[Freeswitch-users] How to pause and resume play an audio file in conference?
Steven Ayre
steveayre at gmail.com
Mon Jan 14 15:49:33 MSK 2013
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
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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
> > 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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> > 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
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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://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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>
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