<div dir="ltr">Hi
<span style="font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">Patrick</span> ,<div><br></div><div>I tryed a lot effort on google but no help i got and even i sent e-mail to lua mail list as well. but no one replied on it.</div>
<div><br></div><div>Sorry but if you reply to me with details on luac then i will be a big help for me...<br><br><div class="gmail_quote">On Sat, Mar 17, 2012 at 2:10 AM, <span dir="ltr"><<a href="mailto:freeswitch-users-request@lists.freeswitch.org">freeswitch-users-request@lists.freeswitch.org</a>></span> wrote:<br>
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<br>Today's Topics:<br>
<br>
1. Re: blocking destination number (Brian Foster)<br>
2. Re: Mod_flite wont load (Jeff Lenk)<br>
3. Re: Any help me to installed Luac compiler (Patrick Lists)<br>
4. mod_shout issue (Balamurugan Mahendran)<br>
5. Strange VLAN problem with freeswitch (Erin O'Meara)<br>
6. Re: Getting real-time amplitude of a channel (Michael Collins)<br>
<br><br>---------- Forwarded message ----------<br>From: Brian Foster <<a href="mailto:bdfoster@endigotech.com">bdfoster@endigotech.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br>Date: Fri, 16 Mar 2012 12:58:27 -0400<br>Subject: Re: [Freeswitch-users] blocking destination number<br><p>The toll_allow variable.would be defined in the actual user xml file located in /usr/local/freeswitch/conf/directory. Then you would evaluate that variable in the dialplan that would end up making the bridge to your gateway. </p>
<p>-BDF</p>
<div class="gmail_quote">On Mar 16, 2012 5:21 AM, "Miha" <<a href="mailto:miha@softnet.si" target="_blank">miha@softnet.si</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi @Gabriel,<br>
<br>
I was looking on wiki how to define new xml or document where I will<br>
have my restricted numbers.<br>
Where are domestic,international,local defined?<br>
<br>
Thanks!<br>
Miha<br>
<br>
On 3/10/2012 11:23 PM, Gabriel Gunderson wrote:<br>
> On Fri, Mar 9, 2012 at 7:37 AM, Miha Zoubek<<a href="mailto:miha@softnet.si" target="_blank">miha@softnet.si</a>> wrote:<br>
>> what is the bast way to block destination number for certain user.<br>
>> Is it possible to do it in user/dir?<br>
> Ultimately, it would be a dialplan configuration, but you could set<br>
> the variable that you match in the directory. See the example of<br>
> 'toll_allow' in the default FreeSWITCH configuration...<br>
><br>
><br>
> This is set on a per-user basis in the DIRECTORY:<br>
><br>
> <variable name="toll_allow" value="domestic,international,local"/><br>
><br>
><br>
><br>
> But it's considered while evaluating the DIALPLAN:<br>
><br>
> <extension name="<a href="http://local.example.com" target="_blank">local.example.com</a>"><br>
> <condition field="${toll_allow}" expression="local"/><br>
> <condition field="destination_number" expression="^(\d{7})$"><br>
> <action application="set"<br>
> data="effective_caller_id_number=${outbound_caller_id_number}"/><br>
> <action application="set"<br>
> data="effective_caller_id_name=${outbound_caller_id_name}"/><br>
> <action application="bridge"<br>
> data="sofia/gateway/${default_gateway}/1${default_areacode}$1"/><br>
> </condition><br>
> </extension><br>
><br>
><br>
><br>
> You could easily do something similar by having a list of numbers that<br>
> they can't call listed in the directory (maybe something like<br>
> 'restricted_numbers') and check to make sure the destination doesn't<br>
> match it in the dialplan.<br>
><br>
> Good luck!<br>
><br>
><br>
> Gabe<br>
><br>
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</blockquote></div>
<br><br>---------- Forwarded message ----------<br>From: Jeff Lenk <<a href="mailto:jeff@jefflenk.com">jeff@jefflenk.com</a>><br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
Cc: <br>Date: Fri, 16 Mar 2012 10:46:10 -0700 (PDT)<br>Subject: Re: [Freeswitch-users] Mod_flite wont load<br>Please try git head and let me know if that works.<br>
<br>
--<br>
View this message in context: <a href="http://freeswitch-users.2379917.n2.nabble.com/Mod-flite-wont-load-tp7376926p7379705.html" target="_blank">http://freeswitch-users.2379917.n2.nabble.com/Mod-flite-wont-load-tp7376926p7379705.html</a><br>
Sent from the freeswitch-users mailing list archive at Nabble.com.<br>
<br>
<br>
<br><br>---------- Forwarded message ----------<br>From: Patrick Lists <<a href="mailto:freeswitch-list@puzzled.xs4all.nl">freeswitch-list@puzzled.xs4all.nl</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br>Date: Fri, 16 Mar 2012 18:54:19 +0100<br>Subject: Re: [Freeswitch-users] Any help me to installed Luac compiler<br>On 16-03-12 06:07, virendra bhati wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi list,<br>
<br>
I don't find any link on which luac compiler is mention. only one link<br>
was found on which description about Luac was mention.<br>
<a href="http://www.lua.org/manual/4.0/luac.html" target="_blank">http://www.lua.org/manual/4.0/<u></u>luac.html</a><br>
<br>
Anyone know how to installed Luac compiler in Centos ?<br>
</blockquote>
<br>
Can't you try a little harder to figure out these basic questions yourself? Did you even try to Google for "luac CentOS"? That would have given you a hint.<br>
<br>
