WLR5cy7oQO9a4p32tgZf=L6x0s=GTtoSA@mail.gmail.com>

Thanks, If there any way i can achieve  this



        {Browser}
                 |
                 |
                 |   (SIP-WS or verto)
          (iptel or sip2sip)
                 |-  -   -  -  - -  -  - - - - -[freeswitch]- - - - - - - -
-  -  -   [Register]




On Sat, Jul 18, 2015 at 1:42 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:

> Both mod_verto and SIP-WS are signalling mechanisms for WebRTC. The former
> is a FreeSWITCH-proprietary protocol, the latter is a standards-based
> approach. The former is probably easier to understand if you've never done
> telephony work, the latter requires you to know more stuff.
>
> On 17 July 2015 at 16:52, Murugan Pandian <manpower13.cse at gmail.com>
> wrote:
>
>> Hi,
>>
>>    Any one can help me what is difference between mod_verto and
>>  WebRTC(ws/wss) mod.
>>
>>
>>
>>
>> _________________________________________________________________________
>> 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 manpower13.cse at gmail.com  Sat Jul 18 00:30:35 2015
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Sat, 18 Jul 2015 02:00:35 +0530
Subject: [Freeswitch-users] Mod_verto vs WebRTC(ws/wss)
In-Reply-To: <CAMRuXGj5Z8LtuRd8_WLR5cy7oQO9a4p32tgZf=L6x0s=GTtoSA@mail.gmail.com>
References: <CAMRuXGjVxqiuT75WhKE9nWtSXewRYtRxh-LCu+g9Tac7EvQXWg@mail.gmail.com>
	<CAAyX+KFJCswqb1k7WN86udhAxuBza9RTLjYat695aG6cqAR-dQ@mail.gmail.com>
	<CAMRuXGj5Z8LtuRd8_WLR5cy7oQO9a4p32tgZf=L6x0s=GTtoSA@mail.gmail.com>
Message-ID: <CAMRuXGheGqw9Q5fAzndjyjxS8P3abd2o6bK0Vv_C-i41hwqZOA@mail.gmail.com>

Sorry for Diagram above one not clear


         {Browser}
                 |
                 |
   (SIP-WS or verto)
                 |
        [freeswitch]
                 |
                 |
         [Register]
        (iptel or sip2sip)

On Sat, Jul 18, 2015 at 1:54 AM, Murugan Pandian <manpower13.cse at gmail.com>
wrote:

> Thanks, If there any way i can achieve  this
>
>
>
>         {Browser}
>                  |
>                  |
>                  |   (SIP-WS or verto)
>           (iptel or sip2sip)
>                  |-  -   -  -  - -  -  - - - - -[freeswitch]- - - - - - -
> - -  -  -   [Register]
>
>
>
>
> On Sat, Jul 18, 2015 at 1:42 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>
>> Both mod_verto and SIP-WS are signalling mechanisms for WebRTC. The
>> former is a FreeSWITCH-proprietary protocol, the latter is a
>> standards-based approach. The former is probably easier to understand if
>> you've never done telephony work, the latter requires you to know more
>> stuff.
>>
>> On 17 July 2015 at 16:52, Murugan Pandian <manpower13.cse at gmail.com>
>> wrote:
>>
>>> Hi,
>>>
>>>    Any one can help me what is difference between mod_verto and
>>>  WebRTC(ws/wss) mod.
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> 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 steven.szeto at mitel.com  Sat Jul 18 00:35:06 2015
From: steven.szeto at mitel.com (Steven Szeto)
Date: Fri, 17 Jul 2015 20:35:06 +0000
Subject: [Freeswitch-users] Is there a way to temporarily disable DTMF
 controls during an eavesdrop (or control an eavesdrop session with non-DTMF
 stimulus)?
Message-ID: <DM2PR12MB0252EA8D47B0463D4BD007308A980@DM2PR12MB0252.namprd12.prod.outlook.com>

Michael Jerris asks:

does it work if you do the set before the eavesdrop?  That set is not running until after the eavesdrop ends.

Answer:

Yes, set:eavesdrop_enable_dtmf=false, does work if it is invoked before the eavesdrop. However there are two issues:


*         The "queue_dtmf" command is now ignored, which means the eavesdrop session starts as silent monitor instead of a whisper/coach session

*         The attempt to set eavesdrop_enable_dtmf=true does not seem to reopen the DTMF listening capabilities of the eavesdrop session. DTMF keystrokes 0,1,2,3 have no effect.

The solution for me would be this:


*         Before the eavesdrop session is started, , set:eavesdrop_enable_dtmf=false

*         Introduce the ability to change the eavesdrop mode via the command line

o   E.g. eavesdrop_change_mode <uuid> [0,1,2,3]

o   Where [0,1,2,3] are the following eavesdrop modes respectively: silent monitor, coach the a-leg, coach the b-leg, barge-in conference
From: Steven Szeto
Sent: Friday, July 17, 2015 4:01 PM
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
Subject: Is there a way to temporarily disable DTMF controls during an eavesdrop (or control an eavesdrop session with non-DTMF stimulus)?

I can successfully create a whisper/coach eavesdrop session with the following commands issued from fs_cli:

originate sofia/internal/5401 at 10.47.41.109<mailto:sofia/internal/5401 at 10.47.41.109> &bridge(sofia/internal/5901 at 10.47.41.109<mailto:sofia/internal/5901 at 10.47.41.109>)

whisper/coach:
originate sofia/internal/5902 at 10.47.41.109<mailto:sofia/internal/5902 at 10.47.41.109> 'queue_dtmf:w2 at 500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861' inline

I would like to temporarily disable DTMF so that the supervisor can not change the eavesdrop mode via DTMF, but this does not work:

originate sofia/internal/5902 at 10.47.41.109<mailto:sofia/internal/5902 at 10.47.41.109> 'queue_dtmf:w2 at 500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861, set:eavesdrop_enable_dtmf=false' inline

Moreover, I have a GUI app that presents the supervisor with the ability to change the eavesdrop mode. When he presses the appropriate icon in the GUI, I would then like to reenable DTMF and change the eavesdrop mode on the session:

Uuid_setvar a28739d0-00f0-4a59-8c82-7a5a74ab6861 eavesdrop_enable_dtmf true;
Uuid_recv_dtmf a28739d0-00f0-4a59-8c82-7a5a74ab6861 w3 at 500

Is there another way to change the mode of an eavesdrop session?



________________________________
NOTE: This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message.
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From gmaruzz at gmail.com  Sat Jul 18 01:08:51 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 17 Jul 2015 23:08:51 +0200
Subject: [Freeswitch-users] Mod_verto vs WebRTC(ws/wss)
In-Reply-To: <CAMRuXGheGqw9Q5fAzndjyjxS8P3abd2o6bK0Vv_C-i41hwqZOA@mail.gmail.com>
References: <CAMRuXGjVxqiuT75WhKE9nWtSXewRYtRxh-LCu+g9Tac7EvQXWg@mail.gmail.com>
	<CAAyX+KFJCswqb1k7WN86udhAxuBza9RTLjYat695aG6cqAR-dQ@mail.gmail.com>
	<CAMRuXGj5Z8LtuRd8_WLR5cy7oQO9a4p32tgZf=L6x0s=GTtoSA@mail.gmail.com>
	<CAMRuXGheGqw9Q5fAzndjyjxS8P3abd2o6bK0Vv_C-i41hwqZOA@mail.gmail.com>
Message-ID: <CALXCt0r3+o5mv08zFX_MD=fM2mpOmU_pH7B6=j-TNyLgd4X1BQ@mail.gmail.com>

Yes, you can do it.

Study webrtc on web, then read about mod-verto and webrtc in
http://confluence.freeswitch.org when you studied that, and you know
webrtc, you can easily do it. Check freeswitch 1.6 again in confluence. Or
you can come back and make specific questions on something that is not
clear in documentation.

-giovanni

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On Jul 17, 2015 10:31 PM, "Murugan Pandian" <manpower13.cse at gmail.com>
wrote:

> Sorry for Diagram above one not clear
>
>
>          {Browser}
>                  |
>                  |
>    (SIP-WS or verto)
>                  |
>         [freeswitch]
>                  |
>                  |
>          [Register]
>         (iptel or sip2sip)
>
> On Sat, Jul 18, 2015 at 1:54 AM, Murugan Pandian <manpower13.cse at gmail.com
> > wrote:
>
>> Thanks, If there any way i can achieve  this
>>
>>
>>
>>         {Browser}
>>                  |
>>                  |
>>                  |   (SIP-WS or verto)
>>             (iptel or sip2sip)
>>                  |-  -   -  -  - -  -  - - - - -[freeswitch]- - - - - - -
>> - -  -  -   [Register]
>>
>>
>>
>>
>> On Sat, Jul 18, 2015 at 1:42 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>>
>>> Both mod_verto and SIP-WS are signalling mechanisms for WebRTC. The
>>> former is a FreeSWITCH-proprietary protocol, the latter is a
>>> standards-based approach. The former is probably easier to understand if
>>> you've never done telephony work, the latter requires you to know more
>>> stuff.
>>>
>>> On 17 July 2015 at 16:52, Murugan Pandian <manpower13.cse at gmail.com>
>>> wrote:
>>>
>>>> Hi,
>>>>
>>>>    Any one can help me what is difference between mod_verto and
>>>>  WebRTC(ws/wss) mod.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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 victor.medina at cibersys.com  Sat Jul 18 01:09:17 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Fri, 17 Jul 2015 16:39:17 -0430
Subject: [Freeswitch-users] Mod_verto vs WebRTC(ws/wss)
In-Reply-To: <CAMRuXGheGqw9Q5fAzndjyjxS8P3abd2o6bK0Vv_C-i41hwqZOA@mail.gmail.com>
References: <CAMRuXGjVxqiuT75WhKE9nWtSXewRYtRxh-LCu+g9Tac7EvQXWg@mail.gmail.com>
	<CAAyX+KFJCswqb1k7WN86udhAxuBza9RTLjYat695aG6cqAR-dQ@mail.gmail.com>
	<CAMRuXGj5Z8LtuRd8_WLR5cy7oQO9a4p32tgZf=L6x0s=GTtoSA@mail.gmail.com>
	<CAMRuXGheGqw9Q5fAzndjyjxS8P3abd2o6bK0Vv_C-i41hwqZOA@mail.gmail.com>
Message-ID: <CACXsapOFUS20qh2NPGgXkT-ck+CFWsUAiL+=L5PuqnkRNMUR_g@mail.gmail.com>

FreeSWITCH will do both without much problems.

Decide what to use. Your choice.
 El 17/07/2015 16:02, "Murugan Pandian" <manpower13.cse at gmail.com> escribi?:

> Sorry for Diagram above one not clear
>
>
>          {Browser}
>                  |
>                  |
>    (SIP-WS or verto)
>                  |
>         [freeswitch]
>                  |
>                  |
>          [Register]
>         (iptel or sip2sip)
>
> On Sat, Jul 18, 2015 at 1:54 AM, Murugan Pandian <manpower13.cse at gmail.com
> > wrote:
>
>> Thanks, If there any way i can achieve  this
>>
>>
>>
>>         {Browser}
>>                  |
>>                  |
>>                  |   (SIP-WS or verto)
>>             (iptel or sip2sip)
>>                  |-  -   -  -  - -  -  - - - - -[freeswitch]- - - - - - -
>> - -  -  -   [Register]
>>
>>
>>
>>
>> On Sat, Jul 18, 2015 at 1:42 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>>
>>> Both mod_verto and SIP-WS are signalling mechanisms for WebRTC. The
>>> former is a FreeSWITCH-proprietary protocol, the latter is a
>>> standards-based approach. The former is probably easier to understand if
>>> you've never done telephony work, the latter requires you to know more
>>> stuff.
>>>
>>> On 17 July 2015 at 16:52, Murugan Pandian <manpower13.cse at gmail.com>
>>> wrote:
>>>
>>>> Hi,
>>>>
>>>>    Any one can help me what is difference between mod_verto and
>>>>  WebRTC(ws/wss) mod.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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 gmaruzz at gmail.com  Sat Jul 18 01:13:41 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 17 Jul 2015 23:13:41 +0200
Subject: [Freeswitch-users] Mod_verto vs WebRTC(ws/wss)
In-Reply-To: <CACXsapOFUS20qh2NPGgXkT-ck+CFWsUAiL+=L5PuqnkRNMUR_g@mail.gmail.com>
References: <CAMRuXGjVxqiuT75WhKE9nWtSXewRYtRxh-LCu+g9Tac7EvQXWg@mail.gmail.com>
	<CAAyX+KFJCswqb1k7WN86udhAxuBza9RTLjYat695aG6cqAR-dQ@mail.gmail.com>
	<CAMRuXGj5Z8LtuRd8_WLR5cy7oQO9a4p32tgZf=L6x0s=GTtoSA@mail.gmail.com>
	<CAMRuXGheGqw9Q5fAzndjyjxS8P3abd2o6bK0Vv_C-i41hwqZOA@mail.gmail.com>
	<CACXsapOFUS20qh2NPGgXkT-ck+CFWsUAiL+=L5PuqnkRNMUR_g@mail.gmail.com>
Message-ID: <CALXCt0p5cMzrY-PhgfkJceDFkV365XkkFQw3KzfMAacNnkcD8g@mail.gmail.com>

I have understood from your other mail to list that you want to use a
sip2sip account from browser.

You can then use sip.js to connect to a kamailio that goes to sip2sip, or
you can use verto to connect to freeswitch that will use your sip2sip
account as gateway.

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On Jul 17, 2015 11:10 PM, "Victor Medina" <victor.medina at cibersys.com>
wrote:

> FreeSWITCH will do both without much problems.
>
> Decide what to use. Your choice.
>  El 17/07/2015 16:02, "Murugan Pandian" <manpower13.cse at gmail.com>
> escribi?:
>
>> Sorry for Diagram above one not clear
>>
>>
>>          {Browser}
>>                  |
>>                  |
>>    (SIP-WS or verto)
>>                  |
>>         [freeswitch]
>>                  |
>>                  |
>>          [Register]
>>         (iptel or sip2sip)
>>
>> On Sat, Jul 18, 2015 at 1:54 AM, Murugan Pandian <
>> manpower13.cse at gmail.com> wrote:
>>
>>> Thanks, If there any way i can achieve  this
>>>
>>>
>>>
>>>         {Browser}
>>>                  |
>>>                  |
>>>                  |   (SIP-WS or verto)
>>>             (iptel or sip2sip)
>>>                  |-  -   -  -  - -  -  - - - - -[freeswitch]- - - - - -
>>> - - -  -  -   [Register]
>>>
>>>
>>>
>>>
>>> On Sat, Jul 18, 2015 at 1:42 AM, Ben Langfeld <ben at langfeld.co.uk>
>>> wrote:
>>>
>>>> Both mod_verto and SIP-WS are signalling mechanisms for WebRTC. The
>>>> former is a FreeSWITCH-proprietary protocol, the latter is a
>>>> standards-based approach. The former is probably easier to understand if
>>>> you've never done telephony work, the latter requires you to know more
>>>> stuff.
>>>>
>>>> On 17 July 2015 at 16:52, Murugan Pandian <manpower13.cse at gmail.com>
>>>> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>>    Any one can help me what is difference between mod_verto and
>>>>>  WebRTC(ws/wss) mod.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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 mike at jerris.com  Sat Jul 18 01:29:23 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 17 Jul 2015 17:29:23 -0400
Subject: [Freeswitch-users] Is there a way to temporarily disable DTMF
	controls during an eavesdrop (or control an eavesdrop session
	with non-DTMF stimulus)?
In-Reply-To: <DM2PR12MB0252EA8D47B0463D4BD007308A980@DM2PR12MB0252.namprd12.prod.outlook.com>
References: <DM2PR12MB0252EA8D47B0463D4BD007308A980@DM2PR12MB0252.namprd12.prod.outlook.com>
Message-ID: <BEF2129F-E2D2-4F0A-AD27-E4871F8FF147@jerris.com>

[master e0ef319] FS-7846: [mod_dptools] add eavesdrop_whisper_aleg=true and eavesdrop_whisper_bleg=true channel variables to allow you to start eavesdrop in whisper mode of specific call leg


> On Jul 17, 2015, at 4:35 PM, Steven Szeto <steven.szeto at mitel.com> wrote:
> 
> Michael Jerris asks:
>  
> does it work if you do the set before the eavesdrop?  That set is not running until after the eavesdrop ends.
>  
> Answer:
>  
> Yes, set:eavesdrop_enable_dtmf=false, does work if it is invoked before the eavesdrop. However there are two issues:
>  
> ?         The ?queue_dtmf? command is now ignored, which means the eavesdrop session starts as silent monitor instead of a whisper/coach session
> ?         The attempt to set eavesdrop_enable_dtmf=true does not seem to reopen the DTMF listening capabilities of the eavesdrop session. DTMF keystrokes 0,1,2,3 have no effect.
>  
> The solution for me would be this:
>  
> ?         Before the eavesdrop session is started, , set:eavesdrop_enable_dtmf=false
> ?         Introduce the ability to change the eavesdrop mode via the command line
> o   E.g. eavesdrop_change_mode <uuid> [0,1,2,3]
> o   Where [0,1,2,3] are the following eavesdrop modes respectively: silent monitor, coach the a-leg, coach the b-leg, barge-in conference
> From: Steven Szeto 
> Sent: Friday, July 17, 2015 4:01 PM
> To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
> Subject: Is there a way to temporarily disable DTMF controls during an eavesdrop (or control an eavesdrop session with non-DTMF stimulus)?
>  
> I can successfully create a whisper/coach eavesdrop session with the following commands issued from fs_cli:
>  
> originate sofia/internal/5401 at 10.47.41.109 <mailto:sofia/internal/5401 at 10.47.41.109> &bridge(sofia/internal/5901 at 10.47.41.109 <mailto:sofia/internal/5901 at 10.47.41.109>)
>  
> whisper/coach:
> originate sofia/internal/5902 at 10.47.41.109 <mailto:sofia/internal/5902 at 10.47.41.109> 'queue_dtmf:w2 at 500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861' inline
>  
> I would like to temporarily disable DTMF so that the supervisor can not change the eavesdrop mode via DTMF, but this does not work:
>  
> originate sofia/internal/5902 at 10.47.41.109 <mailto:sofia/internal/5902 at 10.47.41.109> 'queue_dtmf:w2 at 500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861, set:eavesdrop_enable_dtmf=false' inline
>  
> Moreover, I have a GUI app that presents the supervisor with the ability to change the eavesdrop mode. When he presses the appropriate icon in the GUI, I would then like to reenable DTMF and change the eavesdrop mode on the session:
>  
> Uuid_setvar a28739d0-00f0-4a59-8c82-7a5a74ab6861 eavesdrop_enable_dtmf true;
> Uuid_recv_dtmf a28739d0-00f0-4a59-8c82-7a5a74ab6861 w3 at 500
>  
> Is there another way to change the mode of an eavesdrop session?
>  
>  
> 
> NOTE: This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message.
> 
> _________________________________________________________________________
> 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 yehavi.bourvine at gmail.com  Sat Jul 18 07:40:47 2015
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 18 Jul 2015 06:40:47 +0300
Subject: [Freeswitch-users] shared line or other suggestion
In-Reply-To: <B0AD74FC10B94C6DBFCDB0F34F2A9112@bob>
References: <B0AD74FC10B94C6DBFCDB0F34F2A9112@bob>
Message-ID: <CAAtWKeGJJvNWkKsO9wPWVk6PwijXwzdt7U14cTCHF-gj3=inWg@mail.gmail.com>

Here is works ok in this situation. can you please give an example of what
exactly happens, what you see and what you expect to see?

      Regards, __Yehavi:

2015-07-17 16:41 GMT+03:00 Jason Holden <jason.holden at start.ca>:

>  I am currently trying to have the ability to place a call on hold and be
> able to pick the call up from a displayed list on a group of  Polycom
> phones.
>
> I am currently using shared line but when the main phone places a call on
> hold the shared lines to not show the held call with the remote user CID
> but the primary phonesCID information.
>
> Does anyone have a recommendation that?ll work better.
>
> I want to be able to see the remote CID of the caller who is on hold.
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> 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 smontour at twc.com  Fri Jul 17 22:31:42 2015
From: smontour at twc.com (Admin)
Date: Fri, 17 Jul 2015 13:31:42 -0500
Subject: [Freeswitch-users] ESL PHP error
Message-ID: <1437157902.2460.14.camel@twc.com>

I am getting the following error when trying to test an inbound
connection to FS using ESL with PHP. 

---------------------------------------------------------------
> php test.php 
PHP Warning:  dl(): Dynamically loaded extensions aren't enabled
in /usr/share/php/ESL.php on line 22
PHP Fatal error:  Call to undefined function new_ESLconnection()
in /usr/share/php/ESL.php on line 155
----------------------------------------------------------------

The installation (make phpmod-install) went just fine with no errors.
It copied ESL.so to '/usr/lib/php5/20131226' and created esl.ini under
"/etc/php5/apache2/conf.d/esl.ini"

I had ESL php working last year when I was testing a php application.
But then I was running Debian 7.1 and PHP 5.5.

Now i am using the following:
FreeSWITCH Version 1.4.20+git~20150703T164215Z~b95362f965~64bit
PHP 5.6.9-0+deb8u1 (cli) 
Debian 8.1 64-bit

Any help is very much appreciated. Thanks.





From get.prashant.007 at gmail.com  Fri Jul 17 23:41:12 2015
From: get.prashant.007 at gmail.com (Prashant Choudhary)
Date: Sat, 18 Jul 2015 01:11:12 +0530
Subject: [Freeswitch-users] mod_managed vs eventsocket
Message-ID: <CAFgB_10ZtaKSdQ-rqew3LZO79h9zWFFYi9qo7E_gpDASSrU+QQ@mail.gmail.com>

Hello All,

I have worked with LUA and freeswitch. Now I want to migrate to .Net API as
I want to experiment with machine learning using sample calls.  I am trying
to control around some 2-300 calls directly through my API. This may
require the call audio stream to be forwarded in real time to .NET
 I a few questions that might be very silly.

1. Which one is better to achieve this, mod_managed or ESL.
2. I was able to find some examples related to eventsocket, but same for
mod_Managed was rare. Can you suggest me any doc(of-course except the wiki
link) or any email chain which explains this or anything else. This will be
really a big help for me. As currently I am stuck. And I still find the LUA
way easier.

Thanks
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From get.prashant.007 at gmail.com  Sat Jul 18 13:27:04 2015
From: get.prashant.007 at gmail.com (Prashant Choudhary)
Date: Sat, 18 Jul 2015 14:57:04 +0530
Subject: [Freeswitch-users] mod_managed vs eventsocket
In-Reply-To: <CAFgB_10ZtaKSdQ-rqew3LZO79h9zWFFYi9qo7E_gpDASSrU+QQ@mail.gmail.com>
References: <CAFgB_10ZtaKSdQ-rqew3LZO79h9zWFFYi9qo7E_gpDASSrU+QQ@mail.gmail.com>
Message-ID: <CAFgB_13syeAtznT78KE46xo7XSkt2LiSpRNTNHMdqTWmemq9ZQ@mail.gmail.com>

Sorry for my last mail, which was a little more generic. Thing is, I am
able to receive and manage Inbound calls. But I am unable to figure out,
how to make outbound calls through mod_managed. I was able to do the same
through event socket API. One benefit of event socket was that I could run
it as separate console application , but I am not sure whether it is
possible through mod managed as well.
SO again my question remains the same. But it is more towards outbound
calling.

On Sat, Jul 18, 2015 at 1:11 AM, Prashant Choudhary <
get.prashant.007 at gmail.com> wrote:

> Hello All,
>
> I have worked with LUA and freeswitch. Now I want to migrate to .Net API
> as I want to experiment with machine learning using sample calls.  I am
> trying to control around some 2-300 calls directly through my API. This may
> require the call audio stream to be forwarded in real time to .NET
>  I a few questions that might be very silly.
>
> 1. Which one is better to achieve this, mod_managed or ESL.
> 2. I was able to find some examples related to eventsocket, but same for
> mod_Managed was rare. Can you suggest me any doc(of-course except the wiki
> link) or any email chain which explains this or anything else. This will be
> really a big help for me. As currently I am stuck. And I still find the LUA
> way easier.
>
> Thanks
>
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From idokan at gmail.com  Sat Jul 18 18:03:25 2015
From: idokan at gmail.com (ik)
Date: Sat, 18 Jul 2015 17:03:25 +0300
Subject: [Freeswitch-users] ESL PHP error
In-Reply-To: <1437157902.2460.14.camel@twc.com>
References: <1437157902.2460.14.camel@twc.com>
Message-ID: <CAOccsKDQCgVYdSdVWutsXpwKX=MzVHKcYwm-Zai8D49upFEJsg@mail.gmail.com>

If you are using the embedded version of PHP support that arrives with FS,
you must set a whole list of environment stuff.
You first need to have the esl.so file loaded, but only if you have dl.so
enabled in php.ini or php.d you'll be able to do so.

I used a simpler version of ESL, that was based on FusionPBX and using pure
TCP sockets rather then the shard library, and it works great, but you
require to do additional work yourself.

Ido

On Fri, Jul 17, 2015 at 9:31 PM, Admin <smontour at twc.com> wrote:

> I am getting the following error when trying to test an inbound
> connection to FS using ESL with PHP.
>
> ---------------------------------------------------------------
> > php test.php
> PHP Warning:  dl(): Dynamically loaded extensions aren't enabled
> in /usr/share/php/ESL.php on line 22
> PHP Fatal error:  Call to undefined function new_ESLconnection()
> in /usr/share/php/ESL.php on line 155
> ----------------------------------------------------------------
>
> The installation (make phpmod-install) went just fine with no errors.
> It copied ESL.so to '/usr/lib/php5/20131226' and created esl.ini under
> "/etc/php5/apache2/conf.d/esl.ini"
>
> I had ESL php working last year when I was testing a php application.
> But then I was running Debian 7.1 and PHP 5.5.
>
> Now i am using the following:
> FreeSWITCH Version 1.4.20+git~20150703T164215Z~b95362f965~64bit
> PHP 5.6.9-0+deb8u1 (cli)
> Debian 8.1 64-bit
>
> Any help is very much appreciated. Thanks.
>
>
>
>
> _________________________________________________________________________
> 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 ben at langfeld.co.uk  Sat Jul 18 18:18:53 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Sat, 18 Jul 2015 11:18:53 -0300
Subject: [Freeswitch-users] Mod_verto vs WebRTC(ws/wss)
In-Reply-To: <CALXCt0p5cMzrY-PhgfkJceDFkV365XkkFQw3KzfMAacNnkcD8g@mail.gmail.com>
References: <CAMRuXGjVxqiuT75WhKE9nWtSXewRYtRxh-LCu+g9Tac7EvQXWg@mail.gmail.com>
	<CAAyX+KFJCswqb1k7WN86udhAxuBza9RTLjYat695aG6cqAR-dQ@mail.gmail.com>
	<CAMRuXGj5Z8LtuRd8_WLR5cy7oQO9a4p32tgZf=L6x0s=GTtoSA@mail.gmail.com>
	<CAMRuXGheGqw9Q5fAzndjyjxS8P3abd2o6bK0Vv_C-i41hwqZOA@mail.gmail.com>
	<CACXsapOFUS20qh2NPGgXkT-ck+CFWsUAiL+=L5PuqnkRNMUR_g@mail.gmail.com>
	<CALXCt0p5cMzrY-PhgfkJceDFkV365XkkFQw3KzfMAacNnkcD8g@mail.gmail.com>
Message-ID: <CAAyX+KGgTLYJ5vatqF-L+u2YU3W_FxuVf9QQAgHvGDx5wzPO6w@mail.gmail.com>

To clarify, if you want your WebRTC UA specifically to register to sip2sip,
then you must use SIP all the way through. A Verto endpoint cannot register
with a SIP registrar.

On 17 July 2015 at 18:13, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> I have understood from your other mail to list that you want to use a
> sip2sip account from browser.
>
> You can then use sip.js to connect to a kamailio that goes to sip2sip, or
> you can use verto to connect to freeswitch that will use your sip2sip
> account as gateway.
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On Jul 17, 2015 11:10 PM, "Victor Medina" <victor.medina at cibersys.com>
> wrote:
>
>> FreeSWITCH will do both without much problems.
>>
>> Decide what to use. Your choice.
>>  El 17/07/2015 16:02, "Murugan Pandian" <manpower13.cse at gmail.com>
>> escribi?:
>>
>>> Sorry for Diagram above one not clear
>>>
>>>
>>>          {Browser}
>>>                  |
>>>                  |
>>>    (SIP-WS or verto)
>>>                  |
>>>         [freeswitch]
>>>                  |
>>>                  |
>>>          [Register]
>>>         (iptel or sip2sip)
>>>
>>> On Sat, Jul 18, 2015 at 1:54 AM, Murugan Pandian <
>>> manpower13.cse at gmail.com> wrote:
>>>
>>>> Thanks, If there any way i can achieve  this
>>>>
>>>>
>>>>
>>>>         {Browser}
>>>>                  |
>>>>                  |
>>>>                  |   (SIP-WS or verto)
>>>>               (iptel or sip2sip)
>>>>                  |-  -   -  -  - -  -  - - - - -[freeswitch]- - - - - -
>>>> - - -  -  -   [Register]
>>>>
>>>>
>>>>
>>>>
>>>> On Sat, Jul 18, 2015 at 1:42 AM, Ben Langfeld <ben at langfeld.co.uk>
>>>> wrote:
>>>>
>>>>> Both mod_verto and SIP-WS are signalling mechanisms for WebRTC. The
>>>>> former is a FreeSWITCH-proprietary protocol, the latter is a
>>>>> standards-based approach. The former is probably easier to understand if
>>>>> you've never done telephony work, the latter requires you to know more
>>>>> stuff.
>>>>>
>>>>> On 17 July 2015 at 16:52, Murugan Pandian <manpower13.cse at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>>    Any one can help me what is difference between mod_verto and
>>>>>>  WebRTC(ws/wss) mod.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> 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
>
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From chad at apartmentlines.com  Sun Jul 19 06:33:51 2015
From: chad at apartmentlines.com (Chad Phillips)
Date: Sat, 18 Jul 2015 19:33:51 -0700
Subject: [Freeswitch-users] Serving Verto HTML from a different server
Message-ID: <CAJ0sghQ1zkuHEQxUuB0sa2is2hwUk9XBJavQ9GoxJ9sWnDQuQQ@mail.gmail.com>

I?ve gotten fairly comfortable with the setup offered in the Confluence
wiki for the new Verto video features.

I?m curious to know if I can expect any gotchas if I serve the Verto web
stuff from a different server than the FreeSWITCH instance is running on.
>From what I can tell, as long as I make sure to use the same server certs
on both servers, it *should* work.

Is there anything I?m not taking into consideration? Any better way to
approach the problem besides certificate sharing across servers?

Thanks,

Chad
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From smontour at twc.com  Sun Jul 19 20:46:22 2015
From: smontour at twc.com (Admin)
Date: Sun, 19 Jul 2015 11:46:22 -0500
Subject: [Freeswitch-users]  ESL PHP error
Message-ID: <1437324382.2736.6.camel@twc.com>

Thank you for responding to my question. I am using the standard PHP not
the embedded one. However, i edited the php.ini file and set "enable_dl"
parameter to 'on' but still getting the same error. I am starting to
think that this issue is related to PHP version 5.6 and may be a reswig
is needed as it was mentioned in one the archived posts.



From jack at livecall.com  Sun Jul 19 23:16:28 2015
From: jack at livecall.com (Jack)
Date: Sun, 19 Jul 2015 12:16:28 -0700
Subject: [Freeswitch-users] Serving Verto HTML from a different server
In-Reply-To: <CAJ0sghQ1zkuHEQxUuB0sa2is2hwUk9XBJavQ9GoxJ9sWnDQuQQ@mail.gmail.com>
References: <CAJ0sghQ1zkuHEQxUuB0sa2is2hwUk9XBJavQ9GoxJ9sWnDQuQQ@mail.gmail.com>
Message-ID: <55ABF78C.6000500@livecall.com>

Chad,
I was able to get this to work under windows environment.  I set up a 
secure server www.xxxx.com to serve the html and then pointed 
fs.xxxx.com at my FreeSwitch server.
My certificates were "domain validated" so I had two certificates, one 
for each domain/server .  You could probably get a wildcard certificate 
and use the same certificate on both servers.   The main reason I used 
"domain validated" was I was able to purchase them for $5 a year each.
Jack
On 7/18/2015 7:33 PM, Chad Phillips wrote:
> I?ve gotten fairly comfortable with the setup offered in the 
> Confluence wiki for the new Verto video features.
>
> I?m curious to know if I can expect any gotchas if I serve the Verto 
> web stuff from a different server than the FreeSWITCH instance is 
> running on. From what I can tell, as long as I make sure to use the 
> same server certs on both servers, it *should* work.
>
> Is there anything I?m not taking into consideration? Any better way to 
> approach the problem besides certificate sharing across servers?
>
> Thanks,
>
> Chad
>
>
> _________________________________________________________________________
> 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
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2014.0.4821 / Virus Database: 4365/10262 - Release Date: 07/18/15
>

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From get.prashant.007 at gmail.com  Mon Jul 20 00:24:44 2015
From: get.prashant.007 at gmail.com (Prashant Choudhary)
Date: Mon, 20 Jul 2015 01:54:44 +0530
Subject: [Freeswitch-users] mod_managed vs eventsocket
In-Reply-To: <CAFgB_13syeAtznT78KE46xo7XSkt2LiSpRNTNHMdqTWmemq9ZQ@mail.gmail.com>
References: <CAFgB_10ZtaKSdQ-rqew3LZO79h9zWFFYi9qo7E_gpDASSrU+QQ@mail.gmail.com>
	<CAFgB_13syeAtznT78KE46xo7XSkt2LiSpRNTNHMdqTWmemq9ZQ@mail.gmail.com>
Message-ID: <CAFgB_11tki=JHoAqm+UAj0NhjkYBvSzDM2Kab47biWfjY9Qypg@mail.gmail.com>

Hello,
Now I have even managed to do outbound calls. I have a real problem now.
How can I get the audio stream from the person on the other end. I want the
audio stream to be sent to my API in real time. I want to detect silence as
well, but we can do that by experiment. Only thing is that I am unable to
get the audio stream. And this stream should only have the voice of person
being called, not mine(i.e. the incoming audio). Is it possible??
Please help

On Sat, Jul 18, 2015 at 2:57 PM, Prashant Choudhary <
get.prashant.007 at gmail.com> wrote:

> Sorry for my last mail, which was a little more generic. Thing is, I am
> able to receive and manage Inbound calls. But I am unable to figure out,
> how to make outbound calls through mod_managed. I was able to do the same
> through event socket API. One benefit of event socket was that I could run
> it as separate console application , but I am not sure whether it is
> possible through mod managed as well.
> SO again my question remains the same. But it is more towards outbound
> calling.
>
> On Sat, Jul 18, 2015 at 1:11 AM, Prashant Choudhary <
> get.prashant.007 at gmail.com> wrote:
>
>> Hello All,
>>
>> I have worked with LUA and freeswitch. Now I want to migrate to .Net API
>> as I want to experiment with machine learning using sample calls.  I am
>> trying to control around some 2-300 calls directly through my API. This may
>> require the call audio stream to be forwarded in real time to .NET
>>  I a few questions that might be very silly.
>>
>> 1. Which one is better to achieve this, mod_managed or ESL.
>> 2. I was able to find some examples related to eventsocket, but same for
>> mod_Managed was rare. Can you suggest me any doc(of-course except the wiki
>> link) or any email chain which explains this or anything else. This will be
>> really a big help for me. As currently I am stuck. And I still find the LUA
>> way easier.
>>
>> Thanks
>>
>
>
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From steven.szeto at mitel.com  Mon Jul 20 05:41:47 2015
From: steven.szeto at mitel.com (Steven Szeto)
Date: Mon, 20 Jul 2015 01:41:47 +0000
Subject: [Freeswitch-users] Is there a way to temporarily disable
	DTMF	controls during an eavesdrop (or control an eavesdrop
	session	with non-DTMF stimulus)?
In-Reply-To: <BEF2129F-E2D2-4F0A-AD27-E4871F8FF147@jerris.com>
References: <DM2PR12MB0252EA8D47B0463D4BD007308A980@DM2PR12MB0252.namprd12.prod.outlook.com>,
	<BEF2129F-E2D2-4F0A-AD27-E4871F8FF147@jerris.com>
Message-ID: <BY2PR12MB02424D8823DFAE9DF3AA1D2A8A850@BY2PR12MB0242.namprd12.prod.outlook.com>

Once an eavedrop session is established (e.g. a whisper/coach session), is there a way to change the eavesdrop mode to say barge-in conference (recall that we have disabled DTMF keystrokes)?


________________________________
From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Michael Jerris <mike at jerris.com>
Sent: Friday, July 17, 2015 5:29 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Is there a way to temporarily disable DTMF controls during an eavesdrop (or control an eavesdrop session with non-DTMF stimulus)?

[master e0ef319] FS-7846: [mod_dptools] add eavesdrop_whisper_aleg=true and eavesdrop_whisper_bleg=true channel variables to allow you to start eavesdrop in whisper mode of specific call leg


On Jul 17, 2015, at 4:35 PM, Steven Szeto <steven.szeto at mitel.com<mailto:steven.szeto at mitel.com>> wrote:

Michael Jerris asks:

does it work if you do the set before the eavesdrop?  That set is not running until after the eavesdrop ends.

Answer:

Yes, set:eavesdrop_enable_dtmf=false, does work if it is invoked before the eavesdrop. However there are two issues:

?         The ?queue_dtmf? command is now ignored, which means the eavesdrop session starts as silent monitor instead of a whisper/coach session
?         The attempt to set eavesdrop_enable_dtmf=true does not seem to reopen the DTMF listening capabilities of the eavesdrop session. DTMF keystrokes 0,1,2,3 have no effect.

The solution for me would be this:

?         Before the eavesdrop session is started, , set:eavesdrop_enable_dtmf=false
?         Introduce the ability to change the eavesdrop mode via the command line
o   E.g. eavesdrop_change_mode <uuid> [0,1,2,3]
o   Where [0,1,2,3] are the following eavesdrop modes respectively: silent monitor, coach the a-leg, coach the b-leg, barge-in conference
From: Steven Szeto
Sent: Friday, July 17, 2015 4:01 PM
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Is there a way to temporarily disable DTMF controls during an eavesdrop (or control an eavesdrop session with non-DTMF stimulus)?

I can successfully create a whisper/coach eavesdrop session with the following commands issued from fs_cli:

originate sofia/internal/5401 at 10.47.41.109<mailto:sofia/internal/5401 at 10.47.41.109> &bridge(sofia/internal/5901 at 10.47.41.109<mailto:sofia/internal/5901 at 10.47.41.109>)

whisper/coach:
originate sofia/internal/5902 at 10.47.41.109<mailto:sofia/internal/5902 at 10.47.41.109> 'queue_dtmf:w2 at 500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861' inline

I would like to temporarily disable DTMF so that the supervisor can not change the eavesdrop mode via DTMF, but this does not work:

originate sofia/internal/5902 at 10.47.41.109<mailto:sofia/internal/5902 at 10.47.41.109> 'queue_dtmf:w2 at 500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861, set:eavesdrop_enable_dtmf=false' inline

Moreover, I have a GUI app that presents the supervisor with the ability to change the eavesdrop mode. When he presses the appropriate icon in the GUI, I would then like to reenable DTMF and change the eavesdrop mode on the session:

Uuid_setvar a28739d0-00f0-4a59-8c82-7a5a74ab6861 eavesdrop_enable_dtmf true;
Uuid_recv_dtmf a28739d0-00f0-4a59-8c82-7a5a74ab6861 w3 at 500

Is there another way to change the mode of an eavesdrop session?



________________________________
NOTE: This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message.

_________________________________________________________________________
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://lists.freeswitch.org/mailman/options/freeswitch-users>
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From mike at jerris.com  Mon Jul 20 09:02:31 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 20 Jul 2015 01:02:31 -0400
Subject: [Freeswitch-users] Is there a way to temporarily disable DTMF
 controls during an eavesdrop (or control an eavesdrop session with non-DTMF
 stimulus)?
In-Reply-To: <BY2PR12MB02424D8823DFAE9DF3AA1D2A8A850@BY2PR12MB0242.namprd12.prod.outlook.com>
References: <DM2PR12MB0252EA8D47B0463D4BD007308A980@DM2PR12MB0252.namprd12.prod.outlook.com>
	<BEF2129F-E2D2-4F0A-AD27-E4871F8FF147@jerris.com>
	<BY2PR12MB02424D8823DFAE9DF3AA1D2A8A850@BY2PR12MB0242.namprd12.prod.outlook.com>
Message-ID: <CAEwTsAhZ1QwFxd4kALwBTCywY5g8roUkHzgkxHTL=5cjAHN9SA@mail.gmail.com>

Not currently, no.

On Sunday, July 19, 2015, Steven Szeto <steven.szeto at mitel.com> wrote:

>  Once an eavedrop session is established (e.g. a whisper/coach
> session), is there a way to change the eavesdrop mode to say barge-in
> conference (recall that we have disabled DTMF keystrokes)?
>
>
>  ------------------------------
> *From:* freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>
> <freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>>
> on behalf of Michael Jerris <mike at jerris.com
> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>>
> *Sent:* Friday, July 17, 2015 5:29 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Is there a way to temporarily disable
> DTMF controls during an eavesdrop (or control an eavesdrop session with
> non-DTMF stimulus)?
>
>   [master e0ef319] FS-7846: [mod_dptools] add eavesdrop_whisper_aleg=true
> and eavesdrop_whisper_bleg=true channel variables to allow you to start
> eavesdrop in whisper mode of specific call leg
>
>
>   On Jul 17, 2015, at 4:35 PM, Steven Szeto <steven.szeto at mitel.com
> <javascript:_e(%7B%7D,'cvml','steven.szeto at mitel.com');>> wrote:
>
>   Michael Jerris asks:
>
>  does it work if you do the set before the eavesdrop?  That set is not
> running until after the eavesdrop ends.
>
>  Answer:
>
>  Yes, set:eavesdrop_enable_dtmf=false, does work if it is invoked before
> the eavesdrop. However there are two issues:
>
>  ?         The ?queue_dtmf? command is now ignored, which means the
> eavesdrop session starts as silent monitor instead of a whisper/coach
> session
>  ?         The attempt to set eavesdrop_enable_dtmf=true does not seem to
> reopen the DTMF listening capabilities of the eavesdrop session. DTMF
> keystrokes 0,1,2,3 have no effect.
>
>  The solution for me would be this:
>
>  ?         Before the eavesdrop session is started, ,
> set:eavesdrop_enable_dtmf=false
>  ?         Introduce the ability to change the eavesdrop mode via the
> command line
>  o   E.g. eavesdrop_change_mode <uuid> [0,1,2,3]
>  o   Where [0,1,2,3] are the following eavesdrop modes respectively:
> silent monitor, coach the a-leg, coach the b-leg, barge-in conference
>   *From:* Steven Szeto
> *Sent:* Friday, July 17, 2015 4:01 PM
> *To:* 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users at lists.freeswitch.org');>>
> *Subject:* Is there a way to temporarily disable DTMF controls during an
> eavesdrop (or control an eavesdrop session with non-DTMF stimulus)?
>
>  I can successfully create a whisper/coach eavesdrop session with the
> following commands issued from fs_cli:
>
>  originate sofia/internal/5401 at 10.47.41.109
> <javascript:_e(%7B%7D,'cvml','sofia/internal/5401 at 10.47.41.109');>
> &bridge(sofia/internal/5901 at 10.47.41.109
> <javascript:_e(%7B%7D,'cvml','sofia/internal/5901 at 10.47.41.109');>)
>
>  whisper/coach:
>  originate sofia/internal/5902 at 10.47.41.109
> <javascript:_e(%7B%7D,'cvml','sofia/internal/5902 at 10.47.41.109');>
> 'queue_dtmf:w2 at 500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861' inline
>
>  I would like to temporarily disable DTMF so that the supervisor can not
> change the eavesdrop mode via DTMF, but this does not work:
>
>  originate sofia/internal/5902 at 10.47.41.109
> <javascript:_e(%7B%7D,'cvml','sofia/internal/5902 at 10.47.41.109');>
> 'queue_dtmf:w2 at 500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861,
> set:eavesdrop_enable_dtmf=false' inline
>
>  Moreover, I have a GUI app that presents the supervisor with the ability
> to change the eavesdrop mode. When he presses the appropriate icon in the
> GUI, I would then like to reenable DTMF and change the eavesdrop mode on
> the session:
>
>  Uuid_setvar a28739d0-00f0-4a59-8c82-7a5a74ab6861 eavesdrop_enable_dtmf
> true;
>  Uuid_recv_dtmf a28739d0-00f0-4a59-8c82-7a5a74ab6861 w3 at 500
>
>  Is there another way to change the mode of an eavesdrop session?
>
>
>
>  ------------------------------
> NOTE: This e-mail (including any attachments) is for the sole use of the
> intended recipient(s) and may contain information that is confidential
> and/or protected by legal privilege. Any unauthorized review, use, copy,
> disclosure or distribution of this e-mail is strictly prohibited. If you
> are not the intended recipient, please notify Mitel immediately and destroy
> all copies of this e-mail. Mitel does not accept any liability for breach
> of security, error or virus that may result from the transmission of this
> message.
>
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','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
> <javascript:_e(%7B%7D,'cvml','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 Alexander.Haugg at c4b.de  Mon Jul 20 11:14:47 2015
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Mon, 20 Jul 2015 07:14:47 +0000
Subject: [Freeswitch-users] No SIP UPDATE will send
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC8517C1F33@vc4bex2.corp.c4b.de>

Hi all,
I have the problem, that the freeSwitch don't send a SIP UPDATE to the PBX.

Scenario:
PBX (Siemens H3K) is connected via SIP trunking to the FreeSwitch.
If I have two Legs from the H3K to the Freeswitch I'm using the uuid_bridge command to connect this two legs (works fine), but the call legs will not updated via SIP UPDATE. The PBX have no information to update the displays on the telephones.

The target number (toHeader) was the routing number from the H3K to the freeswitch. Now I need a SIP UPDATE from FreeSwitch to PBX to actualize the toHeader with the new partner number (the partner number is the fromHeader of the other leg).

In the sofia profile I was try it with the parameter <param name="send-display-update" value="true"/> and without this parameter (in the source code the default is "true")

What can I do to force an UPDATE to the call legs?

Version of the FreeSwitch
FreeSWITCH Version 1.5.15b+git~20150121T080154Z~90ab1d16f5~64bit (git 90ab1d1 2015-01-21 08:01:54Z 64bit)

In the INVITE from the PBX the ALLOW header includes the UPDATE
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, INFO, PRACK, UPDATE

Many thanks!

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From yadenis at seznam.cz  Mon Jul 20 11:34:28 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Mon, 20 Jul 2015 09:34:28 +0200
Subject: [Freeswitch-users] FreeSwitch CentOS 7 install
In-Reply-To: <CAEwTsAhZ1QwFxd4kALwBTCywY5g8roUkHzgkxHTL=5cjAHN9SA@mail.gmail.com>
References: <DM2PR12MB0252EA8D47B0463D4BD007308A980@DM2PR12MB0252.namprd12.prod.outlook.com>
	<BEF2129F-E2D2-4F0A-AD27-E4871F8FF147@jerris.com>
	<BY2PR12MB02424D8823DFAE9DF3AA1D2A8A850@BY2PR12MB0242.namprd12.prod.outlook.com>
	<CAEwTsAhZ1QwFxd4kALwBTCywY5g8roUkHzgkxHTL=5cjAHN9SA@mail.gmail.com>
Message-ID: <1447341811.20150720093428@seznam.cz>

Hi All,

I have not a big problem during installation FreeSwitch on CentOS7.

bootstrap: automake --no-force --add-missing --copy
bootstrap: rm -rf autom4te.cache
bootstrap: rm -rf autom4te.cache
bootstrap: rm -rf autom4te.cache
bootstrap: rm -rf autom4te.cache
tests/unit/Makefile.am:27: warning: error You must install libtap-dev to build these unit tests: non-POSIX variable name
tests/unit/Makefile.am:27: (probably a GNU make extension)


What "libtap-dev" ? And where can I get it?

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From yadenis at seznam.cz  Mon Jul 20 13:10:54 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Mon, 20 Jul 2015 11:10:54 +0200
Subject: [Freeswitch-users] CentOS 7 install instructions does not work
In-Reply-To: <CAEwTsAhZ1QwFxd4kALwBTCywY5g8roUkHzgkxHTL=5cjAHN9SA@mail.gmail.com>
References: <DM2PR12MB0252EA8D47B0463D4BD007308A980@DM2PR12MB0252.namprd12.prod.outlook.com>
	<BEF2129F-E2D2-4F0A-AD27-E4871F8FF147@jerris.com>
	<BY2PR12MB02424D8823DFAE9DF3AA1D2A8A850@BY2PR12MB0242.namprd12.prod.outlook.com>
	<CAEwTsAhZ1QwFxd4kALwBTCywY5g8roUkHzgkxHTL=5cjAHN9SA@mail.gmail.com>
Message-ID: <709511198.20150720111054@seznam.cz>

Hi All,


Clean installation CentOS 7. I am trying to install FreeSwitch according to instructions.

https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7 


Makefile:887: *** You must install libyuv-dev to build mod_fsv.  Stop.
make[5]: Leaving directory `/usr/src/freeswitch.git/src/mod/applications/mod_fsv

Where can I get this library?
Here only libyuv. But i need libyuv-dev
https://freeswitch.org/stash/projects/SD/repos/libyuv/browse 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From ben at langfeld.co.uk  Mon Jul 20 15:09:26 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Mon, 20 Jul 2015 08:09:26 -0300
Subject: [Freeswitch-users] CentOS 7 install instructions does not work
In-Reply-To: <709511198.20150720111054@seznam.cz>
References: <DM2PR12MB0252EA8D47B0463D4BD007308A980@DM2PR12MB0252.namprd12.prod.outlook.com>
	<BEF2129F-E2D2-4F0A-AD27-E4871F8FF147@jerris.com>
	<BY2PR12MB02424D8823DFAE9DF3AA1D2A8A850@BY2PR12MB0242.namprd12.prod.outlook.com>
	<CAEwTsAhZ1QwFxd4kALwBTCywY5g8roUkHzgkxHTL=5cjAHN9SA@mail.gmail.com>
	<709511198.20150720111054@seznam.cz>
Message-ID: <CAAyX+KH764HK8bG6y_Tdb_1w0U04CpZov6ZKwbFe5UtCmr9EKg@mail.gmail.com>

Did you try `yum install libyuv-devel`?

On 20 July 2015 at 06:10, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Hi All,
>
>
> Clean installation CentOS 7. I am trying to install FreeSwitch according
> to instructions.
>
> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7
>
>
> Makefile:887: *** You must install libyuv-dev to build mod_fsv.  Stop.
> make[5]: Leaving directory
> `/usr/src/freeswitch.git/src/mod/applications/mod_fsv
>
> Where can I get this library?
> Here only libyuv. But i need libyuv-dev
> https://freeswitch.org/stash/projects/SD/repos/libyuv/browse
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 *
>
> _________________________________________________________________________
> 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 yadenis at seznam.cz  Mon Jul 20 15:27:31 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Mon, 20 Jul 2015 13:27:31 +0200
Subject: [Freeswitch-users] CentOS 7 install instructions does not work
In-Reply-To: <CAAyX+KH764HK8bG6y_Tdb_1w0U04CpZov6ZKwbFe5UtCmr9EKg@mail.gmail.com>
References: <DM2PR12MB0252EA8D47B0463D4BD007308A980@DM2PR12MB0252.namprd12.prod.outlook.com>
	<BEF2129F-E2D2-4F0A-AD27-E4871F8FF147@jerris.com>
	<BY2PR12MB02424D8823DFAE9DF3AA1D2A8A850@BY2PR12MB0242.namprd12.prod.outlook.com>
	<CAEwTsAhZ1QwFxd4kALwBTCywY5g8roUkHzgkxHTL=5cjAHN9SA@mail.gmail.com>
	<709511198.20150720111054@seznam.cz>
	<CAAyX+KH764HK8bG6y_Tdb_1w0U04CpZov6ZKwbFe5UtCmr9EKg@mail.gmail.com>
Message-ID: <86757220.20150720132731@seznam.cz>

Hi,

Of course!

yum install libyuv-devel
Loaded plugins: fastestmirror, refresh-packagekit, security
Setting up Install Process
Loading mirror speeds from cached hostfile
 * base: ftp.cvut.cz
 * extras: ftp.cvut.cz
 * rpmforge: mirror.us.leaseweb.net
 * updates: ftp.agh.edu.pl
No package libyuv-devel available.
Error: Nothing to do


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 20. ?ervence 2015, 13:09:26, napsal jste:


Did you try `yum install libyuv-devel`?

On 20 July 2015 at 06:10, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi All,


Clean installation CentOS 7. I am trying to install FreeSwitch according to instructions.

https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7 


Makefile:887: *** You must install libyuv-dev to build mod_fsv.  Stop.
make[5]: Leaving directory `/usr/src/freeswitch.git/src/mod/applications/mod_fsv

Where can I get this library?
Here only libyuv. But i need libyuv-dev
https://freeswitch.org/stash/projects/SD/repos/libyuv/browse 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382


_________________________________________________________________________
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 anthony.minessale at gmail.com  Mon Jul 20 16:55:27 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 20 Jul 2015 07:55:27 -0500
Subject: [Freeswitch-users] Serving Verto HTML from a different server
In-Reply-To: <CAJ0sghQ1zkuHEQxUuB0sa2is2hwUk9XBJavQ9GoxJ9sWnDQuQQ@mail.gmail.com>
References: <CAJ0sghQ1zkuHEQxUuB0sa2is2hwUk9XBJavQ9GoxJ9sWnDQuQQ@mail.gmail.com>
Message-ID: <CAKbxfG_JFqzMBb4JUujWh=r2bXaCNz14qpu04FB7m09=XTB3sw@mail.gmail.com>

The only requirement is that the cert on FS is trusted.  The html can come
from anywhere.

If its not a trusted cert the wss connection will silently fail.  This is
because there is no dialog asking you to trust the cert on wss connections
so they just PUNT and don't even log an error.  I opened a bug a year ago
mentioning ot and they said thats working as designed.

So the only way to make untrusted certs work is to make an https request to
the wss port one time and agree to manually trust it.  Therefore, serving
that same cert for the html serves the same perpose of manually trusting
the cert before making the wss connection.  With trusted certs, html and
wss can be entirelly different.

On Saturday, July 18, 2015, Chad Phillips <chad at apartmentlines.com> wrote:

> I?ve gotten fairly comfortable with the setup offered in the Confluence
> wiki for the new Verto video features.
>
> I?m curious to know if I can expect any gotchas if I serve the Verto web
> stuff from a different server than the FreeSWITCH instance is running on.
> From what I can tell, as long as I make sure to use the same server certs
> on both servers, it *should* work.
>
> Is there anything I?m not taking into consideration? Any better way to
> approach the problem besides certificate sharing across servers?
>
> Thanks,
>
> Chad
>


-- 
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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From telishisheer at gmail.com  Mon Jul 20 17:06:14 2015
From: telishisheer at gmail.com (Shisheer Teli)
Date: Mon, 20 Jul 2015 18:36:14 +0530
Subject: [Freeswitch-users] Voice call failed from softphone (X-lite) to
 Hard phone ( Grand stream GXP280 )
Message-ID: <CACkRGLyktL8bY0xYb7BPFFjY_cfUAozDLnFvy3jQt1K1d8JqVw@mail.gmail.com>

Hi Team,

I am using softphone: x lite and hard phone: Grand stream GXP280 with
FreeSWITCH.

Both phones registered with FreeSWITCH.

When I called from hard phone to the softphone, I am able to receive a call.

But if I called from softphone to hard phone it gives me an error (failed
to established call)


-- 
Regards,
Shisheer T
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From akibsayyed at gmail.com  Mon Jul 20 16:03:27 2015
From: akibsayyed at gmail.com (Akib Sayyed)
Date: Mon, 20 Jul 2015 17:33:27 +0530
Subject: [Freeswitch-users] CentOS 7 install instructions does not work
In-Reply-To: <86757220.20150720132731@seznam.cz>
References: <DM2PR12MB0252EA8D47B0463D4BD007308A980@DM2PR12MB0252.namprd12.prod.outlook.com>
	<BEF2129F-E2D2-4F0A-AD27-E4871F8FF147@jerris.com>
	<BY2PR12MB02424D8823DFAE9DF3AA1D2A8A850@BY2PR12MB0242.namprd12.prod.outlook.com>
	<CAEwTsAhZ1QwFxd4kALwBTCywY5g8roUkHzgkxHTL=5cjAHN9SA@mail.gmail.com>
	<709511198.20150720111054@seznam.cz>
	<CAAyX+KH764HK8bG6y_Tdb_1w0U04CpZov6ZKwbFe5UtCmr9EKg@mail.gmail.com>
	<86757220.20150720132731@seznam.cz>
Message-ID: <CAMG-S913P0S7NLrQXVDWK0-=z2hToWzA+t1jcs_n3xbR2TGcpQ@mail.gmail.com>

Please look for source

https://code.google.com/p/libyuv/wiki/GettingStarted

On Mon, Jul 20, 2015 at 4:57 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Hi,
>
> Of course!
>
> yum install libyuv-devel
> Loaded plugins: fastestmirror, refresh-packagekit, security
> Setting up Install Process
> Loading mirror speeds from cached hostfile
> * base: ftp.cvut.cz
> * extras: ftp.cvut.cz
> * rpmforge: mirror.us.leaseweb.net
> * updates: ftp.agh.edu.pl
> No package libyuv-devel available.
> Error: Nothing to do
>
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 pond?l? 20. ?ervence 2015, 13:09:26, napsal
> jste: *
>
>  Did you try `yum install libyuv-devel`?
>
> On 20 July 2015 at 06:10, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Hi All,
>
>
> Clean installation CentOS 7. I am trying to install FreeSwitch according
> to instructions.
>
> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7
>
>
> Makefile:887: *** You must install libyuv-dev to build mod_fsv.  Stop.
> make[5]: Leaving directory
> `/usr/src/freeswitch.git/src/mod/applications/mod_fsv
>
> Where can I get this library?
> Here only libyuv. But i need libyuv-dev
> https://freeswitch.org/stash/projects/SD/repos/libyuv/browse
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
>
> *. 775-415-382 *
> _________________________________________________________________________
> 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
>



-- 
Akib Sayyed
Matrix-Shell
akibsayyed at gmail.com
akibsayyed at matrixshell.com
Mob:- +91-966-514-2243
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From akibsayyed at gmail.com  Mon Jul 20 16:05:11 2015
From: akibsayyed at gmail.com (Akib Sayyed)
Date: Mon, 20 Jul 2015 17:35:11 +0530
Subject: [Freeswitch-users] CentOS 7 install instructions does not work
In-Reply-To: <CAMG-S913P0S7NLrQXVDWK0-=z2hToWzA+t1jcs_n3xbR2TGcpQ@mail.gmail.com>
References: <DM2PR12MB0252EA8D47B0463D4BD007308A980@DM2PR12MB0252.namprd12.prod.outlook.com>
	<BEF2129F-E2D2-4F0A-AD27-E4871F8FF147@jerris.com>
	<BY2PR12MB02424D8823DFAE9DF3AA1D2A8A850@BY2PR12MB0242.namprd12.prod.outlook.com>
	<CAEwTsAhZ1QwFxd4kALwBTCywY5g8roUkHzgkxHTL=5cjAHN9SA@mail.gmail.com>
	<709511198.20150720111054@seznam.cz>
	<CAAyX+KH764HK8bG6y_Tdb_1w0U04CpZov6ZKwbFe5UtCmr9EKg@mail.gmail.com>
	<86757220.20150720132731@seznam.cz>
	<CAMG-S913P0S7NLrQXVDWK0-=z2hToWzA+t1jcs_n3xbR2TGcpQ@mail.gmail.com>
Message-ID: <CAMG-S90Spc2mghjHEn_p3GtHdet6A84QaOkF-kKLyQNLhm+-PA@mail.gmail.com>

i just checked repo you mentioned. it clearly gives you code with code you
will get all dev file and lib :)

On Mon, Jul 20, 2015 at 5:33 PM, Akib Sayyed <akibsayyed at gmail.com> wrote:

> Please look for source
>
> https://code.google.com/p/libyuv/wiki/GettingStarted
>
> On Mon, Jul 20, 2015 at 4:57 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
>
>>  Hi,
>>
>> Of course!
>>
>> yum install libyuv-devel
>> Loaded plugins: fastestmirror, refresh-packagekit, security
>> Setting up Install Process
>> Loading mirror speeds from cached hostfile
>> * base: ftp.cvut.cz
>> * extras: ftp.cvut.cz
>> * rpmforge: mirror.us.leaseweb.net
>> * updates: ftp.agh.edu.pl
>> No package libyuv-devel available.
>> Error: Nothing to do
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>> <http://mob.tel>. 775-415-382 pond?l? 20. ?ervence 2015, 13:09:26, napsal
>> jste: *
>>
>>  Did you try `yum install libyuv-devel`?
>>
>> On 20 July 2015 at 06:10, Denis Jakovlev <yadenis at seznam.cz> wrote:
>> Hi All,
>>
>>
>> Clean installation CentOS 7. I am trying to install FreeSwitch according
>> to instructions.
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7
>>
>>
>> Makefile:887: *** You must install libyuv-dev to build mod_fsv.  Stop.
>> make[5]: Leaving directory
>> `/usr/src/freeswitch.git/src/mod/applications/mod_fsv
>>
>> Where can I get this library?
>> Here only libyuv. But i need libyuv-dev
>> https://freeswitch.org/stash/projects/SD/repos/libyuv/browse
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>>
>>
>> *. 775-415-382 *
>> _________________________________________________________________________
>> 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
>>
>
>
>
> --
> Akib Sayyed
> Matrix-Shell
> akibsayyed at gmail.com
> akibsayyed at matrixshell.com
> Mob:- +91-966-514-2243
>
>


-- 
Akib Sayyed
Matrix-Shell
akibsayyed at gmail.com
akibsayyed at matrixshell.com
Mob:- +91-966-514-2243
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From jason.holden at start.ca  Mon Jul 20 17:38:42 2015
From: jason.holden at start.ca (Jason Holden)
Date: Mon, 20 Jul 2015 09:38:42 -0400
Subject: [Freeswitch-users] shared line or other suggestion
In-Reply-To: <CAAtWKeGJJvNWkKsO9wPWVk6PwijXwzdt7U14cTCHF-gj3=inWg@mail.gmail.com>
References: <B0AD74FC10B94C6DBFCDB0F34F2A9112@bob>
	<CAAtWKeGJJvNWkKsO9wPWVk6PwijXwzdt7U14cTCHF-gj3=inWg@mail.gmail.com>
Message-ID: <96899FD041D24E51AB391A6FC13697C2@bob>

If a call comes in and phone a answers the call and places it on hold you
can see all the held calls with the proper CID.

On phone b that has a shared line of phone a you see the held call and are
able to answer is but it shows the caller Id of phone a and not the caller
on hold.

I want to see the CID of the caller on the list of held calls and not the
phones CID that has placed the call on hold.

 

 

 

  _____  

From: Yehavi Bourvine [mailto:yehavi.bourvine at gmail.com] 
Sent: Friday, July 17, 2015 11:41 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] shared line or other suggestion

 

Here is works ok in this situation. can you please give an example of what
exactly happens, what you see and what you expect to see?

 

      Regards, __Yehavi:

 

2015-07-17 16:41 GMT+03:00 Jason Holden <jason.holden at start.ca>:

I am currently trying to have the ability to place a call on hold and be
able to pick the call up from a displayed list on a group of  Polycom
phones.

I am currently using shared line but when the main phone places a call on
hold the shared lines to not show the held call with the remote user CID but
the primary phonesCID information.

Does anyone have a recommendation that'll work better.

I want to be able to see the remote CID of the caller who is on hold.

 

 

 

 


_________________________________________________________________________
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 Jul 20 17:47:11 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 20 Jul 2015 09:47:11 -0400
Subject: [Freeswitch-users] CentOS 7 install instructions does not work
In-Reply-To: <CAMG-S90Spc2mghjHEn_p3GtHdet6A84QaOkF-kKLyQNLhm+-PA@mail.gmail.com>
References: <DM2PR12MB0252EA8D47B0463D4BD007308A980@DM2PR12MB0252.namprd12.prod.outlook.com>
	<BEF2129F-E2D2-4F0A-AD27-E4871F8FF147@jerris.com>
	<BY2PR12MB02424D8823DFAE9DF3AA1D2A8A850@BY2PR12MB0242.namprd12.prod.outlook.com>
	<CAEwTsAhZ1QwFxd4kALwBTCywY5g8roUkHzgkxHTL=5cjAHN9SA@mail.gmail.com>
	<709511198.20150720111054@seznam.cz>
	<CAAyX+KH764HK8bG6y_Tdb_1w0U04CpZov6ZKwbFe5UtCmr9EKg@mail.gmail.com>
	<86757220.20150720132731@seznam.cz>
	<CAMG-S913P0S7NLrQXVDWK0-=z2hToWzA+t1jcs_n3xbR2TGcpQ@mail.gmail.com>
	<CAMG-S90Spc2mghjHEn_p3GtHdet6A84QaOkF-kKLyQNLhm+-PA@mail.gmail.com>
Message-ID: <CAEwTsAh7F6RwAjPwFkS-xiVRy1oCa_fCtEQ7ptqkx9gROrEhtQ@mail.gmail.com>

This is definitely NOT where you want to be getting this code.  If you are
in centos 7 as you said, you should be using our centos 7 repo to get these
packages.  If you are using centos 6 (as you said on jira), you should
instead be using the v1.4 branch in git or our 1.4 tarballs.

On Monday, July 20, 2015, Akib Sayyed <akibsayyed at gmail.com> wrote:

> i just checked repo you mentioned. it clearly gives you code with code you
> will get all dev file and lib :)
>
> On Mon, Jul 20, 2015 at 5:33 PM, Akib Sayyed <akibsayyed at gmail.com
> <javascript:_e(%7B%7D,'cvml','akibsayyed at gmail.com');>> wrote:
>
>> Please look for source
>>
>> https://code.google.com/p/libyuv/wiki/GettingStarted
>>
>> On Mon, Jul 20, 2015 at 4:57 PM, Denis Jakovlev <yadenis at seznam.cz
>> <javascript:_e(%7B%7D,'cvml','yadenis at seznam.cz');>> wrote:
>>
>>>  Hi,
>>>
>>> Of course!
>>>
>>> yum install libyuv-devel
>>> Loaded plugins: fastestmirror, refresh-packagekit, security
>>> Setting up Install Process
>>> Loading mirror speeds from cached hostfile
>>> * base: ftp.cvut.cz
>>> * extras: ftp.cvut.cz
>>> * rpmforge: mirror.us.leaseweb.net
>>> * updates: ftp.agh.edu.pl
>>> No package libyuv-devel available.
>>> Error: Nothing to do
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>>> <http://mob.tel>. 775-415-382 pond?l? 20. ?ervence 2015, 13:09:26, napsal
>>> jste: *
>>>
>>>  Did you try `yum install libyuv-devel`?
>>>
>>> On 20 July 2015 at 06:10, Denis Jakovlev <yadenis at seznam.cz
>>> <javascript:_e(%7B%7D,'cvml','yadenis at seznam.cz');>> wrote:
>>> Hi All,
>>>
>>>
>>> Clean installation CentOS 7. I am trying to install FreeSwitch according
>>> to instructions.
>>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7
>>>
>>>
>>> Makefile:887: *** You must install libyuv-dev to build mod_fsv.  Stop.
>>> make[5]: Leaving directory
>>> `/usr/src/freeswitch.git/src/mod/applications/mod_fsv
>>>
>>> Where can I get this library?
>>> Here only libyuv. But i need libyuv-dev
>>> https://freeswitch.org/stash/projects/SD/repos/libyuv/browse
>>>
>>>
>>>
>>>
>>> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>>>
>>>
>>> *. 775-415-382 *
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','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
>>> <javascript:_e(%7B%7D,'cvml','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
>>> <javascript:_e(%7B%7D,'cvml','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
>>> <javascript:_e(%7B%7D,'cvml','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
>>>
>>
>>
>>
>> --
>> Akib Sayyed
>> Matrix-Shell
>> akibsayyed at gmail.com
>> <javascript:_e(%7B%7D,'cvml','akibsayyed at gmail.com');>
>> akibsayyed at matrixshell.com
>> <javascript:_e(%7B%7D,'cvml','akibsayyed at matrixshell.com');>
>> Mob:- +91-966-514-2243
>>
>>
>
>
> --
> Akib Sayyed
> Matrix-Shell
> akibsayyed at gmail.com
> <javascript:_e(%7B%7D,'cvml','akibsayyed at gmail.com');>
> akibsayyed at matrixshell.com
> <javascript:_e(%7B%7D,'cvml','akibsayyed at matrixshell.com');>
> Mob:- +91-966-514-2243
>
>
-------------- next part --------------
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From yadenis at seznam.cz  Mon Jul 20 17:54:56 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Mon, 20 Jul 2015 15:54:56 +0200
Subject: [Freeswitch-users] CentOS 7 install instructions does not work
In-Reply-To: <CAMG-S913P0S7NLrQXVDWK0-=z2hToWzA+t1jcs_n3xbR2TGcpQ@mail.gmail.com>
References: <DM2PR12MB0252EA8D47B0463D4BD007308A980@DM2PR12MB0252.namprd12.prod.outlook.com>
	<BEF2129F-E2D2-4F0A-AD27-E4871F8FF147@jerris.com>
	<BY2PR12MB02424D8823DFAE9DF3AA1D2A8A850@BY2PR12MB0242.namprd12.prod.outlook.com>
	<CAEwTsAhZ1QwFxd4kALwBTCywY5g8roUkHzgkxHTL=5cjAHN9SA@mail.gmail.com>
	<709511198.20150720111054@seznam.cz>
	<CAAyX+KH764HK8bG6y_Tdb_1w0U04CpZov6ZKwbFe5UtCmr9EKg@mail.gmail.com>
	<86757220.20150720132731@seznam.cz>
	<CAMG-S913P0S7NLrQXVDWK0-=z2hToWzA+t1jcs_n3xbR2TGcpQ@mail.gmail.com>
Message-ID: <1327293394.20150720155456@seznam.cz>

Hi,

Then I do not understand what I'm doing wrong. Coming exactly a manual. The system perfectly clean. there's just no chance to make a mistake

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 20. ?ervence 2015, 14:03:27, napsal jste:


Please look for source

https://code.google.com/p/libyuv/wiki/GettingStarted

On Mon, Jul 20, 2015 at 4:57 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi,

Of course!

yum install libyuv-devel
Loaded plugins: fastestmirror, refresh-packagekit, security
Setting up Install Process
Loading mirror speeds from cached hostfile
* base: ftp.cvut.cz
* extras: ftp.cvut.cz
* rpmforge: mirror.us.leaseweb.net
* updates: ftp.agh.edu.pl
No package libyuv-devel available.
Error: Nothing to do


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 20. ?ervence 2015, 13:09:26, napsal jste:


Did you try `yum install libyuv-devel`?

On 20 July 2015 at 06:10, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi All,


Clean installation CentOS 7. I am trying to install FreeSwitch according to instructions.

https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7 


Makefile:887: *** You must install libyuv-dev to build mod_fsv.  Stop.
make[5]: Leaving directory `/usr/src/freeswitch.git/src/mod/applications/mod_fsv

Where can I get this library?
Here only libyuv. But i need libyuv-dev
https://freeswitch.org/stash/projects/SD/repos/libyuv/browse 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382


_________________________________________________________________________
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



-- 
Akib Sayyed
Matrix-Shell
akibsayyed at gmail.com
akibsayyed at matrixshell.com
Mob:- +91-966-514-2243
-------------- next part --------------
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From yadenis at seznam.cz  Mon Jul 20 18:00:25 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Mon, 20 Jul 2015 16:00:25 +0200
Subject: [Freeswitch-users] CentOS 7 install instructions does not work
In-Reply-To: <CAEwTsAh7F6RwAjPwFkS-xiVRy1oCa_fCtEQ7ptqkx9gROrEhtQ@mail.gmail.com>
References: <DM2PR12MB0252EA8D47B0463D4BD007308A980@DM2PR12MB0252.namprd12.prod.outlook.com>
	<BEF2129F-E2D2-4F0A-AD27-E4871F8FF147@jerris.com>
	<BY2PR12MB02424D8823DFAE9DF3AA1D2A8A850@BY2PR12MB0242.namprd12.prod.outlook.com>
	<CAEwTsAhZ1QwFxd4kALwBTCywY5g8roUkHzgkxHTL=5cjAHN9SA@mail.gmail.com>
	<709511198.20150720111054@seznam.cz>
	<CAAyX+KH764HK8bG6y_Tdb_1w0U04CpZov6ZKwbFe5UtCmr9EKg@mail.gmail.com>
	<86757220.20150720132731@seznam.cz>
	<CAMG-S913P0S7NLrQXVDWK0-=z2hToWzA+t1jcs_n3xbR2TGcpQ@mail.gmail.com>
	<CAMG-S90Spc2mghjHEn_p3GtHdet6A84QaOkF-kKLyQNLhm+-PA@mail.gmail.com>
	<CAEwTsAh7F6RwAjPwFkS-xiVRy1oCa_fCtEQ7ptqkx9gROrEhtQ@mail.gmail.com>
Message-ID: <1496028801.20150720160025@seznam.cz>

Dobr? den,

I tried 6 and 7. Going on manual for version 6 and 7. 

This is from the manual for version 6
https://freeswitch.org/confluence/display/FREESWITCH/CentOS+6  
Install FreeSWITCH
cd /usr/src
# To build from Master, the latest source code: 
git clone https://freeswitch.org/stash/scm/fs/freeswitch.git
##### OR #####
# To build from the current release source code:
git clone -b v1.4 https://freeswitch.org/stash/scm/fs/freeswitch.git

Where does it say that I have to use only 1.4 branch? 




-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 20. ?ervence 2015, 15:47:11, napsal jste:


This is definitely NOT where you want to be getting this code.  If you are in centos 7 as you said, you should be using our centos 7 repo to get these packages.  If you are using centos 6 (as you said on jira), you should instead be using the v1.4 branch in git or our 1.4 tarballs.  

On Monday, July 20, 2015, Akib Sayyed <akibsayyed at gmail.com> wrote:
i just checked repo you mentioned. it clearly gives you code with code you will get all dev file and lib :)

On Mon, Jul 20, 2015 at 5:33 PM, Akib Sayyed <akibsayyed at gmail.com> wrote:
Please look for source

https://code.google.com/p/libyuv/wiki/GettingStarted

On Mon, Jul 20, 2015 at 4:57 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi,

Of course!

yum install libyuv-devel
Loaded plugins: fastestmirror, refresh-packagekit, security
Setting up Install Process
Loading mirror speeds from cached hostfile
* base: ftp.cvut.cz
* extras: ftp.cvut.cz
* rpmforge: mirror.us.leaseweb.net
* updates: ftp.agh.edu.pl
No package libyuv-devel available.
Error: Nothing to do


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 20. ?ervence 2015, 13:09:26, napsal jste:


Did you try `yum install libyuv-devel`?

On 20 July 2015 at 06:10, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi All,


Clean installation CentOS 7. I am trying to install FreeSwitch according to instructions.

https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7 


Makefile:887: *** You must install libyuv-dev to build mod_fsv.  Stop.
make[5]: Leaving directory `/usr/src/freeswitch.git/src/mod/applications/mod_fsv

Where can I get this library?
Here only libyuv. But i need libyuv-dev
https://freeswitch.org/stash/projects/SD/repos/libyuv/browse 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382


_________________________________________________________________________
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



-- 
Akib Sayyed
Matrix-Shell
akibsayyed at gmail.com
akibsayyed at matrixshell.com
Mob:- +91-966-514-2243




-- 
Akib Sayyed
Matrix-Shell
akibsayyed at gmail.com
akibsayyed at matrixshell.com
Mob:- +91-966-514-2243
-------------- next part --------------
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From luis.daniel.lucio at gmail.com  Mon Jul 20 19:00:48 2015
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Mon, 20 Jul 2015 11:00:48 -0400
Subject: [Freeswitch-users] CentOS 7 install instructions does not work
In-Reply-To: <1496028801.20150720160025@seznam.cz>
References: <DM2PR12MB0252EA8D47B0463D4BD007308A980@DM2PR12MB0252.namprd12.prod.outlook.com>
	<BEF2129F-E2D2-4F0A-AD27-E4871F8FF147@jerris.com>
	<BY2PR12MB02424D8823DFAE9DF3AA1D2A8A850@BY2PR12MB0242.namprd12.prod.outlook.com>
	<CAEwTsAhZ1QwFxd4kALwBTCywY5g8roUkHzgkxHTL=5cjAHN9SA@mail.gmail.com>
	<709511198.20150720111054@seznam.cz>
	<CAAyX+KH764HK8bG6y_Tdb_1w0U04CpZov6ZKwbFe5UtCmr9EKg@mail.gmail.com>
	<86757220.20150720132731@seznam.cz>
	<CAMG-S913P0S7NLrQXVDWK0-=z2hToWzA+t1jcs_n3xbR2TGcpQ@mail.gmail.com>
	<CAMG-S90Spc2mghjHEn_p3GtHdet6A84QaOkF-kKLyQNLhm+-PA@mail.gmail.com>
	<CAEwTsAh7F6RwAjPwFkS-xiVRy1oCa_fCtEQ7ptqkx9gROrEhtQ@mail.gmail.com>
	<1496028801.20150720160025@seznam.cz>
Message-ID: <CAFLo2Qx1y2x0xXjA0SXYRPMOJVtMUfscFwL75=+7o3K5XJhgdQ@mail.gmail.com>

I will start working on fs 1.6 rpms as soon as the first tarball is
published.

If it is already, please post where I can download it.
On Jul 20, 2015 10:01 AM, "Denis Jakovlev" <yadenis at seznam.cz> wrote:

>  Dobr? den,
>
> I tried 6 and 7. Going on manual for version 6 and 7.
>
> This is from the manual for version 6
> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+6
>
> *Install FreeSWITCH * cd /usr/src
> # To build from Master, the latest source code:
> git clone https://freeswitch.org/stash/scm/fs/freeswitch.git
>
> *##### OR ##### *# To build from the current release source code:
> git clone -b v1.4 https://freeswitch.org/stash/scm/fs/freeswitch.git
>
> Where does it say that I have to use only 1.4 branch?
>
>
>
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 pond?l? 20. ?ervence 2015, 15:47:11, napsal
> jste: *
>  This is definitely NOT where you want to be getting this code.  If you
> are in centos 7 as you said, you should be using our centos 7 repo to get
> these packages.  If you are using centos 6 (as you said on jira), you
> should instead be using the v1.4 branch in git or our 1.4 tarballs.
>
> On Monday, July 20, 2015, Akib Sayyed <akibsayyed at gmail.com> wrote:
> i just checked repo you mentioned. it clearly gives you code with code you
> will get all dev file and lib :)
>
> On Mon, Jul 20, 2015 at 5:33 PM, Akib Sayyed <akibsayyed at gmail.com> wrote:
> Please look for source
>
> https://code.google.com/p/libyuv/wiki/GettingStarted
>
> On Mon, Jul 20, 2015 at 4:57 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Hi,
>
> Of course!
>
> yum install libyuv-devel
> Loaded plugins: fastestmirror, refresh-packagekit, security
> Setting up Install Process
> Loading mirror speeds from cached hostfile
> * base: ftp.cvut.cz
> * extras: ftp.cvut.cz
> * rpmforge: mirror.us.leaseweb.net
> * updates: ftp.agh.edu.pl
> No package libyuv-devel available.
> Error: Nothing to do
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
>
>
> *. 775-415-382 pond?l? 20. ?ervence 2015, 13:09:26, napsal jste: *
>  Did you try `yum install libyuv-devel`?
>
> On 20 July 2015 at 06:10, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Hi All,
>
>
> Clean installation CentOS 7. I am trying to install FreeSwitch according
> to instructions.
>
> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7
>
>
> Makefile:887: *** You must install libyuv-dev to build mod_fsv.  Stop.
> make[5]: Leaving directory
> `/usr/src/freeswitch.git/src/mod/applications/mod_fsv
>
> Where can I get this library?
> Here only libyuv. But i need libyuv-dev
> https://freeswitch.org/stash/projects/SD/repos/libyuv/browse
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
>
> *. 775-415-382 *
> _________________________________________________________________________
> 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
>
>
>
> --
> Akib Sayyed
> Matrix-Shell
> akibsayyed at gmail.com
> akibsayyed at matrixshell.com
> Mob:- +91-966-514-2243
>
>
>
>
> --
> Akib Sayyed
> Matrix-Shell
> akibsayyed at gmail.com
> akibsayyed at matrixshell.com
> Mob:- +91-966-514-2243
>
> _________________________________________________________________________
> 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 jaflong at yandex.com  Mon Jul 20 19:14:54 2015
From: jaflong at yandex.com (jaflong jaflong)
Date: Mon, 20 Jul 2015 18:14:54 +0300
Subject: [Freeswitch-users] I am using outbound to ivrd. My Call gets
	answered however why do I not get events with
	$conn->events("plain", "ALL"); 
Message-ID: <57281437405294@web15m.yandex.ru>




<include>
        <extension name="demo">
                <condition field="destination_number" expression="^(\d+)$">
					<action application="set" data="ivr_path=/usr/local/freeswitch/scripts/ivr_test.php"/>
					<action application="socket" data="127.0.0.1:8004" async full/>
                </condition>
        </extension>
</include>



ivr_test.php
--------------

#!/usr/bin/php
<?php
require_once('ESL.php');

$conn = new ESLconnection(0);

$music = "/usr/local/freeswitch/sounds/demo.wav";

$conn->execute("answer");
$conn->execute("playback", $music);


while ($conn->connected() == 1) {

	$conn->events("plain", "ALL");

	$e = $conn->recvEvent();
	if ($e) {
		$name = $e->getHeader('event-name');
		//echo "Got event $name\n";
		$out = fopen('php://stderr', 'w+');
		fwrite($out, "Got event $name\n");

	} else {

		//echo "Event empty\n";
		$out = fopen('php://stderr', 'w+');
		fwrite($out, "Event empty\n");
	}

	
}

I am using outbound to ivrd. My Call gets answered however why do I not get events with $conn->events("plain", "ALL");


From chad at apartmentlines.com  Mon Jul 20 19:31:32 2015
From: chad at apartmentlines.com (Chad Phillips)
Date: Mon, 20 Jul 2015 08:31:32 -0700
Subject: [Freeswitch-users] Serving Verto HTML from a different server
In-Reply-To: <CAKbxfG_JFqzMBb4JUujWh=r2bXaCNz14qpu04FB7m09=XTB3sw@mail.gmail.com>
References: <CAJ0sghQ1zkuHEQxUuB0sa2is2hwUk9XBJavQ9GoxJ9sWnDQuQQ@mail.gmail.com>
	<CAKbxfG_JFqzMBb4JUujWh=r2bXaCNz14qpu04FB7m09=XTB3sw@mail.gmail.com>
Message-ID: <CAJ0sghR5XwDNr8Yd+hZFUskYj9Sf+Tnxbn96eEoG5k7b9Lqs8w@mail.gmail.com>

Thanks for the help, guys. I?m making a note to integrate this information
into the wiki page for the 1.6 video installation.

Chad

On Mon, Jul 20, 2015 at 5:55 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> The only requirement is that the cert on FS is trusted.  The html can come
> from anywhere.
>
> If its not a trusted cert the wss connection will silently fail.  This is
> because there is no dialog asking you to trust the cert on wss connections
> so they just PUNT and don't even log an error.  I opened a bug a year ago
> mentioning ot and they said thats working as designed.
>
> So the only way to make untrusted certs work is to make an https request
> to the wss port one time and agree to manually trust it.  Therefore,
> serving that same cert for the html serves the same perpose of manually
> trusting the cert before making the wss connection.  With trusted certs,
> html and wss can be entirelly different.
>
>
> On Saturday, July 18, 2015, Chad Phillips <chad at apartmentlines.com> wrote:
>
>> I?ve gotten fairly comfortable with the setup offered in the Confluence
>> wiki for the new Verto video features.
>>
>> I?m curious to know if I can expect any gotchas if I serve the Verto web
>> stuff from a different server than the FreeSWITCH instance is running on.
>> From what I can tell, as long as I make sure to use the same server certs
>> on both servers, it *should* work.
>>
>> Is there anything I?m not taking into consideration? Any better way to
>> approach the problem besides certificate sharing across servers?
>>
>> Thanks,
>>
>> Chad
>>
>
>
> --
> 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
>
>
> _________________________________________________________________________
> 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 Jul 20 19:47:04 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 20 Jul 2015 11:47:04 -0400
Subject: [Freeswitch-users] CentOS 7 install instructions does not work
In-Reply-To: <CAFLo2Qx1y2x0xXjA0SXYRPMOJVtMUfscFwL75=+7o3K5XJhgdQ@mail.gmail.com>
References: <DM2PR12MB0252EA8D47B0463D4BD007308A980@DM2PR12MB0252.namprd12.prod.outlook.com>
	<BEF2129F-E2D2-4F0A-AD27-E4871F8FF147@jerris.com>
	<BY2PR12MB02424D8823DFAE9DF3AA1D2A8A850@BY2PR12MB0242.namprd12.prod.outlook.com>
	<CAEwTsAhZ1QwFxd4kALwBTCywY5g8roUkHzgkxHTL=5cjAHN9SA@mail.gmail.com>
	<709511198.20150720111054@seznam.cz>
	<CAAyX+KH764HK8bG6y_Tdb_1w0U04CpZov6ZKwbFe5UtCmr9EKg@mail.gmail.com>
	<86757220.20150720132731@seznam.cz>
	<CAMG-S913P0S7NLrQXVDWK0-=z2hToWzA+t1jcs_n3xbR2TGcpQ@mail.gmail.com>
	<CAMG-S90Spc2mghjHEn_p3GtHdet6A84QaOkF-kKLyQNLhm+-PA@mail.gmail.com>
	<CAEwTsAh7F6RwAjPwFkS-xiVRy1oCa_fCtEQ7ptqkx9gROrEhtQ@mail.gmail.com>
	<1496028801.20150720160025@seznam.cz>
	<CAFLo2Qx1y2x0xXjA0SXYRPMOJVtMUfscFwL75=+7o3K5XJhgdQ@mail.gmail.com>
Message-ID: <CAEwTsAhRZBoEigcmLbNSRmcxBtyRGq9JjNJCFbhU+A0n_a5Reg@mail.gmail.com>

The rpms for centos 7 already exist.  Centos 6 is probably not worth
attempting to make work due to the age of dependency libraries.

On Monday, July 20, 2015, Luis Daniel Lucio Quiroz <
luis.daniel.lucio at gmail.com> wrote:

> I will start working on fs 1.6 rpms as soon as the first tarball is
> published.
>
> If it is already, please post where I can download it.
> On Jul 20, 2015 10:01 AM, "Denis Jakovlev" <yadenis at seznam.cz
> <javascript:_e(%7B%7D,'cvml','yadenis at seznam.cz');>> wrote:
>
>>  Dobr? den,
>>
>> I tried 6 and 7. Going on manual for version 6 and 7.
>>
>> This is from the manual for version 6
>> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+6
>>
>> *Install FreeSWITCH * cd /usr/src
>> # To build from Master, the latest source code:
>> git clone https://freeswitch.org/stash/scm/fs/freeswitch.git
>>
>> *##### OR ##### *# To build from the current release source code:
>> git clone -b v1.4 https://freeswitch.org/stash/scm/fs/freeswitch.git
>>
>> Where does it say that I have to use only 1.4 branch?
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>> <http://mob.tel>. 775-415-382 pond?l? 20. ?ervence 2015, 15:47:11, napsal
>> jste: *
>>  This is definitely NOT where you want to be getting this code.  If you
>> are in centos 7 as you said, you should be using our centos 7 repo to get
>> these packages.  If you are using centos 6 (as you said on jira), you
>> should instead be using the v1.4 branch in git or our 1.4 tarballs.
>>
>> On Monday, July 20, 2015, Akib Sayyed <akibsayyed at gmail.com
>> <javascript:_e(%7B%7D,'cvml','akibsayyed at gmail.com');>> wrote:
>> i just checked repo you mentioned. it clearly gives you code with code
>> you will get all dev file and lib :)
>>
>> On Mon, Jul 20, 2015 at 5:33 PM, Akib Sayyed <akibsayyed at gmail.com>
>> wrote:
>> Please look for source
>>
>> https://code.google.com/p/libyuv/wiki/GettingStarted
>>
>> On Mon, Jul 20, 2015 at 4:57 PM, Denis Jakovlev <yadenis at seznam.cz>
>> wrote:
>> Hi,
>>
>> Of course!
>>
>> yum install libyuv-devel
>> Loaded plugins: fastestmirror, refresh-packagekit, security
>> Setting up Install Process
>> Loading mirror speeds from cached hostfile
>> * base: ftp.cvut.cz
>> * extras: ftp.cvut.cz
>> * rpmforge: mirror.us.leaseweb.net
>> * updates: ftp.agh.edu.pl
>> No package libyuv-devel available.
>> Error: Nothing to do
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>>
>>
>>
>> *. 775-415-382 pond?l? 20. ?ervence 2015, 13:09:26, napsal jste: *
>>  Did you try `yum install libyuv-devel`?
>>
>> On 20 July 2015 at 06:10, Denis Jakovlev <yadenis at seznam.cz> wrote:
>> Hi All,
>>
>>
>> Clean installation CentOS 7. I am trying to install FreeSwitch according
>> to instructions.
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7
>>
>>
>> Makefile:887: *** You must install libyuv-dev to build mod_fsv.  Stop.
>> make[5]: Leaving directory
>> `/usr/src/freeswitch.git/src/mod/applications/mod_fsv
>>
>> Where can I get this library?
>> Here only libyuv. But i need libyuv-dev
>> https://freeswitch.org/stash/projects/SD/repos/libyuv/browse
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>>
>>
>> *. 775-415-382 *
>> _________________________________________________________________________
>> 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
>>
>>
>>
>> --
>> Akib Sayyed
>> Matrix-Shell
>> akibsayyed at gmail.com
>> akibsayyed at matrixshell.com
>> Mob:- +91-966-514-2243
>>
>>
>>
>>
>> --
>> Akib Sayyed
>> Matrix-Shell
>> akibsayyed at gmail.com
>> akibsayyed at matrixshell.com
>> Mob:- +91-966-514-2243
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','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
>> <javascript:_e(%7B%7D,'cvml','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 manpower13.cse at gmail.com  Mon Jul 20 19:55:41 2015
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Mon, 20 Jul 2015 21:25:41 +0530
Subject: [Freeswitch-users] Freeswitch+iptel
Message-ID: <CAMRuXGh2VAxb=ZdwVsbp_D0cNFOvu_d7n7iRJkeywW0_s9MfwQ@mail.gmail.com>

Hi,

   I need to use FS only media server and WS-SIP signalling server


  Media-server

        i disabled auth-call in internal.xml so i can make call from iptel
account

  WS-SIP

        But when i try to use WS-SIP i can't  able to  register using
 iptel account getting 408 error(chrome browser)


  If i am doing wrong here pls help me
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From mike at jerris.com  Mon Jul 20 20:20:53 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 20 Jul 2015 12:20:53 -0400
Subject: [Freeswitch-users] Freeswitch+iptel
In-Reply-To: <CAMRuXGh2VAxb=ZdwVsbp_D0cNFOvu_d7n7iRJkeywW0_s9MfwQ@mail.gmail.com>
References: <CAMRuXGh2VAxb=ZdwVsbp_D0cNFOvu_d7n7iRJkeywW0_s9MfwQ@mail.gmail.com>
Message-ID: <CAEwTsAjegHjXMbm20bKXugxQfH0gm0MpZMGkN8_-tBvtFOSGWQ@mail.gmail.com>

Please see the previous thread you started on this same topic where your
questions have already been answered

On Monday, July 20, 2015, Murugan Pandian <manpower13.cse at gmail.com> wrote:

> Hi,
>
>    I need to use FS only media server and WS-SIP signalling server
>
>
>   Media-server
>
>         i disabled auth-call in internal.xml so i can make call from iptel
> account
>
>   WS-SIP
>
>         But when i try to use WS-SIP i can't  able to  register using
>  iptel account getting 408 error(chrome browser)
>
>
>   If i am doing wrong here pls help me
>
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From Hector.Geraldino at ipsoft.com  Mon Jul 20 20:28:23 2015
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Mon, 20 Jul 2015 16:28:23 +0000
Subject: [Freeswitch-users] mod_managed vs eventsocket
In-Reply-To: <CAFgB_11tki=JHoAqm+UAj0NhjkYBvSzDM2Kab47biWfjY9Qypg@mail.gmail.com>
References: <CAFgB_10ZtaKSdQ-rqew3LZO79h9zWFFYi9qo7E_gpDASSrU+QQ@mail.gmail.com>
	<CAFgB_13syeAtznT78KE46xo7XSkt2LiSpRNTNHMdqTWmemq9ZQ@mail.gmail.com>
	<CAFgB_11tki=JHoAqm+UAj0NhjkYBvSzDM2Kab47biWfjY9Qypg@mail.gmail.com>
Message-ID: <D1D298D7.F80B%hgeraldino@ipsoft.com>

I don't think there's an API that you can use to tap on the RTP stream. You can always record your call in stereo and post-process the file, if that's valid for what you're trying to do.

If not, check the MRCP protocol. I guess you can write an MRCP server (or use an existing one), and stream the audio from FS using mod_unimrcp. That's how you integrate 3rd party ASR systems (e.g. Nuance's Recognizer) with FreeSWITCH.

From: <freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> on behalf of Prashant Choudhary <get.prashant.007 at gmail.com<mailto:get.prashant.007 at gmail.com>>
Reply-To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Date: Sunday, July 19, 2015 at 4:24 PM
To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] mod_managed vs eventsocket

Hello,
Now I have even managed to do outbound calls. I have a real problem now.
How can I get the audio stream from the person on the other end. I want the audio stream to be sent to my API in real time. I want to detect silence as well, but we can do that by experiment. Only thing is that I am unable to get the audio stream. And this stream should only have the voice of person being called, not mine(i.e. the incoming audio). Is it possible??
Please help

On Sat, Jul 18, 2015 at 2:57 PM, Prashant Choudhary <get.prashant.007 at gmail.com<mailto:get.prashant.007 at gmail.com>> wrote:
Sorry for my last mail, which was a little more generic. Thing is, I am able to receive and manage Inbound calls. But I am unable to figure out, how to make outbound calls through mod_managed. I was able to do the same through event socket API. One benefit of event socket was that I could run it as separate console application , but I am not sure whether it is possible through mod managed as well.
SO again my question remains the same. But it is more towards outbound calling.

On Sat, Jul 18, 2015 at 1:11 AM, Prashant Choudhary <get.prashant.007 at gmail.com<mailto:get.prashant.007 at gmail.com>> wrote:
Hello All,

I have worked with LUA and freeswitch. Now I want to migrate to .Net API as I want to experiment with machine learning using sample calls.  I am trying to control around some 2-300 calls directly through my API. This may require the call audio stream to be forwarded in real time to .NET
 I a few questions that might be very silly.

1. Which one is better to achieve this, mod_managed or ESL.
2. I was able to find some examples related to eventsocket, but same for mod_Managed was rare. Can you suggest me any doc(of-course except the wiki link) or any email chain which explains this or anything else. This will be really a big help for me. As currently I am stuck. And I still find the LUA way easier.

Thanks


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From mike at jerris.com  Mon Jul 20 21:08:18 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 20 Jul 2015 13:08:18 -0400
Subject: [Freeswitch-users] mod_managed vs eventsocket
In-Reply-To: <D1D298D7.F80B%hgeraldino@ipsoft.com>
References: <CAFgB_10ZtaKSdQ-rqew3LZO79h9zWFFYi9qo7E_gpDASSrU+QQ@mail.gmail.com>
	<CAFgB_13syeAtznT78KE46xo7XSkt2LiSpRNTNHMdqTWmemq9ZQ@mail.gmail.com>
	<CAFgB_11tki=JHoAqm+UAj0NhjkYBvSzDM2Kab47biWfjY9Qypg@mail.gmail.com>
	<D1D298D7.F80B%hgeraldino@ipsoft.com>
Message-ID: <CAEwTsAj+67EX2RLUJCKdOsHbi7_CFbZ5h3pJqgT-E1=jyYnPQQ@mail.gmail.com>

in managed modules if you build an application interface you can access the
media the same way you do in native c modules.  The c api is marshaled into
managed code

On Monday, July 20, 2015, Hector Geraldino <Hector.Geraldino at ipsoft.com>
wrote:

>  I don?t think there?s an API that you can use to tap on the RTP stream.
> You can always record your call in stereo and post-process the file, if
> that?s valid for what you?re trying to do.
>
>  If not, check the MRCP protocol. I guess you can write an MRCP server
> (or use an existing one), and stream the audio from FS using mod_unimrcp.
> That?s how you integrate 3rd party ASR systems (e.g. Nuance?s Recognizer)
> with FreeSWITCH.
>
>   From: <freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>>
> on behalf of Prashant Choudhary <get.prashant.007 at gmail.com
> <javascript:_e(%7B%7D,'cvml','get.prashant.007 at gmail.com');>>
> Reply-To: "freeswitch-users at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users at lists.freeswitch.org');>" <
> freeswitch-users at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users at lists.freeswitch.org');>>
> Date: Sunday, July 19, 2015 at 4:24 PM
> To: "freeswitch-users at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users at lists.freeswitch.org');>" <
> freeswitch-users at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users at lists.freeswitch.org');>>
> Subject: Re: [Freeswitch-users] mod_managed vs eventsocket
>
>   Hello,
> Now I have even managed to do outbound calls. I have a real problem now.
> How can I get the audio stream from the person on the other end. I want
> the audio stream to be sent to my API in real time. I want to detect
> silence as well, but we can do that by experiment. Only thing is that I am
> unable to get the audio stream. And this stream should only have the voice
> of person being called, not mine(i.e. the incoming audio). Is it possible??
> Please help
>
> On Sat, Jul 18, 2015 at 2:57 PM, Prashant Choudhary <
> get.prashant.007 at gmail.com
> <javascript:_e(%7B%7D,'cvml','get.prashant.007 at gmail.com');>> wrote:
>
>> Sorry for my last mail, which was a little more generic. Thing is, I am
>> able to receive and manage Inbound calls. But I am unable to figure out,
>> how to make outbound calls through mod_managed. I was able to do the same
>> through event socket API. One benefit of event socket was that I could run
>> it as separate console application , but I am not sure whether it is
>> possible through mod managed as well.
>> SO again my question remains the same. But it is more towards outbound
>> calling.
>>
>> On Sat, Jul 18, 2015 at 1:11 AM, Prashant Choudhary <
>> get.prashant.007 at gmail.com
>> <javascript:_e(%7B%7D,'cvml','get.prashant.007 at gmail.com');>> wrote:
>>
>>> Hello All,
>>>
>>> I have worked with LUA and freeswitch. Now I want to migrate to .Net API
>>> as I want to experiment with machine learning using sample calls.  I am
>>> trying to control around some 2-300 calls directly through my API. This may
>>> require the call audio stream to be forwarded in real time to .NET
>>>  I a few questions that might be very silly.
>>>
>>> 1. Which one is better to achieve this, mod_managed or ESL.
>>> 2. I was able to find some examples related to eventsocket, but same for
>>> mod_Managed was rare. Can you suggest me any doc(of-course except the wiki
>>> link) or any email chain which explains this or anything else. This will be
>>> really a big help for me. As currently I am stuck. And I still find the LUA
>>> way easier.
>>>
>>> Thanks
>>>
>>
>>
>
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From get.prashant.007 at gmail.com  Mon Jul 20 21:58:00 2015
From: get.prashant.007 at gmail.com (Prashant)
Date: Mon, 20 Jul 2015 23:28:00 +0530
Subject: [Freeswitch-users] mod_managed vs eventsocket
In-Reply-To: <CAEwTsAj+67EX2RLUJCKdOsHbi7_CFbZ5h3pJqgT-E1=jyYnPQQ@mail.gmail.com>
References: <CAFgB_10ZtaKSdQ-rqew3LZO79h9zWFFYi9qo7E_gpDASSrU+QQ@mail.gmail.com>
	<CAFgB_13syeAtznT78KE46xo7XSkt2LiSpRNTNHMdqTWmemq9ZQ@mail.gmail.com>
	<CAFgB_11tki=JHoAqm+UAj0NhjkYBvSzDM2Kab47biWfjY9Qypg@mail.gmail.com>
	<D1D298D7.F80B%hgeraldino@ipsoft.com>
	<CAEwTsAj+67EX2RLUJCKdOsHbi7_CFbZ5h3pJqgT-E1=jyYnPQQ@mail.gmail.com>
Message-ID: <55ad36a9.0566420a.41cc.ffffef06@mx.google.com>

Awesome, thanks for the reply sir.
I was already trying to record it and play to other service. But I guess mod_mrcp seems better.  I will work on it and let you know if I face any problem. Thanks

-----Original Message-----
From: "Michael Jerris" <mike at jerris.com>
Sent: ?20-?07-?2015 10:39 PM
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] mod_managed vs eventsocket

in managed modules if you build an application interface you can access the media the same way you do in native c modules.  The c api is marshaled into managed code

On Monday, July 20, 2015, Hector Geraldino <Hector.Geraldino at ipsoft.com> wrote:

I don?t think there?s an API that you can use to tap on the RTP stream. You can always record your call in stereo and post-process the file, if that?s valid for what you?re trying to do.


If not, check the MRCP protocol. I guess you can write an MRCP server (or use an existing one), and stream the audio from FS using mod_unimrcp. That?s how you integrate 3rd party ASR systems (e.g. Nuance?s Recognizer) with FreeSWITCH.


From: <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Prashant Choudhary <get.prashant.007 at gmail.com>
Reply-To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
Date: Sunday, July 19, 2015 at 4:24 PM
To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] mod_managed vs eventsocket



Hello,
Now I have even managed to do outbound calls. I have a real problem now.
How can I get the audio stream from the person on the other end. I want the audio stream to be sent to my API in real time. I want to detect silence as well, but we can do that by experiment. Only thing is that I am unable to get the audio stream. And this stream should only have the voice of person being called, not mine(i.e. the incoming audio). Is it possible?? 
Please help


On Sat, Jul 18, 2015 at 2:57 PM, Prashant Choudhary <get.prashant.007 at gmail.com> wrote:

Sorry for my last mail, which was a little more generic. Thing is, I am able to receive and manage Inbound calls. But I am unable to figure out, how to make outbound calls through mod_managed. I was able to do the same through event socket API. One benefit of event socket was that I could run it as separate console application , but I am not sure whether it is possible through mod managed as well. 
SO again my question remains the same. But it is more towards outbound calling.


On Sat, Jul 18, 2015 at 1:11 AM, Prashant Choudhary <get.prashant.007 at gmail.com> wrote:

Hello All,

I have worked with LUA and freeswitch. Now I want to migrate to .Net API as I want to experiment with machine learning using sample calls.  I am trying to control around some 2-300 calls directly through my API. This may require the call audio stream to be forwarded in real time to .NET 
 I a few questions that might be very silly.

1. Which one is better to achieve this, mod_managed or ESL.
2. I was able to find some examples related to eventsocket, but same for mod_Managed was rare. Can you suggest me any doc(of-course except the wiki link) or any email chain which explains this or anything else. This will be really a big help for me. As currently I am stuck. And I still find the LUA way easier.

