I suspect it needs a new feature to be added to mod_conference. You could file a wishlist Jira.<div><br></div><div>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.</div>
<div><br></div><div>-Steve</div><div><br></div><div><br></div><div><br><div><br><div class="gmail_quote">On 14 January 2013 12:50, Steven Ayre <span dir="ltr"><<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Nevermind, just reread your original post<div class="HOEnZb"><div class="h5"><div><br><br><div class="gmail_quote">On 14 January 2013 12:49, Steven Ayre <span dir="ltr"><<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Try:<div><div>conference <confname> play <file_path></div><div>conference <confname> pause <file_path></div>
<div>conference <confname> resume <file_path></div><div>conference <confname> stop <file_path></div>
<div><br></div><div><a href="http://wiki.freeswitch.org/wiki/Mod_conference" target="_blank">http://wiki.freeswitch.org/wiki/Mod_conference</a></div><div><div><div><br></div><div><br></div><br><div class="gmail_quote">
On 14 January 2013 07:47, 王迪 <span dir="ltr"><<a href="mailto:wangd@alongtechnology.com.cn" target="_blank">wangd@alongtechnology.com.cn</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><u></u>
<div style="FONT-FAMILY:verdana;FONT-SIZE:10pt">
<div><font color="#c0c0c0" face="Verdana"><span> <font color="#000000">"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. </font></span></font></div>
<div><font color="#c0c0c0" face="Verdana"><span><font color="#000000"></font> </span></font></div><font color="#000080" face="Verdana">
<hr>
</font>
<div><font face="Verdana"><strong>发件人:</strong> freeswitch-users-request
</font></div>
<div><font face="Verdana"><strong>发送时间:</strong> <a href="tel:2013-01-10%C2%A0%2022" value="+12013011022" target="_blank">2013-01-10 22</a>:12:37
</font></div>
<div><font face="Verdana"><strong>收件人:</strong> freeswitch-users
</font></div>
<div><font face="Verdana"><strong>抄送:</strong> </font></div>
<div><font face="Verdana"><strong>主题:</strong> FreeSWITCH-users Digest, Vol
79, Issue 54 </font></div>
<div><font face="Verdana"></font> </div>
<div><font face="Verdana">
<div>Send FreeSWITCH-users mailing list submissions to</div>
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<div>When replying, please edit your Subject line so it is more specific</div>
<div>than "Re: Contents of FreeSWITCH-users digest..."</div>
<div></div>
<div>Today's Topics:</div>
<div></div>
<div> 1. Re: How to pause and resume play an audio file
in conference?</div>
<div> (Avi Marcus)</div>
<div> 2. Re: Best practices question about SIP
registration (Steven Ayre)</div>
<div> 3. Re: mod_com_g729 transcoding (Steven Ayre)</div>
<div> 4. Early media without bridge (Tamas.Cseke )</div>
<div> 5. Loopback Endpoint (Jon Sch?pzinsky)</div>
<div> 6. Re: mod_directory menu-top? (Abaci)</div>
<div></div>
<div>------------------------------------------------------------</div>
<div>From: Avi Marcus <<a href="mailto:avi@avimarcus.net" target="_blank">avi@avimarcus.net</a>></div>
<div>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>></div>
<div>Subject: Re: [Freeswitch-users] How to pause and resume play an audio filein conference?</div>
<div>Date: Thu10 Jan 2013 11:31:49 +0200</div><div>
<div></div>
<div>It looks like you can do it via api command:</div>
<div><a href="http://wiki.freeswitch.org/wiki/Mod_commands#uuid_fileman" target="_blank">http://wiki.freeswitch.org/wiki/Mod_commands#uuid_fileman</a></div>
<div></div>
<div>It seems the conference has stop-talking and start-talking events, so you</div>
<div>can start the file and then have an ESL app that listens for the events and</div>
<div>stops/starts the playback.</div>
<div></div>
<div>Let us know how that goes.</div>
<div></div>
<div>-Avi</div>
<div></div>
<div>On Thu, Jan 10, 2013 at 3:39 AM, 王迪 <<a href="mailto:wangd@alongtechnology.com.cn" target="_blank">wangd@alongtechnology.com.cn</a>> wrote:</div>
<div></div>
</div><div>> **</div><div>
<div>> How to pause and resume play an audio file in conference? For example:</div>
<div>> I create a conference room with some members, and play an audio file to</div>
<div>> them. When someone start-talking, I pause the playing. When nobody talking,</div>
<div>> I resume play the audio file.</div>
<div>></div>
<div>> BTW, In freeswitch-1.2rc, I tryed play,pause,resume commands, but pause</div>
<div>> and resume is for recording, not for my needs.</div>
<div>></div>
<div>></div>
<div>> _________________________________________________________________________</div>
<div>> Professional FreeSWITCH Consulting Services:</div>
<div>> <a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a></div>
<div>> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a></div>
<div>></div>
