similar to: d & e options in meetme()

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

<font size="2">I''m working on a large Rails site and we want to move from RSpec 0.8.2 to the latest and greatest.&nbsp; So we ran the translator and yet we''re having a lot of trouble translating should_render.<br><br>I found this on the web:<br><br>We will NOT be supporting the following in the new syntax:<br>&nbsp;

Does anyone know anything about a 4GB size limit on Samba 3.0.4 running on AIX 5.2 with a 32-bit kernel? We currently have files being transferred from a Windows 2000 server to an AIX machine, and if the files are larger than 4GB, they are getting mangled. Running samba at a high debug level shows the file pointer rewinding or becoming negative once it reaches 4GB and md5sum indicates that the

"In general you can match callerID with the /, but if you don't put anything after the /, then the rule matches "no caller*ID", and if no slash is there at all, it matches "any callerid". " Ok.My question is -> how to match callerid from 001... ? and if don't know how many numbers ? exten => s/0_,Answer don't work- anything else ? tnx Thomas

Hi, there. While waiting for the encoder RC1, I will report some minor problems I experienced with Peter's Winamp decoder. First, I am using Nullsoft Vorbis Decoder v.1.11a with Winamp v.2.76 on Windows 98 SE. 1. When the buttons of Playlist Editor are extended, the part extended in playlist screen seems to be flickering when an Ogg file is being played. 2. I check the option of

Warning: Sligthly off topic. http://shelleytherepublican.com/2006/04/linux-european-threat-to-our-computers.html Quotes: > And guess what software Osama Bin Laden uses on his laptop? > > If you guessed it was Linux you would be 100% right. > Next time somebody asks you how Al Queda agents pay for their > rifles and rocket launchers, you can tell them that foreign hackers >

Le 04/02/2015 18:48, m.roth at 5-cent.us a ?crit : > That directory, and that file, exist in CentOS, also, since 6. And the new > naming... it's*so* much easier to deal with... yeah, right, I'll run the > install, and wait till it hangs, so I can see that the NIC is named, what > was it, on that HP last month, oh, yeah, I need to use ip -f inet link, > far simpler than

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

Newbie here. Need help with a thread that has about 25 posts. No replies in a few days. The thread is in software support for centos 6 and the subject of the thread is Centos 6.4 won't reboot on install. Have had limited success in getting a login screen but once I logout, I can't get back in again. It was mentioned in the thread that the problem is likely in Network Configuration. I

While humorous, let's dial back the trolling at this point. =] This discussion is a largely serious discussion, and shouldn't get derailed. On Wed, Jun 6, 2012 at 12:43 PM, Daniel Berlin <dberlin at dberlin.org> wrote: > > > > Not wanting to clean up the white space is exactly a simple but good > > example of technical debt that we are incurring. > > I

On Sun, Sep 22, 2019 at 09:34:59PM +0200, Nicolas Kovacs wrote: > Le 20/09/2019 ? 14:49, Alexander Dalloz a ?crit : > > Please, forget about so call howtos or tutorials on the net. They > > are either dated, giving wrong ideas (no, you don't need a MySQL > > server to run Postfix), failing to explain why the author has done > > things the way he writes them up, or

It might be a good idea to have a list of standard genres that are suggested for use in the Genre field, just so that (polished) encoders can give the user a list to pick from when encoding a song, instead of making the user come up with one their own. This might seem like an encoder issue at first, but if two different encoders use two different names for a specific genre it's going to

Hi, If there are 5 asterisk servers on the local net and each server runs meetme, eg. 3311,3321,3331,3341,3351 respectively. Can I connect these 5 meetme conferences to one meetme using IAX2? Regards, Zen

Hi all, I'm banging my head against the docs building problem and now I'm getting this: http://docbook.sourceforge.net/release/xsl/current/html/inline.xsl:1: parser error : Start tag expected, '<' not found HTTP/1.0 504 Gateway Time-out ^ compilation error: file http://docbook.sourceforge.net/release/xsl/ current/html/docbook.xsl line 50

Hi all, i have a setup with chatrooms, several MeetMe conferences wich users can change inbetween. 10 users maximum in each room. It seems like when i have more than 40-45 users on the system at the same time asterisk drops abt 20 and continnues buisness as usual. Is there anyone else who have run inte this problem? Any solutions? It would me neat to hear about peoples experiences with MeetMe,

Hi All, I'm still having trouble trying to figure out if it is possible to define (in the dial plan) a context for meetme? I'm using 1.4.4 with dialplan logic of: exten => 123,1,Meetme(,Msa,) This defaults to conferences defined within the rooms context of meetme.conf Is it possible to specify another context as with voicemail? Or can any one think of another

How many simultaneous conference meetme setups can be supported in the same time on Asterisk, and what are the corresponding server's specs for this. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100414/ba26a927/attachment.htm

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