Thanks
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From manpower13.cse at gmail.com  Mon Jul 20 22:29:53 2015
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Mon, 20 Jul 2015 23:59:53 +0530
Subject: [Freeswitch-users] Mod_verto vs WebRTC(ws/wss)
In-Reply-To: <CAAyX+KGgTLYJ5vatqF-L+u2YU3W_FxuVf9QQAgHvGDx5wzPO6w@mail.gmail.com>
References: <CAMRuXGjVxqiuT75WhKE9nWtSXewRYtRxh-LCu+g9Tac7EvQXWg@mail.gmail.com>
	<CAAyX+KFJCswqb1k7WN86udhAxuBza9RTLjYat695aG6cqAR-dQ@mail.gmail.com>
	<CAMRuXGj5Z8LtuRd8_WLR5cy7oQO9a4p32tgZf=L6x0s=GTtoSA@mail.gmail.com>
	<CAMRuXGheGqw9Q5fAzndjyjxS8P3abd2o6bK0Vv_C-i41hwqZOA@mail.gmail.com>
	<CACXsapOFUS20qh2NPGgXkT-ck+CFWsUAiL+=L5PuqnkRNMUR_g@mail.gmail.com>
	<CALXCt0p5cMzrY-PhgfkJceDFkV365XkkFQw3KzfMAacNnkcD8g@mail.gmail.com>
	<CAAyX+KGgTLYJ5vatqF-L+u2YU3W_FxuVf9QQAgHvGDx5wzPO6w@mail.gmail.com>
Message-ID: <CAMRuXGgePd1Z47dtaYY0fJpSo2HaS-OKo8c3ziGRHcRNJBTcLg@mail.gmail.com>

Can i know how i use my sip2sip account gateway in mod_verto

On Sat, Jul 18, 2015 at 7:48 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:

> To clarify, if you want your WebRTC UA specifically to register to
> sip2sip, then you must use SIP all the way through. A Verto endpoint cannot
> register with a SIP registrar.
>
> On 17 July 2015 at 18:13, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
>
>> I have understood from your other mail to list that you want to use a
>> sip2sip account from browser.
>>
>> You can then use sip.js to connect to a kamailio that goes to sip2sip, or
>> you can use verto to connect to freeswitch that will use your sip2sip
>> account as gateway.
>>
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> On Jul 17, 2015 11:10 PM, "Victor Medina" <victor.medina at cibersys.com>
>> wrote:
>>
>>> FreeSWITCH will do both without much problems.
>>>
>>> Decide what to use. Your choice.
>>>  El 17/07/2015 16:02, "Murugan Pandian" <manpower13.cse at gmail.com>
>>> escribi?:
>>>
>>>> Sorry for Diagram above one not clear
>>>>
>>>>
>>>>          {Browser}
>>>>                  |
>>>>                  |
>>>>    (SIP-WS or verto)
>>>>                  |
>>>>         [freeswitch]
>>>>                  |
>>>>                  |
>>>>          [Register]
>>>>         (iptel or sip2sip)
>>>>
>>>> On Sat, Jul 18, 2015 at 1:54 AM, Murugan Pandian <
>>>> manpower13.cse at gmail.com> wrote:
>>>>
>>>>> Thanks, If there any way i can achieve  this
>>>>>
>>>>>
>>>>>
>>>>>         {Browser}
>>>>>                  |
>>>>>                  |
>>>>>                  |   (SIP-WS or verto)
>>>>>               (iptel or sip2sip)
>>>>>                  |-  -   -  -  - -  -  - - - - -[freeswitch]- - - - -
>>>>> - - - -  -  -   [Register]
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Sat, Jul 18, 2015 at 1:42 AM, Ben Langfeld <ben at langfeld.co.uk>
>>>>> wrote:
>>>>>
>>>>>> Both mod_verto and SIP-WS are signalling mechanisms for WebRTC. The
>>>>>> former is a FreeSWITCH-proprietary protocol, the latter is a
>>>>>> standards-based approach. The former is probably easier to understand if
>>>>>> you've never done telephony work, the latter requires you to know more
>>>>>> stuff.
>>>>>>
>>>>>> On 17 July 2015 at 16:52, Murugan Pandian <manpower13.cse at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>>    Any one can help me what is difference between mod_verto and
>>>>>>>  WebRTC(ws/wss) mod.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> 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
>>
>
>
> _________________________________________________________________________
> 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 manpower13.cse at gmail.com  Mon Jul 20 22:30:37 2015
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Tue, 21 Jul 2015 00:00:37 +0530
Subject: [Freeswitch-users] Freeswitch+iptel
In-Reply-To: <CAEwTsAjegHjXMbm20bKXugxQfH0gm0MpZMGkN8_-tBvtFOSGWQ@mail.gmail.com>
References: <CAMRuXGh2VAxb=ZdwVsbp_D0cNFOvu_d7n7iRJkeywW0_s9MfwQ@mail.gmail.com>
	<CAEwTsAjegHjXMbm20bKXugxQfH0gm0MpZMGkN8_-tBvtFOSGWQ@mail.gmail.com>
Message-ID: <CAMRuXGjWZQ=rXvDYSOmQr=cYratFnbStw_Fj40mAbkMX3J9E9w@mail.gmail.com>

Yes please see my question in my previous question

On Mon, Jul 20, 2015 at 9:50 PM, Michael Jerris <mike at jerris.com> wrote:

> Please see the previous thread you started on this same topic where your
> questions have already been answered
>
>
> On Monday, July 20, 2015, Murugan Pandian <manpower13.cse at gmail.com>
> wrote:
>
>> Hi,
>>
>>    I need to use FS only media server and WS-SIP signalling server
>>
>>
>>   Media-server
>>
>>         i disabled auth-call in internal.xml so i can make call from
>> iptel account
>>
>>   WS-SIP
>>
>>         But when i try to use WS-SIP i can't  able to  register using
>>  iptel account getting 408 error(chrome browser)
>>
>>
>>   If i am doing wrong here pls help 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
>
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From mike at jerris.com  Mon Jul 20 22:43:49 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 20 Jul 2015 14:43:49 -0400
Subject: [Freeswitch-users] Mod_verto vs WebRTC(ws/wss)
In-Reply-To: <CAMRuXGgePd1Z47dtaYY0fJpSo2HaS-OKo8c3ziGRHcRNJBTcLg@mail.gmail.com>
References: <CAMRuXGjVxqiuT75WhKE9nWtSXewRYtRxh-LCu+g9Tac7EvQXWg@mail.gmail.com>
	<CAAyX+KFJCswqb1k7WN86udhAxuBza9RTLjYat695aG6cqAR-dQ@mail.gmail.com>
	<CAMRuXGj5Z8LtuRd8_WLR5cy7oQO9a4p32tgZf=L6x0s=GTtoSA@mail.gmail.com>
	<CAMRuXGheGqw9Q5fAzndjyjxS8P3abd2o6bK0Vv_C-i41hwqZOA@mail.gmail.com>
	<CACXsapOFUS20qh2NPGgXkT-ck+CFWsUAiL+=L5PuqnkRNMUR_g@mail.gmail.com>
	<CALXCt0p5cMzrY-PhgfkJceDFkV365XkkFQw3KzfMAacNnkcD8g@mail.gmail.com>
	<CAAyX+KGgTLYJ5vatqF-L+u2YU3W_FxuVf9QQAgHvGDx5wzPO6w@mail.gmail.com>
	<CAMRuXGgePd1Z47dtaYY0fJpSo2HaS-OKo8c3ziGRHcRNJBTcLg@mail.gmail.com>
Message-ID: <AD5E1392-9075-4639-B9B0-EDCFEE86FE65@jerris.com>

you can not use mod_verto for this.  In fact, freeswitch does not support forward registration for sip to sip either, you would use a sip proxy for that.

> On Jul 20, 2015, at 2:29 PM, Murugan Pandian <manpower13.cse at gmail.com> wrote:
> 
> Can i know how i use my sip2sip account gateway in mod_verto
> 
> On Sat, Jul 18, 2015 at 7:48 PM, Ben Langfeld <ben at langfeld.co.uk <mailto:ben at langfeld.co.uk>> wrote:
> To clarify, if you want your WebRTC UA specifically to register to sip2sip, then you must use SIP all the way through. A Verto endpoint cannot register with a SIP registrar.
> 
> On 17 July 2015 at 18:13, Giovanni Maruzzelli <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>> wrote:
> I have understood from your other mail to list that you want to use a sip2sip account from browser.
> 
> You can then use sip.js to connect to a kamailio that goes to sip2sip, or you can use verto to connect to freeswitch that will use your sip2sip account as gateway.
> 
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> 
> On Jul 17, 2015 11:10 PM, "Victor Medina" <victor.medina at cibersys.com <mailto:victor.medina at cibersys.com>> wrote:
> FreeSWITCH will do both without much problems.
> 
> Decide what to use. Your choice.
> El 17/07/2015 16:02, "Murugan Pandian" <manpower13.cse at gmail.com <mailto:manpower13.cse at gmail.com>> escribi?:
> Sorry for Diagram above one not clear 
> 
> 
>          {Browser} 
>                  |     
>                  |
>    (SIP-WS or verto)                                              
>                  |
>         [freeswitch]
>                  |
>                  | 
>          [Register]
>         (iptel or sip2sip)
> 
> On Sat, Jul 18, 2015 at 1:54 AM, Murugan Pandian <manpower13.cse at gmail.com <mailto:manpower13.cse at gmail.com>> wrote:
> Thanks, If there any way i can achieve  this
> 
>        
>          
>         {Browser} 
>                  |     
>                  |
>                  |   (SIP-WS or verto)                                               (iptel or sip2sip)
>                  |-  -   -  -  - -  -  - - - - -[freeswitch]- - - - - - - - -  -  -   [Register]
> 
> 
>    
> 
> On Sat, Jul 18, 2015 at 1:42 AM, Ben Langfeld <ben at langfeld.co.uk <mailto:ben at langfeld.co.uk>> wrote:
> Both mod_verto and SIP-WS are signalling mechanisms for WebRTC. The former is a FreeSWITCH-proprietary protocol, the latter is a standards-based approach. The former is probably easier to understand if you've never done telephony work, the latter requires you to know more stuff.
> 
> On 17 July 2015 at 16:52, Murugan Pandian <manpower13.cse at gmail.com <mailto:manpower13.cse at gmail.com>> wrote:
> Hi,
> 
>    Any one can help me what is difference between mod_verto and  WebRTC(ws/wss) mod.
> 
>  
>                                     
> 
> _________________________________________________________________________
> 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/>
> 
> 
> 
> _________________________________________________________________________
> 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/>
> 
> _________________________________________________________________________
> 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/>
> 
> _________________________________________________________________________
> 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 manpower13.cse at gmail.com  Mon Jul 20 23:05:07 2015
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Tue, 21 Jul 2015 00:35:07 +0530
Subject: [Freeswitch-users] Mod_verto vs WebRTC(ws/wss)
In-Reply-To: <AD5E1392-9075-4639-B9B0-EDCFEE86FE65@jerris.com>
References: <CAMRuXGjVxqiuT75WhKE9nWtSXewRYtRxh-LCu+g9Tac7EvQXWg@mail.gmail.com>
	<CAAyX+KFJCswqb1k7WN86udhAxuBza9RTLjYat695aG6cqAR-dQ@mail.gmail.com>
	<CAMRuXGj5Z8LtuRd8_WLR5cy7oQO9a4p32tgZf=L6x0s=GTtoSA@mail.gmail.com>
	<CAMRuXGheGqw9Q5fAzndjyjxS8P3abd2o6bK0Vv_C-i41hwqZOA@mail.gmail.com>
	<CACXsapOFUS20qh2NPGgXkT-ck+CFWsUAiL+=L5PuqnkRNMUR_g@mail.gmail.com>
	<CALXCt0p5cMzrY-PhgfkJceDFkV365XkkFQw3KzfMAacNnkcD8g@mail.gmail.com>
	<CAAyX+KGgTLYJ5vatqF-L+u2YU3W_FxuVf9QQAgHvGDx5wzPO6w@mail.gmail.com>
	<CAMRuXGgePd1Z47dtaYY0fJpSo2HaS-OKo8c3ziGRHcRNJBTcLg@mail.gmail.com>
	<AD5E1392-9075-4639-B9B0-EDCFEE86FE65@jerris.com>
Message-ID: <CAMRuXGhSgytT6vEpa+3Gi-=DEqomL6ES7z52wi+-xrR45BZ28A@mail.gmail.com>

Can i use SIP-WS to achieve this

On Tue, Jul 21, 2015 at 12:13 AM, Michael Jerris <mike at jerris.com> wrote:

> you can not use mod_verto for this.  In fact, freeswitch does not support
> forward registration for sip to sip either, you would use a sip proxy for
> that.
>
> On Jul 20, 2015, at 2:29 PM, Murugan Pandian <manpower13.cse at gmail.com>
> wrote:
>
> Can i know how i use my sip2sip account gateway in mod_verto
>
> On Sat, Jul 18, 2015 at 7:48 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>
>> To clarify, if you want your WebRTC UA specifically to register to
>> sip2sip, then you must use SIP all the way through. A Verto endpoint cannot
>> register with a SIP registrar.
>>
>> On 17 July 2015 at 18:13, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
>>
>>> I have understood from your other mail to list that you want to use a
>>> sip2sip account from browser.
>>>
>>> You can then use sip.js to connect to a kamailio that goes to sip2sip,
>>> or you can use verto to connect to freeswitch that will use your sip2sip
>>> account as gateway.
>>>
>>> sent from my mobile,
>>> Giovanni Maruzzelli
>>> cell: +39 347 266 56 18
>>> On Jul 17, 2015 11:10 PM, "Victor Medina" <victor.medina at cibersys.com>
>>> wrote:
>>>
>>>> FreeSWITCH will do both without much problems.
>>>>
>>>> Decide what to use. Your choice.
>>>>  El 17/07/2015 16:02, "Murugan Pandian" <manpower13.cse at gmail.com>
>>>> escribi?:
>>>>
>>>>> Sorry for Diagram above one not clear
>>>>>
>>>>>
>>>>>          {Browser}
>>>>>                  |
>>>>>                  |
>>>>>    (SIP-WS or verto)
>>>>>                  |
>>>>>         [freeswitch]
>>>>>                  |
>>>>>                  |
>>>>>          [Register]
>>>>>         (iptel or sip2sip)
>>>>>
>>>>> On Sat, Jul 18, 2015 at 1:54 AM, Murugan Pandian <
>>>>> manpower13.cse at gmail.com> wrote:
>>>>>
>>>>>> Thanks, If there any way i can achieve  this
>>>>>>
>>>>>>
>>>>>>
>>>>>>         {Browser}
>>>>>>                  |
>>>>>>                  |
>>>>>>                  |   (SIP-WS or verto)
>>>>>>                 (iptel or sip2sip)
>>>>>>                  |-  -   -  -  - -  -  - - - - -[freeswitch]- - - - -
>>>>>> - - - -  -  -   [Register]
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Sat, Jul 18, 2015 at 1:42 AM, Ben Langfeld <ben at langfeld.co.uk>
>>>>>> wrote:
>>>>>>
>>>>>>> Both mod_verto and SIP-WS are signalling mechanisms for WebRTC. The
>>>>>>> former is a FreeSWITCH-proprietary protocol, the latter is a
>>>>>>> standards-based approach. The former is probably easier to understand if
>>>>>>> you've never done telephony work, the latter requires you to know more
>>>>>>> stuff.
>>>>>>>
>>>>>>> On 17 July 2015 at 16:52, Murugan Pandian <manpower13.cse at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Hi,
>>>>>>>>
>>>>>>>>    Any one can help me what is difference between mod_verto and
>>>>>>>>  WebRTC(ws/wss) mod.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> 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
>>>
>>
>>
>> _________________________________________________________________________
>> 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 olegstolyar at gmail.com  Mon Jul 20 23:30:22 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Mon, 20 Jul 2015 12:30:22 -0700
Subject: [Freeswitch-users] Websocket creation logging
Message-ID: <CAC8JMOLh5i5iOkvN46Sh0KM6fdwMkZ7hvXpGg-vak2_Tm_gcmg@mail.gmail.com>

Hi guys,

I am experiencing what looks like a memory leak on my FS servers but I
suspect that there is a bug in my WebRTC client code that keep establishing
new websockets and not closing them properly which causes FS to require
more and more memory.

Is there a way in FS to log creation and closing of websockets?

Thank you
Oleg
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From mike at jerris.com  Mon Jul 20 23:37:24 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 20 Jul 2015 15:37:24 -0400
Subject: [Freeswitch-users] Websocket creation logging
In-Reply-To: <CAC8JMOLh5i5iOkvN46Sh0KM6fdwMkZ7hvXpGg-vak2_Tm_gcmg@mail.gmail.com>
References: <CAC8JMOLh5i5iOkvN46Sh0KM6fdwMkZ7hvXpGg-vak2_Tm_gcmg@mail.gmail.com>
Message-ID: <5128BA24-F33E-4B66-A3FE-C54EE2C78474@jerris.com>

In what?  sofia or mod_verto?  in sofia there is probably some tport_debug that would show that, in mod_verto you probably need to add some debug, but you will have to go look at code to be sure.

> On Jul 20, 2015, at 3:30 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> 
> Hi guys,
> 
> I am experiencing what looks like a memory leak on my FS servers but I suspect that there is a bug in my WebRTC client code that keep establishing new websockets and not closing them properly which causes FS to require more and more memory.
> 
> Is there a way in FS to log creation and closing of websockets?
> 
> Thank you
> Oleg
> _________________________________________________________________________
> 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 manpower13.cse at gmail.com  Mon Jul 20 23:42:49 2015
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Tue, 21 Jul 2015 01:12:49 +0530
Subject: [Freeswitch-users] Freeswitch WS+SIPJS+sip2sip
In-Reply-To: <CAAyX+KELEg6St9Ag_CvMbp4Ri-buyyLhM7A-yAPzH5oSEFO=_Q@mail.gmail.com>
References: <CAMRuXGgG+7ps9WEs1UEMwzKC73HMF-6C_JX+ZXkC85h_VzeC_Q@mail.gmail.com>
	<CAEwTsAhzY6spK8VFqCWiqBfTZ-Humutxm7aV8sb7u3o94rFmiw@mail.gmail.com>
	<CAMRuXGj+SfLzxeshtX22tmX3LYto837845T+WnRbDi2GgvaqOw@mail.gmail.com>
	<CAAyX+KELEg6St9Ag_CvMbp4Ri-buyyLhM7A-yAPzH5oSEFO=_Q@mail.gmail.com>
Message-ID: <CAMRuXGgAo1z8YSdMqOR51k9hXTTg=EBcyjn1qo4RKRqgWyAonQ@mail.gmail.com>

how can i achieve this


softphone<-->udp<-->Sip2SIP<------ udp--->FS<------->WS-SIP<--->browser

 |----------------udp-----------|

On Fri, Jul 17, 2015 at 10:51 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:

> What you actually want is a SIP-WS to SIP-UDP proxy like Kamailio. What
> you have setup is not what you think it is, nor is the terminology (JSON)
> you are using correct.
>
> On 17 July 2015 at 13:00, Murugan Pandian <manpower13.cse at gmail.com>
> wrote:
>
>> HI Michael,
>>
>>      Thanks for you answer,Actually i using sip service account to
>> register my browser ,here i am using freeswitch my signalling server (means
>> json to sip because i am using browser) when try to register my sip service
>> account i am getting SIP/2.0 403 Forbidden error
>>
>> On Fri, Jul 17, 2015 at 9:11 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> are you saying you are attempting to register to FreeSWITCH with
>>> credentials for some other service that has nothing to do with your
>>> FreeSWITCH box and its denying you?
>>>
>>> On Friday, July 17, 2015, Murugan Pandian <manpower13.cse at gmail.com>
>>> wrote:
>>>
>>>> Hi,
>>>>
>>>>   I try to register sip2sip.info  account from browser using ws
>>>> signalling,but i am  getting  SIP/2.0 403 Forbidden error  in my console.
>>>>
>>>>  The same sip2sip.info account work in all soft 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
>>
>
>
> _________________________________________________________________________
> 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 manpower13.cse at gmail.com  Mon Jul 20 23:46:58 2015
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Tue, 21 Jul 2015 01:16:58 +0530
Subject: [Freeswitch-users] Mod_verto vs WebRTC(ws/wss)
In-Reply-To: <CAMRuXGhSgytT6vEpa+3Gi-=DEqomL6ES7z52wi+-xrR45BZ28A@mail.gmail.com>
References: <CAMRuXGjVxqiuT75WhKE9nWtSXewRYtRxh-LCu+g9Tac7EvQXWg@mail.gmail.com>
	<CAAyX+KFJCswqb1k7WN86udhAxuBza9RTLjYat695aG6cqAR-dQ@mail.gmail.com>
	<CAMRuXGj5Z8LtuRd8_WLR5cy7oQO9a4p32tgZf=L6x0s=GTtoSA@mail.gmail.com>
	<CAMRuXGheGqw9Q5fAzndjyjxS8P3abd2o6bK0Vv_C-i41hwqZOA@mail.gmail.com>
	<CACXsapOFUS20qh2NPGgXkT-ck+CFWsUAiL+=L5PuqnkRNMUR_g@mail.gmail.com>
	<CALXCt0p5cMzrY-PhgfkJceDFkV365XkkFQw3KzfMAacNnkcD8g@mail.gmail.com>
	<CAAyX+KGgTLYJ5vatqF-L+u2YU3W_FxuVf9QQAgHvGDx5wzPO6w@mail.gmail.com>
	<CAMRuXGgePd1Z47dtaYY0fJpSo2HaS-OKo8c3ziGRHcRNJBTcLg@mail.gmail.com>
	<AD5E1392-9075-4639-B9B0-EDCFEE86FE65@jerris.com>
	<CAMRuXGhSgytT6vEpa+3Gi-=DEqomL6ES7z52wi+-xrR45BZ28A@mail.gmail.com>
Message-ID: <CAMRuXGgvWUWt4DEMH3Yw0F8zx4U8+L70S3idbyCPiUwhBEWE_A@mail.gmail.com>

HI Micheal,

                   Thanks for your response,I try to achieve this
        softphone<--->UDP<---->SIP2SIP<----->FS<------SIP-WS----->browser

 |-----------------UDP------------|


Any one have idea please share here

On Tue, Jul 21, 2015 at 12:35 AM, Murugan Pandian <manpower13.cse at gmail.com>
wrote:

> Can i use SIP-WS to achieve this
>
> On Tue, Jul 21, 2015 at 12:13 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> you can not use mod_verto for this.  In fact, freeswitch does not support
>> forward registration for sip to sip either, you would use a sip proxy for
>> that.
>>
>> On Jul 20, 2015, at 2:29 PM, Murugan Pandian <manpower13.cse at gmail.com>
>> wrote:
>>
>> Can i know how i use my sip2sip account gateway in mod_verto
>>
>> On Sat, Jul 18, 2015 at 7:48 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>>
>>> To clarify, if you want your WebRTC UA specifically to register to
>>> sip2sip, then you must use SIP all the way through. A Verto endpoint cannot
>>> register with a SIP registrar.
>>>
>>> On 17 July 2015 at 18:13, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
>>>
>>>> I have understood from your other mail to list that you want to use a
>>>> sip2sip account from browser.
>>>>
>>>> You can then use sip.js to connect to a kamailio that goes to sip2sip,
>>>> or you can use verto to connect to freeswitch that will use your sip2sip
>>>> account as gateway.
>>>>
>>>> sent from my mobile,
>>>> Giovanni Maruzzelli
>>>> cell: +39 347 266 56 18
>>>> On Jul 17, 2015 11:10 PM, "Victor Medina" <victor.medina at cibersys.com>
>>>> wrote:
>>>>
>>>>> FreeSWITCH will do both without much problems.
>>>>>
>>>>> Decide what to use. Your choice.
>>>>>  El 17/07/2015 16:02, "Murugan Pandian" <manpower13.cse at gmail.com>
>>>>> escribi?:
>>>>>
>>>>>> Sorry for Diagram above one not clear
>>>>>>
>>>>>>
>>>>>>          {Browser}
>>>>>>                  |
>>>>>>                  |
>>>>>>    (SIP-WS or verto)
>>>>>>                  |
>>>>>>         [freeswitch]
>>>>>>                  |
>>>>>>                  |
>>>>>>          [Register]
>>>>>>         (iptel or sip2sip)
>>>>>>
>>>>>> On Sat, Jul 18, 2015 at 1:54 AM, Murugan Pandian <
>>>>>> manpower13.cse at gmail.com> wrote:
>>>>>>
>>>>>>> Thanks, If there any way i can achieve  this
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>         {Browser}
>>>>>>>                  |
>>>>>>>                  |
>>>>>>>                  |   (SIP-WS or verto)
>>>>>>>                 (iptel or sip2sip)
>>>>>>>                  |-  -   -  -  - -  -  - - - - -[freeswitch]- - - -
>>>>>>> - - - - -  -  -   [Register]
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Sat, Jul 18, 2015 at 1:42 AM, Ben Langfeld <ben at langfeld.co.uk>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Both mod_verto and SIP-WS are signalling mechanisms for WebRTC. The
>>>>>>>> former is a FreeSWITCH-proprietary protocol, the latter is a
>>>>>>>> standards-based approach. The former is probably easier to understand if
>>>>>>>> you've never done telephony work, the latter requires you to know more
>>>>>>>> stuff.
>>>>>>>>
>>>>>>>> On 17 July 2015 at 16:52, Murugan Pandian <manpower13.cse at gmail.com
>>>>>>>> > wrote:
>>>>>>>>
>>>>>>>>> Hi,
>>>>>>>>>
>>>>>>>>>    Any one can help me what is difference between mod_verto and
>>>>>>>>>  WebRTC(ws/wss) mod.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> 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
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> 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 olegstolyar at gmail.com  Mon Jul 20 23:55:14 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Mon, 20 Jul 2015 12:55:14 -0700
Subject: [Freeswitch-users] Websocket creation logging
In-Reply-To: <5128BA24-F33E-4B66-A3FE-C54EE2C78474@jerris.com>
References: <CAC8JMOLh5i5iOkvN46Sh0KM6fdwMkZ7hvXpGg-vak2_Tm_gcmg@mail.gmail.com>
	<5128BA24-F33E-4B66-A3FE-C54EE2C78474@jerris.com>
Message-ID: <CAC8JMOL1rzwtN=mR6WWd5t10WteDeXjQwCerP1TwE=vR8+wpmA@mail.gmail.com>

Thanks Michael!

In sofia.  Can you share more about tport_debug?  Where to find it and how
to enable or see?

On Mon, Jul 20, 2015 at 12:37 PM, Michael Jerris <mike at jerris.com> wrote:

> In what?  sofia or mod_verto?  in sofia there is probably some tport_debug
> that would show that, in mod_verto you probably need to add some debug, but
> you will have to go look at code to be sure.
>
> > On Jul 20, 2015, at 3:30 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> >
> > Hi guys,
> >
> > I am experiencing what looks like a memory leak on my FS servers but I
> suspect that there is a bug in my WebRTC client code that keep establishing
> new websockets and not closing them properly which causes FS to require
> more and more memory.
> >
> > Is there a way in FS to log creation and closing of websockets?
> >
> > Thank you
> > Oleg
> > _________________________________________________________________________
> > 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 rtreleaven at bunnykick.ca  Tue Jul 21 00:14:32 2015
From: rtreleaven at bunnykick.ca (Russell Treleaven)
Date: Mon, 20 Jul 2015 16:14:32 -0400
Subject: [Freeswitch-users] Freeswitch WS+SIPJS+sip2sip
In-Reply-To: <CAMRuXGgAo1z8YSdMqOR51k9hXTTg=EBcyjn1qo4RKRqgWyAonQ@mail.gmail.com>
References: <CAMRuXGgG+7ps9WEs1UEMwzKC73HMF-6C_JX+ZXkC85h_VzeC_Q@mail.gmail.com>
	<CAEwTsAhzY6spK8VFqCWiqBfTZ-Humutxm7aV8sb7u3o94rFmiw@mail.gmail.com>
	<CAMRuXGj+SfLzxeshtX22tmX3LYto837845T+WnRbDi2GgvaqOw@mail.gmail.com>
	<CAAyX+KELEg6St9Ag_CvMbp4Ri-buyyLhM7A-yAPzH5oSEFO=_Q@mail.gmail.com>
	<CAMRuXGgAo1z8YSdMqOR51k9hXTTg=EBcyjn1qo4RKRqgWyAonQ@mail.gmail.com>
Message-ID: <CAHzfiZ7Hq03Jth9oyyaj_wG=b8vZfnvNaj=w+1RiH1cC=vxOmQ@mail.gmail.com>

I humbly suggest you consider consulting at freeswitch.org



On Mon, Jul 20, 2015 at 3:42 PM, Murugan Pandian <manpower13.cse at gmail.com>
wrote:

> how can i achieve this
>
>
> softphone<-->udp<-->Sip2SIP<------ udp--->FS<------->WS-SIP<--->browser
>
>  |----------------udp-----------|
>
> On Fri, Jul 17, 2015 at 10:51 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>
>> What you actually want is a SIP-WS to SIP-UDP proxy like Kamailio. What
>> you have setup is not what you think it is, nor is the terminology (JSON)
>> you are using correct.
>>
>> On 17 July 2015 at 13:00, Murugan Pandian <manpower13.cse at gmail.com>
>> wrote:
>>
>>> HI Michael,
>>>
>>>      Thanks for you answer,Actually i using sip service account to
>>> register my browser ,here i am using freeswitch my signalling server (means
>>> json to sip because i am using browser) when try to register my sip service
>>> account i am getting SIP/2.0 403 Forbidden error
>>>
>>> On Fri, Jul 17, 2015 at 9:11 PM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> are you saying you are attempting to register to FreeSWITCH with
>>>> credentials for some other service that has nothing to do with your
>>>> FreeSWITCH box and its denying you?
>>>>
>>>> On Friday, July 17, 2015, Murugan Pandian <manpower13.cse at gmail.com>
>>>> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>>   I try to register sip2sip.info  account from browser using ws
>>>>> signalling,but i am  getting  SIP/2.0 403 Forbidden error  in my console.
>>>>>
>>>>>  The same sip2sip.info account work in all soft 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
>>>
>>
>>
>> _________________________________________________________________________
>> 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 get.prashant.007 at gmail.com  Tue Jul 21 00:22:36 2015
From: get.prashant.007 at gmail.com (Prashant Choudhary)
Date: Tue, 21 Jul 2015 01:52:36 +0530
Subject: [Freeswitch-users] mod_managed vs eventsocket
In-Reply-To: <55ad36a9.0566420a.41cc.ffffef06@mx.google.com>
References: <CAFgB_10ZtaKSdQ-rqew3LZO79h9zWFFYi9qo7E_gpDASSrU+QQ@mail.gmail.com>
	<CAFgB_13syeAtznT78KE46xo7XSkt2LiSpRNTNHMdqTWmemq9ZQ@mail.gmail.com>
	<CAFgB_11tki=JHoAqm+UAj0NhjkYBvSzDM2Kab47biWfjY9Qypg@mail.gmail.com>
	<D1D298D7.F80B%hgeraldino@ipsoft.com>
	<CAEwTsAj+67EX2RLUJCKdOsHbi7_CFbZ5h3pJqgT-E1=jyYnPQQ@mail.gmail.com>
	<55ad36a9.0566420a.41cc.ffffef06@mx.google.com>
Message-ID: <CAFgB_13VFZ7iBozG+vdTdX++apa9XAZfbSVLok4uAPZ3=EFfpg@mail.gmail.com>

I am trying to get mod_unimrcp running.Meanwhile, Can you tell me if
mod_rtmp is meant to do something similar or not.

Thanks

On Mon, Jul 20, 2015 at 11:28 PM, Prashant <get.prashant.007 at gmail.com>
wrote:

> Awesome, thanks for the reply sir.
> I was already trying to record it and play to other service. But I guess
> mod_mrcp seems better.  I will work on it and let you know if I face any
> problem. Thanks
> ------------------------------
> From: Michael Jerris <mike at jerris.com>
> Sent: ?20-?07-?2015 10:39 PM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] mod_managed vs eventsocket
>
> in managed modules if you build an application interface you can access
> the media the same way you do in native c modules.  The c api is marshaled
> into managed code
>
> On Monday, July 20, 2015, Hector Geraldino <Hector.Geraldino at ipsoft.com>
> wrote:
>
>>  I don?t think there?s an API that you can use to tap on the RTP stream.
>> You can always record your call in stereo and post-process the file, if
>> that?s valid for what you?re trying to do.
>>
>>  If not, check the MRCP protocol. I guess you can write an MRCP server
>> (or use an existing one), and stream the audio from FS using mod_unimrcp.
>> That?s how you integrate 3rd party ASR systems (e.g. Nuance?s Recognizer)
>> with FreeSWITCH.
>>
>>   From: <freeswitch-users-bounces at lists.freeswitch.org> on behalf of
>> Prashant Choudhary <get.prashant.007 at gmail.com>
>> Reply-To: "freeswitch-users at lists.freeswitch.org" <
>> freeswitch-users at lists.freeswitch.org>
>> Date: Sunday, July 19, 2015 at 4:24 PM
>> To: "freeswitch-users at lists.freeswitch.org" <
>> freeswitch-users at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-users] mod_managed vs eventsocket
>>
>>   Hello,
>> Now I have even managed to do outbound calls. I have a real problem now.
>> How can I get the audio stream from the person on the other end. I want
>> the audio stream to be sent to my API in real time. I want to detect
>> silence as well, but we can do that by experiment. Only thing is that I am
>> unable to get the audio stream. And this stream should only have the voice
>> of person being called, not mine(i.e. the incoming audio). Is it possible??
>> Please help
>>
>> On Sat, Jul 18, 2015 at 2:57 PM, Prashant Choudhary <
>> get.prashant.007 at gmail.com> wrote:
>>
>>> Sorry for my last mail, which was a little more generic. Thing is, I am
>>> able to receive and manage Inbound calls. But I am unable to figure out,
>>> how to make outbound calls through mod_managed. I was able to do the same
>>> through event socket API. One benefit of event socket was that I could run
>>> it as separate console application , but I am not sure whether it is
>>> possible through mod managed as well.
>>> SO again my question remains the same. But it is more towards outbound
>>> calling.
>>>
>>> On Sat, Jul 18, 2015 at 1:11 AM, Prashant Choudhary <
>>> get.prashant.007 at gmail.com> wrote:
>>>
>>>> Hello All,
>>>>
>>>> I have worked with LUA and freeswitch. Now I want to migrate to .Net
>>>> API as I want to experiment with machine learning using sample calls.  I am
>>>> trying to control around some 2-300 calls directly through my API. This may
>>>> require the call audio stream to be forwarded in real time to .NET
>>>>  I a few questions that might be very silly.
>>>>
>>>> 1. Which one is better to achieve this, mod_managed or ESL.
>>>> 2. I was able to find some examples related to eventsocket, but same
>>>> for mod_Managed was rare. Can you suggest me any doc(of-course except the
>>>> wiki link) or any email chain which explains this or anything else. This
>>>> will be really a big help for me. As currently I am stuck. And I still find
>>>> the LUA way easier.
>>>>
>>>> Thanks
>>>>
>>>
>>>
>>
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From mike at jerris.com  Tue Jul 21 00:27:57 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 20 Jul 2015 16:27:57 -0400
Subject: [Freeswitch-users] Websocket creation logging
In-Reply-To: <CAC8JMOL1rzwtN=mR6WWd5t10WteDeXjQwCerP1TwE=vR8+wpmA@mail.gmail.com>
References: <CAC8JMOLh5i5iOkvN46Sh0KM6fdwMkZ7hvXpGg-vak2_Tm_gcmg@mail.gmail.com>
	<5128BA24-F33E-4B66-A3FE-C54EE2C78474@jerris.com>
	<CAC8JMOL1rzwtN=mR6WWd5t10WteDeXjQwCerP1TwE=vR8+wpmA@mail.gmail.com>
Message-ID: <BF9A62AC-2FAD-40A5-95BF-F7999754A2B7@jerris.com>

check the wiki on sofia debug logging

> On Jul 20, 2015, at 3:55 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> 
> Thanks Michael!
> 
> In sofia.  Can you share more about tport_debug?  Where to find it and how to enable or see?
> 
> On Mon, Jul 20, 2015 at 12:37 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> In what?  sofia or mod_verto?  in sofia there is probably some tport_debug that would show that, in mod_verto you probably need to add some debug, but you will have to go look at code to be sure.
> 
> > On Jul 20, 2015, at 3:30 PM, Oleg Stolyar <olegstolyar at gmail.com <mailto:olegstolyar at gmail.com>> wrote:
> >
> > Hi guys,
> >
> > I am experiencing what looks like a memory leak on my FS servers but I suspect that there is a bug in my WebRTC client code that keep establishing new websockets and not closing them properly which causes FS to require more and more memory.
> >
> > Is there a way in FS to log creation and closing of websockets?
> >
> > Thank you
> > Oleg
> > _________________________________________________________________________
> > 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/>
> 
> _________________________________________________________________________
> 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 Jul 21 00:28:42 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 20 Jul 2015 16:28:42 -0400
Subject: [Freeswitch-users] mod_managed vs eventsocket
In-Reply-To: <CAFgB_13VFZ7iBozG+vdTdX++apa9XAZfbSVLok4uAPZ3=EFfpg@mail.gmail.com>
References: <CAFgB_10ZtaKSdQ-rqew3LZO79h9zWFFYi9qo7E_gpDASSrU+QQ@mail.gmail.com>
	<CAFgB_13syeAtznT78KE46xo7XSkt2LiSpRNTNHMdqTWmemq9ZQ@mail.gmail.com>
	<CAFgB_11tki=JHoAqm+UAj0NhjkYBvSzDM2Kab47biWfjY9Qypg@mail.gmail.com>
	<D1D298D7.F80B%hgeraldino@ipsoft.com>
	<CAEwTsAj+67EX2RLUJCKdOsHbi7_CFbZ5h3pJqgT-E1=jyYnPQQ@mail.gmail.com>
	<55ad36a9.0566420a.41cc.ffffef06@mx.google.com>
	<CAFgB_13VFZ7iBozG+vdTdX++apa9XAZfbSVLok4uAPZ3=EFfpg@mail.gmail.com>
Message-ID: <92B15BA8-A9AE-41D7-AFFF-F0395F9772BC@jerris.com>

rtmp if for using rtmp in the browser as a client

> On Jul 20, 2015, at 4:22 PM, Prashant Choudhary <get.prashant.007 at gmail.com> wrote:
> 
> I am trying to get mod_unimrcp running.Meanwhile, Can you tell me if mod_rtmp is meant to do something similar or not.
> 
> Thanks
> 
> On Mon, Jul 20, 2015 at 11:28 PM, Prashant <get.prashant.007 at gmail.com <mailto:get.prashant.007 at gmail.com>> wrote:
> Awesome, thanks for the reply sir.
> I was already trying to record it and play to other service. But I guess mod_mrcp seems better.  I will work on it and let you know if I face any problem. Thanks
> From: Michael Jerris <mailto:mike at jerris.com>
> Sent: ?20-?07-?2015 10:39 PM
> To: FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] mod_managed vs eventsocket
> 
> in managed modules if you build an application interface you can access the media the same way you do in native c modules.  The c api is marshaled into managed code
> 
> On Monday, July 20, 2015, Hector Geraldino <Hector.Geraldino at ipsoft.com <mailto:Hector.Geraldino at ipsoft.com>> wrote:
> I don?t think there?s an API that you can use to tap on the RTP stream. You can always record your call in stereo and post-process the file, if that?s valid for what you?re trying to do.
> 
> If not, check the MRCP protocol. I guess you can write an MRCP server (or use an existing one), and stream the audio from FS using mod_unimrcp. That?s how you integrate 3rd party ASR systems (e.g. Nuance?s Recognizer) with FreeSWITCH.
> 
> From: <freeswitch-users-bounces at lists.freeswitch.org <>> on behalf of Prashant Choudhary <get.prashant.007 at gmail.com <>>
> Reply-To: "freeswitch-users at lists.freeswitch.org <>" <freeswitch-users at lists.freeswitch.org <>>
> Date: Sunday, July 19, 2015 at 4:24 PM
> To: "freeswitch-users at lists.freeswitch.org <>" <freeswitch-users at lists.freeswitch.org <>>
> Subject: Re: [Freeswitch-users] mod_managed vs eventsocket
> 
> Hello,
> Now I have even managed to do outbound calls. I have a real problem now.
> How can I get the audio stream from the person on the other end. I want the audio stream to be sent to my API in real time. I want to detect silence as well, but we can do that by experiment. Only thing is that I am unable to get the audio stream. And this stream should only have the voice of person being called, not mine(i.e. the incoming audio). Is it possible??
> Please help
> 
> On Sat, Jul 18, 2015 at 2:57 PM, Prashant Choudhary <get.prashant.007 at gmail.com <>> wrote:
> Sorry for my last mail, which was a little more generic. Thing is, I am able to receive and manage Inbound calls. But I am unable to figure out, how to make outbound calls through mod_managed. I was able to do the same through event socket API. One benefit of event socket was that I could run it as separate console application , but I am not sure whether it is possible through mod managed as well.
> SO again my question remains the same. But it is more towards outbound calling.
> 
> On Sat, Jul 18, 2015 at 1:11 AM, Prashant Choudhary <get.prashant.007 at gmail.com <>> wrote:
> Hello All,
> 
> I have worked with LUA and freeswitch. Now I want to migrate to .Net API as I want to experiment with machine learning using sample calls.  I am trying to control around some 2-300 calls directly through my API. This may require the call audio stream to be forwarded in real time to .NET
>  I a few questions that might be very silly.
> 
> 1. Which one is better to achieve this, mod_managed or ESL.
> 2. I was able to find some examples related to eventsocket, but same for mod_Managed was rare. Can you suggest me any doc(of-course except the wiki link) or any email chain which explains this or anything else. This will be really a big help for me. As currently I am stuck. And I still find the LUA way easier.
> 
> Thanks
> 
> 
> 
> _________________________________________________________________________
> 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 olegstolyar at gmail.com  Tue Jul 21 00:59:36 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Mon, 20 Jul 2015 13:59:36 -0700
Subject: [Freeswitch-users] Websocket creation logging
In-Reply-To: <BF9A62AC-2FAD-40A5-95BF-F7999754A2B7@jerris.com>
References: <CAC8JMOLh5i5iOkvN46Sh0KM6fdwMkZ7hvXpGg-vak2_Tm_gcmg@mail.gmail.com>
	<5128BA24-F33E-4B66-A3FE-C54EE2C78474@jerris.com>
	<CAC8JMOL1rzwtN=mR6WWd5t10WteDeXjQwCerP1TwE=vR8+wpmA@mail.gmail.com>
	<BF9A62AC-2FAD-40A5-95BF-F7999754A2B7@jerris.com>
Message-ID: <CAC8JMO+9sc8nNVO1J31L7jvoDk8USK5oiywVv2tAKFxJjOr_Tw@mail.gmail.com>

I already have this in my sofia.conf.xml
<param name="tracelevel" value="DEBUG"/>

How do I find tport related info in my log files?  I searched for t-port
and tport and didn't find anything.

Do I also need to set up log-level to debug?  I thought log-level was for
the console and according to the wiki tracelevel was for the log files.

On Mon, Jul 20, 2015 at 1:27 PM, Michael Jerris <mike at jerris.com> wrote:

> check the wiki on sofia debug logging
>
> On Jul 20, 2015, at 3:55 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
> Thanks Michael!
>
> In sofia.  Can you share more about tport_debug?  Where to find it and
> how to enable or see?
>
> On Mon, Jul 20, 2015 at 12:37 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> In what?  sofia or mod_verto?  in sofia there is probably some
>> tport_debug that would show that, in mod_verto you probably need to add
>> some debug, but you will have to go look at code to be sure.
>>
>> > On Jul 20, 2015, at 3:30 PM, Oleg Stolyar <olegstolyar at gmail.com>
>> wrote:
>> >
>> > Hi guys,
>> >
>> > I am experiencing what looks like a memory leak on my FS servers but I
>> suspect that there is a bug in my WebRTC client code that keep establishing
>> new websockets and not closing them properly which causes FS to require
>> more and more memory.
>> >
>> > Is there a way in FS to log creation and closing of websockets?
>> >
>> > Thank you
>> > Oleg
>> >
>> _________________________________________________________________________
>> > 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 ben at langfeld.co.uk  Tue Jul 21 01:04:45 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Mon, 20 Jul 2015 18:04:45 -0300
Subject: [Freeswitch-users] Mod_verto vs WebRTC(ws/wss)
In-Reply-To: <CAMRuXGgvWUWt4DEMH3Yw0F8zx4U8+L70S3idbyCPiUwhBEWE_A@mail.gmail.com>
References: <CAMRuXGjVxqiuT75WhKE9nWtSXewRYtRxh-LCu+g9Tac7EvQXWg@mail.gmail.com>
	<CAAyX+KFJCswqb1k7WN86udhAxuBza9RTLjYat695aG6cqAR-dQ@mail.gmail.com>
	<CAMRuXGj5Z8LtuRd8_WLR5cy7oQO9a4p32tgZf=L6x0s=GTtoSA@mail.gmail.com>
	<CAMRuXGheGqw9Q5fAzndjyjxS8P3abd2o6bK0Vv_C-i41hwqZOA@mail.gmail.com>
	<CACXsapOFUS20qh2NPGgXkT-ck+CFWsUAiL+=L5PuqnkRNMUR_g@mail.gmail.com>
	<CALXCt0p5cMzrY-PhgfkJceDFkV365XkkFQw3KzfMAacNnkcD8g@mail.gmail.com>
	<CAAyX+KGgTLYJ5vatqF-L+u2YU3W_FxuVf9QQAgHvGDx5wzPO6w@mail.gmail.com>
	<CAMRuXGgePd1Z47dtaYY0fJpSo2HaS-OKo8c3ziGRHcRNJBTcLg@mail.gmail.com>
	<AD5E1392-9075-4639-B9B0-EDCFEE86FE65@jerris.com>
	<CAMRuXGhSgytT6vEpa+3Gi-=DEqomL6ES7z52wi+-xrR45BZ28A@mail.gmail.com>
	<CAMRuXGgvWUWt4DEMH3Yw0F8zx4U8+L70S3idbyCPiUwhBEWE_A@mail.gmail.com>
Message-ID: <CAAyX+KEsH42W2NVQ+x7NmXu=jed374FwEZq6msxc5CyYGj3mdg@mail.gmail.com>

I think your best bet would be to buy some consulting services to help you
understand all of this. Many people on this list would be happy to help
you. Many of us have experience with such scenarios, and some of us have
published materials on relevant issues. You should contact the FreeSWITCH
biz list or one of us off-list to negotiate some paid assistance.

On 20 July 2015 at 16:46, Murugan Pandian <manpower13.cse at gmail.com> wrote:

> HI Micheal,
>
>                    Thanks for your response,I try to achieve this
>         softphone<--->UDP<---->SIP2SIP<----->FS<------SIP-WS----->browser
>
>  |-----------------UDP------------|
>
>
> Any one have idea please share here
>
> On Tue, Jul 21, 2015 at 12:35 AM, Murugan Pandian <
> manpower13.cse at gmail.com> wrote:
>
>> Can i use SIP-WS to achieve this
>>
>> On Tue, Jul 21, 2015 at 12:13 AM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> you can not use mod_verto for this.  In fact, freeswitch does not
>>> support forward registration for sip to sip either, you would use a sip
>>> proxy for that.
>>>
>>> On Jul 20, 2015, at 2:29 PM, Murugan Pandian <manpower13.cse at gmail.com>
>>> wrote:
>>>
>>> Can i know how i use my sip2sip account gateway in mod_verto
>>>
>>> On Sat, Jul 18, 2015 at 7:48 PM, Ben Langfeld <ben at langfeld.co.uk>
>>> wrote:
>>>
>>>> To clarify, if you want your WebRTC UA specifically to register to
>>>> sip2sip, then you must use SIP all the way through. A Verto endpoint cannot
>>>> register with a SIP registrar.
>>>>
>>>> On 17 July 2015 at 18:13, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>>> wrote:
>>>>
>>>>> I have understood from your other mail to list that you want to use a
>>>>> sip2sip account from browser.
>>>>>
>>>>> You can then use sip.js to connect to a kamailio that goes to sip2sip,
>>>>> or you can use verto to connect to freeswitch that will use your sip2sip
>>>>> account as gateway.
>>>>>
>>>>> sent from my mobile,
>>>>> Giovanni Maruzzelli
>>>>> cell: +39 347 266 56 18
>>>>> On Jul 17, 2015 11:10 PM, "Victor Medina" <victor.medina at cibersys.com>
>>>>> wrote:
>>>>>
>>>>>> FreeSWITCH will do both without much problems.
>>>>>>
>>>>>> Decide what to use. Your choice.
>>>>>>  El 17/07/2015 16:02, "Murugan Pandian" <manpower13.cse at gmail.com>
>>>>>> escribi?:
>>>>>>
>>>>>>> Sorry for Diagram above one not clear
>>>>>>>
>>>>>>>
>>>>>>>          {Browser}
>>>>>>>                  |
>>>>>>>                  |
>>>>>>>    (SIP-WS or verto)
>>>>>>>                  |
>>>>>>>         [freeswitch]
>>>>>>>                  |
>>>>>>>                  |
>>>>>>>          [Register]
>>>>>>>         (iptel or sip2sip)
>>>>>>>
>>>>>>> On Sat, Jul 18, 2015 at 1:54 AM, Murugan Pandian <
>>>>>>> manpower13.cse at gmail.com> wrote:
>>>>>>>
>>>>>>>> Thanks, If there any way i can achieve  this
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>         {Browser}
>>>>>>>>                  |
>>>>>>>>                  |
>>>>>>>>                  |   (SIP-WS or verto)
>>>>>>>>                   (iptel or sip2sip)
>>>>>>>>                  |-  -   -  -  - -  -  - - - - -[freeswitch]- - - -
>>>>>>>> - - - - -  -  -   [Register]
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Sat, Jul 18, 2015 at 1:42 AM, Ben Langfeld <ben at langfeld.co.uk>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>>> Both mod_verto and SIP-WS are signalling mechanisms for WebRTC.
>>>>>>>>> The former is a FreeSWITCH-proprietary protocol, the latter is a
>>>>>>>>> standards-based approach. The former is probably easier to understand if
>>>>>>>>> you've never done telephony work, the latter requires you to know more
>>>>>>>>> stuff.
>>>>>>>>>
>>>>>>>>> On 17 July 2015 at 16:52, Murugan Pandian <
>>>>>>>>> manpower13.cse at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Hi,
>>>>>>>>>>
>>>>>>>>>>    Any one can help me what is difference between mod_verto and
>>>>>>>>>>  WebRTC(ws/wss) mod.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> 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
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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 krice at freeswitch.org  Tue Jul 21 02:39:54 2015
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 20 Jul 2015 22:39:54 +0000
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) July
	11th-July 17th
Message-ID: <55ad78baed4fc_779ad9d334351eb@resque-worker-high.0.mail>

New Post on freeswitch.org from Kathleen King
check it out at http://ift.tt/1JtgsMJ
FreeSWITCH Week in Review (Master Branch) July 11th-July 17th
Hello, again. This passed week in the FreeSWITCH master branch we had 43 commits. We had a number of cool new features this week including: added functionality for capturing screenshots from both legs to uuid_write_png, the addition of new multi-canvas and telepresence features in mod_conference, the addition of?vmute member flag to mod_conference, and?an API for removing an active ladspa effect on a channel.

Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support.

New features that were added:

FS-7769 [mod_conference] Add new multi-canvas and telepresence features

FS-7847 [mod_conference] Add layers that do not match the aspect ration of conference by using the new hscale layer param for horizontal scale, and add zoom=true param to crop layer instead of letterbox, add grid-zoom layout group that demonstrates these layouts, and?fix logo ratios and add borders too.

FS-7813 [mod_conference] Add vmute member flag.

FS-7846 [mod_dptools] Add eavesdrop_whisper_aleg=true and eavesdrop_whisper_bleg=true channel variables to allow you to start eavesdrop in whisper mode of specific call leg

FS-7760 [mod_sofia] Revise channel fetch on nightmare transfer and add dial-prefix and absolute-dial-string to the nightmare xml

FS-7829 [mod_opus] Add sprop-stereo fmtp param to specify if a sender is likely to send stereo or not so the receiver can safely downmix to mono to avoid wasting receiver resources

FS-7830 [mod_opus] Added use-dtx param in config file (enables DTX on the encoder, announces in fmtp)

FS-7824?[mod_png] Add functionality for capturing screenshots from both legs to uuid_write_png

FS-7549 [mod_ladspa] Added an API for removing an active ladspa effect on a channel. For conformance reasons, the uuid_ladspa command now accepts ?stop? and ?start?, while the previous functionality (without any verb) which will simply add ladspa remains intact.

Improvements in build system, cross platform support, and packaging:

FS-7845 [mod_conference]?Break up mod_conference into multiple source files to improve build performance

FS-7769 [mod_conference] Fixed a build issue

FS-7820 Fix build system typo. Don?t assign the same variable twice.

FS-7043 Fixed apr1 unresolved symbols in libfreeswitch.so.1.0.0

FS-7130 Make /run/freeswitch persistent in the Debian packages, so it will start under systemd

The following bugs were squashed:

FS-7849?[verto]?Remove extra div breaking full screen in html

FS-7832 [mod_opus] Fixes when comparing local and remote fmtp params

FS-7731 [mod_xml_cdr] Fixed a curl default connection timeout

FS-7844 Fix packet loss fraction when calculating loss average

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From smontour at twc.com  Mon Jul 20 23:54:07 2015
From: smontour at twc.com (Admin)
Date: Mon, 20 Jul 2015 14:54:07 -0500
Subject: [Freeswitch-users] Disable 401 and 407 for internal profile
Message-ID: <1437422047.3145.1.camel@twc.com>

Hi,
 I want to configure FS to not challenge calls that come in from an
internal profile (internal network devices), meaning disable SIP '401
Unauthorized' and '407 Proxy Authentication Required' for calls that
belong to internal profile. I changed 'auth-calls' parameter to 'false',
in internal.xml file but FS still sends 401 method when a user attempts
to register and it sends 407 when an invite comes in. It seems the
parameter is for ACL though. Is there another parameter to change in
order to disable SIP 401 and 407 methods? 

Thank you.




From yadenis at seznam.cz  Tue Jul 21 15:22:24 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Tue, 21 Jul 2015 13:22:24 +0200
Subject: [Freeswitch-users] VideoCall with h264
In-Reply-To: <1437422047.3145.1.camel@twc.com>
References: <1437422047.3145.1.camel@twc.com>
Message-ID: <917907795.20150721132224@seznam.cz>

Hi All,

The question in the following. Is it possible to make calls through freeswitch using h264 instead of using vp8?


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From brian at freeswitch.org  Tue Jul 21 16:03:13 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 21 Jul 2015 07:03:13 -0500
Subject: [Freeswitch-users] Disable 401 and 407 for internal profile
In-Reply-To: <1437422047.3145.1.camel@twc.com>
References: <1437422047.3145.1.camel@twc.com>
Message-ID: <CAEJMVkAUQ-Tz0RKHUBkbKxSP9rMEo0bdSr9sY5CPqqtN=C1MPg@mail.gmail.com>

Remove the inbound acl too

On Monday, July 20, 2015, Admin <smontour at twc.com> wrote:

> Hi,
>  I want to configure FS to not challenge calls that come in from an
> internal profile (internal network devices), meaning disable SIP '401
> Unauthorized' and '407 Proxy Authentication Required' for calls that
> belong to internal profile. I changed 'auth-calls' parameter to 'false',
> in internal.xml file but FS still sends 401 method when a user attempts
> to register and it sends 407 when an invite comes in. It seems the
> parameter is for ACL though. Is there another parameter to change in
> order to disable SIP 401 and 407 methods?
>
> Thank you.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> 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 <javascript:;>
> 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
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From trever at middleearth.sapphiresunday.org  Tue Jul 21 16:58:07 2015
From: trever at middleearth.sapphiresunday.org (Trever L. Adams)
Date: Tue, 21 Jul 2015 06:58:07 -0600
Subject: [Freeswitch-users] script answers one session, creates another,
 need to hand off one to another lua without hanging up
Message-ID: <55AE41DF.80804@middleearth.sapphiresunday.org>

Hello Everyone,

I am stuck. I have a lua script that gets run from a bind digit setup.
The only way to make it work was to have the bind digit call another lua
script that contains:

theret = api:executeString("uuid_dual_transfer " .. my_uuid .. "
'm:^:set:number=" .. call_id_num .. "^ set:name=" .. call_id_name ..
"^lua:lua_script.lua/inline' " .. "ext/XML/context")

The incoming call goes to the ext the lua script is a menu for the
internal user.

I have another script that answers some incoming calls. It then creates
a new session and gives the internal user a menu. One option is to
connect, another to voicemail. I have these working. The problem is that
one option is to call the lua script above and to send the incoming call
to the same ext. At one point I had it to where it would do one and then
the other. I need a way to make them happen simultaneously.

The lua script in the executeString above uses session, but the session
is created as internal_session in the other script.

i have tried bridging with execute_on_answer calling the script with the
dual transfer above, but it executes AFTER hangup.

Has anyone found a way to solve this problem? session:executeString
doesn't seem to exist anymore for lua.

Thank you,
Trever

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From boros at vmtele.com  Tue Jul 21 19:05:38 2015
From: boros at vmtele.com (=?UTF-8?B?VG9tw6HFoSBCb3Jvcw==?=)
Date: Tue, 21 Jul 2015 17:05:38 +0200
Subject: [Freeswitch-users] problem with attended transfer
Message-ID: <55AE5FC2.4090207@vmtele.com>

Hello all,

I think I have found a bug in version 
1.5.final+git~20150528T173517Z~6a2fc5e0f7~64bit

My scenario is the following:


A ----> calls ---> B

B ----> calls ---> C

B ---> sends REFER to join A and C (attended transfer via SIP)


The two calls are connected A is able to talk to C, but after a while, 
the call is disconnected.

When the REFER message arrives, FS sends BYE message to B with cause 
"NORMAL_CLEARING" and after a while (about 10 sec) FS sends a BYE 
message to C with cause "ATTENDED_TRANSFER" and the transferred call is 
hung up.

I assume, that those causes should be swapped. BYE with attended 
transfer to B and no BYE at all for C until the call is ended by the 
endpoints.

Do someone have this issue, how you fixed it? Should I upgrade my 
freeswitch to a newer version?
-- 
Tom?? Boros
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From krice at freeswitch.org  Tue Jul 21 19:08:37 2015
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 21 Jul 2015 10:08:37 -0500
Subject: [Freeswitch-users] problem with attended transfer
In-Reply-To: <55AE5FC2.4090207@vmtele.com>
Message-ID: <D1D3CAA5.A2D4D%krice@freeswitch.org>

You should either be using 1.4 or master... 1.5 was never a release
version... Retest with a proper version and then report any bugs to Jira


On 7/21/15, 10:05 AM, "Tom?? Boros" <boros at vmtele.com> wrote:

>    Hello all,
>  
>  I think I have found a bug in version
> 1.5.final+git~20150528T173517Z~6a2fc5e0f7~64bit
>  
>  My scenario is the following:
>  
>  
>  A ----> calls ---> B
>  
>  B ----> calls ---> C
>  
>  B ---> sends REFER to join A and C (attended transfer via SIP)
>  
>  
>  The two calls are connected A is able to talk to C, but after a while, the
> call is disconnected.
>  
>  When the REFER message arrives, FS sends BYE message to B with cause
> "NORMAL_CLEARING" and after a while (about 10 sec) FS sends a BYE message to C
> with cause "ATTENDED_TRANSFER" and the transferred call is hung up.
>  
>  I assume, that those causes should be swapped. BYE with attended transfer to
> B and no BYE at all for C until the call is ended by the endpoints.
>  
>  Do someone have this issue, how you fixed it? Should I upgrade my freeswitch
> to a newer version?
>  

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From brian at freeswitch.org  Tue Jul 21 19:09:25 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 21 Jul 2015 10:09:25 -0500
Subject: [Freeswitch-users] problem with attended transfer
In-Reply-To: <55AE5FC2.4090207@vmtele.com>
References: <55AE5FC2.4090207@vmtele.com>
Message-ID: <CAEJMVkATHka52KPjp6Qi8J7g1ZxtGHeHjLTR7kZHMWacKNBJZA@mail.gmail.com>

I would give the 1.4.20 release a try, and optionally try master which is
1.7.x, Then if that fails to produce results:
https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA

On Tue, Jul 21, 2015 at 10:05 AM, Tom?? Boros <boros at vmtele.com> wrote:

>  Hello all,
>
> I think I have found a bug in version
> 1.5.final+git~20150528T173517Z~6a2fc5e0f7~64bit
>
> My scenario is the following:
>
>
> A ----> calls ---> B
>
> B ----> calls ---> C
>
> B ---> sends REFER to join A and C (attended transfer via SIP)
>
>
> The two calls are connected A is able to talk to C, but after a while, the
> call is disconnected.
>
> When the REFER message arrives, FS sends BYE message to B with cause
> "NORMAL_CLEARING" and after a while (about 10 sec) FS sends a BYE message
> to C with cause "ATTENDED_TRANSFER" and the transferred call is hung up.
>
> I assume, that those causes should be swapped. BYE with attended transfer
> to B and no BYE at all for C until the call is ended by the endpoints.
>
> Do someone have this issue, how you fixed it? Should I upgrade my
> freeswitch to a newer version?
> --
>  Tom?? Boros
>
> _________________________________________________________________________
> 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
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From mike at jerris.com  Tue Jul 21 19:57:24 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 21 Jul 2015 11:57:24 -0400
Subject: [Freeswitch-users] VideoCall with h264
In-Reply-To: <917907795.20150721132224@seznam.cz>
References: <1437422047.3145.1.camel@twc.com>
	<917907795.20150721132224@seznam.cz>
Message-ID: <CAEwTsAhB7GkoPpOEcyj7jgdardDRoA9ZaakKAWkcvHU7ixqLBQ@mail.gmail.com>

This currently works fine on Debian 8 Jessie.  To my knowledge other
distributions do not include all the required dependencies so it should be
possible but would require you to handle getting all of the dependencies
sorted out yourself.  We will not be putting those additional dependencies
for pattented algorithms in our repositories.

On Tuesday, July 21, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Hi All,
>
> The question in the following. Is it possible to make calls through
> freeswitch using h264 instead of using vp8?
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 * <http://www.freeswitch.org>
>
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From mahdi_shirazi at yahoo.com  Tue Jul 21 16:29:05 2015
From: mahdi_shirazi at yahoo.com (Mehdi Shirazi)
Date: Tue, 21 Jul 2015 12:29:05 +0000 (UTC)
Subject: [Freeswitch-users] mod_fifo in a database shared between multiple
	freswitches
Message-ID: <1910917121.1743240.1437481745831.JavaMail.yahoo@mail.yahoo.com>

HiIs it possible to use mode_fifo in a shared database to have a big call center on multi server freeswitch environment ?
RegardsMehdi
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From 00asgaroth00 at gmail.com  Tue Jul 21 17:55:41 2015
From: 00asgaroth00 at gmail.com (Asgaroth)
Date: Tue, 21 Jul 2015 14:55:41 +0100
Subject: [Freeswitch-users] Enterprise originate not honouring fs_path in
	302 redirect
Message-ID: <55AE4F5D.7030307@gmail.com>

Hi All,

First of all, I'm a relatively new user of FreeSWITCH so, please bear 
with me if I'm asking a silly question.

I have users registering through multiple kamailio 
loadbalancer/registrars, so when we build a bridge command to multiple 
local users we send the initial invite(s) to the kamailio lookup server 
which then responds with a 302 redirect message specifying the contact 
details of the user.

What I am trying to do is get FreeSWITCH to call a particular user via a 
particular loadbalancer based on information recieved in the 302 
redirect. From what I have read in the archives, we can achieve this 
with the fs_path directive on the bridge command. However, how/where do 
I specify the fs_path in the contact header in the 302 redirect in such 
a manner that FreeSWITCH will honour the fs_path directive.

I originally tried setting a fs_path parameter in the contact of the 302 
redirect message, something like the following:

Contact: 
<sip:subscriber at 1.2.3.4:5678;rinstance=9af71926212a25cf;transport=UDP;received=sip:5.6.7.8:9012;fs_path=sip:1.2.3.5>

However, FreeSWITCH ignores the fs_path directive here and attempts to 
send this call directly to 1.2.3.4, instead of sending it via 1.2.3.5.

Is this at all possible using the enterprise originate (:_: seperator) 
feature of the bridge command?

We're attempting this with FreeSWITCH v 1.4.20 ("official" rpm packages 
from the repo)

Any info/tips/suggestions/beatings much appreciate :)

Thanks


From mike at jerris.com  Tue Jul 21 20:25:31 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 21 Jul 2015 12:25:31 -0400
Subject: [Freeswitch-users] mod_fifo in a database shared between
	multiple freswitches
In-Reply-To: <1910917121.1743240.1437481745831.JavaMail.yahoo@mail.yahoo.com>
References: <1910917121.1743240.1437481745831.JavaMail.yahoo@mail.yahoo.com>
Message-ID: <A831E657-9C84-4FFE-B147-833C331BC78A@jerris.com>

no

> On Jul 21, 2015, at 8:29 AM, Mehdi Shirazi <mahdi_shirazi at yahoo.com> wrote:
> 
> Hi
> Is it possible to use mode_fifo in a shared database to have a big call center on multi server freeswitch environment ?
> 
> Regards
> Mehdi
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From smontour at twc.com  Tue Jul 21 20:36:10 2015
From: smontour at twc.com (Admin)
Date: Tue, 21 Jul 2015 11:36:10 -0500
Subject: [Freeswitch-users]  Disable 401 and 407 for internal profile
Message-ID: <1437496570.3313.11.camel@twc.com>

Thanks Brian. That seems to work partially and strangely too. All calls
made to extensions above '1019' stopped working now. When a call is
destined to any extension above '1019' comes back with '480 Temporarily
Unavailable'. Below is a pasted copy of the SIP trace for a call from
1001 to 1019 (works), and 1001 to 1020 (doesn't work). Registration
output is also pasted below.

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

recv 726 bytes from udp/[192.168.1.100]:5062 at 11:18:28.880410:

------------------------------------------------------------------------
   INVITE sip:1020 at 192.168.1.100 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3263-1-0
   From: 1001 <sip:1001 at 192.168.1.100>;tag=32631
   To: 1020 <sip:1020 at 192.168.1.100>
   Call-ID: 1-3263 at 192.168.1.100
   CSeq: 1 INVITE
   Contact: 1001 <sip:1001 at 192.168.1.100:5062>
   Max-Forwards: 70
   Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE,
UPDATE, SUBSCRIBE
   Supported: 100rel, replaces, from-change
   Timestamp: 1
   Content-Type: application/sdp
   Content-Length:   238
   
   v=0
   o=1001 53655765 2353687637 IN IP4 192.168.1.100
   s=Talk
   c=IN IP4 192.168.1.100
   t=0 0
   m=audio 6001 RTP/AVP 0 8 97
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:97 iLBC/8000
   m=video 6001 RTP/AVP 31
   a=rtpmap:31 H261/90000

------------------------------------------------------------------------
send 333 bytes to udp/[192.168.1.100]:5062 at 11:18:28.880719:

------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3263-1-0
   From: 1001 <sip:1001 at 192.168.1.100>;tag=32631
   To: 1020 <sip:1020 at 192.168.1.100>
   Call-ID: 1-3263 at 192.168.1.100
   CSeq: 1 INVITE
   Timestamp: 1 0.000152
   User-Agent: FreeSWITCH-mod_sofia/1.4.20
+git~20150703T164215Z~b95362f965~64bit
   Content-Length: 0

------------------------------------------------------------------------
2015-07-21 11:18:28.863966 [NOTICE] switch_channel.c:1075 New Channel
sofia/internal/1001 at 192.168.1.100 [d4084b55-8351-48fa-87ff-01d4b2355d4e]
2015-07-21 11:18:28.863966 [INFO] mod_dialplan_xml.c:635 Processing 1001
<1001>->1020 in context public
2015-07-21 11:18:28.863966 [NOTICE] switch_core_state_machine.c:315
sofia/internal/1001 at 192.168.1.100 has executed the last dialplan
instruction, hanging up.
2015-07-21 11:18:28.863966 [NOTICE] switch_core_state_machine.c:317
Hangup sofia/internal/1001 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]
send 836 bytes to udp/[192.168.1.100]:5062 at 11:18:28.885154:

------------------------------------------------------------------------
   SIP/2.0 480 Temporarily Unavailable
   Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3263-1-0
   Max-Forwards: 70
   From: 1001 <sip:1001 at 192.168.1.100>;tag=32631
   To: 1020 <sip:1020 at 192.168.1.100>;tag=mrKveDD123K3K
   Call-ID: 1-3263 at 192.168.1.100
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.4.20
+git~20150703T164215Z~b95362f965~64bit
   Accept: application/sdp
   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
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0
   Remote-Party-ID: "1020"
<sip:1020 at 192.168.1.100>;party=calling;privacy=off;screen=no

------------------------------------------------------------------------
2015-07-21 11:18:28.883950 [NOTICE] switch_core_session.c:1641 Session
20 (sofia/internal/1001 at 192.168.1.100) Ended
2015-07-21 11:18:28.883950 [NOTICE] switch_core_session.c:1645 Close
Channel sofia/internal/1001 at 192.168.1.100 [CS_DESTROY]
recv 361 bytes from udp/[192.168.1.100]:5062 at 11:18:28.887876:

------------------------------------------------------------------------
   ACK sip:1020 at 192.168.1.100 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3263-1-0
   From: 1001 <sip:1001 at 192.168.1.100>;tag=32631
   To: 1020 <sip:1020 at 192.168.1.100>;tag=mrKveDD123K3K
   Call-ID: 1-3263 at 192.168.1.100
   CSeq: 1 ACK
   Contact: <sip:sipp at 192.168.1.100:5062;transport=UDP>
   Max-Forwards: 70
   Subject: Performance Test
   Content-Length: 0
   

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

>>>>>>>>>>> 1001 - 1019 - Working >>>>>>>>>>>>>>>>

recv 726 bytes from udp/[192.168.1.100]:5062 at 11:20:10.977891:

------------------------------------------------------------------------
   INVITE sip:1019 at 192.168.1.100 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0
   From: 1001 <sip:1001 at 192.168.1.100>;tag=32811
   To: 1019 <sip:1019 at 192.168.1.100>
   Call-ID: 1-3281 at 192.168.1.100
   CSeq: 1 INVITE
   Contact: 1001 <sip:1001 at 192.168.1.100:5062>
   Max-Forwards: 70
   Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE,
UPDATE, SUBSCRIBE
   Supported: 100rel, replaces, from-change
   Timestamp: 1
   Content-Type: application/sdp
   Content-Length:   238
   
   v=0
   o=1001 53655765 2353687637 IN IP4 192.168.1.100
   s=Talk
   c=IN IP4 192.168.1.100
   t=0 0
   m=audio 6001 RTP/AVP 0 8 97
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:97 iLBC/8000
   m=video 6001 RTP/AVP 31
   a=rtpmap:31 H261/90000

------------------------------------------------------------------------
send 333 bytes to udp/[192.168.1.100]:5062 at 11:20:10.978324:

------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0
   From: 1001 <sip:1001 at 192.168.1.100>;tag=32811
   To: 1019 <sip:1019 at 192.168.1.100>
   Call-ID: 1-3281 at 192.168.1.100
   CSeq: 1 INVITE
   Timestamp: 1 0.000208
   User-Agent: FreeSWITCH-mod_sofia/1.4.20
+git~20150703T164215Z~b95362f965~64bit
   Content-Length: 0
   

------------------------------------------------------------------------
2015-07-21 11:20:10.963951 [NOTICE] switch_channel.c:1075 New Channel
sofia/internal/1001 at 192.168.1.100 [3ea757fa-c80f-471f-909c-21d6b35ccd94]
2015-07-21 11:20:10.963951 [INFO] mod_dialplan_xml.c:635 Processing 1001
<1001>->1019 in context public
2015-07-21 11:20:10.963951 [NOTICE] switch_ivr.c:1863 Transfer
sofia/internal/1001 at 192.168.1.100 to XML[1019 at default]
2015-07-21 11:20:10.963951 [INFO] mod_dialplan_xml.c:635 Processing 1001
<1001>->1019 in context default
2015-07-21 11:20:10.983953 [NOTICE] switch_channel.c:1075 New Channel
sofia/internal/1019 at 192.168.1.100:5061
[2bb31faa-8cc2-400a-8997-3d5126acf148]
send 1189 bytes to udp/[192.168.1.100]:5061 at 11:20:10.990948:

------------------------------------------------------------------------
   INVITE sip:1019 at 192.168.1.100:5061 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bK8829733UK6ttr
   Max-Forwards: 68
   From: "1001" <sip:1001 at 192.168.1.100>;tag=pa6Dj3e8vN08a
   To: <sip:1019 at 192.168.1.100:5061>
   Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3
   CSeq: 78411741 INVITE
   Contact: <sip:mod_sofia at 192.168.1.100:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.4.20
+git~20150703T164215Z~b95362f965~64bit
   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: 249
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "1001"
<sip:1001 at 192.168.1.100>;party=calling;screen=yes;privacy=off
   
   v=0
   o=FreeSWITCH 1437471870 1437471871 IN IP4 192.168.1.100
   s=FreeSWITCH
   c=IN IP4 192.168.1.100
   t=0 0
   m=audio 23740 RTP/AVP 0 8 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20

------------------------------------------------------------------------
recv 325 bytes from udp/[192.168.1.100]:5061 at 11:20:10.994691:

------------------------------------------------------------------------
   SIP/2.0 180 Ringing
   Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bK8829733UK6ttr
   From: "1001" <sip:1001 at 192.168.1.100>;tag=pa6Dj3e8vN08a
   To: <sip:1019 at 192.168.1.100:5061>;tag=32791
   Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3
   CSeq: 78411741 INVITE
   Contact: 1019 <sip:1019 at 192.168.1.100:5061>
   Content-Length: 0
   

------------------------------------------------------------------------
2015-07-21 11:20:10.983953 [NOTICE] sofia.c:6729 Ring-Ready
sofia/internal/1019 at 192.168.1.100:5061!
2015-07-21 11:20:11.003950 [INFO] switch_ivr_originate.c:1193 Sending
early media
2015-07-21 11:20:11.003950 [NOTICE] sofia_media.c:92 Pre-Answer
sofia/internal/1001 at 192.168.1.100!
send 1069 bytes to udp/[192.168.1.100]:5062 at 11:20:11.015428:

------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0
   From: 1001 <sip:1001 at 192.168.1.100>;tag=32811
   To: 1019 <sip:1019 at 192.168.1.100>;tag=N1cNg8X4ZcapF
   Call-ID: 1-3281 at 192.168.1.100
   CSeq: 1 INVITE
   Contact: <sip:1019 at 192.168.1.100:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.4.20
+git~20150703T164215Z~b95362f965~64bit
   Accept: application/sdp
   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: 188
   Remote-Party-ID: "1019"
<sip:1019 at 192.168.1.100>;party=calling;privacy=off;screen=no
   
   v=0
   o=FreeSWITCH 1437471111 1437471112 IN IP4 192.168.1.100
   s=FreeSWITCH
   c=IN IP4 192.168.1.100
   t=0 0
   m=audio 24500 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   a=ptime:20
   m=video 0 RTP/AVP 19

------------------------------------------------------------------------
recv 593 bytes from udp/[192.168.1.100]:5061 at 11:20:11.995950:

------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bK8829733UK6ttr
   From: "1001" <sip:1001 at 192.168.1.100>;tag=pa6Dj3e8vN08a
   To: <sip:1019 at 192.168.1.100:5061>;tag=32791
   Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3
   CSeq: 78411741 INVITE
   Contact: 1019 <sip:1019 at 192.168.1.100:5061>
   Content-Type: application/sdp
   Content-Length:   238
   
   v=0
   o=5001 53655765 2353687637 IN IP4 192.168.1.100
   s=Talk
   c=IN IP4 192.168.1.100
   t=0 0
   m=audio 6000 RTP/AVP 0 8 97
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:97 iLBC/8000
   m=video 6000 RTP/AVP 31
   a=rtpmap:31 H261/90000

------------------------------------------------------------------------
send 360 bytes to udp/[192.168.1.100]:5061 at 11:20:11.998449:

------------------------------------------------------------------------
   ACK sip:1019 at 192.168.1.100:5061 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bK9Hv29ymZgFHDm
   Max-Forwards: 70
   From: "1001" <sip:1001 at 192.168.1.100>;tag=pa6Dj3e8vN08a
   To: <sip:1019 at 192.168.1.100:5061>;tag=32791
   Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3
   CSeq: 78411741 ACK
   Contact: <sip:mod_sofia at 192.168.1.100:5060>
   Content-Length: 0
   

------------------------------------------------------------------------
2015-07-21 11:20:11.983975 [NOTICE] sofia.c:7488 Channel
[sofia/internal/1019 at 192.168.1.100:5061] has been answered
send 1039 bytes to udp/[192.168.1.100]:5062 at 11:20:12.013969:

------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0
   From: 1001 <sip:1001 at 192.168.1.100>;tag=32811
   To: 1019 <sip:1019 at 192.168.1.100>;tag=N1cNg8X4ZcapF
   Call-ID: 1-3281 at 192.168.1.100
   CSeq: 1 INVITE
   Contact: <sip:1019 at 192.168.1.100:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.4.20
+git~20150703T164215Z~b95362f965~64bit
   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: 188
   Remote-Party-ID: "Outbound Call"
<sip:1019 at 192.168.1.100>;party=calling;privacy=off;screen=no
   
   v=0
   o=FreeSWITCH 1437471111 1437471112 IN IP4 192.168.1.100
   s=FreeSWITCH
   c=IN IP4 192.168.1.100
   t=0 0
   m=audio 24500 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   a=ptime:20
   m=video 0 RTP/AVP 19
2015-07-21 11:20:11.983975 [NOTICE] switch_ivr_originate.c:3523 Channel
[sofia/internal/1001 at 192.168.1.100] has been answered

------------------------------------------------------------------------
recv 371 bytes from udp/[192.168.1.100]:5062 at 11:20:12.018300:

------------------------------------------------------------------------
   ACK sip:1019 at 192.168.1.100:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0
   From: 1001 <sip:1001 at 192.168.1.100>;tag=32811
   To: 1019 <sip:1019 at 192.168.1.100>;tag=N1cNg8X4ZcapF
   Call-ID: 1-3281 at 192.168.1.100
   CSeq: 1 ACK
   Contact: 1001 <sip:1001 at 192.168.1.100:5062>
   Max-Forwards: 70
   Subject: Performance Test
   Content-Length: 0
   

------------------------------------------------------------------------
recv 371 bytes from udp/[192.168.1.100]:5062 at 11:20:17.020325:

------------------------------------------------------------------------
   BYE sip:1019 at 192.168.1.100:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0
   From: 1001 <sip:1001 at 192.168.1.100>;tag=32811
   To: 1019 <sip:1019 at 192.168.1.100>;tag=N1cNg8X4ZcapF
   Call-ID: 1-3281 at 192.168.1.100
   CSeq: 2 BYE
   Contact: 1001 <sip:1001 at 192.168.1.100:5062>
   Max-Forwards: 70
   Subject: Performance Test
   Content-Length: 0
   

------------------------------------------------------------------------
2015-07-21 11:20:17.023972 [NOTICE] sofia.c:952 Hangup
sofia/internal/1001 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]
send 465 bytes to udp/[192.168.1.100]:5062 at 11:20:17.035053:

------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0
   From: 1001 <sip:1001 at 192.168.1.100>;tag=32811
   To: 1019 <sip:1019 at 192.168.1.100>;tag=N1cNg8X4ZcapF
   Call-ID: 1-3281 at 192.168.1.100
   CSeq: 2 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.4.20
+git~20150703T164215Z~b95362f965~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Content-Length: 0
   

------------------------------------------------------------------------
2015-07-21 11:20:17.023972 [NOTICE] switch_ivr_bridge.c:758 Hangup
sofia/internal/1019 at 192.168.1.100:5061 [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]
send 585 bytes to udp/[192.168.1.100]:5061 at 11:20:17.042705:

------------------------------------------------------------------------
   BYE sip:1019 at 192.168.1.100:5061 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bKaUNUBt52Dr7ZF
   Max-Forwards: 70
   From: "1001" <sip:1001 at 192.168.1.100>;tag=pa6Dj3e8vN08a
   To: <sip:1019 at 192.168.1.100:5061>;tag=32791
   Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3
   CSeq: 78411742 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.4.20
+git~20150703T164215Z~b95362f965~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0
   

------------------------------------------------------------------------
2015-07-21 11:20:17.023972 [NOTICE] switch_core_session.c:1641 Session
21 (sofia/internal/1001 at 192.168.1.100) Ended
2015-07-21 11:20:17.023972 [NOTICE] switch_core_session.c:1645 Close
Channel sofia/internal/1001 at 192.168.1.100 [CS_DESTROY]
2015-07-21 11:20:17.023972 [NOTICE] switch_core_session.c:1641 Session
22 (sofia/internal/1019 at 192.168.1.100:5061) Ended
2015-07-21 11:20:17.023972 [NOTICE] switch_core_session.c:1645 Close
Channel sofia/internal/1019 at 192.168.1.100:5061 [CS_DESTROY]
recv 317 bytes from udp/[192.168.1.100]:5061 at 11:20:17.051862:

------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bKaUNUBt52Dr7ZF
   From: "1001" <sip:1001 at 192.168.1.100>;tag=pa6Dj3e8vN08a
   To: <sip:1019 at 192.168.1.100:5061>;tag=32791
   Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3
   CSeq: 78411742 BYE
   Contact: 1019 <sip:1019 at 192.168.1.100:5061>
   Content-Length: 0

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

Call-ID:    	1-2629 at 192.168.1.100
User:       	1001 at 192.168.1.100
Contact:    	"" <sip:1001 at 192.168.1.100:5061>
Agent:      	SIPp
Status:     	Registered(UDP)(unknown) EXP(2015-07-21 12:25:01)
EXPSECS(3112)
Ping-Status:	Reachable
Host:       	everest
IP:         	192.168.1.100
Port:       	5061
Auth-User:  	unknown
Auth-Realm: 	192.168.1.100
MWI-Account:	1001 at 192.168.1.100

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

Call-ID:    	1-3203 at 192.168.1.100
User:       	1020 at 192.168.1.100
Contact:    	"" <sip:1020 at 192.168.1.100:5061>
Agent:      	SIPp
Status:     	Registered(UDP)(unknown) EXP(2015-07-21 13:11:07)
EXPSECS(5878)
Ping-Status:	Reachable
Host:       	everest
IP:         	192.168.1.100
Port:       	5061
Auth-User:  	unknown
Auth-Realm: 	192.168.1.100
MWI-Account:	1020 at 192.168.1.100

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

Call-ID:    	1-3216 at 192.168.1.100
User:       	1019 at 192.168.1.100
Contact:    	"" <sip:1019 at 192.168.1.100:5061>
Agent:      	SIPp
Status:     	Registered(UDP)(unknown) EXP(2015-07-21 13:11:51)
EXPSECS(5922)
Ping-Status:	Reachable
Host:       	everest
IP:         	192.168.1.100
Port:       	5061
Auth-User:  	unknown
Auth-Realm: 	192.168.1.100
MWI-Account:	1019 at 192.168.1.100







From yadenis at seznam.cz  Tue Jul 21 22:00:45 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Tue, 21 Jul 2015 20:00:45 +0200
Subject: [Freeswitch-users] VideoCall with h264
In-Reply-To: <CAEwTsAhB7GkoPpOEcyj7jgdardDRoA9ZaakKAWkcvHU7ixqLBQ@mail.gmail.com>
References: <1437422047.3145.1.camel@twc.com>
	<917907795.20150721132224@seznam.cz>
	<CAEwTsAhB7GkoPpOEcyj7jgdardDRoA9ZaakKAWkcvHU7ixqLBQ@mail.gmail.com>
Message-ID: <781319F2-9C91-478F-A612-A3A79EBECB7A@seznam.cz>

Hi.

It is perfectly. Working system I just installed on Debian. 

How it works? I tried to add in my DialPlan
<action application="export" data="nolocal:absolute_codec_string=OPUS,H264?/> 
instead 
<action application="export" data="nolocal:absolute_codec_string=OPUS,VP8?/>

And I do not even get to connect. (incompatible destination)
Although the settings vp8 everything works like clockwork. On both sides Chrome. Module mod_vpx, mod_h26x is loaded.

What am I doing wrong?




-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

> On 21. 7. 2015, at 17:57, Michael Jerris <mike at jerris.com> wrote:
> 
> This currently works fine on Debian 8 Jessie.  To my knowledge other distributions do not include all the required dependencies so it should be possible but would require you to handle getting all of the dependencies sorted out yourself.  We will not be putting those additional dependencies for pattented algorithms in our repositories.
> 
> On Tuesday, July 21, 2015, Denis Jakovlev <yadenis at seznam.cz <mailto:yadenis at seznam.cz>> wrote:
> Hi All,
> 
> The question in the following. Is it possible to make calls through freeswitch using h264 instead of using vp8?
> 
> 
> -- 
> S pozdravem,
> Ing.Denis Jakovlev                           
> mob.tel <http://mob.tel/>. 775-415-382
>  <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

-------------- next part --------------
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From anthony.minessale at gmail.com  Wed Jul 22 00:11:30 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 21 Jul 2015 15:11:30 -0500
Subject: [Freeswitch-users] VideoCall with h264
In-Reply-To: <781319F2-9C91-478F-A612-A3A79EBECB7A@seznam.cz>
References: <1437422047.3145.1.camel@twc.com>
	<917907795.20150721132224@seznam.cz>
	<CAEwTsAhB7GkoPpOEcyj7jgdardDRoA9ZaakKAWkcvHU7ixqLBQ@mail.gmail.com>
	<781319F2-9C91-478F-A612-A3A79EBECB7A@seznam.cz>
Message-ID: <CAKbxfG_FwLXPX8zwgT2x33ZMAGxC0LrmtJAqBp5YFOJ6d00X+Q@mail.gmail.com>

build and use mod_av to install the codec and unload mod_h26x  mod_h26x is
only passthrough.


On Tue, Jul 21, 2015 at 1:00 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:

> Hi.
>
> It is perfectly. Working system I just installed on Debian.
>
> How it works? I tried to add in my DialPlan
> <action application="export"
> data="nolocal:absolute_codec_string=OPUS,H264?/>
> instead
> <action application="export"
> data="nolocal:absolute_codec_string=OPUS,VP8?/>
>
> And I do not even get to connect. (incompatible destination)
> Although the settings vp8 everything works like clockwork. On both sides
> Chrome. Module mod_vpx, mod_h26x is loaded.
>
> What am I doing wrong?
>
>
>
>
> *-- *
> *S pozdravem,*
> *Ing.Denis Jakovlev                           *
> *mob.tel <http://mob.tel>. 775-415-382*
>
> On 21. 7. 2015, at 17:57, Michael Jerris <mike at jerris.com> wrote:
>
> This currently works fine on Debian 8 Jessie.  To my knowledge other
> distributions do not include all the required dependencies so it should be
> possible but would require you to handle getting all of the dependencies
> sorted out yourself.  We will not be putting those additional dependencies
> for pattented algorithms in our repositories.
>
> On Tuesday, July 21, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
>
>>  Hi All,
>>
>> The question in the following. Is it possible to make calls through
>> freeswitch using h264 instead of using vp8?
>>
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>> <http://mob.tel/>. 775-415-382 * <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
>



-- 
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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From yadenis at seznam.cz  Wed Jul 22 00:28:08 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Tue, 21 Jul 2015 22:28:08 +0200
Subject: [Freeswitch-users] VideoCall with h264
In-Reply-To: <CAKbxfG_FwLXPX8zwgT2x33ZMAGxC0LrmtJAqBp5YFOJ6d00X+Q@mail.gmail.com>
References: <1437422047.3145.1.camel@twc.com>
	<917907795.20150721132224@seznam.cz>
	<CAEwTsAhB7GkoPpOEcyj7jgdardDRoA9ZaakKAWkcvHU7ixqLBQ@mail.gmail.com>
	<781319F2-9C91-478F-A612-A3A79EBECB7A@seznam.cz>
	<CAKbxfG_FwLXPX8zwgT2x33ZMAGxC0LrmtJAqBp5YFOJ6d00X+Q@mail.gmail.com>
Message-ID: <E688F729-2CE9-4834-AA3F-8A555A5CA86E@seznam.cz>

Dobr? den.

I'll try. The problem is that I have a great video records only mod_vlc. With mod_av I did not write the video. But I will try of course. Mod_av I have compiled.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

> On 21. 7. 2015, at 22:11, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> 
> build and use mod_av to install the codec and unload mod_h26x  mod_h26x is only passthrough.
> 
> 
> On Tue, Jul 21, 2015 at 1:00 PM, Denis Jakovlev <yadenis at seznam.cz <mailto:yadenis at seznam.cz>> wrote:
> Hi.
> 
> It is perfectly. Working system I just installed on Debian. 
> 
> How it works? I tried to add in my DialPlan
> <action application="export" data="nolocal:absolute_codec_string=OPUS,H264?/> 
> instead 
> <action application="export" data="nolocal:absolute_codec_string=OPUS,VP8?/>
> 
> And I do not even get to connect. (incompatible destination)
> Although the settings vp8 everything works like clockwork. On both sides Chrome. Module mod_vpx, mod_h26x is loaded.
> 
> What am I doing wrong?
> 
> 
> 
> 
> -- 
> S pozdravem,
> Ing.Denis Jakovlev                           
> mob.tel <http://mob.tel/>. 775-415-382
> 
>> On 21. 7. 2015, at 17:57, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> 
>> This currently works fine on Debian 8 Jessie.  To my knowledge other distributions do not include all the required dependencies so it should be possible but would require you to handle getting all of the dependencies sorted out yourself.  We will not be putting those additional dependencies for pattented algorithms in our repositories.
>> 
>> On Tuesday, July 21, 2015, Denis Jakovlev <yadenis at seznam.cz <mailto:yadenis at seznam.cz>> wrote:
>> Hi All,
>> 
>> The question in the following. Is it possible to make calls through freeswitch using h264 instead of using vp8?
>> 
>> 
>> -- 
>> S pozdravem,
>> Ing.Denis Jakovlev                           
>> mob.tel <http://mob.tel/>. 775-415-382
>>  <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/>
> 
> 
> 
> -- 
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
> 
> ? http://freeswitch.org/ <http://freeswitch.org/>  ? http://cluecon.com/ <http://cluecon.com/>  ? http://twitter.com/FreeSWITCH <http://twitter.com/FreeSWITCH>
> ? irc.freenode.net <http://irc.freenode.net/> #freeswitch ? http://freeswitch.org/g+ <http://freeswitch.org/g+>
> 
> ClueCon Weekly Development Call 
> ? sip:888 at conference.freeswitch.org <mailto:sip%3A888 at conference.freeswitch.org>  ? +19193869900 
> 
> https://www.youtube.com/watch?v=9XXgW34t40s <https://www.youtube.com/watch?v=9XXgW34t40s>
> https://www.youtube.com/watch?v=NLaDpGQuZDA <https://www.youtube.com/watch?v=NLaDpGQuZDA>
> _________________________________________________________________________
> 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.org  Wed Jul 22 00:31:45 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 21 Jul 2015 15:31:45 -0500
Subject: [Freeswitch-users] Disable 401 and 407 for internal profile
In-Reply-To: <1437496570.3313.11.camel@twc.com>
References: <1437496570.3313.11.camel@twc.com>
Message-ID: <CAEJMVkB95-=2gBtA4eYV2=+O4gBHQBcW4HWKAzdghgxqM3zyhQ@mail.gmail.com>

There is no 1020 sip in the vanilla config its 1000-1019, and "has executed
the last dialplan instruction, hanging up" is a good clue about what is
going on, if you set the log level to debug you'll see it doing all the
compares in the dialplan and 1020 doesn't match.  The call ran out of stuff
to do.

On Tue, Jul 21, 2015 at 11:36 AM, Admin <smontour at twc.com> wrote:

> Thanks Brian. That seems to work partially and strangely too. All calls
> made to extensions above '1019' stopped working now. When a call is
> destined to any extension above '1019' comes back with '480 Temporarily
> Unavailable'. Below is a pasted copy of the SIP trace for a call from
> 1001 to 1019 (works), and 1001 to 1020 (doesn't work). Registration
> output is also pasted below.
>
> ==========================================================================
>
> recv 726 bytes from udp/[192.168.1.100]:5062 at 11:18:28.880410:
>
> ------------------------------------------------------------------------
>    INVITE sip:1020 at 192.168.1.100 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3263-1-0
>    From: 1001 <sip:1001 at 192.168.1.100>;tag=32631
>    To: 1020 <sip:1020 at 192.168.1.100>
>    Call-ID: 1-3263 at 192.168.1.100
>    CSeq: 1 INVITE
>    Contact: 1001 <sip:1001 at 192.168.1.100:5062>
>    Max-Forwards: 70
>    Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE,
> UPDATE, SUBSCRIBE
>    Supported: 100rel, replaces, from-change
>    Timestamp: 1
>    Content-Type: application/sdp
>    Content-Length:   238
>
>    v=0
>    o=1001 53655765 2353687637 IN IP4 192.168.1.100
>    s=Talk
>    c=IN IP4 192.168.1.100
>    t=0 0
>    m=audio 6001 RTP/AVP 0 8 97
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:97 iLBC/8000
>    m=video 6001 RTP/AVP 31
>    a=rtpmap:31 H261/90000
>
> ------------------------------------------------------------------------
> send 333 bytes to udp/[192.168.1.100]:5062 at 11:18:28.880719:
>
> ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3263-1-0
>    From: 1001 <sip:1001 at 192.168.1.100>;tag=32631
>    To: 1020 <sip:1020 at 192.168.1.100>
>    Call-ID: 1-3263 at 192.168.1.100
>    CSeq: 1 INVITE
>    Timestamp: 1 0.000152
>    User-Agent: FreeSWITCH-mod_sofia/1.4.20
> +git~20150703T164215Z~b95362f965~64bit
>    Content-Length: 0
>
> ------------------------------------------------------------------------
> 2015-07-21 11:18:28.863966 [NOTICE] switch_channel.c:1075 New Channel
> sofia/internal/1001 at 192.168.1.100 [d4084b55-8351-48fa-87ff-01d4b2355d4e]
> 2015-07-21 11:18:28.863966 [INFO] mod_dialplan_xml.c:635 Processing 1001
> <1001>->1020 in context public
> 2015-07-21 11:18:28.863966 [NOTICE] switch_core_state_machine.c:315
> sofia/internal/1001 at 192.168.1.100 has executed the last dialplan
> instruction, hanging up.
> 2015-07-21 11:18:28.863966 [NOTICE] switch_core_state_machine.c:317
> Hangup sofia/internal/1001 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]
> send 836 bytes to udp/[192.168.1.100]:5062 at 11:18:28.885154:
>
> ------------------------------------------------------------------------
>    SIP/2.0 480 Temporarily Unavailable
>    Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3263-1-0
>    Max-Forwards: 70
>    From: 1001 <sip:1001 at 192.168.1.100>;tag=32631
>    To: 1020 <sip:1020 at 192.168.1.100>;tag=mrKveDD123K3K
>    Call-ID: 1-3263 at 192.168.1.100
>    CSeq: 1 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.4.20
> +git~20150703T164215Z~b95362f965~64bit
>    Accept: application/sdp
>    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
>    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>    Content-Length: 0
>    Remote-Party-ID: "1020"
> <sip:1020 at 192.168.1.100>;party=calling;privacy=off;screen=no
>
> ------------------------------------------------------------------------
> 2015-07-21 11:18:28.883950 [NOTICE] switch_core_session.c:1641 Session
> 20 (sofia/internal/1001 at 192.168.1.100) Ended
> 2015-07-21 11:18:28.883950 [NOTICE] switch_core_session.c:1645 Close
> Channel sofia/internal/1001 at 192.168.1.100 [CS_DESTROY]
> recv 361 bytes from udp/[192.168.1.100]:5062 at 11:18:28.887876:
>
> ------------------------------------------------------------------------
>    ACK sip:1020 at 192.168.1.100 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3263-1-0
>    From: 1001 <sip:1001 at 192.168.1.100>;tag=32631
>    To: 1020 <sip:1020 at 192.168.1.100>;tag=mrKveDD123K3K
>    Call-ID: 1-3263 at 192.168.1.100
>    CSeq: 1 ACK
>    Contact: <sip:sipp at 192.168.1.100:5062;transport=UDP>
>    Max-Forwards: 70
>    Subject: Performance Test
>    Content-Length: 0
>
>
> ------------------------------------------------------------------------
>
> >>>>>>>>>>> 1001 - 1019 - Working >>>>>>>>>>>>>>>>
>
> recv 726 bytes from udp/[192.168.1.100]:5062 at 11:20:10.977891:
>
> ------------------------------------------------------------------------
>    INVITE sip:1019 at 192.168.1.100 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0
>    From: 1001 <sip:1001 at 192.168.1.100>;tag=32811
>    To: 1019 <sip:1019 at 192.168.1.100>
>    Call-ID: 1-3281 at 192.168.1.100
>    CSeq: 1 INVITE
>    Contact: 1001 <sip:1001 at 192.168.1.100:5062>
>    Max-Forwards: 70
>    Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE,
> UPDATE, SUBSCRIBE
>    Supported: 100rel, replaces, from-change
>    Timestamp: 1
>    Content-Type: application/sdp
>    Content-Length:   238
>
>    v=0
>    o=1001 53655765 2353687637 IN IP4 192.168.1.100
>    s=Talk
>    c=IN IP4 192.168.1.100
>    t=0 0
>    m=audio 6001 RTP/AVP 0 8 97
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:97 iLBC/8000
>    m=video 6001 RTP/AVP 31
>    a=rtpmap:31 H261/90000
>
> ------------------------------------------------------------------------
> send 333 bytes to udp/[192.168.1.100]:5062 at 11:20:10.978324:
>
> ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0
>    From: 1001 <sip:1001 at 192.168.1.100>;tag=32811
>    To: 1019 <sip:1019 at 192.168.1.100>
>    Call-ID: 1-3281 at 192.168.1.100
>    CSeq: 1 INVITE
>    Timestamp: 1 0.000208
>    User-Agent: FreeSWITCH-mod_sofia/1.4.20
> +git~20150703T164215Z~b95362f965~64bit
>    Content-Length: 0
>
>
> ------------------------------------------------------------------------
> 2015-07-21 11:20:10.963951 [NOTICE] switch_channel.c:1075 New Channel
> sofia/internal/1001 at 192.168.1.100 [3ea757fa-c80f-471f-909c-21d6b35ccd94]
> 2015-07-21 11:20:10.963951 [INFO] mod_dialplan_xml.c:635 Processing 1001
> <1001>->1019 in context public
> 2015-07-21 11:20:10.963951 [NOTICE] switch_ivr.c:1863 Transfer
> sofia/internal/1001 at 192.168.1.100 to XML[1019 at default]
> 2015-07-21 11:20:10.963951 [INFO] mod_dialplan_xml.c:635 Processing 1001
> <1001>->1019 in context default
> 2015-07-21 11:20:10.983953 [NOTICE] switch_channel.c:1075 New Channel
> sofia/internal/1019 at 192.168.1.100:5061
> [2bb31faa-8cc2-400a-8997-3d5126acf148]
> send 1189 bytes to udp/[192.168.1.100]:5061 at 11:20:10.990948:
>
> ------------------------------------------------------------------------
>    INVITE sip:1019 at 192.168.1.100:5061 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bK8829733UK6ttr
>    Max-Forwards: 68
>    From: "1001" <sip:1001 at 192.168.1.100>;tag=pa6Dj3e8vN08a
>    To: <sip:1019 at 192.168.1.100:5061>
>    Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3
>    CSeq: 78411741 INVITE
>    Contact: <sip:mod_sofia at 192.168.1.100:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.4.20
> +git~20150703T164215Z~b95362f965~64bit
>    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: 249
>    X-FS-Support: update_display,send_info
>    Remote-Party-ID: "1001"
> <sip:1001 at 192.168.1.100>;party=calling;screen=yes;privacy=off
>
>    v=0
>    o=FreeSWITCH 1437471870 1437471871 IN IP4 192.168.1.100
>    s=FreeSWITCH
>    c=IN IP4 192.168.1.100
>    t=0 0
>    m=audio 23740 RTP/AVP 0 8 101 13
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>
> ------------------------------------------------------------------------
> recv 325 bytes from udp/[192.168.1.100]:5061 at 11:20:10.994691:
>
> ------------------------------------------------------------------------
>    SIP/2.0 180 Ringing
>    Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bK8829733UK6ttr
>    From: "1001" <sip:1001 at 192.168.1.100>;tag=pa6Dj3e8vN08a
>    To: <sip:1019 at 192.168.1.100:5061>;tag=32791
>    Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3
>    CSeq: 78411741 INVITE
>    Contact: 1019 <sip:1019 at 192.168.1.100:5061>
>    Content-Length: 0
>
>
> ------------------------------------------------------------------------
> 2015-07-21 11:20:10.983953 [NOTICE] sofia.c:6729 Ring-Ready
> sofia/internal/1019 at 192.168.1.100:5061!
> 2015-07-21 11:20:11.003950 [INFO] switch_ivr_originate.c:1193 Sending
> early media
> 2015-07-21 11:20:11.003950 [NOTICE] sofia_media.c:92 Pre-Answer
> sofia/internal/1001 at 192.168.1.100!
> send 1069 bytes to udp/[192.168.1.100]:5062 at 11:20:11.015428:
>
> ------------------------------------------------------------------------
>    SIP/2.0 183 Session Progress
>    Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0
>    From: 1001 <sip:1001 at 192.168.1.100>;tag=32811
>    To: 1019 <sip:1019 at 192.168.1.100>;tag=N1cNg8X4ZcapF
>    Call-ID: 1-3281 at 192.168.1.100
>    CSeq: 1 INVITE
>    Contact: <sip:1019 at 192.168.1.100:5060;transport=udp>
>    User-Agent: FreeSWITCH-mod_sofia/1.4.20
> +git~20150703T164215Z~b95362f965~64bit
>    Accept: application/sdp
>    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: 188
>    Remote-Party-ID: "1019"
> <sip:1019 at 192.168.1.100>;party=calling;privacy=off;screen=no
>
>    v=0
>    o=FreeSWITCH 1437471111 1437471112 IN IP4 192.168.1.100
>    s=FreeSWITCH
>    c=IN IP4 192.168.1.100
>    t=0 0
>    m=audio 24500 RTP/AVP 0
>    a=rtpmap:0 PCMU/8000
>    a=ptime:20
>    m=video 0 RTP/AVP 19
>
> ------------------------------------------------------------------------
> recv 593 bytes from udp/[192.168.1.100]:5061 at 11:20:11.995950:
>
> ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bK8829733UK6ttr
>    From: "1001" <sip:1001 at 192.168.1.100>;tag=pa6Dj3e8vN08a
>    To: <sip:1019 at 192.168.1.100:5061>;tag=32791
>    Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3
>    CSeq: 78411741 INVITE
>    Contact: 1019 <sip:1019 at 192.168.1.100:5061>
>    Content-Type: application/sdp
>    Content-Length:   238
>
>    v=0
>    o=5001 53655765 2353687637 IN IP4 192.168.1.100
>    s=Talk
>    c=IN IP4 192.168.1.100
>    t=0 0
>    m=audio 6000 RTP/AVP 0 8 97
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:97 iLBC/8000
>    m=video 6000 RTP/AVP 31
>    a=rtpmap:31 H261/90000
>
> ------------------------------------------------------------------------
> send 360 bytes to udp/[192.168.1.100]:5061 at 11:20:11.998449:
>
> ------------------------------------------------------------------------
>    ACK sip:1019 at 192.168.1.100:5061 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bK9Hv29ymZgFHDm
>    Max-Forwards: 70
>    From: "1001" <sip:1001 at 192.168.1.100>;tag=pa6Dj3e8vN08a
>    To: <sip:1019 at 192.168.1.100:5061>;tag=32791
>    Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3
>    CSeq: 78411741 ACK
>    Contact: <sip:mod_sofia at 192.168.1.100:5060>
>    Content-Length: 0
>
>
> ------------------------------------------------------------------------
> 2015-07-21 11:20:11.983975 [NOTICE] sofia.c:7488 Channel
> [sofia/internal/1019 at 192.168.1.100:5061] has been answered
> send 1039 bytes to udp/[192.168.1.100]:5062 at 11:20:12.013969:
>
> ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0
>    From: 1001 <sip:1001 at 192.168.1.100>;tag=32811
>    To: 1019 <sip:1019 at 192.168.1.100>;tag=N1cNg8X4ZcapF
>    Call-ID: 1-3281 at 192.168.1.100
>    CSeq: 1 INVITE
>    Contact: <sip:1019 at 192.168.1.100:5060;transport=udp>
>    User-Agent: FreeSWITCH-mod_sofia/1.4.20
> +git~20150703T164215Z~b95362f965~64bit
>    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: 188
>    Remote-Party-ID: "Outbound Call"
> <sip:1019 at 192.168.1.100>;party=calling;privacy=off;screen=no
>
>    v=0
>    o=FreeSWITCH 1437471111 1437471112 IN IP4 192.168.1.100
>    s=FreeSWITCH
>    c=IN IP4 192.168.1.100
>    t=0 0
>    m=audio 24500 RTP/AVP 0
>    a=rtpmap:0 PCMU/8000
>    a=ptime:20
>    m=video 0 RTP/AVP 19
> 2015-07-21 11:20:11.983975 [NOTICE] switch_ivr_originate.c:3523 Channel
> [sofia/internal/1001 at 192.168.1.100] has been answered
>
> ------------------------------------------------------------------------
> recv 371 bytes from udp/[192.168.1.100]:5062 at 11:20:12.018300:
>
> ------------------------------------------------------------------------
>    ACK sip:1019 at 192.168.1.100:5060;transport=udp SIP/2.0
>    Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0
>    From: 1001 <sip:1001 at 192.168.1.100>;tag=32811
>    To: 1019 <sip:1019 at 192.168.1.100>;tag=N1cNg8X4ZcapF
>    Call-ID: 1-3281 at 192.168.1.100
>    CSeq: 1 ACK
>    Contact: 1001 <sip:1001 at 192.168.1.100:5062>
>    Max-Forwards: 70
>    Subject: Performance Test
>    Content-Length: 0
>
>
> ------------------------------------------------------------------------
> recv 371 bytes from udp/[192.168.1.100]:5062 at 11:20:17.020325:
>
> ------------------------------------------------------------------------
>    BYE sip:1019 at 192.168.1.100:5060;transport=udp SIP/2.0
>    Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0
>    From: 1001 <sip:1001 at 192.168.1.100>;tag=32811
>    To: 1019 <sip:1019 at 192.168.1.100>;tag=N1cNg8X4ZcapF
>    Call-ID: 1-3281 at 192.168.1.100
>    CSeq: 2 BYE
>    Contact: 1001 <sip:1001 at 192.168.1.100:5062>
>    Max-Forwards: 70
>    Subject: Performance Test
>    Content-Length: 0
>
>
> ------------------------------------------------------------------------
> 2015-07-21 11:20:17.023972 [NOTICE] sofia.c:952 Hangup
> sofia/internal/1001 at 192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING]
> send 465 bytes to udp/[192.168.1.100]:5062 at 11:20:17.035053:
>
> ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0
>    From: 1001 <sip:1001 at 192.168.1.100>;tag=32811
>    To: 1019 <sip:1019 at 192.168.1.100>;tag=N1cNg8X4ZcapF
>    Call-ID: 1-3281 at 192.168.1.100
>    CSeq: 2 BYE
>    User-Agent: FreeSWITCH-mod_sofia/1.4.20
> +git~20150703T164215Z~b95362f965~64bit
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Content-Length: 0
>
>
> ------------------------------------------------------------------------
> 2015-07-21 11:20:17.023972 [NOTICE] switch_ivr_bridge.c:758 Hangup
> sofia/internal/1019 at 192.168.1.100:5061 [CS_EXCHANGE_MEDIA]
> [NORMAL_CLEARING]
> send 585 bytes to udp/[192.168.1.100]:5061 at 11:20:17.042705:
>
> ------------------------------------------------------------------------
>    BYE sip:1019 at 192.168.1.100:5061 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bKaUNUBt52Dr7ZF
>    Max-Forwards: 70
>    From: "1001" <sip:1001 at 192.168.1.100>;tag=pa6Dj3e8vN08a
>    To: <sip:1019 at 192.168.1.100:5061>;tag=32791
>    Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3
>    CSeq: 78411742 BYE
>    User-Agent: FreeSWITCH-mod_sofia/1.4.20
> +git~20150703T164215Z~b95362f965~64bit
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>    Content-Length: 0
>
>
> ------------------------------------------------------------------------
> 2015-07-21 11:20:17.023972 [NOTICE] switch_core_session.c:1641 Session
> 21 (sofia/internal/1001 at 192.168.1.100) Ended
> 2015-07-21 11:20:17.023972 [NOTICE] switch_core_session.c:1645 Close
> Channel sofia/internal/1001 at 192.168.1.100 [CS_DESTROY]
> 2015-07-21 11:20:17.023972 [NOTICE] switch_core_session.c:1641 Session
> 22 (sofia/internal/1019 at 192.168.1.100:5061) Ended
> 2015-07-21 11:20:17.023972 [NOTICE] switch_core_session.c:1645 Close
> Channel sofia/internal/1019 at 192.168.1.100:5061 [CS_DESTROY]
> recv 317 bytes from udp/[192.168.1.100]:5061 at 11:20:17.051862:
>
> ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bKaUNUBt52Dr7ZF
>    From: "1001" <sip:1001 at 192.168.1.100>;tag=pa6Dj3e8vN08a
>    To: <sip:1019 at 192.168.1.100:5061>;tag=32791
>    Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3
>    CSeq: 78411742 BYE
>    Contact: 1019 <sip:1019 at 192.168.1.100:5061>
>    Content-Length: 0
>
> ==================================================================
>
> Call-ID:        1-2629 at 192.168.1.100
> User:           1001 at 192.168.1.100
> Contact:        "" <sip:1001 at 192.168.1.100:5061>
> Agent:          SIPp
> Status:         Registered(UDP)(unknown) EXP(2015-07-21 12:25:01)
> EXPSECS(3112)
> Ping-Status:    Reachable
> Host:           everest
> IP:             192.168.1.100
> Port:           5061
> Auth-User:      unknown
> Auth-Realm:     192.168.1.100
> MWI-Account:    1001 at 192.168.1.100
>
> --------------------------------------------------------
>
> Call-ID:        1-3203 at 192.168.1.100
> User:           1020 at 192.168.1.100
> Contact:        "" <sip:1020 at 192.168.1.100:5061>
> Agent:          SIPp
> Status:         Registered(UDP)(unknown) EXP(2015-07-21 13:11:07)
> EXPSECS(5878)
> Ping-Status:    Reachable
> Host:           everest
> IP:             192.168.1.100
> Port:           5061
> Auth-User:      unknown
> Auth-Realm:     192.168.1.100
> MWI-Account:    1020 at 192.168.1.100
>
> --------------------------------------------------------
>
> Call-ID:        1-3216 at 192.168.1.100
> User:           1019 at 192.168.1.100
> Contact:        "" <sip:1019 at 192.168.1.100:5061>
> Agent:          SIPp
> Status:         Registered(UDP)(unknown) EXP(2015-07-21 13:11:51)
> EXPSECS(5922)
> Ping-Status:    Reachable
> Host:           everest
> IP:             192.168.1.100
> Port:           5061
> Auth-User:      unknown
> Auth-Realm:     192.168.1.100
> MWI-Account:    1019 at 192.168.1.100
>
>
>
>
>
>
> _________________________________________________________________________
> 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
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From brian at freeswitch.org  Wed Jul 22 00:34:29 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 21 Jul 2015 15:34:29 -0500
Subject: [Freeswitch-users] Enterprise originate not honouring fs_path
 in 302 redirect
In-Reply-To: <55AE4F5D.7030307@gmail.com>
References: <55AE4F5D.7030307@gmail.com>
Message-ID: <CAEJMVkAqT1kD9Xptg=X4nO78x6ckf=qfxvo_XthBy=w09tJydA@mail.gmail.com>

Its not designed to do that, and we never will.  Its an exploitable attack
vector if we did.