<div>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server</div>
<div>> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a></div>
<div>></div>
<div>> Official FreeSWITCH Sites</div>
<div>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a></div>
<div>> <a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a></div>
<div>> <a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a></div>
<div>></div>
<div>> FreeSWITCH-users mailing list</div>
<div>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a></div>
<div>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a></div>
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<div>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a></div>
<div>></div>
<div>></div>
<div></div>
</div><div>------------------------------------------------------------</div>
<div>------------------------------------------------------------</div>
<div>From: Steven Ayre <<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>></div>
<div>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>></div>
<div>Subject: Re: [Freeswitch-users] Best practices question about SIP registration</div>
<div>Date: Thu10 Jan 2013 09:31:29 +0000</div>
<div></div>
<div>> There IS a wrinkle to this special case: If you have a URL of the form <<a href="http://sip:user@example.com:5055" target="_blank">sip:user@example.com:5055</a>> the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host</div>
<div></div>
<div>The reason being that a SRV record includes the port (note that means</div>
<div>it's a very easy way to add/change additional ports as well as</div>
<div>addresses without having to change customer configs).</div>
<div></div>
<div>So obviously a host:port isn't compatible with a SRV, since it</div>
<div>wouldn't make sense to override the port, so the assumption is that</div>
<div>it's specifying an A/AAAA record instead and port on that IP instead.</div>
<div></div>
<div>-Steve</div>
<div></div>
<div></div>
<div></div>
<div></div>
<div></div>
<div>On 8 January 2013 23:35, Lawrence Conroy <<a href="mailto:lconroy@insensate.co.uk" target="_blank">lconroy@insensate.co.uk</a>> wrote:</div>
<div>> Hi Michael, folks,</div>
<div>> if you're going to codify it, I did slightly simplify things:</div>
<div>> (full disclosure -- I disagreed with it at the time which is probably why I forgot to mention it -- honest).</div>
<div>></div>
<div>> There IS a wrinkle to this special case: If you have a URL of the form <<a href="http://sip:user@example.com:5055" target="_blank">sip:user@example.com:5055</a>> the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host <a href="http://example.com" target="_blank">example.com</a> (i.e., there's a machine called <a href="http://example.com" target="_blank">example.com</a>, and it is handling SIP traffic for that domainpart).</div>
<div>> 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>).</div>
<div>></div>
<div>> I'd put that before the paragraph starting "However, relying on ..."</div>
<div>></div>
<div>> 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.</div>
<div>></div>
<div>> all the best,</div>
<div>> Lawrence</div>
<div>></div>
<div>></div>
<div>> On 8 Jan 2013, at 22:56, Michael Collins wrote:</div>
<div>></div>
<div>>> Lawrence,</div>
<div>>></div>
<div>>> Thanks for this explanation. It was very well written. I'm looking for a</div>
<div>>> place to codify this on the wiki so that it gets preserved... :)</div>
<div>>></div>
<div>>> -MC</div>
<div>>></div>
<div>>> On Tue, Jan 8, 2013 at 1:56 PM, Lawrence Conroy <<a href="mailto:lconroy@insensate.co.uk" target="_blank">lconroy@insensate.co.uk</a>>wrote:</div>
<div>>></div>
<div>>>> Hi there,</div>
<div>>>> at the risk of butting in on someone else's party ...</div>
<div>>>> Nope; your interpretations is NOT best practice.</div>
<div>>>> I have some sympathy, as the term domain is overloaded within fS.</div>
<div>>>></div>
<div>>>> A sip address consists of a userpart and a domain part -- e.g.,</div>
<div>>>> <sip:user@sipdomain></div>
<div>>>> The sip domain is similar to an email domain -- e.g., <mailto:</div>
<div>>>> user@maildomain></div>
<div>>>> With email, you need to do a lookup of the MX record in DNS to find the</div>
<div>>>> FQDN of the machine that handles mail for the domain.</div>
<div>>>> With SIP (see RFC 3263), you do a lookup on the SRV record (at</div>
<div>>>> _sip._udp.<sipdomain>) to find the machine that handles SIP</div>
<div>>>> registrations/incalls for the domain. That also gives you the port on</div>
<div>>>> which that machine is listening.</div>
<div>>>> (Yup, you can also have a NAPTR record in the domain to tell you where the</div>
<div>>>> SRV record is, but many folks don't bother -- for Best Practice, you</div>
<div>>>> should, but ...)</div>
