[Freeswitch-users] How to pause and resume play an audio file in conference?

Mon Jan 14 10:47:33 MSK 2013

 "uuid_fileman" is for channel, not for a conference. If I used "uuid_fileman" to pause a channel, and I can not talk to this channel. I wanna play file in a conference, and pause the playing and talk, then stop talking and resume the playing. 

发件人： freeswitch-users-request 
发送时间： 2013-01-10  22:12:37 
收件人： freeswitch-users 
抄送： 
主题： FreeSWITCH-users Digest, Vol 79, Issue 54 

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Today's Topics:
   1. Re: How to pause and resume play an audio file in conference?
      (Avi Marcus)
   2. Re: Best practices question about SIP registration (Steven Ayre)
   3. Re: mod_com_g729 transcoding (Steven Ayre)
   4. Early media without bridge (Tamas.Cseke )
   5. Loopback Endpoint (Jon Sch?pzinsky)
   6. Re: mod_directory menu-top? (Abaci)
------------------------------------------------------------
From:  Avi Marcus <avi at avimarcus.net>
To:  FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] How to pause and resume play an audio filein conference?
Date:  Thu10 Jan 2013 11:31:49 +0200
It looks like you can do it via api command:
http://wiki.freeswitch.org/wiki/Mod_commands#uuid_fileman
It seems the conference has stop-talking and start-talking events, so you
can start the file and then have an ESL app that listens for the events and
stops/starts the playback.
Let us know how that goes.
-Avi
On Thu, Jan 10, 2013 at 3:39 AM, 王迪 <wangd at alongtechnology.com.cn> wrote:
> **
>  How to pause and resume play an audio file in conference?  For example:
> I create a conference room with some members, and play an audio file to
> them. When someone start-talking, I pause the playing. When nobody talking,
> I resume play the audio file.
>
> BTW, In freeswitch-1.2rc, I tryed play,pause,resume commands, but pause
> and resume is for recording, not for my needs.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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> http://wiki.freeswitch.org
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>
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>
------------------------------------------------------------
------------------------------------------------------------
From:  Steven Ayre <steveayre at gmail.com>
To:  FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] Best practices question about SIP registration
Date:  Thu10 Jan 2013 09:31:29 +0000
> There IS a wrinkle to this special case: If you have a URL of the form <sip:user at example.com:5055> the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host
The reason being that a SRV record includes the port (note that means
it's a very easy way to add/change additional ports as well as
addresses without having to change customer configs).
So obviously a host:port isn't compatible with a SRV, since it
wouldn't make sense to override the port, so the assumption is that
it's specifying an A/AAAA record instead and port on that IP instead.
-Steve
On 8 January 2013 23:35, Lawrence Conroy <lconroy at insensate.co.uk> wrote:
> Hi Michael, folks,
>  if you're going to codify it, I did slightly simplify things:
> (full disclosure -- I disagreed with it at the time which is probably why I forgot to mention it -- honest).
>
> There IS a wrinkle to this special case: If you have a URL of the form <sip:user at example.com:5055> the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host example.com (i.e., there's a machine called example.com, and it is handling SIP traffic for that domainpart).
> Basically, if you see a colon in the domainpart, you're looking for a machine -- otherwise you're looking for a NAPTR (and/or a SRV at _sip._udp.<sipdomain>).
>
> I'd put that before the paragraph starting "However, relying on ..."
>
> Curiously enough, the old 2543-compliant servers did hunt the SRV rather than giving up and looking for an A, so this was a change. Such fun was had re-writing implementations (plural) and testing them yet again. Sigh.
>
> all the best,
>   Lawrence
>
>
> On 8 Jan 2013, at 22:56, Michael Collins wrote:
>
>> Lawrence,
>>
>> Thanks for this explanation. It was very well written. I'm looking for a
>> place to codify this on the wiki so that it gets preserved... :)
>>
>> -MC
>>
>> On Tue, Jan 8, 2013 at 1:56 PM, Lawrence Conroy <lconroy at insensate.co.uk>wrote:
>>
>>> Hi there,
>>> at the risk of butting in on someone else's party ...
>>> Nope; your interpretations is NOT best practice.
>>> I have some sympathy, as the term domain is overloaded within fS.
>>>
>>> A sip address consists of a userpart and a domain part -- e.g.,
>>> <sip:user at sipdomain>
>>> The sip domain is similar to an email domain -- e.g., <mailto:
>>> user at maildomain>
>>> With email, you need to do a lookup of the MX record in DNS to find the
>>> FQDN of the machine that handles mail for the domain.
