search for: altaphon

...reflect changes made in zaptel and zapata.conf entries. Any other config files that it doesn't reload? >From: "Asterisk@Home" <asteriskathome@yahoo.com> > >Did you go into AMP and configure some place for >incomming calls to go? > >--- David Josephson <dlj@altaphon.com> wrote: > > > >>Is there a configuration difference for clone X100P >>cards versus >>"compatible"? I have a similar problem to what David >>Shaw posted earlier >> >> > >

Is there a configuration difference for clone X100P cards versus "compatible"? I have a similar problem to what David Shaw posted earlier today. 0.5 installed OK, but mine just with one X100P clone. Default config files, edited zapata.conf per the FAQs so it includes the line channel => 1 without the semicolon. Any outgoing call attempt returns "all circuits are busy"

Steve Blair writes > I can redirect and relay calls to numerous destinations via >SER but because the Octel needs an SMDI interface for mailbox >identification I am stuck, none of the solutions thus far support >SMDI-SIP munging. > > I just started thinking about the possibility of using Asterisk >with a few FXS cards to provide the gateway between SIP and >the Octel.

Rob at draughon.org writes > I recently obtained a Western Electric multi-line phone and am >seeking help with getting this beast working with *. > > The interesting stuff in my * implementation consists of a T100P >card, a TDM400P card, and an Adtran TA750 channel bank with three quad-port >FXS modules and a quad-port FXO. The TA750 is wired to a 24-port Cat 5 patch >panel

Pete VK2YX writes > I've been involved with IRLP for about 5 years and am one of the original > install team. I've gone through the emmotions of allowing "other" networks > connect to IRLP and I know its caused some lots of headache. > > As far as a closed network goes, yes there is LOTS of passion to keep it HAM > only and I'd have to support that notion.

Daniel Nystrom wrote > It seems like the Radio discussions is closing in on something I was > interested in. > How about controlling 30 2-way radios via E1 and 30-channel "Mux" > (channel bank?) with E&M signalling? > I think the Mux uses CAS and each channel has Audio out, PTT, Audio > IN, Busy. 6-wire connection i guess? > That should be a really nice setup

A few bits and pieces that I hope someone here has answers for. Basically, I'm in the audio business and I hope to get better control of the audio performance in *. 1. I seem to recall seeing a tonelist parameter that allowed tones to be set at levels other than 0, for instance !1000/500@-20 would generate 1 kHz for half a second at -20 dBm0. But this doesn't work and I can't

Greetings list, We are having a puzzle with * (asteriskathome 0.5) and SIP phones (SPA2000 ATA's). If callwaiting is enabled, everything (including call waiting) is normal. If callwaiting is turned off, the phone will not accept incoming calls and the call goes straight to whatever is programmed for the busy voicemail response. It doesn't matter whether reinvite is on or off, or

On Fri, 2005-03-25 at 08:00 -0800, Mark W Wood wrote: > > I have searched both the wiki and googled looking for a solution to > > a square key configuration. I need to have C.O. lines to appear on > > the buttons to facilitate a small office. All of the users can see > > each other and calls are put on hold and picked up by the other > > users instead of transferred.

>> >> >>Can I program a specific C.O. line directly to a button? >> >> Adopting the Critchfield style for a moment, no, *you* probably can't. But, depending on one's expertise with the hardware and Asterisk, it can certainly be done. I am by no means an Asterisk expert but have about 35 years of random phone experience. I have an ancient Siemens HiNet

Has no one else experienced the problem I mentioned a few days ago with app_dictate? Or maybe no one is using that app. We're having a problem with choppy audio and failure of the accelerated playback feature which seems to be consistent on a couple of installs, failing with some SIP carriers and working fine with others. MOH and other audio playback features seem to work fine. What's

Has anyone found an inexpensive E&M trunk card that will play with *? Looking for an interface to a legacy electromechanical PBX that's able to pass answer supervision. Docs on the X100P card would be helpful, we could probably pull E&M out of that. Any ideas?

On the various technical issues raised here (OK, we posted the rules, we won't discuss the legality anymore) I think there is only one main obstacle to using FRS radios for extensions on *. They are simplex (push-to-talk, release-to-listen). The protocol for dealing with voice-activated-switching (VOX) has been used in ham and public safety simplex autopatches but it's really tricky

Trying to get some straight info from the VOIP providers is difficult. Say there's a small Asterisk switch and it's registered with Broadvoice or LiveVOIP or someone. There are a couple of people using the switch, one is on an outgoing call with the VOIP provider. What happens when someone else initiates another outgoing call through that provider on the same SIP registry? Does * know

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

Rich Adamson writes >>GMRS, FRS and MURS radios may not be interconnected with the PSTN (47 > > CFR 95.141). There has been a lot of talk from lobbyists to clarify this > > rule, but as it stands you could conceivably connect a *private* network > > to GMRS or MURS radios (you can't make any plugins or modifications to > > an FRS radio that isn't type