<div>>>></div>
<div>>>> There IS a "get out" clause in the SIP specs for RFC 2543 (AKA legacy)</div>
<div>>>> support that means most SIP clients will look for the SRV and, if it can't</div>
<div>>>> be found (or there's an IP address rather than a DNS -style domain, in</div>
<div>>>> which case the SIP client won't bother hunting the SRV), the client will</div>
<div>>>> guess that the domain has a machine (i.e. it will look for an A or AAAA</div>
<div>>>> record), and also guess it's listening on 5060 (the default port).</div>
<div>>>> Email is the same (mail to <a href="mailto:fred@example.com" target="_blank">fred@example.com</a>, and strictly the sender will</div>
<div>>>> do a check for a MX and then look for an A record for <a href="http://example.com" target="_blank">example.com</a>, and</div>
<div>>>> try there).</div>
<div>>>></div>
<div>>>> However, relying on that default "get out" clause is definitely NOT what</div>
<div>>>> you should do for BCP.</div>
<div>>>> Using the hostname as the sip domain is a kludge -- the FQDN with A record</div>
<div>>>> usually works, but it's not what you want to do.</div>
<div>>>></div>
<div>>>> SO ... get yourself a domain, put a D2U NAPTR at that domain, put a SRV at</div>
<div>>>> _sip._udp.<domain>, and you're done. No need for an A record at that domain</div>
<div>>>> at all.</div>
<div>>>></div>
<div>>>> (RFC 3263 is not too hard to read, for a change -- it's certainly shorter</div>
<div>>>> than RFC 3261, and it even has an ASCII art diagram :).</div>
<div>>>></div>
<div>>>> all the best,</div>
<div>>>> Lawrence</div>
<div>>>></div>
<div>>>> On 8 Jan 2013, at 21:05, Steven Schoch wrote:</div>
<div>>>></div>
<div>>>>> On Fri, Dec 28, 2012 at 8:47 PM, Tim St. Pierre <</div>
<div>>>>> <a href="mailto:fs-list@communicatefreely.net" target="_blank">fs-list@communicatefreely.net</a>> wrote:</div>
<div>>>>></div>
<div>>>>>> Hi Steven,</div>
<div>>>>>></div>
<div>>>>>> I would recommend using a proper domain name as much as possible. For</div>
<div>>>>>> one, it looks</div>
<div>>>>>> nicer! A SIP URI is supposed to be user@domain like an e-mail address</div>
<div>>>>>> is, and I hope that</div>
<div>>>>>> one day URI dialing will be common place, so we might as well do it</div>
<div>>>> right</div>
<div>>>>>> the first time.</div>
<div>>>>>></div>
<div>>>>></div>
<div>>>>> What you're saying is that "domain" should really be a fully-qualified</div>
<div>>>> host</div>
<div>>>>> name that points via DNS to the actual host on which FreeSwitch is</div>
<div>>>> running.</div>
<div>>>>> That is, the domain should be "<a href="http://pbx.example.com" target="_blank">pbx.example.com</a>" instead of just "</div>
<div>>>>> <a href="http://example.com" target="_blank">example.com</a>", as the last example would most likely point to a web</div>
<div>>>> server,</div>
<div>>>>> not the SIP server. Do I have that right?</div>
<div>>>>></div>
<div>>>>> Next, in the configuration for Polycom phones (for example), there are 2</div>
<div>>>>> fields that both have the userid. In the example in</div>
<div>>>>> <a href="http://wiki.freeswitch.org/wiki/Polycom_configuration" target="_blank">http://wiki.freeswitch.org/wiki/Polycom_configuration</a> it has:</div>
<div>>>>></div>
<div>>>>> reg.1.auth.userId="1000"</div>
<div>>>>></div>
<div>>>>> and</div>
<div>>>>></div>
<div>>>>> reg.1.address="1000@fs.domain.local"</div>
<div>>>>></div>
<div>>>>> How is the "address" value used? Is that sent in the SIP registration</div>
<div>>>>> message? If that's the case, what does Freeswitch do with it?</div>
<div>>>>></div>
<div>>>>> --</div>
<div>>>>> Steve</div><div><div>
<div>>>>> _________________________________________________________________________</div>
<div>>>>> Professional FreeSWITCH Consulting Services:</div>
<div>>>>> <a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a></div>
<div>>>>> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a></div>
<div>>>>></div>
<div>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server</div>
<div>>>>> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a></div>
<div>>>>></div>
<div>>>>> Official FreeSWITCH Sites</div>
<div>>>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a></div>
<div>>>>> <a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a></div>
<div>>>>> <a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a></div>
<div>>>>></div>
<div>>>>> FreeSWITCH-users mailing list</div>
<div>>>>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a></div>
<div>>>>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a></div>
<div>>>>> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a></div>
<div>>>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a></div>
<div>>>></div>
<div>>>></div>