On Tue, Jul 21, 2015 at 8:55 AM, Asgaroth <00asgaroth00 at gmail.com> wrote:

> Hi All,
>
> First of all, I'm a relatively new user of FreeSWITCH so, please bear
> with me if I'm asking a silly question.
>
> I have users registering through multiple kamailio
> loadbalancer/registrars, so when we build a bridge command to multiple
> local users we send the initial invite(s) to the kamailio lookup server
> which then responds with a 302 redirect message specifying the contact
> details of the user.
>
> What I am trying to do is get FreeSWITCH to call a particular user via a
> particular loadbalancer based on information recieved in the 302
> redirect. From what I have read in the archives, we can achieve this
> with the fs_path directive on the bridge command. However, how/where do
> I specify the fs_path in the contact header in the 302 redirect in such
> a manner that FreeSWITCH will honour the fs_path directive.
>
> I originally tried setting a fs_path parameter in the contact of the 302
> redirect message, something like the following:
>
> Contact:
> <sip:subscriber at 1.2.3.4:5678
> ;rinstance=9af71926212a25cf;transport=UDP;received=sip:5.6.7.8:9012
> ;fs_path=sip:1.2.3.5>
>
> However, FreeSWITCH ignores the fs_path directive here and attempts to
> send this call directly to 1.2.3.4, instead of sending it via 1.2.3.5.
>
> Is this at all possible using the enterprise originate (:_: seperator)
> feature of the bridge command?
>
> We're attempting this with FreeSWITCH v 1.4.20 ("official" rpm packages
> from the repo)
>
> Any info/tips/suggestions/beatings much appreciate :)
>
> Thanks
>
> _________________________________________________________________________
> 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
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From brian at freeswitch.org  Wed Jul 22 00:59:43 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 21 Jul 2015 15:59:43 -0500
Subject: [Freeswitch-users] ClueCon Coder Games Monday Aug. 3rd 2015
Message-ID: <CAEJMVkC3dOe1520Yw9dQsfY0zugEm1Y7iedtaETKkAQo=dHs7Q@mail.gmail.com>

[image: Description: HDD:Users:anthm:Downloads:ccxx.jpg]
 *August 3rd 2015*

August 3rd ? August 6th 2015

877-742-2583  ?  marketing at cluecon.com

Register NOW!

?   $1299 Staying at the Hotel

?   $1599 Staying Elsewhere


*Contact Us*

https://cluecon.com

marketing at cluecon.com








*So You Think You Can Code?*


You?ve seen the presentations, you?ve asked your questions, you have the
resources, now it is your time to shine by using the sponsor APIs to create
something exciting! We want to see what you can do! Bonus points for each
API you can incorporate! Go check out the APIs now to get a head start on
the competition and get those creative juices flowing! You have less than
two weeks to prepare!


Sponsor APIs: FreeSWITCH <http://www.freeswitch.org> - Corvisa
<https://www.corvisa.com> - Tropo <https://www.tropo.com> - Kandy
<https://www.kandy.io> - Twilio <https://www.twilio.com> - Plivo
<https://www.plivo.com> - and more...
IPv6 Round Table
IPv6 and why you should deploy it ASAP: John Brzozowski, Fellow and Chief
Architect, IPv6 at Comcast, Bill Sandiford President of CNOC, Member of the
board at ARIN.
Flowroute - Jeopardy
Think you know about SIP? Do you know enough to beat the competition?
Flowroute is hosting a SIP themed game of Jeopardy! Put your brain to the
test and come see how much you really know!
DTMF-u
Do you like all things games? Well, then this is the game for you! But
there is a twist! Before you can win the game you must build it! Using your
choice of language or API you must build a game! There will be three
categories of DTMF-u; Tic-tac-toe, DTMF pattern recognition, or Freestyle.
You could build something that plays a random DTMF sequence, receives
player input, and then either continues or fails the player. Or maybe a
WebRTC based game of tic-tac-toe? Or surprise us! Have fun with it!
Creative ways to fail a player may give you bonus points. The top three
games will be played by everyone and the winner of each will take home a
prize! All gaming bots will be screened via a Turing test to ensure no
unintended apocalyptic consequences.
Show and Tell
Alright, now is your chance! You have been playing with the code all day
and this is your chance to show off! We want to know what you've done and
how you did it. Use your creativity and skills as a programmer to impress
the judges and win a prize! This is a no holds barred all out free for all!
Any language doing anything! Knock our socks off and take home a fabulous
prize and a year?s supply of bragging rights!
Grand Prize

[image: Description: mac1]The grand prize is a laser engraved commemorative
FreeSWITCH 1.6 Edition dual-core 13" Retina MacBook Pro!  See the Important
Dates Section for Registration details!








 Why I Think You MUST COME To ClueCon!



[image: Description: kk]Hi, I?m Kathleen. I?m the FreeSWITCH and ClueCon
Social Media Correspondent.  I?ve been working hard all year keeping you
all up to date on what?s going on with FreeSWITCH. Today I?m here to let
you know more about the upcoming ClueCon 2015 Conference!



This year we are adding an optional day on Monday with an all-day
Hack-A-Thon with great coding contests, game shows and kick-off fun!



If you are interested in WebRTC, Voice over IP or Open Source projects like
FreeSWITCH, ClueCon is the greatest opportunity you have to gain exposure
to the most knowledge and technology in one place.  Also, it?s the most fun
you can possibly have while still getting a ton of work done!  I really
look forward to seeing you all there and enjoying the amazing talks, the
Epic Annual Kick-Off Pizza Party, The Gigabit Reception and so much more.
Make sure you register today so you can reserve your place among the
attendees!


Be sure to follow us on Facebook and Twitter to get my latest updates in
info!


[image: Inline image 1] <https://facebook.com/cluecon> [image: Inline image
2] <https://twitter.com/cluecon>



[image: Inline image 3] <https://www.youtube.com/watch?v=NLaDpGQuZDA>


 [image: Inline image 4] <https://www.youtube.com/watch?v=9XXgW34t40s>
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From mariogasparoni at gmail.com  Wed Jul 22 01:26:59 2015
From: mariogasparoni at gmail.com (Mario Gasparoni Junior)
Date: Tue, 21 Jul 2015 18:26:59 -0300
Subject: [Freeswitch-users] VideoCall with h264
In-Reply-To: <E688F729-2CE9-4834-AA3F-8A555A5CA86E@seznam.cz>
References: <1437422047.3145.1.camel@twc.com>
	<917907795.20150721132224@seznam.cz>
	<CAEwTsAhB7GkoPpOEcyj7jgdardDRoA9ZaakKAWkcvHU7ixqLBQ@mail.gmail.com>
	<781319F2-9C91-478F-A612-A3A79EBECB7A@seznam.cz>
	<CAKbxfG_FwLXPX8zwgT2x33ZMAGxC0LrmtJAqBp5YFOJ6d00X+Q@mail.gmail.com>
	<E688F729-2CE9-4834-AA3F-8A555A5CA86E@seznam.cz>
Message-ID: <CAKOUqqsPXEayPWsDQ1nkQvhAdd4yvrbx7rQN8qfXfcggYLb-aA@mail.gmail.com>

Did you try adding "H264" in "global_codec_prefs" and
"outbound_codec_prefs" in your vars.xml ?

Sent from mobile
Em 21/07/2015 17:29, "Denis Jakovlev" <yadenis at seznam.cz> escreveu:

> Dobr? den.
>
> I'll try. The problem is that I have a great video records only mod_vlc.
> With mod_av I did not write the video. But I will try of course. Mod_av I
> have compiled.
>
> *-- *
> *S pozdravem,*
> *Ing.Denis Jakovlev                           *
> *mob.tel <http://mob.tel>. 775-415-382*
>
> On 21. 7. 2015, at 22:11, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>
> build and use mod_av to install the codec and unload mod_h26x  mod_h26x is
> only passthrough.
>
>
> On Tue, Jul 21, 2015 at 1:00 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
>
>> Hi.
>>
>> It is perfectly. Working system I just installed on Debian.
>>
>> How it works? I tried to add in my DialPlan
>> <action application="export"
>> data="nolocal:absolute_codec_string=OPUS,H264?/>
>> instead
>> <action application="export"
>> data="nolocal:absolute_codec_string=OPUS,VP8?/>
>>
>> And I do not even get to connect. (incompatible destination)
>> Although the settings vp8 everything works like clockwork. On both sides
>> Chrome. Module mod_vpx, mod_h26x is loaded.
>>
>> What am I doing wrong?
>>
>>
>>
>>
>> *-- *
>> *S pozdravem,*
>> *Ing.Denis Jakovlev                           *
>> *mob.tel <http://mob.tel/>. 775-415-382*
>>
>> On 21. 7. 2015, at 17:57, Michael Jerris <mike at jerris.com> wrote:
>>
>> This currently works fine on Debian 8 Jessie.  To my knowledge other
>> distributions do not include all the required dependencies so it should be
>> possible but would require you to handle getting all of the dependencies
>> sorted out yourself.  We will not be putting those additional dependencies
>> for pattented algorithms in our repositories.
>>
>> On Tuesday, July 21, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
>>
>>>  Hi All,
>>>
>>> The question in the following. Is it possible to make calls through
>>> freeswitch using h264 instead of using vp8?
>>>
>>>
>>>
>>>
>>>
>>>
>>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>>> <http://mob.tel/>. 775-415-382 * <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
>>
>
>
>
> --
> 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
>  _________________________________________________________________________
> 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 krice at freeswitch.org  Wed Jul 22 04:23:04 2015
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 22 Jul 2015 00:23:04 +0000
Subject: [Freeswitch-users] Announcing the ClueCon Coder Games!
Message-ID: <55aee2685a93_3b2a761330978dd@resque-worker.19.mail>

New Post on freeswitch.org from Kathleen King
check it out at http://ift.tt/1CUbP1a
Announcing the ClueCon Coder Games!
So You Think You Can Code?

You?ve seen the presentations, you?ve asked your questions, you have the resources, now it is your time to shine by using the sponsor APIs to create something exciting! We want to see what you can do! Bonus points for each API you can incorporate! Go check out the APIs now to get a head start on the competition and get those creative juices flowing! You have less than two weeks to prepare!

Sponsor APIs: FreeSWITCH, Tropo, Kandy, Twilio, Plivo, and more?

IPv6 Round Table

IPv6 and why you should deploy it ASAP: John Brzozowski, Fellow and Chief Architect, IPv6 at Comcast, Bill Sandiford President of CNOC, Member of the board at ARIN.

Flowroute ? Jeopardy

Think you know about SIP? Do you know enough to beat the competition? Flowroute is hosting a SIP themed game of Jeopardy! Put your brain to the test and come see how much you really know!

DTMF-u

Do you like all things games? Well, then this is the game for you! But there is a twist! Before you can win the game you must build it! Using your choice of language or API you must build a game! There will be three categories of DTMF-u; Tic-tac-toe, DTMF pattern recognition, or Freestyle. You could build something that plays a random DTMF sequence, receives player input, and then either continues or fails the player. Or maybe a WebRTC based game of tic-tac-toe? Or surprise us! Have fun with it! Creative ways to fail a player may give you bonus points. The top three games will be played by everyone and the winner of each will take home a prize! All gaming bots will be screened via a Turing test to ensure no unintended apocalyptic consequences.

Show and Tell

Alright, now is your chance! You have been playing with the code all day and this is your chance to show off! We want to know what you?ve done and how you did it. Use your creativity and skills as a programmer to impress the judges and win a prize! This is a no holds barred all out free for all! Any language doing anything! Knock our socks off and take home a fabulous prize and a year?s supply of bragging rights!

Raffle Grand Prize!

The grand prize is a laser engraved commemorative FreeSWITCH 1.6 Edition dual-core 13? Retina MacBook Pro!

?

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From smontour at twc.com  Wed Jul 22 05:04:24 2015
From: smontour at twc.com (Admin)
Date: Tue, 21 Jul 2015 20:04:24 -0500
Subject: [Freeswitch-users]  Disable 401 and 407 for internal profile
Message-ID: <1437527064.2238.10.camel@twc.com>

I have had the dialplan configured for users 1000-1999
("destination_number" expression="^(1[0-9][0-9][0-9])$"> for months. I
have run thousands of calls using the same dialplan and users config.
files.
To disable auth., i made two changes in internal.xml. I set 'auth-calls'
value to 'false' and removed the line that contains
'apply-inbound-acl'. 
I didn't make any changes to neither dialplan nor the users config.
files.



From brian at freeswitch.org  Wed Jul 22 05:06:59 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 21 Jul 2015 20:06:59 -0500
Subject: [Freeswitch-users] Disable 401 and 407 for internal profile
In-Reply-To: <1437527064.2238.10.camel@twc.com>
References: <1437527064.2238.10.camel@twc.com>
Message-ID: <CAEJMVkBLKfZHNuBNYPM2ErOZcNu-y3W5HCUfcOML5igYsf0EQw@mail.gmail.com>

Probably hitting the wrong context now, because the internal profile is set
to use public and not default, user_context overrides this when they auth.

On Tue, Jul 21, 2015 at 8:04 PM, Admin <smontour at twc.com> wrote:

> I have had the dialplan configured for users 1000-1999
> ("destination_number" expression="^(1[0-9][0-9][0-9])$"> for months. I
> have run thousands of calls using the same dialplan and users config.
> files.
> To disable auth., i made two changes in internal.xml. I set 'auth-calls'
> value to 'false' and removed the line that contains
> 'apply-inbound-acl'.
> I didn't make any changes to neither dialplan nor the users config.
> files.
>
>
> _________________________________________________________________________
> 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
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From anthony.minessale at gmail.com  Wed Jul 22 09:40:49 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 22 Jul 2015 00:40:49 -0500
Subject: [Freeswitch-users] VideoCall with h264
In-Reply-To: <CAKOUqqsPXEayPWsDQ1nkQvhAdd4yvrbx7rQN8qfXfcggYLb-aA@mail.gmail.com>
References: <1437422047.3145.1.camel@twc.com>
	<917907795.20150721132224@seznam.cz>
	<CAEwTsAhB7GkoPpOEcyj7jgdardDRoA9ZaakKAWkcvHU7ixqLBQ@mail.gmail.com>
	<781319F2-9C91-478F-A612-A3A79EBECB7A@seznam.cz>
	<CAKbxfG_FwLXPX8zwgT2x33ZMAGxC0LrmtJAqBp5YFOJ6d00X+Q@mail.gmail.com>
	<E688F729-2CE9-4834-AA3F-8A555A5CA86E@seznam.cz>
	<CAKOUqqsPXEayPWsDQ1nkQvhAdd4yvrbx7rQN8qfXfcggYLb-aA@mail.gmail.com>
Message-ID: <CAKbxfG-85cU4bP3qNmfKXd5oo1gk9uF+CrePxxt8mBrw=S4=9g@mail.gmail.com>

you can record files with mod_vlc if you want and use mod_av for codecs.
You just prefix your record file names with {modname=mod_vlc}

On Tue, Jul 21, 2015 at 4:26 PM, Mario Gasparoni Junior <
mariogasparoni at gmail.com> wrote:

> Did you try adding "H264" in "global_codec_prefs" and
> "outbound_codec_prefs" in your vars.xml ?
>
> Sent from mobile
> Em 21/07/2015 17:29, "Denis Jakovlev" <yadenis at seznam.cz> escreveu:
>
>> Dobr? den.
>>
>> I'll try. The problem is that I have a great video records only mod_vlc.
>> With mod_av I did not write the video. But I will try of course. Mod_av I
>> have compiled.
>>
>> *-- *
>> *S pozdravem,*
>> *Ing.Denis Jakovlev                           *
>> *mob.tel <http://mob.tel>. 775-415-382*
>>
>> On 21. 7. 2015, at 22:11, Anthony Minessale <anthony.minessale at gmail.com>
>> wrote:
>>
>> build and use mod_av to install the codec and unload mod_h26x  mod_h26x
>> is only passthrough.
>>
>>
>> On Tue, Jul 21, 2015 at 1:00 PM, Denis Jakovlev <yadenis at seznam.cz>
>> wrote:
>>
>>> Hi.
>>>
>>> It is perfectly. Working system I just installed on Debian.
>>>
>>> How it works? I tried to add in my DialPlan
>>> <action application="export"
>>> data="nolocal:absolute_codec_string=OPUS,H264?/>
>>> instead
>>> <action application="export"
>>> data="nolocal:absolute_codec_string=OPUS,VP8?/>
>>>
>>> And I do not even get to connect. (incompatible destination)
>>> Although the settings vp8 everything works like clockwork. On both sides
>>> Chrome. Module mod_vpx, mod_h26x is loaded.
>>>
>>> What am I doing wrong?
>>>
>>>
>>>
>>>
>>> *-- *
>>> *S pozdravem,*
>>> *Ing.Denis Jakovlev                           *
>>> *mob.tel <http://mob.tel/>. 775-415-382*
>>>
>>> On 21. 7. 2015, at 17:57, Michael Jerris <mike at jerris.com> wrote:
>>>
>>> This currently works fine on Debian 8 Jessie.  To my knowledge other
>>> distributions do not include all the required dependencies so it should be
>>> possible but would require you to handle getting all of the dependencies
>>> sorted out yourself.  We will not be putting those additional dependencies
>>> for pattented algorithms in our repositories.
>>>
>>> On Tuesday, July 21, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
>>>
>>>>  Hi All,
>>>>
>>>> The question in the following. Is it possible to make calls through
>>>> freeswitch using h264 instead of using vp8?
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>>>> <http://mob.tel/>. 775-415-382 * <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
>>>
>>
>>
>>
>> --
>> 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
>>  _________________________________________________________________________
>> 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
>



-- 
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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From yadenis at seznam.cz  Wed Jul 22 10:29:47 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Wed, 22 Jul 2015 08:29:47 +0200
Subject: [Freeswitch-users] VideoCall with h264
In-Reply-To: <CAKOUqqsPXEayPWsDQ1nkQvhAdd4yvrbx7rQN8qfXfcggYLb-aA@mail.gmail.com>
References: <1437422047.3145.1.camel@twc.com>
	<917907795.20150721132224@seznam.cz> 
	<CAEwTsAhB7GkoPpOEcyj7jgdardDRoA9ZaakKAWkcvHU7ixqLBQ@mail.gmail.com>
	<781319F2-9C91-478F-A612-A3A79EBECB7A@seznam.cz>
	<CAKbxfG_FwLXPX8zwgT2x33ZMAGxC0LrmtJAqBp5YFOJ6d00X+Q@mail.gmail.com>
	<E688F729-2CE9-4834-AA3F-8A555A5CA86E@seznam.cz>
	<CAKOUqqsPXEayPWsDQ1nkQvhAdd4yvrbx7rQN8qfXfcggYLb-aA@mail.gmail.com>
Message-ID: <1835069044.20150722082947@seznam.cz>

Dobr? den,

Yes of course

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

?ter? 21. ?ervence 2015, 23:26:59, napsal jste:


Did you try adding "H264" in "global_codec_prefs" and "outbound_codec_prefs" in your vars.xml ?
Sent from mobile
Em 21/07/2015 17:29, "Denis Jakovlev" <yadenis at seznam.cz> escreveu:
Dobr? den.

I'll try. The problem is that I have a great video records only mod_vlc. With mod_av I did not write the video. But I will try of course. Mod_av I have compiled.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

On 21. 7. 2015, at 22:11, Anthony Minessale <anthony.minessale at gmail.com> wrote:

build and use mod_av to install the codec and unload mod_h26x  mod_h26x is only passthrough.


On Tue, Jul 21, 2015 at 1:00 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi.

It is perfectly. Working system I just installed on Debian. 

How it works? I tried to add in my DialPlan
<action application="export" data="nolocal:absolute_codec_string=OPUS,H264?/> 
instead 
<action application="export" data="nolocal:absolute_codec_string=OPUS,VP8?/>

And I do not even get to connect. (incompatible destination)
Although the settings vp8 everything works like clockwork. On both sides Chrome. Module mod_vpx, mod_h26x is loaded.

What am I doing wrong?




-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

On 21. 7. 2015, at 17:57, Michael Jerris <mike at jerris.com> wrote:

This currently works fine on Debian 8 Jessie.  To my knowledge other distributions do not include all the required dependencies so it should be possible but would require you to handle getting all of the dependencies sorted out yourself.  We will not be putting those additional dependencies for pattented algorithms in our repositories.

On Tuesday, July 21, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi All,

The question in the following. Is it possible to make calls through freeswitch using h264 instead of using vp8?


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
_________________________________________________________________________
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



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ? http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? 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
_________________________________________________________________________
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 yadenis at seznam.cz  Wed Jul 22 10:30:01 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Wed, 22 Jul 2015 08:30:01 +0200
Subject: [Freeswitch-users] VideoCall with h264
In-Reply-To: <CAKbxfG-85cU4bP3qNmfKXd5oo1gk9uF+CrePxxt8mBrw=S4=9g@mail.gmail.com>
References: <1437422047.3145.1.camel@twc.com>
	<917907795.20150721132224@seznam.cz> 
	<CAEwTsAhB7GkoPpOEcyj7jgdardDRoA9ZaakKAWkcvHU7ixqLBQ@mail.gmail.com>
	<781319F2-9C91-478F-A612-A3A79EBECB7A@seznam.cz>
	<CAKbxfG_FwLXPX8zwgT2x33ZMAGxC0LrmtJAqBp5YFOJ6d00X+Q@mail.gmail.com>
	<E688F729-2CE9-4834-AA3F-8A555A5CA86E@seznam.cz>
	<CAKOUqqsPXEayPWsDQ1nkQvhAdd4yvrbx7rQN8qfXfcggYLb-aA@mail.gmail.com>
	<CAKbxfG-85cU4bP3qNmfKXd5oo1gk9uF+CrePxxt8mBrw=S4=9g@mail.gmail.com>
Message-ID: <1008537781.20150722083001@seznam.cz>

Dobr? den,

Thank You. I do not know about it.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

st?eda 22. ?ervence 2015, 7:40:49, napsal jste:


you can record files with mod_vlc if you want and use mod_av for codecs.
You just prefix your record file names with {modname=mod_vlc}

On Tue, Jul 21, 2015 at 4:26 PM, Mario Gasparoni Junior <mariogasparoni at gmail.com> wrote:
Did you try adding "H264" in "global_codec_prefs" and "outbound_codec_prefs" in your vars.xml ?
Sent from mobile
Em 21/07/2015 17:29, "Denis Jakovlev" <yadenis at seznam.cz> escreveu:
Dobr? den.

I'll try. The problem is that I have a great video records only mod_vlc. With mod_av I did not write the video. But I will try of course. Mod_av I have compiled.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

On 21. 7. 2015, at 22:11, Anthony Minessale <anthony.minessale at gmail.com> wrote:

build and use mod_av to install the codec and unload mod_h26x  mod_h26x is only passthrough.


On Tue, Jul 21, 2015 at 1:00 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi.

It is perfectly. Working system I just installed on Debian. 

How it works? I tried to add in my DialPlan
<action application="export" data="nolocal:absolute_codec_string=OPUS,H264?/> 
instead 
<action application="export" data="nolocal:absolute_codec_string=OPUS,VP8?/>

And I do not even get to connect. (incompatible destination)
Although the settings vp8 everything works like clockwork. On both sides Chrome. Module mod_vpx, mod_h26x is loaded.

What am I doing wrong?




-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

On 21. 7. 2015, at 17:57, Michael Jerris <mike at jerris.com> wrote:

This currently works fine on Debian 8 Jessie.  To my knowledge other distributions do not include all the required dependencies so it should be possible but would require you to handle getting all of the dependencies sorted out yourself.  We will not be putting those additional dependencies for pattented algorithms in our repositories.

On Tuesday, July 21, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi All,

The question in the following. Is it possible to make calls through freeswitch using h264 instead of using vp8?


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
_________________________________________________________________________
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
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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+

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
_________________________________________________________________________
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

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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
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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+

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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From yadenis at seznam.cz  Wed Jul 22 10:43:43 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Wed, 22 Jul 2015 08:43:43 +0200
Subject: [Freeswitch-users] VideoCall with h264
In-Reply-To: <CAKbxfG_FwLXPX8zwgT2x33ZMAGxC0LrmtJAqBp5YFOJ6d00X+Q@mail.gmail.com>
References: <1437422047.3145.1.camel@twc.com>
	<917907795.20150721132224@seznam.cz> 
	<CAEwTsAhB7GkoPpOEcyj7jgdardDRoA9ZaakKAWkcvHU7ixqLBQ@mail.gmail.com>
	<781319F2-9C91-478F-A612-A3A79EBECB7A@seznam.cz>
	<CAKbxfG_FwLXPX8zwgT2x33ZMAGxC0LrmtJAqBp5YFOJ6d00X+Q@mail.gmail.com>
Message-ID: <866421435.20150722084343@seznam.cz>

Dobr? den,

Unfortunately with mod_av exactly the same result.
mod_av loaded. 
<action application="export" data="nolocal:absolute_codec_string=OPUS,H264?/> 

cd510fac-5fcc-463c-8acc-dd39781564ba 2015-07-22 08:34:46.869224 [NOTICE] sofia.c:7631 Hangup sofia/internal/onop4853 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]

Some additional settings, even?


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

?ter? 21. ?ervence 2015, 22:11:30, napsal jste:


build and use mod_av to install the codec and unload mod_h26x  mod_h26x is only passthrough.


On Tue, Jul 21, 2015 at 1:00 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi.

It is perfectly. Working system I just installed on Debian. 

How it works? I tried to add in my DialPlan
<action application="export" data="nolocal:absolute_codec_string=OPUS,H264?/> 
instead 
<action application="export" data="nolocal:absolute_codec_string=OPUS,VP8?/>

And I do not even get to connect. (incompatible destination)
Although the settings vp8 everything works like clockwork. On both sides Chrome. Module mod_vpx, mod_h26x is loaded.

What am I doing wrong?




-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

On 21. 7. 2015, at 17:57, Michael Jerris <mike at jerris.com> wrote:

This currently works fine on Debian 8 Jessie.  To my knowledge other distributions do not include all the required dependencies so it should be possible but would require you to handle getting all of the dependencies sorted out yourself.  We will not be putting those additional dependencies for pattented algorithms in our repositories.

On Tuesday, July 21, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi All,

The question in the following. Is it possible to make calls through freeswitch using h264 instead of using vp8?


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
_________________________________________________________________________
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



-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ? http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? 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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From Alexander.Haugg at c4b.de  Wed Jul 22 17:43:05 2015
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Wed, 22 Jul 2015 13:43:05 +0000
Subject: [Freeswitch-users] Build ERROR VS 2013
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC8517C428C@vc4bex2.corp.c4b.de>

Hi,

At the moment they'r bild errors for mod_conference, mod_shout, mod_rtmp and mod_verto. The same errors on "Relese" or "Debug" build for "win32" and "x64".
And the "libspeex" project need the "/FS" compiler flag!

Freeswitch version is from today:

? git rev-parse HEAD
17f8002936cb1797dc8bd46b4bad52800ebfb5cb

?  git describe
v1.5.final-1094-g17f8002


mod_conference:
==============
Error      4             error C1083: Cannot open include file: 'mod_conference.h': No such file or directory                D:\tmp\freeswitch\src\mod\applications\mod_conference\mod_conference.c           42           1                mod_conference

mod_shout:
==========
Error      4             error C1083: Cannot open include file: 'mpg123.h': No such file or directory                D:\tmp\freeswitch\src\mod\formats\mod_shout\mod_shout.c           36           1             mod_shout

mod_rtmp:
=========
Error      4             error C1083: Cannot open include file: 'openssl/hmac.h': No such file or directory                d:\tmp\freeswitch\src\mod\endpoints\mod_rtmp\handshake.h         30           1             mod_rtmp

mod_verto:
=========
Error      5             error C2079: '<unnamed-tag>::recv_addr6' uses undefined struct 'sockaddr_in6'                d:\tmp\freeswitch\src\mod\endpoints\mod_verto\mcast\mcast.h    75           1             mod_verto
Error      4             error C2079: '<unnamed-tag>::send_addr6' uses undefined struct 'sockaddr_in6'                d:\tmp\freeswitch\src\mod\endpoints\mod_verto\mcast\mcast.h    74           1             mod_verto


Thanks a lot!
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From brian at freeswitch.org  Wed Jul 22 17:50:29 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 22 Jul 2015 08:50:29 -0500
Subject: [Freeswitch-users] Build ERROR VS 2013
In-Reply-To: <B60DA91F5EA0E24A96E73CDE43CC9EC8517C428C@vc4bex2.corp.c4b.de>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC8517C428C@vc4bex2.corp.c4b.de>
Message-ID: <CAEJMVkCxH6QsEYDOA28Zf-xc+Uk_kc0Pp7pp0JteEJYpcW2AWA@mail.gmail.com>

Please see https://freeswitch.org/jira/browse/FS-7644

master currently isn't able to build on Windows, work is being done to
accomplish this.

In the future please search thru JIRA, if the issue doesn't appear to be
reported then please file an issue so we can track it properly moving
forward.

Thanks,

On Wed, Jul 22, 2015 at 8:43 AM, Alexander Haugg <Alexander.Haugg at c4b.de>
wrote:

>  Hi,
>
>
>
> At the moment they?r bild errors for mod_conference, mod_shout, mod_rtmp
> and mod_verto. The same errors on ?Relese? or ?Debug? build for ?win32? and
> ?x64?.
>
> And the ?libspeex? project need the ?/FS? compiler flag!
>
>
>
> Freeswitch version is from today:
>
> ? git rev-parse HEAD
>
> 17f8002936cb1797dc8bd46b4bad52800ebfb5cb
>
> ?  git describe
>
> v1.5.final-1094-g17f8002
>
>
>
>
>
> mod_conference:
>
> ==============
>
> Error      4             error C1083: Cannot open include file:
> 'mod_conference.h': No such file or directory
> D:\tmp\freeswitch\src\mod\applications\mod_conference\mod_conference.c
> 42           1                mod_conference
>
>
>
> mod_shout:
>
> ==========
>
> Error      4             error C1083: Cannot open include file:
> 'mpg123.h': No such file or directory
> D:\tmp\freeswitch\src\mod\formats\mod_shout\mod_shout.c
> 36           1             mod_shout
>
>
>
> mod_rtmp:
>
> =========
>
> Error      4             error C1083: Cannot open include file:
> 'openssl/hmac.h': No such file or directory
> d:\tmp\freeswitch\src\mod\endpoints\mod_rtmp\handshake.h
> 30           1             mod_rtmp
>
>
>
> mod_verto:
>
> =========
>
> Error      5             error C2079: '<unnamed-tag>::recv_addr6' uses
> undefined struct 'sockaddr_in6'
> d:\tmp\freeswitch\src\mod\endpoints\mod_verto\mcast\mcast.h    75
> 1             mod_verto
>
> Error      4             error C2079: '<unnamed-tag>::send_addr6' uses
> undefined struct 'sockaddr_in6'
> d:\tmp\freeswitch\src\mod\endpoints\mod_verto\mcast\mcast.h    74
> 1             mod_verto
>
>
>
>
>
> Thanks a lot!
>
> _________________________________________________________________________
> 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
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From brian at freeswitch.org  Wed Jul 22 17:51:26 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 22 Jul 2015 08:51:26 -0500
Subject: [Freeswitch-users] VideoCall with h264
In-Reply-To: <866421435.20150722084343@seznam.cz>
References: <1437422047.3145.1.camel@twc.com>
	<917907795.20150721132224@seznam.cz>
	<CAEwTsAhB7GkoPpOEcyj7jgdardDRoA9ZaakKAWkcvHU7ixqLBQ@mail.gmail.com>
	<781319F2-9C91-478F-A612-A3A79EBECB7A@seznam.cz>
	<CAKbxfG_FwLXPX8zwgT2x33ZMAGxC0LrmtJAqBp5YFOJ6d00X+Q@mail.gmail.com>
	<866421435.20150722084343@seznam.cz>
Message-ID: <CAEJMVkCWJhuifw3FRS7Dd4r8wkLfybxv6SzvrN_nVO6kk8sm8g@mail.gmail.com>

Did you make sure mod_vpx and mod_h26x IS NOT loaded?

On Wed, Jul 22, 2015 at 1:43 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Dobr? den,
>
> Unfortunately with mod_av exactly the same result.
> mod_av loaded.
> <action application="export"
> data="nolocal:absolute_codec_string=OPUS,H264?/>
>
> cd510fac-5fcc-463c-8acc-dd39781564ba 2015-07-22 08:34:46.869224 [NOTICE]
> sofia.c:7631 Hangup sofia/internal/onop4853 [CS_CONSUME_MEDIA]
> [INCOMPATIBLE_DESTINATION]
>
> Some additional settings, even?
>
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 ?ter? 21. ?ervence 2015, 22:11:30, napsal
> jste: *
>  build and use mod_av to install the codec and unload mod_h26x  mod_h26x
> is only passthrough.
>
>
> On Tue, Jul 21, 2015 at 1:00 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Hi.
>
> It is perfectly. Working system I just installed on Debian.
>
> How it works? I tried to add in my DialPlan
> <action application="export"
> data="nolocal:absolute_codec_string=OPUS,H264?/>
> instead
> <action application="export"
> data="nolocal:absolute_codec_string=OPUS,VP8?/>
>
> And I do not even get to connect. (incompatible destination)
> Although the settings vp8 everything works like clockwork. On both sides
> Chrome. Module mod_vpx, mod_h26x is loaded.
>
> What am I doing wrong?
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
> *. 775-415-382 *On 21. 7. 2015, at 17:57, Michael Jerris <mike at jerris.com>
> wrote:
>
> This currently works fine on Debian 8 Jessie.  To my knowledge other
> distributions do not include all the required dependencies so it should be
> possible but would require you to handle getting all of the dependencies
> sorted out yourself.  We will not be putting those additional dependencies
> for pattented algorithms in our repositories.
>
> On Tuesday, July 21, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Hi All,
>
> The question in the following. Is it possible to make calls through
> freeswitch using h264 instead of using vp8?
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
> *. 775-415-382 *
> _________________________________________________________________________
> 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
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? 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
>
> _________________________________________________________________________
> 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
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From yadenis at seznam.cz  Wed Jul 22 18:04:48 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Wed, 22 Jul 2015 16:04:48 +0200
Subject: [Freeswitch-users] VideoCall with h264
In-Reply-To: <CAEJMVkCWJhuifw3FRS7Dd4r8wkLfybxv6SzvrN_nVO6kk8sm8g@mail.gmail.com>
References: <1437422047.3145.1.camel@twc.com>
	<917907795.20150721132224@seznam.cz> 
	<CAEwTsAhB7GkoPpOEcyj7jgdardDRoA9ZaakKAWkcvHU7ixqLBQ@mail.gmail.com>
	<781319F2-9C91-478F-A612-A3A79EBECB7A@seznam.cz>
	<CAKbxfG_FwLXPX8zwgT2x33ZMAGxC0LrmtJAqBp5YFOJ6d00X+Q@mail.gmail.com>
	<866421435.20150722084343@seznam.cz>
	<CAEJMVkCWJhuifw3FRS7Dd4r8wkLfybxv6SzvrN_nVO6kk8sm8g@mail.gmail.com>
Message-ID: <1929949773.20150722160448@seznam.cz>

Dobr? den,

Yes of course. 100%

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

st?eda 22. ?ervence 2015, 15:51:26, napsal jste:


Did you make sure mod_vpx and mod_h26x IS NOT loaded?

On Wed, Jul 22, 2015 at 1:43 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
Dobr? den,

Unfortunately with mod_av exactly the same result.
mod_av loaded. 
<action application="export" data="nolocal:absolute_codec_string=OPUS,H264?/> 

cd510fac-5fcc-463c-8acc-dd39781564ba 2015-07-22 08:34:46.869224 [NOTICE] sofia.c:7631 Hangup sofia/internal/onop4853 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]

Some additional settings, even?


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

?ter? 21. ?ervence 2015, 22:11:30, napsal jste:


build and use mod_av to install the codec and unload mod_h26x  mod_h26x is only passthrough.


On Tue, Jul 21, 2015 at 1:00 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi.

It is perfectly. Working system I just installed on Debian. 

How it works? I tried to add in my DialPlan
<action application="export" data="nolocal:absolute_codec_string=OPUS,H264?/> 
instead 
<action application="export" data="nolocal:absolute_codec_string=OPUS,VP8?/>

And I do not even get to connect. (incompatible destination)
Although the settings vp8 everything works like clockwork. On both sides Chrome. Module mod_vpx, mod_h26x is loaded.

What am I doing wrong?




-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

On 21. 7. 2015, at 17:57, Michael Jerris <mike at jerris.com> wrote:

This currently works fine on Debian 8 Jessie.  To my knowledge other distributions do not include all the required dependencies so it should be possible but would require you to handle getting all of the dependencies sorted out yourself.  We will not be putting those additional dependencies for pattented algorithms in our repositories.

On Tuesday, July 21, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi All,

The question in the following. Is it possible to make calls through freeswitch using h264 instead of using vp8?


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
_________________________________________________________________________
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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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+

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

_________________________________________________________________________
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! | Reddit: /r/freeswitch
T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
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From 00asgaroth00 at gmail.com  Wed Jul 22 18:23:42 2015
From: 00asgaroth00 at gmail.com (Asgaroth)
Date: Wed, 22 Jul 2015 15:23:42 +0100
Subject: [Freeswitch-users] Enterprise originate not honouring fs_path
 in 302 redirect
In-Reply-To: <CAEJMVkAqT1kD9Xptg=X4nO78x6ckf=qfxvo_XthBy=w09tJydA@mail.gmail.com>
References: <55AE4F5D.7030307@gmail.com>
	<CAEJMVkAqT1kD9Xptg=X4nO78x6ckf=qfxvo_XthBy=w09tJydA@mail.gmail.com>
Message-ID: <55AFA76E.5090103@gmail.com>

Thanks for the info Brian, we'll need to find an alternative way to get 
this going then :-)

On 21/07/2015 21:34, Brian West wrote:
> Its not designed to do that, and we never will. Its an exploitable 
> attack vector if we did.
>
> On Tue, Jul 21, 2015 at 8:55 AM, Asgaroth <00asgaroth00 at gmail.com 
> <mailto:00asgaroth00 at gmail.com>> wrote:
>
>     Hi All,
>
>     First of all, I'm a relatively new user of FreeSWITCH so, please bear
>     with me if I'm asking a silly question.
>
>     I have users registering through multiple kamailio
>     loadbalancer/registrars, so when we build a bridge command to multiple
>     local users we send the initial invite(s) to the kamailio lookup
>     server
>     which then responds with a 302 redirect message specifying the contact
>     details of the user.
>
>     What I am trying to do is get FreeSWITCH to call a particular user
>     via a
>     particular loadbalancer based on information recieved in the 302
>     redirect. From what I have read in the archives, we can achieve this
>     with the fs_path directive on the bridge command. However,
>     how/where do
>     I specify the fs_path in the contact header in the 302 redirect in
>     such
>     a manner that FreeSWITCH will honour the fs_path directive.
>
>     I originally tried setting a fs_path parameter in the contact of
>     the 302
>     redirect message, something like the following:
>
>     Contact:
>     <sip:subscriber at 1.2.3.4:5678;rinstance=9af71926212a25cf;transport=UDP;received=sip:5.6.7.8:9012;fs_path=sip:1.2.3.5>
>
>     However, FreeSWITCH ignores the fs_path directive here and attempts to
>     send this call directly to 1.2.3.4, instead of sending it via 1.2.3.5.
>
>     Is this at all possible using the enterprise originate (:_: seperator)
>     feature of the bridge command?
>
>     We're attempting this with FreeSWITCH v 1.4.20 ("official" rpm
>     packages
>     from the repo)
>
>     Any info/tips/suggestions/beatings much appreciate :)
>
>     Thanks
>
>     _________________________________________________________________________
>     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
>
>
>
>
> -- 
>
> */Brian West/*
> brian at freeswitch.org <mailto: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
>
>
>
> _________________________________________________________________________
> 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 jaflong at yandex.com  Wed Jul 22 18:55:31 2015
From: jaflong at yandex.com (jaflong jaflong)
Date: Wed, 22 Jul 2015 17:55:31 +0300
Subject: [Freeswitch-users] I have a php script run by ivrd,
	is it possible to subscribe to events on this outbound eslconnection
Message-ID: <1744091437576931@web13g.yandex.ru>

I have a php script run by ivrd, is it possible to subscribe to
events on this outbound eslconnection 


From brian at freeswitch.org  Wed Jul 22 18:58:27 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 22 Jul 2015 09:58:27 -0500
Subject: [Freeswitch-users] I have a php script run by ivrd,
 is it possible to subscribe to events on this outbound eslconnection
In-Reply-To: <1744091437576931@web13g.yandex.ru>
References: <1744091437576931@web13g.yandex.ru>
Message-ID: <CAEJMVkB-9JUrgfo7fqm-KdfNFb4oYNJm9cwJkGxNefu+L-Uh0w@mail.gmail.com>

You may wish to try using outbound event socket if thats what you're wising
to accomplish.

On Wed, Jul 22, 2015 at 9:55 AM, jaflong jaflong <jaflong at yandex.com> wrote:

> I have a php script run by ivrd, is it possible to subscribe to
> events on this outbound eslconnection
>
> _________________________________________________________________________
> 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
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From yadenis at seznam.cz  Wed Jul 22 19:03:44 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Wed, 22 Jul 2015 17:03:44 +0200
Subject: [Freeswitch-users] Name instead of the caller ID
In-Reply-To: <55AFA76E.5090103@gmail.com>
References: <55AE4F5D.7030307@gmail.com> 
	<CAEJMVkAqT1kD9Xptg=X4nO78x6ckf=qfxvo_XthBy=w09tJydA@mail.gmail.com>
	<55AFA76E.5090103@gmail.com>
Message-ID: <582782477.20150722170344@seznam.cz>

Hi all,

I have a simple dialplan with IVR. If the client will set the correct PIN, it will be forwarded to the right person. 

The question in the following. Can I instead of the caller ID display for a selected name me? Caller's phone number can be different. It is even possible?

Bridge do like this.

<!-- Pripojeni na operatora VIP-->
<extension name="from-pe3ny-connect">
   <condition field="destination_number" expression="^CONNECTVIP$" break="on-true"">
    <action application="export" data="nolocal:absolute_codec_string=PCMA at 20i,PCMU at 20i"/>
    <action application="set" data="zrtp_secure_media=true"/>
    <!--<action application="set" data="inherit_codec=true""/>-->
    <action application="set" data="bypass_media=true"/>
    <action application="set" data="ringback=${us-ring}"/>
    <!--<action application="set" data="transfer_ringback=$${hold_music}"/>-->
    <action application="set" data="call_timeout=15"/>
<!-- na recepci -->
    <action application="set" data="hangup_after_bridge=true"/>
    <action application="set" data="continue_on_fail=true"/>
    <action application="bridge" data="user/1008 at sip1.applicloud.com,user/1007 at sip1.applicloud.com"/>
<!-- Na gsm -->
    <action application="answer"/>
    <action application="sleep" data="1000"/>
    <action application="bridge" data="sofia/gateway/pe3ny/732557771"/>-->
    <action application="hangup"/>
   </condition>
</extension> 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From yadenis at seznam.cz  Wed Jul 22 19:14:26 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Wed, 22 Jul 2015 17:14:26 +0200
Subject: [Freeswitch-users] Name instead of the caller ID
In-Reply-To: <582782477.20150722170344@seznam.cz>
References: <55AE4F5D.7030307@gmail.com>  
	<CAEJMVkAqT1kD9Xptg=X4nO78x6ckf=qfxvo_XthBy=w09tJydA@mail.gmail.com>
	<55AFA76E.5090103@gmail.com> <582782477.20150722170344@seznam.cz>
Message-ID: <3910652412.20150722171426@seznam.cz>

Dobr? den,

<action application="set" data="effective_caller_id_name=Bob Smith"/>

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

st?eda 22. ?ervence 2015, 17:03:44, napsal jste:


Hi all,

I have a simple dialplan with IVR. If the client will set the correct PIN, it will be forwarded to the right person. 

The question in the following. Can I instead of the caller ID display for a selected name me? Caller's phone number can be different. It is even possible?

Bridge do like this.

<!-- Pripojeni na operatora VIP-->
<extension name="from-pe3ny-connect">
  <condition field="destination_number" expression="^CONNECTVIP$" break="on-true"">
   <action application="export" data="nolocal:absolute_codec_string=PCMA at 20i,PCMU at 20i"/>
   <action application="set" data="zrtp_secure_media=true"/>
   <!--<action application="set" data="inherit_codec=true""/>-->
   <action application="set" data="bypass_media=true"/>
   <action application="set" data="ringback=${us-ring}"/>
   <!--<action application="set" data="transfer_ringback=$${hold_music}"/>-->
   <action application="set" data="call_timeout=15"/>
<!-- na recepci -->
   <action application="set" data="hangup_after_bridge=true"/>
   <action application="set" data="continue_on_fail=true"/>
   <action application="bridge" data="user/1008 at sip1.applicloud.com,user/1007 at sip1.applicloud.com"/>
<!-- Na gsm -->
   <action application="answer"/>
   <action application="sleep" data="1000"/>
   <action application="bridge" data="sofia/gateway/pe3ny/732557771"/>-->
   <action application="hangup"/>
  </condition>
</extension> 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From jaflong at yandex.com  Wed Jul 22 19:17:25 2015
From: jaflong at yandex.com (jaflong jaflong)
Date: Wed, 22 Jul 2015 18:17:25 +0300
Subject: [Freeswitch-users] I have a php script run by ivrd,
	is it possible to subscribe to events on this outbound eslconnection
In-Reply-To: <CAEJMVkB-9JUrgfo7fqm-KdfNFb4oYNJm9cwJkGxNefu+L-Uh0w@mail.gmail.com>
References: <1744091437576931@web13g.yandex.ru>
	<CAEJMVkB-9JUrgfo7fqm-KdfNFb4oYNJm9cwJkGxNefu+L-Uh0w@mail.gmail.com>
Message-ID: <1114761437578245@web12g.yandex.ru>

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From mike at jerris.com  Wed Jul 22 19:25:05 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 22 Jul 2015 11:25:05 -0400
Subject: [Freeswitch-users] Name instead of the caller ID
In-Reply-To: <3910652412.20150722171426@seznam.cz>
References: <55AE4F5D.7030307@gmail.com>
	<CAEJMVkAqT1kD9Xptg=X4nO78x6ckf=qfxvo_XthBy=w09tJydA@mail.gmail.com>
	<55AFA76E.5090103@gmail.com> <582782477.20150722170344@seznam.cz>
	<3910652412.20150722171426@seznam.cz>
Message-ID: <CAEwTsAgyh+9C=8O1Z-j2B9GOu25HwaTFL_D9AuRCLVrf62KQTw@mail.gmail.com>

it's probably way better to use origination_caller_id_name in the bridge
line

On Wednesday, July 22, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Dobr? den,
>
> <action application="set" data="effective_caller_id_name=Bob Smith"/>
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 st?eda 22. ?ervence 2015, 17:03:44, napsal
> jste: *
>  Hi all,
>
> I have a simple dialplan with IVR. If the client will set the correct PIN,
> it will be forwarded to the right person.
>
> The question in the following. Can I instead of the caller ID display for
> a selected name me? Caller's phone number can be different. It is even
> possible?
>
> Bridge do like this.
>
> <!-- Pripojeni na operatora VIP-->
> <extension name="from-pe3ny-connect">
>   <condition field="destination_number" expression="^CONNECTVIP$" break=
> "on-true"">
>    <action application="export" data=
> "nolocal:absolute_codec_string=PCMA at 20i,PCMU at 20i"/>
>    <action application="set" data="zrtp_secure_media=true"/>
>    <!--<action application="set" data="inherit_codec=true""/>-->
>    <action application="set" data="bypass_media=true"/>
>    <action application="set" data="ringback=${us-ring}"/>
>    <!--<action application="set"
> data="transfer_ringback=$${hold_music}"/>-->
>    <action application="set" data="call_timeout=15"/>
> <!-- na recepci -->
>    <action application="set" data="hangup_after_bridge=true"/>
>    <action application="set" data="continue_on_fail=true"/>
>    <action application="bridge" data="user/1008 at sip1.applicloud.com
> <javascript:_e(%7B%7D,'cvml','1008 at sip1.applicloud.com');>,user/
> 1007 at sip1.applicloud.com
> <javascript:_e(%7B%7D,'cvml','1007 at sip1.applicloud.com');>"/>
> <!-- Na gsm -->
>    <action application="answer"/>
>    <action application="sleep" data="1000"/>
>    <action application="bridge" data="sofia/gateway/pe3ny/732557771"/>-->
>   <action application="hangup"/>
>   </condition>
> </extension>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382*
>
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From adrottenberg at gmail.com  Wed Jul 22 19:27:16 2015
From: adrottenberg at gmail.com (Duvid Rottenberg)
Date: Wed, 22 Jul 2015 11:27:16 -0400
Subject: [Freeswitch-users] Build ERROR VS 2013
In-Reply-To: <CAEJMVkCxH6QsEYDOA28Zf-xc+Uk_kc0Pp7pp0JteEJYpcW2AWA@mail.gmail.com>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC8517C428C@vc4bex2.corp.c4b.de>
	<CAEJMVkCxH6QsEYDOA28Zf-xc+Uk_kc0Pp7pp0JteEJYpcW2AWA@mail.gmail.com>
Message-ID: <CAC4JBEB2FzhEwN5kq9wJ5mb2tpkSTD=-0Q7NyNMLty0-xyvfdQ@mail.gmail.com>

The problem is that there some modules that use features from C99, the one
I noticed is roundf. Visual Studio 2012 didn't support C99. Even if you
open a VS 2012 project in VS 2013 it will by default use the VS 2012
toolset which links to the older version of the C libraries.

I was able to build it on my machine by converting it to use the Visual
Studio 2013 toolset. There is a way to automatically apply it to all
projects in a solution, I don't remember exactly how I did it. I have it on
my todo list to create new 2013 solution & project files, but I have not
gotten around to it.

Thanks,
Duvid Rottenberg

On Wed, Jul 22, 2015 at 9:50 AM, Brian West <brian at freeswitch.org> wrote:

> Please see https://freeswitch.org/jira/browse/FS-7644
>
> master currently isn't able to build on Windows, work is being done to
> accomplish this.
>
> In the future please search thru JIRA, if the issue doesn't appear to be
> reported then please file an issue so we can track it properly moving
> forward.
>
> Thanks,
>
> On Wed, Jul 22, 2015 at 8:43 AM, Alexander Haugg <Alexander.Haugg at c4b.de>
> wrote:
>
>>  Hi,
>>
>>
>>
>> At the moment they?r bild errors for mod_conference, mod_shout, mod_rtmp
>> and mod_verto. The same errors on ?Relese? or ?Debug? build for ?win32? and
>> ?x64?.
>>
>> And the ?libspeex? project need the ?/FS? compiler flag!
>>
>>
>>
>> Freeswitch version is from today:
>>
>> ? git rev-parse HEAD
>>
>> 17f8002936cb1797dc8bd46b4bad52800ebfb5cb
>>
>> ?  git describe
>>
>> v1.5.final-1094-g17f8002
>>
>>
>>
>>
>>
>> mod_conference:
>>
>> ==============
>>
>> Error      4             error C1083: Cannot open include file:
>> 'mod_conference.h': No such file or directory
>> D:\tmp\freeswitch\src\mod\applications\mod_conference\mod_conference.c
>> 42           1                mod_conference
>>
>>
>>
>> mod_shout:
>>
>> ==========
>>
>> Error      4             error C1083: Cannot open include file:
>> 'mpg123.h': No such file or directory
>> D:\tmp\freeswitch\src\mod\formats\mod_shout\mod_shout.c
>> 36           1             mod_shout
>>
>>
>>
>> mod_rtmp:
>>
>> =========
>>
>> Error      4             error C1083: Cannot open include file:
>> 'openssl/hmac.h': No such file or directory
>> d:\tmp\freeswitch\src\mod\endpoints\mod_rtmp\handshake.h
>> 30           1             mod_rtmp
>>
>>
>>
>> mod_verto:
>>
>> =========
>>
>> Error      5             error C2079: '<unnamed-tag>::recv_addr6' uses
>> undefined struct 'sockaddr_in6'
>> d:\tmp\freeswitch\src\mod\endpoints\mod_verto\mcast\mcast.h    75
>> 1             mod_verto
>>
>> Error      4             error C2079: '<unnamed-tag>::send_addr6' uses
>> undefined struct 'sockaddr_in6'
>> d:\tmp\freeswitch\src\mod\endpoints\mod_verto\mcast\mcast.h    74
>> 1             mod_verto
>>
>>
>>
>>
>>
>> Thanks a lot!
>>
>> _________________________________________________________________________
>> 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
>
> _________________________________________________________________________
> 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 yadenis at seznam.cz  Wed Jul 22 19:40:50 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Wed, 22 Jul 2015 17:40:50 +0200
Subject: [Freeswitch-users] Name instead of the caller ID
In-Reply-To: <CAEwTsAgyh+9C=8O1Z-j2B9GOu25HwaTFL_D9AuRCLVrf62KQTw@mail.gmail.com>
References: <55AE4F5D.7030307@gmail.com> 
	<CAEJMVkAqT1kD9Xptg=X4nO78x6ckf=qfxvo_XthBy=w09tJydA@mail.gmail.com>
	<55AFA76E.5090103@gmail.com> <582782477.20150722170344@seznam.cz>
	<3910652412.20150722171426@seznam.cz>
	<CAEwTsAgyh+9C=8O1Z-j2B9GOu25HwaTFL_D9AuRCLVrf62KQTw@mail.gmail.com>
Message-ID: <1826680753.20150722174050@seznam.cz>

Dobr? den,

Like this?

<action application="bridge" data="{origination_caller_id_name=Bob Smith}user/1008 at sip1.applicloud.com"/>

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

st?eda 22. ?ervence 2015, 17:25:05, napsal jste:


it's probably way better to use origination_caller_id_name in the bridge line

On Wednesday, July 22, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
Dobr? den,

<action application="set" data="effective_caller_id_name=Bob Smith"/>

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

st?eda 22. ?ervence 2015, 17:03:44, napsal jste:


Hi all,

I have a simple dialplan with IVR. If the client will set the correct PIN, it will be forwarded to the right person. 

The question in the following. Can I instead of the caller ID display for a selected name me? Caller's phone number can be different. It is even possible?

Bridge do like this.

<!-- Pripojeni na operatora VIP-->
<extension name="from-pe3ny-connect">
 <condition field="destination_number" expression="^CONNECTVIP$" break="on-true"">
  <action application="export" data="nolocal:absolute_codec_string=PCMA at 20i,PCMU at 20i"/>
  <action application="set" data="zrtp_secure_media=true"/>
  <!--<action application="set" data="inherit_codec=true""/>-->
  <action application="set" data="bypass_media=true"/>
  <action application="set" data="ringback=${us-ring}"/>
  <!--<action application="set" data="transfer_ringback=$${hold_music}"/>-->
  <action application="set" data="call_timeout=15"/>
<!-- na recepci -->
  <action application="set" data="hangup_after_bridge=true"/>
  <action application="set" data="continue_on_fail=true"/>
  <action application="bridge" data="user/1008 at sip1.applicloud.com,user/1007 at sip1.applicloud.com"/>
<!-- Na gsm -->
  <action application="answer"/>
  <action application="sleep" data="1000"/>
  <action application="bridge" data="sofia/gateway/pe3ny/732557771"/>-->
  <action application="hangup"/>
 </condition>
</extension> 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From olegstolyar at gmail.com  Wed Jul 22 21:14:48 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Wed, 22 Jul 2015 10:14:48 -0700
Subject: [Freeswitch-users] Websocket creation logging
In-Reply-To: <CAC8JMO+9sc8nNVO1J31L7jvoDk8USK5oiywVv2tAKFxJjOr_Tw@mail.gmail.com>
References: <CAC8JMOLh5i5iOkvN46Sh0KM6fdwMkZ7hvXpGg-vak2_Tm_gcmg@mail.gmail.com>
	<5128BA24-F33E-4B66-A3FE-C54EE2C78474@jerris.com>
	<CAC8JMOL1rzwtN=mR6WWd5t10WteDeXjQwCerP1TwE=vR8+wpmA@mail.gmail.com>
	<BF9A62AC-2FAD-40A5-95BF-F7999754A2B7@jerris.com>
	<CAC8JMO+9sc8nNVO1J31L7jvoDk8USK5oiywVv2tAKFxJjOr_Tw@mail.gmail.com>
Message-ID: <CAC8JMOJGU0a0fJU3tayW5_E3Y2GNHajW=MHUVkdY0gUtew_VFA@mail.gmail.com>

So, I tried that.  I disconnected my computer from the network while on
call and then connected it back.  WebRTC came back but after that the UA
could not establish new sessions.  Also, I don't think the disconnected
event fired in this case.

On Mon, Jul 20, 2015 at 1:59 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> I already have this in my sofia.conf.xml
> <param name="tracelevel" value="DEBUG"/>
>
> How do I find tport related info in my log files?  I searched for t-port
> and tport and didn't find anything.
>
> Do I also need to set up log-level to debug?  I thought log-level was for
> the console and according to the wiki tracelevel was for the log files.
>
> On Mon, Jul 20, 2015 at 1:27 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> check the wiki on sofia debug logging
>>
>> On Jul 20, 2015, at 3:55 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>
>> Thanks Michael!
>>
>> In sofia.  Can you share more about tport_debug?  Where to find it and
>> how to enable or see?
>>
>> On Mon, Jul 20, 2015 at 12:37 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> In what?  sofia or mod_verto?  in sofia there is probably some
>>> tport_debug that would show that, in mod_verto you probably need to add
>>> some debug, but you will have to go look at code to be sure.
>>>
>>> > On Jul 20, 2015, at 3:30 PM, Oleg Stolyar <olegstolyar at gmail.com>
>>> wrote:
>>> >
>>> > Hi guys,
>>> >
>>> > I am experiencing what looks like a memory leak on my FS servers but I
>>> suspect that there is a bug in my WebRTC client code that keep establishing
>>> new websockets and not closing them properly which causes FS to require
>>> more and more memory.
>>> >
>>> > Is there a way in FS to log creation and closing of websockets?
>>> >
>>> > Thank you
>>> > Oleg
>>> >
>>> _________________________________________________________________________
>>> > 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 olegstolyar at gmail.com  Wed Jul 22 21:19:21 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Wed, 22 Jul 2015 10:19:21 -0700
Subject: [Freeswitch-users] Websocket creation logging
In-Reply-To: <CAC8JMOJGU0a0fJU3tayW5_E3Y2GNHajW=MHUVkdY0gUtew_VFA@mail.gmail.com>
References: <CAC8JMOLh5i5iOkvN46Sh0KM6fdwMkZ7hvXpGg-vak2_Tm_gcmg@mail.gmail.com>
	<5128BA24-F33E-4B66-A3FE-C54EE2C78474@jerris.com>
	<CAC8JMOL1rzwtN=mR6WWd5t10WteDeXjQwCerP1TwE=vR8+wpmA@mail.gmail.com>
	<BF9A62AC-2FAD-40A5-95BF-F7999754A2B7@jerris.com>
	<CAC8JMO+9sc8nNVO1J31L7jvoDk8USK5oiywVv2tAKFxJjOr_Tw@mail.gmail.com>
	<CAC8JMOJGU0a0fJU3tayW5_E3Y2GNHajW=MHUVkdY0gUtew_VFA@mail.gmail.com>
Message-ID: <CAC8JMOKmKox_H0A_kH_8ovTBhV=xt+fG3gQ=LuoH3pjxEUAj5w@mail.gmail.com>

Sorry, wrong email group :-) Please ignore.

On Wed, Jul 22, 2015 at 10:14 AM, Oleg Stolyar <olegstolyar at gmail.com>
wrote:

> So, I tried that.  I disconnected my computer from the network while on
> call and then connected it back.  WebRTC came back but after that the UA
> could not establish new sessions.  Also, I don't think the disconnected
> event fired in this case.
>
> On Mon, Jul 20, 2015 at 1:59 PM, Oleg Stolyar <olegstolyar at gmail.com>
> wrote:
>
>> I already have this in my sofia.conf.xml
>> <param name="tracelevel" value="DEBUG"/>
>>
>> How do I find tport related info in my log files?  I searched for t-port
>> and tport and didn't find anything.
>>
>> Do I also need to set up log-level to debug?  I thought log-level was for
>> the console and according to the wiki tracelevel was for the log files.
>>
>> On Mon, Jul 20, 2015 at 1:27 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> check the wiki on sofia debug logging
>>>
>>> On Jul 20, 2015, at 3:55 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>>
>>> Thanks Michael!
>>>
>>> In sofia.  Can you share more about tport_debug?  Where to find it and
>>> how to enable or see?
>>>
>>> On Mon, Jul 20, 2015 at 12:37 PM, Michael Jerris <mike at jerris.com>
>>> wrote:
>>>
>>>> In what?  sofia or mod_verto?  in sofia there is probably some
>>>> tport_debug that would show that, in mod_verto you probably need to add
>>>> some debug, but you will have to go look at code to be sure.
>>>>
>>>> > On Jul 20, 2015, at 3:30 PM, Oleg Stolyar <olegstolyar at gmail.com>
>>>> wrote:
>>>> >
>>>> > Hi guys,
>>>> >
>>>> > I am experiencing what looks like a memory leak on my FS servers but
>>>> I suspect that there is a bug in my WebRTC client code that keep
>>>> establishing new websockets and not closing them properly which causes FS
>>>> to require more and more memory.
>>>> >
>>>> > Is there a way in FS to log creation and closing of websockets?
>>>> >
>>>> > Thank you
>>>> > Oleg
>>>> >
>>>> _________________________________________________________________________
>>>> > 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 mike at jerris.com  Wed Jul 22 21:35:49 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 22 Jul 2015 13:35:49 -0400
Subject: [Freeswitch-users] Websocket creation logging
In-Reply-To: <CAC8JMOJGU0a0fJU3tayW5_E3Y2GNHajW=MHUVkdY0gUtew_VFA@mail.gmail.com>
References: <CAC8JMOLh5i5iOkvN46Sh0KM6fdwMkZ7hvXpGg-vak2_Tm_gcmg@mail.gmail.com>
	<5128BA24-F33E-4B66-A3FE-C54EE2C78474@jerris.com>
	<CAC8JMOL1rzwtN=mR6WWd5t10WteDeXjQwCerP1TwE=vR8+wpmA@mail.gmail.com>
	<BF9A62AC-2FAD-40A5-95BF-F7999754A2B7@jerris.com>
	<CAC8JMO+9sc8nNVO1J31L7jvoDk8USK5oiywVv2tAKFxJjOr_Tw@mail.gmail.com>
	<CAC8JMOJGU0a0fJU3tayW5_E3Y2GNHajW=MHUVkdY0gUtew_VFA@mail.gmail.com>
Message-ID: <CAEwTsAh77Ks+EFe2y3TnLwnrX0JzT9hL+AF4CKCwBXpW8Z0BmA@mail.gmail.com>

we don't have reconnect features in sip ws.  verto supports this
functionality

On Wednesday, July 22, 2015, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> So, I tried that.  I disconnected my computer from the network while on
> call and then connected it back.  WebRTC came back but after that the UA
> could not establish new sessions.  Also, I don't think the disconnected
> event fired in this case.
>
> On Mon, Jul 20, 2015 at 1:59 PM, Oleg Stolyar <olegstolyar at gmail.com
> <javascript:_e(%7B%7D,'cvml','olegstolyar at gmail.com');>> wrote:
>
>> I already have this in my sofia.conf.xml
>> <param name="tracelevel" value="DEBUG"/>
>>
>> How do I find tport related info in my log files?  I searched for t-port
>> and tport and didn't find anything.
>>
>> Do I also need to set up log-level to debug?  I thought log-level was for
>> the console and according to the wiki tracelevel was for the log files.
>>
>> On Mon, Jul 20, 2015 at 1:27 PM, Michael Jerris <mike at jerris.com
>> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>> wrote:
>>
>>> check the wiki on sofia debug logging
>>>
>>> On Jul 20, 2015, at 3:55 PM, Oleg Stolyar <olegstolyar at gmail.com
>>> <javascript:_e(%7B%7D,'cvml','olegstolyar at gmail.com');>> wrote:
>>>
>>> Thanks Michael!
>>>
>>> In sofia.  Can you share more about tport_debug?  Where to find it and
>>> how to enable or see?
>>>
>>> On Mon, Jul 20, 2015 at 12:37 PM, Michael Jerris <mike at jerris.com
>>> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>> wrote:
>>>
>>>> In what?  sofia or mod_verto?  in sofia there is probably some
>>>> tport_debug that would show that, in mod_verto you probably need to add
>>>> some debug, but you will have to go look at code to be sure.
>>>>
>>>> > On Jul 20, 2015, at 3:30 PM, Oleg Stolyar <olegstolyar at gmail.com
>>>> <javascript:_e(%7B%7D,'cvml','olegstolyar at gmail.com');>> wrote:
>>>> >
>>>> > Hi guys,
>>>> >
>>>> > I am experiencing what looks like a memory leak on my FS servers but
>>>> I suspect that there is a bug in my WebRTC client code that keep
>>>> establishing new websockets and not closing them properly which causes FS
>>>> to require more and more memory.
>>>> >
>>>> > Is there a way in FS to log creation and closing of websockets?
>>>> >
>>>> > Thank you
>>>> > Oleg
>>>> >
>>>> _________________________________________________________________________
>>>> > Professional FreeSWITCH Consulting Services:
>>>> > consulting at freeswitch.org
>>>> <javascript:_e(%7B%7D,'cvml','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
>>>> <javascript:_e(%7B%7D,'cvml','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
>>>> <javascript:_e(%7B%7D,'cvml','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
>>>> <javascript:_e(%7B%7D,'cvml','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
>>> <javascript:_e(%7B%7D,'cvml','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
>>> <javascript:_e(%7B%7D,'cvml','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
>>> <javascript:_e(%7B%7D,'cvml','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
>>> <javascript:_e(%7B%7D,'cvml','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 danny.gershman at gmail.com  Thu Jul 23 01:09:11 2015
From: danny.gershman at gmail.com (Danny Gershman)
Date: Wed, 22 Jul 2015 21:09:11 +0000
Subject: [Freeswitch-users] Endless playback in conference
Message-ID: <CAEorZynEWmqnOAEfY00_k1Rwe6jPjsjNSPxJjVaTq_neOMFozg@mail.gmail.com>

I'm trying to do an endless playback of an mp3 file in a conference.   I
have a couple of ideas, but none seem really solid.

1) Pass a variable on play and monitor from mod_event_socket and play again
if not forcibly terminated.

2) Load up local_stream dynamically from an xmlhttp server, and then
restart the local stream service, however will interrupt MOH for other
users.

Any other ideas?  Is there a way to do looping for "api" through
mod_event_socket?  I know you can with "sendmsg"

--Danny Gershman
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From bote_radio at botecomm.com  Thu Jul 23 02:26:53 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Wed, 22 Jul 2015 18:26:53 -0400
Subject: [Freeswitch-users] Endless playback in conference
In-Reply-To: <CAEorZynEWmqnOAEfY00_k1Rwe6jPjsjNSPxJjVaTq_neOMFozg@mail.gmail.com>
References: <CAEorZynEWmqnOAEfY00_k1Rwe6jPjsjNSPxJjVaTq_neOMFozg@mail.gmail.com>
Message-ID: <01ec01d0c4cd$82fd1bf0$88f753d0$@botecomm.com>

There?s a parameter for mod_conference named ?perpetual-sound? that looks like it would do the trick.

 

It?s about 1/3 of the way down

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

 

PLEASE check further for any changes that might have been made in the latest FreeSWITCH as the conference module has undergone substantial changes and perpetual-sound might have been one of them. 

 

Bote

 

 

 

 

From: Danny Gershman
Sent: Wednesday, 22 July, 2015 17:09
Subject: [Freeswitch-users] Endless playback in conference

 

I'm trying to do an endless playback of an mp3 file in a conference.   I have a couple of ideas, but none seem really solid.

 

1) Pass a variable on play and monitor from mod_event_socket and play again if not forcibly terminated.  

 

2) Load up local_stream dynamically from an xmlhttp server, and then restart the local stream service, however will interrupt MOH for other users.

 

Any other ideas?  Is there a way to do looping for "api" through mod_event_socket?  I know you can with "sendmsg"

 

--Danny Gershman

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From jprangi at gmail.com  Thu Jul 23 03:58:40 2015
From: jprangi at gmail.com (Jai Rangi)
Date: Wed, 22 Jul 2015 16:58:40 -0700
Subject: [Freeswitch-users] Mod_callcenter ODBC vs File
Message-ID: <CAPfjz3gqws5xNFYa8mKfh5NtFrVJo_sCHxqu_a5w3h2a5jcPZw@mail.gmail.com>

Hello,

FS version:  FreeSWITCH Version
1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221 2015-03-12
18:55:23Z 64bit)

I found a very strange behavior on mod_call center when used with odbc.
This was easy to reproduce when the agents are busy and new calls come in.
New member goes in abandoned state immediately and stay in same state
forever. Even if previous call is complete and Agent is in Waiting state.
Infact some times new members go in abandoned state immediately even if the
agents are available.

Everything works perfect once I changed the mod_callcenter.conf.xml config
to file only mode, new calls go in Waiting state and call is connected to
agent as soon as the agent is free.

Any idea why the new member goes in abandoned state randomly in place of
being in waiting state? Can it be due to too small timeout in call center
module as it query the DB every time? Can that be changed?

Anyone else had smiler problem, any experience or any recommendations will
be much appreciated.

Thank you,
-Jai
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From brian at freeswitch.org  Thu Jul 23 05:48:43 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 22 Jul 2015 20:48:43 -0500
Subject: [Freeswitch-users] Mod_callcenter ODBC vs File
In-Reply-To: <CAPfjz3gqws5xNFYa8mKfh5NtFrVJo_sCHxqu_a5w3h2a5jcPZw@mail.gmail.com>
References: <CAPfjz3gqws5xNFYa8mKfh5NtFrVJo_sCHxqu_a5w3h2a5jcPZw@mail.gmail.com>
Message-ID: <CAEJMVkDyLo1iegWVP-gBdtL8LDuvzg0+fBUiru51HFKBfjGqSA@mail.gmail.com>

Once you test our latest release (1.4.20):

[image: Inline image 1]

https://freeswitch.org/jira

Thanks,


On Wed, Jul 22, 2015 at 6:58 PM, Jai Rangi <jprangi at gmail.com> wrote:

> Hello,
>
> FS version:  FreeSWITCH Version
> 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221 2015-03-12
> 18:55:23Z 64bit)
>
> I found a very strange behavior on mod_call center when used with odbc.
> This was easy to reproduce when the agents are busy and new calls come in.
> New member goes in abandoned state immediately and stay in same state
> forever. Even if previous call is complete and Agent is in Waiting state.
> Infact some times new members go in abandoned state immediately even if the
> agents are available.
>
> Everything works perfect once I changed the mod_callcenter.conf.xml config
> to file only mode, new calls go in Waiting state and call is connected to
> agent as soon as the agent is free.
>
> Any idea why the new member goes in abandoned state randomly in place of
> being in waiting state? Can it be due to too small timeout in call center
> module as it query the DB every time? Can that be changed?
>
> Anyone else had smiler problem, any experience or any recommendations will
> be much appreciated.
>
> Thank you,
> -Jai
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> 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
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From brian at freeswitch.org  Thu Jul 23 06:04:08 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 22 Jul 2015 21:04:08 -0500
Subject: [Freeswitch-users] I have a php script run by ivrd,
 is it possible to subscribe to events on this outbound eslconnection
In-Reply-To: <1114761437578245@web12g.yandex.ru>
References: <1744091437576931@web13g.yandex.ru>
	<CAEJMVkB-9JUrgfo7fqm-KdfNFb4oYNJm9cwJkGxNefu+L-Uh0w@mail.gmail.com>
	<1114761437578245@web12g.yandex.ru>
Message-ID: <CAEJMVkAVTycst4C4VQxv7g5BakoRAtvFQMuseoGRC3mq=Vdqnw@mail.gmail.com>

try 'event*s* all plain;

On Wed, Jul 22, 2015 at 10:17 AM, jaflong jaflong <jaflong at yandex.com>
wrote:

>
>
> I am usingoutbount event socket but cannot get it to work
>
> -----------------------------------
>
> require_once('ESL.php');
>
> $audio_location = "/usr/local/freeswitch/sounds/";
> $music = $audio_location . "demo.wav";
>
> $conn = new ESLconnection(0);
>
> $conn->execute("answer");
>
> $conn->events("plain","all");
>
>
> while ($conn->connected() == 1) {
>
>     $e = $conn->recvEvent();
>     if ($e) {
>         $arrEvent = eventParse($e);
>
>     } else {
>
>
>     }
>
> }
>
> -----------------------------------
>
> This does not work, give response
>
> event plain all
>
> Content-Type: command/reply
> Reply-Text: -ERR command not found
>
>
> 22.07.2015, 18:00, "Brian West" <brian at freeswitch.org>:
>
> You may wish to try using outbound event socket if thats what you're
> wising to accomplish.
>
> On Wed, Jul 22, 2015 at 9:55 AM, jaflong jaflong <jaflong at yandex.com>
> wrote:
>
> I have a php script run by ivrd, is it possible to subscribe to
> events on this outbound eslconnection
>
> _________________________________________________________________________
> 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
> ,
>
> _________________________________________________________________________
> 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*
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
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From alexeymelnichuck at gmail.com  Wed Jul 22 18:17:07 2015
From: alexeymelnichuck at gmail.com (Alexey Melnichuk)
Date: Wed, 22 Jul 2015 14:17:07 +0000 (UTC)
Subject: [Freeswitch-users] Break unaswered bridge
Message-ID: <loom.20150722T161450-208@post.gmane.org>

This is simple dialplan that I try to implement

<extension>
  <condition field="destination_number" expression="^\d+$" break="on-true">
    <action application="bind_digit_action" 
      data="conf,*1,exec:execute_extension,xfer"/>
    <action application="digit_action_set_realm" data="conf"/>
    <action application="conference" data="100 at default"/>
  </condition>

  <condition field="destination_number" expression="^xfer$" break="on-true">
    <action application="bind_digit_action" data="xfer,*1,?????????"/>
    <action application="digit_action_set_realm" data="xfer"/>
    <action application="digit_action_set_realm" data="xfer"/>
    <action application="read" 
      data="2 11 'tone_stream://%(10000,0,350,440)' target_num 30000 #"/>
    <action application="bridge" data="user/${target_num}@${domain_name}"/>
  </condition>

</extension>

I have two problem with this dialplan
1. in secon `bind_digit_action` I can not use `exec` only `api`.
e.g. <action application="bind_digit_action" data="xfer,*1,exec:info"/>
does not work. Application executes only when I return from first 
`exec:execute_extension`

2. I can not cancel bridge before answer

I can not use transfer because I do not whant leave conference 
(e.g. user has `end_conference=true` flag).
I try use uuid_kill from Lua like "xfer,*1,lua:kill_legs.lua ${uuid}"

  -- kill_legs.lua
  local uuid = argv[1]
  local sql = ("select uuid from channels where call_uuid='%s' and 
uuid<>'%s'")
    :format(uuid, uuid)
  dbh:query(sql, function(row)
    local res = api:executeString("uuid_kill " .. row.uuid)
  end)

But this is does not work because it kills only active channels, 
but for dial-string like 
`loopback/111/my.domain,[leg_delay_start=20]loopback/222/my.domain`
second channels creates only after 20 sec.
And because I may have more then one channels I can not use `originate_uuid`
In documentation I found `break` application but as I say I can not use it.
`uuid_break` api break only play audio.



From alexeymelnichuck at gmail.com  Thu Jul 23 07:59:18 2015
From: alexeymelnichuck at gmail.com (Alexey Melnichuk)
Date: Thu, 23 Jul 2015 03:59:18 +0000 (UTC)
Subject: [Freeswitch-users] Break unaswered bridge
Message-ID: <loom.20150723T055711-382@post.gmane.org>

This is simple dialplan that I try to implement

<extension>
  <condition field="destination_number" expression="^\d+$" break="on-true">
    <action application="bind_digit_action" 
data="conf,*1,exec:execute_extension,xfer"/>
    <action application="digit_action_set_realm" data="conf"/>
    <action application="conference" data="100 at default"/>
  </condition>

  <condition field="destination_number" expression="^xfer$" break="on-true">
    <action application="bind_digit_action" data="xfer,*1,?????????"/>
    <action application="digit_action_set_realm" data="xfer"/>
    <action application="digit_action_set_realm" data="xfer"/>
    <action application="read" data="2 11 'tone_stream://%(10000,0,350,440)' 
num 30000 #"/>
    <action application="bridge" data="user/${num}@${domain_name}"/>
  </condition>

</extension>

I have two problem with this dialplan
1. in secon `bind_digit_action` I can not use `exec` only `api`.
e.g. <action application="bind_digit_action" data="xfer,*1,exec:info"/> does 
not work.
application executes only when I return from first `exec:execute_extension`

2. I can not cancel bridge before answer

I can not use transfer because I do not whant leave conference
(e.g. user has `end_conference=true` flag).
I try use uuid_kill from Lua like "xfer,*1,lua:kill_legs.lua ${uuid}"

  -- kill_legs.lua
  local uuid = argv[1]
  local sql = ("select uuid from channels where call_uuid='%s' and 
uuid<>'%s'"):format(uuid, uuid)
  dbh:query(sql, function(row)
    local res = api:executeString("uuid_kill " .. row.uuid)
  end)

But this is does not work because it kills only active channels, 
but for dial-string like `loopback/111/my.domain,
[leg_delay_start=20]loopback/222/my.domain`
second channels creates only after 20 sec.
And because I may have more then one channels I can not use `originate_uuid`
In documentation I found `break` application but as I say I can not use it.
`uuid_break` api break only play audio.



From Alexander.Haugg at c4b.de  Thu Jul 23 09:13:14 2015
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Thu, 23 Jul 2015 05:13:14 +0000
Subject: [Freeswitch-users] Build ERROR VS 2013
In-Reply-To: <CAC4JBEB2FzhEwN5kq9wJ5mb2tpkSTD=-0Q7NyNMLty0-xyvfdQ@mail.gmail.com>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC8517C428C@vc4bex2.corp.c4b.de>
	<CAEJMVkCxH6QsEYDOA28Zf-xc+Uk_kc0Pp7pp0JteEJYpcW2AWA@mail.gmail.com>
	<CAC4JBEB2FzhEwN5kq9wJ5mb2tpkSTD=-0Q7NyNMLty0-xyvfdQ@mail.gmail.com>
Message-ID: <B60DA91F5EA0E24A96E73CDE43CC9EC8517C4390@vc4bex2.corp.c4b.de>

Thanks for the answers!

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Duvid Rottenberg
Gesendet: Mittwoch, 22. Juli 2015 17:27
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Build ERROR VS 2013

The problem is that there some modules that use features from C99, the one I noticed is roundf. Visual Studio 2012 didn't support C99. Even if you open a VS 2012 project in VS 2013 it will by default use the VS 2012 toolset which links to the older version of the C libraries.

I was able to build it on my machine by converting it to use the Visual Studio 2013 toolset. There is a way to automatically apply it to all projects in a solution, I don't remember exactly how I did it. I have it on my todo list to create new 2013 solution & project files, but I have not gotten around to it.

Thanks,
Duvid Rottenberg

On Wed, Jul 22, 2015 at 9:50 AM, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
Please see https://freeswitch.org/jira/browse/FS-7644

master currently isn't able to build on Windows, work is being done to accomplish this.

In the future please search thru JIRA, if the issue doesn't appear to be reported then please file an issue so we can track it properly moving forward.

Thanks,

On Wed, Jul 22, 2015 at 8:43 AM, Alexander Haugg <Alexander.Haugg at c4b.de<mailto:Alexander.Haugg at c4b.de>> wrote:
Hi,

At the moment they?r bild errors for mod_conference, mod_shout, mod_rtmp and mod_verto. The same errors on ?Relese? or ?Debug? build for ?win32? and ?x64?.
And the ?libspeex? project need the ?/FS? compiler flag!

Freeswitch version is from today:

==> git rev-parse HEAD
17f8002936cb1797dc8bd46b4bad52800ebfb5cb

==>  git describe
v1.5.final-1094-g17f8002


mod_conference:
==============
Error      4             error C1083: Cannot open include file: 'mod_conference.h': No such file or directory                D:\tmp\freeswitch\src\mod\applications\mod_conference\mod_conference.c           42           1                mod_conference

mod_shout:
==========
Error      4             error C1083: Cannot open include file: 'mpg123.h': No such file or directory                D:\tmp\freeswitch\src\mod\formats\mod_shout\mod_shout.c           36           1             mod_shout

mod_rtmp:
=========
Error      4             error C1083: Cannot open include file: 'openssl/hmac.h': No such file or directory                d:\tmp\freeswitch\src\mod\endpoints\mod_rtmp\handshake.h         30           1             mod_rtmp

mod_verto:
=========
Error      5             error C2079: '<unnamed-tag>::recv_addr6' uses undefined struct 'sockaddr_in6'                d:\tmp\freeswitch\src\mod\endpoints\mod_verto\mcast\mcast.h    75           1             mod_verto
Error      4             error C2079: '<unnamed-tag>::send_addr6' uses undefined struct 'sockaddr_in6'                d:\tmp\freeswitch\src\mod\endpoints\mod_verto\mcast\mcast.h    74           1             mod_verto


Thanks a lot!

_________________________________________________________________________
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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[Das Bild wurde vom Absender entfernt.]

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<tel:%2B19184209001> | F:+19184209002<tel:%2B19184209002> | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

_________________________________________________________________________
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
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From adam.lappe at qsc.de  Thu Jul 23 12:15:24 2015
From: adam.lappe at qsc.de (Lappe, Adam)
Date: Thu, 23 Jul 2015 08:15:24 +0000
Subject: [Freeswitch-users] mod_local_stream moh confusion
Message-ID: <2E67ADAE1D3582409C90EBAE8C64C5070E5569@QSCDEMXP01B.ONE4ALL.LAN>

Hello,

i am testing the latest 1.4.19 Version of FreeSWITCH.
Currently we are running an old 1.2.7 Version.

Everything seems to work fine, but there is 1 error that is very confusing:

When a call gets transfered by the callee (i.e. by the receptionist) the call will be terminated.
All I see is this error line:
[ERR] switch_core_file.c:149 Invalid file format [local_stream] for [moh]!

I don't use the local_stream module.
freeswitch at internal> module_exists mod_local_stream
false

This error does not exists with the old version.

Is this a bug, or am I missing something?

Thanks in advance,
Adam
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From yadenis at seznam.cz  Thu Jul 23 12:25:30 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Thu, 23 Jul 2015 10:25:30 +0200
Subject: [Freeswitch-users] The problem with the TURN server and connection.
In-Reply-To: <loom.20150722T161450-208@post.gmane.org>
References: <loom.20150722T161450-208@post.gmane.org>
Message-ID: <778052174.20150723102530@seznam.cz>

Hi All,

I have one problem. 
One of my clients is behind the serious firewall and they are banned UDP connection at all. His clients are connected to it via the Internet. 
I put my own Turn/Stun server (rfc5766-turn-server from Google). It works the same server as the freeswitch. It works fine.
The problem follows. When all banned UDP connection - you can not connect (INCOMPATIBLE_DESTINATION). If you allow outgoing UDP At least, that is connected with no problems.
Is it even possible connection if one of the parties completely banned UDP connection? 

I thought that the problem is just the decides the turn server. But I is not working for me.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From bhavikpatel14388 at gmail.com  Thu Jul 23 13:36:39 2015
From: bhavikpatel14388 at gmail.com (bhavik patel)
Date: Thu, 23 Jul 2015 15:06:39 +0530
Subject: [Freeswitch-users] BLF Implementation
In-Reply-To: <CAG6TCM_mRoPM-p2Yi1iuLRk8y=wCRTvq4g9Xie3MHWNPjdKjhQ@mail.gmail.com>
References: <CAG6TCM9xynte8HzQxFXQ0VaZKmz4Njh+qd0TtZ6WP+KNgkxHTw@mail.gmail.com>
	<CAG6TCM8v4OjsQ87gxZYCYVz4aHZz-fo-uWJ_CEkA82FMypCfaw@mail.gmail.com>
	<CAEwTsAiA+3Jo_6hjxxt7CVeHFwdd63UsU1G-4-DWve3DfXr2GQ@mail.gmail.com>
	<CAG6TCM_mRoPM-p2Yi1iuLRk8y=wCRTvq4g9Xie3MHWNPjdKjhQ@mail.gmail.com>
Message-ID: <CAG6TCM8cLHpope5jHrQTtDC79FTp6kg3oOUxyK-ToEyaFbhAKg@mail.gmail.com>

Do i need to configure ODBC connection also to make it working ?

On Thu, Jul 16, 2015 at 9:38 AM, bhavik patel <bhavikpatel14388 at gmail.com>
wrote:

> I did try "
> http://www.microweb10.com/index.php/en/tutorials/enable-presence-fs" but
> not getting success.
>
> Please suggest me.
>
> On Thu, Jul 16, 2015 at 9:23 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> manage-presence needs to be enabled
>>
>>
>> On Wednesday, July 15, 2015, bhavik patel <bhavikpatel14388 at gmail.com>
>> wrote:
>>
>>> Any feedback please.
>>>
>>> On Tue, Jul 14, 2015 at 5:38 PM, bhavik patel <
>>> bhavikpatel14388 at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> I am trying to implement BLF in Grandstream (GXP1400) with freeswitch
>>>> 1.4.20.
>>>>
>>>> For that , i used "<param name="send-presence-on-register"
>>>> value="true"/>" in sip profile.
>>>>
>>>> But Not getting any NOTIFY request to sip phone,and BLF is not working.
>>>>
>>>> Can any one suggest me how to enable this function in Freeswitch ?
>>>>
>>>> Any help would be much appreciated.
>>>>
>>>> --
>>>> Thanks,
>>>> Bhavik Patel
>>>>
>>>>
>>>>
>>>>
>>>
>>>
>>> --
>>> Thanks,
>>> Bhavik Patel
>>>
>>>
>> _________________________________________________________________________
>> 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
>>
>
>
>
> --
> Thanks,
> Bhavik Patel
>
>


-- 
Thanks,
Bhavik Patel
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From jaflong at yandex.com  Thu Jul 23 14:12:23 2015
From: jaflong at yandex.com (jaflong jaflong)
Date: Thu, 23 Jul 2015 13:12:23 +0300
Subject: [Freeswitch-users] I have a php script run by ivrd,
	is it possible to subscribe to events on this outbound eslconnection
In-Reply-To: <CAEJMVkAVTycst4C4VQxv7g5BakoRAtvFQMuseoGRC3mq=Vdqnw@mail.gmail.com>
References: <1744091437576931@web13g.yandex.ru>
	<CAEJMVkB-9JUrgfo7fqm-KdfNFb4oYNJm9cwJkGxNefu+L-Uh0w@mail.gmail.com>
	<1114761437578245@web12g.yandex.ru>
	<CAEJMVkAVTycst4C4VQxv7g5BakoRAtvFQMuseoGRC3mq=Vdqnw@mail.gmail.com>
Message-ID: <1181481437646343@web10h.yandex.ru>

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From ben at langfeld.co.uk  Thu Jul 23 15:56:43 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Thu, 23 Jul 2015 08:56:43 -0300
Subject: [Freeswitch-users] The problem with the TURN server and
	connection.
In-Reply-To: <778052174.20150723102530@seznam.cz>
References: <loom.20150722T161450-208@post.gmane.org>
	<778052174.20150723102530@seznam.cz>
Message-ID: <CAAyX+KFmuLQVRkP-Au5cN_Tu5aLD0377r8fwrxx3+2Q3E+hiCw@mail.gmail.com>

Take a look at coturn, a fork of rfc5766-turn-server, which supports
RFC6062, the TCP transport for TURN.

On 23 July 2015 at 05:25, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Hi All,
>
> I have one problem.
> One of my clients is behind the serious firewall and they are banned UDP
> connection at all. His clients are connected to it via the Internet.
> I put my own Turn/Stun server (rfc5766-turn-server from Google). It works
> the same server as the freeswitch. It works fine.
> The problem follows. When all banned UDP connection - you can not connect
> (INCOMPATIBLE_DESTINATION). If you allow outgoing UDP At least, that is
> connected with no problems.
> Is it even possible connection if one of the parties completely banned UDP
> connection?
>
> I thought that the problem is just the decides the turn server. But I is
> not working for me.
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 * <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 gmaruzz at gmail.com  Thu Jul 23 16:32:28 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 23 Jul 2015 14:32:28 +0200
Subject: [Freeswitch-users] The problem with the TURN server and
	connection.
In-Reply-To: <CAAyX+KFmuLQVRkP-Au5cN_Tu5aLD0377r8fwrxx3+2Q3E+hiCw@mail.gmail.com>
References: <loom.20150722T161450-208@post.gmane.org>
	<778052174.20150723102530@seznam.cz>
	<CAAyX+KFmuLQVRkP-Au5cN_Tu5aLD0377r8fwrxx3+2Q3E+hiCw@mail.gmail.com>
Message-ID: <CALXCt0r4Hk9px9ePti8U68+wHSAdx+WLu-mVytHaOoUGzKhLMw@mail.gmail.com>

Also your "normal" turn server supports tcp.

You must configure it.

Also, you must instruct your clients to use your server.

Also, best is to use it as tcp/tls, and on port 443, so firewalls will
allow traffic as it was https.

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On Jul 23, 2015 1:57 PM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:

> Take a look at coturn, a fork of rfc5766-turn-server, which supports
> RFC6062, the TCP transport for TURN.
>
> On 23 July 2015 at 05:25, Denis Jakovlev <yadenis at seznam.cz> wrote:
>
>>  Hi All,
>>
>> I have one problem.
>> One of my clients is behind the serious firewall and they are banned UDP
>> connection at all. His clients are connected to it via the Internet.
>> I put my own Turn/Stun server (rfc5766-turn-server from Google). It works
>> the same server as the freeswitch. It works fine.
>> The problem follows. When all banned UDP connection - you can not connect
>> (INCOMPATIBLE_DESTINATION). If you allow outgoing UDP At least, that is
>> connected with no problems.
>> Is it even possible connection if one of the parties completely banned
>> UDP connection?
>>
>> I thought that the problem is just the decides the turn server. But I is
>> not working for me.
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>> <http://mob.tel>. 775-415-382 * <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 yadenis at seznam.cz  Thu Jul 23 16:57:31 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Thu, 23 Jul 2015 14:57:31 +0200
Subject: [Freeswitch-users] The problem with the TURN server and
	connection.
In-Reply-To: <CALXCt0r4Hk9px9ePti8U68+wHSAdx+WLu-mVytHaOoUGzKhLMw@mail.gmail.com>
References: <loom.20150722T161450-208@post.gmane.org>
	<778052174.20150723102530@seznam.cz> 
	<CAAyX+KFmuLQVRkP-Au5cN_Tu5aLD0377r8fwrxx3+2Q3E+hiCw@mail.gmail.com>
	<CALXCt0r4Hk9px9ePti8U68+wHSAdx+WLu-mVytHaOoUGzKhLMw@mail.gmail.com>
Message-ID: <189685625.20150723145731@seznam.cz>

Dobr? den,

My Turn server is configured. Tests runs without problems. (http://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/)
What should I tell clients that they use it? Is not this work automatically? 

Maybe you advise where I can read about it in detail? I confess that I can not find enough information to me was clear.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

?tvrtek 23. ?ervence 2015, 14:32:28, napsal jste:


Also your "normal" turn server supports tcp.
You must configure it.
Also, you must instruct your clients to use your server.
Also, best is to use it as tcp/tls, and on port 443, so firewalls will allow traffic as it was https.
sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On Jul 23, 2015 1:57 PM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
Take a look at coturn, a fork of rfc5766-turn-server, which supports RFC6062, the TCP transport for TURN.

On 23 July 2015 at 05:25, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi All,

I have one problem. 
One of my clients is behind the serious firewall and they are banned UDP connection at all. His clients are connected to it via the Internet. 
I put my own Turn/Stun server (rfc5766-turn-server from Google). It works the same server as the freeswitch. It works fine.
The problem follows. When all banned UDP connection - you can not connect (INCOMPATIBLE_DESTINATION). If you allow outgoing UDP At least, that is connected with no problems.
Is it even possible connection if one of the parties completely banned UDP connection? 

I thought that the problem is just the decides the turn server. But I is not working for me.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

_________________________________________________________________________
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 findmeinwland at gmail.com  Thu Jul 23 10:51:33 2015
From: findmeinwland at gmail.com (Artur Mega)
Date: Thu, 23 Jul 2015 11:51:33 +0500
Subject: [Freeswitch-users] mod_xml_radius starts accounting twice for
	incoming calls
Message-ID: <CAOLEQ7djZo=Gg=L6tF80BcMvhC2wE-ees4yaRvYXNub9BT60AQ@mail.gmail.com>

When new incoming call comes to tr2.xxxxxxx.ru, new session is being
created and accounting begins. Further, call forwarding to another server,
to fs2.xxxxxxx.ru, and new session is being created again.

2015-07-23 11:22:42.514095 [NOTICE] switch_channel.c:1054 *New Channel*
sofia/external/73472460000 at tr2.xxxxxxx.ru:5080 [
26777290306690-93831426613199 at 192.168.217.156]
...
2015-07-23 11:22:42.574084 [INFO] mod_xml_radius.c:1123
mod_xml_radius:  *Accounting
Start success*
2015-07-23 11:22:42.574084 [DEBUG] switch_core_state_machine.c:164
sofia/external/73472460000 at tr2.xxxxxxx.ru:5080 Standard ROUTING
2015-07-23 11:22:42.574084 [INFO] mod_dialplan_xml.c:558 Processing
73472460000 <73472460000>->79373057071 in context public
Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080 parsing
[public->originate_leg] continue=true
Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080 Absolute Condition
[originate_leg]
Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080 Action
export(nolocal:h323-call-origin=originate)
Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080 parsing
[public->public_mult] continue=false
Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080 Regex (PASS)
[public_mult] destination_number(79373057071) =~ /^(79373057071)$/
break=on-false
Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080 Action
set(hangup_after_bridge=true)
Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080 Action
bridge(sofia/internal/79373057071 at fs2.xxxxxxx.ru:5070)
2015-07-23 11:22:42.574084 [DEBUG] switch_core_state_machine.c:214
(sofia/external/73472460000 at tr2.xxxxxxx.ru:5080) State Change CS_ROUTING ->
CS_EXECUTE
...
2015-07-23 11:22:42.574084 [NOTICE] switch_channel.c:1054 *New Channel*
sofia/internal/79373057071 at fs2.xxxxxxx.ru:5070
[e059ad7f-b2f8-4beb-b1f3-249bc431cfc4]
...
2015-07-23 11:22:42.634059 [INFO] mod_xml_radius.c:1123
mod_xml_radius: * Accounting
Start success*

Thus, one call is being counted twice (for incoming to tr2, and then for
outgoing from tr2 to fs2). But we need to make mod_xml_radius counts this
call only once, how can we handle it?
Thanks
-- 

Arthur
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From findmeinwland at gmail.com  Thu Jul 23 14:39:05 2015
From: findmeinwland at gmail.com (Artur Mega)
Date: Thu, 23 Jul 2015 15:39:05 +0500
Subject: [Freeswitch-users] (no subject)
Message-ID: <CAOLEQ7cWecMp+s22JCeYLK_h3-=gUBmj1La8Fns30=J0qabcyA@mail.gmail.com>

When new incoming call comes to tr2.xxxxxxx.ru, new session is being
created and accounting begins. Further, call forwarding to another server,
to fs2.xxxxxxx.ru, and new session is being created again.

2015-07-23 11:22:42.514095 [NOTICE] switch_channel.c:1054 *New Channel*
 sofia/external/73472460000 at tr2.xxxxxxx.ru:5080 [
26777290306690-93831426613199 at 192.168.217.156]
...
2015-07-23 11:22:42.574084 [INFO] mod_xml_radius.c:1123
mod_xml_radius:  *Accounting
Start success*
2015-07-23 11:22:42.574084 [DEBUG] switch_core_state_machine.c:164
sofia/external/73472460000 at tr2.xxxxxxx.ru:5080 Standard ROUTING
2015-07-23 11:22:42.574084 [INFO] mod_dialplan_xml.c:558 Processing
73472460000 <73472460000>->79373057071 in context public
Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080 parsing
[public->originate_leg] continue=true
Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080 Absolute Condition
[originate_leg]
Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080 Action
export(nolocal:h323-call-origin=originate)
Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080 parsing
[public->public_mult] continue=false
Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080 Regex (PASS)
[public_mult] destination_number(79373057071) =~ /^(79373057071)$/
break=on-false
Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080 Action
set(hangup_after_bridge=true)
Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080 Action
bridge(sofia/internal/79373057071 at fs2.xxxxxxx.ru:5070)
2015-07-23 11:22:42.574084 [DEBUG] switch_core_state_machine.c:214
(sofia/external/73472460000 at tr2.xxxxxxx.ru:5080) State Change CS_ROUTING ->
CS_EXECUTE
...
2015-07-23 11:22:42.574084 [NOTICE] switch_channel.c:1054 *New Channel*
 sofia/internal/79373057071 at fs2.xxxxxxx.ru:5070
 [e059ad7f-b2f8-4beb-b1f3-249bc431cfc4]
...
2015-07-23 11:22:42.634059 [INFO] mod_xml_radius.c:1123
mod_xml_radius: * Accounting
Start success*

Thus, one call is being counted twice (for incoming to tr2, and then for
outgoing from tr2 to fs2). But we need to make mod_xml_radius counts this
call only once, how can we handle it?
Thanks
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From sergio.garcia at quobis.com  Thu Jul 23 15:17:56 2015
From: sergio.garcia at quobis.com (=?ISO-8859-1?Q?Sergio_Garc=EDa?=)
Date: Thu, 23 Jul 2015 13:17:56 +0200
Subject: [Freeswitch-users] ReINVITE - 488 Not Acceptable Here
Message-ID: <CAMxfXLpU8+ku1KfLFB7-U-p5UC_s1K+g4zQQuuk-nr4+ZQqhuw@mail.gmail.com>

Hello all,

I am using *FreeSwitch* as a *WebRTC gateway* thanks to its Websocket
support (in my case WSS), but I'm facing this strange problem. Audio and
Video calls are working perfectly fine, but when I try to set a call On
Hold, FreeSwitch replies with "488 Not Acceptable Here" error to the
ReINVITE I'm sending.

The only error I can see in the logs is:


* [ERR] sofia.c:7280 Reinvite Codec Error!*

The only difference between the original INVITE and this ReINVITE is that I
try to set IP address to *0.0.0.0*, port to *0* and media attribute to
*inactive*. I don't understand what part of the SDP, FS doesn't "like".

Attached you can find a more detailed log file.

Thank you very much in advance.

