similar to: Does anyone manage the wiki?

icecast@xiph.org wrote: > Hello, this is the mailing list anti-spam filter at Xiph.Org. > We need you to confirm your e-mail message with the subject of > "(no subject)". > > Please send a message to the following address, or simply use your > mailer's "Reply" feature. > > icecast+confirm+1053014484.2633.87372b@xiph.org > > Rather than

Hi, I need to create a simple number checking for authorizing the calls. if a person dial 91800555121212345 where 12345 is the authorization code. If the authorization code is correct the call will go through if not it will play something saying wrong authorization code or just hangup. This my dialplan to get the authorization code AUTH=12345 exten => _9.,1,Answer() exten =>

Hi everybody. I''d like to announce that 3.2.11, 3.1.10, 3.0.19, and 2.3.15 have been released. These releases contain two **extremely critical security fixes** so please update **IMMEDIATELY**. You can read about the security fixes by following these links: * [CVE-2013-0155](https://groups.google.com/group/rubyonrails-security/browse_thread/thread/b75585bae4326af2) *

Hi everybody. I''d like to announce that 3.2.11, 3.1.10, 3.0.19, and 2.3.15 have been released. These releases contain two **extremely critical security fixes** so please update **IMMEDIATELY**. You can read about the security fixes by following these links: * [CVE-2013-0155](https://groups.google.com/group/rubyonrails-security/browse_thread/thread/b75585bae4326af2) *

On 20 January 2018 at 23:30, Tim S <tim.strommen at gmail.com> wrote: > I have seen this take over 2 seconds before on a sluggish machine. Thanks - my host uses SSD and everything seems pretty quick, but I'll give it a 1 second pause. > you'd need to pipe that to a Google Speech API tunnel. > That's probably not something you can hack away at with simple > Asterisk

Admittedly this is probably pretty stupid of me, but there are just some things I can't understand by reading documentation. Any suggestions or recommendations about how to handle my problem are greatly appreciated. I'm trying to achieve the same functionality as my Nortel PBX, without rewriting much 'C' code. In my dialplan I'd like to compare two variables as a means of

I''m trying to use a partial from within Markaby. I haven''t been able to figure out how to access a parameter passed into the partial. Here''s the code I''m using: h1 "Create a new note" if @note render :partial => ''form/errors'', :record => @note end ... (that snippet, as well as the partial is stolen shamelessly from Restolog

Hello all, I am looking for a way to restrict users from logging in two separate phones with the same authorization name/password at the same time. Meaning, I only want users to be able to place a call from one phone in one location, but have the ability to move from computer to computer. Has anyone found any sort of solution for this type scenario? Thanks, Bryan Mahin Please visit us @

Hi, We have received the subscription notification. Following is the information required to contribute to the wiki page: Username: PacktPublishing Subject: Book feature on the wiki page Location: https://wiki.centos.org/Books Thanks & Regards Sherwin Sherwin Silveira Key Partner Executive -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a Model whose table name and primary key don''t follow Rails conventions. So I have used set_table_name() and set_primary_key() and can do the Record.find() method ok. My problem is that the Record.new(params[:id]) idiom in a controller complains: undefined method `stringify_keys!'' for "REAL_PKEY":String Is there something else I need to do?

Hi all,   can I freely create an account for the Wiki and do changes? Or should I notify someone (Rowland)? I found a small error in https://wiki.samba.org/index.php/User_Home_Folders . The wiki mentions as rights on the share Read and Execute for Authenticated Users, but on that level I see only Full Control, Change and Read. My assumption is that it should be Change and Read for Authenticated

Hi there, I know this question is kind of stupid. But, I don't know anywhere else to ask. I've received some answers when I asked about the need of having a zaptel interface to make the meetme application work, that said that it was better to have a real hardware then the zaptelrtc software modules. So, my question is, would any of the following cards work with the meetme

I am lost as to what I am missing. I am setting up dovecot 2.0.9 on a RHEL 6.4 machine as provided by my provider. I have IMAP up and running, and I have POP3 up and running. Testing confirms this. Also, if it makes a difference, I enabled dovecot as my LDA. Sendmail was setup as well due to our 1.x version using it and I though I had to. Anyway, everything is working perfectly with the

Folks -- First, apologies for not lurking for weeks or months to get the culture of the list. I read the recent post about improvement to the quality of posts with some amusement and full agreement. The problem is a big and very real one. I hope I'm not deepening it. But my question isn't explicitly asked with this subject line or definitively answered in the archives -- that I have

Hi?all friends here. I am Nathanael Lee from China.All of you can find my ID Nathanael in our community.It's very nice for me to meet all of you here. I am an English interpretor freely worked in China.I am so lucky that in this year,I got Microsoft Community Influencer Certificate. In these 2 years,I keep my mind to focus on centos and make so many progress both in its daily use and server

I have call center PCs that switch between an IBEAM SIP softphone and a NEBU IAX softphone (for reasons that aren't germane here). The SIP softphones work fine, but the IAX softphones get a fast busy unless I give them an IAX trunk to use, instead of the PRI trunks that all the other phones are using. I am using Asterisk 1.2.3. svn rev 47264. I've appended a sample call trace. The

I'm sorry for the way OT post, but here goes. I'm an informatics specialist, and R user. My wife is a secondary school maths teacher. My wife recently tried to explain to her class the link between education and potential salary, and I would love to be able to show this graphically, however, I cannot find any freely available data for this. Does anyone know of a suitable dataset,

Andrew D Kirch Indianapolis, United States Good day, I am Mr. Andrew D Kirch, a native of Indianpolis, United States and I am an Asterisk Hacker with the Summit Open Source Development Group. First and foremost,I apologized using this medium to reach you for a transaction/business of this magnitude, but this is due to Confidentiality and prompt access reposed on this medium. Be informed

I need help translating extensions.conf to AEL: [default] exten => _X.,1,Set(DID=${EXTEN:6}) exten => _X.,n,Goto(continue,1) exten => _1X.,1,Set(DID=${EXTEN:7}) exten => _1X.,n,Goto(continue,1) exten => continue,1,Noop(${DID}) exten => continue,n,Set(GROUP(IAX)=incoming) exten => continue,n,GotoIf($[${MATH(${GROUP_COUNT(incoming at IAX)}+${GROUP_COUNT(outgoing at

I have an asterisk server at home. I'd like to test one just installed elsewhere. Both servers are behind firewalls. I can see the session start in CLI, my congratulations is apparently playing and RTP is being sent. Hearing no audio. Can send key presses and see audio playing changed. "Peer audio RTP is at port 198.145.28.177:10180", but that never shows at the client side, behind