<div>>>> _________________________________________________________________________</div>
<div>>>> Professional FreeSWITCH Consulting Services:</div>
<div>>>> <a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a></div>
<div>>>> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a></div>
<div>>>></div>
<div>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server</div>
<div>>>> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a></div>
<div>>>></div>
<div>>>> Official FreeSWITCH Sites</div>
<div>>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a></div>
<div>>>> <a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a></div>
<div>>>> <a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a></div>
<div>>>></div>
<div>>>> FreeSWITCH-users mailing list</div>
<div>>>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a></div>
<div>>>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a></div>
<div>>>> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a></div>
<div>>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a></div>
<div>>>></div>
<div>>></div>
<div>>></div>
<div>>></div>
</div></div><div>>> --</div>
<div>>> Michael S Collins</div>
<div>>> Twitter: @mercutioviz</div>
<div>>> <a href="http://www.FreeSWITCH.org" target="_blank">http://www.FreeSWITCH.org</a></div>
<div>>> <a href="http://www.ClueCon.com" target="_blank">http://www.ClueCon.com</a></div>
<div>>> <a href="http://www.OSTAG.org" target="_blank">http://www.OSTAG.org</a></div><div><div>
<div>>> _________________________________________________________________________</div>
<div>>> Professional FreeSWITCH Consulting Services:</div>
<div>>> <a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a></div>
<div>>> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a></div>
<div>>></div>
<div>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server</div>
<div>>> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a></div>
<div>>></div>
<div>>> Official FreeSWITCH Sites</div>
<div>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a></div>
<div>>> <a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a></div>
<div>>> <a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a></div>
<div>>></div>
<div>>> FreeSWITCH-users mailing list</div>
<div>>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a></div>
<div>>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a></div>
<div>>> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a></div>
<div>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a></div>
<div>></div>
<div>></div>
<div>> _________________________________________________________________________</div>
<div>> Professional FreeSWITCH Consulting Services:</div>
<div>> <a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a></div>
<div>> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a></div>
<div>></div>
<div>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server</div>
<div>> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a></div>
<div>></div>
<div>> Official FreeSWITCH Sites</div>
<div>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a></div>
<div>> <a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a></div>
<div>> <a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a></div>
<div>></div>
<div>> FreeSWITCH-users mailing list</div>
<div>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a></div>
<div>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a></div>
<div>> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a></div>
<div>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a></div>
<div></div>
<div></div>
<div></div>
</div></div><div>------------------------------------------------------------</div>
<div>------------------------------------------------------------</div>
<div>From: Steven Ayre <<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>></div>
<div>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>></div>
<div>Subject: Re: [Freeswitch-users] mod_com_g729 transcoding</div>
<div>Date: Thu10 Jan 2013 09:38:52 +0000</div>
<div></div>
<div>A more complete debug-level log could be useful to get more context.</div>
<div></div>
<div>Is box 2 running any other calls at the same time that might be using</div>
<div>the license?</div>
<div></div>
<div>Is box 2 doing anything else with the call? Anything like eavesdrop,</div>
<div>recording, start_dtmf / start_dtmf_generate etc which uses the media</div>
<div>will use a license, and I *think* that license only gets released</div>
<div>until the end of the call.</div>
<div></div>
<div>-Steve</div>
<div></div>
<div></div>
<div></div>
<div>On 10 January 2013 07:09, Colin Mason <<a href="mailto:cmason@frontiernetworks.ca" target="_blank">cmason@frontiernetworks.ca</a>> wrote:</div>