Regards,
-- 

*Sergio Garc?a Ramos *
VoIP Engineer @ Quobis <http://www.quobis.com/> | e:
sergio.garcia at quobis.com | t: +34902999465
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freeswitch at internal> tport.c:2773 tport_wakeup() tport_wakeup(0x2a06f20): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x2a06f20)
tport.c:2296 tport_set_secondary_timer() tport(0x2a06f20): reset timer
tport.c:2773 tport_wakeup() tport_wakeup(0x2a06f20): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x2a06f20)
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x2a06f20) msg 0x2aac4d0 from (wss/1.1.1.1:47606) has 2933 bytes, veclen = 1
recv 2933 bytes from wss/[1.1.1.1]:47606 at 12:26:13.086659:
   ------------------------------------------------------------------------
   INVITE sip:mod_sofia at 2.2.2.2:5060 SIP/2.0
   Via: SIP/2.0/WSS s0puhni5qlgc.invalid;branch=z9hG4bK6674273
   Max-Forwards: 69
   To: <sip:1013 at demo.example.com>;tag=7jUFmHeD0pvXF
   From: <sip:b1encs3k at s0puhni5qlgc.invalid;transport=ws>;tag=hl91vpapon
   Call-ID: 0c13533c-abc8-1233-50b5-00505685769c
   CSeq: 6856 INVITE
   Contact: <sip:b1encs3k at s0puhni5qlgc.invalid;transport=ws>
   Content-Type: application/sdp
   Supported: path, outbound, gruu
   User-Agent: WebRTC Client
   Content-Length: 2428
   
   v=0
   o=- 4384031540968842851 2 IN IP4 127.0.0.1
   s=-
   t=0 0
   a=group:BUNDLE audio video
   a=msid-semantic: WMS XwpcRKYmZydaZ7CgK4sgklgodNEj3584GuB7
   m=audio 0 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
   c=IN IP4 0.0.0.0
   a=rtcp:9 IN IP4 0.0.0.0
   a=ice-ufrag:UcnaZ9epkI/tHZKA
   a=ice-pwd:a4LSyQfKaE0Ew0fMuaqwc6d5
   a=fingerprint:sha-256 28:B0:00:4B:84:71:00:59:71:CE:1F:6C:6B:5C:16:E9:C4:F8:68:EF:74:7E:0D:33:6C:03:C6:22:27:98:AA:76
   a=setup:actpass
   a=mid:audio
   a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
   a=inactive
   a=rtcp-mux
   a=rtpmap:111 opus/48000/2
   a=fmtp:111 minptime=10; useinbandfec=1
   a=rtpmap:103 ISAC/16000
   a=rtpmap:104 ISAC/32000
   a=rtpmap:9 G722/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:106 CN/32000
   a=rtpmap:105 CN/16000
   a=rtpmap:13 CN/8000
   a=rtpmap:126 telephone-event/8000
   a=maxptime:60
   a=ssrc:3092381274 cname:e48ww6lOt72QPsQY
   a=ssrc:3092381274 msid:XwpcRKYmZydaZ7CgK4sgklgodNEj3584GuB7 f5acaa72-5a7e-46b6-a36b-21d993912d02
   a=ssrc:3092381274 mslabel:XwpcRKYmZydaZ7CgK4sgklgodNEj3584GuB7
   a=ssrc:3092381274 label:f5acaa72-5a7e-46b6-a36b-21d993912d02
   m=video 0 RTP/SAVPF 100 116 117 96
   c=IN IP4 0.0.0.0
   a=rtcp:9 IN IP4 0.0.0.0
   a=ice-ufrag:UcnaZ9epkI/tHZKA
   a=ice-pwd:a4LSyQfKaE0Ew0fMuaqwc6d5
   a=fingerprint:sha-256 28:B0:00:4B:84:71:00:59:71:CE:1F:6C:6B:5C:16:E9:C4:F8:68:EF:74:7E:0D:33:6C:03:C6:22:27:98:AA:76
   a=setup:actpass
   a=mid:video
   a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
   a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
   a=extmap:4 urn:3gpp:video-orientation
   a=inactive
   a=rtcp-mux
   a=rtpmap:100 VP8/90000
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli
   a=rtcp-fb:100 goog-remb
   a=rtpmap:116 red/90000
   a=rtpmap:117 ulpfec/90000
   a=rtpmap:96 rtx/90000
   a=fmtp:96 apt=100
   a=ssrc-group:FID 850102287 3335913892
   a=ssrc:850102287 cname:e48ww6lOt72QPsQY
   a=ssrc:850102287 msid:XwpcRKYmZydaZ7CgK4sgklgodNEj3584GuB7 f73531ae-a55f-4d53-8cd6-7d336ec7e02b
   a=ssrc:850102287 mslabel:XwpcRKYmZydaZ7CgK4sgklgodNEj3584GuB7
   a=ssrc:850102287 label:f73531ae-a55f-4d53-8cd6-7d336ec7e02b
   a=ssrc:3335913892 cname:e48ww6lOt72QPsQY
   a=ssrc:3335913892 msid:XwpcRKYmZydaZ7CgK4sgklgodNEj3584GuB7 f73531ae-a55f-4d53-8cd6-7d336ec7e02b
   a=ssrc:3335913892 mslabel:XwpcRKYmZydaZ7CgK4sgklgodNEj3584GuB7
   a=ssrc:3335913892 label:f73531ae-a55f-4d53-8cd6-7d336ec7e02b
   ------------------------------------------------------------------------
tport.c:3023 tport_deliver() tport_deliver(0x2a06f20): msg 0x2aac4d0 (2933 bytes) from wss/1.1.1.1:47606/sips next=(nil)
nta.c:2880 agent_recv_request() nta: received INVITE sip:mod_sofia at 2.2.2.2:5060 SIP/2.0 (CSeq 6856)
nta.c:3174 agent_check_request_via() nta: Via check: received=1.1.1.1
nta.c:3060 agent_recv_request() nta: INVITE (6856) going to existing leg
nta.c:1348 set_timeout() nta: timer shortened to 2000 ms
nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering
soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x27af900) called
soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x27af900, (nil), 0x2a39919, 2428) called
tport.c:3257 tport_tsend() tport_tsend(0x2a06f20) tpn = WSS/1.1.1.1:47606
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x2a07110 0x279b050 116 (116)
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x2a07110 0x2a397d1 71 (71)
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x2a07110 0x2a397a0 49 (49)
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x2a07110 0x2a39818 66 (66)
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x2a07110 0x279b0c4 71 (71)
tport.c:3594 tport_vsend() tport_vsend(0x2a06f20): 373 bytes of 373 to wss/1.1.1.1:47606
tport.c:3492 tport_send_msg() tport_vsend returned 373
send 373 bytes to wss/[1.1.1.1]:47606 at 12:26:13.086909:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/WSS s0puhni5qlgc.invalid;branch=z9hG4bK6674273;received=1.1.1.1;rport=47606
   From: <sip:b1encs3k at s0puhni5qlgc.invalid;transport=ws>;tag=hl91vpapon
   To: <sip:1013 at demo.example.com>;tag=7jUFmHeD0pvXF
   Call-ID: 0c13533c-abc8-1233-50b5-00505685769c
   CSeq: 6856 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.4.20-34~64bit
   Content-Length: 0
   
   ------------------------------------------------------------------------
tport.c:2296 tport_set_secondary_timer() tport(0x2a06f20): reset timer
nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (6856)
nua_stack.c:271 nua_stack_event() nua(0x27af3f0): event i_invite 100 Trying
nua_session.c:4145 signal_call_state_change() nua(0x27af3f0): ready call updated: received received offer
soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x27af900, [0x7f5c0b8555c8], [0x7f5c0b8555d0], [(nil)]) called
nua_stack.c:271 nua_stack_event() nua(0x27af3f0): event i_state 100 Trying
tport.c:2296 tport_set_secondary_timer() tport(0x2a06f20): reset timer
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-07-23 12:26:13.080456 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/b1encs3k at s0puhni5qlgc.invalid [BREAK]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-07-23 12:26:13.080456 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/b1encs3k at s0puhni5qlgc.invalid [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-07-23 12:26:13.100449 [DEBUG] sofia.c:6627 Channel sofia/internal/b1encs3k at s0puhni5qlgc.invalid entering state [received][100]
2015-07-23 12:26:13.100449 [DEBUG] sofia.c:6637 Remote SDP:
v=0
o=- 4384031540968842851 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS XwpcRKYmZydaZ7CgK4sgklgodNEj3584GuB7
m=audio 0 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 0.0.0.0
m=video 0 RTP/SAVPF 100 116 117 96
c=IN IP4 0.0.0.0

2015-07-23 12:26:13.100449 [INFO] switch_core_media.c:3047 Activating audio RTCP PORT 38567
2015-07-23 12:26:13.100449 [DEBUG] switch_rtp.c:3919 RTCP send rate is: 10000 and packet rate is: 20000 Remote Port: 38567
2015-07-23 12:26:13.100449 [DEBUG] switch_rtp.c:2349 Setting RTCP remote addr to 2.2.2.3:38567
2015-07-23 12:26:13.100449 [INFO] switch_core_media.c:3047 Activating video RTCP PORT 39403
2015-07-23 12:26:13.100449 [DEBUG] switch_rtp.c:3919 RTCP send rate is: 10000 and packet rate is: 90000 Remote Port: 39403
2015-07-23 12:26:13.100449 [DEBUG] switch_rtp.c:2349 Setting RTCP remote addr to 2.2.2.3:39403
2015-07-23 12:26:13.100449 [ERR] sofia.c:7280 Reinvite Codec Error!
nua.c:879 nua_respond() nua: nua_respond: entering
nua_stack.c:573 nua_stack_signal() nua(0x27af3f0): recv signal r_respond 488 Not Acceptable Here
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x27af900, ...) called
nua_session.c:2320 nua_invite_server_respond() nua: nua_invite_server_respond: entering
soa.c:1214 soa_clear_remote_sdp() soa_clear_remote_sdp(static::0x27af900) called
tport.c:3257 tport_tsend() tport_tsend(0x2a06f20) tpn = WSS/1.1.1.1:47606
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x2a07110 0x27086f0 129 (129)
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x2a07110 0x2a397d1 71 (71)
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x2a07110 0x2a397a0 49 (49)
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x2a07110 0x2a39818 66 (66)
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x2a07110 0x2708771 213 (213)
tport.c:3594 tport_vsend() tport_vsend(0x2a06f20): 528 bytes of 528 to wss/1.1.1.1:47606
tport.c:3492 tport_send_msg() tport_vsend returned 528
send 528 bytes to wss/[1.1.1.1]:47606 at 12:26:13.104835:
   ------------------------------------------------------------------------
   SIP/2.0 488 Not Acceptable Here
   Via: SIP/2.0/WSS s0puhni5qlgc.invalid;branch=z9hG4bK6674273;received=1.1.1.1;rport=47606
   From: <sip:b1encs3k at s0puhni5qlgc.invalid;transport=ws>;tag=hl91vpapon
   To: <sip:1013 at demo.example.com>;tag=7jUFmHeD0pvXF
   Call-ID: 0c13533c-abc8-1233-50b5-00505685769c
   CSeq: 6856 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.4.20-34~64bit
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
   Supported: path, replaces
   Content-Length: 0
   
   ------------------------------------------------------------------------
tport.c:2296 tport_set_secondary_timer() tport(0x2a06f20): reset timer
nta.c:6791 incoming_reply() nta: sent 488 Not Acceptable Here for INVITE (6856)
nua_session.c:4145 signal_call_state_change() nua(0x27af3f0): ready call updated: init
nua_stack.c:271 nua_stack_event() nua(0x27af3f0): event i_state 488 Not Acceptable Here
nua_stack.c:271 nua_stack_event() nua(0x27af3f0): event i_active 488 Call active
soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x27af900) called
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-07-23 12:26:13.100449 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/b1encs3k at s0puhni5qlgc.invalid [BREAK]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-07-23 12:26:13.100449 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/b1encs3k at s0puhni5qlgc.invalid [BREAK]
nua_stack.c:529 nua_signal() nua(0x27af3f0): sent signal r_respond
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-07-23 12:26:13.100449 [DEBUG] sofia.c:6627 Channel sofia/internal/b1encs3k at s0puhni5qlgc.invalid entering state [ready][488]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
tport.c:2773 tport_wakeup() tport_wakeup(0x2a06f20): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x2a06f20)
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x2a06f20) msg 0x2673c20 from (wss/1.1.1.1:47606) has 292 bytes, veclen = 1
recv 292 bytes from wss/[1.1.1.1]:47606 at 12:26:13.145877:
   ------------------------------------------------------------------------
   ACK sip:mod_sofia at 2.2.2.2:5060 SIP/2.0
   Via: SIP/2.0/WSS s0puhni5qlgc.invalid;branch=z9hG4bK6674273
   To: <sip:1013 at demo.example.com>;tag=7jUFmHeD0pvXF
   From: <sip:b1encs3k at s0puhni5qlgc.invalid;transport=ws>;tag=hl91vpapon
   Call-ID: 0c13533c-abc8-1233-50b5-00505685769c
   CSeq: 6856 ACK
   
   ------------------------------------------------------------------------
tport.c:3023 tport_deliver() tport_deliver(0x2a06f20): bad msg 0x2673c20 (292 bytes) from wss/1.1.1.1:47606/sips next=(nil)
nta.c:2880 agent_recv_request() nta: received ACK sip:mod_sofia at 2.2.2.2:5060 SIP/2.0 (CSeq 6856)
tport.c:2296 tport_set_secondary_timer() tport(0x2a06f20): reset timer
nta.c:1296 agent_timer() nta: timer set next to 14116 ms
/exit

From gmaruzz at gmail.com  Thu Jul 23 17:06:40 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 23 Jul 2015 15:06:40 +0200
Subject: [Freeswitch-users] The problem with the TURN server and
	connection.
In-Reply-To: <189685625.20150723145731@seznam.cz>
References: <loom.20150722T161450-208@post.gmane.org>
	<778052174.20150723102530@seznam.cz>
	<CAAyX+KFmuLQVRkP-Au5cN_Tu5aLD0377r8fwrxx3+2Q3E+hiCw@mail.gmail.com>
	<CALXCt0r4Hk9px9ePti8U68+wHSAdx+WLu-mVytHaOoUGzKhLMw@mail.gmail.com>
	<189685625.20150723145731@seznam.cz>
Message-ID: <CALXCt0p-O1ze9zuhLinQHSmVNrSxcqymxmWKRsQQBwi2hhrPTw@mail.gmail.com>

you must tell your clients which stun/turn server(s) to use.
they often default to use google's stun server.
You must write in your clients config which servers to use...

What are you using as clients?


On Thu, Jul 23, 2015 at 2:57 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Dobr? den,
>
> My Turn server is configured. Tests runs without problems. (
> http://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/)
> What should I tell clients that they use it? Is not this work
> automatically?
>
> Maybe you advise where I can read about it in detail? I confess that I can
> not find enough information to me was clear.
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 ?tvrtek 23. ?ervence 2015, 14:32:28, napsal
> jste: *
>  Also your "normal" turn server supports tcp.
> You must configure it.
> Also, you must instruct your clients to use your server.
> Also, best is to use it as tcp/tls, and on port 443, so firewalls will
> allow traffic as it was https.
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On Jul 23, 2015 1:57 PM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
> Take a look at coturn, a fork of rfc5766-turn-server, which supports
> RFC6062, the TCP transport for TURN.
>
> On 23 July 2015 at 05:25, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Hi All,
>
> I have one problem.
> One of my clients is behind the serious firewall and they are banned UDP
> connection at all. His clients are connected to it via the Internet.
> I put my own Turn/Stun server (rfc5766-turn-server from Google). It works
> the same server as the freeswitch. It works fine.
> The problem follows. When all banned UDP connection - you can not connect
> (INCOMPATIBLE_DESTINATION). If you allow outgoing UDP At least, that is
> connected with no problems.
> Is it even possible connection if one of the parties completely banned UDP
> connection?
>
> I thought that the problem is just the decides the turn server. But I is
> not working for me.
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
> *. 775-415-382 *
> _________________________________________________________________________
> 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
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From rtreleaven at bunnykick.ca  Thu Jul 23 17:16:54 2015
From: rtreleaven at bunnykick.ca (Russell Treleaven)
Date: Thu, 23 Jul 2015 09:16:54 -0400
Subject: [Freeswitch-users] The problem with the TURN server and
	connection.
In-Reply-To: <CALXCt0p-O1ze9zuhLinQHSmVNrSxcqymxmWKRsQQBwi2hhrPTw@mail.gmail.com>
References: <loom.20150722T161450-208@post.gmane.org>
	<778052174.20150723102530@seznam.cz>
	<CAAyX+KFmuLQVRkP-Au5cN_Tu5aLD0377r8fwrxx3+2Q3E+hiCw@mail.gmail.com>
	<CALXCt0r4Hk9px9ePti8U68+wHSAdx+WLu-mVytHaOoUGzKhLMw@mail.gmail.com>
	<189685625.20150723145731@seznam.cz>
	<CALXCt0p-O1ze9zuhLinQHSmVNrSxcqymxmWKRsQQBwi2hhrPTw@mail.gmail.com>
Message-ID: <CAHzfiZ4v+4EmB1=Eh5hyb80DhAsyr7dpU6ZYY5LWMkjGCywE8w@mail.gmail.com>

How about finding out why the firewall blocks all udp?
If they don't have a compelling reason to block it maybe they will loosen
the policy.
I a curious to know why they are blocking udp.


On Thu, Jul 23, 2015 at 9:06 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> you must tell your clients which stun/turn server(s) to use.
> they often default to use google's stun server.
> You must write in your clients config which servers to use...
>
> What are you using as clients?
>
>
> On Thu, Jul 23, 2015 at 2:57 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
>
>>  Dobr? den,
>>
>> My Turn server is configured. Tests runs without problems. (
>> http://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/)
>> What should I tell clients that they use it? Is not this work
>> automatically?
>>
>> Maybe you advise where I can read about it in detail? I confess that I
>> can not find enough information to me was clear.
>>
>>
>>
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>> <http://mob.tel>. 775-415-382 ?tvrtek 23. ?ervence 2015, 14:32:28, napsal
>> jste: *
>>  Also your "normal" turn server supports tcp.
>> You must configure it.
>> Also, you must instruct your clients to use your server.
>> Also, best is to use it as tcp/tls, and on port 443, so firewalls will
>> allow traffic as it was https.
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> On Jul 23, 2015 1:57 PM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
>> Take a look at coturn, a fork of rfc5766-turn-server, which supports
>> RFC6062, the TCP transport for TURN.
>>
>> On 23 July 2015 at 05:25, Denis Jakovlev <yadenis at seznam.cz> wrote:
>> Hi All,
>>
>> I have one problem.
>> One of my clients is behind the serious firewall and they are banned UDP
>> connection at all. His clients are connected to it via the Internet.
>> I put my own Turn/Stun server (rfc5766-turn-server from Google). It works
>> the same server as the freeswitch. It works fine.
>> The problem follows. When all banned UDP connection - you can not connect
>> (INCOMPATIBLE_DESTINATION). If you allow outgoing UDP At least, that is
>> connected with no problems.
>> Is it even possible connection if one of the parties completely banned
>> UDP connection?
>>
>> I thought that the problem is just the decides the turn server. But I is
>> not working for me.
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>>
>> *. 775-415-382 *
>> _________________________________________________________________________
>> 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
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> 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 yadenis at seznam.cz  Thu Jul 23 17:20:52 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Thu, 23 Jul 2015 15:20:52 +0200
Subject: [Freeswitch-users] The problem with the TURN server and
	connection.
In-Reply-To: <CALXCt0p-O1ze9zuhLinQHSmVNrSxcqymxmWKRsQQBwi2hhrPTw@mail.gmail.com>
References: <loom.20150722T161450-208@post.gmane.org>
	<778052174.20150723102530@seznam.cz> 
	<CAAyX+KFmuLQVRkP-Au5cN_Tu5aLD0377r8fwrxx3+2Q3E+hiCw@mail.gmail.com>
	<CALXCt0r4Hk9px9ePti8U68+wHSAdx+WLu-mVytHaOoUGzKhLMw@mail.gmail.com>
	<189685625.20150723145731@seznam.cz>
	<CALXCt0p-O1ze9zuhLinQHSmVNrSxcqymxmWKRsQQBwi2hhrPTw@mail.gmail.com>
Message-ID: <101431786.20150723152052@seznam.cz>

Hi,

As a client, I use jssip or sip.js. 
There is, of course, the turn the server is registered


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

?tvrtek 23. ?ervence 2015, 15:06:40, napsal jste:


you must tell your clients which stun/turn server(s) to use.
they often default to use google's stun server.
You must write in your clients config which servers to use...

What are you using as clients?


On Thu, Jul 23, 2015 at 2:57 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
Dobr? den,

My Turn server is configured. Tests runs without problems. (http://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/)
What should I tell clients that they use it? Is not this work automatically? 

Maybe you advise where I can read about it in detail? I confess that I can not find enough information to me was clear.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

?tvrtek 23. ?ervence 2015, 14:32:28, napsal jste:


Also your "normal" turn server supports tcp.
You must configure it.
Also, you must instruct your clients to use your server.
Also, best is to use it as tcp/tls, and on port 443, so firewalls will allow traffic as it was https.
sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On Jul 23, 2015 1:57 PM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
Take a look at coturn, a fork of rfc5766-turn-server, which supports RFC6062, the TCP transport for TURN.

On 23 July 2015 at 05:25, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi All,

I have one problem. 
One of my clients is behind the serious firewall and they are banned UDP connection at all. His clients are connected to it via the Internet. 
I put my own Turn/Stun server (rfc5766-turn-server from Google). It works the same server as the freeswitch. It works fine.
The problem follows. When all banned UDP connection - you can not connect (INCOMPATIBLE_DESTINATION). If you allow outgoing UDP At least, that is connected with no problems.
Is it even possible connection if one of the parties completely banned UDP connection? 

I thought that the problem is just the decides the turn server. But I is not working for me.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

_________________________________________________________________________
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



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From yadenis at seznam.cz  Thu Jul 23 17:24:12 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Thu, 23 Jul 2015 15:24:12 +0200
Subject: [Freeswitch-users] The problem with the TURN server and
	connection.
In-Reply-To: <CAAyX+KFmuLQVRkP-Au5cN_Tu5aLD0377r8fwrxx3+2Q3E+hiCw@mail.gmail.com>
References: <loom.20150722T161450-208@post.gmane.org>
	<778052174.20150723102530@seznam.cz> 
	<CAAyX+KFmuLQVRkP-Au5cN_Tu5aLD0377r8fwrxx3+2Q3E+hiCw@mail.gmail.com>
Message-ID: <762006319.20150723152412@seznam.cz>

Dobr? den,

That's it I use

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

?tvrtek 23. ?ervence 2015, 13:56:43, napsal jste:


Take a look at coturn, a fork of rfc5766-turn-server, which supports RFC6062, the TCP transport for TURN.

On 23 July 2015 at 05:25, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi All,

I have one problem. 
One of my clients is behind the serious firewall and they are banned UDP connection at all. His clients are connected to it via the Internet. 
I put my own Turn/Stun server (rfc5766-turn-server from Google). It works the same server as the freeswitch. It works fine.
The problem follows. When all banned UDP connection - you can not connect (INCOMPATIBLE_DESTINATION). If you allow outgoing UDP At least, that is connected with no problems.
Is it even possible connection if one of the parties completely banned UDP connection? 

I thought that the problem is just the decides the turn server. But I is not working for me.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

_________________________________________________________________________
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 Jul 23 17:25:35 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 23 Jul 2015 15:25:35 +0200
Subject: [Freeswitch-users] The problem with the TURN server and
	connection.
In-Reply-To: <101431786.20150723152052@seznam.cz>
References: <loom.20150722T161450-208@post.gmane.org>
	<778052174.20150723102530@seznam.cz>
	<CAAyX+KFmuLQVRkP-Au5cN_Tu5aLD0377r8fwrxx3+2Q3E+hiCw@mail.gmail.com>
	<CALXCt0r4Hk9px9ePti8U68+wHSAdx+WLu-mVytHaOoUGzKhLMw@mail.gmail.com>
	<189685625.20150723145731@seznam.cz>
	<CALXCt0p-O1ze9zuhLinQHSmVNrSxcqymxmWKRsQQBwi2hhrPTw@mail.gmail.com>
	<101431786.20150723152052@seznam.cz>
Message-ID: <CALXCt0oNz2yh-X6vrkDPE4Rj_pOnrozezO=_uCvqBdK-OPW9FQ@mail.gmail.com>

in both jssip and sip.js you can configure stun and turn to use

eg:

http://sipjs.com/api/0.5.0/ua_configuration_parameters/#turnservers

http://sipjs.com/api/0.5.0/ua_configuration_parameters/#stunservers



On Thu, Jul 23, 2015 at 3:20 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Hi,
>
> As a client, I use jssip or sip.js.
> There is, of course, the turn the server is registered
>
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 ?tvrtek 23. ?ervence 2015, 15:06:40, napsal
> jste: *
>
>  you must tell your clients which stun/turn server(s) to use.
> they often default to use google's stun server.
> You must write in your clients config which servers to use...
>
> What are you using as clients?
>
>
> On Thu, Jul 23, 2015 at 2:57 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Dobr? den,
>
> My Turn server is configured. Tests runs without problems. (
> http://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/)
> What should I tell clients that they use it? Is not this work
> automatically?
>
> Maybe you advise where I can read about it in detail? I confess that I can
> not find enough information to me was clear.
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
>
>
> *. 775-415-382 ?tvrtek 23. ?ervence 2015, 14:32:28, napsal jste: *
>  Also your "normal" turn server supports tcp.
> You must configure it.
> Also, you must instruct your clients to use your server.
> Also, best is to use it as tcp/tls, and on port 443, so firewalls will
> allow traffic as it was https.
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On Jul 23, 2015 1:57 PM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
> Take a look at coturn, a fork of rfc5766-turn-server, which supports
> RFC6062, the TCP transport for TURN.
>
> On 23 July 2015 at 05:25, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Hi All,
>
> I have one problem.
> One of my clients is behind the serious firewall and they are banned UDP
> connection at all. His clients are connected to it via the Internet.
> I put my own Turn/Stun server (rfc5766-turn-server from Google). It works
> the same server as the freeswitch. It works fine.
> The problem follows. When all banned UDP connection - you can not connect
> (INCOMPATIBLE_DESTINATION). If you allow outgoing UDP At least, that is
> connected with no problems.
> Is it even possible connection if one of the parties completely banned UDP
> connection?
>
> I thought that the problem is just the decides the turn server. But I is
> not working for me.
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
> *. 775-415-382 *
> _________________________________________________________________________
> 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
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> 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
Cell : +39-347-2665618
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From gmaruzz at gmail.com  Thu Jul 23 17:28:45 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 23 Jul 2015 15:28:45 +0200
Subject: [Freeswitch-users] The problem with the TURN server and
	connection.
In-Reply-To: <762006319.20150723152412@seznam.cz>
References: <loom.20150722T161450-208@post.gmane.org>
	<778052174.20150723102530@seznam.cz>
	<CAAyX+KFmuLQVRkP-Au5cN_Tu5aLD0377r8fwrxx3+2Q3E+hiCw@mail.gmail.com>
	<762006319.20150723152412@seznam.cz>
Message-ID: <CALXCt0qrhSJ=gHmnwu2pBsNWwnKD_447S1_JL0UBH2fK2KzWgQ@mail.gmail.com>

pay attention, you MUST use the "transport" argument for tcp to work, eg:

turnServers: {
  urls:"turn:exam.org:443?transport=tcp",
  username:"alice",
  password:"racecar"
}



On Thu, Jul 23, 2015 at 3:24 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Dobr? den,
>
> That's it I use
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 ?tvrtek 23. ?ervence 2015, 13:56:43, napsal
> jste: *
>  Take a look at coturn, a fork of rfc5766-turn-server, which supports
> RFC6062, the TCP transport for TURN.
>
> On 23 July 2015 at 05:25, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Hi All,
>
> I have one problem.
> One of my clients is behind the serious firewall and they are banned UDP
> connection at all. His clients are connected to it via the Internet.
> I put my own Turn/Stun server (rfc5766-turn-server from Google). It works
> the same server as the freeswitch. It works fine.
> The problem follows. When all banned UDP connection - you can not connect
> (INCOMPATIBLE_DESTINATION). If you allow outgoing UDP At least, that is
> connected with no problems.
> Is it even possible connection if one of the parties completely banned UDP
> connection?
>
> I thought that the problem is just the decides the turn server. But I is
> not working for me.
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
> *. 775-415-382 *
> _________________________________________________________________________
> 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
Cell : +39-347-2665618
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From jaflong at yandex.com  Thu Jul 23 18:56:53 2015
From: jaflong at yandex.com (jaflong jaflong)
Date: Thu, 23 Jul 2015 17:56:53 +0300
Subject: [Freeswitch-users] I have a php script run by ivrd,
	is it possible to subscribe to events on this outbound eslconnection
In-Reply-To: <1114761437578245@web12g.yandex.ru>
References: <1744091437576931@web13g.yandex.ru>
	<CAEJMVkB-9JUrgfo7fqm-KdfNFb4oYNJm9cwJkGxNefu+L-Uh0w@mail.gmail.com>
	<1114761437578245@web12g.yandex.ru>
Message-ID: <255331437663413@web18g.yandex.ru>

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From danny.gershman at gmail.com  Thu Jul 23 19:07:36 2015
From: danny.gershman at gmail.com (Danny Gershman)
Date: Thu, 23 Jul 2015 15:07:36 +0000
Subject: [Freeswitch-users] Endless playback in conference
In-Reply-To: <01ec01d0c4cd$82fd1bf0$88f753d0$@botecomm.com>
References: <CAEorZynEWmqnOAEfY00_k1Rwe6jPjsjNSPxJjVaTq_neOMFozg@mail.gmail.com>
	<01ec01d0c4cd$82fd1bf0$88f753d0$@botecomm.com>
Message-ID: <CAEorZyn0TGWO9nd8Y4hBpWGPLFnag6k6dVSd7dYZCiCcCPzv8A@mail.gmail.com>

Wouldn't a profile have to be created on the fly for this? From what I can
see you cannot set this for a conference from an api call. Also to stop it,
you would have to change the profile for the conference to remove it.
On Wed, Jul 22, 2015 at 6:28 PM Bote Man <bote_radio at botecomm.com> wrote:

> There?s a parameter for mod_conference named ?perpetual-sound? that looks
> like it would do the trick.
>
>
>
> It?s about 1/3 of the way down
>
> https://freeswitch.org/confluence/display/FREESWITCH/mod_conference
>
>
>
> PLEASE check further for any changes that might have been made in the
> latest FreeSWITCH as the conference module has undergone substantial
> changes and perpetual-sound might have been one of them.
>
>
>
> Bote
>
>
>
>
>
>
>
>
>
> *From:* Danny Gershman
> *Sent:* Wednesday, 22 July, 2015 17:09
> *Subject:* [Freeswitch-users] Endless playback in conference
>
>
>
> I'm trying to do an endless playback of an mp3 file in a conference.   I
> have a couple of ideas, but none seem really solid.
>
>
>
> 1) Pass a variable on play and monitor from mod_event_socket and play
> again if not forcibly terminated.
>
>
>
> 2) Load up local_stream dynamically from an xmlhttp server, and then
> restart the local stream service, however will interrupt MOH for other
> users.
>
>
>
> Any other ideas?  Is there a way to do looping for "api" through
> mod_event_socket?  I know you can with "sendmsg"
>
>
>
> --Danny Gershman
> _________________________________________________________________________
> 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.org  Thu Jul 23 19:09:02 2015
From: brian at freeswitch.org (Brian West)
Date: Thu, 23 Jul 2015 10:09:02 -0500
Subject: [Freeswitch-users] mod_local_stream moh confusion
In-Reply-To: <2E67ADAE1D3582409C90EBAE8C64C5070E5569@QSCDEMXP01B.ONE4ALL.LAN>
References: <2E67ADAE1D3582409C90EBAE8C64C5070E5569@QSCDEMXP01B.ONE4ALL.LAN>
Message-ID: <CAEJMVkAnYcxOzL19E_Zs9Bn73zx8gAtGuTCYGZPrqsFhVb6n3g@mail.gmail.com>

It would indicate that you do not have mod_local_stream loaded.

On Thu, Jul 23, 2015 at 3:15 AM, Lappe, Adam <adam.lappe at qsc.de> wrote:

>  Hello,
>
>
>
> i am testing the latest 1.4.19 Version of FreeSWITCH.
>
> Currently we are running an old 1.2.7 Version.
>
>
>
> Everything seems to work fine, but there is 1 error that is very confusing:
>
>
>
> When a call gets transfered by the callee (i.e. by the receptionist) the
> call will be terminated.
>
> All I see is this error line:
>
> *[ERR] switch_core_file.c:149 Invalid file format [local_stream] for
> [moh]!*
>
>
>
> I don?t use the local_stream module.
>
> freeswitch at internal> module_exists mod_local_stream
>
> false
>
>
>
> This error does not exists with the old version.
>
>
>
> Is this a bug, or am I missing something?
>
>
>
> Thanks in advance,
>
> Adam
>
> _________________________________________________________________________
> 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
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From brian at freeswitch.org  Thu Jul 23 19:09:40 2015
From: brian at freeswitch.org (Brian West)
Date: Thu, 23 Jul 2015 10:09:40 -0500
Subject: [Freeswitch-users] I have a php script run by ivrd,
 is it possible to subscribe to events on this outbound eslconnection
In-Reply-To: <1181481437646343@web10h.yandex.ru>
References: <1744091437576931@web13g.yandex.ru>
	<CAEJMVkB-9JUrgfo7fqm-KdfNFb4oYNJm9cwJkGxNefu+L-Uh0w@mail.gmail.com>
	<1114761437578245@web12g.yandex.ru>
	<CAEJMVkAVTycst4C4VQxv7g5BakoRAtvFQMuseoGRC3mq=Vdqnw@mail.gmail.com>
	<1181481437646343@web10h.yandex.ru>
Message-ID: <CAEJMVkCLd7WNqaLJoOk=MOnP4Bb6Ouu1cd5ix7KZwZObt+VhLw@mail.gmail.com>

on outbound you would use myevents.  See confluence on mod_event_socket.

On Thu, Jul 23, 2015 at 5:12 AM, jaflong jaflong <jaflong at yandex.com> wrote:

>
>
>
> On a inbound connection connecting with user name and password, (the code
> example below - top) according to the tcpdump the command
> $conn->events("plain", "ALL"); sends
> "event plain ALL"
> and is accepted and works.
>
> However using the same command on outbound (the code example below -
> bottom) the response is error
>
> Why does the same command work on esl inbound but not esl outbound?
>
>
>
> INBOUND
>
> <?php
>
> require_once('ESL.php');
>
> $conn = new ESLconnection('127.0.0.1', '8021', 'ClueCon');
>
> while ($conn->connected() == 1) {
>
>     $conn->events("plain", "ALL");
>
>     $e = $conn->recvEvent();
>
>     if ($e) {
>
>     } else {
>
>     }
>
> }
> ?>
>
> tcpdump
>
> Content-Type: auth/request
> auth ClueCon
> Content-Type: command/reply
> Reply-Text: +OK accepted
> event plain ALL
> Content-Type: command/reply
> Reply-Text: +OK event listener enabled plain
>
>
> ----------------------------------------------------------------
>
> OUTBOUND
>
> <?php
>
> require_once('ESL.php');
>
> $conn = new ESLconnection(0);
>
> while ($conn->connected() == 1) {
>
>     $conn->events("plain", "ALL");
>
>     $e = $conn->recvEvent();
>
>     if ($e) {
>
>     } else {
>
>     }
> }
> ?>
>
>
> tcpdump
>
> event plain ALL
> Content-Type: command/reply
> Reply-Text: -ERR command not found
>
>
>
> 23.07.2015, 05:06, "Brian West" <brian at freeswitch.org>:
>
> try 'event*s* all plain;
>
> On Wed, Jul 22, 2015 at 10:17 AM, jaflong jaflong <jaflong at yandex.com>
> wrote:
>
>
>
> I am usingoutbount event socket but cannot get it to work
>
> -----------------------------------
>
> require_once('ESL.php');
>
> $audio_location = "/usr/local/freeswitch/sounds/";
> $music = $audio_location . "demo.wav";
>
> $conn = new ESLconnection(0);
>
> $conn->execute("answer");
>
> $conn->events("plain","all");
>
>
> while ($conn->connected() == 1) {
>
>     $e = $conn->recvEvent();
>     if ($e) {
>         $arrEvent = eventParse($e);
>
>     } else {
>
>
>     }
>
> }
>
> -----------------------------------
>
> This does not work, give response
>
> event plain all
>
> Content-Type: command/reply
> Reply-Text: -ERR command not found
>
>
> 22.07.2015, 18:00, "Brian West" <brian at freeswitch.org>:
>
> You may wish to try using outbound event socket if thats what you're
> wising to accomplish.
>
> On Wed, Jul 22, 2015 at 9:55 AM, jaflong jaflong <jaflong at yandex.com>
> wrote:
>
> I have a php script run by ivrd, is it possible to subscribe to
> events on this outbound eslconnection
>
> _________________________________________________________________________
> 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
> ,
>
> _________________________________________________________________________
> 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*
> 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
> ,
>
> _________________________________________________________________________
> 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*
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
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From mike at jerris.com  Thu Jul 23 19:14:33 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 23 Jul 2015 11:14:33 -0400
Subject: [Freeswitch-users] Build ERROR VS 2013
In-Reply-To: <B60DA91F5EA0E24A96E73CDE43CC9EC8517C4390@vc4bex2.corp.c4b.de>
References: <B60DA91F5EA0E24A96E73CDE43CC9EC8517C428C@vc4bex2.corp.c4b.de>
	<CAEJMVkCxH6QsEYDOA28Zf-xc+Uk_kc0Pp7pp0JteEJYpcW2AWA@mail.gmail.com>
	<CAC4JBEB2FzhEwN5kq9wJ5mb2tpkSTD=-0Q7NyNMLty0-xyvfdQ@mail.gmail.com>
	<B60DA91F5EA0E24A96E73CDE43CC9EC8517C4390@vc4bex2.corp.c4b.de>
Message-ID: <CAEwTsAhW+e4yGiJ84n5EMTm+tvKOZFwZgTG_8_cFdFZTF+F+OQ@mail.gmail.com>

I'm still waiting on a properly formatted pull request to fix this issue.

On Thursday, July 23, 2015, Alexander Haugg <Alexander.Haugg at c4b.de> wrote:

>  Thanks for the answers!
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>]
> *Im Auftrag von *Duvid Rottenberg
> *Gesendet:* Mittwoch, 22. Juli 2015 17:27
> *An:* FreeSWITCH Users Help
> *Betreff:* Re: [Freeswitch-users] Build ERROR VS 2013
>
>
>
> The problem is that there some modules that use features from C99, the one
> I noticed is roundf. Visual Studio 2012 didn't support C99. Even if you
> open a VS 2012 project in VS 2013 it will by default use the VS 2012
> toolset which links to the older version of the C libraries.
>
>
>
> I was able to build it on my machine by converting it to use the Visual
> Studio 2013 toolset. There is a way to automatically apply it to all
> projects in a solution, I don't remember exactly how I did it. I have it on
> my todo list to create new 2013 solution & project files, but I have not
> gotten around to it.
>
>
>
> Thanks,
>
> Duvid Rottenberg
>
>
>
> On Wed, Jul 22, 2015 at 9:50 AM, Brian West <brian at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','brian at freeswitch.org');>> wrote:
>
>  Please see https://freeswitch.org/jira/browse/FS-7644
>
>
>
> master currently isn't able to build on Windows, work is being done to
> accomplish this.
>
>
>
> In the future please search thru JIRA, if the issue doesn't appear to be
> reported then please file an issue so we can track it properly moving
> forward.
>
>
>
> Thanks,
>
>
>
> On Wed, Jul 22, 2015 at 8:43 AM, Alexander Haugg <Alexander.Haugg at c4b.de
> <javascript:_e(%7B%7D,'cvml','Alexander.Haugg at c4b.de');>> wrote:
>
>   Hi,
>
>
>
> At the moment they?r bild errors for mod_conference, mod_shout, mod_rtmp
> and mod_verto. The same errors on ?Relese? or ?Debug? build for ?win32? and
> ?x64?.
>
> And the ?libspeex? project need the ?/FS? compiler flag!
>
>
>
> Freeswitch version is from today:
>
> ? git rev-parse HEAD
>
> 17f8002936cb1797dc8bd46b4bad52800ebfb5cb
>
> ?  git describe
>
> v1.5.final-1094-g17f8002
>
>
>
>
>
> mod_conference:
>
> ==============
>
> Error      4             error C1083: Cannot open include file:
> 'mod_conference.h': No such file or directory
> D:\tmp\freeswitch\src\mod\applications\mod_conference\mod_conference.c
> 42           1                mod_conference
>
>
>
> mod_shout:
>
> ==========
>
> Error      4             error C1083: Cannot open include file:
> 'mpg123.h': No such file or directory
> D:\tmp\freeswitch\src\mod\formats\mod_shout\mod_shout.c
> 36           1             mod_shout
>
>
>
> mod_rtmp:
>
> =========
>
> Error      4             error C1083: Cannot open include file:
> 'openssl/hmac.h': No such file or directory
> d:\tmp\freeswitch\src\mod\endpoints\mod_rtmp\handshake.h
> 30           1             mod_rtmp
>
>
>
> mod_verto:
>
> =========
>
> Error      5             error C2079: '<unnamed-tag>::recv_addr6' uses
> undefined struct 'sockaddr_in6'
> d:\tmp\freeswitch\src\mod\endpoints\mod_verto\mcast\mcast.h    75
> 1             mod_verto
>
> Error      4             error C2079: '<unnamed-tag>::send_addr6' uses
> undefined struct 'sockaddr_in6'
> d:\tmp\freeswitch\src\mod\endpoints\mod_verto\mcast\mcast.h    74
> 1             mod_verto
>
>
>
>
>
> Thanks a lot!
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','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
> <javascript:_e(%7B%7D,'cvml','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
> <javascript:_e(%7B%7D,'cvml','brian at freeswitch.org');>
>
> [image: Das Bild wurde vom Absender entfernt.]
>
> *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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','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
> <javascript:_e(%7B%7D,'cvml','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.org  Thu Jul 23 19:19:13 2015
From: brian at freeswitch.org (Brian West)
Date: Thu, 23 Jul 2015 10:19:13 -0500
Subject: [Freeswitch-users] Endless playback in conference
In-Reply-To: <CAEorZyn0TGWO9nd8Y4hBpWGPLFnag6k6dVSd7dYZCiCcCPzv8A@mail.gmail.com>
References: <CAEorZynEWmqnOAEfY00_k1Rwe6jPjsjNSPxJjVaTq_neOMFozg@mail.gmail.com>
	<01ec01d0c4cd$82fd1bf0$88f753d0$@botecomm.com>
	<CAEorZyn0TGWO9nd8Y4hBpWGPLFnag6k6dVSd7dYZCiCcCPzv8A@mail.gmail.com>
Message-ID: <CAEJMVkDPHWqoPZ5oQ4Pwp5HJQK_eaqSAfZnoz_ZjqvoUpTwRhw@mail.gmail.com>

What is your goal here?  Maybe I missed the entire scenario.

On Thu, Jul 23, 2015 at 10:07 AM, Danny Gershman <danny.gershman at gmail.com>
wrote:

> Wouldn't a profile have to be created on the fly for this? From what I can
> see you cannot set this for a conference from an api call. Also to stop it,
> you would have to change the profile for the conference to remove it.
> On Wed, Jul 22, 2015 at 6:28 PM Bote Man <bote_radio at botecomm.com> wrote:
>
>> There?s a parameter for mod_conference named ?perpetual-sound? that looks
>> like it would do the trick.
>>
>>
>>
>> It?s about 1/3 of the way down
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/mod_conference
>>
>>
>>
>> PLEASE check further for any changes that might have been made in the
>> latest FreeSWITCH as the conference module has undergone substantial
>> changes and perpetual-sound might have been one of them.
>>
>>
>>
>> Bote
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> *From:* Danny Gershman
>> *Sent:* Wednesday, 22 July, 2015 17:09
>> *Subject:* [Freeswitch-users] Endless playback in conference
>>
>>
>>
>> I'm trying to do an endless playback of an mp3 file in a conference.   I
>> have a couple of ideas, but none seem really solid.
>>
>>
>>
>> 1) Pass a variable on play and monitor from mod_event_socket and play
>> again if not forcibly terminated.
>>
>>
>>
>> 2) Load up local_stream dynamically from an xmlhttp server, and then
>> restart the local stream service, however will interrupt MOH for other
>> users.
>>
>>
>>
>> Any other ideas?  Is there a way to do looping for "api" through
>> mod_event_socket?  I know you can with "sendmsg"
>>
>>
>>
>> --Danny Gershman
>> _________________________________________________________________________
>> 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*
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
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From danny.gershman at gmail.com  Thu Jul 23 19:28:12 2015
From: danny.gershman at gmail.com (Danny Gershman)
Date: Thu, 23 Jul 2015 15:28:12 +0000
Subject: [Freeswitch-users] Endless playback in conference
In-Reply-To: <CAEJMVkDPHWqoPZ5oQ4Pwp5HJQK_eaqSAfZnoz_ZjqvoUpTwRhw@mail.gmail.com>
References: <CAEorZynEWmqnOAEfY00_k1Rwe6jPjsjNSPxJjVaTq_neOMFozg@mail.gmail.com>
	<01ec01d0c4cd$82fd1bf0$88f753d0$@botecomm.com>
	<CAEorZyn0TGWO9nd8Y4hBpWGPLFnag6k6dVSd7dYZCiCcCPzv8A@mail.gmail.com>
	<CAEJMVkDPHWqoPZ5oQ4Pwp5HJQK_eaqSAfZnoz_ZjqvoUpTwRhw@mail.gmail.com>
Message-ID: <CAEorZy=PUzhRgMHJq8mWfw923A19uc2oZ1mg2UObEi-JVMoSXA@mail.gmail.com>

I want to be able to play a file into a conference and have it loop
forever, like music on hold does.  And then when I need to I want to stop
it.

On Thu, Jul 23, 2015 at 11:20 AM Brian West <brian at freeswitch.org> wrote:

> What is your goal here?  Maybe I missed the entire scenario.
>
> On Thu, Jul 23, 2015 at 10:07 AM, Danny Gershman <danny.gershman at gmail.com
> > wrote:
>
>> Wouldn't a profile have to be created on the fly for this? From what I
>> can see you cannot set this for a conference from an api call. Also to stop
>> it, you would have to change the profile for the conference to remove it.
>> On Wed, Jul 22, 2015 at 6:28 PM Bote Man <bote_radio at botecomm.com> wrote:
>>
>>> There?s a parameter for mod_conference named ?perpetual-sound? that
>>> looks like it would do the trick.
>>>
>>>
>>>
>>> It?s about 1/3 of the way down
>>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_conference
>>>
>>>
>>>
>>> PLEASE check further for any changes that might have been made in the
>>> latest FreeSWITCH as the conference module has undergone substantial
>>> changes and perpetual-sound might have been one of them.
>>>
>>>
>>>
>>> Bote
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> *From:* Danny Gershman
>>> *Sent:* Wednesday, 22 July, 2015 17:09
>>> *Subject:* [Freeswitch-users] Endless playback in conference
>>>
>>>
>>>
>>> I'm trying to do an endless playback of an mp3 file in a conference.   I
>>> have a couple of ideas, but none seem really solid.
>>>
>>>
>>>
>>> 1) Pass a variable on play and monitor from mod_event_socket and play
>>> again if not forcibly terminated.
>>>
>>>
>>>
>>> 2) Load up local_stream dynamically from an xmlhttp server, and then
>>> restart the local stream service, however will interrupt MOH for other
>>> users.
>>>
>>>
>>>
>>> Any other ideas?  Is there a way to do looping for "api" through
>>> mod_event_socket?  I know you can with "sendmsg"
>>>
>>>
>>>
>>> --Danny Gershman
>>> _________________________________________________________________________
>>> 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*
> 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
>  _________________________________________________________________________
> 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 blasterjr at gmail.com  Thu Jul 23 20:11:32 2015
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Thu, 23 Jul 2015 10:11:32 -0600
Subject: [Freeswitch-users] ReINVITE - 488 Not Acceptable Here
In-Reply-To: <CAMxfXLpU8+ku1KfLFB7-U-p5UC_s1K+g4zQQuuk-nr4+ZQqhuw@mail.gmail.com>
References: <CAMxfXLpU8+ku1KfLFB7-U-p5UC_s1K+g4zQQuuk-nr4+ZQqhuw@mail.gmail.com>
Message-ID: <CAO3hyzGSVuXMmggC9H7TVQRzPHCp908J1Sr2tS=aav6-kn0Luw@mail.gmail.com>

Sergio, please fill in some additional information like which JS WebSocket
client you're using, there are lots of hold issues with SipML5 and JsSIP,
the only one that i know that has working hold music is sip.js (was a fork
of JsSIP) and I've personally contributed towards making the hold function
correctly on sip.js

On Thu, Jul 23, 2015 at 5:17 AM, Sergio Garc?a <sergio.garcia at quobis.com>
wrote:

> Hello all,
>
> I am using *FreeSwitch* as a *WebRTC gateway* thanks to its Websocket
> support (in my case WSS), but I'm facing this strange problem. Audio and
> Video calls are working perfectly fine, but when I try to set a call On
> Hold, FreeSwitch replies with "488 Not Acceptable Here" error to the
> ReINVITE I'm sending.
>
> The only error I can see in the logs is:
>
>
> * [ERR] sofia.c:7280 Reinvite Codec Error!*
>
> The only difference between the original INVITE and this ReINVITE is that
> I try to set IP address to *0.0.0.0*, port to *0* and media attribute to
> *inactive*. I don't understand what part of the SDP, FS doesn't "like".
>
> Attached you can find a more detailed log file.
>
> Thank you very much in advance.
>
> Regards,
> --
>
> *Sergio Garc?a Ramos *
> VoIP Engineer @ Quobis <http://www.quobis.com/> | e:
> sergio.garcia at quobis.com | t: +34902999465
>
>
>
> _________________________________________________________________________
> 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 blasterjr at gmail.com  Thu Jul 23 20:14:02 2015
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Thu, 23 Jul 2015 10:14:02 -0600
Subject: [Freeswitch-users] Endless playback in conference
In-Reply-To: <CAEorZy=PUzhRgMHJq8mWfw923A19uc2oZ1mg2UObEi-JVMoSXA@mail.gmail.com>
References: <CAEorZynEWmqnOAEfY00_k1Rwe6jPjsjNSPxJjVaTq_neOMFozg@mail.gmail.com>
	<01ec01d0c4cd$82fd1bf0$88f753d0$@botecomm.com>
	<CAEorZyn0TGWO9nd8Y4hBpWGPLFnag6k6dVSd7dYZCiCcCPzv8A@mail.gmail.com>
	<CAEJMVkDPHWqoPZ5oQ4Pwp5HJQK_eaqSAfZnoz_ZjqvoUpTwRhw@mail.gmail.com>
	<CAEorZy=PUzhRgMHJq8mWfw923A19uc2oZ1mg2UObEi-JVMoSXA@mail.gmail.com>
Message-ID: <CAO3hyzHWqhFh-QDBJUFzvjFubDNJYFy_V3S=YW-dnNUOG_RVuA@mail.gmail.com>

Danny the fastest solution i can come up with is doing an originate that
connects a hold music extension to the conference, then when you wanna stop
it, you just kick that member out.

On Thu, Jul 23, 2015 at 9:28 AM, Danny Gershman <danny.gershman at gmail.com>
wrote:

> I want to be able to play a file into a conference and have it loop
> forever, like music on hold does.  And then when I need to I want to stop
> it.
>
> On Thu, Jul 23, 2015 at 11:20 AM Brian West <brian at freeswitch.org> wrote:
>
>> What is your goal here?  Maybe I missed the entire scenario.
>>
>> On Thu, Jul 23, 2015 at 10:07 AM, Danny Gershman <
>> danny.gershman at gmail.com> wrote:
>>
>>> Wouldn't a profile have to be created on the fly for this? From what I
>>> can see you cannot set this for a conference from an api call. Also to stop
>>> it, you would have to change the profile for the conference to remove it.
>>> On Wed, Jul 22, 2015 at 6:28 PM Bote Man <bote_radio at botecomm.com>
>>> wrote:
>>>
>>>> There?s a parameter for mod_conference named ?perpetual-sound? that
>>>> looks like it would do the trick.
>>>>
>>>>
>>>>
>>>> It?s about 1/3 of the way down
>>>>
>>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_conference
>>>>
>>>>
>>>>
>>>> PLEASE check further for any changes that might have been made in the
>>>> latest FreeSWITCH as the conference module has undergone substantial
>>>> changes and perpetual-sound might have been one of them.
>>>>
>>>>
>>>>
>>>> Bote
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> *From:* Danny Gershman
>>>> *Sent:* Wednesday, 22 July, 2015 17:09
>>>> *Subject:* [Freeswitch-users] Endless playback in conference
>>>>
>>>>
>>>>
>>>> I'm trying to do an endless playback of an mp3 file in a conference.
>>>> I have a couple of ideas, but none seem really solid.
>>>>
>>>>
>>>>
>>>> 1) Pass a variable on play and monitor from mod_event_socket and play
>>>> again if not forcibly terminated.
>>>>
>>>>
>>>>
>>>> 2) Load up local_stream dynamically from an xmlhttp server, and then
>>>> restart the local stream service, however will interrupt MOH for other
>>>> users.
>>>>
>>>>
>>>>
>>>> Any other ideas?  Is there a way to do looping for "api" through
>>>> mod_event_socket?  I know you can with "sendmsg"
>>>>
>>>>
>>>>
>>>> --Danny Gershman
>>>>
>>>> _________________________________________________________________________
>>>> 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*
>> 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
>>  _________________________________________________________________________
>> 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 danny.gershman at gmail.com  Thu Jul 23 21:40:53 2015
From: danny.gershman at gmail.com (Danny Gershman)
Date: Thu, 23 Jul 2015 17:40:53 +0000
Subject: [Freeswitch-users] Endless playback in conference
In-Reply-To: <CAO3hyzHWqhFh-QDBJUFzvjFubDNJYFy_V3S=YW-dnNUOG_RVuA@mail.gmail.com>
References: <CAEorZynEWmqnOAEfY00_k1Rwe6jPjsjNSPxJjVaTq_neOMFozg@mail.gmail.com>
	<01ec01d0c4cd$82fd1bf0$88f753d0$@botecomm.com>
	<CAEorZyn0TGWO9nd8Y4hBpWGPLFnag6k6dVSd7dYZCiCcCPzv8A@mail.gmail.com>
	<CAEJMVkDPHWqoPZ5oQ4Pwp5HJQK_eaqSAfZnoz_ZjqvoUpTwRhw@mail.gmail.com>
	<CAEorZy=PUzhRgMHJq8mWfw923A19uc2oZ1mg2UObEi-JVMoSXA@mail.gmail.com>
	<CAO3hyzHWqhFh-QDBJUFzvjFubDNJYFy_V3S=YW-dnNUOG_RVuA@mail.gmail.com>
Message-ID: <CAEorZynSUb=kQM2znu7RxDp9gYRoYB0Be0JUJvRBPmNF3G9Szg@mail.gmail.com>

Ok what I did was the following, that seems to work.

1) Created a new dialplan extension

    <extension name="dynamicmoh">
      <condition field="destination_number" expression="^dynamicmoh-(.*)$">
        <action application="endless_playback" data="$1" />
      </condition>
    </extension>

2) From a conference I do this

conference <conf> bgdial sofia/internal/dynamicmoh-/path/to/file at server-ip

3) To end playback, just hup the member in the conference.

Thanks.
--Danny

On Thu, Jul 23, 2015 at 12:15 PM Chris Tunbridge <blasterjr at gmail.com>
wrote:

> Danny the fastest solution i can come up with is doing an originate that
> connects a hold music extension to the conference, then when you wanna stop
> it, you just kick that member out.
>
> On Thu, Jul 23, 2015 at 9:28 AM, Danny Gershman <danny.gershman at gmail.com>
> wrote:
>
>> I want to be able to play a file into a conference and have it loop
>> forever, like music on hold does.  And then when I need to I want to stop
>> it.
>>
> On Thu, Jul 23, 2015 at 11:20 AM Brian West <brian at freeswitch.org> wrote:
>>
> What is your goal here?  Maybe I missed the entire scenario.
>>>
>>> On Thu, Jul 23, 2015 at 10:07 AM, Danny Gershman <
>>> danny.gershman at gmail.com> wrote:
>>>
>>>> Wouldn't a profile have to be created on the fly for this? From what I
>>>> can see you cannot set this for a conference from an api call. Also to stop
>>>> it, you would have to change the profile for the conference to remove it.
>>>> On Wed, Jul 22, 2015 at 6:28 PM Bote Man <bote_radio at botecomm.com>
>>>> wrote:
>>>>
>>>>> There?s a parameter for mod_conference named ?perpetual-sound? that
>>>>> looks like it would do the trick.
>>>>>
>>>>>
>>>>>
>>>>> It?s about 1/3 of the way down
>>>>>
>>>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_conference
>>>>>
>>>>>
>>>>>
>>>>> PLEASE check further for any changes that might have been made in the
>>>>> latest FreeSWITCH as the conference module has undergone substantial
>>>>> changes and perpetual-sound might have been one of them.
>>>>>
>>>>>
>>>>>
>>>>> Bote
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> *From:* Danny Gershman
>>>>> *Sent:* Wednesday, 22 July, 2015 17:09
>>>>> *Subject:* [Freeswitch-users] Endless playback in conference
>>>>>
>>>>>
>>>>>
>>>>> I'm trying to do an endless playback of an mp3 file in a conference.
>>>>> I have a couple of ideas, but none seem really solid.
>>>>>
>>>>>
>>>>>
>>>>> 1) Pass a variable on play and monitor from mod_event_socket and play
>>>>> again if not forcibly terminated.
>>>>>
>>>>>
>>>>>
>>>>> 2) Load up local_stream dynamically from an xmlhttp server, and then
>>>>> restart the local stream service, however will interrupt MOH for other
>>>>> users.
>>>>>
>>>>>
>>>>>
>>>>> Any other ideas?  Is there a way to do looping for "api" through
>>>>> mod_event_socket?  I know you can with "sendmsg"
>>>>>
>>>>>
>>>>>
>>>>> --Danny Gershman
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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*
>>> 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
>>>
>> _________________________________________________________________________
>>> 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 igorolhovskiy at gmail.com  Fri Jul 24 01:31:17 2015
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Fri, 24 Jul 2015 00:31:17 +0300
Subject: [Freeswitch-users] Strange mod_sofia logic
Message-ID: <CAJTkRNucm=eZKTt92oQWWVD767=yPBZAwoEyCzoi45+0FVJ-og@mail.gmail.com>

Hi!

Found really strange behavior, but may be it's correct.
I'm using FusionPBX as a FreeSwitch GUI, but problem more to FS.
So, when I'm setting context of profile 'internal' to 'public' (yes, it's
incorrect, but possible) this profile stop accept registrations.
Answering "SIP/2.0
405 Method Not Allowed"
and gives list of methods like
INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,PRACK,NOTIFY,PUBLISH,SUBSCRIBE
But when changing to 'default, all works correct and list of methods are
INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER,
NOTIFY, PUBLISH, SUBSCRIBE
So the question is: how changing of context can inflict on methods
supported?  And how to control methods supported in more obvious way? (For
ex, disable REFER)
FreeSwitch version is 1.4.20
-- 
Best regards,
Igor
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From dsdee at dsba.net  Fri Jul 24 01:36:33 2015
From: dsdee at dsba.net (David S. Dee)
Date: Thu, 23 Jul 2015 15:36:33 -0600
Subject: [Freeswitch-users] Routing based on CID Name
Message-ID: <1437687393.3971659.331631105.40C1EF09@webmail.messagingengine.com>

Hello;

I've been a freeswitch user for some time, but I am finally getting
around to trying to get idea working again.

I have an in-house CID lookup routine that works fine... it sets
caller_id_name and effective_caller_id_name and those modified names
show up fine on my telephones when the calls come in.

That lookup routine adds a prefix of TM: if its a telemarketer, or VM:
if i want the caller to go directly to voicemail...? Basically, if the
original CID from my provider shows up as "FOO Marketing", then after
the cid-extension runs, the caller_id_name and effective_caller_name to
"TM:FOO Marketing"....

What is -not- working is the routing.

I have an extension right after the CID Lookup that I want to do the
routing directly to voicemail.?? Watching the debug logs, it appears
that the field is already parsed and is set to the *original* callerid
name that was available when the call came in...?? Trying different
variations of field="caller_id_name" and field="${caller_id_name}" (and
same with effective_caller_id_name) still yields no joy.

Any guidance??? The snippets from my dialplan are below.

Thanks!


<extension name="CID Inbound Lookup" continue="true">??????? <condition
field="ani" expression="(\d{10})">??????????? <action application="log"
data="INFO In CID Inbound Lookup '$1' -- The Second Time"/>
<action application="log" data="INFO CIDLookup(before): CID Num:
'${caller_id_number}' CID Name: '${caller_id_name}' Orig CID Name:
'${original_caller_id_name}' Dest: '${destination_number}'"/>

<action application="set" data="url=num=${caller_id_number}&name=${url_-
encode(${caller_id_name})}&did=${destination_number}"/>
<action application="log" data="INFO CIDLookup(before): url: '${url}'"/>

<action application="curl" data="
http://MYDOMAIN.com/MYCIDLOOKUP.php?${url}"/[1]>??????????? <action
application="info"/>

<action application="log" data="INFO CIDLookup(after ):
curl_response_code: ${curl_response_code}"/>??????????? <action
application="log" data="INFO CIDLookup(after ): curl_response_data:
${curl_response_data}"/>

<action application="set" data="caller_id_name=${curl_response_data}"/>
<action application="set"
data="effective_caller_id_name=${curl_response_data}"/>


<action application="log" data="INFO CIDLookup(after ): CID Num:
'${caller_id_number}' CID Name: '${caller_id_name}' Orig CID Name:
'${original_caller_id_name}' Dest: '${destination_number}'"/>
<action application="log" data="INFO CIDLookup(after ): caller_id_name:
'${caller_id_name}', original_caller_id_name:
'${original_caller_id_name}', effective_caller_id_name:
'${effective_caller_id_name}'" />

<!-- This event will be picked up by HB-Bot for announcing -->
<action application="event" data="Event-Name=CUSTOM,Event-
Subclass=CallAnnounce,ani=${destination_number},cidnum=${caller_id_numb-
er},cidname=${caller_id_name}"/>

</condition>??? </extension> and here are my test cases that are *right*
after the above... but yet can't pick up the change in variables.? *ONE*
of these (I expect '3b' or '1') should fire... none do.



<extension name="cid_test_0" continue="true">?????? <condition
field="original_caller_id_name" expression="TM" >??????????? <action
application="log" data="INFO cid_test_0: :
'${original_caller_id_name}'"/>?????? </condition>??? </extension>

<extension name="cid_test_1" continue="true">?????? <condition
field="caller_id_name" expression="TM" >??????????? <action
application="log" data="INFO cid_test_1: : '${caller_id_name}'"/>
</condition>??? </extension>

<extension name="cid_test_2" continue="true">?????? <condition
field="effective_caller_id_name" expression="TM" >??????????? <action
application="log" data="INFO cid_test_2: :
'${effective_caller_id_name}'"/>?????? </condition>??? </extension>

<extension name="cid_test_3" continue="true">?????? <condition
field='${caller_id_name}' expression="TM" >??????????? <action
application="log" data="INFO cid_test_3: : '${caller_id_name}'"/>
</condition>??? </extension>

<extension name="cid_test_3b" continue="true">?????? <condition
field="${caller_id_name}" expression="TM" >??????????? <action
application="log" data="INFO cid_test_3b: : '${caller_id_name}'"/>
</condition>??? </extension>

<extension name="cid_test_4" continue="true">?????? <condition
field='${effective_caller_id_name}' expression="TM" >??????????? <action
application="log" data="INFO cid_test_4: :
'${effective_caller_id_name}'"/>?????? </condition>??? </extension>

<extension name="cid_test_5" continue="true">?????? <condition
field='${effective_caller_id_name}' expression="TM" >??????????? <action
application="log" data="INFO cid_test_5: :
'${effective_caller_id_name}'"/>?????? </condition>??? </extension>


Links:

  1. http://www.dsba.net/voip/cidlookup.php?${url}"/
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From steveayre at gmail.com  Fri Jul 24 01:54:37 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 23 Jul 2015 22:54:37 +0100
Subject: [Freeswitch-users] Routing based on CID Name
In-Reply-To: <1437687393.3971659.331631105.40C1EF09@webmail.messagingengine.com>
References: <1437687393.3971659.331631105.40C1EF09@webmail.messagingengine.com>
Message-ID: <CAFiqYumtDnY8NL7YSujfhZLk4QbXm6Nb5nbqdOfb+x+FkHuMoA@mail.gmail.com>

The dialplan is run in two stages - hunting and execution.

In hunting it will look through the extensions for any where the the
conditions match and puts the actions together into a list. It then
executes that list of actions in order. As a result it's checking
the effective_caller_id_name condition before the action that sets the
variable has been executed.

As a workaround if you run 'transfer' that will perform a new dialplan hunt
generating a new list of actions to execute. If you set the variable then
transfer, then the conditions in the new hunt will have the variable
already set.

Another option is using inline="true" on the set action. That will execute
the action during the hunt itself.



On 23 July 2015 at 22:36, David S. Dee <dsdee at dsba.net> wrote:

>  Hello;
>
> I've been a freeswitch user for some time, but I am finally getting around
> to trying to get idea working again.
>
> I have an in-house CID lookup routine that works fine... it sets
> caller_id_name and effective_caller_id_name and those modified names show
> up fine on my telephones when the calls come in.
>
> That lookup routine adds a prefix of TM: if its a telemarketer, or VM: if
> i want the caller to go directly to voicemail...  Basically, if the
> original CID from my provider shows up as "FOO Marketing", then after the
> cid-extension runs, the caller_id_name and effective_caller_name to "TM:FOO
> Marketing"....
>
> What is -not- working is the routing.
>
> I have an extension right after the CID Lookup that I want to do the
> routing directly to voicemail.   Watching the debug logs, it appears that
> the field is already parsed and is set to the *original* callerid name that
> was available when the call came in...   Trying different variations of
> field="caller_id_name" and field="${caller_id_name}" (and same with
> effective_caller_id_name) still yields no joy.
>
> Any guidance?   The snippets from my dialplan are below.
>
> Thanks!
>
>
>    <extension name="CID Inbound Lookup" continue="true">
>         <condition field="ani" expression="(\d{10})">
>             <action application="log" data="INFO In CID Inbound Lookup
> '$1' -- The Second Time"/>
>             <action application="log" data="INFO CIDLookup(before): CID
> Num: '${caller_id_number}' CID Name: '${caller_id_name}' Orig CID Name:
> '${original_caller_id_name}' Dest: '${destination_number}'"/>
>
>             <action application="set"
> data="url=num=${caller_id_number}&name=${url_encode(${caller_id_name})}&did=${destination_number}"/>
>             <action application="log" data="INFO CIDLookup(before): url:
> '${url}'"/>
>
>             <action application="curl" data="
> http://MYDOMAIN.com/MYCIDLOOKUP.php?${url}"/
> <http://www.dsba.net/voip/cidlookup.php?$%7Burl%7D%22/>>
>             <action application="info"/>
>
>             <action application="log" data="INFO CIDLookup(after ):
> curl_response_code: ${curl_response_code}"/>
>             <action application="log" data="INFO CIDLookup(after ):
> curl_response_data: ${curl_response_data}"/>
>
>             <action application="set"
> data="caller_id_name=${curl_response_data}"/>
>             <action application="set"
> data="effective_caller_id_name=${curl_response_data}"/>
>
>
>             <action application="log" data="INFO CIDLookup(after ): CID
> Num: '${caller_id_number}' CID Name: '${caller_id_name}' Orig CID Name:
> '${original_caller_id_name}' Dest: '${destination_number}'"/>
>             <action application="log" data="INFO CIDLookup(after ):
> caller_id_name: '${caller_id_name}', original_caller_id_name:
> '${original_caller_id_name}', effective_caller_id_name:
> '${effective_caller_id_name}'" />
>
>             <!-- This event will be picked up by HB-Bot for announcing -->
>             <action application="event"
> data="Event-Name=CUSTOM,Event-Subclass=CallAnnounce,ani=${destination_number},cidnum=${caller_id_number},cidname=${caller_id_name}"/>
>
>         </condition>
>     </extension>
> and here are my test cases that are *right* after the above... but yet
> can't pick up the change in variables.  *ONE* of these (I expect '3b' or
> '1') should fire... none do.
>
>
>
>     <extension name="cid_test_0" continue="true">
>        <condition field="original_caller_id_name" expression="TM" >
>             <action application="log" data="INFO cid_test_0: :
> '${original_caller_id_name}'"/>
>        </condition>
>     </extension>
>
>     <extension name="cid_test_1" continue="true">
>        <condition field="caller_id_name" expression="TM" >
>             <action application="log" data="INFO cid_test_1: :
> '${caller_id_name}'"/>
>        </condition>
>     </extension>
>
>     <extension name="cid_test_2" continue="true">
>        <condition field="effective_caller_id_name" expression="TM" >
>             <action application="log" data="INFO cid_test_2: :
> '${effective_caller_id_name}'"/>
>        </condition>
>     </extension>
>
>     <extension name="cid_test_3" continue="true">
>        <condition field='${caller_id_name}' expression="TM" >
>             <action application="log" data="INFO cid_test_3: :
> '${caller_id_name}'"/>
>        </condition>
>     </extension>
>
>     <extension name="cid_test_3b" continue="true">
>        <condition field="${caller_id_name}" expression="TM" >
>             <action application="log" data="INFO cid_test_3b: :
> '${caller_id_name}'"/>
>        </condition>
>     </extension>
>
>     <extension name="cid_test_4" continue="true">
>        <condition field='${effective_caller_id_name}' expression="TM" >
>             <action application="log" data="INFO cid_test_4: :
> '${effective_caller_id_name}'"/>
>        </condition>
>     </extension>
>
>     <extension name="cid_test_5" continue="true">
>        <condition field='${effective_caller_id_name}' expression="TM" >
>             <action application="log" data="INFO cid_test_5: :
> '${effective_caller_id_name}'"/>
>        </condition>
>     </extension>
>
> _________________________________________________________________________
> 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 dsdee at dsba.net  Fri Jul 24 04:31:19 2015
From: dsdee at dsba.net (David S. Dee)
Date: Thu, 23 Jul 2015 18:31:19 -0600
Subject: [Freeswitch-users] Routing based on CID Name
In-Reply-To: <CAFiqYumtDnY8NL7YSujfhZLk4QbXm6Nb5nbqdOfb+x+FkHuMoA@mail.gmail.com>
References: <1437687393.3971659.331631105.40C1EF09@webmail.messagingengine.com>
	<CAFiqYumtDnY8NL7YSujfhZLk4QbXm6Nb5nbqdOfb+x+FkHuMoA@mail.gmail.com>
Message-ID: <1437697879.4007937.331739905.41D6FA6C@webmail.messagingengine.com>

I started playing with the 'inline=true's and finally got them in the
right place...

done.? Thanks!


On Thu, Jul 23, 2015, at 15:54, Steven Ayre wrote:
> The dialplan is run in two stages - hunting and execution.
>
> In hunting it will look through the extensions for any where the the
> conditions match and puts the actions together into a list. It then
> executes that list of actions in order. As a result it's checking the
> effective_caller_id_name condition before the action that sets the
> variable has been executed.
>
> As a workaround if you run 'transfer' that will perform a new dialplan
> hunt generating a new list of actions to execute. If you set the
> variable then transfer, then the conditions in the new hunt will have
> the variable already set.
>
> Another option is using inline="true" on the set action. That will
> execute the action during the hunt itself.
>
>
>
> On 23 July 2015 at 22:36, David S. Dee <dsdee at dsba.net> wrote:
>> __
>> Hello;
>>
>> I've been a freeswitch user for some time, but I am finally getting
>> around to trying to get idea working again.
>>
>> I have an in-house CID lookup routine that works fine... it sets
>> caller_id_name and effective_caller_id_name and those modified names
>> show up fine on my telephones when the calls come in.
>>
>> That lookup routine adds a prefix of TM: if its a telemarketer, or
>> VM: if i want the caller to go directly to voicemail...? Basically,
>> if the original CID from my provider shows up as "FOO Marketing",
>> then after the cid-extension runs, the caller_id_name and
>> effective_caller_name to "TM:FOO Marketing"....
>>
>> What is -not- working is the routing.
>>
>> I have an extension right after the CID Lookup that I want to do the
>> routing directly to voicemail.?? Watching the debug logs, it appears
>> that the field is already parsed and is set to the *original*
>> callerid name that was available when the call came in...?? Trying
>> different variations of field="caller_id_name" and
>> field="${caller_id_name}" (and same with effective_caller_id_name)
>> still yields no joy.
>>
>> Any guidance??? The snippets from my dialplan are below.
>>
>> Thanks!
>>
>>
>> <extension name="CID Inbound Lookup" continue="true">
>> <condition field="ani" expression="(\d{10})">??????????? <action
>> application="log" data="INFO In CID Inbound Lookup '$1' -- The Second
>> Time"/>??????????? <action application="log" data="INFO
>> CIDLookup(before): CID Num: '${caller_id_number}' CID Name:
>> '${caller_id_name}' Orig CID Name: '${original_caller_id_name}' Dest:
>> '${destination_number}'"/>
>>
>> <action application="set" data="url=num=${caller_id_number}&name=${u-
>> rl_encode(${caller_id_name})}&did=${destination_number}"/>
>> <action application="log" data="INFO CIDLookup(before): url:
>> '${url}'"/>
>>
>> <action application="curl" data="
>> http://MYDOMAIN.com/MYCIDLOOKUP.php?${url}"/[1]>??????????? <action
>> application="info"/>
>>
>> <action application="log" data="INFO CIDLookup(after ):
>> curl_response_code: ${curl_response_code}"/>??????????? <action
>> application="log" data="INFO CIDLookup(after ): curl_response_data:
>> ${curl_response_data}"/>
>>
>> <action application="set"
>> data="caller_id_name=${curl_response_data}"/>??????????? <action
>> application="set"
>> data="effective_caller_id_name=${curl_response_data}"/>
>>
>>
>> <action application="log" data="INFO CIDLookup(after ): CID Num:
>> '${caller_id_number}' CID Name: '${caller_id_name}' Orig CID Name:
>> '${original_caller_id_name}' Dest: '${destination_number}'"/>
>> <action application="log" data="INFO CIDLookup(after ):
>> caller_id_name: '${caller_id_name}', original_caller_id_name:
>> '${original_caller_id_name}', effective_caller_id_name:
>> '${effective_caller_id_name}'" />
>>
>> <!-- This event will be picked up by HB-Bot for announcing -->
>> <action application="event" data="Event-Name=CUSTOM,Event-
>> Subclass=CallAnnounce,ani=${destination_number},cidnum=${caller_id_n-
>> umber},cidname=${caller_id_name}"/>
>>
>> </condition>??? </extension> and here are my test cases that are
>> *right* after the above... but yet can't pick up the change in
>> variables.? *ONE* of these (I expect '3b' or '1') should fire...
>> none do.
>>
>>
>>
>> <extension name="cid_test_0" continue="true">?????? <condition
>> field="original_caller_id_name" expression="TM" >??????????? <action
>> application="log" data="INFO cid_test_0: :
>> '${original_caller_id_name}'"/>?????? </condition>??? </extension>
>>
>> <extension name="cid_test_1" continue="true">?????? <condition
>> field="caller_id_name" expression="TM" >??????????? <action
>> application="log" data="INFO cid_test_1: : '${caller_id_name}'"/>
>> </condition>??? </extension>
>>
>> <extension name="cid_test_2" continue="true">?????? <condition
>> field="effective_caller_id_name" expression="TM" >??????????? <action
>> application="log" data="INFO cid_test_2: :
>> '${effective_caller_id_name}'"/>?????? </condition>??? </extension>
>>
>> <extension name="cid_test_3" continue="true">?????? <condition
>> field='${caller_id_name}' expression="TM" >??????????? <action
>> application="log" data="INFO cid_test_3: : '${caller_id_name}'"/>
>> </condition>??? </extension>
>>
>> <extension name="cid_test_3b" continue="true">?????? <condition
>> field="${caller_id_name}" expression="TM" >??????????? <action
>> application="log" data="INFO cid_test_3b: : '${caller_id_name}'"/>
>> </condition>??? </extension>
>>
>> <extension name="cid_test_4" continue="true">?????? <condition
>> field='${effective_caller_id_name}' expression="TM" >
>> <action application="log" data="INFO cid_test_4: :
>> '${effective_caller_id_name}'"/>?????? </condition>??? </extension>
>>
>> <extension name="cid_test_5" continue="true">?????? <condition
>> field='${effective_caller_id_name}' expression="TM" >
>> <action application="log" data="INFO cid_test_5: :
>> '${effective_caller_id_name}'"/>?????? </condition>??? </extension>
>>
>> __________________________________________________________________-
>> _______
>>
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. http://www.dsba.net/voip/cidlookup.php?$%7Burl%7D%22/
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From ssinyagin at gmail.com  Fri Jul 24 04:31:48 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 24 Jul 2015 02:31:48 +0200
Subject: [Freeswitch-users] Routing based on CID Name
In-Reply-To: <1437687393.3971659.331631105.40C1EF09@webmail.messagingengine.com>
References: <1437687393.3971659.331631105.40C1EF09@webmail.messagingengine.com>
Message-ID: <CACOYK=iVgt=KqUkr+S5YqeXhZ9B09kaox8vNwU3LyoWmrc5R5A@mail.gmail.com>

 <condition field="effective_caller_id_name" expression="TM" >

You need to use ${effective_caller_id_name} here.

But I would do the whole logic in mod_lua or mod_perl, as it's much easier
to program there.
On Jul 23, 2015 11:45 PM, "David S. Dee" <dsdee at dsba.net> wrote:

>  Hello;
>
> I've been a freeswitch user for some time, but I am finally getting around
> to trying to get idea working again.
>
> I have an in-house CID lookup routine that works fine... it sets
> caller_id_name and effective_caller_id_name and those modified names show
> up fine on my telephones when the calls come in.
>
> That lookup routine adds a prefix of TM: if its a telemarketer, or VM: if
> i want the caller to go directly to voicemail...  Basically, if the
> original CID from my provider shows up as "FOO Marketing", then after the
> cid-extension runs, the caller_id_name and effective_caller_name to "TM:FOO
> Marketing"....
>
> What is -not- working is the routing.
>
> I have an extension right after the CID Lookup that I want to do the
> routing directly to voicemail.   Watching the debug logs, it appears that
> the field is already parsed and is set to the *original* callerid name that
> was available when the call came in...   Trying different variations of
> field="caller_id_name" and field="${caller_id_name}" (and same with
> effective_caller_id_name) still yields no joy.
>
> Any guidance?   The snippets from my dialplan are below.
>
> Thanks!
>
>
>    <extension name="CID Inbound Lookup" continue="true">
>         <condition field="ani" expression="(\d{10})">
>             <action application="log" data="INFO In CID Inbound Lookup
> '$1' -- The Second Time"/>
>             <action application="log" data="INFO CIDLookup(before): CID
> Num: '${caller_id_number}' CID Name: '${caller_id_name}' Orig CID Name:
> '${original_caller_id_name}' Dest: '${destination_number}'"/>
>
>             <action application="set"
> data="url=num=${caller_id_number}&name=${url_encode(${caller_id_name})}&did=${destination_number}"/>
>             <action application="log" data="INFO CIDLookup(before): url:
> '${url}'"/>
>
>             <action application="curl" data="
> http://MYDOMAIN.com/MYCIDLOOKUP.php?${url}"/
> <http://www.dsba.net/voip/cidlookup.php?$%7Burl%7D%22/>>
>             <action application="info"/>
>
>             <action application="log" data="INFO CIDLookup(after ):
> curl_response_code: ${curl_response_code}"/>
>             <action application="log" data="INFO CIDLookup(after ):
> curl_response_data: ${curl_response_data}"/>
>
>             <action application="set"
> data="caller_id_name=${curl_response_data}"/>
>             <action application="set"
> data="effective_caller_id_name=${curl_response_data}"/>
>
>
>             <action application="log" data="INFO CIDLookup(after ): CID
> Num: '${caller_id_number}' CID Name: '${caller_id_name}' Orig CID Name:
> '${original_caller_id_name}' Dest: '${destination_number}'"/>
>             <action application="log" data="INFO CIDLookup(after ):
> caller_id_name: '${caller_id_name}', original_caller_id_name:
> '${original_caller_id_name}', effective_caller_id_name:
> '${effective_caller_id_name}'" />
>
>             <!-- This event will be picked up by HB-Bot for announcing -->
>             <action application="event"
> data="Event-Name=CUSTOM,Event-Subclass=CallAnnounce,ani=${destination_number},cidnum=${caller_id_number},cidname=${caller_id_name}"/>
>
>         </condition>
>     </extension>
> and here are my test cases that are *right* after the above... but yet
> can't pick up the change in variables.  *ONE* of these (I expect '3b' or
> '1') should fire... none do.
>
>
>
>     <extension name="cid_test_0" continue="true">
>        <condition field="original_caller_id_name" expression="TM" >
>             <action application="log" data="INFO cid_test_0: :
> '${original_caller_id_name}'"/>
>        </condition>
>     </extension>
>
>     <extension name="cid_test_1" continue="true">
>        <condition field="caller_id_name" expression="TM" >
>             <action application="log" data="INFO cid_test_1: :
> '${caller_id_name}'"/>
>        </condition>
>     </extension>
>
>     <extension name="cid_test_2" continue="true">
>        <condition field="effective_caller_id_name" expression="TM" >
>             <action application="log" data="INFO cid_test_2: :
> '${effective_caller_id_name}'"/>
>        </condition>
>     </extension>
>
>     <extension name="cid_test_3" continue="true">
>        <condition field='${caller_id_name}' expression="TM" >
>             <action application="log" data="INFO cid_test_3: :
> '${caller_id_name}'"/>
>        </condition>
>     </extension>
>
>     <extension name="cid_test_3b" continue="true">
>        <condition field="${caller_id_name}" expression="TM" >
>             <action application="log" data="INFO cid_test_3b: :
> '${caller_id_name}'"/>
>        </condition>
>     </extension>
>
>     <extension name="cid_test_4" continue="true">
>        <condition field='${effective_caller_id_name}' expression="TM" >
>             <action application="log" data="INFO cid_test_4: :
> '${effective_caller_id_name}'"/>
>        </condition>
>     </extension>
>
>     <extension name="cid_test_5" continue="true">
>        <condition field='${effective_caller_id_name}' expression="TM" >
>             <action application="log" data="INFO cid_test_5: :
> '${effective_caller_id_name}'"/>
>        </condition>
>     </extension>
>
> _________________________________________________________________________
> 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 ssinyagin at gmail.com  Fri Jul 24 04:34:51 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 24 Jul 2015 02:34:51 +0200
Subject: [Freeswitch-users] Endless playback in conference
In-Reply-To: <CAEorZynSUb=kQM2znu7RxDp9gYRoYB0Be0JUJvRBPmNF3G9Szg@mail.gmail.com>
References: <CAEorZynEWmqnOAEfY00_k1Rwe6jPjsjNSPxJjVaTq_neOMFozg@mail.gmail.com>
	<01ec01d0c4cd$82fd1bf0$88f753d0$@botecomm.com>
	<CAEorZyn0TGWO9nd8Y4hBpWGPLFnag6k6dVSd7dYZCiCcCPzv8A@mail.gmail.com>
	<CAEJMVkDPHWqoPZ5oQ4Pwp5HJQK_eaqSAfZnoz_ZjqvoUpTwRhw@mail.gmail.com>
	<CAEorZy=PUzhRgMHJq8mWfw923A19uc2oZ1mg2UObEi-JVMoSXA@mail.gmail.com>
	<CAO3hyzHWqhFh-QDBJUFzvjFubDNJYFy_V3S=YW-dnNUOG_RVuA@mail.gmail.com>
	<CAEorZynSUb=kQM2znu7RxDp9gYRoYB0Be0JUJvRBPmNF3G9Szg@mail.gmail.com>
Message-ID: <CACOYK=i4NSW0uSdS+4O3J2LbNYyxfG0_M3gvUcuB-AQfYGebpw@mail.gmail.com>

But then other members are still able to hear some sound from each other
through the music. Is that OK with you?
On Jul 23, 2015 7:42 PM, "Danny Gershman" <danny.gershman at gmail.com> wrote:

> Ok what I did was the following, that seems to work.
>
> 1) Created a new dialplan extension
>
>     <extension name="dynamicmoh">
>       <condition field="destination_number" expression="^dynamicmoh
> -(.*)$">
>         <action application="endless_playback" data="$1" />
>       </condition>
>     </extension>
>
> 2) From a conference I do this
>
> conference <conf> bgdial sofia/internal/dynamicmoh-/path/to/file at server-ip
>
> 3) To end playback, just hup the member in the conference.
>
> Thanks.
> --Danny
>
> On Thu, Jul 23, 2015 at 12:15 PM Chris Tunbridge <blasterjr at gmail.com>
> wrote:
>
>> Danny the fastest solution i can come up with is doing an originate that
>> connects a hold music extension to the conference, then when you wanna stop
>> it, you just kick that member out.
>>
>> On Thu, Jul 23, 2015 at 9:28 AM, Danny Gershman <danny.gershman at gmail.com
>> > wrote:
>>
>>> I want to be able to play a file into a conference and have it loop
>>> forever, like music on hold does.  And then when I need to I want to stop
>>> it.
>>>
>> On Thu, Jul 23, 2015 at 11:20 AM Brian West <brian at freeswitch.org> wrote:
>>>
>> What is your goal here?  Maybe I missed the entire scenario.
>>>>
>>>> On Thu, Jul 23, 2015 at 10:07 AM, Danny Gershman <
>>>> danny.gershman at gmail.com> wrote:
>>>>
>>>>> Wouldn't a profile have to be created on the fly for this? From what I
>>>>> can see you cannot set this for a conference from an api call. Also to stop
>>>>> it, you would have to change the profile for the conference to remove it.
>>>>> On Wed, Jul 22, 2015 at 6:28 PM Bote Man <bote_radio at botecomm.com>
>>>>> wrote:
>>>>>
>>>>>> There?s a parameter for mod_conference named ?perpetual-sound? that
>>>>>> looks like it would do the trick.
>>>>>>
>>>>>>
>>>>>>
>>>>>> It?s about 1/3 of the way down
>>>>>>
>>>>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_conference
>>>>>>
>>>>>>
>>>>>>
>>>>>> PLEASE check further for any changes that might have been made in the
>>>>>> latest FreeSWITCH as the conference module has undergone substantial
>>>>>> changes and perpetual-sound might have been one of them.
>>>>>>
>>>>>>
>>>>>>
>>>>>> Bote
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> *From:* Danny Gershman
>>>>>> *Sent:* Wednesday, 22 July, 2015 17:09
>>>>>> *Subject:* [Freeswitch-users] Endless playback in conference
>>>>>>
>>>>>>
>>>>>>
>>>>>> I'm trying to do an endless playback of an mp3 file in a conference.
>>>>>>   I have a couple of ideas, but none seem really solid.
>>>>>>
>>>>>>
>>>>>>
>>>>>> 1) Pass a variable on play and monitor from mod_event_socket and play
>>>>>> again if not forcibly terminated.
>>>>>>
>>>>>>
>>>>>>
>>>>>> 2) Load up local_stream dynamically from an xmlhttp server, and then
>>>>>> restart the local stream service, however will interrupt MOH for other
>>>>>> users.
>>>>>>
>>>>>>
>>>>>>
>>>>>> Any other ideas?  Is there a way to do looping for "api" through
>>>>>> mod_event_socket?  I know you can with "sendmsg"
>>>>>>
>>>>>>
>>>>>>
>>>>>> --Danny Gershman
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> 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*
>>>> 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
>>>>
>>> _________________________________________________________________________
>>>> 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 ssinyagin at gmail.com  Fri Jul 24 05:04:24 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 24 Jul 2015 03:04:24 +0200
Subject: [Freeswitch-users] Routing based on CID Name
In-Reply-To: <CACOYK=iVgt=KqUkr+S5YqeXhZ9B09kaox8vNwU3LyoWmrc5R5A@mail.gmail.com>
References: <1437687393.3971659.331631105.40C1EF09@webmail.messagingengine.com>
	<CACOYK=iVgt=KqUkr+S5YqeXhZ9B09kaox8vNwU3LyoWmrc5R5A@mail.gmail.com>
Message-ID: <CACOYK=js959b7Gy7+0q4a33CwpioAdJBwEb5L_TmyR983CM=fw@mail.gmail.com>

also with mod_perl, you can set a short HTTP timeout, so that your
system still works if the remote HTTP server stops responding.

Here's an example for CID lookup in LDAP:
https://github.com/voxserv/freeswitch-helper-scripts/blob/master/mod_perl/lookup_caller_ldap.pl



On Fri, Jul 24, 2015 at 2:31 AM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>  <condition field="effective_caller_id_name" expression="TM" >
>
> You need to use ${effective_caller_id_name} here.
>
> But I would do the whole logic in mod_lua or mod_perl, as it's much easier
> to program there.
>
> On Jul 23, 2015 11:45 PM, "David S. Dee" <dsdee at dsba.net> wrote:
>>
>> Hello;
>>
>> I've been a freeswitch user for some time, but I am finally getting around
>> to trying to get idea working again.
>>
>> I have an in-house CID lookup routine that works fine... it sets
>> caller_id_name and effective_caller_id_name and those modified names show up
>> fine on my telephones when the calls come in.
>>
>> That lookup routine adds a prefix of TM: if its a telemarketer, or VM: if
>> i want the caller to go directly to voicemail...  Basically, if the original
>> CID from my provider shows up as "FOO Marketing", then after the
>> cid-extension runs, the caller_id_name and effective_caller_name to "TM:FOO
>> Marketing"....
>>
>> What is -not- working is the routing.
>>
>> I have an extension right after the CID Lookup that I want to do the
>> routing directly to voicemail.   Watching the debug logs, it appears that
>> the field is already parsed and is set to the *original* callerid name that
>> was available when the call came in...   Trying different variations of
>> field="caller_id_name" and field="${caller_id_name}" (and same with
>> effective_caller_id_name) still yields no joy.
>>
>> Any guidance?   The snippets from my dialplan are below.
>>
>> Thanks!
>>
>>
>>    <extension name="CID Inbound Lookup" continue="true">
>>         <condition field="ani" expression="(\d{10})">
>>             <action application="log" data="INFO In CID Inbound Lookup
>> '$1' -- The Second Time"/>
>>             <action application="log" data="INFO CIDLookup(before): CID
>> Num: '${caller_id_number}' CID Name: '${caller_id_name}' Orig CID Name:
>> '${original_caller_id_name}' Dest: '${destination_number}'"/>
>>
>>             <action application="set"
>> data="url=num=${caller_id_number}&name=${url_encode(${caller_id_name})}&did=${destination_number}"/>
>>             <action application="log" data="INFO CIDLookup(before): url:
>> '${url}'"/>
>>
>>             <action application="curl"
>> data="http://MYDOMAIN.com/MYCIDLOOKUP.php?${url}"/>
>>             <action application="info"/>
>>
>>             <action application="log" data="INFO CIDLookup(after ):
>> curl_response_code: ${curl_response_code}"/>
>>             <action application="log" data="INFO CIDLookup(after ):
>> curl_response_data: ${curl_response_data}"/>
>>
>>             <action application="set"
>> data="caller_id_name=${curl_response_data}"/>
>>             <action application="set"
>> data="effective_caller_id_name=${curl_response_data}"/>
>>
>>
>>             <action application="log" data="INFO CIDLookup(after ): CID
>> Num: '${caller_id_number}' CID Name: '${caller_id_name}' Orig CID Name:
>> '${original_caller_id_name}' Dest: '${destination_number}'"/>
>>             <action application="log" data="INFO CIDLookup(after ):
>> caller_id_name: '${caller_id_name}', original_caller_id_name:
>> '${original_caller_id_name}', effective_caller_id_name:
>> '${effective_caller_id_name}'" />
>>
>>             <!-- This event will be picked up by HB-Bot for announcing -->
>>             <action application="event"
>> data="Event-Name=CUSTOM,Event-Subclass=CallAnnounce,ani=${destination_number},cidnum=${caller_id_number},cidname=${caller_id_name}"/>
>>
>>         </condition>
>>     </extension>
>> and here are my test cases that are *right* after the above... but yet
>> can't pick up the change in variables.  *ONE* of these (I expect '3b' or
>> '1') should fire... none do.
>>
>>
>>
>>     <extension name="cid_test_0" continue="true">
>>        <condition field="original_caller_id_name" expression="TM" >
>>             <action application="log" data="INFO cid_test_0: :
>> '${original_caller_id_name}'"/>
>>        </condition>
>>     </extension>
>>
>>     <extension name="cid_test_1" continue="true">
>>        <condition field="caller_id_name" expression="TM" >
>>             <action application="log" data="INFO cid_test_1: :
>> '${caller_id_name}'"/>
>>        </condition>
>>     </extension>
>>
>>     <extension name="cid_test_2" continue="true">
>>        <condition field="effective_caller_id_name" expression="TM" >
>>             <action application="log" data="INFO cid_test_2: :
>> '${effective_caller_id_name}'"/>
>>        </condition>
>>     </extension>
>>
>>     <extension name="cid_test_3" continue="true">
>>        <condition field='${caller_id_name}' expression="TM" >
>>             <action application="log" data="INFO cid_test_3: :
>> '${caller_id_name}'"/>
>>        </condition>
>>     </extension>
>>
>>     <extension name="cid_test_3b" continue="true">
>>        <condition field="${caller_id_name}" expression="TM" >
>>             <action application="log" data="INFO cid_test_3b: :
>> '${caller_id_name}'"/>
>>        </condition>
>>     </extension>
>>
>>     <extension name="cid_test_4" continue="true">
>>        <condition field='${effective_caller_id_name}' expression="TM" >
>>             <action application="log" data="INFO cid_test_4: :
>> '${effective_caller_id_name}'"/>
>>        </condition>
>>     </extension>
>>
>>     <extension name="cid_test_5" continue="true">
>>        <condition field='${effective_caller_id_name}' expression="TM" >
>>             <action application="log" data="INFO cid_test_5: :
>> '${effective_caller_id_name}'"/>
>>        </condition>
>>     </extension>
>>
>> _________________________________________________________________________
>> 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 sergio.garcia at quobis.com  Fri Jul 24 11:18:27 2015
From: sergio.garcia at quobis.com (=?ISO-8859-1?Q?Sergio_Garc=EDa?=)
Date: Fri, 24 Jul 2015 09:18:27 +0200
Subject: [Freeswitch-users] ReINVITE - 488 Not Acceptable Here
In-Reply-To: <CAO3hyzGSVuXMmggC9H7TVQRzPHCp908J1Sr2tS=aav6-kn0Luw@mail.gmail.com>
References: <CAMxfXLpU8+ku1KfLFB7-U-p5UC_s1K+g4zQQuuk-nr4+ZQqhuw@mail.gmail.com>
	<CAO3hyzGSVuXMmggC9H7TVQRzPHCp908J1Sr2tS=aav6-kn0Luw@mail.gmail.com>
Message-ID: <CAMxfXLqJHtLcHMQynn7em9Vzo2OBkmig2g6aX9MEdwsogYi+Hg@mail.gmail.com>

Hi Chris,

I'm not using JSSIP or Sipml5, I'm using our own solution. As I said in the
previous email, we are sending IP 0.0.0.0, port 0 and media inactive. It
seems FS doesn't like this and I would like to know the reason.

I've changed our end to send* a=sendonly* (instead of* a=inactive*) and to
not change the port to 0, and FS accepted it. The problem now is that after
one second hearing the music on hold it stops, and I can never get the
media back (I'm using TURN server to avoid NAT/Firewall problems).

Thank you.

2015-07-23 18:11 GMT+02:00 Chris Tunbridge <blasterjr at gmail.com>:

> Sergio, please fill in some additional information like which JS WebSocket
> client you're using, there are lots of hold issues with SipML5 and JsSIP,
> the only one that i know that has working hold music is sip.js (was a fork
> of JsSIP) and I've personally contributed towards making the hold function
> correctly on sip.js
>
> On Thu, Jul 23, 2015 at 5:17 AM, Sergio Garc?a <sergio.garcia at quobis.com>
> wrote:
>
>> Hello all,
>>
>> I am using *FreeSwitch* as a *WebRTC gateway* thanks to its Websocket
>> support (in my case WSS), but I'm facing this strange problem. Audio and
>> Video calls are working perfectly fine, but when I try to set a call On
>> Hold, FreeSwitch replies with "488 Not Acceptable Here" error to the
>> ReINVITE I'm sending.
>>
>> The only error I can see in the logs is:
>>
>>
>> * [ERR] sofia.c:7280 Reinvite Codec Error!*
>>
>> The only difference between the original INVITE and this ReINVITE is that
>> I try to set IP address to *0.0.0.0*, port to *0* and media attribute to
>> *inactive*. I don't understand what part of the SDP, FS doesn't "like".
>>
>> Attached you can find a more detailed log file.
>>
>> Thank you very much in advance.
>>
>> Regards,
>> --
>>
>> *Sergio Garc?a Ramos *
>> VoIP Engineer @ Quobis <http://www.quobis.com/> | e:
>> sergio.garcia at quobis.com | t: +34902999465
>>
>>
>>
>> _________________________________________________________________________
>> 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
>



-- 

*Sergio Garc?a Ramos *
VoIP Engineer @ Quobis <http://www.quobis.com/> | e:
sergio.garcia at quobis.com | t: +34902999465
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From findmeinwland at gmail.com  Fri Jul 24 13:07:49 2015
From: findmeinwland at gmail.com (Artur Mega)
Date: Fri, 24 Jul 2015 14:07:49 +0500
Subject: [Freeswitch-users] Strange mod_sofia logic
In-Reply-To: <CAJTkRNucm=eZKTt92oQWWVD767=yPBZAwoEyCzoi45+0FVJ-og@mail.gmail.com>
References: <CAJTkRNucm=eZKTt92oQWWVD767=yPBZAwoEyCzoi45+0FVJ-og@mail.gmail.com>
Message-ID: <CAOLEQ7fJSmb4=9r3DRD8uSb7VKaRdQiJ6vWwCU0sqtR2knh2eQ@mail.gmail.com>

very interessting

2015-07-24 2:31 GMT+05:00 Igor Olhovskiy <igorolhovskiy at gmail.com>:

> Hi!
>
> Found really strange behavior, but may be it's correct.
> I'm using FusionPBX as a FreeSwitch GUI, but problem more to FS.
> So, when I'm setting context of profile 'internal' to 'public' (yes, it's
> incorrect, but possible) this profile stop accept registrations. Answering "SIP/2.0
> 405 Method Not Allowed"
> and gives list of methods like
>
> INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,PRACK,NOTIFY,PUBLISH,SUBSCRIBE
> But when changing to 'default, all works correct and list of methods are
> INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER,
> NOTIFY, PUBLISH, SUBSCRIBE
> So the question is: how changing of context can inflict on methods
> supported?  And how to control methods supported in more obvious way? (For
> ex, disable REFER)
> FreeSwitch version is 1.4.20
> --
> Best 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
>



-- 

Arthur
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From jaflong at yandex.com  Fri Jul 24 13:10:25 2015
From: jaflong at yandex.com (jaflong jaflong)
Date: Fri, 24 Jul 2015 12:10:25 +0300
Subject: [Freeswitch-users] api commands not working in outbound socket
Message-ID: <2345411437729025@web18j.yandex.ru>

Hi list

I have a php script run through ivrd.

api commands are returned with "ERR command not found" with done through sendRecv or api functions

any suggestion why it fails?

example 1
---------

code:

$command = "api uuid_getvar db39795e-31e1-11e5-9af9-cfe09f60c9f7 variable_sip_from_user";
$conn->sendRecv($command);


tcpdump:

api uuid_getvar db39795e-31e1-11e5-9af9-cfe09f60c9f7 variable_sip_from_user

Content-Type: command/reply
Reply-Text: -ERR command not found




example 2
---------

code:

$command = "api version";
$conn->sendRecv($command);


tcpdump:

api version

Content-Type: command/reply
Reply-Text: -ERR command not found




example 3
---------

code:

$command = "version";
$conn->api($command);

tcpdump:

api version

Content-Type: command/reply
Reply-Text: -ERR command not found





From zoell at zoell.us  Fri Jul 24 13:57:48 2015
From: zoell at zoell.us (=?UTF-8?B?Wm9sdMOhbiBTemFiw7M=?=)
Date: Fri, 24 Jul 2015 10:57:48 +0100
Subject: [Freeswitch-users] Record in WAV IEEE float format
Message-ID: <CAJrD+vc4pv8Hq6wDeHFnPdXZtcavrheVj2tqa390JUpwUejcjg@mail.gmail.com>

Hi,

I would like to record in Wave IEEE float signed 32bit, 8000hz 256kbps,
mono format. For this I can set the record_stereo to false in the session
and tried to modify mod_sndfile.c to configure the format, so far I tried
like:

if (mode & SFM_WRITE) {
  context->sfinfo.channels = handle->channels;
  context->sfinfo.samplerate = handle->samplerate;
  context->sfinfo.format = SF_FORMAT_WAV | SF_FORMAT_FLOAT;
}

But I receive some weird data when trying to load up the wav in goldwave.
The format looks fine but the data is not.

Do you have any suggestion?

Many thanks,
Zoltan
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From adam.lappe at qsc.de  Fri Jul 24 14:30:18 2015
From: adam.lappe at qsc.de (Lappe, Adam)
Date: Fri, 24 Jul 2015 10:30:18 +0000
Subject: [Freeswitch-users] mod_local_stream moh confusion
In-Reply-To: <CAEJMVkAnYcxOzL19E_Zs9Bn73zx8gAtGuTCYGZPrqsFhVb6n3g@mail.gmail.com>
References: <2E67ADAE1D3582409C90EBAE8C64C5070E5569@QSCDEMXP01B.ONE4ALL.LAN>,
	<CAEJMVkAnYcxOzL19E_Zs9Bn73zx8gAtGuTCYGZPrqsFhVb6n3g@mail.gmail.com>
Message-ID: <2E67ADAE1D3582409C90EBAE8C64C5070E58E0@QSCDEMXP01B.ONE4ALL.LAN>

Hello Brian,

thanks for your answer.

You are correct. As you can see in my first post, i do not use mod_local_stream.
The question is: WHY does FreeSWITCH try to use a local_stream for moh, when a user gets transfered (with an reinvite)?

I grep'ed my config files, i don't load or use mod_local_stream anywhere.

Best regards,
Adam
________________________________________
Von: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org]&quot; im Auftrag von &quot;Brian West [brian at freeswitch.org]
Gesendet: Donnerstag, 23. Juli 2015 17:09
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] mod_local_stream moh confusion

It would indicate that you do not have mod_local_stream loaded.

On Thu, Jul 23, 2015 at 3:15 AM, Lappe, Adam <adam.lappe at qsc.de<mailto:adam.lappe at qsc.de>> wrote:
Hello,

i am testing the latest 1.4.19 Version of FreeSWITCH.
Currently we are running an old 1.2.7 Version.

Everything seems to work fine, but there is 1 error that is very confusing:

When a call gets transfered by the callee (i.e. by the receptionist) the call will be terminated.
All I see is this error line:
[ERR] switch_core_file.c:149 Invalid file format [local_stream] for [moh]!

I don?t use the local_stream module.
freeswitch at internal> module_exists mod_local_stream
false

This error does not exists with the old version.

Is this a bug, or am I missing something?

Thanks in advance,
Adam

_________________________________________________________________________
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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

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



From ben at langfeld.co.uk  Fri Jul 24 15:34:09 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Fri, 24 Jul 2015 08:34:09 -0300
Subject: [Freeswitch-users] ReINVITE - 488 Not Acceptable Here
In-Reply-To: <CAMxfXLqJHtLcHMQynn7em9Vzo2OBkmig2g6aX9MEdwsogYi+Hg@mail.gmail.com>
References: <CAMxfXLpU8+ku1KfLFB7-U-p5UC_s1K+g4zQQuuk-nr4+ZQqhuw@mail.gmail.com>
	<CAO3hyzGSVuXMmggC9H7TVQRzPHCp908J1Sr2tS=aav6-kn0Luw@mail.gmail.com>
	<CAMxfXLqJHtLcHMQynn7em9Vzo2OBkmig2g6aX9MEdwsogYi+Hg@mail.gmail.com>
Message-ID: <CAAyX+KH3kqBm=8BJXNxP5k4oLpjq4_Uwk9q1DsVV=YQGxibK6g@mail.gmail.com>

Perhaps you could mock up a small sample app using SIP.js, check that it
works properly, and then analyse the differences between what it does and
what your custom SIP stack does, then?

On 24 July 2015 at 04:18, Sergio Garc?a <sergio.garcia at quobis.com> wrote:

> Hi Chris,
>
> I'm not using JSSIP or Sipml5, I'm using our own solution. As I said in
> the previous email, we are sending IP 0.0.0.0, port 0 and media inactive.
> It seems FS doesn't like this and I would like to know the reason.
>
> I've changed our end to send* a=sendonly* (instead of* a=inactive*) and
> to not change the port to 0, and FS accepted it. The problem now is that
> after one second hearing the music on hold it stops, and I can never get
> the media back (I'm using TURN server to avoid NAT/Firewall problems).
>
> Thank you.
>
> 2015-07-23 18:11 GMT+02:00 Chris Tunbridge <blasterjr at gmail.com>:
>
>> Sergio, please fill in some additional information like which JS
>> WebSocket client you're using, there are lots of hold issues with SipML5
>> and JsSIP, the only one that i know that has working hold music is sip.js
>> (was a fork of JsSIP) and I've personally contributed towards making the
>> hold function correctly on sip.js
>>
>> On Thu, Jul 23, 2015 at 5:17 AM, Sergio Garc?a <sergio.garcia at quobis.com>
>> wrote:
>>
>>> Hello all,
>>>
>>> I am using *FreeSwitch* as a *WebRTC gateway* thanks to its Websocket
>>> support (in my case WSS), but I'm facing this strange problem. Audio and
>>> Video calls are working perfectly fine, but when I try to set a call On
>>> Hold, FreeSwitch replies with "488 Not Acceptable Here" error to the
>>> ReINVITE I'm sending.
>>>
>>> The only error I can see in the logs is:
>>>
>>>
>>> * [ERR] sofia.c:7280 Reinvite Codec Error!*
>>>
>>> The only difference between the original INVITE and this ReINVITE is
>>> that I try to set IP address to *0.0.0.0*, port to *0* and media
>>> attribute to *inactive*. I don't understand what part of the SDP, FS
>>> doesn't "like".
>>>
>>> Attached you can find a more detailed log file.
>>>
>>> Thank you very much in advance.
>>>
>>> Regards,
>>> --
>>>
>>> *Sergio Garc?a Ramos *
>>> VoIP Engineer @ Quobis <http://www.quobis.com/> | e:
>>> sergio.garcia at quobis.com | t: +34902999465
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> 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
>>
>
>
>
> --
>
> *Sergio Garc?a Ramos *
> VoIP Engineer @ Quobis <http://www.quobis.com/> | e:
> sergio.garcia at quobis.com | t: +34902999465
>
>
>
> _________________________________________________________________________
> 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  Fri Jul 24 15:38:22 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 24 Jul 2015 13:38:22 +0200
Subject: [Freeswitch-users] ReINVITE - 488 Not Acceptable Here
In-Reply-To: <CAAyX+KH3kqBm=8BJXNxP5k4oLpjq4_Uwk9q1DsVV=YQGxibK6g@mail.gmail.com>
References: <CAMxfXLpU8+ku1KfLFB7-U-p5UC_s1K+g4zQQuuk-nr4+ZQqhuw@mail.gmail.com>
	<CAO3hyzGSVuXMmggC9H7TVQRzPHCp908J1Sr2tS=aav6-kn0Luw@mail.gmail.com>
	<CAMxfXLqJHtLcHMQynn7em9Vzo2OBkmig2g6aX9MEdwsogYi+Hg@mail.gmail.com>
	<CAAyX+KH3kqBm=8BJXNxP5k4oLpjq4_Uwk9q1DsVV=YQGxibK6g@mail.gmail.com>
Message-ID: <CALXCt0pAN0aBze44vDB6g6LfneYFuzQdb=WrKF-zOF2ZWm0bRQ@mail.gmail.com>

why don't you check what sip.js do (sip.js is the implementation of sip in
javascript better working with FS), then you model your solution behavior
on it...

