similar to: for review

There is plenty of information on the wiki for setting asterisk up for transferring calls both from the Dail() command, and features.conf. What really seems to be missing, is simply how do you actually perform the transfer? Blind transfers are pretty simple as you only have two obvious steps. How though do you do attended transfers? 1.) You have a call 2.) You dial *2 or whatever you have

For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up from there. What would be really great is if I could control how many calls by the context. So if a call was routed via [overload] Then the ext wouldn't report busy it would just keep ringing available

Hey, I would like in the course of dial plan logic, to trigger a separate outbound call. If that outbound call is answered, and if that certain key response is detected then it will bridge the incoming call to the newly dialed outbound call. What I want to accomplish is that when a caller dials in, they can enter enter an extension that will call out to a callee's cell phone. When the

I'm really confused on how to use these two options together: A while back: JustRumours edited this page: http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe and added a little section about dynamic conferences. the 'e' option is repeated all over the page as the savior of dynamic conferences, maybe I'm just dumb, but can someone tell me if a conference is created with the e

You don't need to reinvent anything to tie radios to *. Ham systems like IRLP, Echolink, eqso etc all have fairly tight controls to keep from being abused (although with a little Linux knowledge, the IRLP package can easily be used to set up your own network using their protocol). Jim Dixon seems to have done the work to integrate radios with * already. See

Hi I can't for the life of me work out why this is not working. When in the campon contect if you hit a DTMF key 2 you get moved to the exten => 2 defined in the mainmenu context not the exten => 2 defined in the campon context. What is wrong? The same happens if you hit key 1. [campon] exten => _*1XXX,1,Answer exten => _*1XXX,2,SetCallerID(${CALLERIDNUM}) exten =>

Hi all, Having fixed my problems with the call waiting ringing on the GS phones, I needed to extend that with a campon facility (available on some legacy systems - sort of callwaiting without phone ringing). I've managed to implement that by adding/modifying app_queue.c. Basically, when calling the SIP phone, and if busy, I can camp the call onto that extension, with MOH, allowing the caller

Miles Scruggs wrote: >>Miles Scruggs wrote: >> >>>How do I shorten my servername that is displayed to windows clients >>>currently it is >>>'Samba 3.0.2a-Debian (netbios name)' How do I change this to netbios >> >>name >> >>>only? >> >>Look for the "Server string" line in your smb.conf - this is where

Hello All, I have 1M rows of time stamped information about delivery trucks and their trip related information from simulation .Detailed info the column names and attributes for clarity 1. id: alphanumeric factor/character 2. datetime.of.trip.start: POSIXct yyyy-mm-dd hh:mm:ss 3. datetime.of.trip.end: POSIXct yyyy-mm-dd hh:mm:ss 4. trip.distance: numeric, miles 5.

Just starting to enjoy the full features of asterisk, I do have a couple questions though, that I can't seem to find answers for in the wiki, just wondering if someone could light my way. after a caller has made their choice of options in the dial plan, I would like them to be placed on "hold" (music, not ringing) while the system processes through the rest of the dial plan

I'm having a real problem with one of my linksys pap2. On outgoing calls the callee will ring, but caller (pap2) will not here it ring When the callee answers, no audio is transmitted either way. Asterisk reports the call connected and bridged correctly. Now the kicker is that sometimes it works and other times it doesn't. I have had the most luck calling land lines, but sometime

Campon, mini-queues, see asterisk tips and tricks on voipinfo... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Zach A Sent: Monday, February 06, 2006 1:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Help on queues I need practical examples showing

So in order to answer the background and backbone questions here is the system as it is. I hope this isn't too much for the list but I'll post it in response to a few inquiries. The current system is quite interesting. We have an office in a town that is about 50 miles from the ski area. The ski area is powered 100% of of generators and the telephone access and internet access goes from

<headline>Short question</headline>: Do people have advice on debugging R programs running after CGIwithR inputting of data from forms? Is there a way of setting up fast local versions if your local machine has to be a windoze (2k) machine (R 2.1.0) and your server is a Debian, ssh shell only set up running R 1.8.0? Are there simple guides to ways of not having to invoke R each

I have a few weird problems with profiles on my samba PDC. Right now I'm just testing with two XP pro clients. Samba is Samba version 3.0.2a-Debian The problems that I'm having and I believe are related are: 1.) Profiles are saved to the server, but don't migrate to different clients. This is very odd, I can make all sorts of changes to the profile and I can see those changes

Dave, I have been interested in some of the same questions that you have been asking. A while back I asked a person at VLSI ( http://www.vlsi.fi/ ) in Finland if they had a product that might work. The exchange of email is included below. In summary, the answer was probably no, but maybe in the future. The VS1005 chip he mentions does not seem to have appeared yet. I currently have a pair

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 As far as a backup provider goes I wouldn't expect them to use rsync over SSL unless that were built into rsync in the future (and has been around long enough that most users would have it). I would expect them to either use rsync over ssh secured by rrsync or rsyncd over ssh with them managing the rsyncd.conf file. Either way the server side

As others have mentioned, Chapter 4 of David A. Black''s book _Ruby for Rails_ is now available for download from the Manning site. The fact that the book is being released one chapter at a time makes it easy to read in small, digestible chunks. I never intended to review every chapter individually here, but now that I''ve done the first three, I feel like I ought to continue.

Don't know how that last email took off, but that wasn't suppose to happen. Any the root 3 59.6 0.0 0 0 ? R< 10:16 11:04 [events/0] is a kernel thread that handles weird irq stuff. I just disabled onboard sound and USB and it went away. I'm still left with my other problems though, here they are 2.) I'm having LMB and DMB problems on the server,

Dear Sir/Madam: I simply want to draw x-y plot with Julian dates (x) and numbers (y). Please see below for my program. In the older version of R, the plot(jdat, miles) worked without any problem. But, with the new version of R, plot(jdat, miles) does not work any more. So, I added log="" option (as far as I know, it is a default setting, so log="" should not be needed).