<div>> I have phone A connected to freeswitch box 1 and phone B connected to</div>
<div>> freeswitch box 2.</div>
<div>></div>
<div>></div>
<div>></div>
<div>> Phone A wants to dial phone B using G729. Codec is always G729.</div>
<div>></div>
<div>> The path RTP follows is:</div>
<div>></div>
<div>></div>
<div>></div>
<div>></div>
<div>></div>
<div>></div>
<div>></div>
<div>> Phone A -------> FreeSWITCH 1 -------> FreeSWITCH 2 -------> Phone B</div>
<div>></div>
<div>> (g729) (g729)</div>
<div>> (g729)</div>
<div>></div>
<div>></div>
<div>></div>
<div>></div>
<div>></div>
<div>></div>
<div>></div>
<div>> My question is, why is it that the freeswitch box receiving the call uses up</div>
<div>> an encoder/decoder when the codec is G729 along the path? If I reverse the</div>
<div>> call and call Phone A from Phone B, FreeSWITCH Box 1 uses up 1 license.</div>
<div>></div>
<div>> if I dial the PSTN to a carrier who supports G729, I don’t use up a license.</div>
<div>> Any thoughts? Maybe this is normal.</div>
<div>></div>
<div>></div>
<div>></div>
<div>> FreeSWITCH Box 1:</div>
<div>></div>
<div>> <a href="tel:2013-01-10%C2%A001" value="+12013011001" target="_blank">2013-01-10 01</a>:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP</div>
<div>> [sofia/mpls/<a href="mailto:I888_1@172.17.17.17" target="_blank">I888_1@172.17.17.17</a>] 172.17.17.17 port 32014 -> 10.253.200.6</div>
<div>> port 16466 codec: 18 ms: 20</div>
<div>></div>
<div>> <a href="tel:2013-01-10%C2%A001" value="+12013011001" target="_blank">2013-01-10 01</a>:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP</div>
<div>> [sofia/transport/2996] 172.17.17.17 port 27276 -> 172.16.16.16 port 17056</div>
<div>> codec: 18 ms: 20</div>
<div>></div>
<div>> freeswitch@internal> g729_info</div>
<div>></div>
<div>> Permitted G729 channels: 40</div>
<div>></div>
<div>> Encoders in use: 0</div>
<div>></div>
<div>> Decoders in use: 0</div>
<div>></div>
<div>></div>
<div>></div>
<div>> FreeSWITCH Box 2:</div>
<div>></div>
<div>> <a href="tel:2013-01-10%C2%A001" value="+12013011001" target="_blank">2013-01-10 01</a>:50:14.184757 [DEBUG] sofia_glue.c:3351 AUDIO RTP</div>
<div>> [sofia/transport/<a href="mailto:3888@172.17.17.17" target="_blank">3888@172.17.17.17</a>] 172.16.16.16 port 17056 -> 172.17.17.17</div>
<div>> port 27276 codec: 18 ms: 20</div>
<div>></div>
<div>> <a href="tel:2013-01-10%C2%A001" value="+12013011001" target="_blank">2013-01-10 01</a>:50:15.664733 [DEBUG] sofia_glue.c:3351 AUDIO RTP</div>
<div>> [sofia/mpls/<a href="http://sip:C996_1@10.253.200.10:5060" target="_blank">sip:C996_1@10.253.200.10:5060</a>] 172.16.16.16 port 18966 -></div>
<div>> 10.253.200.10 port 16486 codec: 18 ms: 20</div>
<div>></div>
<div>> freeswitch@internal> g729_info</div>
<div>></div>
<div>> Permitted G729 channels: 40</div>
<div>></div>
<div>> Encoders in use: 1</div>
<div>></div>
<div>> Decoders in use: 1</div>
<div>></div>
<div>></div>
<div>></div>
<div>></div>
<div>></div>
<div>></div>
<div>></div>
<div>> Thanks in advance guys.</div>
<div>></div>
<div>> Colin</div><div>
<div>></div>
<div>></div>
<div>> _________________________________________________________________________</div>
<div>> Professional FreeSWITCH Consulting Services:</div>
<div>> <a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a></div>
<div>> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a></div>
<div>></div>
<div>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server</div>
<div>> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a></div>
<div>></div>
<div>> Official FreeSWITCH Sites</div>
<div>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a></div>
<div>> <a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a></div>
<div>> <a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a></div>
<div>></div>
<div>> FreeSWITCH-users mailing list</div>
<div>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a></div>
<div>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a></div>
<div>> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a></div>
<div>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a></div>
<div>></div>
<div></div>
<div></div>
<div></div>
</div><div>------------------------------------------------------------</div>
<div>------------------------------------------------------------</div>
<div>From: "Tamas.Cseke " <<a href="mailto:cstomi.levlist@gmail.com" target="_blank">cstomi.levlist@gmail.com</a>></div>
<div>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>></div>
<div>Subject: [Freeswitch-users] Early media without bridge</div>
<div>Date: Thu10 Jan 2013 12:24:00 +0100</div>
<div></div>
<div>Hello,</div>
<div></div>
<div>We would like to hear early media without CHANNEL_BRIDGE event</div>
<div>These are failed calls and callers would like to hear the message that </div>
