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<DIV><FONT color=#c0c0c0 size=2 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 size=2 face=Verdana><SPAN><FONT
color=#000000></FONT> </DIV></SPAN></FONT><FONT color=#000080 size=2
face=Verdana>
<HR>
</FONT>
<DIV><FONT size=2 face=Verdana><STRONG>发件人：</STRONG> freeswitch-users-request
</FONT></DIV>
<DIV><FONT size=2 face=Verdana><STRONG>发送时间：</STRONG> 2013-01-10 22:12:37
</FONT></DIV>
<DIV><FONT size=2 face=Verdana><STRONG>收件人：</STRONG> freeswitch-users
</FONT></DIV>
<DIV><FONT size=2 face=Verdana><STRONG>抄送：</STRONG> </FONT></DIV>
<DIV><FONT size=2 face=Verdana><STRONG>主题：</STRONG> FreeSWITCH-users Digest, Vol
79, Issue 54 </FONT></DIV>
<DIV><FONT size=2 face=Verdana></FONT> </DIV>
<DIV><FONT size=2 face=Verdana>
<DIV>Send FreeSWITCH-users mailing list submissions to</DIV>
<DIV>freeswitch-users@lists.freeswitch.org</DIV>
<DIV></DIV>
<DIV>To subscribe or unsubscribe via the World Wide Web, visit</DIV>
<DIV>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</DIV>
<DIV>or, via email, send a message with subject or body 'help' to</DIV>
<DIV>freeswitch-users-request@lists.freeswitch.org</DIV>
<DIV></DIV>
<DIV>You can reach the person managing the list at</DIV>
<DIV>freeswitch-users-owner@lists.freeswitch.org</DIV>
<DIV></DIV>
<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 <avi@avimarcus.net></DIV>
<DIV>To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org></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>It looks like you can do it via api command:</DIV>
<DIV>http://wiki.freeswitch.org/wiki/Mod_commands#uuid_fileman</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 <wangd@alongtechnology.com.cn> wrote:</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>> consulting@freeswitch.org</DIV>
<DIV>> http://www.freeswitchsolutions.com</DIV>
<DIV>></DIV>
<DIV>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server</DIV>
<DIV>> http://www.cudatel.com</DIV>
<DIV>></DIV>
<DIV>> Official FreeSWITCH Sites</DIV>
<DIV>> http://www.freeswitch.org</DIV>
<DIV>> http://wiki.freeswitch.org</DIV>
<DIV>> http://www.cluecon.com</DIV>
<DIV>></DIV>
<DIV>> FreeSWITCH-users mailing list</DIV>
<DIV>> FreeSWITCH-users@lists.freeswitch.org</DIV>
<DIV>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</DIV>
<DIV>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users</DIV>
<DIV>> http://www.freeswitch.org</DIV>
<DIV>></DIV>
<DIV>></DIV>
<DIV></DIV>
<DIV>------------------------------------------------------------</DIV>
<DIV>------------------------------------------------------------</DIV>
<DIV>From: Steven Ayre <steveayre@gmail.com></DIV>
<DIV>To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org></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 <sip:user@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</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 <lconroy@insensate.co.uk> 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 <sip:user@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).</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 <lconroy@insensate.co.uk>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 fred@example.com, and strictly the sender will</DIV>
<DIV>>>> do a check for a MX and then look for an A record for example.com, 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>>>>> fs-list@communicatefreely.net> 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 "pbx.example.com" instead of just "</DIV>
<DIV>>>>> example.com", 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>>>>> http://wiki.freeswitch.org/wiki/Polycom_configuration 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>>>>> Professional FreeSWITCH Consulting Services:</DIV>
<DIV>>>>> consulting@freeswitch.org</DIV>
<DIV>>>>> http://www.freeswitchsolutions.com</DIV>
<DIV>>>>></DIV>
<DIV>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server</DIV>
<DIV>>>>> http://www.cudatel.com</DIV>
<DIV>>>>></DIV>
<DIV>>>>> Official FreeSWITCH Sites</DIV>
<DIV>>>>> http://www.freeswitch.org</DIV>
<DIV>>>>> http://wiki.freeswitch.org</DIV>
<DIV>>>>> http://www.cluecon.com</DIV>
<DIV>>>>></DIV>
<DIV>>>>> FreeSWITCH-users mailing list</DIV>
<DIV>>>>> FreeSWITCH-users@lists.freeswitch.org</DIV>
<DIV>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</DIV>
<DIV>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users</DIV>
<DIV>>>>> http://www.freeswitch.org</DIV>
<DIV>>>></DIV>
<DIV>>>></DIV>
<DIV>>>> _________________________________________________________________________</DIV>
<DIV>>>> Professional FreeSWITCH Consulting Services:</DIV>
<DIV>>>> consulting@freeswitch.org</DIV>
<DIV>>>> http://www.freeswitchsolutions.com</DIV>
<DIV>>>></DIV>
<DIV>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server</DIV>
<DIV>>>> http://www.cudatel.com</DIV>
<DIV>>>></DIV>
<DIV>>>> Official FreeSWITCH Sites</DIV>