On Fri, Jul 24, 2015 at 1:34 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:

> Perhaps you could mock up a small sample app using SIP.js, check that it
> works properly, and then analyse the differences between what it does and
> what your custom SIP stack does, then?
>
> On 24 July 2015 at 04:18, Sergio Garc?a <sergio.garcia at quobis.com> wrote:
>
>> Hi Chris,
>>
>> I'm not using JSSIP or Sipml5, I'm using our own solution. As I said in
>> the previous email, we are sending IP 0.0.0.0, port 0 and media inactive.
>> It seems FS doesn't like this and I would like to know the reason.
>>
>> I've changed our end to send* a=sendonly* (instead of* a=inactive*) and
>> to not change the port to 0, and FS accepted it. The problem now is that
>> after one second hearing the music on hold it stops, and I can never get
>> the media back (I'm using TURN server to avoid NAT/Firewall problems).
>>
>> Thank you.
>>
>> 2015-07-23 18:11 GMT+02:00 Chris Tunbridge <blasterjr at gmail.com>:
>>
>>> Sergio, please fill in some additional information like which JS
>>> WebSocket client you're using, there are lots of hold issues with SipML5
>>> and JsSIP, the only one that i know that has working hold music is sip.js
>>> (was a fork of JsSIP) and I've personally contributed towards making the
>>> hold function correctly on sip.js
>>>
>>> On Thu, Jul 23, 2015 at 5:17 AM, Sergio Garc?a <sergio.garcia at quobis.com
>>> > wrote:
>>>
>>>> Hello all,
>>>>
>>>> I am using *FreeSwitch* as a *WebRTC gateway* thanks to its Websocket
>>>> support (in my case WSS), but I'm facing this strange problem. Audio and
>>>> Video calls are working perfectly fine, but when I try to set a call On
>>>> Hold, FreeSwitch replies with "488 Not Acceptable Here" error to the
>>>> ReINVITE I'm sending.
>>>>
>>>> The only error I can see in the logs is:
>>>>
>>>>
>>>> * [ERR] sofia.c:7280 Reinvite Codec Error!*
>>>>
>>>> The only difference between the original INVITE and this ReINVITE is
>>>> that I try to set IP address to *0.0.0.0*, port to *0* and media
>>>> attribute to *inactive*. I don't understand what part of the SDP, FS
>>>> doesn't "like".
>>>>
>>>> Attached you can find a more detailed log file.
>>>>
>>>> Thank you very much in advance.
>>>>
>>>> Regards,
>>>> --
>>>>
>>>> *Sergio Garc?a Ramos *
>>>> VoIP Engineer @ Quobis <http://www.quobis.com/> | e:
>>>> sergio.garcia at quobis.com | t: +34902999465
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>>>
>>
>>
>>
>> --
>>
>> *Sergio Garc?a Ramos *
>> VoIP Engineer @ Quobis <http://www.quobis.com/> | e:
>> sergio.garcia at quobis.com | t: +34902999465
>>
>>
>>
>> _________________________________________________________________________
>> 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
Cell : +39-347-2665618
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From menendez.garcia at gmail.com  Fri Jul 24 17:08:47 2015
From: menendez.garcia at gmail.com (Javier Menendez)
Date: Fri, 24 Jul 2015 15:08:47 +0200
Subject: [Freeswitch-users] Firefox 39, webrtc
In-Reply-To: <CA+3gBcnCz3tgcyiDgXdnY7dddbnbY9s+kbG8TrDR9BJHExN-Gw@mail.gmail.com>
References: <CAJTkRNsXxSV_WYRk6dY9WN=kPuv_h2Y+By+j7KUmCBUx8k8zyw@mail.gmail.com>
	<CAEJMVkCqc8y-ne0SOAuZNhxYe159HjNGi0acxMnZQm-+FQtEaQ@mail.gmail.com>
	<CAJTkRNugkoY6c_kod0E+bZsKGDN23c9bXhvfh+C8bw7QOoCCow@mail.gmail.com>
	<CA+3gBcnCz3tgcyiDgXdnY7dddbnbY9s+kbG8TrDR9BJHExN-Gw@mail.gmail.com>
Message-ID: <CAK59s4wmqUwFRn_b1XdH3t6n56jFwqAVnrj_YRPqdAu=bJDo7g@mail.gmail.com>

I agree with Carlos, it's a bug, I have adapted the patch to FS v 1.4.20
and it is working,

On Wed, Jul 15, 2015 at 6:01 PM, Carlos Ruiz D?az <carlos.ruizdiaz at gmail.com
> wrote:

> There's a high change this being a bug in FS, caused by FF removing lots
> of cipher suites it used to use for the handshake.
>
> I experienced this myself with rtpengine.
>
> This is the commit that fixed the issue on that project, in case it may be
> useful for the maintainer(s) in charge of webRTC in FS. [1]
>
> [1]
> https://github.com/sipwise/rtpengine/commit/21e1fb680762f421e05fab036f8138b2276f5037
>
> On Wed, Jul 15, 2015 at 10:47 AM, Igor Olhovskiy <igorolhovskiy at gmail.com>
> wrote:
>
>> I'm trying to figure out if it is a bug of FS :)
>> So asking if someone is having the same )
>>
>> 2015-07-15 18:08 GMT+03:00 Brian West <brian at freeswitch.org>:
>>
>>> I would like to know what we can do to get the point across about filing
>>> bugs on JIRA?
>>>
>>> Even if you file something and it ends up not being a bug, we can at the
>>> very least close it as 'not a bug'.  What we can't do is mentally track
>>> every last issue thats reported on the list.
>>>
>>> https://www.youtube.com/watch?v=lxB57TtX27I
>>>
>>> Thanks,
>>>
>>>
>>>
>>> On Wed, Jul 15, 2015 at 5:55 AM, Igor Olhovskiy <igorolhovskiy at gmail.com
>>> > wrote:
>>>
>>>> Hi!
>>>>
>>>> I'm using sip.js with FS (tried FS 1.5.15 and 1.7). Upgrading to
>>>> Firefox 39 is seems to be bad idea, cause call through it fails due to
>>>>
>>>> 2015-07-15 10:48:39.168250 [ERR] switch_rtp.c:2907 audio Handshake
>>>> failure 1
>>>> 2015-07-15 10:48:39.168250 [INFO] switch_rtp.c:2908 Changing audio DTLS
>>>> state from HANDSHAKE to FAIL
>>>>
>>>> On FF 38 and Chrome all is ok.
>>>> Am I only one with this issue of it's really something wrong with FF?
>>>>
>>>> --
>>>> Best 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
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *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
>>>
>>> _________________________________________________________________________
>>> 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
>>>
>>
>>
>>
>> --
>> Best 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
>>
>
>
>
> --
> Carlos
> http://caruizdiaz.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
>
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From rafael.kaihatu at 8x8.com  Fri Jul 24 17:19:57 2015
From: rafael.kaihatu at 8x8.com (Rafael Kaihatu)
Date: Fri, 24 Jul 2015 10:19:57 -0300
Subject: [Freeswitch-users] question about eavesdrop
Message-ID: <CAKMRjL1jhBoVpTMcCvGuWsRQ+b+BKSWyPDzUuy_0rRrY1szttg@mail.gmail.com>

Hi All,

Let's say I have a 2-party call (leg A and B talking with each other)
I would like to connect a 3rd leg to hear only A

I understood that I can configure with eavesdrop, if the 3rd leg can speak
with A or B or both or neither.

is it possible to do that with eavesdrop?
is there another command?

and eventually, I want to connect a 4th leg to hear only B

Thanks
Rafael Kaihatu
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From steven.szeto at mitel.com  Fri Jul 24 17:36:04 2015
From: steven.szeto at mitel.com (Steven Szeto)
Date: Fri, 24 Jul 2015 13:36:04 +0000
Subject: [Freeswitch-users] question about eavesdrop
In-Reply-To: <CAKMRjL1jhBoVpTMcCvGuWsRQ+b+BKSWyPDzUuy_0rRrY1szttg@mail.gmail.com>
References: <CAKMRjL1jhBoVpTMcCvGuWsRQ+b+BKSWyPDzUuy_0rRrY1szttg@mail.gmail.com>
Message-ID: <BY2PR12MB02423670535BE7899FC2D0188A810@BY2PR12MB0242.namprd12.prod.outlook.com>

Rafael,


When creating the eavesdrop session, first try setting eavesdrop_bridge_aleg=true or eavesdrop_bridge_bleg=true


/Steve


________________________________
From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Rafael Kaihatu <rafael.kaihatu at 8x8.com>
Sent: Friday, July 24, 2015 9:19 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] question about eavesdrop

Hi All,

Let's say I have a 2-party call (leg A and B talking with each other)
I would like to connect a 3rd leg to hear only A

I understood that I can configure with eavesdrop, if the 3rd leg can speak with A or B or both or neither.

is it possible to do that with eavesdrop?
is there another command?

and eventually, I want to connect a 4th leg to hear only B

Thanks
Rafael Kaihatu

________________________________
NOTE: This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message.
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From mike at jerris.com  Fri Jul 24 17:49:20 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 24 Jul 2015 09:49:20 -0400
Subject: [Freeswitch-users] api commands not working in outbound socket
In-Reply-To: <2345411437729025@web18j.yandex.ru>
References: <2345411437729025@web18j.yandex.ru>
Message-ID: <CAEwTsAisTJCkiFc4jZkZ7y-E2GiVVCBfOQdYf1h_piArrkEEAQ@mail.gmail.com>

do the commands work in fs_cli?  if not, are you sure mod_commands is
loaded?

On Friday, July 24, 2015, jaflong jaflong <jaflong at yandex.com> wrote:

> Hi list
>
> I have a php script run through ivrd.
>
> api commands are returned with "ERR command not found" with done through
> sendRecv or api functions
>
> any suggestion why it fails?
>
> example 1
> ---------
>
> code:
>
> $command = "api uuid_getvar db39795e-31e1-11e5-9af9-cfe09f60c9f7
> variable_sip_from_user";
> $conn->sendRecv($command);
>
>
> tcpdump:
>
> api uuid_getvar db39795e-31e1-11e5-9af9-cfe09f60c9f7 variable_sip_from_user
>
> Content-Type: command/reply
> Reply-Text: -ERR command not found
>
>
>
>
> example 2
> ---------
>
> code:
>
> $command = "api version";
> $conn->sendRecv($command);
>
>
> tcpdump:
>
> api version
>
> Content-Type: command/reply
> Reply-Text: -ERR command not found
>
>
>
>
> example 3
> ---------
>
> code:
>
> $command = "version";
> $conn->api($command);
>
> tcpdump:
>
> api version
>
> Content-Type: command/reply
> Reply-Text: -ERR command not found
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> 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 <javascript:;>
> 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 igorolhovskiy at gmail.com  Fri Jul 24 17:53:49 2015
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Fri, 24 Jul 2015 16:53:49 +0300
Subject: [Freeswitch-users] Firefox 39, webrtc
In-Reply-To: <CAK59s4wmqUwFRn_b1XdH3t6n56jFwqAVnrj_YRPqdAu=bJDo7g@mail.gmail.com>
References: <CAJTkRNsXxSV_WYRk6dY9WN=kPuv_h2Y+By+j7KUmCBUx8k8zyw@mail.gmail.com>
	<CAEJMVkCqc8y-ne0SOAuZNhxYe159HjNGi0acxMnZQm-+FQtEaQ@mail.gmail.com>
	<CAJTkRNugkoY6c_kod0E+bZsKGDN23c9bXhvfh+C8bw7QOoCCow@mail.gmail.com>
	<CA+3gBcnCz3tgcyiDgXdnY7dddbnbY9s+kbG8TrDR9BJHExN-Gw@mail.gmail.com>
	<CAK59s4wmqUwFRn_b1XdH3t6n56jFwqAVnrj_YRPqdAu=bJDo7g@mail.gmail.com>
Message-ID: <CAJTkRNuZLupoO7jB5mDAEe_QhL7n=3L4hK9guxSctavhPZieLA@mail.gmail.com>

I've post bug in jira already )

2015-07-24 16:08 GMT+03:00 Javier Menendez <menendez.garcia at gmail.com>:

> I agree with Carlos, it's a bug, I have adapted the patch to FS v 1.4.20
> and it is working,
>
> On Wed, Jul 15, 2015 at 6:01 PM, Carlos Ruiz D?az <
> carlos.ruizdiaz at gmail.com> wrote:
>
>> There's a high change this being a bug in FS, caused by FF removing lots
>> of cipher suites it used to use for the handshake.
>>
>> I experienced this myself with rtpengine.
>>
>> This is the commit that fixed the issue on that project, in case it may
>> be useful for the maintainer(s) in charge of webRTC in FS. [1]
>>
>> [1]
>> https://github.com/sipwise/rtpengine/commit/21e1fb680762f421e05fab036f8138b2276f5037
>>
>> On Wed, Jul 15, 2015 at 10:47 AM, Igor Olhovskiy <igorolhovskiy at gmail.com
>> > wrote:
>>
>>> I'm trying to figure out if it is a bug of FS :)
>>> So asking if someone is having the same )
>>>
>>> 2015-07-15 18:08 GMT+03:00 Brian West <brian at freeswitch.org>:
>>>
>>>> I would like to know what we can do to get the point across about
>>>> filing bugs on JIRA?
>>>>
>>>> Even if you file something and it ends up not being a bug, we can at
>>>> the very least close it as 'not a bug'.  What we can't do is mentally track
>>>> every last issue thats reported on the list.
>>>>
>>>> https://www.youtube.com/watch?v=lxB57TtX27I
>>>>
>>>> Thanks,
>>>>
>>>>
>>>>
>>>> On Wed, Jul 15, 2015 at 5:55 AM, Igor Olhovskiy <
>>>> igorolhovskiy at gmail.com> wrote:
>>>>
>>>>> Hi!
>>>>>
>>>>> I'm using sip.js with FS (tried FS 1.5.15 and 1.7). Upgrading to
>>>>> Firefox 39 is seems to be bad idea, cause call through it fails due to
>>>>>
>>>>> 2015-07-15 10:48:39.168250 [ERR] switch_rtp.c:2907 audio Handshake
>>>>> failure 1
>>>>> 2015-07-15 10:48:39.168250 [INFO] switch_rtp.c:2908 Changing audio
>>>>> DTLS state from HANDSHAKE to FAIL
>>>>>
>>>>> On FF 38 and Chrome all is ok.
>>>>> Am I only one with this issue of it's really something wrong with FF?
>>>>>
>>>>> --
>>>>> Best 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
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> *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
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>>>>
>>>
>>>
>>>
>>> --
>>> Best 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
>>>
>>
>>
>>
>> --
>> Carlos
>> http://caruizdiaz.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
>>
>
>
> _________________________________________________________________________
> 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
>



-- 
Best regards,
Igor
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From mike at jerris.com  Fri Jul 24 17:55:10 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 24 Jul 2015 09:55:10 -0400
Subject: [Freeswitch-users] Firefox 39, webrtc
In-Reply-To: <CAK59s4wmqUwFRn_b1XdH3t6n56jFwqAVnrj_YRPqdAu=bJDo7g@mail.gmail.com>
References: <CAJTkRNsXxSV_WYRk6dY9WN=kPuv_h2Y+By+j7KUmCBUx8k8zyw@mail.gmail.com>
	<CAEJMVkCqc8y-ne0SOAuZNhxYe159HjNGi0acxMnZQm-+FQtEaQ@mail.gmail.com>
	<CAJTkRNugkoY6c_kod0E+bZsKGDN23c9bXhvfh+C8bw7QOoCCow@mail.gmail.com>
	<CA+3gBcnCz3tgcyiDgXdnY7dddbnbY9s+kbG8TrDR9BJHExN-Gw@mail.gmail.com>
	<CAK59s4wmqUwFRn_b1XdH3t6n56jFwqAVnrj_YRPqdAu=bJDo7g@mail.gmail.com>
Message-ID: <CAEwTsAggtyUV=XDokQ4io4d=te6R4ZtAjTqs4oNp9jL_yL4pdA@mail.gmail.com>

could someone submit a pull request against master please

On Friday, July 24, 2015, Javier Menendez <menendez.garcia at gmail.com> wrote:

> I agree with Carlos, it's a bug, I have adapted the patch to FS v 1.4.20
> and it is working,
>
> On Wed, Jul 15, 2015 at 6:01 PM, Carlos Ruiz D?az <
> carlos.ruizdiaz at gmail.com
> <javascript:_e(%7B%7D,'cvml','carlos.ruizdiaz at gmail.com');>> wrote:
>
>> There's a high change this being a bug in FS, caused by FF removing lots
>> of cipher suites it used to use for the handshake.
>>
>> I experienced this myself with rtpengine.
>>
>> This is the commit that fixed the issue on that project, in case it may
>> be useful for the maintainer(s) in charge of webRTC in FS. [1]
>>
>> [1]
>> https://github.com/sipwise/rtpengine/commit/21e1fb680762f421e05fab036f8138b2276f5037
>>
>> On Wed, Jul 15, 2015 at 10:47 AM, Igor Olhovskiy <igorolhovskiy at gmail.com
>> <javascript:_e(%7B%7D,'cvml','igorolhovskiy at gmail.com');>> wrote:
>>
>>> I'm trying to figure out if it is a bug of FS :)
>>> So asking if someone is having the same )
>>>
>>> 2015-07-15 18:08 GMT+03:00 Brian West <brian at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','brian at freeswitch.org');>>:
>>>
>>>> I would like to know what we can do to get the point across about
>>>> filing bugs on JIRA?
>>>>
>>>> Even if you file something and it ends up not being a bug, we can at
>>>> the very least close it as 'not a bug'.  What we can't do is mentally track
>>>> every last issue thats reported on the list.
>>>>
>>>> Daffy Duck Goes Completely Insane
>>>> <https://www.youtube.com/watch?v=lxB57TtX27I>
>>>>
>>>> Thanks,
>>>>
>>>>
>>>>
>>>> On Wed, Jul 15, 2015 at 5:55 AM, Igor Olhovskiy <
>>>> igorolhovskiy at gmail.com
>>>> <javascript:_e(%7B%7D,'cvml','igorolhovskiy at gmail.com');>> wrote:
>>>>
>>>>> Hi!
>>>>>
>>>>> I'm using sip.js with FS (tried FS 1.5.15 and 1.7). Upgrading to
>>>>> Firefox 39 is seems to be bad idea, cause call through it fails due to
>>>>>
>>>>> 2015-07-15 10:48:39.168250 [ERR] switch_rtp.c:2907 audio Handshake
>>>>> failure 1
>>>>> 2015-07-15 10:48:39.168250 [INFO] switch_rtp.c:2908 Changing audio
>>>>> DTLS state from HANDSHAKE to FAIL
>>>>>
>>>>> On FF 38 and Chrome all is ok.
>>>>> Am I only one with this issue of it's really something wrong with FF?
>>>>>
>>>>> --
>>>>> Best regards,
>>>>> Igor
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','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
>>>>> <javascript:_e(%7B%7D,'cvml','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
>>>> <javascript:_e(%7B%7D,'cvml','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
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> <javascript:_e(%7B%7D,'cvml','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
>>>> <javascript:_e(%7B%7D,'cvml','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
>>>>
>>>
>>>
>>>
>>> --
>>> Best regards,
>>> Igor
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','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
>>> <javascript:_e(%7B%7D,'cvml','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
>> http://caruizdiaz.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','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
>> <javascript:_e(%7B%7D,'cvml','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 freeswitch.org  Fri Jul 24 18:03:19 2015
From: krice at freeswitch.org (krice at freeswitch.org)
Date: Fri, 24 Jul 2015 14:03:19 +0000
Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <55b245a7f01c_fc78a47338334f2@resque-worker-high.0.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
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From yadenis at seznam.cz  Fri Jul 24 18:03:48 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Fri, 24 Jul 2015 16:03:48 +0200
Subject: [Freeswitch-users] The problem with the TURN server and
	connection.
In-Reply-To: <CALXCt0qrhSJ=gHmnwu2pBsNWwnKD_447S1_JL0UBH2fK2KzWgQ@mail.gmail.com>
References: <loom.20150722T161450-208@post.gmane.org>
	<778052174.20150723102530@seznam.cz> 
	<CAAyX+KFmuLQVRkP-Au5cN_Tu5aLD0377r8fwrxx3+2Q3E+hiCw@mail.gmail.com>
	<762006319.20150723152412@seznam.cz>
	<CALXCt0qrhSJ=gHmnwu2pBsNWwnKD_447S1_JL0UBH2fK2KzWgQ@mail.gmail.com>
Message-ID: <823909165.20150724160348@seznam.cz>

Dobr? den,

many thanks. It works as it should. I have been a mistake in configuring the turn server.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

?tvrtek 23. ?ervence 2015, 15:28:45, napsal jste:


pay attention, you MUST use the "transport" argument for tcp to work, eg:

turnServers: {
  urls:"turn:exam.org:443?transport=tcp",
  username:"alice",
  password:"racecar"
}

On Thu, Jul 23, 2015 at 3:24 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
Dobr? den,

That's it I use

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

?tvrtek 23. ?ervence 2015, 13:56:43, napsal jste:


Take a look at coturn, a fork of rfc5766-turn-server, which supports RFC6062, the TCP transport for TURN.

On 23 July 2015 at 05:25, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi All,

I have one problem. 
One of my clients is behind the serious firewall and they are banned UDP connection at all. His clients are connected to it via the Internet. 
I put my own Turn/Stun server (rfc5766-turn-server from Google). It works the same server as the freeswitch. It works fine.
The problem follows. When all banned UDP connection - you can not connect (INCOMPATIBLE_DESTINATION). If you allow outgoing UDP At least, that is connected with no problems.
Is it even possible connection if one of the parties completely banned UDP connection? 

I thought that the problem is just the decides the turn server. But I is not working for me.

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

_________________________________________________________________________
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
Cell : +39-347-2665618
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From nzaytsevc at gmail.com  Fri Jul 24 17:26:55 2015
From: nzaytsevc at gmail.com (Nikolay Zaytsev)
Date: Fri, 24 Jul 2015 17:26:55 +0400
Subject: [Freeswitch-users] force-register-domain
Message-ID: <CADeoHuoyYZ86Hw-qsvfM25NyTbGeV_YbFANfPW1j25nTQh6_1g@mail.gmail.com>

Hello,everyone.
Ive tried to use a1-hash instead of username and password.(when i use plain
username and password everything works fine).
However, freeswitch does not allow users to register,when i use domain
which is different from ip_address of the Freeswitch server.
I am using domain - nick.sfedu.
Domain in vars.xml,directory/default.conf is set as nick.sfedu and
force-register-domain in internal profile is set to the same value.
When I use plain username and password, i can see the following:

Call-ID:        NGJhMDU5NTM0MDllZWY2ODllZjFjYzMwYzAzMzQxMTc
User:           11007 at nick.sfedu
Contact:        "Nick" <sip:11007 at 10.132.144.157:49610
;rinstance=fd2236fab37d0ba1;fs_nat=yes;fs_path=sip%3A11007%4010.132.144.157%3A49610%3Brinstance%3Dfd2236fab37d0ba1>
Agent:          X-Lite 4.7.1 74247-a844961e-W6.1
Status:         Registered(UDP-NAT)(unknown) EXP(2015-07-24 14:55:08)
EXPSECS(144)
Host:           fsw1-test
IP:             10.132.144.157
Port:           49610
Auth-User:      11007
Auth-Realm:     10.160.1.23
MWI-Account:    11007 at nick.sfedu

where 10.160.1.23 is ip address of the Freeswitch server,but it should be
nick.sfedu,as far as i understand.
Best regards,
Nikolay Zaytsev
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From ben at langfeld.co.uk  Fri Jul 24 18:30:59 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Fri, 24 Jul 2015 11:30:59 -0300
Subject: [Freeswitch-users] The problem with the TURN server and
	connection.
In-Reply-To: <823909165.20150724160348@seznam.cz>
References: <loom.20150722T161450-208@post.gmane.org>
	<778052174.20150723102530@seznam.cz>
	<CAAyX+KFmuLQVRkP-Au5cN_Tu5aLD0377r8fwrxx3+2Q3E+hiCw@mail.gmail.com>
	<762006319.20150723152412@seznam.cz>
	<CALXCt0qrhSJ=gHmnwu2pBsNWwnKD_447S1_JL0UBH2fK2KzWgQ@mail.gmail.com>
	<823909165.20150724160348@seznam.cz>
Message-ID: <CAAyX+KGQtgZ7yPNLh7S8ADjTMYCV3rQ1-dy21Jn+3xAsWGR6LA@mail.gmail.com>

Could you explain, for the benefit of those finding this thread in the
future, what you did to fix it?

On 24 July 2015 at 11:03, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Dobr? den,
>
> many thanks. It works as it should. I have been a mistake in configuring
> the turn server.
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 ?tvrtek 23. ?ervence 2015, 15:28:45, napsal
> jste: *
>
>  pay attention, you MUST use the "transport" argument for tcp to work, eg:
>
> turnServers: {
>  urls:"turn:exam.org:443?transport=tcp",
>  username:"alice",
>  password:"racecar"
> }
>
> On Thu, Jul 23, 2015 at 3:24 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Dobr? den,
>
> That's it I use
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
>
>
> *. 775-415-382 ?tvrtek 23. ?ervence 2015, 13:56:43, napsal jste: *
>  Take a look at coturn, a fork of rfc5766-turn-server, which supports
> RFC6062, the TCP transport for TURN.
>
> On 23 July 2015 at 05:25, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Hi All,
>
> I have one problem.
> One of my clients is behind the serious firewall and they are banned UDP
> connection at all. His clients are connected to it via the Internet.
> I put my own Turn/Stun server (rfc5766-turn-server from Google). It works
> the same server as the freeswitch. It works fine.
> The problem follows. When all banned UDP connection - you can not connect
> (INCOMPATIBLE_DESTINATION). If you allow outgoing UDP At least, that is
> connected with no problems.
> Is it even possible connection if one of the parties completely banned UDP
> connection?
>
> I thought that the problem is just the decides the turn server. But I is
> not working for me.
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
> *. 775-415-382 *
> _________________________________________________________________________
> 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
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> 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 rafael.kaihatu at 8x8.com  Fri Jul 24 19:42:00 2015
From: rafael.kaihatu at 8x8.com (Rafael Kaihatu)
Date: Fri, 24 Jul 2015 12:42:00 -0300
Subject: [Freeswitch-users] question about eavesdrop
In-Reply-To: <BY2PR12MB02423670535BE7899FC2D0188A810@BY2PR12MB0242.namprd12.prod.outlook.com>
References: <CAKMRjL1jhBoVpTMcCvGuWsRQ+b+BKSWyPDzUuy_0rRrY1szttg@mail.gmail.com>
	<BY2PR12MB02423670535BE7899FC2D0188A810@BY2PR12MB0242.namprd12.prod.outlook.com>
Message-ID: <CAKMRjL3oXKXBMsjSBu78gmM0MvP461ChxZf5TVp-C1_Tc8zj_Q@mail.gmail.com>

Thanks Steve,

that is what I needed

Note for others:
feature implemented by: https://freeswitch.org/jira/browse/FS-7285
available in 1.4.16


On Fri, Jul 24, 2015 at 10:36 AM, Steven Szeto <steven.szeto at mitel.com>
wrote:

>  Rafael,
>
>
>  When creating the eavesdrop session, first try
> setting eavesdrop_bridge_aleg=true or eavesdrop_bridge_bleg=true
>
>
>  /Steve
>
>
>  ------------------------------
> *From:* freeswitch-users-bounces at lists.freeswitch.org <
> freeswitch-users-bounces at lists.freeswitch.org> on behalf of Rafael
> Kaihatu <rafael.kaihatu at 8x8.com>
> *Sent:* Friday, July 24, 2015 9:19 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] question about eavesdrop
>
>  Hi All,
>
>  Let's say I have a 2-party call (leg A and B talking with each other)
> I would like to connect a 3rd leg to hear only A
>
>  I understood that I can configure with eavesdrop, if the 3rd leg can
> speak with A or B or both or neither.
>
>  is it possible to do that with eavesdrop?
> is there another command?
>
>  and eventually, I want to connect a 4th leg to hear only B
>
>  Thanks
> Rafael Kaihatu
>
> ------------------------------
> NOTE: This e-mail (including any attachments) is for the sole use of the
> intended recipient(s) and may contain information that is confidential
> and/or protected by legal privilege. Any unauthorized review, use, copy,
> disclosure or distribution of this e-mail is strictly prohibited. If you
> are not the intended recipient, please notify Mitel immediately and destroy
> all copies of this e-mail. Mitel does not accept any liability for breach
> of security, error or virus that may result from the transmission of this
> message.
>
>
> _________________________________________________________________________
> 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 Peter.Stevens at bbc.co.uk  Fri Jul 24 19:55:31 2015
From: Peter.Stevens at bbc.co.uk (Peter Stevens)
Date: Fri, 24 Jul 2015 15:55:31 +0000
Subject: [Freeswitch-users] FS 1.6 video dependency build issue
Message-ID: <2F938AAE444ADB47AFEE6307955A30CE35DAE63B@BGB01XUD1002.national.core.bbc.co.uk>

Hi,



I have a clean install of Debian 8 on a 32 bit machine.



I'm following information on FS 1.6 video wiki (https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video) to<https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video)%20to> build this version, but the installation of the dependencies part doesn't install all of the required dependencies.



After running:

DEBIAN_FRONTEND=none APT_LISTCHANGES_FRONTEND=none apt-get install -y --force-yes freeswitch-video-deps-most

I had to then:

apt-get install libavcodec-extra

apt-get install portaudio19-dev

Which were both ok, but running:

apt-get install libsngtc-dev



I received following:

Package libsngtc-dev is not available, but is referred to by another package.

This may mean that the package is missing, has been obsoleted, or

is only available from another source

E: Package 'libsngtc-dev' has no installation candidate



Re-ran:

DEBIAN_FRONTEND=none APT_LISTCHANGES_FRONTEND=none apt-get install -y --force-yes freeswitch-video-deps-most

Reading package lists... Done

Building dependency tree

Reading state information... Done

Some packages could not be installed. This may mean that you have

requested an impossible situation or if you are using the unstable

distribution that some required packages have not yet been created

or been moved out of Incoming.

The following information may help to resolve the situation:

The following packages have unmet dependencies:

freeswitch-video-deps-most : Depends: libsngtc-dev but it is not installable

E: Unable to correct problems, you have held broken packages.



I've had a look around amongst the http://files.freeswitch.org/repo/deb/debian/dists/jessie/main directory structure and have discovered that the libsngtc-dev is listed in both the binary-amd64/packages.gz and the binary-amd64/packages.gz files, but is only listed in the Contents-amd64.gz file and not in the Contents-i386.gz file.

Don't know whether this is significant or not, but how/where can I obtain the libsngtc-dev file to complete the dependency installation for freeswitch-video-deps-most?



Any help appreciated.



Peter





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

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From krice at freeswitch.org  Fri Jul 24 20:05:42 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 24 Jul 2015 11:05:42 -0500
Subject: [Freeswitch-users] FS 1.6 video dependency build issue
Message-ID: <37E12F60-F093-4717-B2CE-55244C5FFCB4@freeswitch.org>

Unless you are using sangoma transcoding hardware you don?t need libsngtc

And why 32bit? 

From:  <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Peter Stevens
Reply-To:  FreeSWITCH Users Help
Date:  Friday, July 24, 2015 at 10:55 AM
To:  "freeswitch-users at lists.freeswitch.org"
Subject:  [Freeswitch-users] FS 1.6 video dependency build issue

Hi,

 

I have a clean install of Debian 8 on a 32 bit machine.

 

I'm following information on FS 1.6 video wiki (https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video) to build this version, but the installation of the dependencies part doesn't install all of the required dependencies.

 

After running:

DEBIAN_FRONTEND=none APT_LISTCHANGES_FRONTEND=none apt-get install -y --force-yesfreeswitch-video-deps-most
 
I had to then:
apt-get install libavcodec-extra

apt-get install portaudio19-dev

Which were both ok, but running:

apt-get install libsngtc-dev

 

I received following:

Package libsngtc-dev is not available, but is referred to by another package.

This may mean that the package is missing, has been obsoleted, or

is only available from another source

E: Package 'libsngtc-dev' has no installation candidate

 

Re-ran:

DEBIAN_FRONTEND=none APT_LISTCHANGES_FRONTEND=none apt-get install -y --force-yes freeswitch-video-deps-most

Reading package lists... Done

Building dependency tree 

Reading state information... Done

Some packages could not be installed. This may mean that you have

requested an impossible situation or if you are using the unstable

distribution that some required packages have not yet been created

or been moved out of Incoming.

The following information may help to resolve the situation:

The following packages have unmet dependencies:

freeswitch-video-deps-most : Depends: libsngtc-dev but it is not installable

E: Unable to correct problems, you have held broken packages.

 

I've had a look around amongst the http://files.freeswitch.org/repo/deb/debian/dists/jessie/main directory structure and have discovered that the libsngtc-dev is listed in both the binary-amd64/packages.gz and the binary-amd64/packages.gz files, but is only listed in the Contents-amd64.gz file and not in the Contents-i386.gz file.

Don't know whether this is significant or not, but how/where can I obtain the libsngtc-dev file to complete the dependency installation for freeswitch-video-deps-most?

 

Any help appreciated.

 

Peter

 

 

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

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Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately.
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From mike at jerris.com  Fri Jul 24 20:17:10 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 24 Jul 2015 12:17:10 -0400
Subject: [Freeswitch-users] FS 1.6 video dependency build issue
In-Reply-To: <2F938AAE444ADB47AFEE6307955A30CE35DAE63B@BGB01XUD1002.national.core.bbc.co.uk>
References: <2F938AAE444ADB47AFEE6307955A30CE35DAE63B@BGB01XUD1002.national.core.bbc.co.uk>
Message-ID: <622D40C0-177F-4DD5-AB59-32C9ECAD7892@jerris.com>

please open a jira on this, we should fix that meta package... that being said, strongly recommend you use 64 bit unless there is a very compelling reason not to.

> On Jul 24, 2015, at 11:55 AM, Peter Stevens <Peter.Stevens at bbc.co.uk> wrote:
> 
> Hi,
>  
> I have a clean install of Debian 8 on a 32 bit machine.
>  
> I'm following information on FS 1.6 video wiki (https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video) to <https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video)%20to> build this version, but the installation of the dependencies part doesn't install all of the required dependencies.
>  
> After running:
> DEBIAN_FRONTEND=none APT_LISTCHANGES_FRONTEND=none apt-get install -y --force-yes freeswitch-video-deps-most
>  
> I had to then:
> apt-get install libavcodec-extra
> apt-get install portaudio19-dev
> Which were both ok, but running:
> apt-get install libsngtc-dev
>  
> I received following:
> Package libsngtc-dev is not available, but is referred to by another package.
> This may mean that the package is missing, has been obsoleted, or
> is only available from another source
> E: Package 'libsngtc-dev' has no installation candidate
>  
> Re-ran:
> DEBIAN_FRONTEND=none APT_LISTCHANGES_FRONTEND=none apt-get install -y --force-yes freeswitch-video-deps-most
> Reading package lists... Done
> Building dependency tree 
> Reading state information... Done
> Some packages could not be installed. This may mean that you have
> requested an impossible situation or if you are using the unstable
> distribution that some required packages have not yet been created
> or been moved out of Incoming.
> The following information may help to resolve the situation:
> The following packages have unmet dependencies:
> freeswitch-video-deps-most : Depends: libsngtc-dev but it is not installable
> E: Unable to correct problems, you have held broken packages.
>  
> I've had a look around amongst the http://files.freeswitch.org/repo/deb/debian/dists/jessie/main <http://files.freeswitch.org/repo/deb/debian/dists/jessie/main> directory structure and have discovered that the libsngtc-dev is listed in both the binary-amd64/packages.gz and the binary-amd64/packages.gz files, but is only listed in the Contents-amd64.gz file and not in the Contents-i386.gz file.
> Don't know whether this is significant or not, but how/where can I obtain the libsngtc-dev file to complete the dependency installation forfreeswitch-video-deps-most?
>  
> Any help appreciated.
>  
> Peter
>  
>  
> ----------------------------
> 
> http://www.bbc.co.uk <http://www.bbc.co.uk/>
> This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated.
> If you have received it in error, please delete it from your system.
> Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately.
> Please note that the BBC monitors e-mails sent or received.
> Further communication will signify your consent to this.
> ---------------------
> _________________________________________________________________________
> 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>
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From jaflong at yandex.com  Fri Jul 24 20:17:31 2015
From: jaflong at yandex.com (jaflong jaflong)
Date: Fri, 24 Jul 2015 19:17:31 +0300
Subject: [Freeswitch-users] beep to valer_parked call on bridge
Message-ID: <233191437754651@web3h.yandex.ru>



On a call that has been valet_parked how can I play a beep/audio to it when a incoming call is brdige to it


From italorossib at gmail.com  Fri Jul 24 21:55:52 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Fri, 24 Jul 2015 14:55:52 -0300
Subject: [Freeswitch-users] beep to valer_parked call on bridge
In-Reply-To: <233191437754651@web3h.yandex.ru>
References: <233191437754651@web3h.yandex.ru>
Message-ID: <CAJR7f2GzQMxcZAv+DScOL+t55De+GWXXWXt_g6bqZPmDW+qa9Q@mail.gmail.com>

Try bridge_pre_execute variables

https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-bridge_pre_execute_aleg_app

On Fri, Jul 24, 2015 at 1:17 PM, jaflong jaflong <jaflong at yandex.com> wrote:

>
>
> On a call that has been valet_parked how can I play a beep/audio to it
> when a incoming call is brdige to it
>
> _________________________________________________________________________
> 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
>



-- 
?talo Rossi
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From vipkilla at gmail.com  Fri Jul 24 22:10:50 2015
From: vipkilla at gmail.com (Vik Killa)
Date: Fri, 24 Jul 2015 14:10:50 -0400
Subject: [Freeswitch-users] beep to valer_parked call on bridge
In-Reply-To: <CAJR7f2GzQMxcZAv+DScOL+t55De+GWXXWXt_g6bqZPmDW+qa9Q@mail.gmail.com>
References: <233191437754651@web3h.yandex.ru>
	<CAJR7f2GzQMxcZAv+DScOL+t55De+GWXXWXt_g6bqZPmDW+qa9Q@mail.gmail.com>
Message-ID: <CAC-LwPPk_yWy3Nu50eO2JOU6i8sb11XvcRN+uPJnioCJWwyavA@mail.gmail.com>

sched_broadcast

On Fri, Jul 24, 2015 at 1:55 PM, ?talo Rossi <italorossib at gmail.com> wrote:

> Try bridge_pre_execute variables
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-bridge_pre_execute_aleg_app
>
> On Fri, Jul 24, 2015 at 1:17 PM, jaflong jaflong <jaflong at yandex.com>
> wrote:
>
>>
>>
>> On a call that has been valet_parked how can I play a beep/audio to it
>> when a incoming call is brdige to it
>>
>> _________________________________________________________________________
>> 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
>>
>
>
>
> --
> ?talo Rossi
>
> _________________________________________________________________________
> 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 william.king at quentustech.com  Fri Jul 24 22:45:15 2015
From: william.king at quentustech.com (William King)
Date: Fri, 24 Jul 2015 11:45:15 -0700
Subject: [Freeswitch-users] mod_xml_radius starts accounting twice for
 incoming calls
In-Reply-To: <CAOLEQ7djZo=Gg=L6tF80BcMvhC2wE-ees4yaRvYXNub9BT60AQ@mail.gmail.com>
References: <CAOLEQ7djZo=Gg=L6tF80BcMvhC2wE-ees4yaRvYXNub9BT60AQ@mail.gmail.com>
Message-ID: <55B287BB.8040801@quentustech.com>

The configurations allow you to exclude channels based on regex for
channel variables. By default xml_radius will log(accounting) each
individual channel/session.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 7/22/15 11:51 PM, Artur Mega wrote:
> When new incoming call comes to tr2.xxxxxxx.ru <http://tr2.xxxxxxx.ru>,
> new session is being created and accounting begins. Further, call
> forwarding to another server, to fs2.xxxxxxx.ru <http://fs2.xxxxxxx.ru>,
> and new session is being created again.
> 
> 2015-07-23 11:22:42.514095 [NOTICE] switch_channel.c:1054 *New Channel*
> sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> <http://73472460000 at tr2.xxxxxxx.ru:5080>
> [26777290306690-93831426613199 at 192.168.217.156
> <mailto:26777290306690-93831426613199 at 192.168.217.156>]
> ...
> 2015-07-23 11:22:42.574084 [INFO] mod_xml_radius.c:1123 mod_xml_radius:
>  *Accounting Start success*
> 2015-07-23 11:22:42.574084 [DEBUG] switch_core_state_machine.c:164
> sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> <http://73472460000 at tr2.xxxxxxx.ru:5080> Standard ROUTING
> 2015-07-23 11:22:42.574084 [INFO] mod_dialplan_xml.c:558 Processing
> 73472460000 <73472460000>->79373057071 in context public
> Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> <http://73472460000 at tr2.xxxxxxx.ru:5080> parsing [public->originate_leg]
> continue=true
> Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> <http://73472460000 at tr2.xxxxxxx.ru:5080> Absolute Condition [originate_leg]
> Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> <http://73472460000 at tr2.xxxxxxx.ru:5080> Action
> export(nolocal:h323-call-origin=originate) 
> Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> <http://73472460000 at tr2.xxxxxxx.ru:5080> parsing [public->public_mult]
> continue=false
> Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> <http://73472460000 at tr2.xxxxxxx.ru:5080> Regex (PASS) [public_mult]
> destination_number(79373057071) =~ /^(79373057071)$/ break=on-false
> Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> <http://73472460000 at tr2.xxxxxxx.ru:5080> Action
> set(hangup_after_bridge=true) 
> Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> <http://73472460000 at tr2.xxxxxxx.ru:5080> Action
> bridge(sofia/internal/79373057071 at fs2.xxxxxxx.ru:5070
> <http://79373057071 at fs2.xxxxxxx.ru:5070>) 
> 2015-07-23 11:22:42.574084 [DEBUG] switch_core_state_machine.c:214
> (sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> <http://73472460000 at tr2.xxxxxxx.ru:5080>) State Change CS_ROUTING ->
> CS_EXECUTE
> ...
> 2015-07-23 11:22:42.574084 [NOTICE] switch_channel.c:1054 *New Channel*
> sofia/internal/79373057071 at fs2.xxxxxxx.ru:5070
> <http://79373057071 at fs2.xxxxxxx.ru:5070>
> [e059ad7f-b2f8-4beb-b1f3-249bc431cfc4]
> ...
> 2015-07-23 11:22:42.634059 [INFO] mod_xml_radius.c:1123 mod_xml_radius:
> * Accounting Start success*
> 
> Thus, one call is being counted twice (for incoming to tr2, and then for
> outgoing from tr2 to fs2). But we need to make mod_xml_radius counts
> this call only once, how can we handle it?
> Thanks
> -- 
> 
> Arthur
> 
> 
> _________________________________________________________________________
> 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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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From mike at jerris.com  Fri Jul 24 23:26:34 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 24 Jul 2015 15:26:34 -0400
Subject: [Freeswitch-users] Firefox 39, webrtc
In-Reply-To: <CAJTkRNuZLupoO7jB5mDAEe_QhL7n=3L4hK9guxSctavhPZieLA@mail.gmail.com>
References: <CAJTkRNsXxSV_WYRk6dY9WN=kPuv_h2Y+By+j7KUmCBUx8k8zyw@mail.gmail.com>
	<CAEJMVkCqc8y-ne0SOAuZNhxYe159HjNGi0acxMnZQm-+FQtEaQ@mail.gmail.com>
	<CAJTkRNugkoY6c_kod0E+bZsKGDN23c9bXhvfh+C8bw7QOoCCow@mail.gmail.com>
	<CA+3gBcnCz3tgcyiDgXdnY7dddbnbY9s+kbG8TrDR9BJHExN-Gw@mail.gmail.com>
	<CAK59s4wmqUwFRn_b1XdH3t6n56jFwqAVnrj_YRPqdAu=bJDo7g@mail.gmail.com>
	<CAJTkRNuZLupoO7jB5mDAEe_QhL7n=3L4hK9guxSctavhPZieLA@mail.gmail.com>
Message-ID: <F80F96B8-1985-4973-8BC8-D64DDF7ACD4D@jerris.com>

It would be very helpful if someone can put together a pull request for this issue against master.  Otherwise it will have to wait for someone to have time to look at it and formulate a patch, which is not likely to happen for a little while.

> On Jul 24, 2015, at 9:53 AM, Igor Olhovskiy <igorolhovskiy at gmail.com> wrote:
> 
> I've post bug in jira already )
> 
> 2015-07-24 16:08 GMT+03:00 Javier Menendez <menendez.garcia at gmail.com <mailto:menendez.garcia at gmail.com>>:
> I agree with Carlos, it's a bug, I have adapted the patch to FS v 1.4.20 and it is working, 
> 
> On Wed, Jul 15, 2015 at 6:01 PM, Carlos Ruiz D?az <carlos.ruizdiaz at gmail.com <mailto:carlos.ruizdiaz at gmail.com>> wrote:
> There's a high change this being a bug in FS, caused by FF removing lots of cipher suites it used to use for the handshake.
> 
> I experienced this myself with rtpengine. 
> 
> This is the commit that fixed the issue on that project, in case it may be useful for the maintainer(s) in charge of webRTC in FS. [1]
> 
> [1] https://github.com/sipwise/rtpengine/commit/21e1fb680762f421e05fab036f8138b2276f5037 <https://github.com/sipwise/rtpengine/commit/21e1fb680762f421e05fab036f8138b2276f5037>
> 
> On Wed, Jul 15, 2015 at 10:47 AM, Igor Olhovskiy <igorolhovskiy at gmail.com <mailto:igorolhovskiy at gmail.com>> wrote:
> I'm trying to figure out if it is a bug of FS :) 
> So asking if someone is having the same )
> 
> 2015-07-15 18:08 GMT+03:00 Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>>:
> I would like to know what we can do to get the point across about filing bugs on JIRA?
> 
> Even if you file something and it ends up not being a bug, we can at the very least close it as 'not a bug'.  What we can't do is mentally track every last issue thats reported on the list.
> 
> https://www.youtube.com/watch?v=lxB57TtX27I <https://www.youtube.com/watch?v=lxB57TtX27I>
> 
> Thanks,
> 
> 
> 
> On Wed, Jul 15, 2015 at 5:55 AM, Igor Olhovskiy <igorolhovskiy at gmail.com <mailto:igorolhovskiy at gmail.com>> wrote:
> Hi!
> 
> I'm using sip.js with FS (tried FS 1.5.15 and 1.7). Upgrading to Firefox 39 is seems to be bad idea, cause call through it fails due to 
> 
> 2015-07-15 10:48:39.168250 [ERR] switch_rtp.c:2907 audio Handshake failure 1
> 2015-07-15 10:48:39.168250 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL
> 
> On FF 38 and Chrome all is ok.
> Am I only one with this issue of it's really something wrong with FF?
> 
> -- 
> Best regards,
> Igor
> 
> _________________________________________________________________________
> 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>
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> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Brian West
> brian at freeswitch.org <mailto:brian at freeswitch.org>
> 
> Twitter: @FreeSWITCH , @briankwest
> http://www.freeswitchbook.com <http://www.freeswitchbook.com/>
> http://www.freeswitchcookbook.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 <tel:%2B19184209001> | F:+19184209002 <tel:%2B19184209002> | M:+1918424WEST (9378)
> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
> 
> 
> _________________________________________________________________________
> 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/>
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> 
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> 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/>
> 
> 
> 
> -- 
> Best regards,
> Igor
> 
> _________________________________________________________________________
> 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/>
> 
> 
> 
> -- 
> Carlos
> http://caruizdiaz.com <http://caruizdiaz.com/>
> _________________________________________________________________________
> 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/>
> 
> 
> 
> -- 
> Best 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 findmeinwland at gmail.com  Fri Jul 24 23:32:30 2015
From: findmeinwland at gmail.com (Artur Mega)
Date: Sat, 25 Jul 2015 00:32:30 +0500
Subject: [Freeswitch-users] mod_xml_radius starts accounting twice for
 incoming calls
In-Reply-To: <55B287BB.8040801@quentustech.com>
References: <CAOLEQ7djZo=Gg=L6tF80BcMvhC2wE-ees4yaRvYXNub9BT60AQ@mail.gmail.com>
	<55B287BB.8040801@quentustech.com>
Message-ID: <CAOLEQ7eh9YS-f8WUEpyDeeVD6h66ZZtDSjSzs4Pjr_BF79-8Gg@mail.gmail.com>

?Thanks, William,
i dont know what to edit in mod_xml_radius. I cannot exclude channel from
dialplan, right? So what i have to change in configs?

2015-07-24 23:45 GMT+05:00 William King <william.king at quentustech.com>:

> The configurations allow you to exclude channels based on regex for
> channel variables. By default xml_radius will log(accounting) each
> individual channel/session.
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
>
> On 7/22/15 11:51 PM, Artur Mega wrote:
> > When new incoming call comes to tr2.xxxxxxx.ru <http://tr2.xxxxxxx.ru>,
> > new session is being created and accounting begins. Further, call
> > forwarding to another server, to fs2.xxxxxxx.ru <http://fs2.xxxxxxx.ru>,
> > and new session is being created again.
> >
> > 2015-07-23 11:22:42.514095 [NOTICE] switch_channel.c:1054 *New Channel*
> > sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> > <http://73472460000 at tr2.xxxxxxx.ru:5080>
> > [26777290306690-93831426613199 at 192.168.217.156
> > <mailto:26777290306690-93831426613199 at 192.168.217.156>]
> > ...
> > 2015-07-23 11:22:42.574084 [INFO] mod_xml_radius.c:1123 mod_xml_radius:
> >  *Accounting Start success*
> > 2015-07-23 11:22:42.574084 [DEBUG] switch_core_state_machine.c:164
> > sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> > <http://73472460000 at tr2.xxxxxxx.ru:5080> Standard ROUTING
> > 2015-07-23 11:22:42.574084 [INFO] mod_dialplan_xml.c:558 Processing
> > 73472460000 <73472460000>->79373057071 in context public
> > Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> > <http://73472460000 at tr2.xxxxxxx.ru:5080> parsing [public->originate_leg]
> > continue=true
> > Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> > <http://73472460000 at tr2.xxxxxxx.ru:5080> Absolute Condition
> [originate_leg]
> > Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> > <http://73472460000 at tr2.xxxxxxx.ru:5080> Action
> > export(nolocal:h323-call-origin=originate)
> > Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> > <http://73472460000 at tr2.xxxxxxx.ru:5080> parsing [public->public_mult]
> > continue=false
> > Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> > <http://73472460000 at tr2.xxxxxxx.ru:5080> Regex (PASS) [public_mult]
> > destination_number(79373057071) =~ /^(79373057071)$/ break=on-false
> > Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> > <http://73472460000 at tr2.xxxxxxx.ru:5080> Action
> > set(hangup_after_bridge=true)
> > Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> > <http://73472460000 at tr2.xxxxxxx.ru:5080> Action
> > bridge(sofia/internal/79373057071 at fs2.xxxxxxx.ru:5070
> > <http://79373057071 at fs2.xxxxxxx.ru:5070>)
> > 2015-07-23 11:22:42.574084 [DEBUG] switch_core_state_machine.c:214
> > (sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
> > <http://73472460000 at tr2.xxxxxxx.ru:5080>) State Change CS_ROUTING ->
> > CS_EXECUTE
> > ...
> > 2015-07-23 11:22:42.574084 [NOTICE] switch_channel.c:1054 *New Channel*
> > sofia/internal/79373057071 at fs2.xxxxxxx.ru:5070
> > <http://79373057071 at fs2.xxxxxxx.ru:5070>
> > [e059ad7f-b2f8-4beb-b1f3-249bc431cfc4]
> > ...
> > 2015-07-23 11:22:42.634059 [INFO] mod_xml_radius.c:1123 mod_xml_radius:
> > * Accounting Start success*
> >
> > Thus, one call is being counted twice (for incoming to tr2, and then for
> > outgoing from tr2 to fs2). But we need to make mod_xml_radius counts
> > this call only once, how can we handle it?
> > Thanks
> > --
> >
> > Arthur
> >
> >
> > _________________________________________________________________________
> > 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
>



-- 

Arthur
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From william.king at quentustech.com  Fri Jul 24 23:56:52 2015
From: william.king at quentustech.com (William King)
Date: Fri, 24 Jul 2015 12:56:52 -0700
Subject: [Freeswitch-users] mod_xml_radius starts accounting twice for
 incoming calls
In-Reply-To: <CAOLEQ7eh9YS-f8WUEpyDeeVD6h66ZZtDSjSzs4Pjr_BF79-8Gg@mail.gmail.com>
References: <CAOLEQ7djZo=Gg=L6tF80BcMvhC2wE-ees4yaRvYXNub9BT60AQ@mail.gmail.com>
	<55B287BB.8040801@quentustech.com>
	<CAOLEQ7eh9YS-f8WUEpyDeeVD6h66ZZtDSjSzs4Pjr_BF79-8Gg@mail.gmail.com>
Message-ID: <55B29884.6070000@quentustech.com>

Please read the module config in autoload_configs/xml_radius.conf.xml
for examples.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 7/24/15 12:32 PM, Artur Mega wrote:
> ?Thanks, William,
> i dont know what to edit in mod_xml_radius. I cannot exclude channel
> from dialplan, right? So what i have to change in configs?
> 
> 2015-07-24 23:45 GMT+05:00 William King <william.king at quentustech.com
> <mailto:william.king at quentustech.com>>:
> 
>     The configurations allow you to exclude channels based on regex for
>     channel variables. By default xml_radius will log(accounting) each
>     individual channel/session.
> 
>     William King
>     Senior Engineer
>     Quentus Technologies, INC
>     1037 NE 65th St Suite 273
>     Seattle, WA 98115
>     Main:   (877) 211-9337
>     Office: (206) 388-4772
>     Cell:   (253) 686-5518
>     william.king at quentustech.com <mailto:william.king at quentustech.com>
> 
>     On 7/22/15 11:51 PM, Artur Mega wrote:
>     > When new incoming call comes to tr2.xxxxxxx.ru
>     <http://tr2.xxxxxxx.ru> <http://tr2.xxxxxxx.ru>,
>     > new session is being created and accounting begins. Further, call
>     > forwarding to another server, to fs2.xxxxxxx.ru
>     <http://fs2.xxxxxxx.ru> <http://fs2.xxxxxxx.ru>,
>     > and new session is being created again.
>     >
>     > 2015-07-23 11:22:42.514095 [NOTICE] switch_channel.c:1054 *New
>     Channel*
>     > sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
>     <http://73472460000 at tr2.xxxxxxx.ru:5080>
>     > <http://73472460000 at tr2.xxxxxxx.ru:5080>
>     > [26777290306690-93831426613199 at 192.168.217.156
>     <mailto:26777290306690-93831426613199 at 192.168.217.156>
>     > <mailto:26777290306690-93831426613199 at 192.168.217.156
>     <mailto:26777290306690-93831426613199 at 192.168.217.156>>]
>     > ...
>     > 2015-07-23 11:22:42.574084 [INFO] mod_xml_radius.c:1123 mod_xml_radius:
>     >  *Accounting Start success*
>     > 2015-07-23 11:22:42.574084 [DEBUG] switch_core_state_machine.c:164
>     > sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
>     <http://73472460000 at tr2.xxxxxxx.ru:5080>
>     > <http://73472460000 at tr2.xxxxxxx.ru:5080> Standard ROUTING
>     > 2015-07-23 11:22:42.574084 [INFO] mod_dialplan_xml.c:558 Processing
>     > 73472460000 <73472460000>->79373057071 in context public
>     > Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
>     <http://73472460000 at tr2.xxxxxxx.ru:5080>
>     > <http://73472460000 at tr2.xxxxxxx.ru:5080> parsing
>     [public->originate_leg]
>     > continue=true
>     > Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
>     <http://73472460000 at tr2.xxxxxxx.ru:5080>
>     > <http://73472460000 at tr2.xxxxxxx.ru:5080> Absolute Condition
>     [originate_leg]
>     > Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
>     <http://73472460000 at tr2.xxxxxxx.ru:5080>
>     > <http://73472460000 at tr2.xxxxxxx.ru:5080> Action
>     > export(nolocal:h323-call-origin=originate)
>     > Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
>     <http://73472460000 at tr2.xxxxxxx.ru:5080>
>     > <http://73472460000 at tr2.xxxxxxx.ru:5080> parsing [public->public_mult]
>     > continue=false
>     > Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
>     <http://73472460000 at tr2.xxxxxxx.ru:5080>
>     > <http://73472460000 at tr2.xxxxxxx.ru:5080> Regex (PASS) [public_mult]
>     > destination_number(79373057071) =~ /^(79373057071)$/ break=on-false
>     > Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
>     <http://73472460000 at tr2.xxxxxxx.ru:5080>
>     > <http://73472460000 at tr2.xxxxxxx.ru:5080> Action
>     > set(hangup_after_bridge=true)
>     > Dialplan: sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
>     <http://73472460000 at tr2.xxxxxxx.ru:5080>
>     > <http://73472460000 at tr2.xxxxxxx.ru:5080> Action
>     > bridge(sofia/internal/79373057071 at fs2.xxxxxxx.ru:5070
>     <http://79373057071 at fs2.xxxxxxx.ru:5070>
>     > <http://79373057071 at fs2.xxxxxxx.ru:5070>)
>     > 2015-07-23 11:22:42.574084 [DEBUG] switch_core_state_machine.c:214
>     > (sofia/external/73472460000 at tr2.xxxxxxx.ru:5080
>     <http://73472460000 at tr2.xxxxxxx.ru:5080>
>     > <http://73472460000 at tr2.xxxxxxx.ru:5080>) State Change CS_ROUTING ->
>     > CS_EXECUTE
>     > ...
>     > 2015-07-23 11:22:42.574084 [NOTICE] switch_channel.c:1054 *New
>     Channel*
>     > sofia/internal/79373057071 at fs2.xxxxxxx.ru:5070
>     <http://79373057071 at fs2.xxxxxxx.ru:5070>
>     > <http://79373057071 at fs2.xxxxxxx.ru:5070>
>     > [e059ad7f-b2f8-4beb-b1f3-249bc431cfc4]
>     > ...
>     > 2015-07-23 11:22:42.634059 [INFO] mod_xml_radius.c:1123 mod_xml_radius:
>     > * Accounting Start success*
>     >
>     > Thus, one call is being counted twice (for incoming to tr2, and then for
>     > outgoing from tr2 to fs2). But we need to make mod_xml_radius counts
>     > this call only once, how can we handle it?
>     > Thanks
>     > --
>     >
>     > Arthur
>     >
>     >
>     >
>     _________________________________________________________________________
>     > 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
> 
> 
> 
> 
> -- 
> 
> Arthur
> 
> 
> _________________________________________________________________________
> 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 tru083 at yahoo.com  Sat Jul 25 02:55:10 2015
From: tru083 at yahoo.com (D D)
Date: Fri, 24 Jul 2015 22:55:10 +0000 (UTC)
Subject: [Freeswitch-users] Can I have a dialplan action that runs when a
	call hangs up?
Message-ID: <611377972.2372793.1437778510710.JavaMail.yahoo@mail.yahoo.com>

Hi,
I need to have an action take place when a call hangs up. ?
The call will have been in a conference, and I need to take extra actions after it hangs up.How can I do this using the dialplan?
Thanks!David
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From yadenis at seznam.cz  Sat Jul 25 15:05:04 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Sat, 25 Jul 2015 13:05:04 +0200
Subject: [Freeswitch-users] The problem with the TURN server and
	connection.
In-Reply-To: <CAAyX+KGQtgZ7yPNLh7S8ADjTMYCV3rQ1-dy21Jn+3xAsWGR6LA@mail.gmail.com>
References: <loom.20150722T161450-208@post.gmane.org>
	<778052174.20150723102530@seznam.cz>
	<CAAyX+KFmuLQVRkP-Au5cN_Tu5aLD0377r8fwrxx3+2Q3E+hiCw@mail.gmail.com>
	<762006319.20150723152412@seznam.cz>
	<CALXCt0qrhSJ=gHmnwu2pBsNWwnKD_447S1_JL0UBH2fK2KzWgQ@mail.gmail.com>
	<823909165.20150724160348@seznam.cz>
	<CAAyX+KGQtgZ7yPNLh7S8ADjTMYCV3rQ1-dy21Jn+3xAsWGR6LA@mail.gmail.com>
Message-ID: <490DCFF4-B0E7-419B-9C75-749B2DFF15D9@seznam.cz>

Dobr? den.

Yes of course. The problem was authentication setting on my Turn server. I have not entered the correct password generated by the server.
Plus it was not correctly specified port. It all became clear after switching on logs


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

> On 24. 7. 2015, at 16:30, Ben Langfeld <ben at langfeld.co.uk> wrote:
> 
> Could you explain, for the benefit of those finding this thread in the future, what you did to fix it?
> 
> On 24 July 2015 at 11:03, Denis Jakovlev <yadenis at seznam.cz <mailto:yadenis at seznam.cz>> wrote:
> Dobr? den,
> 
> many thanks. It works as it should. I have been a mistake in configuring the turn server.
> 
> -- 
> S pozdravem,
> Ing.Denis Jakovlev                           
> mob.tel <http://mob.tel/>. 775-415-382
> 
> ?tvrtek 23. ?ervence 2015, 15:28:45, napsal jste:
> 
> 
> pay attention, you MUST use the "transport" argument for tcp to work, eg:
> 
> turnServers: {
>  urls:"turn:exam.org:443 <http://exam.org:443/>?transport=tcp",
>  username:"alice",
>  password:"racecar"
> }
> 
> On Thu, Jul 23, 2015 at 3:24 PM, Denis Jakovlev <yadenis at seznam.cz <mailto:yadenis at seznam.cz>> wrote:
> Dobr? den,
> 
> That's it I use
> 
> -- 
> S pozdravem,
> Ing.Denis Jakovlev                           
> mob.tel <http://mob.tel/>. 775-415-382
> 
> ?tvrtek 23. ?ervence 2015, 13:56:43, napsal jste:
> 
> 
> Take a look at coturn, a fork of rfc5766-turn-server, which supports RFC6062, the TCP transport for TURN.
> 
> On 23 July 2015 at 05:25, Denis Jakovlev <yadenis at seznam.cz <mailto:yadenis at seznam.cz>> wrote:
> Hi All,
> 
> I have one problem. 
> One of my clients is behind the serious firewall and they are banned UDP connection at all. His clients are connected to it via the Internet. 
> I put my own Turn/Stun server (rfc5766-turn-server from Google). It works the same server as the freeswitch. It works fine.
> The problem follows. When all banned UDP connection - you can not connect (INCOMPATIBLE_DESTINATION). If you allow outgoing UDP At least, that is connected with no problems.
> Is it even possible connection if one of the parties completely banned UDP connection? 
> 
> I thought that the problem is just the decides the turn server. But I is not working for me.
> 
> -- 
> S pozdravem,
> Ing.Denis Jakovlev                           
> mob.tel <http://mob.tel/>. 775-415-382
> 
> _________________________________________________________________________
> 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/>
> 
> 
> 
> -- 
> Sincerely,
> 
> Giovanni Maruzzelli
> Cell : +39-347-2665618 <tel:%2B39-347-2665618>
> _________________________________________________________________________
> 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
> 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 Peter.Stevens at bbc.co.uk  Mon Jul 27 14:55:09 2015
From: Peter.Stevens at bbc.co.uk (Peter Stevens)
Date: Mon, 27 Jul 2015 10:55:09 +0000
Subject: [Freeswitch-users] FS 1.6 video dependency build issue
In-Reply-To: <37E12F60-F093-4717-B2CE-55244C5FFCB4@freeswitch.org>
References: <37E12F60-F093-4717-B2CE-55244C5FFCB4@freeswitch.org>
Message-ID: <2F938AAE444ADB47AFEE6307955A30CE35DB21CE@BGB01XUD1002.national.core.bbc.co.uk>

Hi Ken,



Ok, didn't know/investigate what that library was used for.



32bit - that's all I have at the moment (but will change to 64 bit later) and is just for a small test.



________________________________
From: Ken Rice [krice at freeswitch.org]
Sent: 24 July 2015 17:05
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FS 1.6 video dependency build issue

Unless you are using sangoma transcoding hardware you don?t need libsngtc

And why 32bit?

From: <freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> on behalf of Peter Stevens
Reply-To: FreeSWITCH Users Help
Date: Friday, July 24, 2015 at 10:55 AM
To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>"
Subject: [Freeswitch-users] FS 1.6 video dependency build issue


Hi,



I have a clean install of Debian 8 on a 32 bit machine.



I'm following information on FS 1.6 video wiki (https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video) to<https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video)%20to> build this version, but the installation of the dependencies part doesn't install all of the required dependencies.



After running:

DEBIAN_FRONTEND=none APT_LISTCHANGES_FRONTEND=none apt-get install -y --force-yesfreeswitch-video-deps-most

I had to then:

apt-get install libavcodec-extra

apt-get install portaudio19-dev

Which were both ok, but running:

apt-get install libsngtc-dev



I received following:

Package libsngtc-dev is not available, but is referred to by another package.

This may mean that the package is missing, has been obsoleted, or

is only available from another source

E: Package 'libsngtc-dev' has no installation candidate



Re-ran:

DEBIAN_FRONTEND=none APT_LISTCHANGES_FRONTEND=none apt-get install -y --force-yes freeswitch-video-deps-most

Reading package lists... Done

Building dependency tree

Reading state information... Done

Some packages could not be installed. This may mean that you have

requested an impossible situation or if you are using the unstable

distribution that some required packages have not yet been created

or been moved out of Incoming.

The following information may help to resolve the situation:

The following packages have unmet dependencies:

freeswitch-video-deps-most : Depends: libsngtc-dev but it is not installable

E: Unable to correct problems, you have held broken packages.



I've had a look around amongst the http://files.freeswitch.org/repo/deb/debian/dists/jessie/main directory structure and have discovered that the libsngtc-dev is listed in both the binary-amd64/packages.gz and the binary-amd64/packages.gz files, but is only listed in the Contents-amd64.gz file and not in the Contents-i386.gz file.

Don't know whether this is significant or not, but how/where can I obtain the libsngtc-dev file to complete the dependency installation for freeswitch-video-deps-most?



Any help appreciated.



Peter





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_________________________________________________________________________ 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 alxpol at gmail.com  Mon Jul 27 10:33:07 2015
From: alxpol at gmail.com (Alex Polischuk)
Date: Mon, 27 Jul 2015 09:33:07 +0300
Subject: [Freeswitch-users]  audio Handshake failure 1
Message-ID: <CAGf2z7NE0a7mPsPXjZK9w06jnTJTh--R01O2dvBRyhQPxRmxmw@mail.gmail.com>

Hi All,

I use FreeSwitch for transcoding SRTP from WebRTC to SIP with RTP and vice
versa.
When Firefox version 38 add requires for Perfect Forward Secrecy (PFS) I
have problem with Handshake in FS when Firefox Browser terminate a call. I
found that that problem was fixed in the version 1.4.19
https://freeswitch.org/jira/browse/FS-7425

I have the same problem in the version 1.4.20

eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
switch_rtp.c:2924 Changing audio DTLS state from HANDSHAKE to SETUP
eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
switch_rtp.c:2832 audio Fingerprint Verified.
eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
switch_rtp.c:3384 Activating Audio Secure RTP SEND
eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
switch_rtp.c:3362 Activating Audio Secure RTP RECV
eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
switch_rtp.c:2872 Changing audio DTLS state from SETUP to READY
eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.604193 [DEBUG]
switch_rtp.c:1937 rtcp_stats_init: ssrc[-1374535478] base_seq[6179]
f01c46d8-33a0-11e5-ac83-891cb7dea826 2015-07-26 14:16:46.644210 [DEBUG]
switch_rtp.c:5884 Correct ip/port confirmed.
f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [ERR]
switch_rtp.c:2917 audio Handshake failure 1
f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [INFO]
switch_rtp.c:2918 Changing audio DTLS state from HANDSHAKE to FAIL
f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [NOTICE]
switch_rtp.c:2899 Hangup sofia/external/992555500004 at voip.webrtc.jajah.com
[CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER]


what additional definition I should make to resolve this problem?

Thanks,
Alex Polischuk
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From david.fu at oocl.com  Mon Jul 27 11:25:50 2015
From: david.fu at oocl.com (david.fu at oocl.com)
Date: Mon, 27 Jul 2015 15:25:50 +0800
Subject: [Freeswitch-users] Implementing <Form> Tag of VXML in Freeswtich
	using Javascript
Message-ID: <61E1B4F8DE845A4F986F448F6D50A38504BE869D67@E2K7CCR03.corp.oocl.com>

Dear Freeswitch experts,

I would like to implement something like <Form> and <submit> tag of VXML in Freeswtich using Javascript. The purpose is to sending request back to the application server from Freeswtich.  However,  I couldn't find the related API(s) in the Freeswitch official web site. Would you please help advise ? Thank you so much.

Yours faithfully,
David




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From sendmeallyouroffers at googlemail.com  Mon Jul 27 12:52:11 2015
From: sendmeallyouroffers at googlemail.com (Berthold Karl)
Date: Mon, 27 Jul 2015 10:52:11 +0200
Subject: [Freeswitch-users] freeswitch never seems to call sendmail when
 trying to email voicemails
In-Reply-To: <CAEJMVkBY-3nOfXijxhED_byC_K5cNnS8LVW3iRjGmOFserDmjw@mail.gmail.com>
References: <54E6B9A1.20708@dickson.st>
	<CAHtxdDd8CRENTyNoi7FOn3yuy+fyCshEiknfhvhZcg36q=JFuA@mail.gmail.com>
	<54EA5B78.3010504@dickson.st>
	<CAHtxdDcD0fqTWR9mA8=-eT9rm2a1M1p3Qab7OVXCfH=wYnHvvw@mail.gmail.com>
	<54EBE76A.2040709@dickson.st>
	<CAHtxdDfsm2pd8HDM-WwU8HKJT5AQj3isqjWF4aWVbXkRrtB0zg@mail.gmail.com>
	<CAEJMVkBY-3nOfXijxhED_byC_K5cNnS8LVW3iRjGmOFserDmjw@mail.gmail.com>
Message-ID: <CAN6oSBuxXbf=uecMk1UO=ZP9xxpvoXA65Ox4pyZVmPxA-BNwnQ@mail.gmail.com>

Hi,

my FS is calling the sendmail-command, but sendmail ends with an segfault.
It seems like there is no email under /tmp/. Did your sendmail also ends in
a segfault?

2015-02-24 16:51 GMT+01:00 Brian West <brian at freeswitch.org>:

> When you file the JIRA please attach your voicemail.conf.xml.
>
> On Tue, Feb 24, 2015 at 8:21 AM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> If you want to FS correctly processed parameters vm-mailfrom
>> and email-from, write a request to https://freeswitch.org/jira/
>>
>> On Tue, Feb 24, 2015 at 5:52 AM, Jason Lewis <jason at dickson.st> wrote:
>>
>>>  Thanks Sergey,
>>>
>>> I had configured the domain variable to the fqdn of the machine. I
>>> eventually got it working though, I was missing two key lines from my user
>>> config:
>>>       <param name="vm-email-all-messages" value="true"/>
>>>       <param name="vm-attach-file" value="true" />
>>>
>>> I'm not sure how I managed to miss those but anyway, that seems to have
>>> resolved things.
>>>
>>> It seems as though vm-mailfrom is still being ignored though. Currently
>>> I have it set to:
>>>
>>>        <param name="vm-mailfrom" value="vm at freeswitch.xyz.com.au"
>>> <vm at freeswitch.xyz.com.au>/>
>>>
>>> but voicemails get delivered from:
>>>
>>> 1001 at freeswitch.xyz.com.au
>>>
>>> Is this worth investigating further?
>>>
>>> Jason
>>>
>>> Sergey Safarov wrote on 23/02/2015 5:05 PM:
>>>
>>> Try configure "domain" variable in vars.xml
>>>
>>>  <X-PRE-PROCESS cmd="set" data="domain=you_domain_name"/>
>>>
>>>  After it verify that user registered with domain name
>>>
>>>  freeswitch at internal> sofia status profile internal reg
>>>
>>>  Registrations:
>>>
>>> =================================================================================================
>>> Call-ID:     1B26-2327-466848134BEBC9719CDE-002 at SipHost
>>> User:       1201 at you_domain_name
>>> Contact:     "1201"
>>> <sip:1201 at 10.21.18.22:5060;fs_nat=yes;fs_path=sip%3A1201%4010.21.18.22%3A5060>
>>> Agent:       204 12-3868-2416-0.10.56.1-DS
>>> Status:     Registered(UDP-NAT)(unknown) EXP(2015-02-23 06:05:22)
>>> EXPSECS(139)
>>> Ping-Status: Reachable
>>> Host:       fs1.you_domain_name
>>> IP:         10.21.18.22
>>> Port:       5060
>>> Auth-User:   1201
>>> Auth-Realm: you_domain_name
>>> MWI-Account: 1201 at you_domain_name
>>>
>>>  Sergey
>>>
>>>
>>> On Mon, Feb 23, 2015 at 1:43 AM, Jason Lewis <jason at dickson.st> wrote:
>>>
>>>>  So I've managed to see some output from the sendmail program in the
>>>> FS logs. It appears that my fs instance isn't correctly setting its domain?
>>>>
>>>> the FS box has a fqdn, and I also set the domain parameter in the
>>>> vars.xml file, but still the voicemail is sent with a from address of an IP
>>>> address.
>>>>
>>>> Any ideas?
>>>>
>>>> Net::SMTP>>> Net::SMTP(2.33)
>>>> Net::SMTP>>>   Net::Cmd(2.30)
>>>> Net::SMTP>>>     Exporter(5.71)
>>>> Net::SMTP>>>   IO::Socket::INET(1.35)
>>>> Net::SMTP>>>     IO::Socket(1.37)
>>>> Net::SMTP>>>       IO::Handle(1.35)
>>>> Net::SMTP=GLOB(0x23f7748)<<< 220 mb.xyz.com.au ESMTP Postfix
>>>> (Debian/GNU)
>>>> Net::SMTP=GLOB(0x23f7748)>>> EHLO localhost.localdomain
>>>> Net::SMTP=GLOB(0x23f7748)<<< 250-mb.bongalong.st
>>>> Net::SMTP=GLOB(0x23f7748)<<< 250-PIPELINING
>>>> Net::SMTP=GLOB(0x23f7748)<<< 250-SIZE 10240000
>>>> Net::SMTP=GLOB(0x23f7748)<<< 250-VRFY
>>>> Net::SMTP=GLOB(0x23f7748)<<< 250-ETRN
>>>> Net::SMTP=GLOB(0x23f7748)<<< 250-STARTTLS
>>>> Net::SMTP=GLOB(0x23f7748)<<< 250-ENHANCEDSTATUSCODES
>>>> Net::SMTP=GLOB(0x23f7748)<<< 250-8BITMIME
>>>> Net::SMTP=GLOB(0x23f7748)<<< 250 DSN
>>>> Net::SMTP=GLOB(0x23f7748)>>> MAIL FROM:<1002 at 192.168.1.3>
>>>> <1002 at 192.168.1.3>
>>>> Net::SMTP=GLOB(0x23f7748)<<< 501 5.1.7 Bad sender address syntax
>>>>  Net::SMTP=GLOB(0x23f7748)>>> RCPT TO:<jason at xyz.com.au>
>>>> <jason at xyz.com.au>
>>>> Net::SMTP=GLOB(0x23f7748)<<< 503 5.5.1 Error: need MAIL command
>>>> Net::SMTP=GLOB(0x23f7748)>>> DATA
>>>> Net::SMTP=GLOB(0x23f7748)<<< 503 5.5.1 Error: need RCPT command
>>>> Net::SMTP=GLOB(0x23f7748)>>> QUIT
>>>> Net::SMTP=GLOB(0x23f7748)<<< 221 2.0.0 Bye
>>>> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1633 Session
>>>> 3 (loopback/voicemail-a) Ended
>>>> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1637 Close
>>>> Channel loopback/voicemail-a [CS_DESTROY]
>>>> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1633 Session
>>>> 4 (loopback/voicemail-b) Ended
>>>> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1637 Close
>>>> Channel loopback/voicemail-b [CS_DESTROY]
>>>>
>>>>
>>>> On 20/02/2015 7:10 pm, Sergey Safarov wrote:
>>>>
>>>>  You mailer is not understand "mailer-app-args" has been configured in
>>>> "switch.conf.xml"
>>>>
>>>> Remove extra arg or add required
>>>>
>>>> ??, 20 ????. 2015, 7:49, Jason Lewis <jason at dickson.st>:
>>>>
>>>>> Hi,
>>>>>
>>>>> I've been trying to make freeswitch email voicemails but as far as I
>>>>> can
>>>>> tell, it never even calls sendmail.
>>>>>
>>>>> I have setting mailer-app to "sendmail" and "/usr/sbin/sendmail" to no
>>>>> avail. I can successfully send an email from the commandline using
>>>>> sendmail. (sendmail in this case is provided by postfix)
>>>>>
>>>>> I see no emails in the postfix mail logs when I leave a voicemail
>>>>> message.
>>>>>
>>>>> I also tried creating a shell just to see if it even gets called from
>>>>> fs, but it does not get called when a voicemail is deposited:
>>>>> #!/bin/bash
>>>>> echo $(date --rfc-3339=ns): $* >> /tmp/freeswitchsendmail.log
>>>>>
>>>>> After every change, I have run reloadxml and reload mod_voicemail. I
>>>>> have also tried restarting freeswitch.
>>>>>
>>>>> I am running the debian packages of FreeSWITCH Version 1.4.15-1~64bit
>>>>> (-1 64bit)
>>>>>
>>>>> my configuration is based on the vanilla configuration with only very
>>>>> minor changes.
>>>>>
>>>>> I'm at a loss as to how to debug further, but I'm pretty sure the
>>>>> mailer-app is never called. Is there some setting I'm missing or
>>>>> something obvious I'm not doing?
>>>>>
>>>>>
>>>>> My config:
>>>>> 1001.xml:
>>>>> <include>
>>>>>   <user id="1001">
>>>>>     <params>
>>>>>       <param name="password" value="$${default_password}"/>
>>>>>       <param name="vm-password" value="1001"/>
>>>>>       <param name="vm-mailto" value="jason at xyz.com.au"/>
>>>>>       <param name="vm-mailfrom" value="vm at freeswitch.xyz.com.au"/>
>>>>>       <!-- <param name="vm-message-ext" value="wav"/> -->
>>>>>     </params>
>>>>>     <variables>
>>>>>       <variable name="toll_allow"
>>>>> value="domestic,international,local"/>
>>>>>       <variable name="accountcode" value="1001"/>
>>>>>       <variable name="user_context" value="default"/>
>>>>>       <variable name="effective_caller_id_name" value="Extension
>>>>> 1001"/>
>>>>>       <variable name="effective_caller_id_number" value="1001"/>
>>>>>       <variable name="outbound_caller_id_name"
>>>>> value="$${outbound_caller_name}"/>
>>>>>       <variable name="outbound_caller_id_number"
>>>>> value="$${outbound_caller_id}"/>
>>>>>       <variable name="callgroup" value="techsupport"/>
>>>>>     </variables>
>>>>>   </user>
>>>>> </include>
>>>>>
>>>>> and in switch.conf.xml I have the following set:
>>>>>
>>>>>     <param name="mailer-app" value="/usr/sbin/sendmail"/>
>>>>>     <param name="mailer-app-args" value="-t"/>
>>>>>
>>>>>
>>>>> I made a log at level 7 and put it on the pastebin:
>>>>>
>>>>> https://pastebin.freeswitch.org/23921
>>>>>
>>>>>
>>>>> Jason Lewis
>>>>> http://emacstragic.net
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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.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
>>>>
>>>>
>>>>
>>>>   --
>>>> Jason Lewishttp://emacstragic.net
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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.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
>>>
>>>
>>> --
>>> Jason Lewishttp://emacstragic.net
>>>
>>>
>>> _________________________________________________________________________
>>> 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*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> 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 568691 at gmail.com  Mon Jul 27 13:42:39 2015
From: 568691 at gmail.com (Alexandru Covalschi)
Date: Mon, 27 Jul 2015 12:42:39 +0300
Subject: [Freeswitch-users] Video conferencing
Message-ID: <CAD_iGr9nKLZ9_M2Sc-nNC58xS579aYbX62EDUcNj2ynftjsUNA@mail.gmail.com>

Hello! Started to figure out the theme in topic. Didn't find any
descriptions of what is "video-floor" and etc. Can you please provide a
link to the theory I need to read to understand all definitions?

-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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From findmeinwland at gmail.com  Mon Jul 27 17:32:45 2015
From: findmeinwland at gmail.com (Artur Mega)
Date: Mon, 27 Jul 2015 18:32:45 +0500
Subject: [Freeswitch-users] Implementing <Form> Tag of VXML in
 Freeswtich using Javascript
In-Reply-To: <61E1B4F8DE845A4F986F448F6D50A38504BE869D67@E2K7CCR03.corp.oocl.com>
References: <61E1B4F8DE845A4F986F448F6D50A38504BE869D67@E2K7CCR03.corp.oocl.com>
Message-ID: <CAOLEQ7eYA8Hawm_Ud_1ZJ5KLVh1aweycBvX8VrjQ9QVQSEoDyQ@mail.gmail.com>

i cant understand what you want...

2015-07-27 12:25 GMT+05:00 <david.fu at oocl.com>:

>  Dear Freeswitch experts,
>
> I would like to implement something like <Form> and <submit> tag of VXML
> in Freeswtich using Javascript. The purpose is to sending request back to
> the application server from Freeswtich.  However,  I couldn?t find the
> related API(s) in the Freeswitch official web site. Would you please help
> advise ? Thank you so much.
>
> Yours faithfully,
> David
>
>
>
>
> IMPORTANT NOTICE
> Email from OOCL is confidential and may be legally privileged.  If it is
> not
> intended for you, please delete it immediately unread.  The internet
> cannot guarantee that this communication is free of viruses, interception
> or interference and anyone who communicates with us by email is taken
> to accept the risks in doing so.  Without limitation, OOCL and its
> affiliates
> accept no liability whatsoever and howsoever arising in connection with
> the use of this email.  Under no circumstances shall this email constitute
> a binding agreement to carry or for provision of carriage services by OOCL,
> which is subject to the availability of carrier's equipment and vessels and
> the terms and conditions of OOCL's standard bill of lading which is also
> available at http://www.oocl.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
>



-- 

Arthur
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From brian at freeswitch.org  Mon Jul 27 17:53:03 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 27 Jul 2015 08:53:03 -0500
Subject: [Freeswitch-users] Implementing <Form> Tag of VXML in
 Freeswtich using Javascript
In-Reply-To: <61E1B4F8DE845A4F986F448F6D50A38504BE869D67@E2K7CCR03.corp.oocl.com>
References: <61E1B4F8DE845A4F986F448F6D50A38504BE869D67@E2K7CCR03.corp.oocl.com>
Message-ID: <CAEJMVkAy=jUh2JHM9enqH4wkiUO4tyxG6Fy76zJ7VdiqW=1BCA@mail.gmail.com>

There is no support for VXML in FreeSWITCH, We have all the parts and I'm
sure it could probably be implemented, but its never been a high priority
item for us as nobody asks for it.

On Mon, Jul 27, 2015 at 2:25 AM, <david.fu at oocl.com> wrote:

>  Dear Freeswitch experts,
>
> I would like to implement something like <Form> and <submit> tag of VXML
> in Freeswtich using Javascript. The purpose is to sending request back to
> the application server from Freeswtich.  However,  I couldn?t find the
> related API(s) in the Freeswitch official web site. Would you please help
> advise ? Thank you so much.
>
> Yours faithfully,
> David
>
>
>
>
> IMPORTANT NOTICE
> Email from OOCL is confidential and may be legally privileged.  If it is
> not
> intended for you, please delete it immediately unread.  The internet
> cannot guarantee that this communication is free of viruses, interception
> or interference and anyone who communicates with us by email is taken
> to accept the risks in doing so.  Without limitation, OOCL and its
> affiliates
> accept no liability whatsoever and howsoever arising in connection with
> the use of this email.  Under no circumstances shall this email constitute
> a binding agreement to carry or for provision of carriage services by OOCL,
> which is subject to the availability of carrier's equipment and vessels and
> the terms and conditions of OOCL's standard bill of lading which is also
> available at http://www.oocl.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*
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
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From gmaruzz at gmail.com  Mon Jul 27 17:56:49 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 27 Jul 2015 15:56:49 +0200
Subject: [Freeswitch-users] Video conferencing
In-Reply-To: <CAD_iGr9nKLZ9_M2Sc-nNC58xS579aYbX62EDUcNj2ynftjsUNA@mail.gmail.com>
References: <CAD_iGr9nKLZ9_M2Sc-nNC58xS579aYbX62EDUcNj2ynftjsUNA@mail.gmail.com>
Message-ID: <CALXCt0pB5xbD1kFBEurZ2NRLea47JWdCEKxp4o2cSzsnzRPL3g@mail.gmail.com>

video-floor means when you are the one that get's the camera, the one from
which the main video stream is generated, the one the show is focused on,
the one "on the floor"

there is no theory, those are just spoken terms, eg terms to be understood
when you talk with fellow developers.

Experiments with different features and layouts, and you'll get it all

On Mon, Jul 27, 2015 at 11:42 AM, Alexandru Covalschi <568691 at gmail.com>
wrote:

> Hello! Started to figure out the theme in topic. Didn't find any
> descriptions of what is "video-floor" and etc. Can you please provide a
> link to the theory I need to read to understand all definitions?
>
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> phone: +37367398493
> web: http://abs-telecom.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
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From menendez.garcia at gmail.com  Mon Jul 27 18:45:24 2015
From: menendez.garcia at gmail.com (Javier Menendez)
Date: Mon, 27 Jul 2015 16:45:24 +0200
Subject: [Freeswitch-users] audio Handshake failure 1
In-Reply-To: <CAGf2z7NE0a7mPsPXjZK9w06jnTJTh--R01O2dvBRyhQPxRmxmw@mail.gmail.com>
References: <CAGf2z7NE0a7mPsPXjZK9w06jnTJTh--R01O2dvBRyhQPxRmxmw@mail.gmail.com>
Message-ID: <CAK59s4w87K5dftvFVwJmNRmTqy29A8rM5UmfSyPnzMaFjs0cYA@mail.gmail.com>

Please check *https://freeswitch.org/jira/browse/FS-7839
<https://freeswitch.org/jira/browse/FS-7839> *

On Mon, Jul 27, 2015 at 8:33 AM, Alex Polischuk <alxpol at gmail.com> wrote:

> Hi All,
>
> I use FreeSwitch for transcoding SRTP from WebRTC to SIP with RTP and vice
> versa.
> When Firefox version 38 add requires for Perfect Forward Secrecy (PFS) I
> have problem with Handshake in FS when Firefox Browser terminate a call. I
> found that that problem was fixed in the version 1.4.19
> https://freeswitch.org/jira/browse/FS-7425
>
> I have the same problem in the version 1.4.20
>
> eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
> switch_rtp.c:2924 Changing audio DTLS state from HANDSHAKE to SETUP
> eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
> switch_rtp.c:2832 audio Fingerprint Verified.
> eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
> switch_rtp.c:3384 Activating Audio Secure RTP SEND
> eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
> switch_rtp.c:3362 Activating Audio Secure RTP RECV
> eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
> switch_rtp.c:2872 Changing audio DTLS state from SETUP to READY
> eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.604193 [DEBUG]
> switch_rtp.c:1937 rtcp_stats_init: ssrc[-1374535478] base_seq[6179]
> f01c46d8-33a0-11e5-ac83-891cb7dea826 2015-07-26 14:16:46.644210 [DEBUG]
> switch_rtp.c:5884 Correct ip/port confirmed.
> f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [ERR]
> switch_rtp.c:2917 audio Handshake failure 1
> f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [INFO]
> switch_rtp.c:2918 Changing audio DTLS state from HANDSHAKE to FAIL
> f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [NOTICE]
> switch_rtp.c:2899 Hangup sofia/external/992555500004 at voip.webrtc.jajah.com
> [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER]
>
>
> what additional definition I should make to resolve this problem?
>
> Thanks,
> Alex Polischuk
>
>
> _________________________________________________________________________
> 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.org  Mon Jul 27 18:52:44 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 27 Jul 2015 09:52:44 -0500
Subject: [Freeswitch-users] Coder Games Monday August 3rd! Remote
	Participants needed!
Message-ID: <CAEJMVkCh9eyr++hFbvtZVtuVxz0Zc-csjV53=XEoAacn7KGdDg@mail.gmail.com>

We are playing a version of Hollywood Squares, but it will be VoIP related
questions, If you would like to participate, you'll need a camera, headset
and Chrome to participate, Please RSVP to kathleen at freeswitch.org your
details, and we'll do a tech check this week sometime.

https://en.wikipedia.org/wiki/Hollywood_Squares

If you get a chance watch some videos on Youtube.

Thanks,

-- 

*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
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From kamil.nigmatullin at gmail.com  Mon Jul 27 19:11:10 2015
From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin)
Date: Mon, 27 Jul 2015 21:11:10 +0600
Subject: [Freeswitch-users] freeswitch never seems to call sendmail when
 trying to email voicemails
In-Reply-To: <CAN6oSBuxXbf=uecMk1UO=ZP9xxpvoXA65Ox4pyZVmPxA-BNwnQ@mail.gmail.com>
References: <54E6B9A1.20708@dickson.st>
	<CAHtxdDd8CRENTyNoi7FOn3yuy+fyCshEiknfhvhZcg36q=JFuA@mail.gmail.com>
	<54EA5B78.3010504@dickson.st>
	<CAHtxdDcD0fqTWR9mA8=-eT9rm2a1M1p3Qab7OVXCfH=wYnHvvw@mail.gmail.com>
	<54EBE76A.2040709@dickson.st>
	<CAHtxdDfsm2pd8HDM-WwU8HKJT5AQj3isqjWF4aWVbXkRrtB0zg@mail.gmail.com>
	<CAEJMVkBY-3nOfXijxhED_byC_K5cNnS8LVW3iRjGmOFserDmjw@mail.gmail.com>
	<CAN6oSBuxXbf=uecMk1UO=ZP9xxpvoXA65Ox4pyZVmPxA-BNwnQ@mail.gmail.com>
Message-ID: <CA+Xtiefu7T+LBFD11k38wPG-QCPcrWJuaAaLGTOmS_RTmk8m7A@mail.gmail.com>

Why can.t you create your own script and use it to send vm over email?
27 ???? 2015 ?. 18:32 ???????????? "Berthold Karl" <
sendmeallyouroffers at googlemail.com> ???????:

> Hi,
>
> my FS is calling the sendmail-command, but sendmail ends with an segfault.
> It seems like there is no email under /tmp/. Did your sendmail also ends in
> a segfault?
>
> 2015-02-24 16:51 GMT+01:00 Brian West <brian at freeswitch.org>:
>
>> When you file the JIRA please attach your voicemail.conf.xml.
>>
>> On Tue, Feb 24, 2015 at 8:21 AM, Sergey Safarov <s.safarov at gmail.com>
>> wrote:
>>
>>> If you want to FS correctly processed parameters vm-mailfrom
>>> and email-from, write a request to https://freeswitch.org/jira/
>>>
>>> On Tue, Feb 24, 2015 at 5:52 AM, Jason Lewis <jason at dickson.st> wrote:
>>>
>>>>  Thanks Sergey,
>>>>
>>>> I had configured the domain variable to the fqdn of the machine. I
>>>> eventually got it working though, I was missing two key lines from my user
>>>> config:
>>>>       <param name="vm-email-all-messages" value="true"/>
>>>>       <param name="vm-attach-file" value="true" />
>>>>
>>>> I'm not sure how I managed to miss those but anyway, that seems to have
>>>> resolved things.
>>>>
>>>> It seems as though vm-mailfrom is still being ignored though. Currently
>>>> I have it set to:
>>>>
>>>>        <param name="vm-mailfrom" value="vm at freeswitch.xyz.com.au"
>>>> <vm at freeswitch.xyz.com.au>/>
>>>>
>>>> but voicemails get delivered from:
>>>>
>>>> 1001 at freeswitch.xyz.com.au
>>>>
>>>> Is this worth investigating further?
>>>>
>>>> Jason
>>>>
>>>> Sergey Safarov wrote on 23/02/2015 5:05 PM:
>>>>
>>>> Try configure "domain" variable in vars.xml
>>>>
>>>>  <X-PRE-PROCESS cmd="set" data="domain=you_domain_name"/>
>>>>
>>>>  After it verify that user registered with domain name
>>>>
>>>>  freeswitch at internal> sofia status profile internal reg
>>>>
>>>>  Registrations:
>>>>
>>>> =================================================================================================
>>>> Call-ID:     1B26-2327-466848134BEBC9719CDE-002 at SipHost
>>>> User:       1201 at you_domain_name
>>>> Contact:     "1201"
>>>> <sip:1201 at 10.21.18.22:5060;fs_nat=yes;fs_path=sip%3A1201%4010.21.18.22%3A5060>
>>>> Agent:       204 12-3868-2416-0.10.56.1-DS
>>>> Status:     Registered(UDP-NAT)(unknown) EXP(2015-02-23 06:05:22)
>>>> EXPSECS(139)
>>>> Ping-Status: Reachable
>>>> Host:       fs1.you_domain_name
>>>> IP:         10.21.18.22
>>>> Port:       5060
>>>> Auth-User:   1201
>>>> Auth-Realm: you_domain_name
>>>> MWI-Account: 1201 at you_domain_name
>>>>
>>>>  Sergey
>>>>
>>>>
>>>> On Mon, Feb 23, 2015 at 1:43 AM, Jason Lewis <jason at dickson.st> wrote:
>>>>
>>>>>  So I've managed to see some output from the sendmail program in the
>>>>> FS logs. It appears that my fs instance isn't correctly setting its domain?
>>>>>
>>>>> the FS box has a fqdn, and I also set the domain parameter in the
>>>>> vars.xml file, but still the voicemail is sent with a from address of an IP
>>>>> address.
>>>>>
>>>>> Any ideas?
>>>>>
>>>>> Net::SMTP>>> Net::SMTP(2.33)
>>>>> Net::SMTP>>>   Net::Cmd(2.30)
>>>>> Net::SMTP>>>     Exporter(5.71)
>>>>> Net::SMTP>>>   IO::Socket::INET(1.35)
>>>>> Net::SMTP>>>     IO::Socket(1.37)
>>>>> Net::SMTP>>>       IO::Handle(1.35)
>>>>> Net::SMTP=GLOB(0x23f7748)<<< 220 mb.xyz.com.au ESMTP Postfix
>>>>> (Debian/GNU)
>>>>> Net::SMTP=GLOB(0x23f7748)>>> EHLO localhost.localdomain
>>>>> Net::SMTP=GLOB(0x23f7748)<<< 250-mb.bongalong.st
>>>>> Net::SMTP=GLOB(0x23f7748)<<< 250-PIPELINING
>>>>> Net::SMTP=GLOB(0x23f7748)<<< 250-SIZE 10240000
>>>>> Net::SMTP=GLOB(0x23f7748)<<< 250-VRFY
>>>>> Net::SMTP=GLOB(0x23f7748)<<< 250-ETRN
>>>>> Net::SMTP=GLOB(0x23f7748)<<< 250-STARTTLS
>>>>> Net::SMTP=GLOB(0x23f7748)<<< 250-ENHANCEDSTATUSCODES
>>>>> Net::SMTP=GLOB(0x23f7748)<<< 250-8BITMIME
>>>>> Net::SMTP=GLOB(0x23f7748)<<< 250 DSN
>>>>> Net::SMTP=GLOB(0x23f7748)>>> MAIL FROM:<1002 at 192.168.1.3>
>>>>> <1002 at 192.168.1.3>
>>>>> Net::SMTP=GLOB(0x23f7748)<<< 501 5.1.7 Bad sender address syntax
>>>>>  Net::SMTP=GLOB(0x23f7748)>>> RCPT TO:<jason at xyz.com.au>
>>>>> <jason at xyz.com.au>
>>>>> Net::SMTP=GLOB(0x23f7748)<<< 503 5.5.1 Error: need MAIL command
>>>>> Net::SMTP=GLOB(0x23f7748)>>> DATA
>>>>> Net::SMTP=GLOB(0x23f7748)<<< 503 5.5.1 Error: need RCPT command
>>>>> Net::SMTP=GLOB(0x23f7748)>>> QUIT
>>>>> Net::SMTP=GLOB(0x23f7748)<<< 221 2.0.0 Bye
>>>>> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1633 Session
>>>>> 3 (loopback/voicemail-a) Ended
>>>>> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1637 Close
>>>>> Channel loopback/voicemail-a [CS_DESTROY]
>>>>> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1633 Session
>>>>> 4 (loopback/voicemail-b) Ended
>>>>> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1637 Close
>>>>> Channel loopback/voicemail-b [CS_DESTROY]
>>>>>
>>>>>
>>>>> On 20/02/2015 7:10 pm, Sergey Safarov wrote:
>>>>>
>>>>>  You mailer is not understand "mailer-app-args" has been configured
>>>>> in "switch.conf.xml"
>>>>>
>>>>> Remove extra arg or add required
>>>>>
>>>>> ??, 20 ????. 2015, 7:49, Jason Lewis <jason at dickson.st>:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I've been trying to make freeswitch email voicemails but as far as I
>>>>>> can
>>>>>> tell, it never even calls sendmail.
>>>>>>
>>>>>> I have setting mailer-app to "sendmail" and "/usr/sbin/sendmail" to no
>>>>>> avail. I can successfully send an email from the commandline using
>>>>>> sendmail. (sendmail in this case is provided by postfix)
>>>>>>
>>>>>> I see no emails in the postfix mail logs when I leave a voicemail
>>>>>> message.
>>>>>>
>>>>>> I also tried creating a shell just to see if it even gets called from
>>>>>> fs, but it does not get called when a voicemail is deposited:
>>>>>> #!/bin/bash
>>>>>> echo $(date --rfc-3339=ns): $* >> /tmp/freeswitchsendmail.log
>>>>>>
>>>>>> After every change, I have run reloadxml and reload mod_voicemail. I
>>>>>> have also tried restarting freeswitch.
>>>>>>
>>>>>> I am running the debian packages of FreeSWITCH Version 1.4.15-1~64bit
>>>>>> (-1 64bit)
>>>>>>
>>>>>> my configuration is based on the vanilla configuration with only very
>>>>>> minor changes.
>>>>>>
>>>>>> I'm at a loss as to how to debug further, but I'm pretty sure the
>>>>>> mailer-app is never called. Is there some setting I'm missing or
>>>>>> something obvious I'm not doing?
>>>>>>
>>>>>>
>>>>>> My config:
>>>>>> 1001.xml:
>>>>>> <include>
>>>>>>   <user id="1001">
>>>>>>     <params>
>>>>>>       <param name="password" value="$${default_password}"/>
>>>>>>       <param name="vm-password" value="1001"/>
>>>>>>       <param name="vm-mailto" value="jason at xyz.com.au"/>
>>>>>>       <param name="vm-mailfrom" value="vm at freeswitch.xyz.com.au"/>
>>>>>>       <!-- <param name="vm-message-ext" value="wav"/> -->
>>>>>>     </params>
>>>>>>     <variables>
>>>>>>       <variable name="toll_allow"
>>>>>> value="domestic,international,local"/>
>>>>>>       <variable name="accountcode" value="1001"/>
>>>>>>       <variable name="user_context" value="default"/>
>>>>>>       <variable name="effective_caller_id_name" value="Extension
>>>>>> 1001"/>
>>>>>>       <variable name="effective_caller_id_number" value="1001"/>
>>>>>>       <variable name="outbound_caller_id_name"
>>>>>> value="$${outbound_caller_name}"/>
>>>>>>       <variable name="outbound_caller_id_number"
>>>>>> value="$${outbound_caller_id}"/>
>>>>>>       <variable name="callgroup" value="techsupport"/>
>>>>>>     </variables>
>>>>>>   </user>
>>>>>> </include>
>>>>>>
>>>>>> and in switch.conf.xml I have the following set:
>>>>>>
>>>>>>     <param name="mailer-app" value="/usr/sbin/sendmail"/>
>>>>>>     <param name="mailer-app-args" value="-t"/>
>>>>>>
>>>>>>
>>>>>> I made a log at level 7 and put it on the pastebin:
>>>>>>
>>>>>> https://pastebin.freeswitch.org/23921
>>>>>>
>>>>>>
>>>>>> Jason Lewis
>>>>>> http://emacstragic.net
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> 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.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
>>>>>
>>>>>
>>>>>
>>>>>   --
>>>>> Jason Lewishttp://emacstragic.net
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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.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
>>>>
>>>>
>>>> --
>>>> Jason Lewishttp://emacstragic.net
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> 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 yadenis at seznam.cz  Mon Jul 27 19:33:02 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Mon, 27 Jul 2015 17:33:02 +0200
Subject: [Freeswitch-users] freeswitch never seems to call sendmail when
	trying to email voicemails
In-Reply-To: <CA+Xtiefu7T+LBFD11k38wPG-QCPcrWJuaAaLGTOmS_RTmk8m7A@mail.gmail.com>
References: <54E6B9A1.20708@dickson.st> 
	<CAHtxdDd8CRENTyNoi7FOn3yuy+fyCshEiknfhvhZcg36q=JFuA@mail.gmail.com>
	<54EA5B78.3010504@dickson.st>
	<CAHtxdDcD0fqTWR9mA8=-eT9rm2a1M1p3Qab7OVXCfH=wYnHvvw@mail.gmail.com>
	<54EBE76A.2040709@dickson.st>
	<CAHtxdDfsm2pd8HDM-WwU8HKJT5AQj3isqjWF4aWVbXkRrtB0zg@mail.gmail.com>
	<CAEJMVkBY-3nOfXijxhED_byC_K5cNnS8LVW3iRjGmOFserDmjw@mail.gmail.com>
	<CAN6oSBuxXbf=uecMk1UO=ZP9xxpvoXA65Ox4pyZVmPxA-BNwnQ@mail.gmail.com>
	<CA+Xtiefu7T+LBFD11k38wPG-QCPcrWJuaAaLGTOmS_RTmk8m7A@mail.gmail.com>
Message-ID: <29646517.20150727173302@seznam.cz>

Hi,

Why do not use some simple script after the ends session?

For example simple bash script send_voicemail.sh

#!/bin/sh
cd /usr/local/freeswitch/recordings/voicemail/
f=`find -name \*.wav`
for file in $f
do
echo "Processing ${file}"
CURRENT=$(date +%d.%m.%y_%H:%M:%S)
datetime=$CURRENT
echo $datetime
sendemail -f voicemail at mail.com -t somemail at mail.com -m "Voicemail" -u "Voicemail od $datetime" -a ${file} -s smtp.mail.com -xu user -xp pass
echo "BackUP ${file}"
mv ${file} /usr/local/freeswitch/recordings/backup_voicemail/
done


And in dialplan start like this
<action application="export" data="api_hangup_hook=system /usr/local/freeswitch/scripts/send_voicemail.sh"/>



-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

pond?l? 27. ?ervence 2015, 17:11:10, napsal jste:


Why can.t you create your own script and use it to send vm over email?
27 ???? 2015 ?. 18:32 ???????????? "Berthold Karl" <sendmeallyouroffers at googlemail.com> ???????:
Hi,

my FS is calling the sendmail-command, but sendmail ends with an segfault. It seems like there is no email under /tmp/. Did your sendmail also ends in a segfault?

2015-02-24 16:51 GMT+01:00 Brian West <brian at freeswitch.org>:
When you file the JIRA please attach your voicemail.conf.xml.

On Tue, Feb 24, 2015 at 8:21 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
If you want to FS correctly processed parameters vm-mailfrom and email-from, write a request to https://freeswitch.org/jira/

On Tue, Feb 24, 2015 at 5:52 AM, Jason Lewis <jason at dickson.st> wrote:
Thanks Sergey,

I had configured the domain variable to the fqdn of the machine. I eventually got it working though, I was missing two key lines from my user config: 
      <param name="vm-email-all-messages" value="true"/>
      <param name="vm-attach-file" value="true" />

I'm not sure how I managed to miss those but anyway, that seems to have resolved things.