>>> With SIP (see RFC 3263), you do a lookup on the SRV record (at
>>> _sip._udp.<sipdomain>) to find the machine that handles SIP
>>> registrations/incalls for the domain. That also gives you the port on
>>> which that machine is listening.
>>> (Yup, you can also have a NAPTR record in the domain to tell you where the
>>> SRV record is, but many folks don't bother -- for Best Practice, you
>>> should, but ...)
>>>
>>> There IS a "get out" clause in the SIP specs for RFC 2543 (AKA legacy)
>>> support that means most SIP clients will look for the SRV and, if it can't
>>> be found (or there's an IP address rather than a DNS -style domain, in
>>> which case the SIP client won't bother hunting the SRV), the client will
>>> guess that the domain has a machine (i.e. it will look for an A or AAAA
>>> record), and also guess it's listening on 5060 (the default port).
>>> Email is the same (mail to fred at example.com, and strictly the sender will
>>> do a check for a MX and then look for an A record for example.com, and
>>> try there).
>>>
>>> However, relying on that default "get out" clause is definitely NOT what
>>> you should do for BCP.
>>> Using the hostname as the sip domain is a kludge -- the FQDN with A record
>>> usually works, but it's not what you want to do.
>>>
>>> SO ... get yourself a domain, put a D2U NAPTR at that domain, put a SRV at
>>> _sip._udp.<domain>, and you're done. No need for an A record at that domain
>>> at all.
>>>
>>> (RFC 3263 is not too hard to read, for a change -- it's certainly shorter
>>> than RFC 3261, and it even has an ASCII art diagram :).
>>>
>>> all the best,
>>>  Lawrence
>>>
>>> On 8 Jan 2013, at 21:05, Steven Schoch wrote:
>>>
>>>> On Fri, Dec 28, 2012 at 8:47 PM, Tim St. Pierre <
>>>> fs-list at communicatefreely.net> wrote:
>>>>
>>>>> Hi Steven,
>>>>>
>>>>> I would recommend using a proper domain name as much as possible.  For
>>>>> one, it looks
>>>>> nicer!  A SIP URI is supposed to be user at domain like an e-mail address
>>>>> is, and I hope that
>>>>> one day URI dialing will be common place, so we might as well do it
>>> right
>>>>> the first time.
>>>>>
>>>>
>>>> What you're saying is that "domain" should really be a fully-qualified
>>> host
>>>> name that points via DNS to the actual host on which FreeSwitch is
>>> running.
>>>> That is, the domain should be "pbx.example.com" instead of just "
>>>> example.com", as the last example would most likely point to a web
>>> server,
>>>> not the SIP server.  Do I have that right?
>>>>
>>>> Next, in the configuration for Polycom phones (for example), there are 2
>>>> fields that both have the userid.  In the example in
>>>> http://wiki.freeswitch.org/wiki/Polycom_configuration it has:
>>>>
>>>> reg.1.auth.userId="1000"
>>>>
>>>> and
>>>>
>>>> reg.1.address="1000 at fs.domain.local"
>>>>
>>>> How is the "address" value used?  Is that sent in the SIP registration
>>>> message?  If that's the case, what does Freeswitch do with it?
>>>>
>>>> --
>>>> Steve
>>>> _________________________________________________________________________
>>>> 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-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
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>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> 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-users mailing list
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>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
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>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
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>
>
> _________________________________________________________________________
> 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-users mailing list
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------------------------------------------------------------
------------------------------------------------------------
From:  Steven Ayre <steveayre at gmail.com>
To:  FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] mod_com_g729 transcoding
Date:  Thu10 Jan 2013 09:38:52 +0000
A more complete debug-level log could be useful to get more context.
Is box 2 running any other calls at the same time that might be using
the license?
Is box 2 doing anything else with the call? Anything like eavesdrop,
recording, start_dtmf / start_dtmf_generate etc which uses the media
will use a license, and I *think* that license only gets released
until the end of the call.
-Steve
On 10 January 2013 07:09, Colin Mason <cmason at frontiernetworks.ca> wrote:
> I have phone A connected to freeswitch box 1 and phone B connected to
> freeswitch box 2.
>
>
>
> Phone A wants to dial phone B using G729. Codec is always G729.
