[Freeswitch-users] How to pause and resume play an audio file in conference?
Steven Ayre
steveayre at gmail.com
Mon Jan 14 15:51:14 MSK 2013
I suspect it needs a new feature to be added to mod_conference. You could
file a wishlist Jira.
Or as a workaround, originate a channel that calls into the conference and
play the audio on that channel. You'd then be able to use uuid_fileman on
that channel.
-Steve
On 14 January 2013 12:50, Steven Ayre <steveayre at gmail.com> wrote:
> 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
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>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
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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
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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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