It seems as though vm-mailfrom is still being ignored though. Currently I have it set to:

       <param name="vm-mailfrom" value="vm at freeswitch.xyz.com.au"/>

but voicemails get delivered from:

1001 at freeswitch.xyz.com.au

Is this worth investigating further?

Jason

Sergey Safarov wrote on 23/02/2015 5:05 PM:
Try configure "domain" variable in vars.xml 

<X-PRE-PROCESS cmd="set" data="domain=you_domain_name"/>

After it verify that user registered with domain name

freeswitch at internal> sofia status profile internal reg

Registrations:
=================================================================================================
Call-ID:    1B26-2327-466848134BEBC9719CDE-002 at SipHost
User:       1201 at you_domain_name
Contact:    "1201" <sip:1201 at 10.21.18.22:5060;fs_nat=yes;fs_path=sip%3A1201%4010.21.18.22%3A5060>
Agent:      204 12-3868-2416-0.10.56.1-DS
Status:     Registered(UDP-NAT)(unknown) EXP(2015-02-23 06:05:22) EXPSECS(139)
Ping-Status:Reachable
Host:       fs1.you_domain_name
IP:         10.21.18.22
Port:       5060
Auth-User:  1201
Auth-Realm: you_domain_name
MWI-Account:1201 at you_domain_name

Sergey


On Mon, Feb 23, 2015 at 1:43 AM, Jason Lewis <jason at dickson.st> wrote:
So I've managed to see some output from the sendmail program in the FS logs. It appears that my fs instance isn't correctly setting its domain?

the FS box has a fqdn, and I also set the domain parameter in the vars.xml file, but still the voicemail is sent with a from address of an IP address.

Any ideas?

Net::SMTP>>> Net::SMTP(2.33)
Net::SMTP>>>   Net::Cmd(2.30)
Net::SMTP>>>     Exporter(5.71)
Net::SMTP>>>   IO::Socket::INET(1.35)
Net::SMTP>>>     IO::Socket(1.37)
Net::SMTP>>>       IO::Handle(1.35)
Net::SMTP=GLOB(0x23f7748)<<< 220 mb.xyz.com.au ESMTP Postfix (Debian/GNU)
Net::SMTP=GLOB(0x23f7748)>>> EHLO localhost.localdomain
Net::SMTP=GLOB(0x23f7748)<<< 250-mb.bongalong.st
Net::SMTP=GLOB(0x23f7748)<<< 250-PIPELINING
Net::SMTP=GLOB(0x23f7748)<<< 250-SIZE 10240000
Net::SMTP=GLOB(0x23f7748)<<< 250-VRFY
Net::SMTP=GLOB(0x23f7748)<<< 250-ETRN
Net::SMTP=GLOB(0x23f7748)<<< 250-STARTTLS
Net::SMTP=GLOB(0x23f7748)<<< 250-ENHANCEDSTATUSCODES
Net::SMTP=GLOB(0x23f7748)<<< 250-8BITMIME
Net::SMTP=GLOB(0x23f7748)<<< 250 DSN
Net::SMTP=GLOB(0x23f7748)>>> MAIL FROM:<1002 at 192.168.1.3>
Net::SMTP=GLOB(0x23f7748)<<< 501 5.1.7 Bad sender address syntax
Net::SMTP=GLOB(0x23f7748)>>> RCPT TO:<jason at xyz.com.au>
Net::SMTP=GLOB(0x23f7748)<<< 503 5.5.1 Error: need MAIL command
Net::SMTP=GLOB(0x23f7748)>>> DATA
Net::SMTP=GLOB(0x23f7748)<<< 503 5.5.1 Error: need RCPT command
Net::SMTP=GLOB(0x23f7748)>>> QUIT
Net::SMTP=GLOB(0x23f7748)<<< 221 2.0.0 Bye
2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1633 Session 3 (loopback/voicemail-a) Ended
2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1637 Close Channel loopback/voicemail-a [CS_DESTROY]
2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1633 Session 4 (loopback/voicemail-b) Ended
2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1637 Close Channel loopback/voicemail-b [CS_DESTROY]


On 20/02/2015 7:10 pm, Sergey Safarov wrote:
You mailer is not understand "mailer-app-args" has been configured in "switch.conf.xml"
Remove extra arg or add required

??, 20 ????. 2015, 7:49, Jason Lewis <jason at dickson.st>:
Hi,

I've been trying to make freeswitch email voicemails but as far as I can
tell, it never even calls sendmail.

I have setting mailer-app to "sendmail" and "/usr/sbin/sendmail" to no
avail. I can successfully send an email from the commandline using
sendmail. (sendmail in this case is provided by postfix)

I see no emails in the postfix mail logs when I leave a voicemail message.

I also tried creating a shell just to see if it even gets called from
fs, but it does not get called when a voicemail is deposited:
#!/bin/bash
echo $(date --rfc-3339=ns): $* >> /tmp/freeswitchsendmail.log

After every change, I have run reloadxml and reload mod_voicemail. I
have also tried restarting freeswitch.

I am running the debian packages of FreeSWITCH Version 1.4.15-1~64bit
(-1 64bit)

my configuration is based on the vanilla configuration with only very
minor changes.

I'm at a loss as to how to debug further, but I'm pretty sure the
mailer-app is never called. Is there some setting I'm missing or
something obvious I'm not doing?


My config:
1001.xml:
<include>
  <user id="1001">
    <params>
      <param name="password" value="$${default_password}"/>
      <param name="vm-password" value="1001"/>
      <param name="vm-mailto" value="jason at xyz.com.au"/>
      <param name="vm-mailfrom" value="vm at freeswitch.xyz.com.au"/>
      <!-- <param name="vm-message-ext" value="wav"/> -->
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1001"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 1001"/>
      <variable name="effective_caller_id_number" value="1001"/>
      <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number"
value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

and in switch.conf.xml I have the following set:

    <param name="mailer-app" value="/usr/sbin/sendmail"/>
    <param name="mailer-app-args" value="-t"/>


I made a log at level 7 and put it on the pastebin:

https://pastebin.freeswitch.org/23921


Jason Lewis
http://emacstragic.net


_________________________________________________________________________
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


-- 

Jason Lewis

http://emacstragic.net

_________________________________________________________________________
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

-- 

Jason Lewis

http://emacstragic.net

_________________________________________________________________________
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
brian at freeswitch.org
Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com
T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

_________________________________________________________________________
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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_________________________________________________________________________
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 mario_fs at mgtech.com  Mon Jul 27 21:01:37 2015
From: mario_fs at mgtech.com (Mario)
Date: Mon, 27 Jul 2015 10:01:37 -0700
Subject: [Freeswitch-users] freeswitch never seems to call sendmail when
	trying to email voicemails
In-Reply-To: <29646517.20150727173302@seznam.cz>
References: <54E6B9A1.20708@dickson.st>
	<CAHtxdDd8CRENTyNoi7FOn3yuy+fyCshEiknfhvhZcg36q=JFuA@mail.gmail.com>
	<54EA5B78.3010504@dickson.st>
	<CAHtxdDcD0fqTWR9mA8=-eT9rm2a1M1p3Qab7OVXCfH=wYnHvvw@mail.gmail.com>
	<54EBE76A.2040709@dickson.st>
	<CAHtxdDfsm2pd8HDM-WwU8HKJT5AQj3isqjWF4aWVbXkRrtB0zg@mail.gmail.com>
	<CAEJMVkBY-3nOfXijxhED_byC_K5cNnS8LVW3iRjGmOFserDmjw@mail.gmail.com>
	<CAN6oSBuxXbf=uecMk1UO=ZP9xxpvoXA65Ox4pyZVmPxA-BNwnQ@mail.gmail.com>
	<CA+Xtiefu7T+LBFD11k38wPG-QCPcrWJuaAaLGTOmS_RTmk8m7A@mail.gmail.com>
	<29646517.20150727173302@seznam.cz>
Message-ID: <C18BE15B-86F3-40C1-ABED-CA328565C0C1@mgtech.com>

Although this is in the Mac OS X page I wrote this up which may help: https://freeswitch.org/confluence/display/FREESWITCH/Installation+and+Setup+on+OS+X#InstallationandSetuponOSX-EmailVoicemailtoaniPhone

To test your email you can do this from a command line:
printf "Subject: TestnHello" | sendmail -f you at domain.com you at domain.com
Mareio G
> On Jul 27, 2015, at 8:33 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
> 
> Hi,
> 
> Why do not use some simple script after the ends session?
> 
> For example simple bash script send_voicemail.sh
> 
> #!/bin/sh
> cd /usr/local/freeswitch/recordings/voicemail/
> f=`find -name \*.wav`
> for file in $f
> do
> echo "Processing ${file}"
> CURRENT=$(date +%d.%m.%y_%H:%M:%S)
> datetime=$CURRENT
> echo $datetime
> sendemail -f voicemail at mail.com -t somemail at mail.com -m "Voicemail" -u "Voicemail od $datetime" -a ${file} -s smtp.mail.com -xu user -xp pass
> echo "BackUP ${file}"
> mv ${file} /usr/local/freeswitch/recordings/backup_voicemail/
> done
> 
> 
> And in dialplan start like this
> <action application="export" data="api_hangup_hook=system /usr/local/freeswitch/scripts/send_voicemail.sh"/>
> 
> 
> 
> -- 
> S pozdravem,
> Ing.Denis Jakovlev                           
> mob.tel. 775-415-382
> 
> pond?l? 27. ?ervence 2015, 17:11:10, napsal jste:
> 
> 
> Why can.t you create your own script and use it to send vm over email?
> 27 ???? 2015 ?. 18:32 ???????????? "Berthold Karl" <sendmeallyouroffers at googlemail.com <mailto:sendmeallyouroffers at googlemail.com>> ???????:
> Hi,
> 
> my FS is calling the sendmail-command, but sendmail ends with an segfault. It seems like there is no email under /tmp/. Did your sendmail also ends in a segfault?
> 
> 2015-02-24 16:51 GMT+01:00 Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>>:
> When you file the JIRA please attach your voicemail.conf.xml.
> 
> On Tue, Feb 24, 2015 at 8:21 AM, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
> If you want to FS correctly processed parameters vm-mailfrom and email-from, write a request to https://freeswitch.org/jira/ <https://freeswitch.org/jira/>
> 
> On Tue, Feb 24, 2015 at 5:52 AM, Jason Lewis <jason at dickson.st <mailto:jason at dickson.st>> wrote:
> Thanks Sergey,
> 
> I had configured the domain variable to the fqdn of the machine. I eventually got it working though, I was missing two key lines from my user config: 
>      <param name="vm-email-all-messages" value="true"/>
>      <param name="vm-attach-file" value="true" />
> 
> I'm not sure how I managed to miss those but anyway, that seems to have resolved things.
> 
> It seems as though vm-mailfrom is still being ignored though. Currently I have it set to:
> 
>       <param name="vm-mailfrom" value="vm at freeswitch.xyz.com.au" <mailto:vm at freeswitch.xyz.com.au>/>
> 
> but voicemails get delivered from:
> 
> 1001 at freeswitch.xyz.com.au <mailto:1001 at freeswitch.xyz.com.au>
> 
> Is this worth investigating further?
> 
> Jason
> 
> Sergey Safarov wrote on 23/02/2015 5:05 PM:
> Try configure "domain" variable in vars.xml 
> 
> <X-PRE-PROCESS cmd="set" data="domain=you_domain_name"/>
> 
> After it verify that user registered with domain name
> 
> freeswitch at internal> sofia status profile internal reg
> 
> Registrations:
> =================================================================================================
> Call-ID:    1B26-2327-466848134BEBC9719CDE-002 at SipHost
> User:       1201 at you_domain_name
> Contact:    "1201" <sip:1201 at 10.21.18.22:5060;fs_nat=yes;fs_path=sip%3A1201%4010.21.18.22%3A5060>
> Agent:      204 12-3868-2416-0.10.56.1-DS
> Status:     Registered(UDP-NAT)(unknown) EXP(2015-02-23 06:05:22) EXPSECS(139)
> Ping-Status:Reachable
> Host:       fs1.you_domain_name
> IP:         10.21.18.22
> Port:       5060
> Auth-User:  1201
> Auth-Realm: you_domain_name
> MWI-Account:1201 at you_domain_name
> 
> Sergey
> 
> 
> On Mon, Feb 23, 2015 at 1:43 AM, Jason Lewis <jason at dickson.st <mailto:jason at dickson.st>> wrote:
> So I've managed to see some output from the sendmail program in the FS logs. It appears that my fs instance isn't correctly setting its domain?
> 
> the FS box has a fqdn, and I also set the domain parameter in the vars.xml file, but still the voicemail is sent with a from address of an IP address.
> 
> Any ideas?
> 
> Net::SMTP>>> Net::SMTP(2.33)
> Net::SMTP>>>   Net::Cmd(2.30)
> Net::SMTP>>>     Exporter(5.71)
> Net::SMTP>>>   IO::Socket::INET(1.35)
> Net::SMTP>>>     IO::Socket(1.37)
> Net::SMTP>>>       IO::Handle(1.35)
> Net::SMTP=GLOB(0x23f7748)<<< 220 mb.xyz.com.au <http://mb.xyz.com.au/> ESMTP Postfix (Debian/GNU)
> Net::SMTP=GLOB(0x23f7748)>>> EHLO localhost.localdomain
> Net::SMTP=GLOB(0x23f7748)<<< 250-mb.bongalong.st <http://250-mb.bongalong.st/>
> Net::SMTP=GLOB(0x23f7748)<<< 250-PIPELINING
> Net::SMTP=GLOB(0x23f7748)<<< 250-SIZE 10240000
> Net::SMTP=GLOB(0x23f7748)<<< 250-VRFY
> Net::SMTP=GLOB(0x23f7748)<<< 250-ETRN
> Net::SMTP=GLOB(0x23f7748)<<< 250-STARTTLS
> Net::SMTP=GLOB(0x23f7748)<<< 250-ENHANCEDSTATUSCODES
> Net::SMTP=GLOB(0x23f7748)<<< 250-8BITMIME
> Net::SMTP=GLOB(0x23f7748)<<< 250 DSN
> Net::SMTP=GLOB(0x23f7748)>>> MAIL FROM:<1002 at 192.168.1.3> <mailto:1002 at 192.168.1.3>
> Net::SMTP=GLOB(0x23f7748)<<< 501 5.1.7 Bad sender address syntax
> Net::SMTP=GLOB(0x23f7748)>>> RCPT TO:<jason at xyz.com.au> <mailto:jason at xyz.com.au>
> Net::SMTP=GLOB(0x23f7748)<<< 503 5.5.1 Error: need MAIL command
> Net::SMTP=GLOB(0x23f7748)>>> DATA
> Net::SMTP=GLOB(0x23f7748)<<< 503 5.5.1 Error: need RCPT command
> Net::SMTP=GLOB(0x23f7748)>>> QUIT
> Net::SMTP=GLOB(0x23f7748)<<< 221 2.0.0 Bye
> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1633 Session 3 (loopback/voicemail-a) Ended
> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1637 Close Channel loopback/voicemail-a [CS_DESTROY]
> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1633 Session 4 (loopback/voicemail-b) Ended
> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1637 Close Channel loopback/voicemail-b [CS_DESTROY]
> 
> 
> On 20/02/2015 7:10 pm, Sergey Safarov wrote:
> You mailer is not understand "mailer-app-args" has been configured in "switch.conf.xml"
> Remove extra arg or add required
> 
> ??, 20 ????. 2015, 7:49, Jason Lewis <jason at dickson.st <mailto:jason at dickson.st>>:
> Hi,
> 
> I've been trying to make freeswitch email voicemails but as far as I can
> tell, it never even calls sendmail.
> 
> I have setting mailer-app to "sendmail" and "/usr/sbin/sendmail" to no
> avail. I can successfully send an email from the commandline using
> sendmail. (sendmail in this case is provided by postfix)
> 
> I see no emails in the postfix mail logs when I leave a voicemail message.
> 
> I also tried creating a shell just to see if it even gets called from
> fs, but it does not get called when a voicemail is deposited:
> #!/bin/bash
> echo $(date --rfc-3339=ns): $* >> /tmp/freeswitchsendmail.log
> 
> After every change, I have run reloadxml and reload mod_voicemail. I
> have also tried restarting freeswitch.
> 
> I am running the debian packages of FreeSWITCH Version 1.4.15-1~64bit
> (-1 64bit)
> 
> my configuration is based on the vanilla configuration with only very
> minor changes.
> 
> I'm at a loss as to how to debug further, but I'm pretty sure the
> mailer-app is never called. Is there some setting I'm missing or
> something obvious I'm not doing?
> 
> 
> My config:
> 1001.xml:
> <include>
>  <user id="1001">
>    <params>
>      <param name="password" value="$${default_password}"/>
>      <param name="vm-password" value="1001"/>
>      <param name="vm-mailto" value="jason at xyz.com.au <mailto:jason at xyz.com.au>"/>
>      <param name="vm-mailfrom" value="vm at freeswitch.xyz.com.au <mailto:vm at freeswitch.xyz.com.au>"/>
>      <!-- <param name="vm-message-ext" value="wav"/> -->
>    </params>
>    <variables>
>      <variable name="toll_allow" value="domestic,international,local"/>
>      <variable name="accountcode" value="1001"/>
>      <variable name="user_context" value="default"/>
>      <variable name="effective_caller_id_name" value="Extension 1001"/>
>      <variable name="effective_caller_id_number" value="1001"/>
>      <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>      <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>      <variable name="callgroup" value="techsupport"/>
>    </variables>
>  </user>
> </include>
> 
> and in switch.conf.xml I have the following set:
> 
>    <param name="mailer-app" value="/usr/sbin/sendmail"/>
>    <param name="mailer-app-args" value="-t"/>
> 
> 
> I made a log at level 7 and put it on the pastebin:
> 
> https://pastebin.freeswitch.org/23921 <https://pastebin.freeswitch.org/23921>
> 
> 
> Jason Lewis
> http://emacstragic.net <http://emacstragic.net/>
> 
> 
> _________________________________________________________________________
> 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/>
> 
> 
> -- 
> Jason Lewis
> http://emacstragic.net <http://emacstragic.net/>
> 
> _________________________________________________________________________
> 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/>
> 
> -- 
> Jason Lewis
> http://emacstragic.net <http://emacstragic.net/>
> 
> _________________________________________________________________________
> 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/>
> 
> 
> 
> -- 
> Brian West
> brian at freeswitch.org <mailto:brian at freeswitch.org>
> Twitter: @FreeSWITCH , @briankwest
> http://www.freeswitchbook.com <http://www.freeswitchbook.com/>
> http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/>
> T:+19184209001 <tel:%2B19184209001> | F:+19184209002 <tel:%2B19184209002> | M:+1918424WEST (9378)
> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
> 
> _________________________________________________________________________
> 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/>_________________________________________________________________________
> 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.org  Mon Jul 27 21:46:09 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 27 Jul 2015 12:46:09 -0500
Subject: [Freeswitch-users] FreeSWITCH Automated Testing
Message-ID: <CAEJMVkDGD_mP6kUVToG4W=i6o1c8+MBxKz0dhX1JShhhDGDxbQ@mail.gmail.com>

FreeSWITCHers,

So after a bit discussion, a case of red bull, an approach to automating
functional testing has now solidified.

Audio Path is verified via DTMF in-band, Video Path is verified by using QR
codes.

Here are two VERY crude examples:

https://www.dropbox.com/s/4fkdcgzo8dt0hdr/conference_test.pl?dl=0

https://www.dropbox.com/s/497rm2hgopp5u7h/eavesdrop_test.pl?dl=0

Input?

-- 

*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
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From nzaytsevc at gmail.com  Mon Jul 27 22:02:36 2015
From: nzaytsevc at gmail.com (Nikolay Zaytsev)
Date: Mon, 27 Jul 2015 22:02:36 +0400
Subject: [Freeswitch-users] Fwd:
In-Reply-To: <CABo0-L6JD7UfU_s1uZaM1m7RQRsdChHsBKEPOLhb5uckkBMFdw@mail.gmail.com>
References: <CABo0-L7rFYYh0ax1QUsCXgULziWV8PVYr=MScVcxuXUeOYxVHQ@mail.gmail.com>
	<CABo0-L6JD7UfU_s1uZaM1m7RQRsdChHsBKEPOLhb5uckkBMFdw@mail.gmail.com>
Message-ID: <CADeoHuqnvidY9yoWt-7sruviobTYPbK86zm7dJvA6GKkjWEAPg@mail.gmail.com>

Good day to you.
I have a problem with the setting up of the freetdm module.
I have sangoma a102de card.
The system recognizes the card correctly.
Audiocodes mediant 600 with the same trunk configuration works corrctly.
But during the startup freeswitch writes
[DEBUG] ftdm_io.c:5469 Module sangoma_isdn does not support configuration.
and after that
[INFO] ftmod_sangoma_isdn_stack_rcv.c:1098 sng_isdn->s2: Link is down,
dropping transmit frame.
Config files are attached.
Please, help me:)
Best regards,
Nikolay Zaytsev
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From nzaytsevc at gmail.com  Mon Jul 27 22:03:29 2015
From: nzaytsevc at gmail.com (Nikolay Zaytsev)
Date: Mon, 27 Jul 2015 22:03:29 +0400
Subject: [Freeswitch-users] Freetdm configuration
Message-ID: <CADeoHupM02LzV-S+n=+pMe0o1C1tEAOVbE9=EredLg2Xm5qqQw@mail.gmail.com>

Good day to you.
I have a problem with the setting up of the freetdm module.
I have sangoma a102de card.
The system recognizes the card correctly.
Audiocodes mediant 600 with the same trunk configuration works corrctly.
But during the startup freeswitch writes
[DEBUG] ftdm_io.c:5469 Module sangoma_isdn does not support configuration.
and after that
[INFO] ftmod_sangoma_isdn_stack_rcv.c:1098 sng_isdn->s2: Link is down,
dropping transmit frame.
Config files are attached.
Please, help me:)
Best regards,
Nikolay Zaytsev
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From gmaruzz at gmail.com  Mon Jul 27 22:04:50 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 27 Jul 2015 20:04:50 +0200
Subject: [Freeswitch-users] FreeSWITCH Automated Testing
In-Reply-To: <CAEJMVkDGD_mP6kUVToG4W=i6o1c8+MBxKz0dhX1JShhhDGDxbQ@mail.gmail.com>
References: <CAEJMVkDGD_mP6kUVToG4W=i6o1c8+MBxKz0dhX1JShhhDGDxbQ@mail.gmail.com>
Message-ID: <CALXCt0q7LunnyO7TL9eSdO0crrO1-14mJ=xeapU03r1o5w461Q@mail.gmail.com>

Very cool!
QR trick is clever!


On Mon, Jul 27, 2015 at 7:46 PM, Brian West <brian at freeswitch.org> wrote:

> FreeSWITCHers,
>
> So after a bit discussion, a case of red bull, an approach to automating
> functional testing has now solidified.
>
> Audio Path is verified via DTMF in-band, Video Path is verified by using
> QR codes.
>
> Here are two VERY crude examples:
>
> https://www.dropbox.com/s/4fkdcgzo8dt0hdr/conference_test.pl?dl=0
>
> https://www.dropbox.com/s/497rm2hgopp5u7h/eavesdrop_test.pl?dl=0
>
> Input?
>
> --
>
> *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
>
> _________________________________________________________________________
> 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
Cell : +39-347-2665618
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From brian at freeswitch.org  Mon Jul 27 22:21:48 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 27 Jul 2015 13:21:48 -0500
Subject: [Freeswitch-users] FreeSWITCH Automated Testing
In-Reply-To: <CALXCt0q7LunnyO7TL9eSdO0crrO1-14mJ=xeapU03r1o5w461Q@mail.gmail.com>
References: <CAEJMVkDGD_mP6kUVToG4W=i6o1c8+MBxKz0dhX1JShhhDGDxbQ@mail.gmail.com>
	<CALXCt0q7LunnyO7TL9eSdO0crrO1-14mJ=xeapU03r1o5w461Q@mail.gmail.com>
Message-ID: <CAEJMVkD4q3hOPrTvRT3d6Tf2WHHoKPYvE57Mh=RqYBKjsXLggg@mail.gmail.com>

https://twitter.com/briankwest/status/625720099429691393

Has an example of an 8x8 layout with 64 conf callers broadcasting 64 uniq
QR codes, 128 sessions in total. :)

On Mon, Jul 27, 2015 at 1:04 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Very cool!
> QR trick is clever!
>
>
> On Mon, Jul 27, 2015 at 7:46 PM, Brian West <brian at freeswitch.org> wrote:
>
>> FreeSWITCHers,
>>
>> So after a bit discussion, a case of red bull, an approach to automating
>> functional testing has now solidified.
>>
>> Audio Path is verified via DTMF in-band, Video Path is verified by using
>> QR codes.
>>
>> Here are two VERY crude examples:
>>
>> https://www.dropbox.com/s/4fkdcgzo8dt0hdr/conference_test.pl?dl=0
>>
>> https://www.dropbox.com/s/497rm2hgopp5u7h/eavesdrop_test.pl?dl=0
>>
>> Input?
>>
>> --
>>
>> *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
>>
>> _________________________________________________________________________
>> 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
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> 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
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From jaybinks at gmail.com  Tue Jul 28 02:48:44 2015
From: jaybinks at gmail.com (jay binks)
Date: Tue, 28 Jul 2015 08:48:44 +1000
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Automated Testing
In-Reply-To: <CAEJMVkDGD_mP6kUVToG4W=i6o1c8+MBxKz0dhX1JShhhDGDxbQ@mail.gmail.com>
References: <CAEJMVkDGD_mP6kUVToG4W=i6o1c8+MBxKz0dhX1JShhhDGDxbQ@mail.gmail.com>
Message-ID: <CAO47HpdfV3dJ3PoWOPsU8JiYk5f1zGdrDd3nv+4Q1k2ttqxxGA@mail.gmail.com>

Thats AWESOME ! :)
Well done.

On 28 July 2015 at 03:46, Brian West <brian at freeswitch.org> wrote:

> FreeSWITCHers,
>
> So after a bit discussion, a case of red bull, an approach to automating
> functional testing has now solidified.
>
> Audio Path is verified via DTMF in-band, Video Path is verified by using
> QR codes.
>
> Here are two VERY crude examples:
>
> https://www.dropbox.com/s/4fkdcgzo8dt0hdr/conference_test.pl?dl=0
>
> https://www.dropbox.com/s/497rm2hgopp5u7h/eavesdrop_test.pl?dl=0
>
> Input?
>
> --
>
> *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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From brian at freeswitch.org  Tue Jul 28 02:52:34 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 27 Jul 2015 17:52:34 -0500
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Automated Testing
In-Reply-To: <CAO47HpdfV3dJ3PoWOPsU8JiYk5f1zGdrDd3nv+4Q1k2ttqxxGA@mail.gmail.com>
References: <CAEJMVkDGD_mP6kUVToG4W=i6o1c8+MBxKz0dhX1JShhhDGDxbQ@mail.gmail.com>
	<CAO47HpdfV3dJ3PoWOPsU8JiYk5f1zGdrDd3nv+4Q1k2ttqxxGA@mail.gmail.com>
Message-ID: <CAEJMVkD0VO-dmoH6_Qw3mxe9pa6arEqWAa4waz8N7hHzSLxsDQ@mail.gmail.com>

Now we need to start a community project to build on this and drive it
forward. :)

SEE YOU ALL AT CLUECON!

On Mon, Jul 27, 2015 at 5:48 PM, jay binks <jaybinks at gmail.com> wrote:

> Thats AWESOME ! :)
> Well done.
>
> On 28 July 2015 at 03:46, Brian West <brian at freeswitch.org> wrote:
>
>> FreeSWITCHers,
>>
>> So after a bit discussion, a case of red bull, an approach to automating
>> functional testing has now solidified.
>>
>> Audio Path is verified via DTMF in-band, Video Path is verified by using
>> QR codes.
>>
>> Here are two VERY crude examples:
>>
>> https://www.dropbox.com/s/4fkdcgzo8dt0hdr/conference_test.pl?dl=0
>>
>> https://www.dropbox.com/s/497rm2hgopp5u7h/eavesdrop_test.pl?dl=0
>>
>> Input?
>>
>> --
>>
>> *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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Sincerely
>
> Jay
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> 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
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From rajil.s at gmail.com  Tue Jul 28 03:58:05 2015
From: rajil.s at gmail.com (Rajil Saraswat)
Date: Mon, 27 Jul 2015 18:58:05 -0500
Subject: [Freeswitch-users] Should vpn address space be defined as part of
	local network?
Message-ID: <CAFoaQoDpAwGkT7EbSCjC2Fo7Lfo=wP6R9FSjnYrNxcF1ZX0wwQ@mail.gmail.com>

Hello all,

I am trying to get my head around the nat.auto and localnet.auto acls.

I have a VPN server using the 10.8.0.0/24 address space with gateway
on 10.8.0.1. The PBX is on the local lan (172.16.5.0/24) with ip
172.16.5.5.  When freeswitch starts i see it builds the following acls

nat.auto
Created ip list nat.auto default (deny)
Adding 172.16.5.5/255.255.255.0 (deny) to list nat.auto
Adding 10.0.0.0/8 (allow) [] to list nat.auto
Adding 172.16.0.0/12 (allow) [] to list nat.auto

localnet.auto
Created ip list localnet.auto default (deny)
Adding 172.16.5.5/255.255.255.0 (allow) to list localnet.auto


Do i need to move my vpn address space (10.8.0.0/16) from nat.auto to
the localnet.auto so that it not natted? Something like this:

nat.auto
172.16.5.5/255.255.255.0 (deny)
10.0.0.0/8 (allow)
172.16.0.0/12 (allow)
10.8.0.0/16 (deny)

localnet.auto
172.16.5.5/255.255.255.0 (allow)
10.8.0.0/16 (allow)

Thanks


From brian at freeswitch.org  Tue Jul 28 04:01:15 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 27 Jul 2015 19:01:15 -0500
Subject: [Freeswitch-users] Should vpn address space be defined as part
 of local network?
In-Reply-To: <CAFoaQoDpAwGkT7EbSCjC2Fo7Lfo=wP6R9FSjnYrNxcF1ZX0wwQ@mail.gmail.com>
References: <CAFoaQoDpAwGkT7EbSCjC2Fo7Lfo=wP6R9FSjnYrNxcF1ZX0wwQ@mail.gmail.com>
Message-ID: <CAEJMVkBpFd=G2AXBM+NN=xZMDWuvPHBJPe6N9k8x3j+_sXD5MQ@mail.gmail.com>

Or create your OWN ACL that covers you local network space.

On Mon, Jul 27, 2015 at 6:58 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:

> Hello all,
>
> I am trying to get my head around the nat.auto and localnet.auto acls.
>
> I have a VPN server using the 10.8.0.0/24 address space with gateway
> on 10.8.0.1. The PBX is on the local lan (172.16.5.0/24) with ip
> 172.16.5.5.  When freeswitch starts i see it builds the following acls
>
> nat.auto
> Created ip list nat.auto default (deny)
> Adding 172.16.5.5/255.255.255.0 (deny) to list nat.auto
> Adding 10.0.0.0/8 (allow) [] to list nat.auto
> Adding 172.16.0.0/12 (allow) [] to list nat.auto
>
> localnet.auto
> Created ip list localnet.auto default (deny)
> Adding 172.16.5.5/255.255.255.0 (allow) to list localnet.auto
>
>
> Do i need to move my vpn address space (10.8.0.0/16) from nat.auto to
> the localnet.auto so that it not natted? Something like this:
>
> nat.auto
> 172.16.5.5/255.255.255.0 (deny)
> 10.0.0.0/8 (allow)
> 172.16.0.0/12 (allow)
> 10.8.0.0/16 (deny)
>
> localnet.auto
> 172.16.5.5/255.255.255.0 (allow)
> 10.8.0.0/16 (allow)
>
> Thanks
>
> _________________________________________________________________________
> 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
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From rajil.s at gmail.com  Tue Jul 28 04:21:55 2015
From: rajil.s at gmail.com (Rajil Saraswat)
Date: Mon, 27 Jul 2015 19:21:55 -0500
Subject: [Freeswitch-users] Should vpn address space be defined as part
 of local network?
In-Reply-To: <CAEJMVkBpFd=G2AXBM+NN=xZMDWuvPHBJPe6N9k8x3j+_sXD5MQ@mail.gmail.com>
References: <CAFoaQoDpAwGkT7EbSCjC2Fo7Lfo=wP6R9FSjnYrNxcF1ZX0wwQ@mail.gmail.com>
	<CAEJMVkBpFd=G2AXBM+NN=xZMDWuvPHBJPe6N9k8x3j+_sXD5MQ@mail.gmail.com>
Message-ID: <CAFoaQoD3SkUa+B-mr8+rnvt-zf6_J_Txi2AuCev+stndAPDhkQ@mail.gmail.com>

On 27 July 2015 at 19:01, Brian West <brian at freeswitch.org> wrote:

> Or create your OWN ACL that covers you local network space.
>
> On Mon, Jul 27, 2015 at 6:58 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:
>
>> Hello all,
>>
>> I am trying to get my head around the nat.auto and localnet.auto acls.
>>
>> I have a VPN server using the 10.8.0.0/24 address space with gateway
>> on 10.8.0.1. The PBX is on the local lan (172.16.5.0/24) with ip
>> 172.16.5.5.  When freeswitch starts i see it builds the following acls
>>
>> nat.auto
>> Created ip list nat.auto default (deny)
>> Adding 172.16.5.5/255.255.255.0 (deny) to list nat.auto
>> Adding 10.0.0.0/8 (allow) [] to list nat.auto
>> Adding 172.16.0.0/12 (allow) [] to list nat.auto
>>
>> localnet.auto
>> Created ip list localnet.auto default (deny)
>> Adding 172.16.5.5/255.255.255.0 (allow) to list localnet.auto
>>
>>
>> Do i need to move my vpn address space (10.8.0.0/16) from nat.auto to
>> the localnet.auto so that it not natted? Something like this:
>>
>> nat.auto
>> 172.16.5.5/255.255.255.0 (deny)
>> 10.0.0.0/8 (allow)
>> 172.16.0.0/12 (allow)
>> 10.8.0.0/16 (deny)
>>
>> localnet.auto
>> 172.16.5.5/255.255.255.0 (allow)
>> 10.8.0.0/16 (allow)
>>
>> Thanks
>>
>> _________________________________________________________________________
>> 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
>>
>
>
>
Is the local network space only (for the purpose of NAT) defined as the LAN
on which PBX is running or should it include all the class C address space
which it is connected to via VPN?

Thanks
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From brian at freeswitch.org  Tue Jul 28 04:46:24 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 27 Jul 2015 19:46:24 -0500
Subject: [Freeswitch-users] Should vpn address space be defined as part
 of local network?
In-Reply-To: <CAFoaQoD3SkUa+B-mr8+rnvt-zf6_J_Txi2AuCev+stndAPDhkQ@mail.gmail.com>
References: <CAFoaQoDpAwGkT7EbSCjC2Fo7Lfo=wP6R9FSjnYrNxcF1ZX0wwQ@mail.gmail.com>
	<CAEJMVkBpFd=G2AXBM+NN=xZMDWuvPHBJPe6N9k8x3j+_sXD5MQ@mail.gmail.com>
	<CAFoaQoD3SkUa+B-mr8+rnvt-zf6_J_Txi2AuCev+stndAPDhkQ@mail.gmail.com>
Message-ID: <CAEJMVkBCZv1EbdxVVf5D5ShMfBRCx-9JcMa4BB7V9_2y8MauHA@mail.gmail.com>

If there is no nat between you and that network consider it local.

On Mon, Jul 27, 2015 at 7:21 PM, Rajil Saraswat <rajil.s at gmail.com> wrote:

> On 27 July 2015 at 19:01, Brian West <brian at freeswitch.org> wrote:
>
>> Or create your OWN ACL that covers you local network space.
>>
>> On Mon, Jul 27, 2015 at 6:58 PM, Rajil Saraswat <rajil.s at gmail.com>
>> wrote:
>>
>>> Hello all,
>>>
>>> I am trying to get my head around the nat.auto and localnet.auto acls.
>>>
>>> I have a VPN server using the 10.8.0.0/24 address space with gateway
>>> on 10.8.0.1. The PBX is on the local lan (172.16.5.0/24) with ip
>>> 172.16.5.5.  When freeswitch starts i see it builds the following acls
>>>
>>> nat.auto
>>> Created ip list nat.auto default (deny)
>>> Adding 172.16.5.5/255.255.255.0 (deny) to list nat.auto
>>> Adding 10.0.0.0/8 (allow) [] to list nat.auto
>>> Adding 172.16.0.0/12 (allow) [] to list nat.auto
>>>
>>> localnet.auto
>>> Created ip list localnet.auto default (deny)
>>> Adding 172.16.5.5/255.255.255.0 (allow) to list localnet.auto
>>>
>>>
>>> Do i need to move my vpn address space (10.8.0.0/16) from nat.auto to
>>> the localnet.auto so that it not natted? Something like this:
>>>
>>> nat.auto
>>> 172.16.5.5/255.255.255.0 (deny)
>>> 10.0.0.0/8 (allow)
>>> 172.16.0.0/12 (allow)
>>> 10.8.0.0/16 (deny)
>>>
>>> localnet.auto
>>> 172.16.5.5/255.255.255.0 (allow)
>>> 10.8.0.0/16 (allow)
>>>
>>> Thanks
>>>
>>> _________________________________________________________________________
>>> 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
>>>
>>
>>
>>
> Is the local network space only (for the purpose of NAT) defined as the
> LAN on which PBX is running or should it include all the class C address
> space which it is connected to via VPN?
>
> Thanks
>
>
>
> _________________________________________________________________________
> 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
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From dmitry.muntean at gmail.com  Tue Jul 28 04:59:52 2015
From: dmitry.muntean at gmail.com (Dmitri Muntean)
Date: Tue, 28 Jul 2015 10:59:52 +1000
Subject: [Freeswitch-users] Incorrect Caller-Context in the events
Message-ID: <CAEKHyetF+Kjo7cfthubAzmmx67dS2AGZP7YMjYyyJ9oQ_X3QDw@mail.gmail.com>

Hey guys,

I'm running freeswitch 1.4.20 and 1.4.13 with the number of conferences
setup on them and have 3 profiles and contexts defined, none of those are
called "default". I read the events via inbound event socket, and most of
the events happening inside those conferences have correct Caller-Context
set to the profile that the call came through. But sometimes (very rarely)
the Caller-Context is set to "default" - I was wondering if anyone seen
such behaviour.

I don't have "default" profile or context defined anywhere in my
configuration.

The events I am reading off freeswitch are Event-Name: DTMF and Event-Name:
CUSTOM with Event-Subclass: conference::maintenance. When the
Caller-Context is "default" it appears so in both type of events.

Thanks
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From david.fu at oocl.com  Tue Jul 28 06:00:46 2015
From: david.fu at oocl.com (david.fu at oocl.com)
Date: Tue, 28 Jul 2015 10:00:46 +0800
Subject: [Freeswitch-users] Implementing <Form> Tag of VXML in
 Freeswtich using Javascript
In-Reply-To: <CAEJMVkAy=jUh2JHM9enqH4wkiUO4tyxG6Fy76zJ7VdiqW=1BCA@mail.gmail.com>
References: <61E1B4F8DE845A4F986F448F6D50A38504BE869D67@E2K7CCR03.corp.oocl.com>
	<CAEJMVkAy=jUh2JHM9enqH4wkiUO4tyxG6Fy76zJ7VdiqW=1BCA@mail.gmail.com>
Message-ID: <61E1B4F8DE845A4F986F448F6D50A38504BE869E7C@E2K7CCR03.corp.oocl.com>

Hi Brian,

Thanks for your prompt reply so much.  Actually, we have an existing VXML application. The voice browser sends HTTP request to the Application server, which returns VXML to the voice browser in IVR server. Now, we would like to migrate it to Freeswitch.  As we used <FORM> and <SUBMIT> tag in VXML, we are searching how to implement this in Freeswitch to send request to Application Server, and then return XML or Javascript instead of VXML to Freeswtich. Would you please give us some hints ? Thanks.

Best Regards,
David

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Monday, July 27, 2015 9:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Implementing <Form> Tag of VXML in Freeswtich using Javascript

There is no support for VXML in FreeSWITCH, We have all the parts and I'm sure it could probably be implemented, but its never been a high priority item for us as nobody asks for it.

On Mon, Jul 27, 2015 at 2:25 AM, <david.fu at oocl.com<mailto:david.fu at oocl.com>> wrote:
Dear Freeswitch experts,

I would like to implement something like <Form> and <submit> tag of VXML in Freeswtich using Javascript. The purpose is to sending request back to the application server from Freeswtich.  However,  I couldn?t find the related API(s) in the Freeswitch official web site. Would you please help advise ? Thank you so much.

Yours faithfully,
David




IMPORTANT NOTICE
Email from OOCL is confidential and may be legally privileged.  If it is not
intended for you, please delete it immediately unread.  The internet
cannot guarantee that this communication is free of viruses, interception
or interference and anyone who communicates with us by email is taken
to accept the risks in doing so.  Without limitation, OOCL and its affiliates
accept no liability whatsoever and howsoever arising in connection with
the use of this email.  Under no circumstances shall this email constitute
a binding agreement to carry or for provision of carriage services by OOCL,
which is subject to the availability of carrier's equipment and vessels and
the terms and conditions of OOCL's standard bill of lading which is also
available at http://www.oocl.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
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

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


IMPORTANT NOTICE
Email from OOCL is confidential and may be legally privileged.  If it is not
intended for you, please delete it immediately unread.  The internet
cannot guarantee that this communication is free of viruses, interception
or interference and anyone who communicates with us by email is taken
to accept the risks in doing so.  Without limitation, OOCL and its affiliates
accept no liability whatsoever and howsoever arising in connection with
the use of this email.  Under no circumstances shall this email constitute
a binding agreement to carry or for provision of carriage services by OOCL,
which is subject to the availability of carrier's equipment and vessels and
the terms and conditions of OOCL's standard bill of lading which is also
available at http://www.oocl.com.
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From brian at freeswitch.org  Tue Jul 28 06:15:57 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 27 Jul 2015 21:15:57 -0500
Subject: [Freeswitch-users] Incorrect Caller-Context in the events
In-Reply-To: <CAEKHyetF+Kjo7cfthubAzmmx67dS2AGZP7YMjYyyJ9oQ_X3QDw@mail.gmail.com>
References: <CAEKHyetF+Kjo7cfthubAzmmx67dS2AGZP7YMjYyyJ9oQ_X3QDw@mail.gmail.com>
Message-ID: <CAEJMVkDukjcpD-=aTbVAirfN3sYk_ngPzDVr6-N1SWfgwh2PTg@mail.gmail.com>

load xml_cdr and look at the XML once he call is complete, it'll give you
the full details of what happened.

On Mon, Jul 27, 2015 at 7:59 PM, Dmitri Muntean <dmitry.muntean at gmail.com>
wrote:

> Hey guys,
>
> I'm running freeswitch 1.4.20 and 1.4.13 with the number of conferences
> setup on them and have 3 profiles and contexts defined, none of those are
> called "default". I read the events via inbound event socket, and most of
> the events happening inside those conferences have correct Caller-Context
> set to the profile that the call came through. But sometimes (very rarely)
> the Caller-Context is set to "default" - I was wondering if anyone seen
> such behaviour.
>
> I don't have "default" profile or context defined anywhere in my
> configuration.
>
> The events I am reading off freeswitch are Event-Name: DTMF and
> Event-Name: CUSTOM with Event-Subclass: conference::maintenance. When the
> Caller-Context is "default" it appears so in both type of events.
>
> Thanks
>
>
>
> _________________________________________________________________________
> 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
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From cmrienzo at gmail.com  Tue Jul 28 06:59:36 2015
From: cmrienzo at gmail.com (cmrienzo at gmail.com)
Date: Mon, 27 Jul 2015 22:59:36 -0400
Subject: [Freeswitch-users] Implementing <Form> Tag of VXML in
	Freeswtich using Javascript
In-Reply-To: <61E1B4F8DE845A4F986F448F6D50A38504BE869E7C@E2K7CCR03.corp.oocl.com>
References: <61E1B4F8DE845A4F986F448F6D50A38504BE869D67@E2K7CCR03.corp.oocl.com>
	<CAEJMVkAy=jUh2JHM9enqH4wkiUO4tyxG6Fy76zJ7VdiqW=1BCA@mail.gmail.com>
	<61E1B4F8DE845A4F986F448F6D50A38504BE869E7C@E2K7CCR03.corp.oocl.com>
Message-ID: <7516D1CA-D0B1-4A31-9088-604BD48FF276@gmail.com>

If you have the voice browser source code you could try and port it to control FS over event socket.  Otherwise, you're in for a lot of work to turn FS into a voice browser from scratch. 

Chris



> On Jul 27, 2015, at 22:00, <david.fu at oocl.com> <david.fu at oocl.com> wrote:
> 
> Hi Brian,
>  
> Thanks for your prompt reply so much.  Actually, we have an existing VXML application. The voice browser sends HTTP request to the Application server, which returns VXML to the voice browser in IVR server. Now, we would like to migrate it to Freeswitch.  As we used <FORM> and <SUBMIT> tag in VXML, we are searching how to implement this in Freeswitch to send request to Application Server, and then return XML or Javascript instead of VXML to Freeswtich. Would you please give us some hints ? Thanks.
>  
> Best Regards,
> David
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
> Sent: Monday, July 27, 2015 9:53 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Implementing <Form> Tag of VXML in Freeswtich using Javascript
>  
> There is no support for VXML in FreeSWITCH, We have all the parts and I'm sure it could probably be implemented, but its never been a high priority item for us as nobody asks for it.
>  
> On Mon, Jul 27, 2015 at 2:25 AM, <david.fu at oocl.com> wrote:
> Dear Freeswitch experts,
>  
> I would like to implement something like <Form> and <submit> tag of VXML in Freeswtich using Javascript. The purpose is to sending request back to the application server from Freeswtich.  However,  I couldn?t find the related API(s) in the Freeswitch official web site. Would you please help advise ? Thank you so much.
>  
> Yours faithfully,
> David
>  
>  
> 
> 
> IMPORTANT NOTICE
> Email from OOCL is confidential and may be legally privileged.  If it is not
> intended for you, please delete it immediately unread.  The internet
> cannot guarantee that this communication is free of viruses, interception
> or interference and anyone who communicates with us by email is taken
> to accept the risks in doing so.  Without limitation, OOCL and its affiliates
> accept no liability whatsoever and howsoever arising in connection with
> the use of this email.  Under no circumstances shall this email constitute
> a binding agreement to carry or for provision of carriage services by OOCL,
> which is subject to the availability of carrier's equipment and vessels and
> the terms and conditions of OOCL's standard bill of lading which is also
> available at http://www.oocl.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
> brian at freeswitch.org
> 
> 
> 
> Twitter: @FreeSWITCH , @briankwest
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
> 
> Got Bugs? Report them here! | Reddit: /r/freeswitch
> 
> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
> 
> 
> 
> IMPORTANT NOTICE
> Email from OOCL is confidential and may be legally privileged.  If it is not
> intended for you, please delete it immediately unread.  The internet
> cannot guarantee that this communication is free of viruses, interception
> or interference and anyone who communicates with us by email is taken
> to accept the risks in doing so.  Without limitation, OOCL and its affiliates
> accept no liability whatsoever and howsoever arising in connection with
> the use of this email.  Under no circumstances shall this email constitute
> a binding agreement to carry or for provision of carriage services by OOCL,
> which is subject to the availability of carrier's equipment and vessels and
> the terms and conditions of OOCL's standard bill of lading which is also
> available at http://www.oocl.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
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From david.fu at oocl.com  Tue Jul 28 07:23:08 2015
From: david.fu at oocl.com (david.fu at oocl.com)
Date: Tue, 28 Jul 2015 11:23:08 +0800
Subject: [Freeswitch-users] Implementing <Form> Tag of VXML
	in	Freeswtich using Javascript
In-Reply-To: <7516D1CA-D0B1-4A31-9088-604BD48FF276@gmail.com>
References: <61E1B4F8DE845A4F986F448F6D50A38504BE869D67@E2K7CCR03.corp.oocl.com>
	<CAEJMVkAy=jUh2JHM9enqH4wkiUO4tyxG6Fy76zJ7VdiqW=1BCA@mail.gmail.com>
	<61E1B4F8DE845A4F986F448F6D50A38504BE869E7C@E2K7CCR03.corp.oocl.com>
	<7516D1CA-D0B1-4A31-9088-604BD48FF276@gmail.com>
Message-ID: <61E1B4F8DE845A4F986F448F6D50A38504BE869EC2@E2K7CCR03.corp.oocl.com>

Hi Chris,

Thanks Chris for the advice. You ?re right that we may need a lot of effort to turn FS into a voice Brower. Hence,  I am thinking how to turn the returned VXML in our existing application server into something that FS can interpret. For example, rewrite the code in our existing application to return XML/Javascript instead of VXML to FS.   However, I don?t know how to implement <SUBMIT> in Javascript or XML according to the Javascript reference site at https://wiki.freeswitch.org/wiki/Category:Javascript.  Would you please give me some hints ?

Existing:

IVR Voice Browser(Send HTTP request) <-- --> Application server(return VXML to the voice browser of IVR). The VXML mainly stores callflow of each hotline.

Future:

FS(Send HTTP request) <-- --> Application server(return Javascript/XML to the dialplan of FS)


Best Regards,
David

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of cmrienzo at gmail.com
Sent: Tuesday, July 28, 2015 11:00 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Implementing <Form> Tag of VXML in Freeswtich using Javascript

If you have the voice browser source code you could try and port it to control FS over event socket.  Otherwise, you're in for a lot of work to turn FS into a voice browser from scratch.

Chris



On Jul 27, 2015, at 22:00, <david.fu at oocl.com<mailto:david.fu at oocl.com>> <david.fu at oocl.com<mailto:david.fu at oocl.com>> wrote:
Hi Brian,

Thanks for your prompt reply so much.  Actually, we have an existing VXML application. The voice browser sends HTTP request to the Application server, which returns VXML to the voice browser in IVR server. Now, we would like to migrate it to Freeswitch.  As we used <FORM> and <SUBMIT> tag in VXML, we are searching how to implement this in Freeswitch to send request to Application Server, and then return XML or Javascript instead of VXML to Freeswtich. Would you please give us some hints ? Thanks.

Best Regards,
David

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Monday, July 27, 2015 9:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Implementing <Form> Tag of VXML in Freeswtich using Javascript

There is no support for VXML in FreeSWITCH, We have all the parts and I'm sure it could probably be implemented, but its never been a high priority item for us as nobody asks for it.

On Mon, Jul 27, 2015 at 2:25 AM, <david.fu at oocl.com<mailto:david.fu at oocl.com>> wrote:
Dear Freeswitch experts,

I would like to implement something like <Form> and <submit> tag of VXML in Freeswtich using Javascript. The purpose is to sending request back to the application server from Freeswtich.  However,  I couldn?t find the related API(s) in the Freeswitch official web site. Would you please help advise ? Thank you so much.

Yours faithfully,
David




IMPORTANT NOTICE
Email from OOCL is confidential and may be legally privileged.  If it is not
intended for you, please delete it immediately unread.  The internet
cannot guarantee that this communication is free of viruses, interception
or interference and anyone who communicates with us by email is taken
to accept the risks in doing so.  Without limitation, OOCL and its affiliates
accept no liability whatsoever and howsoever arising in connection with
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a binding agreement to carry or for provision of carriage services by OOCL,
which is subject to the availability of carrier's equipment and vessels and
the terms and conditions of OOCL's standard bill of lading which is also
available at http://www.oocl.com.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

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


IMPORTANT NOTICE
Email from OOCL is confidential and may be legally privileged.  If it is not
intended for you, please delete it immediately unread.  The internet
cannot guarantee that this communication is free of viruses, interception
or interference and anyone who communicates with us by email is taken
to accept the risks in doing so.  Without limitation, OOCL and its affiliates
accept no liability whatsoever and howsoever arising in connection with
the use of this email.  Under no circumstances shall this email constitute
a binding agreement to carry or for provision of carriage services by OOCL,
which is subject to the availability of carrier's equipment and vessels and
the terms and conditions of OOCL's standard bill of lading which is also
available at http://www.oocl.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
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http://www.cluecon.com

FreeSWITCH-users mailing list
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http://www.freeswitch.org


IMPORTANT NOTICE
Email from OOCL is confidential and may be legally privileged.  If it is not
intended for you, please delete it immediately unread.  The internet
cannot guarantee that this communication is free of viruses, interception
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From cmrienzo at gmail.com  Tue Jul 28 07:47:31 2015
From: cmrienzo at gmail.com (cmrienzo at gmail.com)
Date: Mon, 27 Jul 2015 23:47:31 -0400
Subject: [Freeswitch-users] Implementing <Form> Tag of VXML
	in	Freeswtich using Javascript
In-Reply-To: <61E1B4F8DE845A4F986F448F6D50A38504BE869EC2@E2K7CCR03.corp.oocl.com>
References: <61E1B4F8DE845A4F986F448F6D50A38504BE869D67@E2K7CCR03.corp.oocl.com>
	<CAEJMVkAy=jUh2JHM9enqH4wkiUO4tyxG6Fy76zJ7VdiqW=1BCA@mail.gmail.com>
	<61E1B4F8DE845A4F986F448F6D50A38504BE869E7C@E2K7CCR03.corp.oocl.com>
	<7516D1CA-D0B1-4A31-9088-604BD48FF276@gmail.com>
	<61E1B4F8DE845A4F986F448F6D50A38504BE869EC2@E2K7CCR03.corp.oocl.com>
Message-ID: <37ED4000-EC4C-4449-AAFA-435C00C5539B@gmail.com>

Look here: https://wiki.freeswitch.org/wiki/Run

Or check mod_curl

Chris


> On Jul 27, 2015, at 23:23, <david.fu at oocl.com> <david.fu at oocl.com> wrote:
> 
> Hi Chris,
>  
> Thanks Chris for the advice. You ?re right that we may need a lot of effort to turn FS into a voice Brower. Hence,  I am thinking how to turn the returned VXML in our existing application server into something that FS can interpret. For example, rewrite the code in our existing application to return XML/Javascript instead of VXML to FS.   However, I don?t know how to implement <SUBMIT> in Javascript or XML according to the Javascript reference site at https://wiki.freeswitch.org/wiki/Category:Javascript.  Would you please give me some hints ?
>  
> Existing:
>  
> IVR Voice Browser(Send HTTP request) <-- --> Application server(return VXML to the voice browser of IVR). The VXML mainly stores callflow of each hotline.
>  
> Future:
>  
> FS(Send HTTP request) <-- --> Application server(return Javascript/XML to the dialplan of FS)
>  
>  
> Best Regards,
> David
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of cmrienzo at gmail.com
> Sent: Tuesday, July 28, 2015 11:00 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Implementing <Form> Tag of VXML in Freeswtich using Javascript
>  
> If you have the voice browser source code you could try and port it to control FS over event socket.  Otherwise, you're in for a lot of work to turn FS into a voice browser from scratch. 
>  
> Chris
>  
>  
> 
> On Jul 27, 2015, at 22:00, <david.fu at oocl.com> <david.fu at oocl.com> wrote:
> 
> Hi Brian,
>  
> Thanks for your prompt reply so much.  Actually, we have an existing VXML application. The voice browser sends HTTP request to the Application server, which returns VXML to the voice browser in IVR server. Now, we would like to migrate it to Freeswitch.  As we used <FORM> and <SUBMIT> tag in VXML, we are searching how to implement this in Freeswitch to send request to Application Server, and then return XML or Javascript instead of VXML to Freeswtich. Would you please give us some hints ? Thanks.
>  
> Best Regards,
> David
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
> Sent: Monday, July 27, 2015 9:53 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Implementing <Form> Tag of VXML in Freeswtich using Javascript
>  
> There is no support for VXML in FreeSWITCH, We have all the parts and I'm sure it could probably be implemented, but its never been a high priority item for us as nobody asks for it.
>  
> On Mon, Jul 27, 2015 at 2:25 AM, <david.fu at oocl.com> wrote:
> Dear Freeswitch experts,
>  
> I would like to implement something like <Form> and <submit> tag of VXML in Freeswtich using Javascript. The purpose is to sending request back to the application server from Freeswtich.  However,  I couldn?t find the related API(s) in the Freeswitch official web site. Would you please help advise ? Thank you so much.
>  
> Yours faithfully,
> David
>  
>  
> 
> 
> IMPORTANT NOTICE
> Email from OOCL is confidential and may be legally privileged.  If it is not
> intended for you, please delete it immediately unread.  The internet
> cannot guarantee that this communication is free of viruses, interception
> or interference and anyone who communicates with us by email is taken
> to accept the risks in doing so.  Without limitation, OOCL and its affiliates
> accept no liability whatsoever and howsoever arising in connection with
> the use of this email.  Under no circumstances shall this email constitute
> a binding agreement to carry or for provision of carriage services by OOCL,
> which is subject to the availability of carrier's equipment and vessels and
> the terms and conditions of OOCL's standard bill of lading which is also
> available at http://www.oocl.com.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
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> 
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> 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! | Reddit: /r/freeswitch
> 
> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
> 
> 
> 
> IMPORTANT NOTICE
> Email from OOCL is confidential and may be legally privileged.  If it is not
> intended for you, please delete it immediately unread.  The internet
> cannot guarantee that this communication is free of viruses, interception
> or interference and anyone who communicates with us by email is taken
> to accept the risks in doing so.  Without limitation, OOCL and its affiliates
> accept no liability whatsoever and howsoever arising in connection with
> the use of this email.  Under no circumstances shall this email constitute
> a binding agreement to carry or for provision of carriage services by OOCL,
> which is subject to the availability of carrier's equipment and vessels and
> the terms and conditions of OOCL's standard bill of lading which is also
> available at http://www.oocl.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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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> IMPORTANT NOTICE
> Email from OOCL is confidential and may be legally privileged.  If it is not
> intended for you, please delete it immediately unread.  The internet
> cannot guarantee that this communication is free of viruses, interception
> or interference and anyone who communicates with us by email is taken
> to accept the risks in doing so.  Without limitation, OOCL and its affiliates
> accept no liability whatsoever and howsoever arising in connection with
> the use of this email.  Under no circumstances shall this email constitute
> a binding agreement to carry or for provision of carriage services by OOCL,
> which is subject to the availability of carrier's equipment and vessels and
> the terms and conditions of OOCL's standard bill of lading which is also
> available at http://www.oocl.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 sendmeallyouroffers at googlemail.com  Tue Jul 28 11:19:22 2015
From: sendmeallyouroffers at googlemail.com (Berthold Karl)
Date: Tue, 28 Jul 2015 09:19:22 +0200
Subject: [Freeswitch-users] freeswitch never seems to call sendmail when
 trying to email voicemails
In-Reply-To: <C18BE15B-86F3-40C1-ABED-CA328565C0C1@mgtech.com>
References: <54E6B9A1.20708@dickson.st>
	<CAHtxdDd8CRENTyNoi7FOn3yuy+fyCshEiknfhvhZcg36q=JFuA@mail.gmail.com>
	<54EA5B78.3010504@dickson.st>
	<CAHtxdDcD0fqTWR9mA8=-eT9rm2a1M1p3Qab7OVXCfH=wYnHvvw@mail.gmail.com>
	<54EBE76A.2040709@dickson.st>
	<CAHtxdDfsm2pd8HDM-WwU8HKJT5AQj3isqjWF4aWVbXkRrtB0zg@mail.gmail.com>
	<CAEJMVkBY-3nOfXijxhED_byC_K5cNnS8LVW3iRjGmOFserDmjw@mail.gmail.com>
	<CAN6oSBuxXbf=uecMk1UO=ZP9xxpvoXA65Ox4pyZVmPxA-BNwnQ@mail.gmail.com>
	<CA+Xtiefu7T+LBFD11k38wPG-QCPcrWJuaAaLGTOmS_RTmk8m7A@mail.gmail.com>
	<29646517.20150727173302@seznam.cz>
	<C18BE15B-86F3-40C1-ABED-CA328565C0C1@mgtech.com>
Message-ID: <CAN6oSBt4VqjViDaXdkund+Ch9W=5JF__n55sOx5PVXYdAn3Zgw@mail.gmail.com>

Thanks for the replies!

I've tested sendmail, it works for the user which is running freeswitch.

I'am running FS 1.4.8 under Debian.

I want to use the FS internal VM-to-E-Mail because all informations about
To-, From- etc are configured in the user.xml. If I'am running my own
script, I have to setup everything in a second configuration or have to
parse the user.xml a second time..

Thx Dennis for the Sample script,

Any ideas?
I've tested sendmail with few arguments... everytime its ends with a
segfault... It seems FS isn't creating the Mail under /tmp/. /tmp is World
writable/readable.

Regards



2015-07-27 19:01 GMT+02:00 Mario <mario_fs at mgtech.com>:

> Although this is in the Mac OS X page I wrote this up which may help:
> https://freeswitch.org/confluence/display/FREESWITCH/Installation+and+Setup+on+OS+X#InstallationandSetuponOSX-EmailVoicemailtoaniPhone
>
> To test your email you can do this from a command line:
> printf "Subject: TestnHello" | sendmail -f you at domain.com you at domain.com
> Mareio G
>
> On Jul 27, 2015, at 8:33 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
>
>  Hi,
>
> Why do not use some simple script after the ends session?
>
> For example simple bash script send_voicemail.sh
>
> #!/bin/sh
> cd /usr/local/freeswitch/recordings/voicemail/
> f=`find -name \*.wav`
> for file in $f
> do
> echo "Processing ${file}"
> CURRENT=$(date +%d.%m.%y_%H:%M:%S)
> datetime=$CURRENT
> echo $datetime
> sendemail -f voicemail at mail.com -t somemail at mail.com -m "Voicemail" -u
> "Voicemail od $datetime" -a ${file} -s smtp.mail.com -xu user -xp pass
> echo "BackUP ${file}"
> mv ${file} /usr/local/freeswitch/recordings/backup_voicemail/
> done
>
>
> And in dialplan start like this
> <action application="export" data="api_hangup_hook=system
> /usr/local/freeswitch/scripts/send_voicemail.sh"/>
>
>
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 pond?l? 27. ?ervence 2015, 17:11:10, napsal
> jste: *
>
>  Why can.t you create your own script and use it to send vm over email?
> 27 ???? 2015 ?. 18:32 ???????????? "Berthold Karl" <
> sendmeallyouroffers at googlemail.com> ???????:
> Hi,
>
> my FS is calling the sendmail-command, but sendmail ends with an segfault.
> It seems like there is no email under /tmp/. Did your sendmail also ends in
> a segfault?
>
> 2015-02-24 16:51 GMT+01:00 Brian West <brian at freeswitch.org>:
> When you file the JIRA please attach your voicemail.conf.xml.
>
> On Tue, Feb 24, 2015 at 8:21 AM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
> If you want to FS correctly processed parameters vm-mailfrom and
> email-from, write a request to https://freeswitch.org/jira/
>
> On Tue, Feb 24, 2015 at 5:52 AM, Jason Lewis <jason at dickson.st> wrote:
> Thanks Sergey,
>
> I had configured the domain variable to the fqdn of the machine. I
> eventually got it working though, I was missing two key lines from my user
> config:
>      <param name="vm-email-all-messages" value="true"/>
>      <param name="vm-attach-file" value="true" />
>
> I'm not sure how I managed to miss those but anyway, that seems to have
> resolved things.
>
> It seems as though vm-mailfrom is still being ignored though. Currently I
> have it set to:
>
>       <param name="vm-mailfrom" value="vm at freeswitch.xyz.com.au"
> <vm at freeswitch.xyz.com.au>/>
>
> but voicemails get delivered from:
>
> 1001 at freeswitch.xyz.com.au
>
> Is this worth investigating further?
>
> Jason
>
> Sergey Safarov wrote on 23/02/2015 5:05 PM:
> Try configure "domain" variable in vars.xml
>
> <X-PRE-PROCESS cmd="set" data="domain=you_domain_name"/>
>
> After it verify that user registered with domain name
>
> freeswitch at internal> sofia status profile internal reg
>
> Registrations:
>
> =================================================================================================
> Call-ID:    1B26-2327-466848134BEBC9719CDE-002 at SipHost
> User:       1201 at you_domain_name
> Contact:    "1201" <
> sip:1201 at 10.21.18.22:5060;fs_nat=yes;fs_path=sip%3A1201%4010.21.18.22%3A5060
> >
> Agent:      204 12-3868-2416-0.10.56.1-DS
> Status:     Registered(UDP-NAT)(unknown) EXP(2015-02-23 06:05:22)
> EXPSECS(139)
> Ping-Status:Reachable
> Host:       fs1.you_domain_name
> IP:         10.21.18.22
> Port:       5060
> Auth-User:  1201
> Auth-Realm: you_domain_name
> MWI-Account:1201 at you_domain_name
>
> Sergey
>
>
> On Mon, Feb 23, 2015 at 1:43 AM, Jason Lewis <jason at dickson.st
> > wrote:
>
> So I've managed to see some output from the sendmail program in the FS
> logs. It appears that my fs instance isn't correctly setting its domain?
>
> the FS box has a fqdn, and I also set the domain parameter in the vars.xml
> file, but still the voicemail is sent with a from address of an IP address.
>
> Any ideas?
>
> Net::SMTP>>> Net::SMTP(2.33)
> Net::SMTP>>>   Net::Cmd(2.30)
> Net::SMTP>>>     Exporter(5.71)
> Net::SMTP>>>   IO::Socket::INET(1.35)
> Net::SMTP>>>     IO::Socket(1.37)
> Net::SMTP>>>       IO::Handle(1.35)
> Net::SMTP=GLOB(0x23f7748)<<< 220 mb.xyz.com.au ESMTP Postfix (Debian/GNU)
> Net::SMTP=GLOB(0x23f7748)>>> EHLO localhost.localdomain
> Net::SMTP=GLOB(0x23f7748)<<< 250-mb.bongalong.st
>
> Net::SMTP=GLOB(0x23f7748)<<< 250-PIPELINING
> Net::SMTP=GLOB(0x23f7748)<<< 250-SIZE 10240000
> Net::SMTP=GLOB(0x23f7748)<<< 250-VRFY
> Net::SMTP=GLOB(0x23f7748)<<< 250-ETRN
> Net::SMTP=GLOB(0x23f7748)<<< 250-STARTTLS
> Net::SMTP=GLOB(0x23f7748)<<< 250-ENHANCEDSTATUSCODES
> Net::SMTP=GLOB(0x23f7748)<<< 250-8BITMIME
> Net::SMTP=GLOB(0x23f7748)<<< 250 DSN
> Net::SMTP=GLOB(0x23f7748)>>> MAIL FROM:<1002 at 192.168.1.3>
> <1002 at 192.168.1.3>
> Net::SMTP=GLOB(0x23f7748)<<< 501 5.1.7 Bad sender address syntax
> Net::SMTP=GLOB(0x23f7748)>>> RCPT TO:<jason at xyz.com.au> <jason at xyz.com.au>
> Net::SMTP=GLOB(0x23f7748)<<< 503 5.5.1 Error: need MAIL command
> Net::SMTP=GLOB(0x23f7748)>>> DATA
> Net::SMTP=GLOB(0x23f7748)<<< 503 5.5.1 Error: need RCPT command
> Net::SMTP=GLOB(0x23f7748)>>> QUIT
> Net::SMTP=GLOB(0x23f7748)<<< 221 2.0.0 Bye
> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1633 Session 3
> (loopback/voicemail-a) Ended
> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1637 Close
> Channel loopback/voicemail-a [CS_DESTROY]
> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1633 Session 4
> (loopback/voicemail-b) Ended
> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1637 Close
> Channel loopback/voicemail-b [CS_DESTROY]
>
>
> On 20/02/2015 7:10 pm, Sergey Safarov wrote:
> You mailer is not understand "mailer-app-args" has been configured in
> "switch.conf.xml"
> Remove extra arg or add required
>
> ??, 20 ????. 2015, 7:49, Jason Lewis <jason at dickson.st>:
> Hi,
>
> I've been trying to make freeswitch email voicemails but as far as I can
> tell, it never even calls sendmail.
>
> I have setting mailer-app to "sendmail" and "/usr/sbin/sendmail" to no
> avail. I can successfully send an email from the commandline using
> sendmail. (sendmail in this case is provided by postfix)
>
> I see no emails in the postfix mail logs when I leave a voicemail message.
>
> I also tried creating a shell just to see if it even gets called from
> fs, but it does not get called when a voicemail is deposited:
> #!/bin/bash
> echo $(date --rfc-3339=ns): $* >> /tmp/freeswitchsendmail.log
>
> After every change, I have run reloadxml and reload mod_voicemail. I
> have also tried restarting freeswitch.
>
> I am running the debian packages of FreeSWITCH Version 1.4.15-1~64bit
> (-1 64bit)
>
> my configuration is based on the vanilla configuration with only very
> minor changes.
>
> I'm at a loss as to how to debug further, but I'm pretty sure the
> mailer-app is never called. Is there some setting I'm missing or
> something obvious I'm not doing?
>
>
> My config:
> 1001.xml:
> <include>
>  <user id="1001">
>    <params>
>      <param name="password" value="$${default_password}"/>
>      <param name="vm-password" value="1001"/>
>      <param name="vm-mailto" value="jason at xyz.com.au"/>
>      <param name="vm-mailfrom" value="vm at freeswitch.xyz.com.au"/>
>      <!-- <param name="vm-message-ext" value="wav"/> -->
>    </params>
>    <variables>
>      <variable name="toll_allow" value="domestic,international,local"/>
>      <variable name="accountcode" value="1001"/>
>      <variable name="user_context" value="default"/>
>      <variable name="effective_caller_id_name" value="Extension 1001"/>
>      <variable name="effective_caller_id_number" value="1001"/>
>      <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>      <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>      <variable name="callgroup" value="techsupport"/>
>    </variables>
>  </user>
> </include>
>
> and in switch.conf.xml I have the following set:
>
>    <param name="mailer-app" value="/usr/sbin/sendmail"/>
>    <param name="mailer-app-args" value="-t"/>
>
>
> I made a log at level 7 and put it on the pastebin:
>
> https://pastebin.freeswitch.org/23921
>
>
> Jason Lewis
> http://emacstragic.net
>
>
> _________________________________________________________________________
> 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
>
>
> --
> Jason Lewis
> http://emacstragic.net
>
> _________________________________________________________________________
> 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
>
> --
> Jason Lewis
> http://emacstragic.net
>
> _________________________________________________________________________
> 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 *brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest *http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> 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 ssinyagin at gmail.com  Tue Jul 28 11:40:39 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 28 Jul 2015 09:40:39 +0200
Subject: [Freeswitch-users] FreeSWITCH Automated Testing
In-Reply-To: <CAEJMVkDGD_mP6kUVToG4W=i6o1c8+MBxKz0dhX1JShhhDGDxbQ@mail.gmail.com>
References: <CAEJMVkDGD_mP6kUVToG4W=i6o1c8+MBxKz0dhX1JShhhDGDxbQ@mail.gmail.com>
Message-ID: <CACOYK=jJwYb4B=QtGYXP+C1NrLLLjKAmraH_X3SB0vzb53bGJQ@mail.gmail.com>

here I started a project for a customer. It validates the SIP message flow
for a remote server, sends and receives DTMF, and is intended to be used
for automated testing. FreeSWITCH is used as a dialer and a callee, and the
remote server can also be... a FreeSWITCH server :)
The documentation is not there yet, but it will be finalized during Summer:
https://github.com/voxserv/rring

Also here is an example of using Sevana AQuA software for comparing the
input and output audio.
https://github.com/ssinyagin-freeswitch-jira/FS-7805

Also based on FS-7805 experience, I'm making a Perl script which will
process the histogram produced by "sox stat -freq", so that it can check if
the recorded audio files are identical between each other. It's slower than
AQuA, but is completely free and allows to verify the clock precision on
the host.

I'll be happy if these things also go into your automated test environment
(after I finalize them).





On Mon, Jul 27, 2015 at 7:46 PM, Brian West <brian at freeswitch.org> wrote:

> FreeSWITCHers,
>
> So after a bit discussion, a case of red bull, an approach to automating
> functional testing has now solidified.
>
> Audio Path is verified via DTMF in-band, Video Path is verified by using
> QR codes.
>
> Here are two VERY crude examples:
>
> https://www.dropbox.com/s/4fkdcgzo8dt0hdr/conference_test.pl?dl=0
>
> https://www.dropbox.com/s/497rm2hgopp5u7h/eavesdrop_test.pl?dl=0
>
> Input?
>
> --
>
> *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
>
> _________________________________________________________________________
> 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 sendmeallyouroffers at googlemail.com  Tue Jul 28 12:40:02 2015
From: sendmeallyouroffers at googlemail.com (Berthold Karl)
Date: Tue, 28 Jul 2015 10:40:02 +0200
Subject: [Freeswitch-users] freeswitch never seems to call sendmail when
 trying to email voicemails
In-Reply-To: <CAN6oSBt4VqjViDaXdkund+Ch9W=5JF__n55sOx5PVXYdAn3Zgw@mail.gmail.com>
References: <54E6B9A1.20708@dickson.st>
	<CAHtxdDd8CRENTyNoi7FOn3yuy+fyCshEiknfhvhZcg36q=JFuA@mail.gmail.com>
	<54EA5B78.3010504@dickson.st>
	<CAHtxdDcD0fqTWR9mA8=-eT9rm2a1M1p3Qab7OVXCfH=wYnHvvw@mail.gmail.com>
	<54EBE76A.2040709@dickson.st>
	<CAHtxdDfsm2pd8HDM-WwU8HKJT5AQj3isqjWF4aWVbXkRrtB0zg@mail.gmail.com>
	<CAEJMVkBY-3nOfXijxhED_byC_K5cNnS8LVW3iRjGmOFserDmjw@mail.gmail.com>
	<CAN6oSBuxXbf=uecMk1UO=ZP9xxpvoXA65Ox4pyZVmPxA-BNwnQ@mail.gmail.com>
	<CA+Xtiefu7T+LBFD11k38wPG-QCPcrWJuaAaLGTOmS_RTmk8m7A@mail.gmail.com>
	<29646517.20150727173302@seznam.cz>
	<C18BE15B-86F3-40C1-ABED-CA328565C0C1@mgtech.com>
	<CAN6oSBt4VqjViDaXdkund+Ch9W=5JF__n55sOx5PVXYdAn3Zgw@mail.gmail.com>
Message-ID: <CAN6oSBsVjND7y_m=APj+QRr3yvfT=9hPLxbietUnAsdiiid8dA@mail.gmail.com>

Hi,

so I wrote a hangup hook script... I'am not able to get sendmail running
with freeswitch.

I wrote a perl script which is called as hangup_hook.
Thanks to $env, its serves all the informations I needed.
After some testings and playing with hangup_hook I've added some more
features.


Only one Problem left. How can I tell FS, thats MWI is not needed.

Regards

2015-07-28 9:19 GMT+02:00 Berthold Karl <sendmeallyouroffers at googlemail.com>
:

> Thanks for the replies!
>
> I've tested sendmail, it works for the user which is running freeswitch.
>
> I'am running FS 1.4.8 under Debian.
>
> I want to use the FS internal VM-to-E-Mail because all informations about
> To-, From- etc are configured in the user.xml. If I'am running my own
> script, I have to setup everything in a second configuration or have to
> parse the user.xml a second time..
>
> Thx Dennis for the Sample script,
>
> Any ideas?
> I've tested sendmail with few arguments... everytime its ends with a
> segfault... It seems FS isn't creating the Mail under /tmp/. /tmp is World
> writable/readable.
>
> Regards
>
>
>
> 2015-07-27 19:01 GMT+02:00 Mario <mario_fs at mgtech.com>:
>
>> Although this is in the Mac OS X page I wrote this up which may help:
>> https://freeswitch.org/confluence/display/FREESWITCH/Installation+and+Setup+on+OS+X#InstallationandSetuponOSX-EmailVoicemailtoaniPhone
>>
>> To test your email you can do this from a command line:
>> printf "Subject: TestnHello" | sendmail -f you at domain.com you at domain.com
>> Mareio G
>>
>> On Jul 27, 2015, at 8:33 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
>>
>>  Hi,
>>
>> Why do not use some simple script after the ends session?
>>
>> For example simple bash script send_voicemail.sh
>>
>> #!/bin/sh
>> cd /usr/local/freeswitch/recordings/voicemail/
>> f=`find -name \*.wav`
>> for file in $f
>> do
>> echo "Processing ${file}"
>> CURRENT=$(date +%d.%m.%y_%H:%M:%S)
>> datetime=$CURRENT
>> echo $datetime
>> sendemail -f voicemail at mail.com -t somemail at mail.com -m "Voicemail" -u
>> "Voicemail od $datetime" -a ${file} -s smtp.mail.com -xu user -xp pass
>> echo "BackUP ${file}"
>> mv ${file} /usr/local/freeswitch/recordings/backup_voicemail/
>> done
>>
>>
>> And in dialplan start like this
>> <action application="export" data="api_hangup_hook=system
>> /usr/local/freeswitch/scripts/send_voicemail.sh"/>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>> <http://mob.tel>. 775-415-382 pond?l? 27. ?ervence 2015, 17:11:10, napsal
>> jste: *
>>
>>  Why can.t you create your own script and use it to send vm over email?
>> 27 ???? 2015 ?. 18:32 ???????????? "Berthold Karl" <
>> sendmeallyouroffers at googlemail.com> ???????:
>> Hi,
>>
>> my FS is calling the sendmail-command, but sendmail ends with an
>> segfault. It seems like there is no email under /tmp/. Did your sendmail
>> also ends in a segfault?
>>
>> 2015-02-24 16:51 GMT+01:00 Brian West <brian at freeswitch.org>:
>> When you file the JIRA please attach your voicemail.conf.xml.
>>
>> On Tue, Feb 24, 2015 at 8:21 AM, Sergey Safarov <s.safarov at gmail.com>
>> wrote:
>> If you want to FS correctly processed parameters vm-mailfrom and
>> email-from, write a request to https://freeswitch.org/jira/
>>
>> On Tue, Feb 24, 2015 at 5:52 AM, Jason Lewis <jason at dickson.st> wrote:
>> Thanks Sergey,
>>
>> I had configured the domain variable to the fqdn of the machine. I
>> eventually got it working though, I was missing two key lines from my user
>> config:
>>      <param name="vm-email-all-messages" value="true"/>
>>      <param name="vm-attach-file" value="true" />
>>
>> I'm not sure how I managed to miss those but anyway, that seems to have
>> resolved things.
>>
>> It seems as though vm-mailfrom is still being ignored though. Currently I
>> have it set to:
>>
>>       <param name="vm-mailfrom" value="vm at freeswitch.xyz.com.au"
>> <vm at freeswitch.xyz.com.au>/>
>>
>> but voicemails get delivered from:
>>
>> 1001 at freeswitch.xyz.com.au
>>
>> Is this worth investigating further?
>>
>> Jason
>>
>> Sergey Safarov wrote on 23/02/2015 5:05 PM:
>> Try configure "domain" variable in vars.xml
>>
>> <X-PRE-PROCESS cmd="set" data="domain=you_domain_name"/>
>>
>> After it verify that user registered with domain name
>>
>> freeswitch at internal> sofia status profile internal reg
>>
>> Registrations:
>>
>> =================================================================================================
>> Call-ID:    1B26-2327-466848134BEBC9719CDE-002 at SipHost
>> User:       1201 at you_domain_name
>> Contact:    "1201" <
>> sip:1201 at 10.21.18.22:5060;fs_nat=yes;fs_path=sip%3A1201%4010.21.18.22%3A5060
>> >
>> Agent:      204 12-3868-2416-0.10.56.1-DS
>> Status:     Registered(UDP-NAT)(unknown) EXP(2015-02-23 06:05:22)
>> EXPSECS(139)
>> Ping-Status:Reachable
>> Host:       fs1.you_domain_name
>> IP:         10.21.18.22
>> Port:       5060
>> Auth-User:  1201
>> Auth-Realm: you_domain_name
>> MWI-Account:1201 at you_domain_name
>>
>> Sergey
>>
>>
>> On Mon, Feb 23, 2015 at 1:43 AM, Jason Lewis <jason at dickson.st
>> > wrote:
>>
>> So I've managed to see some output from the sendmail program in the FS
>> logs. It appears that my fs instance isn't correctly setting its domain?
>>
>> the FS box has a fqdn, and I also set the domain parameter in the
>> vars.xml file, but still the voicemail is sent with a from address of an IP
>> address.
>>
>> Any ideas?
>>
>> Net::SMTP>>> Net::SMTP(2.33)
>> Net::SMTP>>>   Net::Cmd(2.30)
>> Net::SMTP>>>     Exporter(5.71)
>> Net::SMTP>>>   IO::Socket::INET(1.35)
>> Net::SMTP>>>     IO::Socket(1.37)
>> Net::SMTP>>>       IO::Handle(1.35)
>> Net::SMTP=GLOB(0x23f7748)<<< 220 mb.xyz.com.au ESMTP Postfix (Debian/GNU)
>> Net::SMTP=GLOB(0x23f7748)>>> EHLO localhost.localdomain
>> Net::SMTP=GLOB(0x23f7748)<<< 250-mb.bongalong.st
>>
>> Net::SMTP=GLOB(0x23f7748)<<< 250-PIPELINING
>> Net::SMTP=GLOB(0x23f7748)<<< 250-SIZE 10240000
>> Net::SMTP=GLOB(0x23f7748)<<< 250-VRFY
>> Net::SMTP=GLOB(0x23f7748)<<< 250-ETRN
>> Net::SMTP=GLOB(0x23f7748)<<< 250-STARTTLS
>> Net::SMTP=GLOB(0x23f7748)<<< 250-ENHANCEDSTATUSCODES
>> Net::SMTP=GLOB(0x23f7748)<<< 250-8BITMIME
>> Net::SMTP=GLOB(0x23f7748)<<< 250 DSN
>> Net::SMTP=GLOB(0x23f7748)>>> MAIL FROM:<1002 at 192.168.1.3>
>> <1002 at 192.168.1.3>
>> Net::SMTP=GLOB(0x23f7748)<<< 501 5.1.7 Bad sender address syntax
>> Net::SMTP=GLOB(0x23f7748)>>> RCPT TO:<jason at xyz.com.au>
>> <jason at xyz.com.au>
>> Net::SMTP=GLOB(0x23f7748)<<< 503 5.5.1 Error: need MAIL command
>> Net::SMTP=GLOB(0x23f7748)>>> DATA
>> Net::SMTP=GLOB(0x23f7748)<<< 503 5.5.1 Error: need RCPT command
>> Net::SMTP=GLOB(0x23f7748)>>> QUIT
>> Net::SMTP=GLOB(0x23f7748)<<< 221 2.0.0 Bye
>> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1633 Session 3
>> (loopback/voicemail-a) Ended
>> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1637 Close
>> Channel loopback/voicemail-a [CS_DESTROY]
>> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1633 Session 4
>> (loopback/voicemail-b) Ended
>> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1637 Close
>> Channel loopback/voicemail-b [CS_DESTROY]
>>
>>
>> On 20/02/2015 7:10 pm, Sergey Safarov wrote:
>> You mailer is not understand "mailer-app-args" has been configured in
>> "switch.conf.xml"
>> Remove extra arg or add required
>>
>> ??, 20 ????. 2015, 7:49, Jason Lewis <jason at dickson.st>:
>> Hi,
>>
>> I've been trying to make freeswitch email voicemails but as far as I can
>> tell, it never even calls sendmail.
>>
>> I have setting mailer-app to "sendmail" and "/usr/sbin/sendmail" to no
>> avail. I can successfully send an email from the commandline using
>> sendmail. (sendmail in this case is provided by postfix)
>>
>> I see no emails in the postfix mail logs when I leave a voicemail message.
>>
>> I also tried creating a shell just to see if it even gets called from
>> fs, but it does not get called when a voicemail is deposited:
>> #!/bin/bash
>> echo $(date --rfc-3339=ns): $* >> /tmp/freeswitchsendmail.log
>>
>> After every change, I have run reloadxml and reload mod_voicemail. I
>> have also tried restarting freeswitch.
>>
>> I am running the debian packages of FreeSWITCH Version 1.4.15-1~64bit
>> (-1 64bit)
>>
>> my configuration is based on the vanilla configuration with only very
>> minor changes.
>>
>> I'm at a loss as to how to debug further, but I'm pretty sure the
>> mailer-app is never called. Is there some setting I'm missing or
>> something obvious I'm not doing?
>>
>>
>> My config:
>> 1001.xml:
>> <include>
>>  <user id="1001">
>>    <params>
>>      <param name="password" value="$${default_password}"/>
>>      <param name="vm-password" value="1001"/>
>>      <param name="vm-mailto" value="jason at xyz.com.au"/>
>>      <param name="vm-mailfrom" value="vm at freeswitch.xyz.com.au"/>
>>      <!-- <param name="vm-message-ext" value="wav"/> -->
>>    </params>
>>    <variables>
>>      <variable name="toll_allow" value="domestic,international,local"/>
>>      <variable name="accountcode" value="1001"/>
>>      <variable name="user_context" value="default"/>
>>      <variable name="effective_caller_id_name" value="Extension 1001"/>
>>      <variable name="effective_caller_id_number" value="1001"/>
>>      <variable name="outbound_caller_id_name"
>> value="$${outbound_caller_name}"/>
>>      <variable name="outbound_caller_id_number"
>> value="$${outbound_caller_id}"/>
>>      <variable name="callgroup" value="techsupport"/>
>>    </variables>
>>  </user>
>> </include>
>>
>> and in switch.conf.xml I have the following set:
>>
>>    <param name="mailer-app" value="/usr/sbin/sendmail"/>
>>    <param name="mailer-app-args" value="-t"/>
>>
>>
>> I made a log at level 7 and put it on the pastebin:
>>
>> https://pastebin.freeswitch.org/23921
>>
>>
>> Jason Lewis
>> http://emacstragic.net
>>
>>
>> _________________________________________________________________________
>> 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
>>
>>
>> --
>> Jason Lewis
>> http://emacstragic.net
>>
>> _________________________________________________________________________
>> 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
>>
>> --
>> Jason Lewis
>> http://emacstragic.net
>>
>> _________________________________________________________________________
>> 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 *brian at freeswitch.org
>>
>> *Twitter: @FreeSWITCH , @briankwest *http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> 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 mkvonarx at gmail.com  Tue Jul 28 13:27:06 2015
From: mkvonarx at gmail.com (Markus von Arx)
Date: Tue, 28 Jul 2015 11:27:06 +0200
Subject: [Freeswitch-users] how to add mod_event_socket and mod_conference
	activities to the logfile
Message-ID: <CAHA5ALeUAOhuQAhJU38mh_qbomY3BAwE_pVPNGm8raUhoQuodg@mail.gmail.com>

Hallo

I'd like to see more details about what's going on in the mod_event_socket
and in the mod_conference modules in the FreeSWITCH log files. With the
default (vanilla) configuration, nothing gets logged to the freeswitch.log
logfile from inside these modules. I scanned all config files and the FS
confluence wiki, but could not find any hint on how this can be achieved.

The information I'd like to see in the freeswitch.log logfile is e.g. the
following:
* state changes in the mod_conference conferences, e.g. conference created,
conference destroyed, member added, member removed, member muted, member
unmuted, ... (including affected conference name and member id of course)
* requests, replies & events in the mod_event_socket - if possible
"shortened" and not including the whole event texts

Any hints how this can be configured?

Thanks a lot,
Markus
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From adam.lappe at qsc.de  Tue Jul 28 13:53:00 2015
From: adam.lappe at qsc.de (Lappe, Adam)
Date: Tue, 28 Jul 2015 09:53:00 +0000
Subject: [Freeswitch-users] mod_local_stream moh confusion
In-Reply-To: <2E67ADAE1D3582409C90EBAE8C64C5070E58E0@QSCDEMXP01B.ONE4ALL.LAN>
References: <2E67ADAE1D3582409C90EBAE8C64C5070E5569@QSCDEMXP01B.ONE4ALL.LAN>, 
	<CAEJMVkAnYcxOzL19E_Zs9Bn73zx8gAtGuTCYGZPrqsFhVb6n3g@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C5070E58E0@QSCDEMXP01B.ONE4ALL.LAN>
Message-ID: <2E67ADAE1D3582409C90EBAE8C64C5070E60E4@QSCDEMXP01B.ONE4ALL.LAN>

Other Question:

how can i set a other file to use for moh.

I make an outgoing call and set the moh path to a wav file. This file is only used to play it to the callee, if I set him  on hold.
Now, when the callee transfers my call, I receive a RE-INVITE and my FreeSWITCH tries to play a music on hold using the mod_local_stream.

Again the question: why does FreeSWITCH try to play a moh und why does he try to use mod_local_stream?
Can I change local_stream to another wav file somehow?

Thanks in advance,
Adam

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Lappe, Adam
Gesendet: Freitag, 24. Juli 2015 12:30
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] mod_local_stream moh confusion

Hello Brian,

thanks for your answer.

You are correct. As you can see in my first post, i do not use mod_local_stream.
The question is: WHY does FreeSWITCH try to use a local_stream for moh, when a user gets transfered (with an reinvite)?

I grep'ed my config files, i don't load or use mod_local_stream anywhere.

Best regards,
Adam
________________________________________
Von: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org]&quot; im Auftrag von &quot;Brian West [brian at freeswitch.org]
Gesendet: Donnerstag, 23. Juli 2015 17:09
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] mod_local_stream moh confusion

It would indicate that you do not have mod_local_stream loaded.

On Thu, Jul 23, 2015 at 3:15 AM, Lappe, Adam <adam.lappe at qsc.de<mailto:adam.lappe at qsc.de>> wrote:
Hello,

i am testing the latest 1.4.19 Version of FreeSWITCH.
Currently we are running an old 1.2.7 Version.

Everything seems to work fine, but there is 1 error that is very confusing:

When a call gets transfered by the callee (i.e. by the receptionist) the call will be terminated.
All I see is this error line:
[ERR] switch_core_file.c:149 Invalid file format [local_stream] for [moh]!

I don't use the local_stream module.
freeswitch at internal> module_exists mod_local_stream false

This error does not exists with the old version.

Is this a bug, or am I missing something?

Thanks in advance,
Adam

_________________________________________________________________________
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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

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


_________________________________________________________________________
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 alxpol at gmail.com  Tue Jul 28 15:12:03 2015
From: alxpol at gmail.com (Alex Polischuk)
Date: Tue, 28 Jul 2015 14:12:03 +0300
Subject: [Freeswitch-users] audio Handshake failure 1
In-Reply-To: <CAK59s4w87K5dftvFVwJmNRmTqy29A8rM5UmfSyPnzMaFjs0cYA@mail.gmail.com>
References: <CAGf2z7NE0a7mPsPXjZK9w06jnTJTh--R01O2dvBRyhQPxRmxmw@mail.gmail.com>
	<CAK59s4w87K5dftvFVwJmNRmTqy29A8rM5UmfSyPnzMaFjs0cYA@mail.gmail.com>
Message-ID: <CAGf2z7NZkq47S0wT+kbT_ptNXGWb3J6uYWX_RVmWeRnB_ufhHA@mail.gmail.com>

Thanks


On Mon, Jul 27, 2015 at 5:45 PM, Javier Menendez <menendez.garcia at gmail.com>
wrote:

> Please check *https://freeswitch.org/jira/browse/FS-7839
> <https://freeswitch.org/jira/browse/FS-7839> *
>
> On Mon, Jul 27, 2015 at 8:33 AM, Alex Polischuk <alxpol at gmail.com> wrote:
>
>> Hi All,
>>
>> I use FreeSwitch for transcoding SRTP from WebRTC to SIP with RTP and
>> vice versa.
>> When Firefox version 38 add requires for Perfect Forward Secrecy (PFS) I
>> have problem with Handshake in FS when Firefox Browser terminate a call. I
>> found that that problem was fixed in the version 1.4.19
>> https://freeswitch.org/jira/browse/FS-7425
>>
>> I have the same problem in the version 1.4.20
>>
>> eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
>> switch_rtp.c:2924 Changing audio DTLS state from HANDSHAKE to SETUP
>> eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
>> switch_rtp.c:2832 audio Fingerprint Verified.
>> eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
>> switch_rtp.c:3384 Activating Audio Secure RTP SEND
>> eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
>> switch_rtp.c:3362 Activating Audio Secure RTP RECV
>> eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO]
>> switch_rtp.c:2872 Changing audio DTLS state from SETUP to READY
>> eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.604193 [DEBUG]
>> switch_rtp.c:1937 rtcp_stats_init: ssrc[-1374535478] base_seq[6179]
>> f01c46d8-33a0-11e5-ac83-891cb7dea826 2015-07-26 14:16:46.644210 [DEBUG]
>> switch_rtp.c:5884 Correct ip/port confirmed.
>> f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [ERR]
>> switch_rtp.c:2917 audio Handshake failure 1
>> f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [INFO]
>> switch_rtp.c:2918 Changing audio DTLS state from HANDSHAKE to FAIL
>> f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [NOTICE]
>> switch_rtp.c:2899 Hangup sofia/external/
>> 992555500004 at voip.webrtc.jajah.com [CS_EXCHANGE_MEDIA]
>> [DESTINATION_OUT_OF_ORDER]
>>
>>
>> what additional definition I should make to resolve this problem?
>>
>> Thanks,
>> Alex Polischuk
>>
>>
>> _________________________________________________________________________
>> 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 italorossib at gmail.com  Tue Jul 28 16:37:32 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Tue, 28 Jul 2015 09:37:32 -0300
Subject: [Freeswitch-users] mod_local_stream moh confusion
In-Reply-To: <2E67ADAE1D3582409C90EBAE8C64C5070E58E0@QSCDEMXP01B.ONE4ALL.LAN>
References: <2E67ADAE1D3582409C90EBAE8C64C5070E5569@QSCDEMXP01B.ONE4ALL.LAN>
	<CAEJMVkAnYcxOzL19E_Zs9Bn73zx8gAtGuTCYGZPrqsFhVb6n3g@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C5070E58E0@QSCDEMXP01B.ONE4ALL.LAN>
Message-ID: <CAJR7f2Hb0VfBVdzH7fiTzS3DLMtEoBNGWqMArsDT4xZKMi5q0Q@mail.gmail.com>

Probably because you have

<X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>

in your vars.xml and you have

<param name="hold-music" value="$${hold_music}"/>

in your sofia profiles.

On Fri, Jul 24, 2015 at 7:30 AM, Lappe, Adam <adam.lappe at qsc.de> wrote:

> Hello Brian,
>
> thanks for your answer.
>
> You are correct. As you can see in my first post, i do not use
> mod_local_stream.
> The question is: WHY does FreeSWITCH try to use a local_stream for moh,
> when a user gets transfered (with an reinvite)?
>
> I grep'ed my config files, i don't load or use mod_local_stream anywhere.
>
> Best regards,
> Adam
> ________________________________________
> Von: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org]&quot; im Auftrag von
> &quot;Brian West [brian at freeswitch.org]
> Gesendet: Donnerstag, 23. Juli 2015 17:09
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] mod_local_stream moh confusion
>
> It would indicate that you do not have mod_local_stream loaded.
>
> On Thu, Jul 23, 2015 at 3:15 AM, Lappe, Adam <adam.lappe at qsc.de<mailto:
> adam.lappe at qsc.de>> wrote:
> Hello,
>
> i am testing the latest 1.4.19 Version of FreeSWITCH.
> Currently we are running an old 1.2.7 Version.
>
> Everything seems to work fine, but there is 1 error that is very confusing:
>
> When a call gets transfered by the callee (i.e. by the receptionist) the
> call will be terminated.
> All I see is this error line:
> [ERR] switch_core_file.c:149 Invalid file format [local_stream] for [moh]!
>
> I don?t use the local_stream module.
> freeswitch at internal> module_exists mod_local_stream
> false
>
> This error does not exists with the old version.
>
> Is this a bug, or am I missing something?
>
> Thanks in advance,
> Adam
>
> _________________________________________________________________________
> 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
>
>
>
> --
>
> Brian West
> brian at freeswitch.org<mailto:brian at freeswitch.org>
>
> [http://billing.freeswitch.org/templates/default/img/whmcslogo.png]
>
> 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
>
>
> _________________________________________________________________________
> 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
>



-- 
?talo Rossi
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From italorossib at gmail.com  Tue Jul 28 16:39:12 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Tue, 28 Jul 2015 09:39:12 -0300
Subject: [Freeswitch-users] mod_local_stream moh confusion
In-Reply-To: <2E67ADAE1D3582409C90EBAE8C64C5070E60E4@QSCDEMXP01B.ONE4ALL.LAN>
References: <2E67ADAE1D3582409C90EBAE8C64C5070E5569@QSCDEMXP01B.ONE4ALL.LAN>
	<CAEJMVkAnYcxOzL19E_Zs9Bn73zx8gAtGuTCYGZPrqsFhVb6n3g@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C5070E58E0@QSCDEMXP01B.ONE4ALL.LAN>
	<2E67ADAE1D3582409C90EBAE8C64C5070E60E4@QSCDEMXP01B.ONE4ALL.LAN>
Message-ID: <CAJR7f2FccPRC7Acpzhjx1CbF=dAxyunt3XrQre7j1bC9xQAkTA@mail.gmail.com>

https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-MusicOnHoldRelated

On Tue, Jul 28, 2015 at 6:53 AM, Lappe, Adam <adam.lappe at qsc.de> wrote:

> Other Question:
>
> how can i set a other file to use for moh.
>
> I make an outgoing call and set the moh path to a wav file. This file is
> only used to play it to the callee, if I set him  on hold.
> Now, when the callee transfers my call, I receive a RE-INVITE and my
> FreeSWITCH tries to play a music on hold using the mod_local_stream.
>
> Again the question: why does FreeSWITCH try to play a moh und why does he
> try to use mod_local_stream?
> Can I change local_stream to another wav file somehow?
>
> Thanks in advance,
> Adam
>
> -----Urspr?ngliche Nachricht-----
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Lappe, Adam
> Gesendet: Freitag, 24. Juli 2015 12:30
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] mod_local_stream moh confusion
>
> Hello Brian,
>
> thanks for your answer.
>
> You are correct. As you can see in my first post, i do not use
> mod_local_stream.
> The question is: WHY does FreeSWITCH try to use a local_stream for moh,
> when a user gets transfered (with an reinvite)?
>
> I grep'ed my config files, i don't load or use mod_local_stream anywhere.
>
> Best regards,
> Adam
> ________________________________________
> Von: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org]&quot; im Auftrag von
> &quot;Brian West [brian at freeswitch.org]
> Gesendet: Donnerstag, 23. Juli 2015 17:09
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] mod_local_stream moh confusion
>
> It would indicate that you do not have mod_local_stream loaded.
>
> On Thu, Jul 23, 2015 at 3:15 AM, Lappe, Adam <adam.lappe at qsc.de<mailto:
> adam.lappe at qsc.de>> wrote:
> Hello,
>
> i am testing the latest 1.4.19 Version of FreeSWITCH.
> Currently we are running an old 1.2.7 Version.
>
> Everything seems to work fine, but there is 1 error that is very confusing:
>
> When a call gets transfered by the callee (i.e. by the receptionist) the
> call will be terminated.
> All I see is this error line:
> [ERR] switch_core_file.c:149 Invalid file format [local_stream] for [moh]!
>
> I don't use the local_stream module.
> freeswitch at internal> module_exists mod_local_stream false
>
> This error does not exists with the old version.
>
> Is this a bug, or am I missing something?
>
> Thanks in advance,
> Adam
>
> _________________________________________________________________________
> 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
>
>
>
> --
>
> Brian West
> brian at freeswitch.org<mailto:brian at freeswitch.org>
>
> [http://billing.freeswitch.org/templates/default/img/whmcslogo.png]
>
> 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
>
>
> _________________________________________________________________________
> 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
>



-- 
?talo Rossi
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From vma at 440hz.fr  Tue Jul 28 16:54:45 2015
From: vma at 440hz.fr (Vallimamod Abdullah)
Date: Tue, 28 Jul 2015 14:54:45 +0200
Subject: [Freeswitch-users] Freetdm configuration
In-Reply-To: <CADeoHupM02LzV-S+n=+pMe0o1C1tEAOVbE9=EredLg2Xm5qqQw@mail.gmail.com>
References: <CADeoHupM02LzV-S+n=+pMe0o1C1tEAOVbE9=EredLg2Xm5qqQw@mail.gmail.com>
Message-ID: <9734427A-3A0F-44D3-8A6F-C5929DADFD90@440hz.fr>

Hi,

You are apparently getting physical alarms with the link. Check the card and your cables first to see if the link is stable and you are not getting any error on wanpipe side. Sangoma wiki may help: http://wiki.sangoma.com/Wanpipemon-T1-E1-physical-Line-alarms <http://wiki.sangoma.com/Wanpipemon-T1-E1-physical-Line-alarms>

Best Regards,
Vallimamod
.

> On 27 Jul 2015, at 20:03, Nikolay Zaytsev <nzaytsevc at gmail.com> wrote:
> 
> 
> 
> Good day to you.
> I have a problem with the setting up of the freetdm module.
> I have sangoma a102de card.
> The system recognizes the card correctly.
> Audiocodes mediant 600 with the same trunk configuration works corrctly.
> But during the startup freeswitch writes
> [DEBUG] ftdm_io.c:5469 Module sangoma_isdn does not support configuration.
> and after that
> [INFO] ftmod_sangoma_isdn_stack_rcv.c:1098 sng_isdn->s2: Link is down, dropping transmit frame.
> Config files are attached.
> Please, help me:)
> Best regards,
> Nikolay Zaytsev
> 
> 
> 
> <freeswitch.log><freetdm.conf><freetdm.conf.xml>_________________________________________________________________________
> 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 adam.lappe at qsc.de  Tue Jul 28 18:14:23 2015
From: adam.lappe at qsc.de (Lappe, Adam)
Date: Tue, 28 Jul 2015 14:14:23 +0000
Subject: [Freeswitch-users] mod_local_stream moh confusion
In-Reply-To: <CAJR7f2FccPRC7Acpzhjx1CbF=dAxyunt3XrQre7j1bC9xQAkTA@mail.gmail.com>
References: <2E67ADAE1D3582409C90EBAE8C64C5070E5569@QSCDEMXP01B.ONE4ALL.LAN>
	<CAEJMVkAnYcxOzL19E_Zs9Bn73zx8gAtGuTCYGZPrqsFhVb6n3g@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C5070E58E0@QSCDEMXP01B.ONE4ALL.LAN>
	<2E67ADAE1D3582409C90EBAE8C64C5070E60E4@QSCDEMXP01B.ONE4ALL.LAN>
	<CAJR7f2FccPRC7Acpzhjx1CbF=dAxyunt3XrQre7j1bC9xQAkTA@mail.gmail.com>
Message-ID: <2E67ADAE1D3582409C90EBAE8C64C5070E6140@QSCDEMXP01B.ONE4ALL.LAN>

Hi,

thanks for your suggestions:

I neither have
<X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
nor
<param name="hold-music" value="$${hold_music}"/>
in my config file. I also don?t have vars.xml or X-PRE-PROCESSES but 1 large freeswitch.xml.

When I get transferred I receive a RE-INVITE with a=sendonly media attribute in SDP.
Then FreeSWITCH tries to use local_stream://moh (which is not configured anywhere)!!

Also I tried
<action application="set" data="temp_hold_music=silence"/>
or
<action application="set" data=" temp_hold_music= (path)/blub.wav "/>
but this does not work.

How can I disable the local_stream://moh here?
Any other suggestions?

Thanks,
Adam

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von ?talo Rossi
Gesendet: Dienstag, 28. Juli 2015 14:39
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] mod_local_stream moh confusion

https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-MusicOnHoldRelated

On Tue, Jul 28, 2015 at 6:53 AM, Lappe, Adam <adam.lappe at qsc.de<mailto:adam.lappe at qsc.de>> wrote:
Other Question:

how can i set a other file to use for moh.

I make an outgoing call and set the moh path to a wav file. This file is only used to play it to the callee, if I set him  on hold.
Now, when the callee transfers my call, I receive a RE-INVITE and my FreeSWITCH tries to play a music on hold using the mod_local_stream.

Again the question: why does FreeSWITCH try to play a moh und why does he try to use mod_local_stream?
Can I change local_stream to another wav file somehow?

Thanks in advance,
Adam

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] Im Auftrag von Lappe, Adam
Gesendet: Freitag, 24. Juli 2015 12:30
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] mod_local_stream moh confusion

Hello Brian,

thanks for your answer.

You are correct. As you can see in my first post, i do not use mod_local_stream.
The question is: WHY does FreeSWITCH try to use a local_stream for moh, when a user gets transfered (with an reinvite)?

I grep'ed my config files, i don't load or use mod_local_stream anywhere.

Best regards,
Adam
________________________________________
Von: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>]&quot; im Auftrag von &quot;Brian West [brian at freeswitch.org<mailto:brian at freeswitch.org>]
Gesendet: Donnerstag, 23. Juli 2015 17:09
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] mod_local_stream moh confusion

It would indicate that you do not have mod_local_stream loaded.

On Thu, Jul 23, 2015 at 3:15 AM, Lappe, Adam <adam.lappe at qsc.de<mailto:adam.lappe at qsc.de><mailto:adam.lappe at qsc.de<mailto:adam.lappe at qsc.de>>> wrote:
Hello,

i am testing the latest 1.4.19 Version of FreeSWITCH.
Currently we are running an old 1.2.7 Version.

