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

Steven Ayre steveayre at gmail.com
Mon Jan 14 15:50:02 MSK 2013 

Nevermind, just reread your original post


On 14 January 2013 12:49, Steven Ayre <steveayre at gmail.com> wrote:

> Try:
> conference <confname> play <file_path>
> conference <confname> pause <file_path>
> conference <confname> resume <file_path>
> conference <confname> stop <file_path>
>
> http://wiki.freeswitch.org/wiki/Mod_conference
>
>
>
> On 14 January 2013 07:47, 王迪 <wangd at alongtechnology.com.cn> wrote:
>
>> **
>>  "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
>>   Send FreeSWITCH-users mailing list submissions to
>> freeswitch-users at lists.freeswitch.org
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>>  When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of FreeSWITCH-users digest..."
>>  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
>> > 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
>> >
>> >
>>  ------------------------------------------------------------
>> ------------------------------------------------------------
>> 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
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> 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
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> 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
>> >> 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://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
>>   ------------------------------------------------------------
>> ------------------------------------------------------------
>> 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
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>   ------------------------------------------------------------
>> ------------------------------------------------------------
>> 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
>>   ------------------------------------------------------------
>> ------------------------------------------------------------
>> 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
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>   ------------------------------------------------------------
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://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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