Since your question is not a FreeSWITCH question but it is a lua question why not ask on the lua mailinglist or on the lua irc channel if they have one.<br>
<br>
Regards,<br>
Patrick<br>
<br>
<br>
<br><br>---------- Forwarded message ----------<br>From: Balamurugan Mahendran <<a href="mailto:b2m@a-cti.com">b2m@a-cti.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br>Date: Fri, 16 Mar 2012 23:34:23 +0530<br>Subject: [Freeswitch-users] mod_shout issue<br>Hi all,<div><br></div><div><div>2012-03-16 18:02:32.957534 [ERR] mod_shout.c:859 Error: MPG123 Error at /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:659.</div>
<div>2012-03-16 18:02:32.957534 [ERR] mod_shout.c:862 Error from mpg123: Invalid mpg123 handle. (code 10)</div>
</div><div><br></div><div>Please let me know what Iam missing here.</div><div><br></div><div>Thanks,</div><div>Bala</div>
<br><br>---------- Forwarded message ----------<br>From: "Erin O'Meara" <<a href="mailto:erin.omeara@salmonbaytechnology.com">erin.omeara@salmonbaytechnology.com</a>><br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
Cc: <br>Date: Fri, 16 Mar 2012 09:58:51 -0700<br>Subject: [Freeswitch-users] Strange VLAN problem with freeswitch<br>I have moved offices down the street and extended my office network via vlan (old office was using VLAN also). I have two networks one internal and one wireless that I sell wireless service with.<div>
<br></div><div>FusionPBX (In the cloud) -> Cable Modem -> PFSense -> VLAN -> Phone.</div>
<div><br></div><div>my phone works fine for a few days on either network then calls stop working, if I switch the phone VLAN to the other VLAN it starts working again. I noticed this issue from my office that is connect of a Ubiquiti wireless link and VLAN extended to my new office.</div>
<div><br></div><div>What happens is the calls ring and I can pickup but no audio in either direction.</div><div><br></div><div>here is the pastebin of a call failing then I plugged the same phone into the other vlan and reboot and made a successful call.</div>
<div><br></div><div><a href="http://pastebin.freeswitch.org/18675" target="_blank">http://pastebin.freeswitch.org/18675</a> <br></div><div><br></div><div><span style="font-family:Arial,Verdana,sans-serif"><div style="text-align:-webkit-auto;display:block">
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<br><br>---------- Forwarded message ----------<br>From: Michael Collins <<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br>Date: Fri, 16 Mar 2012 13:39:55 -0700<br>Subject: Re: [Freeswitch-users] Getting real-time amplitude of a channel<br>Post the bounty on the wiki and then shoot an email to the -dev list letting them know there's a new bounty available...<br>
<br>-MC<br><br><div class="gmail_quote">On Fri, Mar 16, 2012 at 9:59 AM, Matt Stockton <span dir="ltr"><<a href="mailto:mstockton@harqen.com" target="_blank">mstockton@harqen.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi Michael,<div><br></div><div>Thanks for the info!! Is there information I can find on how the bounties work? I'm interesting in getting this feature, but it would probably take me much longer to implement than someone who knows the code well. I see a bounty wiki entry...is that active? or is there a better place to post such things?</div>
<div><br></div><div>Thanks,</div><div>Matt<div><div><br><br><div class="gmail_quote">On Wed, Mar 14, 2012 at 12:30 PM, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">AFAIK there isn't anything pre-rolled that you can use. However, I know that Tony's auto gain control (agc) logic checks the "volume" level coming in on a channel. You might be able to look in mod_conference.c and see where the agc calculates the incoming volume level and write an API that displays that value.<br>
<br>I'd start by looking at the function check_agc_levels() and for references to member->agc_volume_in_level. Hopefully that will give you a place to start.<br><br>-MC<br><br><div class="gmail_quote"><div><div>
On Wed, Mar 14, 2012 at 10:26 AM, Matt Stockton <span dir="ltr"><<a href="mailto:mstockton@harqen.com" target="_blank">mstockton@harqen.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>Hi all,<div><br></div><div>I am interested in providing a visualization of a channel in a conference. I am able to process the start talking and stop talking events from the conference, but I'm wondering, has anyone done anything that can get the real-time amplitude of a channel? I didn't see any built-in API commands to do this, but wondering if there's any modules or something else that I'm overlooking? Or maybe now is the time for me to dig into the API documentation?</div>
<div><br></div><div>Just curious if anyone has done something like this before, and if so, if you could guide me in the right direction.</div><div><br></div><div>Thanks!</div><span><font color="#888888"><div>
Matt</div>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br><div dir="ltr"><br>Thanks and regards<br><br> Virendra Bhati<br>+91-8885268942<br>Software Engineer<br>E-mail-: <a href="mailto:virbhati@gmail.com" target="_blank">virbhati@gmail.com</a><br>
Skype id:- virbhati2<br>Hyderabad(India)<br></div><br>
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