Everything seems to work fine, but there is 1 error that is very confusing:

When a call gets transfered by the callee (i.e. by the receptionist) the call will be terminated.
All I see is this error line:
[ERR] switch_core_file.c:149 Invalid file format [local_stream] for [moh]!

I don't use the local_stream module.
freeswitch at internal> module_exists mod_local_stream false

This error does not exists with the old version.

Is this a bug, or am I missing something?

Thanks in advance,
Adam

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org><mailto: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><mailto: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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org><mailto:brian at freeswitch.org<mailto:brian at freeswitch.org>>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

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<tel:%2B19184209001> | F:+19184209002<tel:%2B19184209002> | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001<tel:%2B883%205100%201420%209001> | ISN:410*543 | Skype:briankwest


_________________________________________________________________________
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



--
?talo Rossi
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From italorossib at gmail.com  Tue Jul 28 18:36:32 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Tue, 28 Jul 2015 11:36:32 -0300
Subject: [Freeswitch-users] mod_local_stream moh confusion
In-Reply-To: <2E67ADAE1D3582409C90EBAE8C64C5070E6140@QSCDEMXP01B.ONE4ALL.LAN>
References: <2E67ADAE1D3582409C90EBAE8C64C5070E5569@QSCDEMXP01B.ONE4ALL.LAN>
	<CAEJMVkAnYcxOzL19E_Zs9Bn73zx8gAtGuTCYGZPrqsFhVb6n3g@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C5070E58E0@QSCDEMXP01B.ONE4ALL.LAN>
	<2E67ADAE1D3582409C90EBAE8C64C5070E60E4@QSCDEMXP01B.ONE4ALL.LAN>
	<CAJR7f2FccPRC7Acpzhjx1CbF=dAxyunt3XrQre7j1bC9xQAkTA@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C5070E6140@QSCDEMXP01B.ONE4ALL.LAN>
Message-ID: <CAJR7f2E6aMK_do1g0S9SF1y5eyFt77Vh8AjemWpQRcfyh=r8KA@mail.gmail.com>

You'll probably need to export it, or set it globally like fs_cli -x
'global_setvar temp_hold_music=silence'

local_stream://moh is set in switch_core_media if temp_hold_music and
hold_music aren't set

On Tue, Jul 28, 2015 at 11:14 AM, Lappe, Adam <adam.lappe at qsc.de> wrote:

>  Hi,
>
>
>
> thanks for your suggestions:
>
>
>
> I neither have
>
> <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
>
> nor
>
> <param name="hold-music" value="$${hold_music}"/>
>
> in my config file. I also don?t have vars.xml or X-PRE-PROCESSES but 1
> large freeswitch.xml.
>
>
>
> When I get transferred I receive a RE-INVITE with *a=sendonly* media
> attribute in SDP.
>
> Then FreeSWITCH tries to use local_stream://moh (which is not configured
> anywhere)!!
>
>
>
> Also I tried
>
> <action application="set" data="temp_hold_music=silence"/>
>
> or
>
> <action application="set" data=" temp_hold_music= (path)/blub.wav "/>
>
> but this does not work.
>
>
>
> How can I disable the local_stream://moh here?
>
> Any other suggestions?
>
>
>
> Thanks,
>
> Adam
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *?talo
> Rossi
> *Gesendet:* Dienstag, 28. Juli 2015 14:39
>
> *An:* FreeSWITCH Users Help
> *Betreff:* Re: [Freeswitch-users] mod_local_stream moh confusion
>
>
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-MusicOnHoldRelated
>
>
>
> On Tue, Jul 28, 2015 at 6:53 AM, Lappe, Adam <adam.lappe at qsc.de> wrote:
>
> Other Question:
>
> how can i set a other file to use for moh.
>
> I make an outgoing call and set the moh path to a wav file. This file is
> only used to play it to the callee, if I set him  on hold.
> Now, when the callee transfers my call, I receive a RE-INVITE and my
> FreeSWITCH tries to play a music on hold using the mod_local_stream.
>
> Again the question: why does FreeSWITCH try to play a moh und why does he
> try to use mod_local_stream?
> Can I change local_stream to another wav file somehow?
>
> Thanks in advance,
> Adam
>
> -----Urspr?ngliche Nachricht-----
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Lappe, Adam
> Gesendet: Freitag, 24. Juli 2015 12:30
>
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] mod_local_stream moh confusion
>
> Hello Brian,
>
> thanks for your answer.
>
> You are correct. As you can see in my first post, i do not use
> mod_local_stream.
> The question is: WHY does FreeSWITCH try to use a local_stream for moh,
> when a user gets transfered (with an reinvite)?
>
> I grep'ed my config files, i don't load or use mod_local_stream anywhere.
>
> Best regards,
> Adam
> ________________________________________
> Von: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org]&quot; im Auftrag von
> &quot;Brian West [brian at freeswitch.org]
> Gesendet: Donnerstag, 23. Juli 2015 17:09
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] mod_local_stream moh confusion
>
> It would indicate that you do not have mod_local_stream loaded.
>
> On Thu, Jul 23, 2015 at 3:15 AM, Lappe, Adam <adam.lappe at qsc.de<mailto:
> adam.lappe at qsc.de>> wrote:
> Hello,
>
> i am testing the latest 1.4.19 Version of FreeSWITCH.
> Currently we are running an old 1.2.7 Version.
>
> Everything seems to work fine, but there is 1 error that is very confusing:
>
> When a call gets transfered by the callee (i.e. by the receptionist) the
> call will be terminated.
> All I see is this error line:
> [ERR] switch_core_file.c:149 Invalid file format [local_stream] for [moh]!
>
> I don't use the local_stream module.
> freeswitch at internal> module_exists mod_local_stream false
>
> This error does not exists with the old version.
>
> Is this a bug, or am I missing something?
>
> Thanks in advance,
> Adam
>
> _________________________________________________________________________
> 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
>
>
>
> --
>
> Brian West
> brian at freeswitch.org<mailto:brian at freeswitch.org>
>
> [http://billing.freeswitch.org/templates/default/img/whmcslogo.png]
>
> 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
>
>
> _________________________________________________________________________
> 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
>
>
>
>
>
> --
>
> ?talo Rossi
>
> _________________________________________________________________________
> 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
>



-- 
?talo Rossi
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From brian at freeswitch.org  Tue Jul 28 19:11:24 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 28 Jul 2015 10:11:24 -0500
Subject: [Freeswitch-users] how to add mod_event_socket and
 mod_conference activities to the logfile
In-Reply-To: <CAHA5ALeUAOhuQAhJU38mh_qbomY3BAwE_pVPNGm8raUhoQuodg@mail.gmail.com>
References: <CAHA5ALeUAOhuQAhJU38mh_qbomY3BAwE_pVPNGm8raUhoQuodg@mail.gmail.com>
Message-ID: <CAEJMVkCEwTPukLgUENduF5G5XPNJoj-auq+B_=wqdASOAkx7mw@mail.gmail.com>

https://freeswitch.org/jira/browse/FS-6473

We're happy to review patches for this functionality.

/b


On Tue, Jul 28, 2015 at 4:27 AM, Markus von Arx <mkvonarx at gmail.com> wrote:

> Hallo
>
> I'd like to see more details about what's going on in the mod_event_socket
> and in the mod_conference modules in the FreeSWITCH log files. With the
> default (vanilla) configuration, nothing gets logged to the freeswitch.log
> logfile from inside these modules. I scanned all config files and the FS
> confluence wiki, but could not find any hint on how this can be achieved.
>
> The information I'd like to see in the freeswitch.log logfile is e.g. the
> following:
> * state changes in the mod_conference conferences, e.g. conference
> created, conference destroyed, member added, member removed, member muted,
> member unmuted, ... (including affected conference name and member id of
> course)
> * requests, replies & events in the mod_event_socket - if possible
> "shortened" and not including the whole event texts
>
> Any hints how this can be configured?
>
> Thanks a lot,
> 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
>



-- 

*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
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From yadenis at seznam.cz  Tue Jul 28 19:39:18 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Tue, 28 Jul 2015 17:39:18 +0200
Subject: [Freeswitch-users] FreeSwitch and stunnel
In-Reply-To: <CAEJMVkCEwTPukLgUENduF5G5XPNJoj-auq+B_=wqdASOAkx7mw@mail.gmail.com>
References: <CAHA5ALeUAOhuQAhJU38mh_qbomY3BAwE_pVPNGm8raUhoQuodg@mail.gmail.com>
	<CAEJMVkCEwTPukLgUENduF5G5XPNJoj-auq+B_=wqdASOAkx7mw@mail.gmail.com>
Message-ID: <1567939808.20150728173918@seznam.cz>

Hi all,

The question in the following. Somebody succeeded to start freeswitch with stunnel?

I set up stunnel. Allow the rules. SSH is working properly. But when I run frisvitch, he did not even want to start internal profile.

15-07-28 17:30:46.746538 [ERR] sofia.c:2935 Error Creating SIP UA for profile: internal (sip:mod_sofia at 84.242.71.194:61698;maddr=192.168.144.130;transport=udp,tcp) ATTEMPT 3 (RETRY IN 5 SEC)
2015-07-28 17:30:46.746538 [ERR] sofia.c:2945 Error Creating SIP UA for profile: internal (sip:mod_sofia at 84.242.71.194:61698;maddr=192.168.144.130;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.


Does anyone have any experience with this? what am I doing wrong?


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From tru083 at yahoo.com  Tue Jul 28 19:40:04 2015
From: tru083 at yahoo.com (D D)
Date: Tue, 28 Jul 2015 15:40:04 +0000 (UTC)
Subject: [Freeswitch-users] How can I have FS participate in a shared UDP
 multicast session/conference?
Message-ID: <1088312232.4438368.1438098004698.JavaMail.yahoo@mail.yahoo.com>

Hi,
I want to have FS participate in a multicast session with other members.
I see there is the esf_page_group application available, but I think it will constantly send rtp and therefor will monopolize the port.I think I would need to use VAD my call into esf_page_group.? Is this the right approach?? If so, how can I use VAD?
Thanks,David
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From manpower13.cse at gmail.com  Tue Jul 28 19:42:32 2015
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Tue, 28 Jul 2015 21:12:32 +0530
Subject: [Freeswitch-users] IP-Based Authentication
Message-ID: <CAMRuXGjf5avzhPDVa1KFs6pt0Uwt10CasLnxmrKhz=j4V+qDRA@mail.gmail.com>

Hi,


      How i auth agent based on IP Address
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From victor.chukalovskiy at gmail.com  Tue Jul 28 19:52:37 2015
From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy)
Date: Tue, 28 Jul 2015 11:52:37 -0400
Subject: [Freeswitch-users] IP-Based Authentication
In-Reply-To: <CAMRuXGjf5avzhPDVa1KFs6pt0Uwt10CasLnxmrKhz=j4V+qDRA@mail.gmail.com>
References: <CAMRuXGjf5avzhPDVa1KFs6pt0Uwt10CasLnxmrKhz=j4V+qDRA@mail.gmail.com>
Message-ID: <55B7A545.3020301@gmail.com>

Hi, you might want to be more specific with your question.

For incoming calls (FreeSWITCH receiving calls from an IP):

Use Sofia profile param apply-inbound-acl=my_list. Then define agent 
IP(s) directly in the acl list "my_list" that you define in acl.conf.xml
or use Sofia profile param apply-inbound-acl=domains. Then define 
"domains" ACL  in acl.conf.xml pointed to your user directory. Put agent 
IPs as a cidr param in user directory

For outgoing calls (FreeSWITCH sending calls to an IP):

Construct  dial-string that includes agent's IP and use it in the dialplan.
or create a gateway that points to the IP. This way you can also do 
OPTIONS ping in case far-end is down.


On 15-07-28 11:42 AM, Murugan Pandian wrote:
> Hi,
>
>
>       How i auth agent based on IP Address
>
>
> _________________________________________________________________________
> 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 freeswitch.org  Tue Jul 28 20:24:42 2015
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 28 Jul 2015 16:24:42 +0000
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) July
	18th-July 24th
Message-ID: <55b7accabb73f_e7cbc2f3305792a@resque-worker-high.1.mail>

New Post on freeswitch.org from Kathleen King
check it out at http://ift.tt/1U48OzU
FreeSWITCH Week in Review (Master Branch) July 18th-July 24th
Hello, again. This passed week in the FreeSWITCH master branch we had 46 commits. The new features this week are: the addition of?getcputime to retrieve FreeSWITCH process CPU usage, added support for?80 ms, 100 ms, 120 ms packetization to mod_opus, ?and added H.263 codec support to mod_av.

Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support.

New features that were added:

FS-7848 [mod_opus] Add?support for?80 ms, 100 ms, 120 ms packetization

FS-7519 FS-7677 [mod_av] Add H.263 codec support

FS-7885 Add getcputime to retrieve FreeSWITCH process CPU usage

FS-7889 [mod_conference] Move conference chat to use an event channel so messages only go to the right ?room? for the conference and move conference chat functionality to use event_channel.

Improvements in build system, cross platform support, and packaging:

FS-7860 Prevent a switch_rtp header conflict

FS-7130?Make /run/freeswitch persistent, so it will start under systemd

The following bugs were squashed:

FS-7789 [mod_av] Fixed issue with audio dropping out partway through recordings

FS-7854 Add task_runtime to tasks table in core database

FS-7856 [mod_av] Fix some segfaults and leaks.

FS-7866 Fixed a crash when running incorrect var api expansion syntax ?eval ${${external_sip_ip}:4}?

FS-7861 FS-7862 [mod_conference] Fixed a crash and other issues caused by multi canvas feature

FS-7681 [mod_conference] Factor out conference-&gt;canvas and allow per canvas record and play

FS-7869 [mod_conference] Fixed a deadlock on shutdown after playing video file that will not display video

FS-7654 Fixed regressions on eavesdropping on channels playing a file and on channels with unlike rates

FS-7872 [mod_verto] Gracefully fail attempting to transfer 1 legged call

FS-7874 [mod_conference] Fixed incorrect layout group count

FS-7870 [mod_conference] Allow jsonapi commands to pass the string id field to pass special ID?s like ?last?

FS-7882 [mod_conference] Allow JSON API commands to send third arg for muting

FS-7888 [mod_verto] Fixed namespacing problems in javascript library masked by global verto object

FS-7811 Use more common format CIF for blank image

FS-7902 [mod_local_stream] Fix for queue filling up when you have a mix of video and non video files

FS-7891 [mod_spandsp] Allow spandsp dtmf detector to work on rates other than 8k

FS-7839 Correct firefox &gt; 38 DTLS behavior to match new EC requirements

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From brian at freeswitch.org  Tue Jul 28 21:21:28 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 28 Jul 2015 12:21:28 -0500
Subject: [Freeswitch-users] FreeSwitch and stunnel
In-Reply-To: <1567939808.20150728173918@seznam.cz>
References: <CAHA5ALeUAOhuQAhJU38mh_qbomY3BAwE_pVPNGm8raUhoQuodg@mail.gmail.com>
	<CAEJMVkCEwTPukLgUENduF5G5XPNJoj-auq+B_=wqdASOAkx7mw@mail.gmail.com>
	<1567939808.20150728173918@seznam.cz>
Message-ID: <CAEJMVkCqvJqyt7jOzb8-hzUR88Z_2yuRbZpRMqKA2d5CytHj5g@mail.gmail.com>

Why are you trying to use stunnel ?  Sofia is capable of speaking TLS.

On Tue, Jul 28, 2015 at 10:39 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Hi all,
>
> The question in the following. Somebody succeeded to start freeswitch with
> stunnel?
>
> I set up stunnel. Allow the rules. SSH is working properly. But when I run
> frisvitch, he did not even want to start internal profile.
>
> 15-07-28 17:30:46.746538 [ERR] sofia.c:2935 Error Creating SIP UA for
> profile: internal (sip:mod_sofia at 84.242.71.194:61698;maddr=192.168.144.130;transport=udp,tcp)
> ATTEMPT 3 (RETRY IN 5 SEC)
> 2015-07-28 17:30:46.746538 [ERR] sofia.c:2945 Error Creating SIP UA for
> profile: internal (sip:mod_sofia at 84.242.71.194:61698
> ;maddr=192.168.144.130;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.
>
>
> Does anyone have any experience with this? what am I doing wrong?
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 *
>
> _________________________________________________________________________
> 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
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From yadenis at seznam.cz  Tue Jul 28 22:01:23 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Tue, 28 Jul 2015 20:01:23 +0200
Subject: [Freeswitch-users] FreeSwitch and stunnel
In-Reply-To: <CAEJMVkCqvJqyt7jOzb8-hzUR88Z_2yuRbZpRMqKA2d5CytHj5g@mail.gmail.com>
References: <CAHA5ALeUAOhuQAhJU38mh_qbomY3BAwE_pVPNGm8raUhoQuodg@mail.gmail.com>
	<CAEJMVkCEwTPukLgUENduF5G5XPNJoj-auq+B_=wqdASOAkx7mw@mail.gmail.com>
	<1567939808.20150728173918@seznam.cz>
	<CAEJMVkCqvJqyt7jOzb8-hzUR88Z_2yuRbZpRMqKA2d5CytHj5g@mail.gmail.com>
Message-ID: <A2625AEC-42C0-4497-852A-F7995B736084@seznam.cz>

Dobr? den.

At my client the only thing that is allowed is stunnel. I would be happy to use standard tools. But I can not. Banks fairly complex security policy

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382

> On 28. 7. 2015, at 19:21, Brian West <brian at freeswitch.org> wrote:
> 
> Why are you trying to use stunnel ?  Sofia is capable of speaking TLS.
> 
> On Tue, Jul 28, 2015 at 10:39 AM, Denis Jakovlev <yadenis at seznam.cz <mailto:yadenis at seznam.cz>> wrote:
> Hi all,
> 
> The question in the following. Somebody succeeded to start freeswitch with stunnel?
> 
> I set up stunnel. Allow the rules. SSH is working properly. But when I run frisvitch, he did not even want to start internal profile.
> 
> 15-07-28 17:30:46.746538 [ERR] sofia.c:2935 Error Creating SIP UA for profile: internal (sip:mod_sofia at 84.242.71.194:61698;maddr=192.168.144.130;transport=udp,tcp) ATTEMPT 3 (RETRY IN 5 SEC)
> 2015-07-28 17:30:46.746538 [ERR] sofia.c:2945 Error Creating SIP UA for profile: internal (sip:mod_sofia at 84.242.71.194:61698;maddr=192.168.144.130;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.
> 
> 
> Does anyone have any experience with this? what am I doing wrong?
> 
> 
> -- 
> S pozdravem,
> Ing.Denis Jakovlev                           
> mob.tel <http://mob.tel/>. 775-415-382
> 
> 
> _________________________________________________________________________
> 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/>
> 
> 
> 
> -- 
> Brian West
> brian at freeswitch.org <mailto:brian at freeswitch.org>
> 
> Twitter: @FreeSWITCH , @briankwest
> http://www.freeswitchbook.com <http://www.freeswitchbook.com/>
> http://www.freeswitchcookbook.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
> 
> _________________________________________________________________________
> 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 yehavi.bourvine at gmail.com  Tue Jul 28 22:02:03 2015
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 28 Jul 2015 21:02:03 +0300
Subject: [Freeswitch-users] Ad hoc conference
Message-ID: <CAAtWKeFidWLen_RpPJG3YPMu6-XnLUKbG2i6zVCjKh585TqNPQ@mail.gmail.com>

Hello,

It has been some time since the last time I wrote a dialplan for
Freeswitch, so my mind is slightly rusted :-)

I've been asked to do ad-hoc conferencing support for multi users
conference: A calls B, puts him on hold, calls C, connects all together,
put them on hold, call D and connects all together, and so on. This is the
way they want it, like the old PBX we have.

Is there some example of how to do such a thing? if not, I thought of the
following:

- When A presses *1 the first time, a conference room is created and the
other party is transfered to it.
- For the second and others: A initiates a new call, and when connected
press *2 to connect the
  new party as well as A into the conference room,

What I don;t know how to do is: How to take a bridged call, split it into
two legs and move one or both into the conference room. Any idea will help.

        Thanks, __Yehavi:
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From mike at jerris.com  Tue Jul 28 22:19:53 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 28 Jul 2015 14:19:53 -0400
Subject: [Freeswitch-users] Ad hoc conference
In-Reply-To: <CAAtWKeFidWLen_RpPJG3YPMu6-XnLUKbG2i6zVCjKh585TqNPQ@mail.gmail.com>
References: <CAAtWKeFidWLen_RpPJG3YPMu6-XnLUKbG2i6zVCjKh585TqNPQ@mail.gmail.com>
Message-ID: <CAEwTsAj=n-zim0NG3dPoaU3T+ppCu+LyY5NTumFN7AP-KQB0oA@mail.gmail.com>

there are different ways to do this depending on client support.  what
client are you using?

On Tuesday, July 28, 2015, Yehavi Bourvine <yehavi.bourvine at gmail.com>
wrote:

> Hello,
>
> It has been some time since the last time I wrote a dialplan for
> Freeswitch, so my mind is slightly rusted :-)
>
> I've been asked to do ad-hoc conferencing support for multi users
> conference: A calls B, puts him on hold, calls C, connects all together,
> put them on hold, call D and connects all together, and so on. This is the
> way they want it, like the old PBX we have.
>
> Is there some example of how to do such a thing? if not, I thought of the
> following:
>
> - When A presses *1 the first time, a conference room is created and the
> other party is transfered to it.
> - For the second and others: A initiates a new call, and when connected
> press *2 to connect the
>   new party as well as A into the conference room,
>
> What I don;t know how to do is: How to take a bridged call, split it into
> two legs and move one or both into the conference room. Any idea will help.
>
>         Thanks, __Yehavi:
>
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From yehavi.bourvine at gmail.com  Tue Jul 28 22:31:32 2015
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 28 Jul 2015 21:31:32 +0300
Subject: [Freeswitch-users] Ad hoc conference
In-Reply-To: <CAEwTsAj=n-zim0NG3dPoaU3T+ppCu+LyY5NTumFN7AP-KQB0oA@mail.gmail.com>
References: <CAAtWKeFidWLen_RpPJG3YPMu6-XnLUKbG2i6zVCjKh585TqNPQ@mail.gmail.com>
	<CAEwTsAj=n-zim0NG3dPoaU3T+ppCu+LyY5NTumFN7AP-KQB0oA@mail.gmail.com>
Message-ID: <CAAtWKeHrDgcPA--j6aeW4e4Qxz=N9MfZqRVW6oBMLfnhGOBW7Q@mail.gmail.com>

Hi,

  I am using various clients: Polycoms, AudioCodes (both phones and ATAs),
Yealink.
That's why I thought of using * codes, so I can program most of the phones
to send them.

I want it to be phone independent, so it will work on all phones.

        Thanks, __Yehavi:

2015-07-28 21:19 GMT+03:00 Michael Jerris <mike at jerris.com>:

> there are different ways to do this depending on client support.  what
> client are you using?
>
>
> On Tuesday, July 28, 2015, Yehavi Bourvine <yehavi.bourvine at gmail.com>
> wrote:
>
>> Hello,
>>
>> It has been some time since the last time I wrote a dialplan for
>> Freeswitch, so my mind is slightly rusted :-)
>>
>> I've been asked to do ad-hoc conferencing support for multi users
>> conference: A calls B, puts him on hold, calls C, connects all together,
>> put them on hold, call D and connects all together, and so on. This is the
>> way they want it, like the old PBX we have.
>>
>> Is there some example of how to do such a thing? if not, I thought of the
>> following:
>>
>> - When A presses *1 the first time, a conference room is created and the
>> other party is transfered to it.
>> - For the second and others: A initiates a new call, and when connected
>> press *2 to connect the
>>   new party as well as A into the conference room,
>>
>> What I don;t know how to do is: How to take a bridged call, split it into
>> two legs and move one or both into the conference room. Any idea will help.
>>
>>         Thanks, __Yehavi:
>>
>
> _________________________________________________________________________
> 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 blasterjr at gmail.com  Tue Jul 28 22:49:22 2015
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Tue, 28 Jul 2015 12:49:22 -0600
Subject: [Freeswitch-users] IP-Based Authentication
In-Reply-To: <55B7A545.3020301@gmail.com>
References: <CAMRuXGjf5avzhPDVa1KFs6pt0Uwt10CasLnxmrKhz=j4V+qDRA@mail.gmail.com>
	<55B7A545.3020301@gmail.com>
Message-ID: <CAO3hyzHgbHFHEjRP3wnwGLuWS23kFXJ+89c4pG-oz-Syn2yjtQ@mail.gmail.com>

Another option to this is on your user, you can use a CIDR attribute to
bind them to that CIDR, this is useful when dealing with fixed address
customers (like sip trunks)

On Tue, Jul 28, 2015 at 9:52 AM, Victor Chukalovskiy <
victor.chukalovskiy at gmail.com> wrote:

>  Hi, you might want to be more specific with your question.
>
> For incoming calls (FreeSWITCH receiving calls from an IP):
>
> Use Sofia profile param apply-inbound-acl=my_list. Then define agent IP(s)
> directly in the acl list "my_list" that you define in acl.conf.xml
> or use Sofia profile param apply-inbound-acl=domains. Then define
> "domains" ACL  in acl.conf.xml pointed to your user directory. Put agent
> IPs as a cidr param in user directory
>
> For outgoing calls (FreeSWITCH sending calls to an IP):
>
> Construct  dial-string that includes agent's IP and use it in the dialplan.
> or create a gateway that points to the IP. This way you can also do
> OPTIONS ping in case far-end is down.
>
>
>  On 15-07-28 11:42 AM, Murugan Pandian wrote:
>
> Hi,
>
>
>        How i auth agent based on IP Address
>
>
> _________________________________________________________________________
> 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 carlos.ruizdiaz at gmail.com  Wed Jul 29 00:22:04 2015
From: carlos.ruizdiaz at gmail.com (=?UTF-8?Q?Carlos_Ruiz_D=C3=ADaz?=)
Date: Tue, 28 Jul 2015 15:22:04 -0500
Subject: [Freeswitch-users] Unpark call
Message-ID: <CA+3gBcnFg6LdqkZjrp3bC0+Z5rCC26=pbQ+4f+gvh6Y02uzk6Q@mail.gmail.com>

Hi,

I'm looking for a way to unpark a call remotely.

I have this scenario that involves Kamailio as registrar and proxy, and I'm
sending calls to FS whenever an user is busy and can't pick up a call. This
call is parked by FS.

When Kamailio detects the user in question got available, I want to unpark
the previous call and connect it to the original destination.

Any ideas on how to do this in a clean way?

Regards,
-- 
Carlos
http://caruizdiaz.com
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From govoiper at gmail.com  Wed Jul 29 01:30:02 2015
From: govoiper at gmail.com (SamyGo)
Date: Tue, 28 Jul 2015 17:30:02 -0400
Subject: [Freeswitch-users] Unpark call
In-Reply-To: <CA+3gBcnFg6LdqkZjrp3bC0+Z5rCC26=pbQ+4f+gvh6Y02uzk6Q@mail.gmail.com>
References: <CA+3gBcnFg6LdqkZjrp3bC0+Z5rCC26=pbQ+4f+gvh6Y02uzk6Q@mail.gmail.com>
Message-ID: <CAJUJwtj+pVLU-K_U1Ezf_smuE3ryaEgK4gy0HZRBVOktWoi_Cw@mail.gmail.com>

Hi,

Can you tell if FS is involved in all calls or Kamailio dials A and B party
and only sends calls to a parking lot in failure route ?

We did this long time ago with OpenSIPS where A party goes into fake
ringing at a FS server and keeps dialing B party number at OpenSIPS gw every
after 45 or so seconds, if B party is available FS bridges the call, if not
then again stay at the parking lot. You need to set some valet_parking_
variables for timeout and orbit extension.

Given some more details I think I can help you out here.

BR,
Sammy






On Tue, Jul 28, 2015 at 4:22 PM, Carlos Ruiz D?az <carlos.ruizdiaz at gmail.com
> wrote:

> Hi,
>
> I'm looking for a way to unpark a call remotely.
>
> I have this scenario that involves Kamailio as registrar and proxy, and
> I'm sending calls to FS whenever an user is busy and can't pick up a call.
> This call is parked by FS.
>
> When Kamailio detects the user in question got available, I want to unpark
> the previous call and connect it to the original destination.
>
> Any ideas on how to do this in a clean way?
>
> Regards,
> --
> Carlos
> http://caruizdiaz.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
>
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From carlos.ruizdiaz at gmail.com  Wed Jul 29 01:41:14 2015
From: carlos.ruizdiaz at gmail.com (=?UTF-8?Q?Carlos_Ruiz_D=C3=ADaz?=)
Date: Tue, 28 Jul 2015 16:41:14 -0500
Subject: [Freeswitch-users] Unpark call
In-Reply-To: <CAJUJwtj+pVLU-K_U1Ezf_smuE3ryaEgK4gy0HZRBVOktWoi_Cw@mail.gmail.com>
References: <CA+3gBcnFg6LdqkZjrp3bC0+Z5rCC26=pbQ+4f+gvh6Y02uzk6Q@mail.gmail.com>
	<CAJUJwtj+pVLU-K_U1Ezf_smuE3ryaEgK4gy0HZRBVOktWoi_Cw@mail.gmail.com>
Message-ID: <CA+3gBcneq4J-rAQFnG33a1kA_mRRto-R-6VYGU-F4=2QD7BKuw@mail.gmail.com>

On Tue, Jul 28, 2015 at 4:30 PM, SamyGo <govoiper at gmail.com> wrote:

> Hi,
>
> Can you tell if FS is involved in all calls or Kamailio dials A and B
> party and only sends calls to a parking lot in failure route ?
>

Yes. I maintain state of which calls went to FS, and which ones didn't.

I was thinking of using something like a notifier that should be fired from
Kamailio (using http maybe?) and be received by FS somewhere. This would
tell which parking queue to process and later "deflect" or "bridge" the
call again to Kamailio which will forward it to the right agent.

Is this achievable?


>
> We did this long time ago with OpenSIPS where A party goes into fake
> ringing at a FS server and keeps dialing B party number at OpenSIPS gw every
> after 45 or so seconds, if B party is available FS bridges the call, if not
> then again stay at the parking lot. You need to set some valet_parking_
> variables for timeout and orbit extension.
>

This looks like a polling procedure, right? I will have to study valet
parking first to see what I can do with it.

Thank you Sammy!


>
> Given some more details I think I can help you out here.
>
> BR,
> Sammy
>
>
>
>
>
>
> On Tue, Jul 28, 2015 at 4:22 PM, Carlos Ruiz D?az <
> carlos.ruizdiaz at gmail.com> wrote:
>
>> Hi,
>>
>> I'm looking for a way to unpark a call remotely.
>>
>> I have this scenario that involves Kamailio as registrar and proxy, and
>> I'm sending calls to FS whenever an user is busy and can't pick up a call.
>> This call is parked by FS.
>>
>> When Kamailio detects the user in question got available, I want to
>> unpark the previous call and connect it to the original destination.
>>
>> Any ideas on how to do this in a clean way?
>>
>> Regards,
>> --
>> Carlos
>> http://caruizdiaz.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
>>
>
>
> _________________________________________________________________________
> 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
http://caruizdiaz.com
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From jack at livecall.com  Wed Jul 29 01:53:34 2015
From: jack at livecall.com (Jack)
Date: Tue, 28 Jul 2015 14:53:34 -0700
Subject: [Freeswitch-users] Implementing <Form> Tag of VXML in
 Freeswtich using Javascript
In-Reply-To: <61E1B4F8DE845A4F986F448F6D50A38504BE869EC2@E2K7CCR03.corp.oocl.com>
References: <61E1B4F8DE845A4F986F448F6D50A38504BE869D67@E2K7CCR03.corp.oocl.com>
	<CAEJMVkAy=jUh2JHM9enqH4wkiUO4tyxG6Fy76zJ7VdiqW=1BCA@mail.gmail.com>
	<61E1B4F8DE845A4F986F448F6D50A38504BE869E7C@E2K7CCR03.corp.oocl.com>
	<7516D1CA-D0B1-4A31-9088-604BD48FF276@gmail.com>
	<61E1B4F8DE845A4F986F448F6D50A38504BE869EC2@E2K7CCR03.corp.oocl.com>
Message-ID: <55B7F9DE.9040004@livecall.com>

I think the approach I would take would be to build a stand alone server 
on top of MiniHttpd , assign it a port and then receive my POSTs there, 
process them,  then send the api commands from there into freeswitch.
Jack


On 7/27/2015 8:23 PM, david.fu at oocl.com wrote:
>
> Hi Chris,
>
> Thanks Chris for the advice. You ?re right that we may need a lot of 
> effort to turn FS into a voice Brower. Hence,  I am thinking how to 
> turn the returned VXML in our existing application server into 
> something that FS can interpret. For example, rewrite the code in our 
> existing application to return XML/Javascript instead of VXML to FS. 
>   However, I don?t know how to implement <SUBMIT> in Javascript or XML 
> according to the Javascript reference site at 
> https://wiki.freeswitch.org/wiki/Category:Javascript.  Would you 
> please give me some hints ?
>
> *_Existing:_*
>
> IVR Voice Browser(Send HTTP request) <-- --> Application server(return 
> VXML to the voice browser of IVR). The VXML mainly stores callflow of 
> each hotline.
>
> *_Future:_*
>
> FS(Send HTTP request) <-- --> Application server(return Javascript/XML 
> to the dialplan of FS)
>
> Best Regards,
>
> David
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *cmrienzo at gmail.com
> *Sent:* Tuesday, July 28, 2015 11:00 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Implementing <Form> Tag of VXML in 
> Freeswtich using Javascript
>
> If you have the voice browser source code you could try and port it to 
> control FS over event socket.  Otherwise, you're in for a lot of work 
> to turn FS into a voice browser from scratch.
>
> Chris
>
>
> On Jul 27, 2015, at 22:00, <david.fu at oocl.com 
> <mailto:david.fu at oocl.com>> <david.fu at oocl.com 
> <mailto:david.fu at oocl.com>> wrote:
>
>     Hi Brian,
>
>     Thanks for your prompt reply so much.  Actually, we have an
>     existing VXML application. The voice browser sends HTTP request to
>     the Application server, which returns VXML to the voice browser in
>     IVR server. Now, we would like to migrate it to Freeswitch.  As we
>     used <FORM> and <SUBMIT> tag in VXML, we are searching how to
>     implement this in Freeswitch to send request to Application
>     Server, and then return XML or Javascript instead of VXML to
>     Freeswtich. Would you please give us some hints ? Thanks.
>
>     Best Regards,
>
>     David
>
>     *From:*freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>     [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf
>     Of *Brian West
>     *Sent:* Monday, July 27, 2015 9:53 PM
>     *To:* FreeSWITCH Users Help
>     *Subject:* Re: [Freeswitch-users] Implementing <Form> Tag of VXML
>     in Freeswtich using Javascript
>
>     There is no support for VXML in FreeSWITCH, We have all the parts
>     and I'm sure it could probably be implemented, but its never been
>     a high priority item for us as nobody asks for it.
>
>     On Mon, Jul 27, 2015 at 2:25 AM, <david.fu at oocl.com
>     <mailto:david.fu at oocl.com>> wrote:
>
>     Dear Freeswitch experts,
>
>     I would like to implement something like <Form> and <submit> tag
>     of VXML in Freeswtich using Javascript. The purpose is to sending
>     request back to the application server from Freeswtich.  However, 
>     I couldn?t find the related API(s) in the Freeswitch official web
>     site. Would you please help advise ? Thank you so much.
>
>     Yours faithfully,
>
>     David
>
>
>
>     IMPORTANT NOTICE
>     Email from OOCL is confidential and may be legally privileged.  If
>     it is not
>     intended for you, please delete it immediately unread.  The internet
>     cannot guarantee that this communication is free of viruses,
>     interception
>     or interference and anyone who communicates with us by email is taken
>     to accept the risks in doing so.  Without limitation, OOCL and its
>     affiliates
>     accept no liability whatsoever and howsoever arising in connection
>     with
>     the use of this email.  Under no circumstances shall this email
>     constitute
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>     which is subject to the availability of carrier's equipment and
>     vessels and
>     the terms and conditions of OOCL's standard bill of lading which
>     is also
>     available at http://www.oocl.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
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>     -- 
>
>     */Brian West/*
>     brian at freeswitch.org <mailto: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
>
>
>
>     IMPORTANT NOTICE
>     Email from OOCL is confidential and may be legally privileged.  If
>     it is not
>     intended for you, please delete it immediately unread. The internet
>     cannot guarantee that this communication is free of viruses,
>     interception
>     or interference and anyone who communicates with us by email is taken
>     to accept the risks in doing so.  Without limitation, OOCL and its
>     affiliates
>     accept no liability whatsoever and howsoever arising in connection
>     with
>     the use of this email.  Under no circumstances shall this email
>     constitute
>     a binding agreement to carry or for provision of carriage services
>     by OOCL,
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From govoiper at gmail.com  Wed Jul 29 02:34:00 2015
From: govoiper at gmail.com (SamyGo)
Date: Tue, 28 Jul 2015 18:34:00 -0400
Subject: [Freeswitch-users] Unpark call
In-Reply-To: <CA+3gBcneq4J-rAQFnG33a1kA_mRRto-R-6VYGU-F4=2QD7BKuw@mail.gmail.com>
References: <CA+3gBcnFg6LdqkZjrp3bC0+Z5rCC26=pbQ+4f+gvh6Y02uzk6Q@mail.gmail.com>
	<CAJUJwtj+pVLU-K_U1Ezf_smuE3ryaEgK4gy0HZRBVOktWoi_Cw@mail.gmail.com>
	<CA+3gBcneq4J-rAQFnG33a1kA_mRRto-R-6VYGU-F4=2QD7BKuw@mail.gmail.com>
Message-ID: <CAJUJwtiTw8LGtjD4VVZU953N4F4Euq13RgAiacGLCpFdrAMnEA@mail.gmail.com>

my idea in line.

On Tue, Jul 28, 2015 at 5:41 PM, Carlos Ruiz D?az <carlos.ruizdiaz at gmail.com
> wrote:

> On Tue, Jul 28, 2015 at 4:30 PM, SamyGo <govoiper at gmail.com> wrote:
>
>> Hi,
>>
>> Can you tell if FS is involved in all calls or Kamailio dials A and B
>> party and only sends calls to a parking lot in failure route ?
>>
>
> Yes. I maintain state of which calls went to FS, and which ones didn't.
>
> I was thinking of using something like a notifier that should be fired
> from Kamailio (using http maybe?) and be received by FS somewhere. This
> would tell which parking queue to process and later "deflect" or "bridge"
> the call again to Kamailio which will forward it to the right agent.
>
> Is this achievable?
>
Yes. I beleive if you've the uuid of the call stored in a redis hash
corresponding to the SIP Call-ID header , and you tell you maintain state
of which calls sent ..so as soon as B party gets diconnected you can
trigger an ESL/API command to FS to "bridge" that uuid with a particular
destination_number.


>
>>
>> We did this long time ago with OpenSIPS where A party goes into fake
>> ringing at a FS server and keeps dialing B party number at OpenSIPS gw
>> every after 45 or so seconds, if B party is available FS bridges the call,
>> if not then again stay at the parking lot. You need to set some
>> valet_parking_ variables for timeout and orbit extension.
>>
>
> This looks like a polling procedure, right? I will have to study valet
> parking first to see what I can do with it.
>

Yes exactly what it is, that how valet_parking_timeout works. Its upto you
whichever way you like to go with.

>
> Thank you Sammy!
>
>
>>
>> Given some more details I think I can help you out here.
>>
>> BR,
>> Sammy
>>
>>
>>
>>
>>
>>
>> On Tue, Jul 28, 2015 at 4:22 PM, Carlos Ruiz D?az <
>> carlos.ruizdiaz at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> I'm looking for a way to unpark a call remotely.
>>>
>>> I have this scenario that involves Kamailio as registrar and proxy, and
>>> I'm sending calls to FS whenever an user is busy and can't pick up a call.
>>> This call is parked by FS.
>>>
>>> When Kamailio detects the user in question got available, I want to
>>> unpark the previous call and connect it to the original destination.
>>>
>>> Any ideas on how to do this in a clean way?
>>>
>>> Regards,
>>> --
>>> Carlos
>>> http://caruizdiaz.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
>>>
>>
>>
>> _________________________________________________________________________
>> 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
> http://caruizdiaz.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
>
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From govoiper at gmail.com  Wed Jul 29 02:46:48 2015
From: govoiper at gmail.com (SamyGo)
Date: Tue, 28 Jul 2015 18:46:48 -0400
Subject: [Freeswitch-users] Unpark call
In-Reply-To: <CAJUJwtiTw8LGtjD4VVZU953N4F4Euq13RgAiacGLCpFdrAMnEA@mail.gmail.com>
References: <CA+3gBcnFg6LdqkZjrp3bC0+Z5rCC26=pbQ+4f+gvh6Y02uzk6Q@mail.gmail.com>
	<CAJUJwtj+pVLU-K_U1Ezf_smuE3ryaEgK4gy0HZRBVOktWoi_Cw@mail.gmail.com>
	<CA+3gBcneq4J-rAQFnG33a1kA_mRRto-R-6VYGU-F4=2QD7BKuw@mail.gmail.com>
	<CAJUJwtiTw8LGtjD4VVZU953N4F4Euq13RgAiacGLCpFdrAMnEA@mail.gmail.com>
Message-ID: <CAJUJwtjFyKPwviOc-0aNr7uS1d81v8Zz5q2tSbJgsMz6wOqL9w@mail.gmail.com>

*Stage-1 Kamailio*
You'll have three things at Kamailio
1-  Call-ID and
2-  FS Server IP hosting this parking lot
3-  Destination Number which was busy.

store 1,and 2 in redis hash with Call-ID as Key. Store 3 at a redis hash
"TriggerUnPark"

*Stage-2: FreeSwitch*

At FS the first thing you do is get the sip header Call-ID, and update the
redis Call-ID Hash with the FS uuid of this call.

*Stage-3: Kamailio if(is_method("BYE")) {*
check the redis hash "TriggerUnPark" if the $tU is found in there,
if yes delete it from the hash and execute a small lua script "unpark.lua"
with Call-ID as its argument
*}*


*Stage-4:*
the unpark.lua will retrieve the Call-ID, find the relevant FS Server IP,
connect to its ESL layer, and originate a call towards the Kamailio Server
for the destination number.

If destination number is available call gets bridged in the lua script else
it drops back to the parking lot and Stage-1 repeats.


I just wrote this as I could possibly think of how this would work, so I
imagine there must be easier and efficient ways to do this all.

Regards,
Sammy


On Tue, Jul 28, 2015 at 6:34 PM, SamyGo <govoiper at gmail.com> wrote:

> my idea in line.
>
> On Tue, Jul 28, 2015 at 5:41 PM, Carlos Ruiz D?az <
> carlos.ruizdiaz at gmail.com> wrote:
>
>> On Tue, Jul 28, 2015 at 4:30 PM, SamyGo <govoiper at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> Can you tell if FS is involved in all calls or Kamailio dials A and B
>>> party and only sends calls to a parking lot in failure route ?
>>>
>>
>> Yes. I maintain state of which calls went to FS, and which ones didn't.
>>
>> I was thinking of using something like a notifier that should be fired
>> from Kamailio (using http maybe?) and be received by FS somewhere. This
>> would tell which parking queue to process and later "deflect" or "bridge"
>> the call again to Kamailio which will forward it to the right agent.
>>
>> Is this achievable?
>>
> Yes. I beleive if you've the uuid of the call stored in a redis hash
> corresponding to the SIP Call-ID header , and you tell you maintain state
> of which calls sent ..so as soon as B party gets diconnected you can
> trigger an ESL/API command to FS to "bridge" that uuid with a particular
> destination_number.
>
>
>>
>>>
>>> We did this long time ago with OpenSIPS where A party goes into fake
>>> ringing at a FS server and keeps dialing B party number at OpenSIPS gw
>>> every after 45 or so seconds, if B party is available FS bridges the call,
>>> if not then again stay at the parking lot. You need to set some
>>> valet_parking_ variables for timeout and orbit extension.
>>>
>>
>> This looks like a polling procedure, right? I will have to study valet
>> parking first to see what I can do with it.
>>
>
> Yes exactly what it is, that how valet_parking_timeout works. Its upto you
> whichever way you like to go with.
>
>>
>> Thank you Sammy!
>>
>>
>>>
>>> Given some more details I think I can help you out here.
>>>
>>> BR,
>>> Sammy
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Tue, Jul 28, 2015 at 4:22 PM, Carlos Ruiz D?az <
>>> carlos.ruizdiaz at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> I'm looking for a way to unpark a call remotely.
>>>>
>>>> I have this scenario that involves Kamailio as registrar and proxy, and
>>>> I'm sending calls to FS whenever an user is busy and can't pick up a call.
>>>> This call is parked by FS.
>>>>
>>>> When Kamailio detects the user in question got available, I want to
>>>> unpark the previous call and connect it to the original destination.
>>>>
>>>> Any ideas on how to do this in a clean way?
>>>>
>>>> Regards,
>>>> --
>>>> Carlos
>>>> http://caruizdiaz.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
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> 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
>> http://caruizdiaz.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
>>
>
>
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From carlos.ruizdiaz at gmail.com  Wed Jul 29 02:54:48 2015
From: carlos.ruizdiaz at gmail.com (=?UTF-8?Q?Carlos_Ruiz_D=C3=ADaz?=)
Date: Tue, 28 Jul 2015 17:54:48 -0500
Subject: [Freeswitch-users] Unpark call
In-Reply-To: <CAJUJwtjFyKPwviOc-0aNr7uS1d81v8Zz5q2tSbJgsMz6wOqL9w@mail.gmail.com>
References: <CA+3gBcnFg6LdqkZjrp3bC0+Z5rCC26=pbQ+4f+gvh6Y02uzk6Q@mail.gmail.com>
	<CAJUJwtj+pVLU-K_U1Ezf_smuE3ryaEgK4gy0HZRBVOktWoi_Cw@mail.gmail.com>
	<CA+3gBcneq4J-rAQFnG33a1kA_mRRto-R-6VYGU-F4=2QD7BKuw@mail.gmail.com>
	<CAJUJwtiTw8LGtjD4VVZU953N4F4Euq13RgAiacGLCpFdrAMnEA@mail.gmail.com>
	<CAJUJwtjFyKPwviOc-0aNr7uS1d81v8Zz5q2tSbJgsMz6wOqL9w@mail.gmail.com>
Message-ID: <CA+3gBckH8RQK8Pphrw9VODeW7oTRdLgYORsxsTFomJUHyqXy-g@mail.gmail.com>

Thank you very much Sammy.

Some of the alternatives you suggested crossed my mind already, but I lack
the required FS experience to implement them so I'm getting it now :).

I will let you know what solution I ended up using so that everyone could
benefit from it.

Have a nice day.
Carlos


On Tue, Jul 28, 2015 at 5:46 PM, SamyGo <govoiper at gmail.com> wrote:

> *Stage-1 Kamailio*
> You'll have three things at Kamailio
> 1-  Call-ID and
> 2-  FS Server IP hosting this parking lot
> 3-  Destination Number which was busy.
>
> store 1,and 2 in redis hash with Call-ID as Key. Store 3 at a redis hash
> "TriggerUnPark"
>
> *Stage-2: FreeSwitch*
>
> At FS the first thing you do is get the sip header Call-ID, and update the
> redis Call-ID Hash with the FS uuid of this call.
>
> *Stage-3: Kamailio if(is_method("BYE")) {*
> check the redis hash "TriggerUnPark" if the $tU is found in there,
> if yes delete it from the hash and execute a small lua script "unpark.lua"
> with Call-ID as its argument
> *}*
>
>
> *Stage-4:*
> the unpark.lua will retrieve the Call-ID, find the relevant FS Server IP,
> connect to its ESL layer, and originate a call towards the Kamailio Server
> for the destination number.
>
> If destination number is available call gets bridged in the lua script
> else it drops back to the parking lot and Stage-1 repeats.
>
>
> I just wrote this as I could possibly think of how this would work, so I
> imagine there must be easier and efficient ways to do this all.
>
> Regards,
> Sammy
>
>
> On Tue, Jul 28, 2015 at 6:34 PM, SamyGo <govoiper at gmail.com> wrote:
>
>> my idea in line.
>>
>> On Tue, Jul 28, 2015 at 5:41 PM, Carlos Ruiz D?az <
>> carlos.ruizdiaz at gmail.com> wrote:
>>
>>> On Tue, Jul 28, 2015 at 4:30 PM, SamyGo <govoiper at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> Can you tell if FS is involved in all calls or Kamailio dials A and B
>>>> party and only sends calls to a parking lot in failure route ?
>>>>
>>>
>>> Yes. I maintain state of which calls went to FS, and which ones didn't.
>>>
>>> I was thinking of using something like a notifier that should be fired
>>> from Kamailio (using http maybe?) and be received by FS somewhere. This
>>> would tell which parking queue to process and later "deflect" or "bridge"
>>> the call again to Kamailio which will forward it to the right agent.
>>>
>>> Is this achievable?
>>>
>> Yes. I beleive if you've the uuid of the call stored in a redis hash
>> corresponding to the SIP Call-ID header , and you tell you maintain state
>> of which calls sent ..so as soon as B party gets diconnected you can
>> trigger an ESL/API command to FS to "bridge" that uuid with a particular
>> destination_number.
>>
>>
>>>
>>>>
>>>> We did this long time ago with OpenSIPS where A party goes into fake
>>>> ringing at a FS server and keeps dialing B party number at OpenSIPS gw
>>>> every after 45 or so seconds, if B party is available FS bridges the call,
>>>> if not then again stay at the parking lot. You need to set some
>>>> valet_parking_ variables for timeout and orbit extension.
>>>>
>>>
>>> This looks like a polling procedure, right? I will have to study valet
>>> parking first to see what I can do with it.
>>>
>>
>> Yes exactly what it is, that how valet_parking_timeout works. Its upto
>> you whichever way you like to go with.
>>
>>>
>>> Thank you Sammy!
>>>
>>>
>>>>
>>>> Given some more details I think I can help you out here.
>>>>
>>>> BR,
>>>> Sammy
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Tue, Jul 28, 2015 at 4:22 PM, Carlos Ruiz D?az <
>>>> carlos.ruizdiaz at gmail.com> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> I'm looking for a way to unpark a call remotely.
>>>>>
>>>>> I have this scenario that involves Kamailio as registrar and proxy,
>>>>> and I'm sending calls to FS whenever an user is busy and can't pick up a
>>>>> call. This call is parked by FS.
>>>>>
>>>>> When Kamailio detects the user in question got available, I want to
>>>>> unpark the previous call and connect it to the original destination.
>>>>>
>>>>> Any ideas on how to do this in a clean way?
>>>>>
>>>>> Regards,
>>>>> --
>>>>> Carlos
>>>>> http://caruizdiaz.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
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>>> http://caruizdiaz.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
>>>
>>
>>
>
> _________________________________________________________________________
> 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
http://caruizdiaz.com
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From kamil.nigmatullin at gmail.com  Wed Jul 29 11:06:21 2015
From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin)
Date: Wed, 29 Jul 2015 13:06:21 +0600
Subject: [Freeswitch-users] [ERR] sofia_glue.c:313 Invalid tls-verify-policy
	value: none
Message-ID: <CA+XtiedweUqHZwNtvYrTus79XS4g96-5hCGpUkq=FwbUYFwSRQ@mail.gmail.com>

After fresh installation of 4.20 I keep getting this erroe sometimes
2015-07-29 08:54:37.376432 [ERR] sofia_glue.c:313 Invalid tls-verify-policy
value: none


Can I fix it somehow?

Thank you

-- 
Kamil Nigmatullin
Tel: 77272323748
mob: 7 (707) 2517003
Skype: kamil.nigmatullin
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From adam.lappe at qsc.de  Wed Jul 29 11:28:29 2015
From: adam.lappe at qsc.de (Lappe, Adam)
Date: Wed, 29 Jul 2015 07:28:29 +0000
Subject: [Freeswitch-users] mod_local_stream moh confusion
In-Reply-To: <CAJR7f2E6aMK_do1g0S9SF1y5eyFt77Vh8AjemWpQRcfyh=r8KA@mail.gmail.com>
References: <2E67ADAE1D3582409C90EBAE8C64C5070E5569@QSCDEMXP01B.ONE4ALL.LAN>
	<CAEJMVkAnYcxOzL19E_Zs9Bn73zx8gAtGuTCYGZPrqsFhVb6n3g@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C5070E58E0@QSCDEMXP01B.ONE4ALL.LAN>
	<2E67ADAE1D3582409C90EBAE8C64C5070E60E4@QSCDEMXP01B.ONE4ALL.LAN>
	<CAJR7f2FccPRC7Acpzhjx1CbF=dAxyunt3XrQre7j1bC9xQAkTA@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C5070E6140@QSCDEMXP01B.ONE4ALL.LAN>
	<CAJR7f2E6aMK_do1g0S9SF1y5eyFt77Vh8AjemWpQRcfyh=r8KA@mail.gmail.com>
Message-ID: <2E67ADAE1D3582409C90EBAE8C64C5070E629E@QSCDEMXP01B.ONE4ALL.LAN>

Thanks, this works!!!

fs_cli -x 'global_setvar temp_hold_music=silence'

Now I can hear hold music and the call does not drop.
The next question ist:
Why does
<action application="set" data="temp_hold_music=silence"/>
not work?

Shouldn?t it be the same result?

Best regards
Adam

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von ?talo Rossi
Gesendet: Dienstag, 28. Juli 2015 16:37
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] mod_local_stream moh confusion

You'll probably need to export it, or set it globally like fs_cli -x 'global_setvar temp_hold_music=silence'

local_stream://moh is set in switch_core_media if temp_hold_music and hold_music aren't set

On Tue, Jul 28, 2015 at 11:14 AM, Lappe, Adam <adam.lappe at qsc.de<mailto:adam.lappe at qsc.de>> wrote:
Hi,

thanks for your suggestions:

I neither have
<X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
nor
<param name="hold-music" value="$${hold_music}"/>
in my config file. I also don?t have vars.xml or X-PRE-PROCESSES but 1 large freeswitch.xml.

When I get transferred I receive a RE-INVITE with a=sendonly media attribute in SDP.
Then FreeSWITCH tries to use local_stream://moh (which is not configured anywhere)!!

Also I tried
<action application="set" data="temp_hold_music=silence"/>
or
<action application="set" data=" temp_hold_music= (path)/blub.wav "/>
but this does not work.

How can I disable the local_stream://moh here?
Any other suggestions?

Thanks,
Adam

Von: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] Im Auftrag von ?talo Rossi
Gesendet: Dienstag, 28. Juli 2015 14:39

An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] mod_local_stream moh confusion

https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-MusicOnHoldRelated

On Tue, Jul 28, 2015 at 6:53 AM, Lappe, Adam <adam.lappe at qsc.de<mailto:adam.lappe at qsc.de>> wrote:
Other Question:

how can i set a other file to use for moh.

I make an outgoing call and set the moh path to a wav file. This file is only used to play it to the callee, if I set him  on hold.
Now, when the callee transfers my call, I receive a RE-INVITE and my FreeSWITCH tries to play a music on hold using the mod_local_stream.

Again the question: why does FreeSWITCH try to play a moh und why does he try to use mod_local_stream?
Can I change local_stream to another wav file somehow?

Thanks in advance,
Adam

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] Im Auftrag von Lappe, Adam
Gesendet: Freitag, 24. Juli 2015 12:30
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] mod_local_stream moh confusion

Hello Brian,

thanks for your answer.

You are correct. As you can see in my first post, i do not use mod_local_stream.
The question is: WHY does FreeSWITCH try to use a local_stream for moh, when a user gets transfered (with an reinvite)?

I grep'ed my config files, i don't load or use mod_local_stream anywhere.

Best regards,
Adam
________________________________________
Von: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>]&quot; im Auftrag von &quot;Brian West [brian at freeswitch.org<mailto:brian at freeswitch.org>]
Gesendet: Donnerstag, 23. Juli 2015 17:09
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] mod_local_stream moh confusion

It would indicate that you do not have mod_local_stream loaded.

On Thu, Jul 23, 2015 at 3:15 AM, Lappe, Adam <adam.lappe at qsc.de<mailto:adam.lappe at qsc.de><mailto:adam.lappe at qsc.de<mailto:adam.lappe at qsc.de>>> wrote:
Hello,

i am testing the latest 1.4.19 Version of FreeSWITCH.
Currently we are running an old 1.2.7 Version.

Everything seems to work fine, but there is 1 error that is very confusing:

When a call gets transfered by the callee (i.e. by the receptionist) the call will be terminated.
All I see is this error line:
[ERR] switch_core_file.c:149 Invalid file format [local_stream] for [moh]!

I don't use the local_stream module.
freeswitch at internal> module_exists mod_local_stream false

This error does not exists with the old version.

Is this a bug, or am I missing something?

Thanks in advance,
Adam

_________________________________________________________________________
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--

Brian West
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Got Bugs? Report them here<https://freeswitch.org/jira>! | Reddit: /r/freeswitch<https://www.reddit.com/r/freeswitch>

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_________________________________________________________________________
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--
?talo Rossi

_________________________________________________________________________
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From sendmeallyouroffers at googlemail.com  Wed Jul 29 12:43:09 2015
From: sendmeallyouroffers at googlemail.com (Berthold Karl)
Date: Wed, 29 Jul 2015 10:43:09 +0200
Subject: [Freeswitch-users] freeswitch never seems to call sendmail when
 trying to email voicemails
In-Reply-To: <CAN6oSBsVjND7y_m=APj+QRr3yvfT=9hPLxbietUnAsdiiid8dA@mail.gmail.com>
References: <54E6B9A1.20708@dickson.st>
	<CAHtxdDd8CRENTyNoi7FOn3yuy+fyCshEiknfhvhZcg36q=JFuA@mail.gmail.com>
	<54EA5B78.3010504@dickson.st>
	<CAHtxdDcD0fqTWR9mA8=-eT9rm2a1M1p3Qab7OVXCfH=wYnHvvw@mail.gmail.com>
	<54EBE76A.2040709@dickson.st>
	<CAHtxdDfsm2pd8HDM-WwU8HKJT5AQj3isqjWF4aWVbXkRrtB0zg@mail.gmail.com>
	<CAEJMVkBY-3nOfXijxhED_byC_K5cNnS8LVW3iRjGmOFserDmjw@mail.gmail.com>
	<CAN6oSBuxXbf=uecMk1UO=ZP9xxpvoXA65Ox4pyZVmPxA-BNwnQ@mail.gmail.com>
	<CA+Xtiefu7T+LBFD11k38wPG-QCPcrWJuaAaLGTOmS_RTmk8m7A@mail.gmail.com>
	<29646517.20150727173302@seznam.cz>
	<C18BE15B-86F3-40C1-ABED-CA328565C0C1@mgtech.com>
	<CAN6oSBt4VqjViDaXdkund+Ch9W=5JF__n55sOx5PVXYdAn3Zgw@mail.gmail.com>
	<CAN6oSBsVjND7y_m=APj+QRr3yvfT=9hPLxbietUnAsdiiid8dA@mail.gmail.com>
Message-ID: <CAN6oSBsfixEAjMe=PyCn8_9SoKsyqniu=hrhdpqTe2tiPO_4=w@mail.gmail.com>

Hi,

can't find Informations about how to disable the MWI.

So I've build a workaournd.

When a voicemail is left, the script sends the recording to the user and
than deletes the voicemail via API. The api command vm_delete deletes the
Recording and also the MWI :)
$api->executeString("vm_delete $voicemail_user\@$voicemail_domain $uuid");

perl hangup_hook is real fun.

:)

Regards


2015-07-28 10:40 GMT+02:00 Berthold Karl <sendmeallyouroffers at googlemail.com
>:

> Hi,
>
> so I wrote a hangup hook script... I'am not able to get sendmail running
> with freeswitch.
>
> I wrote a perl script which is called as hangup_hook.
> Thanks to $env, its serves all the informations I needed.
> After some testings and playing with hangup_hook I've added some more
> features.
>
>
> Only one Problem left. How can I tell FS, thats MWI is not needed.
>
> Regards
>
> 2015-07-28 9:19 GMT+02:00 Berthold Karl <
> sendmeallyouroffers at googlemail.com>:
>
>> Thanks for the replies!
>>
>> I've tested sendmail, it works for the user which is running freeswitch.
>>
>> I'am running FS 1.4.8 under Debian.
>>
>> I want to use the FS internal VM-to-E-Mail because all informations about
>> To-, From- etc are configured in the user.xml. If I'am running my own
>> script, I have to setup everything in a second configuration or have to
>> parse the user.xml a second time..
>>
>> Thx Dennis for the Sample script,
>>
>> Any ideas?
>> I've tested sendmail with few arguments... everytime its ends with a
>> segfault... It seems FS isn't creating the Mail under /tmp/. /tmp is World
>> writable/readable.
>>
>> Regards
>>
>>
>>
>> 2015-07-27 19:01 GMT+02:00 Mario <mario_fs at mgtech.com>:
>>
>>> Although this is in the Mac OS X page I wrote this up which may help:
>>> https://freeswitch.org/confluence/display/FREESWITCH/Installation+and+Setup+on+OS+X#InstallationandSetuponOSX-EmailVoicemailtoaniPhone
>>>
>>> To test your email you can do this from a command line:
>>> printf "Subject: TestnHello" | sendmail -f you at domain.com you at domain.com
>>> Mareio G
>>>
>>> On Jul 27, 2015, at 8:33 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
>>>
>>>  Hi,
>>>
>>> Why do not use some simple script after the ends session?
>>>
>>> For example simple bash script send_voicemail.sh
>>>
>>> #!/bin/sh
>>> cd /usr/local/freeswitch/recordings/voicemail/
>>> f=`find -name \*.wav`
>>> for file in $f
>>> do
>>> echo "Processing ${file}"
>>> CURRENT=$(date +%d.%m.%y_%H:%M:%S)
>>> datetime=$CURRENT
>>> echo $datetime
>>> sendemail -f voicemail at mail.com -t somemail at mail.com -m "Voicemail" -u
>>> "Voicemail od $datetime" -a ${file} -s smtp.mail.com -xu user -xp pass
>>> echo "BackUP ${file}"
>>> mv ${file} /usr/local/freeswitch/recordings/backup_voicemail/
>>> done
>>>
>>>
>>> And in dialplan start like this
>>> <action application="export" data="api_hangup_hook=system
>>> /usr/local/freeswitch/scripts/send_voicemail.sh"/>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>>> <http://mob.tel>. 775-415-382 pond?l? 27. ?ervence 2015, 17:11:10, napsal
>>> jste: *
>>>
>>>  Why can.t you create your own script and use it to send vm over email?
>>> 27 ???? 2015 ?. 18:32 ???????????? "Berthold Karl" <
>>> sendmeallyouroffers at googlemail.com> ???????:
>>> Hi,
>>>
>>> my FS is calling the sendmail-command, but sendmail ends with an
>>> segfault. It seems like there is no email under /tmp/. Did your sendmail
>>> also ends in a segfault?
>>>
>>> 2015-02-24 16:51 GMT+01:00 Brian West <brian at freeswitch.org>:
>>> When you file the JIRA please attach your voicemail.conf.xml.
>>>
>>> On Tue, Feb 24, 2015 at 8:21 AM, Sergey Safarov <s.safarov at gmail.com>
>>> wrote:
>>> If you want to FS correctly processed parameters vm-mailfrom and
>>> email-from, write a request to https://freeswitch.org/jira/
>>>
>>> On Tue, Feb 24, 2015 at 5:52 AM, Jason Lewis <jason at dickson.st> wrote:
>>> Thanks Sergey,
>>>
>>> I had configured the domain variable to the fqdn of the machine. I
>>> eventually got it working though, I was missing two key lines from my user
>>> config:
>>>      <param name="vm-email-all-messages" value="true"/>
>>>      <param name="vm-attach-file" value="true" />
>>>
>>> I'm not sure how I managed to miss those but anyway, that seems to have
>>> resolved things.
>>>
>>> It seems as though vm-mailfrom is still being ignored though. Currently
>>> I have it set to:
>>>
>>>       <param name="vm-mailfrom" value="vm at freeswitch.xyz.com.au"
>>> <vm at freeswitch.xyz.com.au>/>
>>>
>>> but voicemails get delivered from:
>>>
>>> 1001 at freeswitch.xyz.com.au
>>>
>>> Is this worth investigating further?
>>>
>>> Jason
>>>
>>> Sergey Safarov wrote on 23/02/2015 5:05 PM:
>>> Try configure "domain" variable in vars.xml
>>>
>>> <X-PRE-PROCESS cmd="set" data="domain=you_domain_name"/>
>>>
>>> After it verify that user registered with domain name
>>>
>>> freeswitch at internal> sofia status profile internal reg
>>>
>>> Registrations:
>>>
>>> =================================================================================================
>>> Call-ID:    1B26-2327-466848134BEBC9719CDE-002 at SipHost
>>> User:       1201 at you_domain_name
>>> Contact:    "1201" <
>>> sip:1201 at 10.21.18.22:5060;fs_nat=yes;fs_path=sip%3A1201%4010.21.18.22%3A5060
>>> >
>>> Agent:      204 12-3868-2416-0.10.56.1-DS
>>> Status:     Registered(UDP-NAT)(unknown) EXP(2015-02-23 06:05:22)
>>> EXPSECS(139)
>>> Ping-Status:Reachable
>>> Host:       fs1.you_domain_name
>>> IP:         10.21.18.22
>>> Port:       5060
>>> Auth-User:  1201
>>> Auth-Realm: you_domain_name
>>> MWI-Account:1201 at you_domain_name
>>>
>>> Sergey
>>>
>>>
>>> On Mon, Feb 23, 2015 at 1:43 AM, Jason Lewis <jason at dickson.st
>>> > wrote:
>>>
>>> So I've managed to see some output from the sendmail program in the FS
>>> logs. It appears that my fs instance isn't correctly setting its domain?
>>>
>>> the FS box has a fqdn, and I also set the domain parameter in the
>>> vars.xml file, but still the voicemail is sent with a from address of an IP
>>> address.
>>>
>>> Any ideas?
>>>
>>> Net::SMTP>>> Net::SMTP(2.33)
>>> Net::SMTP>>>   Net::Cmd(2.30)
>>> Net::SMTP>>>     Exporter(5.71)
>>> Net::SMTP>>>   IO::Socket::INET(1.35)
>>> Net::SMTP>>>     IO::Socket(1.37)
>>> Net::SMTP>>>       IO::Handle(1.35)
>>> Net::SMTP=GLOB(0x23f7748)<<< 220 mb.xyz.com.au ESMTP Postfix
>>> (Debian/GNU)
>>> Net::SMTP=GLOB(0x23f7748)>>> EHLO localhost.localdomain
>>> Net::SMTP=GLOB(0x23f7748)<<< 250-mb.bongalong.st
>>>
>>> Net::SMTP=GLOB(0x23f7748)<<< 250-PIPELINING
>>> Net::SMTP=GLOB(0x23f7748)<<< 250-SIZE 10240000
>>> Net::SMTP=GLOB(0x23f7748)<<< 250-VRFY
>>> Net::SMTP=GLOB(0x23f7748)<<< 250-ETRN
>>> Net::SMTP=GLOB(0x23f7748)<<< 250-STARTTLS
>>> Net::SMTP=GLOB(0x23f7748)<<< 250-ENHANCEDSTATUSCODES
>>> Net::SMTP=GLOB(0x23f7748)<<< 250-8BITMIME
>>> Net::SMTP=GLOB(0x23f7748)<<< 250 DSN
>>> Net::SMTP=GLOB(0x23f7748)>>> MAIL FROM:<1002 at 192.168.1.3>
>>> <1002 at 192.168.1.3>
>>> Net::SMTP=GLOB(0x23f7748)<<< 501 5.1.7 Bad sender address syntax
>>> Net::SMTP=GLOB(0x23f7748)>>> RCPT TO:<jason at xyz.com.au>
>>> <jason at xyz.com.au>
>>> Net::SMTP=GLOB(0x23f7748)<<< 503 5.5.1 Error: need MAIL command
>>> Net::SMTP=GLOB(0x23f7748)>>> DATA
>>> Net::SMTP=GLOB(0x23f7748)<<< 503 5.5.1 Error: need RCPT command
>>> Net::SMTP=GLOB(0x23f7748)>>> QUIT
>>> Net::SMTP=GLOB(0x23f7748)<<< 221 2.0.0 Bye
>>> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1633 Session 3
>>> (loopback/voicemail-a) Ended
>>> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1637 Close
>>> Channel loopback/voicemail-a [CS_DESTROY]
>>> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1633 Session 4
>>> (loopback/voicemail-b) Ended
>>> 2015-02-22 11:08:14.145852 [NOTICE] switch_core_session.c:1637 Close
>>> Channel loopback/voicemail-b [CS_DESTROY]
>>>
>>>
>>> On 20/02/2015 7:10 pm, Sergey Safarov wrote:
>>> You mailer is not understand "mailer-app-args" has been configured in
>>> "switch.conf.xml"
>>> Remove extra arg or add required
>>>
>>> ??, 20 ????. 2015, 7:49, Jason Lewis <jason at dickson.st>:
>>> Hi,
>>>
>>> I've been trying to make freeswitch email voicemails but as far as I can
>>> tell, it never even calls sendmail.
>>>
>>> I have setting mailer-app to "sendmail" and "/usr/sbin/sendmail" to no
>>> avail. I can successfully send an email from the commandline using
>>> sendmail. (sendmail in this case is provided by postfix)
>>>
>>> I see no emails in the postfix mail logs when I leave a voicemail
>>> message.
>>>
>>> I also tried creating a shell just to see if it even gets called from
>>> fs, but it does not get called when a voicemail is deposited:
>>> #!/bin/bash
>>> echo $(date --rfc-3339=ns): $* >> /tmp/freeswitchsendmail.log
>>>
>>> After every change, I have run reloadxml and reload mod_voicemail. I
>>> have also tried restarting freeswitch.
>>>
>>> I am running the debian packages of FreeSWITCH Version 1.4.15-1~64bit
>>> (-1 64bit)
>>>
>>> my configuration is based on the vanilla configuration with only very
>>> minor changes.
>>>
>>> I'm at a loss as to how to debug further, but I'm pretty sure the
>>> mailer-app is never called. Is there some setting I'm missing or
>>> something obvious I'm not doing?
>>>
>>>
>>> My config:
>>> 1001.xml:
>>> <include>
>>>  <user id="1001">
>>>    <params>
>>>      <param name="password" value="$${default_password}"/>
>>>      <param name="vm-password" value="1001"/>
>>>      <param name="vm-mailto" value="jason at xyz.com.au"/>
>>>      <param name="vm-mailfrom" value="vm at freeswitch.xyz.com.au"/>
>>>      <!-- <param name="vm-message-ext" value="wav"/> -->
>>>    </params>
>>>    <variables>
>>>      <variable name="toll_allow" value="domestic,international,local"/>
>>>      <variable name="accountcode" value="1001"/>
>>>      <variable name="user_context" value="default"/>
>>>      <variable name="effective_caller_id_name" value="Extension 1001"/>
>>>      <variable name="effective_caller_id_number" value="1001"/>
>>>      <variable name="outbound_caller_id_name"
>>> value="$${outbound_caller_name}"/>
>>>      <variable name="outbound_caller_id_number"
>>> value="$${outbound_caller_id}"/>
>>>      <variable name="callgroup" value="techsupport"/>
>>>    </variables>
>>>  </user>
>>> </include>
>>>
>>> and in switch.conf.xml I have the following set:
>>>
>>>    <param name="mailer-app" value="/usr/sbin/sendmail"/>
>>>    <param name="mailer-app-args" value="-t"/>
>>>
>>>
>>> I made a log at level 7 and put it on the pastebin:
>>>
>>> https://pastebin.freeswitch.org/23921
>>>
>>>
>>> Jason Lewis
>>> http://emacstragic.net
>>>
>>>
>>> _________________________________________________________________________
>>> 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
>>>
>>>
>>> --
>>> Jason Lewis
>>> http://emacstragic.net
>>>
>>> _________________________________________________________________________
>>> 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
>>>
>>> --
>>> Jason Lewis
>>> http://emacstragic.net
>>>
>>> _________________________________________________________________________
>>> 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 *brian at freeswitch.org
>>>
>>> *Twitter: @FreeSWITCH , @briankwest *http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>> _________________________________________________________________________
>>> 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 alex at digitalmail.com  Wed Jul 29 15:42:08 2015
From: alex at digitalmail.com (Alex Lake)
Date: Wed, 29 Jul 2015 12:42:08 +0100
Subject: [Freeswitch-users] Complex phrase macro
Message-ID: <55B8BC10.9000002@digitalmail.com>

I'd like a macro that can play multiple sound files from a specific 
directory.

eg. here's one that should (though I've not tested it!) play 3 sound files:

       phrase "xplay3:/home/pabx/004-3774/x,001,002,003"

     <macro name="xplay3">
       <input pattern="(.*),(.*),(.*),(.*)">
         <match>
           <action function="play-file" data="$1/$2.wav"/>
           <action function="play-file" data="$1/$3.wav"/>
           <action function="play-file" data="$1/$4.wav"/>
         </match>
       </input>
     </macro>

What I'd like to do is one that can play an arbitrary number of files.


From alex at digitalmail.com  Wed Jul 29 15:46:58 2015
From: alex at digitalmail.com (Alex Lake)
Date: Wed, 29 Jul 2015 12:46:58 +0100
Subject: [Freeswitch-users] Complex phrase macro
In-Reply-To: <55B8BC10.9000002@digitalmail.com>
References: <55B8BC10.9000002@digitalmail.com>
Message-ID: <55B8BD32.1040708@digitalmail.com>

BTW - Is something like this possible (using recursion):

     <macro name="xplayN">
         <input pattern="(.*),(.*),(.*)">
             <match>
                 <action function="play-file" data="$1/$2.wav"/>
                 <action function="play-file" data="phrase:xplayN:$1,$3"/>
             </match>
         </input>
         <input pattern="(.*),(.*)">
             <match>
                 <action function="play-file" data="$1/$2.wav"/>
             </match>
         </input>
     </macro>


On 29/07/2015 12:42, Alex Lake wrote:
> I'd like a macro that can play multiple sound files from a specific
> directory.
>
> eg. here's one that should (though I've not tested it!) play 3 sound files:
>
>         phrase "xplay3:/home/pabx/004-3774/x,001,002,003"
>
>       <macro name="xplay3">
>         <input pattern="(.*),(.*),(.*),(.*)">
>           <match>
>             <action function="play-file" data="$1/$2.wav"/>
>             <action function="play-file" data="$1/$3.wav"/>
>             <action function="play-file" data="$1/$4.wav"/>
>           </match>
>         </input>
>       </macro>
>
> What I'd like to do is one that can play an arbitrary number of files.
>
> _________________________________________________________________________
> 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 petedao at gmail.com  Wed Jul 29 16:37:52 2015
From: petedao at gmail.com (Pete Kay)
Date: Wed, 29 Jul 2015 05:37:52 -0700
Subject: [Freeswitch-users] recording ring tong
Message-ID: <CAPxXe8WxxQpP03aFAHbZs76-5CONW1iWva+7Kh-QvO9XusphaQ@mail.gmail.com>

Hi

I am trying to record the incoming ring tone.  Does anyone know how to do
that?

The record app does not seem to be able to record ring tone.

Thanks,
Pete
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From alex at digitalmail.com  Wed Jul 29 17:06:34 2015
From: alex at digitalmail.com (Alex Lake)
Date: Wed, 29 Jul 2015 14:06:34 +0100
Subject: [Freeswitch-users] Complex phrase macro
In-Reply-To: <55B8BC10.9000002@digitalmail.com>
References: <55B8BC10.9000002@digitalmail.com>
Message-ID: <55B8CFDA.6000001@digitalmail.com>

OK - Got this working. Just in case anyone is interested:

     <macro name="xplayN">
         <input pattern="^([^,]*),([^,]*),(.*)$">
             <match>
                 <action function="play-file" data="$1/$2.wav"/>
                 <action function="play-file" data="phrase:xplayN:$1,$3"/>
             </match>
         </input>
         <input pattern="^([^,]*),([^,]*)$">
             <match>
                 <action function="play-file" data="$1/$2.wav"/>
             </match>
         </input>
     </macro>

Example of use:

       <action application="playback" 
data="phrase:xplayN:/home/pabx/004-3774/x,001,0211,0221,0231"/>




From gmaruzz at gmail.com  Wed Jul 29 17:13:03 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 29 Jul 2015 15:13:03 +0200
Subject: [Freeswitch-users] Complex phrase macro
In-Reply-To: <55B8CFDA.6000001@digitalmail.com>
References: <55B8BC10.9000002@digitalmail.com>
	<55B8CFDA.6000001@digitalmail.com>
Message-ID: <CALXCt0osZ=zoeJ+2U1r7jqOR_Aw_twoBk2iSGEvRc6j9-AxPtw@mail.gmail.com>

clever!

On Wed, Jul 29, 2015 at 3:06 PM, Alex Lake <alex at digitalmail.com> wrote:

> OK - Got this working. Just in case anyone is interested:
>
>      <macro name="xplayN">
>          <input pattern="^([^,]*),([^,]*),(.*)$">
>              <match>
>                  <action function="play-file" data="$1/$2.wav"/>
>                  <action function="play-file" data="phrase:xplayN:$1,$3"/>
>              </match>
>          </input>
>          <input pattern="^([^,]*),([^,]*)$">
>              <match>
>                  <action function="play-file" data="$1/$2.wav"/>
>              </match>
>          </input>
>      </macro>
>
> Example of use:
>
>        <action application="playback"
> data="phrase:xplayN:/home/pabx/004-3774/x,001,0211,0221,0231"/>
>
>
>
> _________________________________________________________________________
> 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
Cell : +39-347-2665618
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From italorossib at gmail.com  Wed Jul 29 18:21:39 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Wed, 29 Jul 2015 11:21:39 -0300
Subject: [Freeswitch-users] mod_local_stream moh confusion
In-Reply-To: <2E67ADAE1D3582409C90EBAE8C64C5070E629E@QSCDEMXP01B.ONE4ALL.LAN>
References: <2E67ADAE1D3582409C90EBAE8C64C5070E5569@QSCDEMXP01B.ONE4ALL.LAN>
	<CAEJMVkAnYcxOzL19E_Zs9Bn73zx8gAtGuTCYGZPrqsFhVb6n3g@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C5070E58E0@QSCDEMXP01B.ONE4ALL.LAN>
	<2E67ADAE1D3582409C90EBAE8C64C5070E60E4@QSCDEMXP01B.ONE4ALL.LAN>
	<CAJR7f2FccPRC7Acpzhjx1CbF=dAxyunt3XrQre7j1bC9xQAkTA@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C5070E6140@QSCDEMXP01B.ONE4ALL.LAN>
	<CAJR7f2E6aMK_do1g0S9SF1y5eyFt77Vh8AjemWpQRcfyh=r8KA@mail.gmail.com>
	<2E67ADAE1D3582409C90EBAE8C64C5070E629E@QSCDEMXP01B.ONE4ALL.LAN>
Message-ID: <CAJR7f2Ha7h2cZOvBzB_6Av8fY1DngFxVuJRRRH5XkfR--pq=LA@mail.gmail.com>

On Wed, Jul 29, 2015 at 4:28 AM, Lappe, Adam <adam.lappe at qsc.de> wrote:

>  Thanks, this works!!!
>
>
>
> fs_cli -x 'global_setvar temp_hold_music=silence'
>
>
>
> Now I can hear hold music and the call does not drop.
>
> The next question ist:
>
> Why does
>
> <action application="set" data="temp_hold_music=silence"/>
>
> not work?
>

>
> Shouldn?t it be the same result?
>

No, because you're only setting it to your A-leg, this way your B-leg
doesn't have the var set. You need to export it (application="export"
data="temp_hold_music=silence") or set it globally like you did (it'll
affect all legs).


>
>
> Best regards
>
> Adam
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *?talo
> Rossi
> *Gesendet:* Dienstag, 28. Juli 2015 16:37
>
> *An:* FreeSWITCH Users Help
> *Betreff:* Re: [Freeswitch-users] mod_local_stream moh confusion
>
>
>
> You'll probably need to export it, or set it globally like fs_cli -x
> 'global_setvar temp_hold_music=silence'
>
>
>
> local_stream://moh is set in switch_core_media if temp_hold_music and
> hold_music aren't set
>
>
>
> On Tue, Jul 28, 2015 at 11:14 AM, Lappe, Adam <adam.lappe at qsc.de> wrote:
>
> Hi,
>
>
>
> thanks for your suggestions:
>
>
>
> I neither have
>
> <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
>
> nor
>
> <param name="hold-music" value="$${hold_music}"/>
>
> in my config file. I also don?t have vars.xml or X-PRE-PROCESSES but 1
> large freeswitch.xml.
>
>
>
> When I get transferred I receive a RE-INVITE with *a=sendonly* media
> attribute in SDP.
>
> Then FreeSWITCH tries to use local_stream://moh (which is not configured
> anywhere)!!
>
>
>
> Also I tried
>
> <action application="set" data="temp_hold_music=silence"/>
>
> or
>
> <action application="set" data=" temp_hold_music= (path)/blub.wav "/>
>
> but this does not work.
>
>
>
> How can I disable the local_stream://moh here?
>
> Any other suggestions?
>
>
>
> Thanks,
>
> Adam
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *?talo
> Rossi
> *Gesendet:* Dienstag, 28. Juli 2015 14:39
>
>
> *An:* FreeSWITCH Users Help
> *Betreff:* Re: [Freeswitch-users] mod_local_stream moh confusion
>
>
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-MusicOnHoldRelated
>
>
>
> On Tue, Jul 28, 2015 at 6:53 AM, Lappe, Adam <adam.lappe at qsc.de> wrote:
>
> Other Question:
>
> how can i set a other file to use for moh.
>
> I make an outgoing call and set the moh path to a wav file. This file is
> only used to play it to the callee, if I set him  on hold.
> Now, when the callee transfers my call, I receive a RE-INVITE and my
> FreeSWITCH tries to play a music on hold using the mod_local_stream.
>
> Again the question: why does FreeSWITCH try to play a moh und why does he
> try to use mod_local_stream?
> Can I change local_stream to another wav file somehow?
>
> Thanks in advance,
> Adam
>
> -----Urspr?ngliche Nachricht-----
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Lappe, Adam
> Gesendet: Freitag, 24. Juli 2015 12:30
>
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] mod_local_stream moh confusion
>
> Hello Brian,
>
> thanks for your answer.
>
> You are correct. As you can see in my first post, i do not use
> mod_local_stream.
> The question is: WHY does FreeSWITCH try to use a local_stream for moh,
> when a user gets transfered (with an reinvite)?
>
> I grep'ed my config files, i don't load or use mod_local_stream anywhere.
>
> Best regards,
> Adam
> ________________________________________
> Von: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org]&quot; im Auftrag von
> &quot;Brian West [brian at freeswitch.org]
> Gesendet: Donnerstag, 23. Juli 2015 17:09
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] mod_local_stream moh confusion
>
> It would indicate that you do not have mod_local_stream loaded.
>
> On Thu, Jul 23, 2015 at 3:15 AM, Lappe, Adam <adam.lappe at qsc.de<mailto:
> adam.lappe at qsc.de>> wrote:
> Hello,
>
> i am testing the latest 1.4.19 Version of FreeSWITCH.
> Currently we are running an old 1.2.7 Version.
>
> Everything seems to work fine, but there is 1 error that is very confusing:
>
> When a call gets transfered by the callee (i.e. by the receptionist) the
> call will be terminated.
> All I see is this error line:
> [ERR] switch_core_file.c:149 Invalid file format [local_stream] for [moh]!
>
> I don't use the local_stream module.
> freeswitch at internal> module_exists mod_local_stream false
>
> This error does not exists with the old version.
>
> Is this a bug, or am I missing something?
>
> Thanks in advance,
> Adam
>
> _________________________________________________________________________
> 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
>
>
>
> --
>
> Brian West
> brian at freeswitch.org<mailto:brian at freeswitch.org>
>
> [http://billing.freeswitch.org/templates/default/img/whmcslogo.png]
>
> 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
>
>
> _________________________________________________________________________
> 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
>
>
>
>
>
> --
>
> ?talo Rossi
>
>
> _________________________________________________________________________
> 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
>
>
>
>
>
> --
>
> ?talo Rossi
>
> _________________________________________________________________________
> 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
>



-- 
?talo Rossi
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From mike at jerris.com  Wed Jul 29 18:28:32 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 29 Jul 2015 10:28:32 -0400
Subject: [Freeswitch-users] recording ring tong
In-Reply-To: <CAPxXe8WxxQpP03aFAHbZs76-5CONW1iWva+7Kh-QvO9XusphaQ@mail.gmail.com>
References: <CAPxXe8WxxQpP03aFAHbZs76-5CONW1iWva+7Kh-QvO9XusphaQ@mail.gmail.com>
Message-ID: <CAEwTsAgXBuB7ivUzfau_2SZaBhyCY5hAeqd0cpzDRg-cGUyaBQ@mail.gmail.com>

Sometimes a ring tone is a tone that is played in band as audio, other
times it is just signal of indication of ringing, and your end device
generates its own tone for you to hear. In the latter case recording would
not capture it, because there is no tone to capture.

On Wednesday, July 29, 2015, Pete Kay <petedao at gmail.com> wrote:

> Hi
>
> I am trying to record the incoming ring tone.  Does anyone know how to do
> that?
>
> The record app does not seem to be able to record ring tone.
>
> Thanks,
> Pete
>
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From grcamauer at gmail.com  Wed Jul 29 18:52:19 2015
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Wed, 29 Jul 2015 11:52:19 -0300
Subject: [Freeswitch-users] G729 both in hardware and bypass
Message-ID: <CAPNTnnopFtvY5MTBxE1ovPo4UUupwvo_rZaQ3YftdGt16yS89A@mail.gmail.com>

OK, this is probably not a usual setup, but I would like to know if it is
possible:

I have a FS server with 2 IP trunks configured.  It also has a Digium
hardware transcoding card which I use for 120 sessions of G729.

I would like to set FS up in such a way that one of the IP trunks uses G729
in bypass mode.  It is an outbound IVR and all voice prompts are
pre-recorded in G729.

I would like that the other trunk use the Digium transcoding hardware.  Is
there any way to configure this?


Thanks in advance,
-- 
Guillermo Ruiz Camauer
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From ssinyagin at gmail.com  Wed Jul 29 21:02:38 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 29 Jul 2015 19:02:38 +0200
Subject: [Freeswitch-users] Complex phrase macro
In-Reply-To: <55B8CFDA.6000001@digitalmail.com>
References: <55B8BC10.9000002@digitalmail.com>
	<55B8CFDA.6000001@digitalmail.com>
Message-ID: <CACOYK=g+foyFRheKrPfB1iwLt+snBp1NERAQwpt1q5r_DUDrEQ@mail.gmail.com>

Why not just using Lua instead?
 On Jul 29, 2015 3:08 PM, "Alex Lake" <alex at digitalmail.com> wrote:

> OK - Got this working. Just in case anyone is interested:
>
>      <macro name="xplayN">
>          <input pattern="^([^,]*),([^,]*),(.*)$">
>              <match>
>                  <action function="play-file" data="$1/$2.wav"/>
>                  <action function="play-file" data="phrase:xplayN:$1,$3"/>
>              </match>
>          </input>
>          <input pattern="^([^,]*),([^,]*)$">
>              <match>
>                  <action function="play-file" data="$1/$2.wav"/>
>              </match>
>          </input>
>      </macro>
>
> Example of use:
>
>        <action application="playback"
> data="phrase:xplayN:/home/pabx/004-3774/x,001,0211,0221,0231"/>
>
>
>
> _________________________________________________________________________
> 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  Wed Jul 29 21:16:17 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 29 Jul 2015 13:16:17 -0400
Subject: [Freeswitch-users] Complex phrase macro
In-Reply-To: <CACOYK=g+foyFRheKrPfB1iwLt+snBp1NERAQwpt1q5r_DUDrEQ@mail.gmail.com>
References: <55B8BC10.9000002@digitalmail.com>
	<55B8CFDA.6000001@digitalmail.com>
	<CACOYK=g+foyFRheKrPfB1iwLt+snBp1NERAQwpt1q5r_DUDrEQ@mail.gmail.com>
Message-ID: <C0109557-B6AB-45C9-841D-035F865ACF05@jerris.com>

Or even dialplan. When would this actually add any value over maybe just calling playback multiple times in a row?

> On Jul 29, 2015, at 1:02 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> 
> Why not just using Lua instead?
> On Jul 29, 2015 3:08 PM, "Alex Lake" <alex at digitalmail.com <mailto:alex at digitalmail.com>> wrote:
> OK - Got this working. Just in case anyone is interested:
> 
>      <macro name="xplayN">
>          <input pattern="^([^,]*),([^,]*),(.*)$">
>              <match>
>                  <action function="play-file" data="$1/$2.wav"/>
>                  <action function="play-file" data="phrase:xplayN:$1,$3"/>
>              </match>
>          </input>
>          <input pattern="^([^,]*),([^,]*)$">
>              <match>
>                  <action function="play-file" data="$1/$2.wav"/>
>              </match>
>          </input>
>      </macro>
> 
> Example of use:
> 
>        <action application="playback"
> data="phrase:xplayN:/home/pabx/004-3774/x,001,0211,0221,0231"/>
> 
> 
> 
> _________________________________________________________________________
> 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
> 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 agoulis at opensips.org  Wed Jul 29 22:08:37 2015
From: agoulis at opensips.org (Alex Goulis)
Date: Wed, 29 Jul 2015 13:08:37 -0500
Subject: [Freeswitch-users] It's not too late to register for the OpenSIPS
 Workshop @Cluecon Monday August 3, 2015
In-Reply-To: <CAKbxfG8hNhDouktyOE=DdFdbP=EMZVvcVWeZ2tVakikibd66+g@mail.gmail.com>
References: <CAKbxfG8hNhDouktyOE=DdFdbP=EMZVvcVWeZ2tVakikibd66+g@mail.gmail.com>
Message-ID: <55B916A5.70409@opensips.org>

There's still room for the OpenSIPS Workshop @ClueCon.

The OpenSIPS Workshop is a new format that will focus more on providing 
attendees with more of a technical approach to integrating OpenSIPS 
rather than case studies - the sessions will provide working setups 
(with explanations) for OpenSIPS in different scenarios . The Workshops 
will cover new OpenSIPS release and features, FreeSwitch integration , 
Edge Proxy setup, Asynchronous I/O support and Fraud prevention.

The Official schedule can be found at:
http://www.opensips.org/Community/Workshop-2015Chicago-Schedule

But we're happy to announce the following workshop leaders:

Razvan Crainea - OpenSIPS project
Dan Christian Bogos - ITsysCOM
Vlad Paiu - OpenSIPS Project
Eric Tamme - OnSIP
Alex Goulis - Ratetel
Pete Kelly - SourceVox

Whether you are a beginner or a seasoned professional, the OpenSIPS 
Workshop @ClueCon has something for you.

For beginners, get answers to questions like:

* how to extend, expand and enhance FreeSwitch with OpenSIPS
* intelligent load balancing of freeswitch servers
* why OpenSIPS is the right choice for your network
* Building a calling card solution using OpenSIPS and Freeswitch

Seasoned professionals can:

* Learn about new modules and features for the 2.x release
* Discuss advanced topics
* Discuss the development roadmap for OpenSIPS

Once we've wrapped up the workshop, head over to the Hackathon with the 
OpenSIPS Team.


The registration fees (per person) are:

     $199 for those attending only the OpenSIPS Workshop
     $149 for those attending ClueCon as well or having a discount code 
(*CLUECON2015*)  or are registering 3 or more participants

Space is limited, so don't wait too long to register.

More information can be found at:
http://www.opensips.org/Community/Workshop-2015Chicago

We're looking forward to seeing you in Chicago!

- The OpenSIPS Team
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From vbvbrj at gmail.com  Wed Jul 29 22:25:01 2015
From: vbvbrj at gmail.com (Mimiko)
Date: Wed, 29 Jul 2015 21:25:01 +0300
Subject: [Freeswitch-users] FS and T.38 fax
Message-ID: <55B91A7D.3070403@gmail.com>

Hello.

I'm on FS 1.3.13b_cad607d72e. I have two D-Link VoIP gateways DVG-5008S 
and DVG-2024. Also there are two Canon MFP with faxes. Each DVG have on 
one port connected a Canon MFP. The FXS ports are configured to use T.38 
for fax. Sending fax from one MFP to another starts, but ends with 
error. Switching ports to use T.30 allows to send and receive fax 
between MFPs.

Doesn't this version of FS support T.38? Does latest version support? Or 
I miss something in configuration?


-- 
Mimiko desu.


From gmaruzz at gmail.com  Wed Jul 29 22:30:23 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 29 Jul 2015 20:30:23 +0200
Subject: [Freeswitch-users] FS and T.38 fax
In-Reply-To: <55B91A7D.3070403@gmail.com>
References: <55B91A7D.3070403@gmail.com>
Message-ID: <CALXCt0o2oQpCCYs25Kv2MTWqzTfcad6MYeD+Uw4sr3JCc0P+Bg@mail.gmail.com>

upgrade to 1.4.x (currently 1.4.20), your version is very old, and no more
supported...

1.4.x fully supports t38


On Wed, Jul 29, 2015 at 8:25 PM, Mimiko <vbvbrj at gmail.com> wrote:

> Hello.
>
> I'm on FS 1.3.13b_cad607d72e. I have two D-Link VoIP gateways DVG-5008S
> and DVG-2024. Also there are two Canon MFP with faxes. Each DVG have on
> one port connected a Canon MFP. The FXS ports are configured to use T.38
> for fax. Sending fax from one MFP to another starts, but ends with
> error. Switching ports to use T.30 allows to send and receive fax
> between MFPs.
>
> Doesn't this version of FS support T.38? Does latest version support? Or
> I miss something in configuration?
>
>
> --
> Mimiko desu.
>
> _________________________________________________________________________
> 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
Cell : +39-347-2665618
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From vbvbrj at gmail.com  Wed Jul 29 22:35:36 2015
From: vbvbrj at gmail.com (Mimiko)
Date: Wed, 29 Jul 2015 21:35:36 +0300
Subject: [Freeswitch-users] FS and T.38 fax
In-Reply-To: <CALXCt0o2oQpCCYs25Kv2MTWqzTfcad6MYeD+Uw4sr3JCc0P+Bg@mail.gmail.com>
References: <55B91A7D.3070403@gmail.com>
	<CALXCt0o2oQpCCYs25Kv2MTWqzTfcad6MYeD+Uw4sr3JCc0P+Bg@mail.gmail.com>
Message-ID: <55B91CF8.7090408@gmail.com>

On 29.07.2015 21:30, Giovanni Maruzzelli wrote:
> 1.4.x fully supports t38

Oh, is it indeed true? I've read about some multipart of SDP is not 
supported by FS. If latest version supports T.38 without problems, I'll 
consider to upgrade. I need to setup another server with Jessie (am on 
Squeezy now) and recompile FS.

Thank you. :)

-- 
Mimiko desu.


From gmaruzz at gmail.com  Wed Jul 29 22:37:47 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 29 Jul 2015 20:37:47 +0200
Subject: [Freeswitch-users] FS and T.38 fax
In-Reply-To: <55B91CF8.7090408@gmail.com>
References: <55B91A7D.3070403@gmail.com>
	<CALXCt0o2oQpCCYs25Kv2MTWqzTfcad6MYeD+Uw4sr3JCc0P+Bg@mail.gmail.com>
	<55B91CF8.7090408@gmail.com>
Message-ID: <CALXCt0pmi13OXBNLMTA-11cnp0d9BFceNnmkXyyfGXAywoe9zQ@mail.gmail.com>

1.4 does not require jessie
1.6 requires jessie
You dont need 1.6 for t38

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On Jul 29, 2015 8:36 PM, "Mimiko" <vbvbrj at gmail.com> wrote:

> On 29.07.2015 21:30, Giovanni Maruzzelli wrote:
> > 1.4.x fully supports t38
>
> Oh, is it indeed true? I've read about some multipart of SDP is not
> supported by FS. If latest version supports T.38 without problems, I'll
> consider to upgrade. I need to setup another server with Jessie (am on
> Squeezy now) and recompile FS.
>
> Thank you. :)
>
> --
> Mimiko desu.
>
> _________________________________________________________________________
> 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 vbvbrj at gmail.com  Wed Jul 29 22:40:32 2015
From: vbvbrj at gmail.com (Mimiko)
Date: Wed, 29 Jul 2015 21:40:32 +0300
Subject: [Freeswitch-users] FS and T.38 fax
In-Reply-To: <CALXCt0pmi13OXBNLMTA-11cnp0d9BFceNnmkXyyfGXAywoe9zQ@mail.gmail.com>
References: <55B91A7D.3070403@gmail.com>
	<CALXCt0o2oQpCCYs25Kv2MTWqzTfcad6MYeD+Uw4sr3JCc0P+Bg@mail.gmail.com>
	<55B91CF8.7090408@gmail.com>
	<CALXCt0pmi13OXBNLMTA-11cnp0d9BFceNnmkXyyfGXAywoe9zQ@mail.gmail.com>
Message-ID: <55B91E20.3050005@gmail.com>

On 29.07.2015 21:37, Giovanni Maruzzelli wrote:
> 1.4 does not require jessie
> 1.6 requires jessie
> You dont need 1.6 for t38

Giovanni, I want to move to Jessie anyway. I didn't follow FS relizes 
history for two years. Latest stable is 1.6 already?

-- 
Mimiko desu.


From gmaruzz at gmail.com  Wed Jul 29 22:44:30 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 29 Jul 2015 20:44:30 +0200
Subject: [Freeswitch-users] FS and T.38 fax
In-Reply-To: <55B91E20.3050005@gmail.com>
References: <55B91A7D.3070403@gmail.com>
	<CALXCt0o2oQpCCYs25Kv2MTWqzTfcad6MYeD+Uw4sr3JCc0P+Bg@mail.gmail.com>
	<55B91CF8.7090408@gmail.com>
	<CALXCt0pmi13OXBNLMTA-11cnp0d9BFceNnmkXyyfGXAywoe9zQ@mail.gmail.com>
	<55B91E20.3050005@gmail.com>
Message-ID: <CALXCt0peNumvOiP8row1aFAjxLpMesFJSCzPB3EgV77Amftb4w@mail.gmail.com>

1.6 is not yet released, albeit it will probably be very soon.

1.6 has a lot of features for videoconferencing and webrtc.

many people use 1.6 in production

-giovanni

On Wed, Jul 29, 2015 at 8:40 PM, Mimiko <vbvbrj at gmail.com> wrote:

> On 29.07.2015 21:37, Giovanni Maruzzelli wrote:
> > 1.4 does not require jessie
> > 1.6 requires jessie
> > You dont need 1.6 for t38
>
> Giovanni, I want to move to Jessie anyway. I didn't follow FS relizes
> history for two years. Latest stable is 1.6 already?
>
> --
> Mimiko desu.
>
> _________________________________________________________________________
> 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
Cell : +39-347-2665618
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From vbvbrj at gmail.com  Wed Jul 29 22:54:05 2015
From: vbvbrj at gmail.com (Mimiko)
Date: Wed, 29 Jul 2015 21:54:05 +0300
Subject: [Freeswitch-users] FS and T.38 fax
In-Reply-To: <CALXCt0peNumvOiP8row1aFAjxLpMesFJSCzPB3EgV77Amftb4w@mail.gmail.com>
References: <55B91A7D.3070403@gmail.com>
	<CALXCt0o2oQpCCYs25Kv2MTWqzTfcad6MYeD+Uw4sr3JCc0P+Bg@mail.gmail.com>
	<55B91CF8.7090408@gmail.com>
	<CALXCt0pmi13OXBNLMTA-11cnp0d9BFceNnmkXyyfGXAywoe9zQ@mail.gmail.com>
	<55B91E20.3050005@gmail.com>
	<CALXCt0peNumvOiP8row1aFAjxLpMesFJSCzPB3EgV77Amftb4w@mail.gmail.com>
Message-ID: <55B9214D.5000709@gmail.com>

On 29.07.2015 21:44, Giovanni Maruzzelli wrote:
> many people use 1.6 in production

When I compiled 1.3.13b it also was latest version from git.

Thank you Giovanni.

-- 
Mimiko desu.


From mandra at gmail.com  Wed Jul 29 23:21:51 2015
From: mandra at gmail.com (Chris Mandra)
Date: Wed, 29 Jul 2015 15:21:51 -0400
Subject: [Freeswitch-users] webrtc/Sip issue
Message-ID: <CAJdRzN6Nmr1v9p4_Yg3saxPmsi6m5Ke1boEoWCZR+eC-7JsNFQ@mail.gmail.com>

Hi guys - one of my users keeps getting this error in  fs_cli  when he
tries to connect via Chrome (web-rtc) on a macbook running yosemite.

2015-07-28 00:42:20.669463 [ERR] switch_core_media.c:6538 AUDIO RTP REPORTS
ERROR: [Remote Address Error!]

I'm using essentially the same setup and never have an issue. Any ideas? He
doesn't get anything very useful in the chrome console, just this:

*********************

Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | closing
INVITE session edsmilh98mlqt0a48qo0tvmlnm0u3m

sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) |
sip.invitecontext.mediahandler | closing PeerConnection

sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.dialog |
dialog edsmilh98mlqt0a48qo0qf3p9f1n97g6QD4ZtFHXgrH deleted

sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) |
sip.inviteclientcontext | emitting event failed

sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) |
sip.inviteclientcontext | emitting event rejected

sc-webrtc.js:931 _rejected called

*******************


Any ideas? We're using latest FS and sip.js


thanks,

chris
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From aqsyounas at gmail.com  Thu Jul 30 00:47:26 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Wed, 29 Jul 2015 13:47:26 -0700
Subject: [Freeswitch-users] how to extract messages(SIP) from large pcap
	file.
Message-ID: <CAGVOjVV+a9HWMw14SLWvaeXA+qc6X5h1UQatRN=0sWSaFpVktA@mail.gmail.com>

Hi,

I know this not a relevant forum for this type of question but hope some of
you guys could help me with some pointers.

I have a large pcap(dump) file with calls of multiple clients having
different IPs.

I want to extract messages based on different IPs and dump into separate
dump file.

How could I achieve this?

Any help would be much appreciated.

Thanks.
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From gmaruzz at gmail.com  Thu Jul 30 01:01:33 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 29 Jul 2015 23:01:33 +0200
Subject: [Freeswitch-users] how to extract messages(SIP) from large pcap
	file.
In-Reply-To: <CAGVOjVV+a9HWMw14SLWvaeXA+qc6X5h1UQatRN=0sWSaFpVktA@mail.gmail.com>
References: <CAGVOjVV+a9HWMw14SLWvaeXA+qc6X5h1UQatRN=0sWSaFpVktA@mail.gmail.com>
Message-ID: <CALXCt0rutZqoWLJ-40q7XTdPvJrPaicCemt0CdeLwBVuFX+Z8Q@mail.gmail.com>

Pcapsipdump

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On Jul 29, 2015 10:48 PM, "Aqs Younas" <aqsyounas at gmail.com> wrote:

> Hi,
>
> I know this not a relevant forum for this type of question but hope some
> of you guys could help me with some pointers.
>
> I have a large pcap(dump) file with calls of multiple clients having
> different IPs.
>
> I want to extract messages based on different IPs and dump into separate
> dump file.
>
> How could I achieve this?
>
> Any help would be much appreciated.
>
> Thanks.
>
> _________________________________________________________________________
> 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 eduar47 at gmail.com  Tue Jul 28 19:47:53 2015
From: eduar47 at gmail.com (Eduar Cardona)
Date: Tue, 28 Jul 2015 10:47:53 -0500
Subject: [Freeswitch-users] IP-Based Authentication
In-Reply-To: <CAMRuXGjf5avzhPDVa1KFs6pt0Uwt10CasLnxmrKhz=j4V+qDRA@mail.gmail.com>
References: <CAMRuXGjf5avzhPDVa1KFs6pt0Uwt10CasLnxmrKhz=j4V+qDRA@mail.gmail.com>
Message-ID: <CAHqHMeuyDLF9Rv=yWvXqwjpRSfns8NoU8ExeeubS7Ers+9qBgA@mail.gmail.com>

You could use cidr in your user definition. Something like:

<user id="my_user_id" cidr="10.20.30.40/32">

After that, reloadacl.


2015-07-28 10:42 GMT-05:00 Murugan Pandian <manpower13.cse at gmail.com>:

> Hi,
>
>
>       How i auth agent based on IP Address
>
> _________________________________________________________________________
> 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 alexeymelnichuck at gmail.com  Wed Jul 29 11:00:27 2015
From: alexeymelnichuck at gmail.com (Alexey Melnichuk)
Date: Wed, 29 Jul 2015 07:00:27 +0000 (UTC)
Subject: [Freeswitch-users] Ad hoc conference
References: <CAAtWKeFidWLen_RpPJG3YPMu6-XnLUKbG2i6zVCjKh585TqNPQ@mail.gmail.com>
Message-ID: <loom.20150729T084058-267@post.gmane.org>

Yehavi Bourvine <yehavi.bourvine at ...> writes:

> 
> - When A presses *1 the first time, a conference room is created and the
>other party is transfered to it.
> - For the second and others: A initiates a new call, and when 
> connected press *2  to connect the  new party as well as A into the 
>conference room,
> 
> What I don;t know how to do is: How to take a bridged call, split it into 
two 
? legs and move one or both into the conference room. Any idea will help.
> 

I just do same thing. This is my first attempt to write FS dialplan.
Some notes. I use FustionPBX and it has extension called `page` that
create conf. So I use same name to my extension.
Also I have problem with implementation `unbridge` so for now 
I use `transfer` application. This is my variant[1].