>
> The path RTP follows is:
>
>
>
>
>
>
>
> Phone A -------> FreeSWITCH 1 -------> FreeSWITCH 2 -------> Phone B
>
>                    (g729)                                (g729)
> (g729)
>
>
>
>
>
>
>
> My question is, why is it that the freeswitch box receiving the call uses up
> an encoder/decoder when the codec is G729 along the path? If I reverse the
> call and call Phone A from Phone B, FreeSWITCH Box 1 uses up 1 license.
>
> if I dial the PSTN to a carrier who supports G729, I don’t use up a license.
> Any thoughts? Maybe this is normal.
>
>
>
> FreeSWITCH Box 1:
>
> 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP
> [sofia/mpls/I888_1 at 172.17.17.17] 172.17.17.17 port 32014 -> 10.253.200.6
> port 16466 codec: 18 ms: 20
>
> 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP
> [sofia/transport/2996] 172.17.17.17 port 27276 -> 172.16.16.16 port 17056
> codec: 18 ms: 20
>
> freeswitch at internal> g729_info
>
> Permitted G729 channels: 40
>
> Encoders in use: 0
>
> Decoders in use: 0
>
>
>
> FreeSWITCH Box 2:
>
> 2013-01-10 01:50:14.184757 [DEBUG] sofia_glue.c:3351 AUDIO RTP
> [sofia/transport/3888 at 172.17.17.17] 172.16.16.16 port 17056 -> 172.17.17.17
> port 27276 codec: 18 ms: 20
>
> 2013-01-10 01:50:15.664733 [DEBUG] sofia_glue.c:3351 AUDIO RTP
> [sofia/mpls/sip:C996_1 at 10.253.200.10:5060] 172.16.16.16 port 18966 ->
> 10.253.200.10 port 16486 codec: 18 ms: 20
>
> freeswitch at internal> g729_info
>
> Permitted G729 channels: 40
>
> Encoders in use: 1
>
> Decoders in use: 1
>
>
>
>
>
>
>
> Thanks in advance guys.
>
> Colin
>
>
> _________________________________________________________________________
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
------------------------------------------------------------
------------------------------------------------------------
From:  "Tamas.Cseke " <cstomi.levlist at gmail.com>
To:  FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject:  [Freeswitch-users] Early media without bridge
Date:  Thu10 Jan 2013 12:24:00 +0100
Hello,
We would like to hear early media without CHANNEL_BRIDGE event
These are failed calls and callers would like to hear the message that 
the provider plays
Because the caller thinks the call is answered if originate returns.
as far as I understand:
  -early media makes the originate return
  -if we ignore early media the bridge  won't return, but we don't hear it
we would like both of them, is it possible somehow?
I 'm not sure I fully understand all of the ignore_early_media options
but I haven't find solution for this,
Could you please advise me one, if there is any?
I'm thinking about we maybe need a new ignore_early_media option
like "consume" but sending the media to the caller instead of dropping it
If there isn't already a solution I also would appreciate if you let me 
know your opinion about this idea
Thanks advance,
Tamas
------------------------------------------------------------
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From:  Jon_Sch鴓zinsky<jos at firstcom.dk>
To:  "freeswitch-users at lists.freeswitch.org"<freeswitch-users at lists.freeswitch.org>
Subject:  [Freeswitch-users] Loopback Endpoint
Date:  Thu10 Jan 2013 14:36:06 +0100
Hello List,
I can see that the "this will destroy the world and this may kill your pets" warning has been removed from the documentation for the Loopback endpoint.
Is this an indication that it has become safer to use?
I have a specific problem where I essentially have to dial a dialplan for each user I am trying to reach, in parallel.
Is it correctly understood that loopback would be the best/only way of implementing this, or is there another way?
Kind Regards
Jon Schøpzinsky
------------------------------------------------------------
------------------------------------------------------------
From:  Abaci <abaci64 at gmail.com>
To:  FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] mod_directory menu-top?
Date:  Thu10 Jan 2013 09:11:09 -0500
use 'execute_extension' to start the directory application so that you 
get back to the ivr when you exit the directory application.
On 1/9/2013 6:41 PM, Phillip Warner wrote:
> Hi, is there a parameter in mod_directory to have it transfer (back) to an ivr if the user decides not to search by directory instead of having to hang-up and call again?
>
> For example: ivr -->  directory --> user changes mind about dialling by name and wants to return to ivr --> ivr
>
> Thanks. Phil.
> _________________________________________________________________________
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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