<DIV>>>> http://www.freeswitch.org</DIV>
<DIV>>>> http://wiki.freeswitch.org</DIV>
<DIV>>>> http://www.cluecon.com</DIV>
<DIV>>>></DIV>
<DIV>>>> FreeSWITCH-users mailing list</DIV>
<DIV>>>> FreeSWITCH-users@lists.freeswitch.org</DIV>
<DIV>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</DIV>
<DIV>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users</DIV>
<DIV>>>> http://www.freeswitch.org</DIV>
<DIV>>>></DIV>
<DIV>>></DIV>
<DIV>>></DIV>
<DIV>>></DIV>
<DIV>>> --</DIV>
<DIV>>> Michael S Collins</DIV>
<DIV>>> Twitter: @mercutioviz</DIV>
<DIV>>> http://www.FreeSWITCH.org</DIV>
<DIV>>> http://www.ClueCon.com</DIV>
<DIV>>> http://www.OSTAG.org</DIV>
<DIV>>> _________________________________________________________________________</DIV>
<DIV>>> Professional FreeSWITCH Consulting Services:</DIV>
<DIV>>> consulting@freeswitch.org</DIV>
<DIV>>> http://www.freeswitchsolutions.com</DIV>
<DIV>>></DIV>
<DIV>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server</DIV>
<DIV>>> http://www.cudatel.com</DIV>
<DIV>>></DIV>
<DIV>>> Official FreeSWITCH Sites</DIV>
<DIV>>> http://www.freeswitch.org</DIV>
<DIV>>> http://wiki.freeswitch.org</DIV>
<DIV>>> http://www.cluecon.com</DIV>
<DIV>>></DIV>
<DIV>>> FreeSWITCH-users mailing list</DIV>
<DIV>>> FreeSWITCH-users@lists.freeswitch.org</DIV>
<DIV>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</DIV>
<DIV>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users</DIV>
<DIV>>> http://www.freeswitch.org</DIV>
<DIV>></DIV>
<DIV>></DIV>
<DIV>> _________________________________________________________________________</DIV>
<DIV>> Professional FreeSWITCH Consulting Services:</DIV>
<DIV>> consulting@freeswitch.org</DIV>
<DIV>> http://www.freeswitchsolutions.com</DIV>
<DIV>></DIV>
<DIV>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server</DIV>
<DIV>> http://www.cudatel.com</DIV>
<DIV>></DIV>
<DIV>> Official FreeSWITCH Sites</DIV>
<DIV>> http://www.freeswitch.org</DIV>
<DIV>> http://wiki.freeswitch.org</DIV>
<DIV>> http://www.cluecon.com</DIV>
<DIV>></DIV>
<DIV>> FreeSWITCH-users mailing list</DIV>
<DIV>> FreeSWITCH-users@lists.freeswitch.org</DIV>
<DIV>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</DIV>
<DIV>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users</DIV>
<DIV>> http://www.freeswitch.org</DIV>
<DIV></DIV>
<DIV></DIV>
<DIV></DIV>
<DIV>------------------------------------------------------------</DIV>
<DIV>------------------------------------------------------------</DIV>
<DIV>From: Steven Ayre <steveayre@gmail.com></DIV>
<DIV>To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org></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 <cmason@frontiernetworks.ca> 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>> 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP</DIV>
<DIV>> [sofia/mpls/I888_1@172.17.17.17] 172.17.17.17 port 32014 -> 10.253.200.6</DIV>
<DIV>> port 16466 codec: 18 ms: 20</DIV>
<DIV>></DIV>
<DIV>> 2013-01-10 01:50:14.195277 [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>> 2013-01-10 01:50:14.184757 [DEBUG] sofia_glue.c:3351 AUDIO RTP</DIV>
<DIV>> [sofia/transport/3888@172.17.17.17] 172.16.16.16 port 17056 -> 172.17.17.17</DIV>
<DIV>> port 27276 codec: 18 ms: 20</DIV>
<DIV>></DIV>
<DIV>> 2013-01-10 01:50:15.664733 [DEBUG] sofia_glue.c:3351 AUDIO RTP</DIV>
<DIV>> [sofia/mpls/sip:C996_1@10.253.200.10:5060] 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>> Professional FreeSWITCH Consulting Services:</DIV>
<DIV>> consulting@freeswitch.org</DIV>
<DIV>> http://www.freeswitchsolutions.com</DIV>
<DIV>></DIV>
<DIV>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server</DIV>
<DIV>> http://www.cudatel.com</DIV>
<DIV>></DIV>
<DIV>> Official FreeSWITCH Sites</DIV>
<DIV>> http://www.freeswitch.org</DIV>
<DIV>> http://wiki.freeswitch.org</DIV>
<DIV>> http://www.cluecon.com</DIV>
<DIV>></DIV>
<DIV>> FreeSWITCH-users mailing list</DIV>
<DIV>> FreeSWITCH-users@lists.freeswitch.org</DIV>
<DIV>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</DIV>
<DIV>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users</DIV>
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<DIV></DIV>
<DIV></DIV>
<DIV></DIV>
<DIV>------------------------------------------------------------</DIV>
<DIV>------------------------------------------------------------</DIV>
<DIV>From: "Tamas.Cseke " <cstomi.levlist@gmail.com></DIV>
<DIV>To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org></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<jos@firstcom.dk></DIV>
<DIV>To: "freeswitch-users@lists.freeswitch.org"<freeswitch-users@lists.freeswitch.org></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 <abaci64@gmail.com></DIV>
<DIV>To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org></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>
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