To transfer both legs to conf I use

	export page_extension=page-${create_uuid foo}-${domain_name}
	bind_digit_action local,*0,exec:transfer,-both 
 page_enter_to XML page.${domain_name},both

[1] https://gist.github.com/moteus/7cf4cd6336f5026e176a




From vivat at uxis.de  Wed Jul 29 16:59:21 2015
From: vivat at uxis.de (Vivat)
Date: Wed, 29 Jul 2015 12:59:21 +0000 (UTC)
Subject: [Freeswitch-users] Bridge incoming call to internal extension and
	to external gateway
Message-ID: <loom.20150729T144304-825@post.gmane.org>

Hi,

I try to ring an incoming call simultanously at my office phone (Snom IP
phone) and at my mobile (via a second SIP gateway). Actually it should be
simple: bridge with two targets.
So I set up in the dialplan something like
<action application="bridge" data="user/1000,sofia/gateway/MyTelco/[my
mobile no]"/>

But the office phone rings only for less then a second and ends then with
"Call completed elsewhere". That means I can not answer the call at the
office phone. The forwarding to the mobile works perfect.

Each single target works (i.e. if I remove the mobile gateway from "bridge"
then my office phone rings for longer and I can answer the call there).

I read the wiki for the FollowMe dialplan but that is actually not what I
want. There should be no first phone with a timeout and then ringing the
second phone but both phones should ring in parallel and the phone with the
first answer wins.
Any hints how to solve this? Thanks in advance.



From bferrell at baywinds.org  Thu Jul 30 00:59:05 2015
From: bferrell at baywinds.org (Bruce Ferrell)
Date: Wed, 29 Jul 2015 13:59:05 -0700
Subject: [Freeswitch-users] how to extract messages(SIP) from large pcap
 file.
In-Reply-To: <CAGVOjVV+a9HWMw14SLWvaeXA+qc6X5h1UQatRN=0sWSaFpVktA@mail.gmail.com>
References: <CAGVOjVV+a9HWMw14SLWvaeXA+qc6X5h1UQatRN=0sWSaFpVktA@mail.gmail.com>
Message-ID: <55B93E99.2090400@baywinds.org>

The best place to get this assistance would be in the wireshark forums.

The said, the key to this is what is called the berkeley packet filter 
"language.

for finding the traffic in your large pcap file, in wireshark the format 
would look something like this:

ip.addr==<host ip address> and sip

in tcpdump it would look something like this

host <host ip address> and sip



On 7/29/15 1:47 PM, Aqs Younas wrote:
> Hi,
>
> I know this not a relevant forum for this type of question but hope 
> some of you guys could help me with some pointers.
>
> I have a large pcap(dump) file with calls of multiple clients having 
> different IPs.
>
> I want to extract messages based on different IPs and dump into 
> separate dump file.
>
> How could I achieve this?
>
> Any help would be much appreciated.
>
> Thanks.
>
>
> _________________________________________________________________________
> 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 ssinyagin at gmail.com  Thu Jul 30 01:05:56 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 29 Jul 2015 23:05:56 +0200
Subject: [Freeswitch-users] Bridge incoming call to internal extension
 and to external gateway
In-Reply-To: <loom.20150729T144304-825@post.gmane.org>
References: <loom.20150729T144304-825@post.gmane.org>
Message-ID: <CACOYK=iTna4ESXrWcXwiEkRdPr1ME=LfFVrbpkpCot4cJroWrA@mail.gmail.com>

You need to set ignore_early_media to true.
On Jul 29, 2015 11:02 PM, "Vivat" <vivat at uxis.de> wrote:

> Hi,
>
> I try to ring an incoming call simultanously at my office phone (Snom IP
> phone) and at my mobile (via a second SIP gateway). Actually it should be
> simple: bridge with two targets.
> So I set up in the dialplan something like
> <action application="bridge" data="user/1000,sofia/gateway/MyTelco/[my
> mobile no]"/>
>
> But the office phone rings only for less then a second and ends then with
> "Call completed elsewhere". That means I can not answer the call at the
> office phone. The forwarding to the mobile works perfect.
>
> Each single target works (i.e. if I remove the mobile gateway from "bridge"
> then my office phone rings for longer and I can answer the call there).
>
> I read the wiki for the FollowMe dialplan but that is actually not what I
> want. There should be no first phone with a timeout and then ringing the
> second phone but both phones should ring in parallel and the phone with the
> first answer wins.
> Any hints how to solve this? 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
>
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From mike at jerris.com  Thu Jul 30 01:07:24 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 29 Jul 2015 17:07:24 -0400
Subject: [Freeswitch-users] FS and T.38 fax
In-Reply-To: <55B9214D.5000709@gmail.com>
References: <55B91A7D.3070403@gmail.com>
	<CALXCt0o2oQpCCYs25Kv2MTWqzTfcad6MYeD+Uw4sr3JCc0P+Bg@mail.gmail.com>
	<55B91CF8.7090408@gmail.com>
	<CALXCt0pmi13OXBNLMTA-11cnp0d9BFceNnmkXyyfGXAywoe9zQ@mail.gmail.com>
	<55B91E20.3050005@gmail.com>
	<CALXCt0peNumvOiP8row1aFAjxLpMesFJSCzPB3EgV77Amftb4w@mail.gmail.com>
	<55B9214D.5000709@gmail.com>
Message-ID: <269DC1C4-8165-403E-88AA-E1CB1E9BC681@jerris.com>

You are not using the right git repo then.  

git clone https://freeswitch.org/stash/scm/fs/freeswitch.git


> On Jul 29, 2015, at 2:54 PM, Mimiko <vbvbrj at gmail.com> wrote:
> 
> On 29.07.2015 21:44, Giovanni Maruzzelli wrote:
>> many people use 1.6 in production
> 
> When I compiled 1.3.13b it also was latest version from git.
> 
> Thank you Giovanni.
> 
> -- 
> Mimiko desu.
> 



From mike at jerris.com  Thu Jul 30 01:08:52 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 29 Jul 2015 17:08:52 -0400
Subject: [Freeswitch-users] webrtc/Sip issue
In-Reply-To: <CAJdRzN6Nmr1v9p4_Yg3saxPmsi6m5Ke1boEoWCZR+eC-7JsNFQ@mail.gmail.com>
References: <CAJdRzN6Nmr1v9p4_Yg3saxPmsi6m5Ke1boEoWCZR+eC-7JsNFQ@mail.gmail.com>
Message-ID: <456E33B6-86AC-45B9-8406-F333C46D6276@jerris.com>

a debug trace including the sip traffic might tell us more.

> On Jul 29, 2015, at 3:21 PM, Chris Mandra <mandra at gmail.com> wrote:
> 
> Hi guys - one of my users keeps getting this error in  fs_cli  when he tries to connect via Chrome (web-rtc) on a macbook running yosemite.
> 2015-07-28 00:42:20.669463 [ERR] switch_core_media.c:6538 AUDIO RTP REPORTS ERROR: [Remote Address Error!]
> 
> I'm using essentially the same setup and never have an issue. Any ideas? He doesn't get anything very useful in the chrome console, just this:
> 
> *********************
> 
> Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | closing INVITE session edsmilh98mlqt0a48qo0tvmlnm0u3m
> 
> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.invitecontext.mediahandler | closing PeerConnection
> 
> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.dialog | dialog edsmilh98mlqt0a48qo0qf3p9f1n97g6QD4ZtFHXgrH deleted
> 
> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | emitting event failed
> 
> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | emitting event rejected
> 
> 
> sc-webrtc.js:931 _rejected called
> 
> *******************
> 
> 
> 
> Any ideas? We're using latest FS and sip.js
> 
> 
> 
> thanks,
> 
> chris
> 

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From rajil.s at gmail.com  Thu Jul 30 07:38:34 2015
From: rajil.s at gmail.com (Rajil Saraswat)
Date: Wed, 29 Jul 2015 22:38:34 -0500
Subject: [Freeswitch-users] [CS_NEW] [WRONG_CALL_STATE]
Message-ID: <CAFoaQoAb6CA0rjGcfE6BY_GLMjjExW58BgqabSJuAiK7RX_YcA@mail.gmail.com>

Hello,

I have two freeswitch servers connected over vpn like the following:

FS A <> openvpn<> FS B

Server A has the ip address 172.16.1.2
Server B has the ip address 192.16.1.2
openvpn gateway is on 10.8.0.1

Both the servers have the common acl defined to not nat the above
addresses like following:

<list name="nat.auto" default="deny">
    <node type="deny" cidr="10.8.0.0/24"/>
    <node type="deny" cidr="172.16.1.0/24"/>
    <node type="deny" cidr="192.168.1.0/24"/>
</list>


<list name="localnet.auto" default="deny">
       <node type="allow" cidr="172.16.1.0/24"/>
       <node type="allow" cidr="10.8.0.0/24"/>
       <node type="allow" cidr="192.168.1.0/24"/>
</list>


I am able to call the extensions on server B from server A. However if
i try calling the extension on server A from server B i get an error
of [CS_NEW] [WRONG_CALL_STATE]. The only way i can make a successful
call from B to A is when i add the network of B in the acl on server A
as follows:

<list name="domains" default="deny">
<node type="allow" domain="$${domain}"/>
<node type="allow" cidr="192.168.1.0/24"/>


Is there a reason why i need to do this?

Thanks
Rajil


From k.presler at megafit.su  Thu Jul 30 08:47:23 2015
From: k.presler at megafit.su (=?UTF-8?B?0JrQuNGA0LjQu9C7INCf0YDQtdGB0LvQtdGA?=)
Date: Thu, 30 Jul 2015 10:47:23 +0600
Subject: [Freeswitch-users]  Fork( ) and Exec ( ) functions
Message-ID: <55B9AC5B.1070300@megafit.su>

Abdul, have you any success?

  
-- 
? ?????????, ??????? ?????? ?????????
??????? ??? ?????????
e-mail: 112 at megafit.su
???: +7 (383) 311-09-09 (???.112)
  
  



From vbvbrj at gmail.com  Thu Jul 30 09:29:12 2015
From: vbvbrj at gmail.com (Mimiko)
Date: Thu, 30 Jul 2015 08:29:12 +0300
Subject: [Freeswitch-users] FS and T.38 fax
In-Reply-To: <269DC1C4-8165-403E-88AA-E1CB1E9BC681@jerris.com>
References: <55B91A7D.3070403@gmail.com>
	<CALXCt0o2oQpCCYs25Kv2MTWqzTfcad6MYeD+Uw4sr3JCc0P+Bg@mail.gmail.com>
	<55B91CF8.7090408@gmail.com>
	<CALXCt0pmi13OXBNLMTA-11cnp0d9BFceNnmkXyyfGXAywoe9zQ@mail.gmail.com>
	<55B91E20.3050005@gmail.com>
	<CALXCt0peNumvOiP8row1aFAjxLpMesFJSCzPB3EgV77Amftb4w@mail.gmail.com>
	<55B9214D.5000709@gmail.com>
	<269DC1C4-8165-403E-88AA-E1CB1E9BC681@jerris.com>
Message-ID: <55B9B628.9050702@gmail.com>

On 30.07.2015 00:07, Michael Jerris wrote:
> You are not using the right git repo then.

Oh, Michael, it was two years ago. :)

-- 
Mimiko desu.


From john.nash778 at gmail.com  Thu Jul 30 10:34:21 2015
From: john.nash778 at gmail.com (John Nash)
Date: Thu, 30 Jul 2015 12:04:21 +0530
Subject: [Freeswitch-users] Display live calls in web pages
Message-ID: <CAA4rGPwPu01X41=DdLD5Yb+VX0d28Q4T9jM6+Of0B+ki1PKywg@mail.gmail.com>

Is there any way of storing "live" calls (Ongoing calls) in DB to a
specific table? where we can fetch these records and display in web page?

I planned to store calls using some lua script hooks but cli command show
calls already shows calls which we can use?
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From s.safarov at gmail.com  Thu Jul 30 10:46:13 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 30 Jul 2015 06:46:13 +0000
Subject: [Freeswitch-users] [CS_NEW] [WRONG_CALL_STATE]
In-Reply-To: <CAFoaQoAb6CA0rjGcfE6BY_GLMjjExW58BgqabSJuAiK7RX_YcA@mail.gmail.com>
References: <CAFoaQoAb6CA0rjGcfE6BY_GLMjjExW58BgqabSJuAiK7RX_YcA@mail.gmail.com>
Message-ID: <CAHtxdDe34ScODL6Mz498NbB9oHHzi2_MpMOzAiyyLnaJuxWERQ@mail.gmail.com>

Try analyze callflow via enabled siptrace in FS or via wireshark on vpn
interface.

On Thu, Jul 30, 2015, 06:40 Rajil Saraswat <rajil.s at gmail.com> wrote:

> Hello,
>
> I have two freeswitch servers connected over vpn like the following:
>
> FS A <> openvpn<> FS B
>
> Server A has the ip address 172.16.1.2
> Server B has the ip address 192.16.1.2
> openvpn gateway is on 10.8.0.1
>
> Both the servers have the common acl defined to not nat the above
> addresses like following:
>
> <list name="nat.auto" default="deny">
>     <node type="deny" cidr="10.8.0.0/24"/>
>     <node type="deny" cidr="172.16.1.0/24"/>
>     <node type="deny" cidr="192.168.1.0/24"/>
> </list>
>
>
> <list name="localnet.auto" default="deny">
>        <node type="allow" cidr="172.16.1.0/24"/>
>        <node type="allow" cidr="10.8.0.0/24"/>
>        <node type="allow" cidr="192.168.1.0/24"/>
> </list>
>
>
> I am able to call the extensions on server B from server A. However if
> i try calling the extension on server A from server B i get an error
> of [CS_NEW] [WRONG_CALL_STATE]. The only way i can make a successful
> call from B to A is when i add the network of B in the acl on server A
> as follows:
>
> <list name="domains" default="deny">
> <node type="allow" domain="$${domain}"/>
> <node type="allow" cidr="192.168.1.0/24"/>
>
>
> Is there a reason why i need to do this?
>
> Thanks
> Rajil
>
> _________________________________________________________________________
> 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 vbvbrj at gmail.com  Thu Jul 30 10:57:14 2015
From: vbvbrj at gmail.com (Mimiko)
Date: Thu, 30 Jul 2015 09:57:14 +0300
Subject: [Freeswitch-users] Display live calls in web pages
In-Reply-To: <CAA4rGPwPu01X41=DdLD5Yb+VX0d28Q4T9jM6+Of0B+ki1PKywg@mail.gmail.com>
References: <CAA4rGPwPu01X41=DdLD5Yb+VX0d28Q4T9jM6+Of0B+ki1PKywg@mail.gmail.com>
Message-ID: <55B9CACA.3010804@gmail.com>

On 30.07.2015 09:34, John Nash wrote:
> Is there any way of storing "live" calls (Ongoing calls) in DB to a
> specific table? where we can fetch these records and display in web page?
>
> I planned to store calls using some lua script hooks but cli command
> show calls already shows calls which we can use?

Hello.

If using LUA, then you can hook a lua script to start-up of FS which 
will hook itself to events for new incomming, but not yet established 
call, or established bridges. Third option is to periodically call cli 
command: `show calls`

-- 
Mimiko desu.


From omortimer at gmail.com  Thu Jul 30 11:02:55 2015
From: omortimer at gmail.com (Oz Mortimer)
Date: Thu, 30 Jul 2015 08:02:55 +0100
Subject: [Freeswitch-users] Display live calls in web pages
In-Reply-To: <55B9CACA.3010804@gmail.com>
References: <CAA4rGPwPu01X41=DdLD5Yb+VX0d28Q4T9jM6+Of0B+ki1PKywg@mail.gmail.com>
	<55B9CACA.3010804@gmail.com>
Message-ID: <95E01F53-459D-447D-9361-4CB457C72A3C@gmail.com>

Why not query core.db? You can use pdo or whatever. Command line is;
Sqlite3 /usr/local/freeswitch/db/core.db
Select * from channels
Or select * from calls




> On 30 Jul 2015, at 07:57, Mimiko <vbvbrj at gmail.com> wrote:
> 
>> On 30.07.2015 09:34, John Nash wrote:
>> Is there any way of storing "live" calls (Ongoing calls) in DB to a
>> specific table? where we can fetch these records and display in web page?
>> 
>> I planned to store calls using some lua script hooks but cli command
>> show calls already shows calls which we can use?
> 
> Hello.
> 
> If using LUA, then you can hook a lua script to start-up of FS which 
> will hook itself to events for new incomming, but not yet established 
> call, or established bridges. Third option is to periodically call cli 
> command: `show calls`
> 
> -- 
> Mimiko desu.
> 
> _________________________________________________________________________
> 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 s.safarov at gmail.com  Thu Jul 30 11:08:42 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 30 Jul 2015 10:08:42 +0300
Subject: [Freeswitch-users] Display live calls in web pages
In-Reply-To: <CAA4rGPwPu01X41=DdLD5Yb+VX0d28Q4T9jM6+Of0B+ki1PKywg@mail.gmail.com>
References: <CAA4rGPwPu01X41=DdLD5Yb+VX0d28Q4T9jM6+Of0B+ki1PKywg@mail.gmail.com>
Message-ID: <CAHtxdDeSHxE7MH=V3NH4JU=KArQbN0r98HCkrq75v5SBBcQ3Pw@mail.gmail.com>

You can call "show calls" via ESL or mod_erlang and parse output.

Sergey

On Thu, Jul 30, 2015 at 9:34 AM, John Nash <john.nash778 at gmail.com> wrote:

> Is there any way of storing "live" calls (Ongoing calls) in DB to a
> specific table? where we can fetch these records and display in web page?
>
> I planned to store calls using some lua script hooks but cli command show
> calls already shows calls which we can use?
>
> _________________________________________________________________________
> 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 petedao at gmail.com  Thu Jul 30 11:46:20 2015
From: petedao at gmail.com (Pete Kay)
Date: Thu, 30 Jul 2015 00:46:20 -0700
Subject: [Freeswitch-users] recording ring tong
In-Reply-To: <CAEwTsAgXBuB7ivUzfau_2SZaBhyCY5hAeqd0cpzDRg-cGUyaBQ@mail.gmail.com>
References: <CAPxXe8WxxQpP03aFAHbZs76-5CONW1iWva+7Kh-QvO9XusphaQ@mail.gmail.com>
	<CAEwTsAgXBuB7ivUzfau_2SZaBhyCY5hAeqd0cpzDRg-cGUyaBQ@mail.gmail.com>
Message-ID: <CAPxXe8XdGLLKSsprWCeEgN7g25Xv08NHGwv-p83s3rvcXrLvtQ@mail.gmail.com>

Hi

How to record the ring tone media if 183  is received with SDP showing the
media IP and port of the ring tone media?

Thanks,
Pete

On Wed, Jul 29, 2015 at 7:28 AM, Michael Jerris <mike at jerris.com> wrote:

> Sometimes a ring tone is a tone that is played in band as audio, other
> times it is just signal of indication of ringing, and your end device
> generates its own tone for you to hear. In the latter case recording
> would not capture it, because there is no tone to capture.
>
>
> On Wednesday, July 29, 2015, Pete Kay <petedao at gmail.com> wrote:
>
>> Hi
>>
>> I am trying to record the incoming ring tone.  Does anyone know how to do
>> that?
>>
>> The record app does not seem to be able to record ring tone.
>>
>> Thanks,
>> Pete
>>
>
> _________________________________________________________________________
> 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 s.safarov at gmail.com  Thu Jul 30 11:49:27 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 30 Jul 2015 10:49:27 +0300
Subject: [Freeswitch-users] recording ring tong
In-Reply-To: <CAPxXe8XdGLLKSsprWCeEgN7g25Xv08NHGwv-p83s3rvcXrLvtQ@mail.gmail.com>
References: <CAPxXe8WxxQpP03aFAHbZs76-5CONW1iWva+7Kh-QvO9XusphaQ@mail.gmail.com>
	<CAEwTsAgXBuB7ivUzfau_2SZaBhyCY5hAeqd0cpzDRg-cGUyaBQ@mail.gmail.com>
	<CAPxXe8XdGLLKSsprWCeEgN7g25Xv08NHGwv-p83s3rvcXrLvtQ@mail.gmail.com>
Message-ID: <CAHtxdDcSxGSW8Tv=r0otZpqe0GiwaakTP0jc5n48kexKNL-71A@mail.gmail.com>

https://supportforums.cisco.com/discussion/11517891/how-save-rtp-streams-wireshark-and-play-it-using-application-called-audacity

On Thu, Jul 30, 2015 at 10:46 AM, Pete Kay <petedao at gmail.com> wrote:

> Hi
>
> How to record the ring tone media if 183  is received with SDP showing the
> media IP and port of the ring tone media?
>
> Thanks,
> Pete
>
> On Wed, Jul 29, 2015 at 7:28 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> Sometimes a ring tone is a tone that is played in band as audio, other
>> times it is just signal of indication of ringing, and your end device
>> generates its own tone for you to hear. In the latter case recording
>> would not capture it, because there is no tone to capture.
>>
>>
>> On Wednesday, July 29, 2015, Pete Kay <petedao at gmail.com> wrote:
>>
>>> Hi
>>>
>>> I am trying to record the incoming ring tone.  Does anyone know how to
>>> do that?
>>>
>>> The record app does not seem to be able to record ring tone.
>>>
>>> Thanks,
>>> Pete
>>>
>>
>> _________________________________________________________________________
>> 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 gmaruzz at gmail.com  Thu Jul 30 12:18:14 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 30 Jul 2015 10:18:14 +0200
Subject: [Freeswitch-users] Display live calls in web pages
In-Reply-To: <CAHtxdDeSHxE7MH=V3NH4JU=KArQbN0r98HCkrq75v5SBBcQ3Pw@mail.gmail.com>
References: <CAA4rGPwPu01X41=DdLD5Yb+VX0d28Q4T9jM6+Of0B+ki1PKywg@mail.gmail.com>
	<CAHtxdDeSHxE7MH=V3NH4JU=KArQbN0r98HCkrq75v5SBBcQ3Pw@mail.gmail.com>
Message-ID: <CALXCt0p-2YsPTsOoa+A1Czo-5Z3Z1UY90JhXCDGCeEGmDN=Hug@mail.gmail.com>

Mod-rpc-xml has an embedded http server, you can post and get "show calls"
to it, and it will return a nice formatted html table :)

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On Jul 30, 2015 9:09 AM, "Sergey Safarov" <s.safarov at gmail.com> wrote:

> You can call "show calls" via ESL or mod_erlang and parse output.
>
> Sergey
>
> On Thu, Jul 30, 2015 at 9:34 AM, John Nash <john.nash778 at gmail.com> wrote:
>
>> Is there any way of storing "live" calls (Ongoing calls) in DB to a
>> specific table? where we can fetch these records and display in web page?
>>
>> I planned to store calls using some lua script hooks but cli command show
>> calls already shows calls which we can use?
>>
>> _________________________________________________________________________
>> 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 john.nash778 at gmail.com  Thu Jul 30 12:25:00 2015
From: john.nash778 at gmail.com (John Nash)
Date: Thu, 30 Jul 2015 13:55:00 +0530
Subject: [Freeswitch-users] Display live calls in web pages
In-Reply-To: <CALXCt0p-2YsPTsOoa+A1Czo-5Z3Z1UY90JhXCDGCeEGmDN=Hug@mail.gmail.com>
References: <CAA4rGPwPu01X41=DdLD5Yb+VX0d28Q4T9jM6+Of0B+ki1PKywg@mail.gmail.com>
	<CAHtxdDeSHxE7MH=V3NH4JU=KArQbN0r98HCkrq75v5SBBcQ3Pw@mail.gmail.com>
	<CALXCt0p-2YsPTsOoa+A1Czo-5Z3Z1UY90JhXCDGCeEGmDN=Hug@mail.gmail.com>
Message-ID: <CAA4rGPzMN8=C283eTq9G558cDnQe1rEcRP8ZCeaQH=Pcen0rXw@mail.gmail.com>

Thank you everyone. I also need to fetch some extra channel variables
(Which are set in dial plans and some message headers) as columns, SQLLite
stores only fixed number of columns right?

On Thu, Jul 30, 2015 at 1:48 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Mod-rpc-xml has an embedded http server, you can post and get "show calls"
> to it, and it will return a nice formatted html table :)
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On Jul 30, 2015 9:09 AM, "Sergey Safarov" <s.safarov at gmail.com> wrote:
>
>> You can call "show calls" via ESL or mod_erlang and parse output.
>>
>> Sergey
>>
>> On Thu, Jul 30, 2015 at 9:34 AM, John Nash <john.nash778 at gmail.com>
>> wrote:
>>
>>> Is there any way of storing "live" calls (Ongoing calls) in DB to a
>>> specific table? where we can fetch these records and display in web page?
>>>
>>> I planned to store calls using some lua script hooks but cli command
>>> show calls already shows calls which we can use?
>>>
>>> _________________________________________________________________________
>>> 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 steveayre at gmail.com  Thu Jul 30 12:28:03 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 30 Jul 2015 09:28:03 +0100
Subject: [Freeswitch-users] Display live calls in web pages
In-Reply-To: <CAA4rGPwPu01X41=DdLD5Yb+VX0d28Q4T9jM6+Of0B+ki1PKywg@mail.gmail.com>
References: <CAA4rGPwPu01X41=DdLD5Yb+VX0d28Q4T9jM6+Of0B+ki1PKywg@mail.gmail.com>
Message-ID: <CAFiqYu=tNhvX7j2gHTtWdpOFU=pd+wi9kbdMwKOByzBdSJCK0Q@mail.gmail.com>

Not to a specific table, but you can use an ODBC database for the core DB.
You'll then be able to query the calls and channels tables. 'show calls' is
just a SQL query that database, which is sqlite by default. Make sure you
only read though, and don't lock any tables/rows.

Alternatively you can connect into FS using ESL from the web server.

On 30 July 2015 at 07:34, John Nash <john.nash778 at gmail.com> wrote:

> Is there any way of storing "live" calls (Ongoing calls) in DB to a
> specific table? where we can fetch these records and display in web page?
>
> I planned to store calls using some lua script hooks but cli command show
> calls already shows calls which we can use?
>
> _________________________________________________________________________
> 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 steveayre at gmail.com  Thu Jul 30 12:29:45 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 30 Jul 2015 09:29:45 +0100
Subject: [Freeswitch-users] Display live calls in web pages
In-Reply-To: <CAA4rGPzMN8=C283eTq9G558cDnQe1rEcRP8ZCeaQH=Pcen0rXw@mail.gmail.com>
References: <CAA4rGPwPu01X41=DdLD5Yb+VX0d28Q4T9jM6+Of0B+ki1PKywg@mail.gmail.com>
	<CAHtxdDeSHxE7MH=V3NH4JU=KArQbN0r98HCkrq75v5SBBcQ3Pw@mail.gmail.com>
	<CALXCt0p-2YsPTsOoa+A1Czo-5Z3Z1UY90JhXCDGCeEGmDN=Hug@mail.gmail.com>
	<CAA4rGPzMN8=C283eTq9G558cDnQe1rEcRP8ZCeaQH=Pcen0rXw@mail.gmail.com>
Message-ID: <CAFiqYumfmSOBBR-cO9hEGLPvKtfSpWD_nwomOiwDrBmzYjs-HA@mail.gmail.com>

If you need extra channel variables then your best bet would be ESL, where
you can 'show calls' / 'show channels' and then uuid_getvar to get the
extra variables.

On 30 July 2015 at 09:25, John Nash <john.nash778 at gmail.com> wrote:

> Thank you everyone. I also need to fetch some extra channel variables
> (Which are set in dial plans and some message headers) as columns, SQLLite
> stores only fixed number of columns right?
>
> On Thu, Jul 30, 2015 at 1:48 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> Mod-rpc-xml has an embedded http server, you can post and get "show
>> calls" to it, and it will return a nice formatted html table :)
>>
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> On Jul 30, 2015 9:09 AM, "Sergey Safarov" <s.safarov at gmail.com> wrote:
>>
>>> You can call "show calls" via ESL or mod_erlang and parse output.
>>>
>>> Sergey
>>>
>>> On Thu, Jul 30, 2015 at 9:34 AM, John Nash <john.nash778 at gmail.com>
>>> wrote:
>>>
>>>> Is there any way of storing "live" calls (Ongoing calls) in DB to a
>>>> specific table? where we can fetch these records and display in web page?
>>>>
>>>> I planned to store calls using some lua script hooks but cli command
>>>> show calls already shows calls which we can use?
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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 vishal.sharma at knowlarity.com  Thu Jul 30 14:55:32 2015
From: vishal.sharma at knowlarity.com (Vishal Sharma)
Date: Thu, 30 Jul 2015 16:25:32 +0530
Subject: [Freeswitch-users] ignore_early_media=true behaviour
Message-ID: <CAEaAqfjr4OQ6cyW+=KWbiFGZAAwTGZyNivPM7nE09QaPFykK6w@mail.gmail.com>

Hi All,
I see a strange behaviour for ignore_early_media variable.
I need to play a hold music file on LegA while Freeswitch tries to bridge
call to one of the listed numbers. Here in dial plan I have used only one
number.

If I use following dialplan without ignore_early_media=true in bridge, Leg
A is able to disconnect the call when Freeswitch is trying to bridge the
call to one of the number in legB.

    <extension name="SIP-TO-SIP">
      <condition field="destination_number" expression="^91(\d+)$">
        <action application="answer"/>
        <action application="set" data="hold_music=/srv/sounds/123.mp3"/>
        <action application="bridge"
data="{caller_id=+2000,ignore_early_media=true,RECORD_STEREO=true,originate_timeout=150,hangup_after_bridge=true,legA_UUID=${uuid}}sofia/gateway/anveo/${destination_number}"/>
        <action application="hangup"/>
      </condition>
    </extension>


caller ------------------FS-------------------Callee
INVITE-------------->|
<-------------------100
<-------------------200
ACK------------------->
###############| INVITE------------------>
###############| <------------------------100
BYE------------------>|
<-------------------200|
###############| CANCEL------------------>
###############| <------------------------200
###############| <---------------------------487
###############| ACK----------------------->



But when I include ignore_early_media=true in bridge command, Freeswitch
ignores BYE sent by legA ,when it is trying to bridge the call with leg B.


caller ------------------FS-------------------Callee
INVITE-------------->|
<-------------------100
<-------------------200
ACK------------------->
###############| INVITE------------------>
###############| <------------------------100
BYE------------------>|
BYE------------------>|
BYE------------------>|
BYE------------------>|
BYE------------------>|
BYE------------------>|
###############| <---------------------------486
###############| ACK----------------------->
<------200 ok  BYE



I would like to end the call if Leg A disconnects the call while I want to
keep ignore_early_media = True so that I can play a hold music file instead
of letting legA user listen to call failure message when FS tries multiple
numbers

Any suggestions ....


Thanks a lot

Vishal Sharma

-- 
SuperReceptionist is now available on Android mobiles. Track your business 
on the go with call analytics, recordings, insights and more: Download the 
app here 
<https://play.google.com/store/apps/details?id=com.superreceptionist>
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From mandra at gmail.com  Thu Jul 30 15:22:57 2015
From: mandra at gmail.com (Chris Mandra)
Date: Thu, 30 Jul 2015 07:22:57 -0400
Subject: [Freeswitch-users] webrtc/Sip issue
In-Reply-To: <456E33B6-86AC-45B9-8406-F333C46D6276@jerris.com>
References: <CAJdRzN6Nmr1v9p4_Yg3saxPmsi6m5Ke1boEoWCZR+eC-7JsNFQ@mail.gmail.com>
	<456E33B6-86AC-45B9-8406-F333C46D6276@jerris.com>
Message-ID: <CAJdRzN4n0HR4VzQkGnuQ9_54d1tAiUba+GX7vcp=auxKs2s26A@mail.gmail.com>

Hi Michael, here's some more debug info: I just want to point out that only
a couple users have this issue. The same hardware and software with diff
users doesn't exhibit this issue.

2015-07-30 10:42:56.431486 [ALERT] switch_core_session.c:2760
sofia/internal/1000 at xxx.xxxxx.xxx receive message [PROGRESS]
2015-07-30 10:42:56.431486 [INFO] switch_core_session.c:2760 Sending early
media
2015-07-30 10:42:57.741478 [DEBUG] switch_core_media.c:5412 STUN Success
[104.197.44.217]:[32306]
2015-07-30 10:42:57.741478 [DEBUG] switch_core_media.c:5884 AUDIO RTP
[sofia/internal/1000 at xxx.xxxxx.xxx] 10.240.125.130 port 32306 ->
2601:88:8003:8a00:1d80:49af:20a6:3e8c port 63602 codec: 111 ms: 20
2015-07-30 10:42:57.741478 [DEBUG] switch_rtp.c:3694 Starting timer [soft]
960 bytes per 20ms
2015-07-30 10:42:57.741478 [ERR] switch_core_media.c:6566 AUDIO RTP REPORTS
ERROR: [Remote Address Error!]
2015-07-30 10:42:57.741478 [NOTICE] switch_core_media.c:6567 Hangup
sofia/internal/1000 at xxx.xxxxx.xxx [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
2015-07-30 10:42:57.741478 [ALERT] switch_channel.c:3277 Send signal
sofia/internal/1000 at xxx.xxxxx.xxx [KILL]
2015-07-30 10:42:57.741478 [ALERT] switch_core_session.c:1412 Send signal
sofia/internal/1000 at xxx.xxxxx.xxx [BREAK]


Startup ipv6 complaints:

015-07-30 10:57:10.336255 [ERR] sofia.c:2935 Error Creating SIP UA for
profile: external-ipv6 (sip:mod_sofia@[::1]:5080;transport=udp,tcp) ATTEMPT
2 (RETRY IN 5 SEC)

2015-07-30 10:57:10.436224 [ERR] sofia.c:2935 Error Creating SIP UA for
profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;transport=udp,tcp) ATTEMPT
2 (RETRY IN 5 SEC)

freeswitch at internal>

2015-07-30 10:57:13.556251 [WARNING] sofia_reg.c:1744 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [1000 at xxx.xxxxx.xxx] from ip
50.242.9.206

2015-07-30 10:57:13.616221 [CONSOLE] mod_voicemail.c:4066 Event Thread
Started

2015-07-30 10:57:15.336251 [ERR] sofia.c:2935 Error Creating SIP UA for
profile: external-ipv6 (sip:mod_sofia@[::1]:5080;transport=udp,tcp) ATTEMPT
3 (RETRY IN 5 SEC)

2015-07-30 10:57:15.336251 [ERR] sofia.c:2945 Error Creating SIP UA for
profile: external-ipv6 (sip:mod_sofia@[::1]:5080;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.

2015-07-30 10:57:15.436256 [ERR] sofia.c:2935 Error Creating SIP UA for
profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;transport=udp,tcp) ATTEMPT
3 (RETRY IN 5 SEC)

2015-07-30 10:57:15.436256 [ERR] sofia.c:2945 Error Creating SIP UA for
profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;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.



And after disabling ipv6

2015-07-30 11:12:50.116372 [DEBUG] switch_core_session.c:2758 Application
ladspa_run Requires media! pre_answering channel sofia/internal/
1000 at xxx.xxxxx.xxx

2015-07-30 11:12:50.116372 [INFO] switch_core_session.c:2760 Sending early
media

2015-07-30 11:12:50.176368 [DEBUG] switch_core_media.c:5412 STUN Success
[xxx.197.44.217]:[23022]

2015-07-30 11:12:50.176368 [DEBUG] switch_core_media.c:5884 AUDIO RTP
[sofia/internal/1000 at xxx.xxxxx.xxx] 10.xxx.xxx.130 port 23022 ->
fdae:f681:faac:a135:b84e:e923:21d0:144a port 52098 codec: 111 ms: 20

2015-07-30 11:12:50.176368 [DEBUG] switch_rtp.c:3694 Starting timer [soft]
960 bytes per 20ms

2015-07-30 11:12:50.176368 [ERR] switch_core_media.c:6566 AUDIO RTP REPORTS
ERROR: [Remote Address Error!]

2015-07-30 11:12:50.176368 [NOTICE] switch_core_media.c:6567 Hangup
sofia/internal/1000 at xxx.xxxxx.xxx [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]



On Wednesday, July 29, 2015, Michael Jerris <mike at jerris.com> wrote:

> a debug trace including the sip traffic might tell us more.
>
> On Jul 29, 2015, at 3:21 PM, Chris Mandra <mandra at gmail.com> wrote:
>
> Hi guys - one of my users keeps getting this error in  fs_cli  when he
> tries to connect via Chrome (web-rtc) on a macbook running yosemite.
>
> 2015-07-28 00:42:20.669463 [ERR] switch_core_media.c:6538 AUDIO RTP
> REPORTS ERROR: [Remote Address Error!]
>
> I'm using essentially the same setup and never have an issue. Any ideas?
> He doesn't get anything very useful in the chrome console, just this:
>
> *********************
>
> Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext |
> closing INVITE session edsmilh98mlqt0a48qo0tvmlnm0u3m
>
> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) |
> sip.invitecontext.mediahandler | closing PeerConnection
>
> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.dialog |
> dialog edsmilh98mlqt0a48qo0qf3p9f1n97g6QD4ZtFHXgrH deleted
>
> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) |
> sip.inviteclientcontext | emitting event failed
>
> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) |
> sip.inviteclientcontext | emitting event rejected
>
> sc-webrtc.js:931 _rejected called
>
> *******************
>
>
> Any ideas? We're using latest FS and sip.js
>
>
> thanks,
>
> chris
>
>
>
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From danny.gershman at gmail.com  Thu Jul 30 16:28:50 2015
From: danny.gershman at gmail.com (Danny Gershman)
Date: Thu, 30 Jul 2015 12:28:50 +0000
Subject: [Freeswitch-users] Endless playback in conference
In-Reply-To: <CACOYK=i4NSW0uSdS+4O3J2LbNYyxfG0_M3gvUcuB-AQfYGebpw@mail.gmail.com>
References: <CAEorZynEWmqnOAEfY00_k1Rwe6jPjsjNSPxJjVaTq_neOMFozg@mail.gmail.com>
	<01ec01d0c4cd$82fd1bf0$88f753d0$@botecomm.com>
	<CAEorZyn0TGWO9nd8Y4hBpWGPLFnag6k6dVSd7dYZCiCcCPzv8A@mail.gmail.com>
	<CAEJMVkDPHWqoPZ5oQ4Pwp5HJQK_eaqSAfZnoz_ZjqvoUpTwRhw@mail.gmail.com>
	<CAEorZy=PUzhRgMHJq8mWfw923A19uc2oZ1mg2UObEi-JVMoSXA@mail.gmail.com>
	<CAO3hyzHWqhFh-QDBJUFzvjFubDNJYFy_V3S=YW-dnNUOG_RVuA@mail.gmail.com>
	<CAEorZynSUb=kQM2znu7RxDp9gYRoYB0Be0JUJvRBPmNF3G9Szg@mail.gmail.com>
	<CACOYK=i4NSW0uSdS+4O3J2LbNYyxfG0_M3gvUcuB-AQfYGebpw@mail.gmail.com>
Message-ID: <CAEorZymLJW3LhuEuM3hTCsw2UZmX_bPYo+_PLj4YDVs6cMmTjw@mail.gmail.com>

Hmmm good point, didn't think of that.  I could deaf the line which would
prevent it from picking up any audio and playing it back into the
conference.

On Thu, Jul 23, 2015 at 8:35 PM Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> But then other members are still able to hear some sound from each other
> through the music. Is that OK with you?
> On Jul 23, 2015 7:42 PM, "Danny Gershman" <danny.gershman at gmail.com>
> wrote:
>
>> Ok what I did was the following, that seems to work.
>>
>> 1) Created a new dialplan extension
>>
>>     <extension name="dynamicmoh">
>>       <condition field="destination_number" expression="^dynamicmoh
>> -(.*)$">
>>         <action application="endless_playback" data="$1" />
>>       </condition>
>>     </extension>
>>
>> 2) From a conference I do this
>>
>> conference <conf> bgdial sofia/internal/dynamicmoh-/path/to/file at server-ip
>>
>> 3) To end playback, just hup the member in the conference.
>>
>> Thanks.
>> --Danny
>>
>> On Thu, Jul 23, 2015 at 12:15 PM Chris Tunbridge <blasterjr at gmail.com>
>> wrote:
>>
>>> Danny the fastest solution i can come up with is doing an originate that
>>> connects a hold music extension to the conference, then when you wanna stop
>>> it, you just kick that member out.
>>>
>>> On Thu, Jul 23, 2015 at 9:28 AM, Danny Gershman <
>>> danny.gershman at gmail.com> wrote:
>>>
>>>> I want to be able to play a file into a conference and have it loop
>>>> forever, like music on hold does.  And then when I need to I want to stop
>>>> it.
>>>>
>>> On Thu, Jul 23, 2015 at 11:20 AM Brian West <brian at freeswitch.org>
>>>> wrote:
>>>>
>>> What is your goal here?  Maybe I missed the entire scenario.
>>>>>
>>>>> On Thu, Jul 23, 2015 at 10:07 AM, Danny Gershman <
>>>>> danny.gershman at gmail.com> wrote:
>>>>>
>>>>>> Wouldn't a profile have to be created on the fly for this? From what
>>>>>> I can see you cannot set this for a conference from an api call. Also to
>>>>>> stop it, you would have to change the profile for the conference to remove
>>>>>> it.
>>>>>> On Wed, Jul 22, 2015 at 6:28 PM Bote Man <bote_radio at botecomm.com>
>>>>>> wrote:
>>>>>>
>>>>>>> There?s a parameter for mod_conference named ?perpetual-sound? that
>>>>>>> looks like it would do the trick.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> It?s about 1/3 of the way down
>>>>>>>
>>>>>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_conference
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> PLEASE check further for any changes that might have been made in
>>>>>>> the latest FreeSWITCH as the conference module has undergone substantial
>>>>>>> changes and perpetual-sound might have been one of them.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Bote
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> *From:* Danny Gershman
>>>>>>> *Sent:* Wednesday, 22 July, 2015 17:09
>>>>>>> *Subject:* [Freeswitch-users] Endless playback in conference
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> I'm trying to do an endless playback of an mp3 file in a conference.
>>>>>>>   I have a couple of ideas, but none seem really solid.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> 1) Pass a variable on play and monitor from mod_event_socket and
>>>>>>> play again if not forcibly terminated.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> 2) Load up local_stream dynamically from an xmlhttp server, and then
>>>>>>> restart the local stream service, however will interrupt MOH for other
>>>>>>> users.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Any other ideas?  Is there a way to do looping for "api" through
>>>>>>> mod_event_socket?  I know you can with "sendmsg"
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --Danny Gershman
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> 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*
>>>>> 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
>>>>>
>>>>
>>>>> _________________________________________________________________________
>>>>> 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
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From mike at jerris.com  Thu Jul 30 18:05:30 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 30 Jul 2015 10:05:30 -0400
Subject: [Freeswitch-users] ignore_early_media=true behaviour
In-Reply-To: <CAEaAqfjr4OQ6cyW+=KWbiFGZAAwTGZyNivPM7nE09QaPFykK6w@mail.gmail.com>
References: <CAEaAqfjr4OQ6cyW+=KWbiFGZAAwTGZyNivPM7nE09QaPFykK6w@mail.gmail.com>
Message-ID: <BDEB407D-C21B-4A85-8219-F7F97582B2AB@jerris.com>

Please file a bug with all the details how to reproduce the issue, and full debug logs with sip trace.

> On Jul 30, 2015, at 6:55 AM, Vishal Sharma <vishal.sharma at knowlarity.com> wrote:
> 
> Hi All,
> I see a strange behaviour for ignore_early_media variable.
> I need to play a hold music file on LegA while Freeswitch tries to bridge call to one of the listed numbers. Here in dial plan I have used only one number.
> 
> If I use following dialplan without ignore_early_media=true in bridge, Leg A is able to disconnect the call when Freeswitch is trying to bridge the call to one of the number in legB.
> 
>     <extension name="SIP-TO-SIP">
>       <condition field="destination_number" expression="^91(\d+)$">
>         <action application="answer"/>
>         <action application="set" data="hold_music=/srv/sounds/123.mp3"/>
>         <action application="bridge" data="{caller_id=+2000,ignore_early_media=true,RECORD_STEREO=true,originate_timeout=150,hangup_after_bridge=true,legA_UUID=${uuid}}sofia/gateway/anveo/${destination_number}"/>
>         <action application="hangup"/>
>       </condition>
>     </extension>
> 
> 
> caller ------------------FS-------------------Callee
> INVITE-------------->|
> <-------------------100
> <-------------------200
> ACK------------------->
> ###############| INVITE------------------>
> ###############| <------------------------100
> BYE------------------>|
> <-------------------200|
> ###############| CANCEL------------------>
> ###############| <------------------------200 
> ###############| <---------------------------487
> ###############| ACK----------------------->
> 
> 
> 
> But when I include ignore_early_media=true in bridge command, Freeswitch ignores BYE sent by legA ,when it is trying to bridge the call with leg B.
> 
> 
> caller ------------------FS-------------------Callee
> INVITE-------------->|
> <-------------------100
> <-------------------200
> ACK------------------->
> ###############| INVITE------------------>
> ###############| <------------------------100
> BYE------------------>|
> BYE------------------>|
> BYE------------------>|
> BYE------------------>|
> BYE------------------>|
> BYE------------------>|
> ###############| <---------------------------486
> ###############| ACK----------------------->
> <------200 ok  BYE
> 
> 
> 
> I would like to end the call if Leg A disconnects the call while I want to keep ignore_early_media = True so that I can play a hold music file instead of letting legA user listen to call failure message when FS tries multiple numbers 
> 
> Any suggestions ....
> 
> 
> Thanks a lot 
> 
> Vishal Sharma 
> 

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From mike at jerris.com  Thu Jul 30 18:07:50 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 30 Jul 2015 10:07:50 -0400
Subject: [Freeswitch-users] webrtc/Sip issue
In-Reply-To: <CAJdRzN4n0HR4VzQkGnuQ9_54d1tAiUba+GX7vcp=auxKs2s26A@mail.gmail.com>
References: <CAJdRzN6Nmr1v9p4_Yg3saxPmsi6m5Ke1boEoWCZR+eC-7JsNFQ@mail.gmail.com>
	<456E33B6-86AC-45B9-8406-F333C46D6276@jerris.com>
	<CAJdRzN4n0HR4VzQkGnuQ9_54d1tAiUba+GX7vcp=auxKs2s26A@mail.gmail.com>
Message-ID: <34D043B9-DDDC-4BA7-876D-C3C90AA91C94@jerris.com>

As i said before.  a debug trace including the sip traffic might tell us more.  A partial log that you have edited to remove data, without sip trace will not tell us anything useful.

> On Jul 30, 2015, at 7:22 AM, Chris Mandra <mandra at gmail.com> wrote:
> 
> Hi Michael, here's some more debug info: I just want to point out that only a couple users have this issue. The same hardware and software with diff users doesn't exhibit this issue. 
> 
> 2015-07-30 10:42:56.431486 [ALERT] switch_core_session.c:2760 sofia/internal/1000 at xxx.xxxxx.xxx receive message [PROGRESS]
> 2015-07-30 10:42:56.431486 [INFO] switch_core_session.c:2760 Sending early media
> 2015-07-30 10:42:57.741478 [DEBUG] switch_core_media.c:5412 STUN Success [104.197.44.217]:[32306]
> 2015-07-30 10:42:57.741478 [DEBUG] switch_core_media.c:5884 AUDIO RTP [sofia/internal/1000 at xxx.xxxxx.xxx] 10.240.125.130 port 32306 -> 2601:88:8003:8a00:1d80:49af:20a6:3e8c port 63602 codec: 111 ms: 20
> 2015-07-30 10:42:57.741478 [DEBUG] switch_rtp.c:3694 Starting timer [soft] 960 bytes per 20ms
> 2015-07-30 10:42:57.741478 [ERR] switch_core_media.c:6566 AUDIO RTP REPORTS ERROR: [Remote Address Error!]
> 2015-07-30 10:42:57.741478 [NOTICE] switch_core_media.c:6567 Hangup sofia/internal/1000 at xxx.xxxxx.xxx [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
> 2015-07-30 10:42:57.741478 [ALERT] switch_channel.c:3277 Send signal sofia/internal/1000 at xxx.xxxxx.xxx [KILL]
> 2015-07-30 10:42:57.741478 [ALERT] switch_core_session.c:1412 Send signal sofia/internal/1000 at xxx.xxxxx.xxx [BREAK]
> 
> 
> Startup ipv6 complaints:
> 
> 015-07-30 10:57:10.336255 [ERR] sofia.c:2935 Error Creating SIP UA for profile: external-ipv6 (sip:mod_sofia@[::1]:5080;transport=udp,tcp) ATTEMPT 2 (RETRY IN 5 SEC)
> 
> 2015-07-30 10:57:10.436224 [ERR] sofia.c:2935 Error Creating SIP UA for profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;transport=udp,tcp) ATTEMPT 2 (RETRY IN 5 SEC)
> 
> freeswitch at internal>
> 
> 2015-07-30 10:57:13.556251 [WARNING] sofia_reg.c:1744 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1000 at xxx.xxxxx.xxx] from ip 50.242.9.206
> 
> 2015-07-30 10:57:13.616221 [CONSOLE] mod_voicemail.c:4066 Event Thread Started
> 
> 2015-07-30 10:57:15.336251 [ERR] sofia.c:2935 Error Creating SIP UA for profile: external-ipv6 (sip:mod_sofia@[::1]:5080;transport=udp,tcp) ATTEMPT 3 (RETRY IN 5 SEC)
> 
> 2015-07-30 10:57:15.336251 [ERR] sofia.c:2945 Error Creating SIP UA for profile: external-ipv6 (sip:mod_sofia@[::1]:5080;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.
> 
> 2015-07-30 10:57:15.436256 [ERR] sofia.c:2935 Error Creating SIP UA for profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;transport=udp,tcp) ATTEMPT 3 (RETRY IN 5 SEC)
> 
> 2015-07-30 10:57:15.436256 [ERR] sofia.c:2945 Error Creating SIP UA for profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;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.
> 
> 
> 
> 
> And after disabling ipv6
> 2015-07-30 11:12:50.116372 [DEBUG] switch_core_session.c:2758 Application ladspa_run Requires media! pre_answering channel sofia/internal/1000 at xxx.xxxxx.xxx
> 
> 2015-07-30 11:12:50.116372 [INFO] switch_core_session.c:2760 Sending early media
> 
> 2015-07-30 11:12:50.176368 [DEBUG] switch_core_media.c:5412 STUN Success [xxx.197.44.217]:[23022]
> 
> 2015-07-30 11:12:50.176368 [DEBUG] switch_core_media.c:5884 AUDIO RTP [sofia/internal/1000 at xxx.xxxxx.xxx] 10.xxx.xxx.130 port 23022 -> fdae:f681:faac:a135:b84e:e923:21d0:144a port 52098 codec: 111 ms: 20
> 
> 2015-07-30 11:12:50.176368 [DEBUG] switch_rtp.c:3694 Starting timer [soft] 960 bytes per 20ms
> 
> 2015-07-30 11:12:50.176368 [ERR] switch_core_media.c:6566 AUDIO RTP REPORTS ERROR: [Remote Address Error!]
> 
> 2015-07-30 11:12:50.176368 [NOTICE] switch_core_media.c:6567 Hangup sofia/internal/1000 at xxx.xxxxx.xxx [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
> 
> 
> 
> 
> On Wednesday, July 29, 2015, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> a debug trace including the sip traffic might tell us more.
> 
>> On Jul 29, 2015, at 3:21 PM, Chris Mandra <mandra at gmail.com <>> wrote:
>> 
>> Hi guys - one of my users keeps getting this error in  fs_cli  when he tries to connect via Chrome (web-rtc) on a macbook running yosemite.
>> 2015-07-28 00:42:20.669463 [ERR] switch_core_media.c:6538 AUDIO RTP REPORTS ERROR: [Remote Address Error!]
>> 
>> I'm using essentially the same setup and never have an issue. Any ideas? He doesn't get anything very useful in the chrome console, just this:
>> 
>> *********************
>> 
>> Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | closing INVITE session edsmilh98mlqt0a48qo0tvmlnm0u3m
>> 
>> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.invitecontext.mediahandler | closing PeerConnection
>> 
>> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.dialog | dialog edsmilh98mlqt0a48qo0qf3p9f1n97g6QD4ZtFHXgrH deleted
>> 
>> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | emitting event failed
>> 
>> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | emitting event rejected
>> 
>> 
>> sc-webrtc.js:931 _rejected called
>> 
>> *******************
>> 
>> 
>> 
>> Any ideas? We're using latest FS and sip.js
>> 
>> 
>> 
>> thanks,
>> 
>> chris
>> 

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From nh45cmi at gmail.com  Thu Jul 30 16:12:03 2015
From: nh45cmi at gmail.com (NH45 CMI)
Date: Thu, 30 Jul 2015 17:42:03 +0530
Subject: [Freeswitch-users] ESL Freeswitch cluster
Message-ID: <CAAy72=Xjy5C67Z_XThAYa=tQmoU9Z1TdhFuhcHJmbYTGpkkoHA@mail.gmail.com>

Hi Guys,


           I am using opensips for load balancing and freeswitch for
register,IVR,Callcenter ,Using mod_xml_curl for realtime ivr and dialplan
but callcenter i have problem because it load only once so i use ESL to add
and remove agent ,if i use one FS it's ok  but more then one FS how can i
add or remove agent to queue all FS using ESL





Regard's
NH45 CMI
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From mandra at gmail.com  Thu Jul 30 18:27:57 2015
From: mandra at gmail.com (Chris Mandra)
Date: Thu, 30 Jul 2015 10:27:57 -0400
Subject: [Freeswitch-users] webrtc/Sip issue
In-Reply-To: <34D043B9-DDDC-4BA7-876D-C3C90AA91C94@jerris.com>
References: <CAJdRzN6Nmr1v9p4_Yg3saxPmsi6m5Ke1boEoWCZR+eC-7JsNFQ@mail.gmail.com>
	<456E33B6-86AC-45B9-8406-F333C46D6276@jerris.com>
	<CAJdRzN4n0HR4VzQkGnuQ9_54d1tAiUba+GX7vcp=auxKs2s26A@mail.gmail.com>
	<34D043B9-DDDC-4BA7-876D-C3C90AA91C94@jerris.com>
Message-ID: <CAJdRzN6LpzQdzyrUXHbZQxXxUQwH_VEF2yDV805gYJx4yATY6Q@mail.gmail.com>

Let me make sure I understand: You want me to use the GDB and get that info
to you as described here:
https://wiki.freeswitch.org/wiki/Debugging_Freeswitch ?

Do you want wireshark info as well? I know that there have been some issues
with IPv6 - perhaps this is related?

thank you,
chris

---------- Forwarded message ----------
From: Michael Jerris <mike at jerris.com>
Date: Thu, Jul 30, 2015 at 10:07 AM
Subject: Re: [Freeswitch-users] webrtc/Sip issue
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>


As i said before.  a debug trace including the sip traffic might tell us
more.  A partial log that you have edited to remove data, without sip trace
will not tell us anything useful.

On Jul 30, 2015, at 7:22 AM, Chris Mandra <mandra at gmail.com> wrote:

Hi Michael, here's some more debug info: I just want to point out that only
a couple users have this issue. The same hardware and software with diff
users doesn't exhibit this issue.

2015-07-30 10:42:56.431486 [ALERT] switch_core_session.c:2760
sofia/internal/1000 at xxx.xxxxx.xxx receive message [PROGRESS]
2015-07-30 10:42:56.431486 [INFO] switch_core_session.c:2760 Sending early
media
2015-07-30 10:42:57.741478 [DEBUG] switch_core_media.c:5412 STUN Success
[104.197.44.217]:[32306]
2015-07-30 10:42:57.741478 [DEBUG] switch_core_media.c:5884 AUDIO RTP [
sofia/internal/1000 at xxx.xxxxx.xxx] 10.240.125.130 port 32306 ->
2601:88:8003:8a00:1d80:49af:20a6:3e8c port 63602 codec: 111 ms: 20
2015-07-30 10:42:57.741478 [DEBUG] switch_rtp.c:3694 Starting timer [soft]
960 bytes per 20ms
2015-07-30 10:42:57.741478 [ERR] switch_core_media.c:6566 AUDIO RTP REPORTS
ERROR: [Remote Address Error!]
2015-07-30 10:42:57.741478 [NOTICE] switch_core_media.c:6567 Hangup
sofia/internal/1000 at xxx.xxxxx.xxx [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
2015-07-30 10:42:57.741478 [ALERT] switch_channel.c:3277 Send signal
sofia/internal/1000 at xxx.xxxxx.xxx [KILL]
2015-07-30 10:42:57.741478 [ALERT] switch_core_session.c:1412 Send signal
sofia/internal/1000 at xxx.xxxxx.xxx [BREAK]


Startup ipv6 complaints:

015-07-30 10:57:10.336255 [ERR] sofia.c:2935 Error Creating SIP UA for
profile: external-ipv6 (sip:mod_sofia@[::1]:5080;transport=udp,tcp) ATTEMPT
2 (RETRY IN 5 SEC)

2015-07-30 10:57:10.436224 [ERR] sofia.c:2935 Error Creating SIP UA for
profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;transport=udp,tcp) ATTEMPT
2 (RETRY IN 5 SEC)

freeswitch at internal>

2015-07-30 10:57:13.556251 [WARNING] sofia_reg.c:1744 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [1000 at xxx.xxxxx.xxx] from ip
50.242.9.206

2015-07-30 10:57:13.616221 [CONSOLE] mod_voicemail.c:4066 Event Thread
Started

2015-07-30 10:57:15.336251 [ERR] sofia.c:2935 Error Creating SIP UA for
profile: external-ipv6 (sip:mod_sofia@[::1]:5080;transport=udp,tcp) ATTEMPT
3 (RETRY IN 5 SEC)

2015-07-30 10:57:15.336251 [ERR] sofia.c:2945 Error Creating SIP UA for
profile: external-ipv6 (sip:mod_sofia@[::1]:5080;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.

2015-07-30 10:57:15.436256 [ERR] sofia.c:2935 Error Creating SIP UA for
profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;transport=udp,tcp) ATTEMPT
3 (RETRY IN 5 SEC)

2015-07-30 10:57:15.436256 [ERR] sofia.c:2945 Error Creating SIP UA for
profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;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.



And after disabling ipv6

2015-07-30 11:12:50.116372 [DEBUG] switch_core_session.c:2758 Application
ladspa_run Requires media! pre_answering channel sofia/internal/
1000 at xxx.xxxxx.xxx

2015-07-30 11:12:50.116372 [INFO] switch_core_session.c:2760 Sending early
media

2015-07-30 11:12:50.176368 [DEBUG] switch_core_media.c:5412 STUN Success
[xxx.197.44.217]:[23022]

2015-07-30 11:12:50.176368 [DEBUG] switch_core_media.c:5884 AUDIO RTP [
sofia/internal/1000 at xxx.xxxxx.xxx] 10.xxx.xxx.130 port 23022 ->
fdae:f681:faac:a135:b84e:e923:21d0:144a port 52098 codec: 111 ms: 20

2015-07-30 11:12:50.176368 [DEBUG] switch_rtp.c:3694 Starting timer [soft]
960 bytes per 20ms

2015-07-30 11:12:50.176368 [ERR] switch_core_media.c:6566 AUDIO RTP REPORTS
ERROR: [Remote Address Error!]

2015-07-30 11:12:50.176368 [NOTICE] switch_core_media.c:6567 Hangup
sofia/internal/1000 at xxx.xxxxx.xxx [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]



On Wednesday, July 29, 2015, Michael Jerris <mike at jerris.com> wrote:

> a debug trace including the sip traffic might tell us more.
>
> On Jul 29, 2015, at 3:21 PM, Chris Mandra <mandra at gmail.com> wrote:
>
> Hi guys - one of my users keeps getting this error in  fs_cli  when he
> tries to connect via Chrome (web-rtc) on a macbook running yosemite.
>
> 2015-07-28 00:42:20.669463 [ERR] switch_core_media.c:6538 AUDIO RTP
> REPORTS ERROR: [Remote Address Error!]
>
> I'm using essentially the same setup and never have an issue. Any ideas?
> He doesn't get anything very useful in the chrome console, just this:
>
> *********************
>
> Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext |
> closing INVITE session edsmilh98mlqt0a48qo0tvmlnm0u3m
>
> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) |
> sip.invitecontext.mediahandler | closing PeerConnection
>
> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.dialog |
> dialog edsmilh98mlqt0a48qo0qf3p9f1n97g6QD4ZtFHXgrH deleted
>
> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) |
> sip.inviteclientcontext | emitting event failed
>
> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) |
> sip.inviteclientcontext | emitting event rejected
>
> sc-webrtc.js:931 _rejected called
>
> *******************
>
>
> Any ideas? We're using latest FS and sip.js
>
>
> thanks,
>
> chris
>
>

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

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
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FreeSWITCH-users mailing list
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-- 
mandra
c:410.258.5281
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From gmaruzz at gmail.com  Thu Jul 30 19:02:06 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 30 Jul 2015 17:02:06 +0200
Subject: [Freeswitch-users] webrtc/Sip issue
In-Reply-To: <CAJdRzN6LpzQdzyrUXHbZQxXxUQwH_VEF2yDV805gYJx4yATY6Q@mail.gmail.com>
References: <CAJdRzN6Nmr1v9p4_Yg3saxPmsi6m5Ke1boEoWCZR+eC-7JsNFQ@mail.gmail.com>
	<456E33B6-86AC-45B9-8406-F333C46D6276@jerris.com>
	<CAJdRzN4n0HR4VzQkGnuQ9_54d1tAiUba+GX7vcp=auxKs2s26A@mail.gmail.com>
	<34D043B9-DDDC-4BA7-876D-C3C90AA91C94@jerris.com>
	<CAJdRzN6LpzQdzyrUXHbZQxXxUQwH_VEF2yDV805gYJx4yATY6Q@mail.gmail.com>
Message-ID: <CALXCt0r54=ScaWwUewQvb2Nuh2GJ8HTn0F1vQimJBsXOnbT3PQ@mail.gmail.com>

no GDB (gdb is for segfaults)

complete debug log from console, from fs_cli:

fs_cli>sofia global siptrace on
fs_cli>sofia debug all 9
fs_cli>fsctl loglevel 9
fs_cli>fsctl console 9

then reproduce what is your problem

then attach it all, complete of all relevant info like configurations,
topology, whatever


On Thu, Jul 30, 2015 at 4:27 PM, Chris Mandra <mandra at gmail.com> wrote:

> Let me make sure I understand: You want me to use the GDB and get that
> info to you as described here:
> https://wiki.freeswitch.org/wiki/Debugging_Freeswitch ?
>
> Do you want wireshark info as well? I know that there have been some
> issues with IPv6 - perhaps this is related?
>
> thank you,
> chris
>
> ---------- Forwarded message ----------
> From: Michael Jerris <mike at jerris.com>
> Date: Thu, Jul 30, 2015 at 10:07 AM
> Subject: Re: [Freeswitch-users] webrtc/Sip issue
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
>
> As i said before.  a debug trace including the sip traffic might tell us
> more.  A partial log that you have edited to remove data, without sip trace
> will not tell us anything useful.
>
> On Jul 30, 2015, at 7:22 AM, Chris Mandra <mandra at gmail.com> wrote:
>
> Hi Michael, here's some more debug info: I just want to point out that
> only a couple users have this issue. The same hardware and software with
> diff users doesn't exhibit this issue.
>
> 2015-07-30 10:42:56.431486 [ALERT] switch_core_session.c:2760
> sofia/internal/1000 at xxx.xxxxx.xxx receive message [PROGRESS]
> 2015-07-30 10:42:56.431486 [INFO] switch_core_session.c:2760 Sending early
> media
> 2015-07-30 10:42:57.741478 [DEBUG] switch_core_media.c:5412 STUN Success
> [104.197.44.217]:[32306]
> 2015-07-30 10:42:57.741478 [DEBUG] switch_core_media.c:5884 AUDIO RTP [
> sofia/internal/1000 at xxx.xxxxx.xxx] 10.240.125.130 port 32306 ->
> 2601:88:8003:8a00:1d80:49af:20a6:3e8c port 63602 codec: 111 ms: 20
> 2015-07-30 10:42:57.741478 [DEBUG] switch_rtp.c:3694 Starting timer [soft]
> 960 bytes per 20ms
> 2015-07-30 10:42:57.741478 [ERR] switch_core_media.c:6566 AUDIO RTP
> REPORTS ERROR: [Remote Address Error!]
> 2015-07-30 10:42:57.741478 [NOTICE] switch_core_media.c:6567 Hangup
> sofia/internal/1000 at xxx.xxxxx.xxx [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
> 2015-07-30 10:42:57.741478 [ALERT] switch_channel.c:3277 Send signal
> sofia/internal/1000 at xxx.xxxxx.xxx [KILL]
> 2015-07-30 10:42:57.741478 [ALERT] switch_core_session.c:1412 Send signal
> sofia/internal/1000 at xxx.xxxxx.xxx [BREAK]
>
>
> Startup ipv6 complaints:
>
> 015-07-30 10:57:10.336255 [ERR] sofia.c:2935 Error Creating SIP UA for
> profile: external-ipv6 (sip:mod_sofia@[::1]:5080;transport=udp,tcp)
> ATTEMPT 2 (RETRY IN 5 SEC)
>
> 2015-07-30 10:57:10.436224 [ERR] sofia.c:2935 Error Creating SIP UA for
> profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;transport=udp,tcp)
> ATTEMPT 2 (RETRY IN 5 SEC)
>
> freeswitch at internal>
>
> 2015-07-30 10:57:13.556251 [WARNING] sofia_reg.c:1744 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [1000 at xxx.xxxxx.xxx] from ip
> 50.242.9.206
>
> 2015-07-30 10:57:13.616221 [CONSOLE] mod_voicemail.c:4066 Event Thread
> Started
>
> 2015-07-30 10:57:15.336251 [ERR] sofia.c:2935 Error Creating SIP UA for
> profile: external-ipv6 (sip:mod_sofia@[::1]:5080;transport=udp,tcp)
> ATTEMPT 3 (RETRY IN 5 SEC)
>
> 2015-07-30 10:57:15.336251 [ERR] sofia.c:2945 Error Creating SIP UA for
> profile: external-ipv6 (sip:mod_sofia@[::1]:5080;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.
>
> 2015-07-30 10:57:15.436256 [ERR] sofia.c:2935 Error Creating SIP UA for
> profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;transport=udp,tcp)
> ATTEMPT 3 (RETRY IN 5 SEC)
>
> 2015-07-30 10:57:15.436256 [ERR] sofia.c:2945 Error Creating SIP UA for
> profile: internal-ipv6 (sip:mod_sofia@[::1]:5060;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.
>
>
>
> And after disabling ipv6
>
> 2015-07-30 11:12:50.116372 [DEBUG] switch_core_session.c:2758 Application
> ladspa_run Requires media! pre_answering channel sofia/internal/
> 1000 at xxx.xxxxx.xxx
>
> 2015-07-30 11:12:50.116372 [INFO] switch_core_session.c:2760 Sending early
> media
>
> 2015-07-30 11:12:50.176368 [DEBUG] switch_core_media.c:5412 STUN Success
> [xxx.197.44.217]:[23022]
>
> 2015-07-30 11:12:50.176368 [DEBUG] switch_core_media.c:5884 AUDIO RTP [
> sofia/internal/1000 at xxx.xxxxx.xxx] 10.xxx.xxx.130 port 23022 ->
> fdae:f681:faac:a135:b84e:e923:21d0:144a port 52098 codec: 111 ms: 20
>
> 2015-07-30 11:12:50.176368 [DEBUG] switch_rtp.c:3694 Starting timer [soft]
> 960 bytes per 20ms
>
> 2015-07-30 11:12:50.176368 [ERR] switch_core_media.c:6566 AUDIO RTP
> REPORTS ERROR: [Remote Address Error!]
>
> 2015-07-30 11:12:50.176368 [NOTICE] switch_core_media.c:6567 Hangup
> sofia/internal/1000 at xxx.xxxxx.xxx [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
>
>
>
> On Wednesday, July 29, 2015, Michael Jerris <mike at jerris.com> wrote:
>
>> a debug trace including the sip traffic might tell us more.
>>
>> On Jul 29, 2015, at 3:21 PM, Chris Mandra <mandra at gmail.com> wrote:
>>
>> Hi guys - one of my users keeps getting this error in  fs_cli  when he
>> tries to connect via Chrome (web-rtc) on a macbook running yosemite.
>>
>> 2015-07-28 00:42:20.669463 [ERR] switch_core_media.c:6538 AUDIO RTP
>> REPORTS ERROR: [Remote Address Error!]
>>
>> I'm using essentially the same setup and never have an issue. Any ideas?
>> He doesn't get anything very useful in the chrome console, just this:
>>
>> *********************
>>
>> Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext |
>> closing INVITE session edsmilh98mlqt0a48qo0tvmlnm0u3m
>>
>> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) |
>> sip.invitecontext.mediahandler | closing PeerConnection
>>
>> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.dialog
>> | dialog edsmilh98mlqt0a48qo0qf3p9f1n97g6QD4ZtFHXgrH deleted
>>
>> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) |
>> sip.inviteclientcontext | emitting event failed
>>
>> sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) |
>> sip.inviteclientcontext | emitting event rejected
>>
>> sc-webrtc.js:931 _rejected called
>>
>> *******************
>>
>>
>> Any ideas? We're using latest FS and sip.js
>>
>>
>> thanks,
>>
>> chris
>>
>>
>
> _________________________________________________________________________
> 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
>
>
>
> --
> mandra
> c:410.258.5281
>
> _________________________________________________________________________
> 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
Cell : +39-347-2665618
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From idokan at gmail.com  Thu Jul 30 19:52:38 2015
From: idokan at gmail.com (ik)
Date: Thu, 30 Jul 2015 18:52:38 +0300
Subject: [Freeswitch-users] how to extract messages(SIP) from large pcap
	file.
In-Reply-To: <CAGVOjVV+a9HWMw14SLWvaeXA+qc6X5h1UQatRN=0sWSaFpVktA@mail.gmail.com>
References: <CAGVOjVV+a9HWMw14SLWvaeXA+qc6X5h1UQatRN=0sWSaFpVktA@mail.gmail.com>
Message-ID: <CAOccsKDZDqnx8HqnJ7p=O+8uAPoe5=6c182LtxZ_jW8r_c=paw@mail.gmail.com>

In wireshark, you can choose to follow UDP (or TCP) stream, and then you
can export that UDP stream to a new pcap file.

BTW, when capturing stuff on pcap related tool, you can create multiple
files based on some rules, such as size of file, or even timestamp like so:
tcpdump -nq -s 0 -A -vvv -i eth0 -G3600 -w /tmp/trace/sip-%F--%H-%M-%S.pcap

The -G is the number of seconds that the file will be rotated.

Ido

On Wed, Jul 29, 2015 at 11:47 PM, Aqs Younas <aqsyounas at gmail.com> wrote:

> Hi,
>
> I know this not a relevant forum for this type of question but hope some
> of you guys could help me with some pointers.
>
> I have a large pcap(dump) file with calls of multiple clients having
> different IPs.
>
> I want to extract messages based on different IPs and dump into separate
> dump file.
>
> How could I achieve this?
>
> Any help would be much appreciated.
>
> Thanks.
>
> _________________________________________________________________________
> 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 Jul 30 19:58:29 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 30 Jul 2015 17:58:29 +0200
Subject: [Freeswitch-users] how to extract messages(SIP) from large pcap
	file.
In-Reply-To: <CAOccsKDZDqnx8HqnJ7p=O+8uAPoe5=6c182LtxZ_jW8r_c=paw@mail.gmail.com>
References: <CAGVOjVV+a9HWMw14SLWvaeXA+qc6X5h1UQatRN=0sWSaFpVktA@mail.gmail.com>
	<CAOccsKDZDqnx8HqnJ7p=O+8uAPoe5=6c182LtxZ_jW8r_c=paw@mail.gmail.com>
Message-ID: <CALXCt0p_cVFse8QCh_yvkaZ5zmrJuMzPUezXh1FdmK0UcGM_ZA@mail.gmail.com>

use pcapsipdump ( http://pcapsipdump.sourceforge.net/ ) it has been
designed exactly for your needs.

-giovanni

On Thu, Jul 30, 2015 at 5:52 PM, ik <idokan at gmail.com> wrote:

> In wireshark, you can choose to follow UDP (or TCP) stream, and then you
> can export that UDP stream to a new pcap file.
>
> BTW, when capturing stuff on pcap related tool, you can create multiple
> files based on some rules, such as size of file, or even timestamp like so:
> tcpdump -nq -s 0 -A -vvv -i eth0 -G3600 -w /tmp/trace/sip-%F--%H-%M-%S.pcap
>
> The -G is the number of seconds that the file will be rotated.
>
> Ido
>
> On Wed, Jul 29, 2015 at 11:47 PM, Aqs Younas <aqsyounas at gmail.com> wrote:
>
>> Hi,
>>
>> I know this not a relevant forum for this type of question but hope some
>> of you guys could help me with some pointers.
>>
>> I have a large pcap(dump) file with calls of multiple clients having
>> different IPs.
>>
>> I want to extract messages based on different IPs and dump into separate
>> dump file.
>>
>> How could I achieve this?
>>
>> Any help would be much appreciated.
>>
>> Thanks.
>>
>> _________________________________________________________________________
>> 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
Cell : +39-347-2665618
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From govoiper at gmail.com  Thu Jul 30 20:03:22 2015
From: govoiper at gmail.com (SamyGo)
Date: Thu, 30 Jul 2015 12:03:22 -0400
Subject: [Freeswitch-users] ESL Freeswitch cluster
In-Reply-To: <CAAy72=Xjy5C67Z_XThAYa=tQmoU9Z1TdhFuhcHJmbYTGpkkoHA@mail.gmail.com>
References: <CAAy72=Xjy5C67Z_XThAYa=tQmoU9Z1TdhFuhcHJmbYTGpkkoHA@mail.gmail.com>
Message-ID: <CAJUJwtikQ5qv8A+YR5hqfy4eCr_iV5gm2BwJsWtKpdw2A_NOhw@mail.gmail.com>

Hi NH45,

Do you want to send the same ESL action to all of the FS Servers, something
like a ESL Proxy that can relay your event to the FS Servers ?

Regards,
Sammy

On Thu, Jul 30, 2015 at 8:12 AM, NH45 CMI <nh45cmi at gmail.com> wrote:

> Hi Guys,
>
>
>            I am using opensips for load balancing and freeswitch for
> register,IVR,Callcenter ,Using mod_xml_curl for realtime ivr and dialplan
> but callcenter i have problem because it load only once so i use ESL to add
> and remove agent ,if i use one FS it's ok  but more then one FS how can i
> add or remove agent to queue all FS using ESL
>
>
>
>
>
> Regard's
> NH45 CMI
>
> _________________________________________________________________________
> 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 Jul 30 20:10:48 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 30 Jul 2015 18:10:48 +0200
Subject: [Freeswitch-users] ESL Freeswitch cluster
In-Reply-To: <CAJUJwtikQ5qv8A+YR5hqfy4eCr_iV5gm2BwJsWtKpdw2A_NOhw@mail.gmail.com>
References: <CAAy72=Xjy5C67Z_XThAYa=tQmoU9Z1TdhFuhcHJmbYTGpkkoHA@mail.gmail.com>
	<CAJUJwtikQ5qv8A+YR5hqfy4eCr_iV5gm2BwJsWtKpdw2A_NOhw@mail.gmail.com>
Message-ID: <CALXCt0rffE_poVtTk6Ay=LKDkjOXGWPwCfEkCGirzxOqL9pzNg@mail.gmail.com>

You will probably need to have postgresql in core, but please check if the
modules you're using for callcenter functionalities support clustering via
db.

At the end of the day, if it becomes too much for your internal resources
to design and implement a solution, you may want to write
consulting at freeswitch.org for commercial (eg: paid) help on this.

-giovanni



On Thu, Jul 30, 2015 at 6:03 PM, SamyGo <govoiper at gmail.com> wrote:

> Hi NH45,
>
> Do you want to send the same ESL action to all of the FS Servers,
> something like a ESL Proxy that can relay your event to the FS Servers ?
>
> Regards,
> Sammy
>
> On Thu, Jul 30, 2015 at 8:12 AM, NH45 CMI <nh45cmi at gmail.com> wrote:
>
>> Hi Guys,
>>
>>
>>            I am using opensips for load balancing and freeswitch for
>> register,IVR,Callcenter ,Using mod_xml_curl for realtime ivr and dialplan
>> but callcenter i have problem because it load only once so i use ESL to add
>> and remove agent ,if i use one FS it's ok  but more then one FS how can i
>> add or remove agent to queue all FS using ESL
>>
>>
>>
>>
>>
>> Regard's
>> NH45 CMI
>>
>> _________________________________________________________________________
>> 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
Cell : +39-347-2665618
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From itsusama at gmail.com  Thu Jul 30 21:07:49 2015
From: itsusama at gmail.com (Usama Zaidi)
Date: Thu, 30 Jul 2015 13:07:49 -0400
Subject: [Freeswitch-users] Virtual DB FreeSWITCH
Message-ID: <CAKuGxCj_G1o0UMN-nWyJZOBe9nE+rngVU5qAvQRybKh-XPi52Q@mail.gmail.com>

Hi all,

I wanted to know if there's something similar to
http://www.opensips.org/About/News0035 available in freeSWITCH, I want to
write CDRs etc to DB but my environment doesn't support multicast/floating
IPs.

Regards.

-- 
I'd love to change the world, but they wont gimme the source code to it
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From aronp at guaranteedplus.com  Thu Jul 30 21:14:40 2015
From: aronp at guaranteedplus.com (Podrigal, Aron)
Date: Thu, 30 Jul 2015 13:14:40 -0400
Subject: [Freeswitch-users] ClueCon Extra ticket
Message-ID: <CANJp-yg254xth=7u0TMvdAnUg9UGuAGe9+fgiK+g+QmgE8L7hg@mail.gmail.com>

Hi All,

One of our team is unable to attend so we have an extra ticket to give
away. First come first serve. Please contact me in private  aron at
mongotel.com.

-- 
Aron Podrigal
-
'1000001', '1110010', '1101111', '1101110'   '1010000', '1101111',
'1100100', '1110010', '1101001', '1100111', '1100001', '1101100'

P: '2b', '31', '33', '34', '37', '34', '35', '38', '36', '30', '39', '39'
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From danny.gershman at gmail.com  Fri Jul 31 00:50:59 2015
From: danny.gershman at gmail.com (Danny Gershman)
Date: Thu, 30 Jul 2015 20:50:59 +0000
Subject: [Freeswitch-users] Endless playback in conference
In-Reply-To: <CAEorZymLJW3LhuEuM3hTCsw2UZmX_bPYo+_PLj4YDVs6cMmTjw@mail.gmail.com>
References: <CAEorZynEWmqnOAEfY00_k1Rwe6jPjsjNSPxJjVaTq_neOMFozg@mail.gmail.com>
	<01ec01d0c4cd$82fd1bf0$88f753d0$@botecomm.com>
	<CAEorZyn0TGWO9nd8Y4hBpWGPLFnag6k6dVSd7dYZCiCcCPzv8A@mail.gmail.com>
	<CAEJMVkDPHWqoPZ5oQ4Pwp5HJQK_eaqSAfZnoz_ZjqvoUpTwRhw@mail.gmail.com>
	<CAEorZy=PUzhRgMHJq8mWfw923A19uc2oZ1mg2UObEi-JVMoSXA@mail.gmail.com>
	<CAO3hyzHWqhFh-QDBJUFzvjFubDNJYFy_V3S=YW-dnNUOG_RVuA@mail.gmail.com>
	<CAEorZynSUb=kQM2znu7RxDp9gYRoYB0Be0JUJvRBPmNF3G9Szg@mail.gmail.com>
	<CACOYK=i4NSW0uSdS+4O3J2LbNYyxfG0_M3gvUcuB-AQfYGebpw@mail.gmail.com>
	<CAEorZymLJW3LhuEuM3hTCsw2UZmX_bPYo+_PLj4YDVs6cMmTjw@mail.gmail.com>
Message-ID: <CAEorZynTXMi5NajrX+Vyo_ZkF7jpaX-6Q=L_JcOc5szgvmA-WA@mail.gmail.com>

All the participants are muted while the leg "MOH leg" is there.

On Thu, Jul 30, 2015 at 8:28 AM Danny Gershman <danny.gershman at gmail.com>
wrote:

> Hmmm good point, didn't think of that.  I could deaf the line which would
> prevent it from picking up any audio and playing it back into the
> conference.
>
> On Thu, Jul 23, 2015 at 8:35 PM Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
>> But then other members are still able to hear some sound from each other
>> through the music. Is that OK with you?
>> On Jul 23, 2015 7:42 PM, "Danny Gershman" <danny.gershman at gmail.com>
>> wrote:
>>
>>> Ok what I did was the following, that seems to work.
>>>
>>> 1) Created a new dialplan extension
>>>
>>>     <extension name="dynamicmoh">
>>>       <condition field="destination_number" expression="^dynamicmoh
>>> -(.*)$">
>>>         <action application="endless_playback" data="$1" />
>>>       </condition>
>>>     </extension>
>>>
>>> 2) From a conference I do this
>>>
>>> conference <conf> bgdial
>>> sofia/internal/dynamicmoh-/path/to/file at server-ip
>>>
>>> 3) To end playback, just hup the member in the conference.
>>>
>>> Thanks.
>>> --Danny
>>>
>>> On Thu, Jul 23, 2015 at 12:15 PM Chris Tunbridge <blasterjr at gmail.com>
>>> wrote:
>>>
>>>> Danny the fastest solution i can come up with is doing an originate
>>>> that connects a hold music extension to the conference, then when you wanna
>>>> stop it, you just kick that member out.
>>>>
>>>> On Thu, Jul 23, 2015 at 9:28 AM, Danny Gershman <
>>>> danny.gershman at gmail.com> wrote:
>>>>
>>>>> I want to be able to play a file into a conference and have it loop
>>>>> forever, like music on hold does.  And then when I need to I want to stop
>>>>> it.
>>>>>
>>>> On Thu, Jul 23, 2015 at 11:20 AM Brian West <brian at freeswitch.org>
>>>>> wrote:
>>>>>
>>>> What is your goal here?  Maybe I missed the entire scenario.
>>>>>>
>>>>>> On Thu, Jul 23, 2015 at 10:07 AM, Danny Gershman <
>>>>>> danny.gershman at gmail.com> wrote:
>>>>>>
>>>>>>> Wouldn't a profile have to be created on the fly for this? From what
>>>>>>> I can see you cannot set this for a conference from an api call. Also to
>>>>>>> stop it, you would have to change the profile for the conference to remove
>>>>>>> it.
>>>>>>> On Wed, Jul 22, 2015 at 6:28 PM Bote Man <bote_radio at botecomm.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> There?s a parameter for mod_conference named ?perpetual-sound? that
>>>>>>>> looks like it would do the trick.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> It?s about 1/3 of the way down
>>>>>>>>
>>>>>>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_conference
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> PLEASE check further for any changes that might have been made in
>>>>>>>> the latest FreeSWITCH as the conference module has undergone substantial
>>>>>>>> changes and perpetual-sound might have been one of them.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Bote
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> *From:* Danny Gershman
>>>>>>>> *Sent:* Wednesday, 22 July, 2015 17:09
>>>>>>>> *Subject:* [Freeswitch-users] Endless playback in conference
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> I'm trying to do an endless playback of an mp3 file in a
>>>>>>>> conference.   I have a couple of ideas, but none seem really solid.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> 1) Pass a variable on play and monitor from mod_event_socket and
>>>>>>>> play again if not forcibly terminated.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> 2) Load up local_stream dynamically from an xmlhttp server, and
>>>>>>>> then restart the local stream service, however will interrupt MOH for other
>>>>>>>> users.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Any other ideas?  Is there a way to do looping for "api" through
>>>>>>>> mod_event_socket?  I know you can with "sendmsg"
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --Danny Gershman
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> 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*
>>>>>> 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
>>>>>>
>>>>>
>>>>>> _________________________________________________________________________
>>>>>> 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
>
>
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From italorossib at gmail.com  Fri Jul 31 05:39:21 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 30 Jul 2015 22:39:21 -0300
Subject: [Freeswitch-users] [CS_NEW] [WRONG_CALL_STATE]
In-Reply-To: <CAHtxdDe34ScODL6Mz498NbB9oHHzi2_MpMOzAiyyLnaJuxWERQ@mail.gmail.com>
References: <CAFoaQoAb6CA0rjGcfE6BY_GLMjjExW58BgqabSJuAiK7RX_YcA@mail.gmail.com>
	<CAHtxdDe34ScODL6Mz498NbB9oHHzi2_MpMOzAiyyLnaJuxWERQ@mail.gmail.com>
Message-ID: <CAJR7f2EKseCY9HgwV6g4p51roV+3i+HP_2_c-PLxcxTQuJXgUw@mail.gmail.com>

you changed auth-calls profile param?

On Thu, Jul 30, 2015 at 3:46 AM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Try analyze callflow via enabled siptrace in FS or via wireshark on vpn
> interface.
>
> On Thu, Jul 30, 2015, 06:40 Rajil Saraswat <rajil.s at gmail.com> wrote:
>
>> Hello,
>>
>> I have two freeswitch servers connected over vpn like the following:
>>
>> FS A <> openvpn<> FS B
>>
>> Server A has the ip address 172.16.1.2
>> Server B has the ip address 192.16.1.2
>> openvpn gateway is on 10.8.0.1
>>
>> Both the servers have the common acl defined to not nat the above
>> addresses like following:
>>
>> <list name="nat.auto" default="deny">
>>     <node type="deny" cidr="10.8.0.0/24"/>
>>     <node type="deny" cidr="172.16.1.0/24"/>
>>     <node type="deny" cidr="192.168.1.0/24"/>
>> </list>
>>
>>
>> <list name="localnet.auto" default="deny">
>>        <node type="allow" cidr="172.16.1.0/24"/>
>>        <node type="allow" cidr="10.8.0.0/24"/>
>>        <node type="allow" cidr="192.168.1.0/24"/>
>> </list>
>>
>>
>> I am able to call the extensions on server B from server A. However if
>> i try calling the extension on server A from server B i get an error
>> of [CS_NEW] [WRONG_CALL_STATE]. The only way i can make a successful
>> call from B to A is when i add the network of B in the acl on server A
>> as follows:
>>
>> <list name="domains" default="deny">
>> <node type="allow" domain="$${domain}"/>
>> <node type="allow" cidr="192.168.1.0/24"/>
>>
>>
>> Is there a reason why i need to do this?
>>
>> Thanks
>> Rajil
>>
>> _________________________________________________________________________
>> 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
>



-- 
?talo Rossi
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From nh45cmi at gmail.com  Fri Jul 31 07:41:00 2015
From: nh45cmi at gmail.com (NH45 CMI)
Date: Fri, 31 Jul 2015 09:11:00 +0530
Subject: [Freeswitch-users] ESL Freeswitch cluster
In-Reply-To: <CALXCt0rffE_poVtTk6Ay=LKDkjOXGWPwCfEkCGirzxOqL9pzNg@mail.gmail.com>
References: <CAAy72=Xjy5C67Z_XThAYa=tQmoU9Z1TdhFuhcHJmbYTGpkkoHA@mail.gmail.com>
	<CAJUJwtikQ5qv8A+YR5hqfy4eCr_iV5gm2BwJsWtKpdw2A_NOhw@mail.gmail.com>
	<CALXCt0rffE_poVtTk6Ay=LKDkjOXGWPwCfEkCGirzxOqL9pzNg@mail.gmail.com>
Message-ID: <CAAy72=W9D4OUXveCcUOd65SK6rp9pJLXjWgtKC+=3i3OT5T6dA@mail.gmail.com>

Hi Sammy,

             Yes i need to send samee ESL action to every FS

On Thu, Jul 30, 2015 at 9:40 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> You will probably need to have postgresql in core, but please check if the
> modules you're using for callcenter functionalities support clustering via
> db.
>
> At the end of the day, if it becomes too much for your internal resources
> to design and implement a solution, you may want to write
> consulting at freeswitch.org for commercial (eg: paid) help on this.
>
> -giovanni
>
>
>
> On Thu, Jul 30, 2015 at 6:03 PM, SamyGo <govoiper at gmail.com> wrote:
>
>> Hi NH45,
>>
>> Do you want to send the same ESL action to all of the FS Servers,
>> something like a ESL Proxy that can relay your event to the FS Servers ?
>>
>> Regards,
>> Sammy
>>
>> On Thu, Jul 30, 2015 at 8:12 AM, NH45 CMI <nh45cmi at gmail.com> wrote:
>>
>>> Hi Guys,
>>>
>>>
>>>            I am using opensips for load balancing and freeswitch for
>>> register,IVR,Callcenter ,Using mod_xml_curl for realtime ivr and dialplan
>>> but callcenter i have problem because it load only once so i use ESL to add
>>> and remove agent ,if i use one FS it's ok  but more then one FS how can i
>>> add or remove agent to queue all FS using ESL
>>>
>>>
>>>
>>>
>>>
>>> Regard's
>>> NH45 CMI
>>>
>>> _________________________________________________________________________
>>> 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
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> 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 govoiper at gmail.com  Fri Jul 31 08:04:05 2015
From: govoiper at gmail.com (SamyGo)
Date: Fri, 31 Jul 2015 00:04:05 -0400
Subject: [Freeswitch-users] ESL Freeswitch cluster
In-Reply-To: <CAAy72=W9D4OUXveCcUOd65SK6rp9pJLXjWgtKC+=3i3OT5T6dA@mail.gmail.com>
References: <CAAy72=Xjy5C67Z_XThAYa=tQmoU9Z1TdhFuhcHJmbYTGpkkoHA@mail.gmail.com>
	<CAJUJwtikQ5qv8A+YR5hqfy4eCr_iV5gm2BwJsWtKpdw2A_NOhw@mail.gmail.com>
	<CALXCt0rffE_poVtTk6Ay=LKDkjOXGWPwCfEkCGirzxOqL9pzNg@mail.gmail.com>
	<CAAy72=W9D4OUXveCcUOd65SK6rp9pJLXjWgtKC+=3i3OT5T6dA@mail.gmail.com>
Message-ID: <CAJUJwtgE+j00WZps1bAt2riFQ3SANGYmR_+HDRB2QtN6=mwxvA@mail.gmail.com>

I'd prefer the way Giovanni mentioned, if there is a common backend then
you dont need to send event to everyone, just to one and push the agent
value in the backend.
Givoanni, even if there is a common backend we still need to send API
signal to each FS to reload the stuff. right ?

On Thu, Jul 30, 2015 at 11:41 PM, NH45 CMI <nh45cmi at gmail.com> wrote:

> Hi Sammy,
>
>              Yes i need to send samee ESL action to every FS
>
> On Thu, Jul 30, 2015 at 9:40 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> You will probably need to have postgresql in core, but please check if
>> the modules you're using for callcenter functionalities support clustering
>> via db.
>>
>> At the end of the day, if it becomes too much for your internal resources
>> to design and implement a solution, you may want to write
>> consulting at freeswitch.org for commercial (eg: paid) help on this.
>>
>> -giovanni
>>
>>
>>
>> On Thu, Jul 30, 2015 at 6:03 PM, SamyGo <govoiper at gmail.com> wrote:
>>
>>> Hi NH45,
>>>
>>> Do you want to send the same ESL action to all of the FS Servers,
>>> something like a ESL Proxy that can relay your event to the FS Servers ?
>>>
>>> Regards,
>>> Sammy
>>>
>>> On Thu, Jul 30, 2015 at 8:12 AM, NH45 CMI <nh45cmi at gmail.com> wrote:
>>>
>>>> Hi Guys,
>>>>
>>>>
>>>>            I am using opensips for load balancing and freeswitch for
>>>> register,IVR,Callcenter ,Using mod_xml_curl for realtime ivr and dialplan
>>>> but callcenter i have problem because it load only once so i use ESL to add
>>>> and remove agent ,if i use one FS it's ok  but more then one FS how can i
>>>> add or remove agent to queue all FS using ESL
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Regard's
>>>> NH45 CMI
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>> Cell : +39-347-2665618
>>
>> _________________________________________________________________________
>> 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 ssinyagin at gmail.com  Fri Jul 31 12:44:07 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 31 Jul 2015 10:44:07 +0200
Subject: [Freeswitch-users] ESL Freeswitch cluster
In-Reply-To: <CAAy72=W9D4OUXveCcUOd65SK6rp9pJLXjWgtKC+=3i3OT5T6dA@mail.gmail.com>
References: <CAAy72=Xjy5C67Z_XThAYa=tQmoU9Z1TdhFuhcHJmbYTGpkkoHA@mail.gmail.com>
	<CAJUJwtikQ5qv8A+YR5hqfy4eCr_iV5gm2BwJsWtKpdw2A_NOhw@mail.gmail.com>
	<CALXCt0rffE_poVtTk6Ay=LKDkjOXGWPwCfEkCGirzxOqL9pzNg@mail.gmail.com>
	<CAAy72=W9D4OUXveCcUOd65SK6rp9pJLXjWgtKC+=3i3OT5T6dA@mail.gmail.com>
Message-ID: <CACOYK=jifpVimxPK5X_Uc-OOVHtcbRpL0BjxyvdNfqZxSctkew@mail.gmail.com>

you still need to control the result of action execution on every FS
node, so you need to create a dispatcher architecture.

For example, a message queue manager, like zeromq, would receive the
command, and send it to local FS nodes, and a locally installed daemon
would communicate to the local FS and send back the results via
zeromq.

Or it could be some multithreaded daemon, in Golang for example, which
would connect to all FS instances and send them the commands. You
would then manage the responses in your Go program.

Nothing impossible, you just need to match your requirements with your
budget and with the skills inside your operations team.





On Fri, Jul 31, 2015 at 5:41 AM, NH45 CMI <nh45cmi at gmail.com> wrote:
> Hi Sammy,
>
>              Yes i need to send samee ESL action to every FS
>
> On Thu, Jul 30, 2015 at 9:40 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>>
>> You will probably need to have postgresql in core, but please check if the
>> modules you're using for callcenter functionalities support clustering via
>> db.
>>
>> At the end of the day, if it becomes too much for your internal resources
>> to design and implement a solution, you may want to write
>> consulting at freeswitch.org for commercial (eg: paid) help on this.
>>
>> -giovanni
>>
>>
>>
>> On Thu, Jul 30, 2015 at 6:03 PM, SamyGo <govoiper at gmail.com> wrote:
>>>
>>> Hi NH45,
>>>
>>> Do you want to send the same ESL action to all of the FS Servers,
>>> something like a ESL Proxy that can relay your event to the FS Servers ?
>>>
>>> Regards,
>>> Sammy
>>>
>>> On Thu, Jul 30, 2015 at 8:12 AM, NH45 CMI <nh45cmi at gmail.com> wrote:
>>>>
>>>> Hi Guys,
>>>>
>>>>
>>>>            I am using opensips for load balancing and freeswitch for
>>>> register,IVR,Callcenter ,Using mod_xml_curl for realtime ivr and dialplan
>>>> but callcenter i have problem because it load only once so i use ESL to add
>>>> and remove agent ,if i use one FS it's ok  but more then one FS how can i
>>>> add or remove agent to queue all FS using ESL
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Regard's
>>>> NH45 CMI
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>> Cell : +39-347-2665618
>>
>> _________________________________________________________________________
>> 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


From alex at digitalmail.com  Fri Jul 31 14:03:31 2015
From: alex at digitalmail.com (Alex Lake)
Date: Fri, 31 Jul 2015 11:03:31 +0100
Subject: [Freeswitch-users] Complex phrase macro
In-Reply-To: <C0109557-B6AB-45C9-841D-035F865ACF05@jerris.com>
References: <55B8BC10.9000002@digitalmail.com>	<55B8CFDA.6000001@digitalmail.com>	<CACOYK=g+foyFRheKrPfB1iwLt+snBp1NERAQwpt1q5r_DUDrEQ@mail.gmail.com>
	<C0109557-B6AB-45C9-841D-035F865ACF05@jerris.com>
Message-ID: <55BB47F3.9070207@digitalmail.com>

It's because the whole dialplan is templated and I wanted to have one 
piece of data that could neatly go in.
Lua would have been a possibility, I guess, but I quite like the phrase 
macro system.

On 29/07/2015 18:16, Michael Jerris wrote:
> Or even dialplan. When would this actually add any value over maybe 
> just calling playback multiple times in a row?
>
>> On Jul 29, 2015, at 1:02 PM, Stanislav Sinyagin <ssinyagin at gmail.com 
>> <mailto:ssinyagin at gmail.com>> wrote:
>>
>> Why not just using Lua instead?
>>
>> On Jul 29, 2015 3:08 PM, "Alex Lake" <alex at digitalmail.com 
>> <mailto:alex at digitalmail.com>> wrote:
>>
>>     OK - Got this working. Just in case anyone is interested:
>>
>>          <macro name="xplayN">
>>              <input pattern="^([^,]*),([^,]*),(.*)$">
>>                  <match>
>>                      <action function="play-file" data="$1/$2.wav"/>
>>                      <action function="play-file"
>>     data="phrase:xplayN:$1,$3"/>
>>                  </match>
>>              </input>
>>              <input pattern="^([^,]*),([^,]*)$">
>>                  <match>
>>                      <action function="play-file" data="$1/$2.wav"/>
>>                  </match>
>>              </input>
>>          </macro>
>>
>>     Example of use:
>>
>>            <action application="playback"
>>     data="phrase:xplayN:/home/pabx/004-3774/x,001,0211,0221,0231"/>
>>
>>
>>
>>     _________________________________________________________________________
>>     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
>>     UNSUBSCRIBE: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
>>
>> 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 itsusama at gmail.com  Fri Jul 31 17:35:51 2015
From: itsusama at gmail.com (Usama Zaidi)
Date: Fri, 31 Jul 2015 09:35:51 -0400
Subject: [Freeswitch-users] Virtual DB FreeSWITCH
Message-ID: <CAKuGxCihiTJZW563oiV2Y-8OZoAc1uCr8wT6FvCKyPp-FnrRiQ@mail.gmail.com>

So I found the solution, using HAProxy would be a viable fix.

Regards!
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From krice at freeswitch.org  Fri Jul 31 18:01:42 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 31 Jul 2015 14:01:42 +0000
Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <55bb7fc685596_f526dcd32c888c4@resque-worker-high.2.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
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From rajil.s at gmail.com  Fri Jul 31 19:01:00 2015
From: rajil.s at gmail.com (Rajil Saraswat)
Date: Fri, 31 Jul 2015 10:01:00 -0500
Subject: [Freeswitch-users] [CS_NEW] [WRONG_CALL_STATE]
In-Reply-To: <CAJR7f2EKseCY9HgwV6g4p51roV+3i+HP_2_c-PLxcxTQuJXgUw@mail.gmail.com>
References: <CAFoaQoAb6CA0rjGcfE6BY_GLMjjExW58BgqabSJuAiK7RX_YcA@mail.gmail.com>
	<CAHtxdDe34ScODL6Mz498NbB9oHHzi2_MpMOzAiyyLnaJuxWERQ@mail.gmail.com>
	<CAJR7f2EKseCY9HgwV6g4p51roV+3i+HP_2_c-PLxcxTQuJXgUw@mail.gmail.com>
Message-ID: <CAFoaQoCGqTNb+UxjG0dVM3kg6UhOZYWH_y8dxefSPMEy6Qk0Cg@mail.gmail.com>

On Jul 30, 2015 8:41 PM, "?talo Rossi" <italorossib at gmail.com> wrote:
>
> you changed auth-calls profile param?
>
> On Thu, Jul 30, 2015 at 3:46 AM, Sergey Safarov <s.safarov at gmail.com>
wrote:
>>
>> Try analyze callflow via enabled siptrace in FS or via wireshark on vpn
interface.
>>
>>
>> On Thu, Jul 30, 2015, 06:40 Rajil Saraswat <rajil.s at gmail.com> wrote:
>>>
>>> Hello,
>>>
>>> I have two freeswitch servers connected over vpn like the following:
>>>
>>> FS A <> openvpn<> FS B
>>>
>>> Server A has the ip address 172.16.1.2
>>> Server B has the ip address 192.16.1.2
>>> openvpn gateway is on 10.8.0.1
>>>
>>> Both the servers have the common acl defined to not nat the above
>>> addresses like following:
>>>
>>> <list name="nat.auto" default="deny">
>>>     <node type="deny" cidr="10.8.0.0/24"/>
>>>     <node type="deny" cidr="172.16.1.0/24"/>
>>>     <node type="deny" cidr="192.168.1.0/24"/>
>>> </list>
>>>
>>>
>>> <list name="localnet.auto" default="deny">
>>>        <node type="allow" cidr="172.16.1.0/24"/>
>>>        <node type="allow" cidr="10.8.0.0/24"/>
>>>        <node type="allow" cidr="192.168.1.0/24"/>
>>> </list>
>>>
>>>
>>> I am able to call the extensions on server B from server A. However if
>>> i try calling the extension on server A from server B i get an error
>>> of [CS_NEW] [WRONG_CALL_STATE]. The only way i can make a successful
>>> call from B to A is when i add the network of B in the acl on server A
>>> as follows:
>>>
>>> <list name="domains" default="deny">
>>> <node type="allow" domain="$${domain}"/>
>>> <node type="allow" cidr="192.168.1.0/24"/>
>>>
>>>
>>> Is there a reason why i need to do this?
>>>
>>> Thanks
>>> Rajil
>>>
>>>
_________________________________________________________________________
>>> 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
>
>
>
>
> --
> ?talo Rossi
>
> _________________________________________________________________________
> 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

swk helped me solve this. The issue was my mtu was more than 1500. Once I
changed from using udp to tcp, the call started working again.
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From mario_fs at mgtech.com  Fri Jul 31 21:32:47 2015
From: mario_fs at mgtech.com (Mario)
Date: Fri, 31 Jul 2015 10:32:47 -0700
Subject: [Freeswitch-users] Complex phrase macro
In-Reply-To: <55BB47F3.9070207@digitalmail.com>
References: <55B8BC10.9000002@digitalmail.com>
	<55B8CFDA.6000001@digitalmail.com>
	<CACOYK=g+foyFRheKrPfB1iwLt+snBp1NERAQwpt1q5r_DUDrEQ@mail.gmail.com>
	<C0109557-B6AB-45C9-841D-035F865ACF05@jerris.com>
	<55BB47F3.9070207@digitalmail.com>
Message-ID: <BB8412C5-6EE5-452B-91CA-4D58AD4BEDB2@mgtech.com>

I like LUA for this. I have a LUA that builds long sentences that even include music in different places. The LUA builds the sound string from multiple files dynamically and even adjusts the length of various sections (like music) so the message is a fairly consistent length even though the parts are different. Yes, LUA is more work up front but then you can do much more and use the logic for multiple sound purposes.
Mario
> On Jul 31, 2015, at 3:03 AM, Alex Lake <alex at digitalmail.com> wrote:
> 
> It's because the whole dialplan is templated and I wanted to have one piece of data that could neatly go in.
> Lua would have been a possibility, I guess, but I quite like the phrase macro system.
> 
> On 29/07/2015 18:16, Michael Jerris wrote:
>> Or even dialplan. When would this actually add any value over maybe just calling playback multiple times in a row?
>> 
>>> On Jul 29, 2015, at 1:02 PM, Stanislav Sinyagin <ssinyagin at gmail.com <mailto:ssinyagin at gmail.com>> wrote:
>>> 
>>> Why not just using Lua instead?
>>> On Jul 29, 2015 3:08 PM, "Alex Lake" <alex at digitalmail.com <mailto:alex at digitalmail.com>> wrote:
>>> OK - Got this working. Just in case anyone is interested:
>>> 
>>>      <macro name="xplayN">
>>>          <input pattern="^([^,]*),([^,]*),(.*)$">
>>>              <match>
>>>                  <action function="play-file" data="$1/$2.wav"/>
>>>                  <action function="play-file" data="phrase:xplayN:$1,$3"/>
>>>              </match>
>>>          </input>
>>>          <input pattern="^([^,]*),([^,]*)$">
>>>              <match>
>>>                  <action function="play-file" data="$1/$2.wav"/>
>>>              </match>
>>>          </input>
>>>      </macro>
>>> 
>>> Example of use:
>>> 
>>>        <action application="playback"
>>> data="phrase:xplayN:/home/pabx/004-3774/x,001,0211,0221,0231"/>
>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> 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/>
>> 
>> 
>> _________________________________________________________________________
>> 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
> 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 ollie at w4les.net  Thu Jul 30 21:32:07 2015
From: ollie at w4les.net (Ollie Owens)
Date: Thu, 30 Jul 2015 17:32:07 +0000
Subject: [Freeswitch-users] ZRTP Mandatory for Calls
Message-ID: <9D11FE37990E404ABB5A83053088DAD1119FB0B7@Exch-17.MessageExchange.com>

Hello,

I'm currently struggling to find a way to force Freeswitch to only allow ZRTP based calls without a trusted MiTM.

I've tried to use "is_zrtp_secure" from the features.xml to hang the call up with the Dialplan but it isn't getting me anywhere.

I've also looked at using LUA scripts to detect when the call has been answered and then check on the variables set in switch_channel.c but this also doesn't work.

I can see from switch_channel.c that it prints "ZRTP not negotiated on both sides; disabling ZRTP passthru mode." and sets zrtp_passthru_active to false when this happens.

Can someone point me in the right direction for looking up the value of this var once the call has been answered to determine whether ZRTP support is available or if there is a better way to approach this?

Cheers,
Ollie
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