<div>the provider plays</div>
<div>Because the caller thinks the call is answered if originate returns.</div>
<div></div>
<div>as far as I understand:</div>
<div> -early media makes the originate return</div>
<div> -if we ignore early media the bridge won't return, but we don't hear it</div>
<div></div>
<div>we would like both of them, is it possible somehow?</div>
<div></div>
<div>I 'm not sure I fully understand all of the ignore_early_media options</div>
<div>but I haven't find solution for this,</div>
<div>Could you please advise me one, if there is any?</div>
<div></div>
<div>I'm thinking about we maybe need a new ignore_early_media option</div>
<div>like "consume" but sending the media to the caller instead of dropping it</div>
<div>If there isn't already a solution I also would appreciate if you let me </div>
<div>know your opinion about this idea</div>
<div></div>
<div>Thanks advance,</div>
<div>Tamas</div>
<div></div>
<div></div>
<div></div>
<div>------------------------------------------------------------</div>
<div>------------------------------------------------------------</div>
<div>From: Jon_Sch鴓zinsky<<a href="mailto:jos@firstcom.dk" target="_blank">jos@firstcom.dk</a>></div>
<div>To: "<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>"<<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>></div>
<div>Subject: [Freeswitch-users] Loopback Endpoint</div>
<div>Date: Thu10 Jan 2013 14:36:06 +0100</div>
<div></div>
<div>Hello List,</div>
<div></div>
<div>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.</div>
<div></div>
<div>Is this an indication that it has become safer to use?</div>
<div></div>
<div>I have a specific problem where I essentially have to dial a dialplan for each user I am trying to reach, in parallel.</div>
<div>Is it correctly understood that loopback would be the best/only way of implementing this, or is there another way?</div>
<div></div>
<div></div>
<div>Kind Regards</div>
<div></div>
<div>Jon Schøpzinsky</div>
<div></div>
<div></div>
<div></div>
<div></div>
<div>------------------------------------------------------------</div>
<div>------------------------------------------------------------</div>
<div>From: Abaci <<a href="mailto:abaci64@gmail.com" target="_blank">abaci64@gmail.com</a>></div>
<div>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>></div>
<div>Subject: Re: [Freeswitch-users] mod_directory menu-top?</div>
<div>Date: Thu10 Jan 2013 09:11:09 -0500</div>
<div></div>
<div>use 'execute_extension' to start the directory application so that you </div>
<div>get back to the ivr when you exit the directory application.</div>
<div></div>
<div>On 1/9/2013 6:41 PM, Phillip Warner wrote:</div>
<div>> 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?</div>
<div>></div>
<div>> For example: ivr --> directory --> user changes mind about dialling by name and wants to return to ivr --> ivr</div>
<div>></div>
<div>> Thanks. Phil.</div><div>
<div>> _________________________________________________________________________</div>
<div>> Professional FreeSWITCH Consulting Services:</div>
<div>> <a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a></div>
<div>> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a></div>
<div>></div>
<div>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server</div>
<div>> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a></div>
<div>></div>
<div>> Official FreeSWITCH Sites</div>
<div>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a></div>
<div>> <a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a></div>
<div>> <a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a></div>
<div>></div>
<div>> FreeSWITCH-users mailing list</div>
<div>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a></div>
<div>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a></div>
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<div>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a></div>
<div></div>
<div></div>
<div></div>
<div></div>
</div><div>------------------------------------------------------------</div>
<div>_______________________________________________</div><div>
<div>FreeSWITCH-users mailing list</div>
<div><a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a></div>
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<div></div></div></font></div></div>
<br>_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
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<br>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
<a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
<br>
Official FreeSWITCH Sites<br>
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<br></blockquote></div><br></div></div></div>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div></div>