similar to: Quick Newbie Question - Auto Call Routing

I have about 5 incoming USA SIP lines, but my provider does not have any sort of roll-over or huntgroup feature. Does anybody have an idea on how I can create a general number that will ring to the next available, non-busy SIP line that I have? Is there a provider out there that would do this? Any suggestions would be greatly welcome. Thank you. -------------- next part --------------

Hello, I'm wondering if it is possible to ring X number of extensions simultaneously, and each answered call can be handled with some code. I can do a huntgroup-esque way of dialling, but I want all the dialled numbers to be picked up. I hope this makes sense.. If not please say.. Many thanks! Andy -------------- next part -------------- An HTML attachment was scrubbed... URL:

i have a client that wants a function that will automatically logout an agent from a queue if they do not answer a call. This would prevent future calls from being sent to that phone if the agent forgot to logout. Any ideas? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com>

Stefan Gofferje wrote: > Hi Folks, > > on my home asterisk, I have a "huntgroup" for incoming calls on the > private line which first let ring my phones in my office and living > room, after a while then office, living room and bedroom. > I do this by simply putting two dial statements in sequence: > > > [private_huntgroup_day] > exten =>

Dear All I want to integrate sugarcrm and asterisk , so when customer call the call center the agent or extension which answers the call , before pickup the phone and talk to customer , view his/her information if it is available. I do this as described below : 1-Setup login username for sugarcrm for each extension 2-Extension Users will login to the sugarcrm. 3-Develop php script to handle new

On a TDM2400 with 3 FXO modules, is there a way to split each line into basically being its own trunk or another way to pull off the following scenerio: PBX has 12 inbound PSTN lines 1,3,5,7 are the 714 phone number hunt group 2,4,6,8 are the 888 phone number hunt group 9-12 are fax lines Customer wants outbound calls to go out in the following order: 8,7,6,5,4,3,2,1,12,10,11,9 Kerry

Thanks Tomas, I confirm the R Under development (unstable) (2018-04-26 r74651) version works for Rscript when the file name has a space, and no arguments are specified. C:\>"C:\Program Files\R\R-devel\bin\x64\Rscript.exe" "C:\foo bar.R" R Under development (unstable) (2018-04-26 r74651) Platform: x86_64-w64-mingw32/x64 (64-bit) Running under: Windows 7 x64 (build 7601)

Hi Tomas, Thanks for the info about the binary builds; I did install it, however the bug still seems to be there in the current build. The workaround you suggested does work: C:\>"C:\Program Files\R\R-devel\bin\x64\Rscript.exe" "C:\foo bar.R" Fatal error: cannot open file 'C:\foo': No such file or directory C:\>"C:\Program

Howdy. I just posted an unofficial HOWTO on configuring icecast2 on a Linux or other UNIX machine. It's just a draft in DocBook format, but I was hoping it might help someone. I've worked enough with icecast over the past few years that I thought it time to share what I've learned. The URL is as follows: http://quasi.ksl.com/icecast/ I have older documentation in

Absolutely!! That was one thing that I was missing and would have like to have included. You can either send me the data or let us know the link. If you wouldn't mind adding to this HOWTO and I will post a link to your page so that people would know where I got the original files. Sounds good. KJ <p>On Wed, 2003-02-12 at 10:37, adam wrote: > looks great! :-) > > if you like i

According to this list I was told that it was available to download? The only thing on the download page is the info on icecast 2 from cvs? Perhaps I'm missing something?? -- Ted Lynn <ted@linuxmds.com> --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to

Has anyone got any information on bulk provisioning of Linksys SPA-941/94s? There is an overview in the admin guide but it refers to a different provisioning guide that I haven't found anywhere. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com

I am ready to pull my hair out. I cannot seem to get the Polycoms to read the time properly. Regardless of the server they are pointed to our the offset, i am getting the correct time, but 24 hours ahead. So for today it is showing Friday April 28 but with the correct time. Any clues? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider

Fixed in R-devel. I will port to R-patched after more testing. Tomas On 04/26/2018 01:52 AM, Tomas Kalibera wrote: > Thanks for the report. A quick workaround before this gets fixed is to > add an extra first argument that has no space in it, e.g. > > Rscript --vanilla "foo bar.R" > > The problem exists on all systems, not just Windows. > > Best > Tomas >

Thanks Tomas. I confirm the quick workaround works for me in the DOS prompt, and when having a shortcut to RScript in SendTo, and when used in the Task Scheduler. I have not tested the R-devel version, due to my unfamiliarity with installing from source code. -----Original Message----- From: Tomas Kalibera [mailto:tomas.kalibera at gmail.com] Sent: Thursday, April 26, 2018 6:34 AM To: Kerry

Is there any setting in the voicemail that will send the voicemail file in a type that is recognized on a Blackberry? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com <http://www.techdatapros.com/> http://www.techdatapros.com --------------

How will this fare? I am planning on putting an asterisk box for my brother in the Philippines but they only have dialup internet. I want them to be able to use a telephone set on a phonejack or linejack card and call me and vice versa via VOIP. My setup in the US is working already with a broadband cable connection. I am thinking that dialup may not work because of the bandwidth required

I have been able to successfully stream ogg vorbis as both a playlist and as a live stream off my icecast2 server here using port 8000. Works great. I can also stream MP3 live audio using darkice and icecast2 off the same box using port 8001. But I have been unable to create a situation where I can stream *both" ogg and mp3 off the same machine. I have even created a new icecast directory to

Hey Rado, Ack... Yep, I'll give that a shot. Can't just shut down the streams at any given time. This may have to wait for testing. The friend I am helping set this up has decided the quality of Ogg Vorbis FAR outweighs the benefits and performance of MP3. Plus, there are the licensing issues involved with MP3. Any ideas on how to do set up a dual stream, one for low bandwidth users and

Has anyone actually used these USB speakerphones http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_ W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewIte m Seems to get a pretty good review here http://voipspeak.net/index.php?option=com_content&task=view&id=39&Itemid =27 But looking for real world feedback. Cheers,