similar to: Is there always data at /dev/zap/1?

Hey all...I'm brand new to * and I want to convert my home into a pbx type setup. I've figured out that I want a Wildcard X100P to bring my single POTS CO into my Linux box. My problem is that I'm sure sure what I need to do to get my analog phones connected up into a structured phone system. It *looks* like I can go the route of the Cisco Analog -> VOIP for about $100 on ebay.

Hey all...I'm just getting my * setup and right now all I have is an FXO but no FXS. I wan't to get rid of telemarketers by having * pick up the phone if there is no CID present, give the caller the Zapateller tones and then ask the user to input their phone number via Privacy Manager (yes I realize that this won't get us any where given that I can't re-ring the phones without FXS

Does anybody out there run * on an AT&T broadband phone line? I'm not seeing any callerid and I can't tell if its AT&T doing something funky or if its my setup. I do see CID on my normal phones.... Thanks, Chris -- The face of a child can say it all, especially the mouth part of the face. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!

Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT.

Hey all...for whatever reason my caller id doesn't appear to be working. My setup is simple (Wildcard FXO and thats it) and I'm just expecting the Caller ID to show up on the console. I'm seeing this: *CLI> -- Starting simple switch on 'Zap/1-1' NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID failed checksum NOTICE[262161]: File chan_zap.c, Line

I'm working on migrating my home POTS phone system into an Asterisk PBX. Currently I have my FXO and FXS setup and working great. Zapateller is nice! I'm working on my voicemail now and currently have my 3 mailbox answering machine plugged into my FXS along with all my other analog phones. I'd like to slowly migrate away from the FXS answering machine in favor of Asterisk Voice Mail.

Hi everybody... I'm having an odd problem with voice mail on a recent CVS of * where it appears not to detect a hangup on FXO and * will keep treating the call as new and continue leaving voicemails until the max has been reached. It will then continue trying to leave voice mails and basically makes the system unavailble to any further incoming or outgoing calls on that FXO..has anybody

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

Hey all..I'm looking to start with a single FXS card but with 3 extensions for VM purposes only. I'd like to know if there is a way that you can have different stuttering dialtones depending on which extension has a VM. For example If x103 and x104 have VM can there be a distinctly different stutter for each mail box and have them play back at once or back to back so that when you

I'm having a problem with Asterisk picking up the Zap/1 and thinking its a new call when instead I've already been on the phone talking to a person. This is not my ideal setup and currently I have just an FXO card and Asterisk is in parallel with my phone system instead of being in the front. I'm not sure if the problem would be fixed by adding an FXS card and putting my phones on

Hey all...I can't seem to figure out *exactly* what needs to be done when I can't flash over to an incoming callwaiting call on FXO from an FXS card. Right now if I get a callwaiting call from the FXO and hit flash nothing happens. I've been over the archives and google and it appears that this is possible and I guess it can be done with *0 but I haven't found the config on

hi all i'm running debian sid..and i tried to compile the las version of compiz because the old version of compiz did not supported compiz settings.. btw here is what i did: 1) followed the official compiling guide => http://go-compiz.org/index.php?title=Compiling 2) try to run by launching compiz...but the only results was to not view the windows' borders.. so i searched in the

> On 6/21/19 10:56 AM, Tierney, Luke wrote: > [...] > Something that would be useful and is being considered is having a > mechanism for registering default condition handlers. This would allow > the condition to be re-signaled with a custom class and then having > a custom conditionMessage method is less likely to cause conflicts. Is it correct that you are proposing something

Ok so I can use any ID3v2 or v1 tagger (say the perl one) for FLAC files and things will be kosher? I'd LOVE to be able roll my own tagging scheme like you can in Vorbis and I don't mind doing some code either..I wasn't aware that FLAC was able to do that..or at least I didn't userstand how to do that. I guess that is where you register for your own developer ID eh? Josh

So there are no command line options like in oggenc to do artist and misc tags? Sure if you've got a solution that I don't have to do from scratch then post away! :-) Thanks! Chris benny k. wrote: >hi chris, > >On Thu, Mar 14, 2002 at 09:22:06AM -0700, Chris Hirsch wrote: > >>Hey all..I'm brand new to FLAC and learning to use it. Can somebody tell >>me how

On Fri, Dec 27, 2019 at 7:04 AM Eddie O'Connor <eoconnor25 at gmail.com> wrote: > > Wouldn't a "find grub / find grub*"....or "locate grub / locate grub*" > work?... > locate would require to install that package and run updatedb first. But, it would save time looking for files that have been already added to its db, specially compared to running

Hello List - I've done some reading on voip-info regarding hardware requirements for an Asterisk server; but I haven't been able to find anyone doing what we plan to, so I am hoping you can assist. We are looking to provide a voice mail only Asterisk solution for approx. 100 homeless people, a customer of ours is planning to provide the service. The Asterisk service will reside in our

Hello, I have a graphics-related question: I was wondering if anyone knows of a way to create a bar chart that is colored with a three-part gradient that changes at fixed y-values. Each bar needs to fade green-to-yellow at Y=.10 and from yellow-to-red at Y=.20. Is there an option in a package somewhere that offers an easy way to do this? Attached is a chart I macgyvered together in Excel using a

Hi all, I''m trying to figure out if I can serve a certain template that matches a hostname or serve the default like I could if this was just a file. Works: file { ''/etc/yum.conf'': ensure => file, source => ["puppet://puppet/yum/yum.conf.$hostname", "puppet://puppet/yum/yum.conf"], } Trying to figure out:

Hey all...I'm trying to find a phone that will support multiple MWI so that I can have a shared central phone with say 4 users who can see visually that hey have messages waiting. Is there any phone that will do this possibly by re-assigning a soft-button? Can the Polycoms do this since those seem to be the phone of choice these days? Thanks!