similar to: Asterisk/VOIP to PSTN (?)

Try this subject line if you will. On 11/14/06, joe a. <joea@j4computers.com> wrote: > > Did not know how to make up a subject line for this. > > I have a dial plan that allows a caller can try my cell phone. And that's > fine. If the call cannot be made, it sends caller back to voice menu. > > However, I'd like a way for the caller to elect to go back to the

I love TrixBox, with the custom config files you can tweak pretty much with TrixBox too, I have at least done some. Plan to do a plain Asterisk install later, but for now I learn a lot about the config files just with TrixBox. Some things might be a bit harder with TrixBox due to some of the premade dial plans, but can get it to work :-) _____ Fra: asterisk-users-bounces@lists.digium.com

Take a look at freepbx 2.2 beta. We have made both ringgroups and follow-me have a call confirmation option. When used, the ringgroup/follow-me extensions that are outside lines (like your cell phone) must confirm they want the call (press 1 to accept, 2 to decline). All the while the caller hears ringing (or MoH if chosen). If no answer, they are sent on to where ever else you want them to go

Experiencing one way audio using IAX2. I did see some other posts on this, and see there may be some internal issues with asterisk and one way audio. Can this be a widespread problem? So many seem to be using IAX, I find it puzzling. Some information points to this being a problem on asymmetrical connections. This is a decidedly asymmetrical connection, with 1.5 Mbs download and 256 kbs,

This is, hopefully, just a case of brain fade. With zapata.conf and zaptel.conf in place, asterisk loaded, no dial plan and all LEDS on the card lit, I get no dial tone, plugging an analog phone into ports 1 or 2, only a buzz and click. zaptel.conf - defaultzone=us loadzone=us fxoks=1,2 fxsks=3,4 zapata.conf [channels] signalling=fxo_ks language=us context=phones-1 group=0

I have a setup with a polycom 601 and an act p160s. All on local segment, no NAT. Can call the act p160s, from the polycom, rings, connects, and a conversation can take place. The reverse is not true, Dialing from the act to the polycom does not work. SIP debug shows, at the end, "Incoming call: got sip response 416 "unsupported URI Scheme" back from 192.168.0.xxx. Which is

On Feb 2, 2014, at 6:18 PM, Andrew Trick <atrick at apple.com> wrote: >> The result of such a system would produce weights for every block in the above CFG as '1.0', or equivalent to the entry block weight. This to me is a really useful metric -- it indicates that no block in the CFG is really more or less likely than any other. Only *biases* in a specific direction would

Hi;? When connecting via VoipBuster or VoipStunt, "I can hear them but they can't hear me"?. This happens with VoipBuster or Voipstunt. Registration is done correctly. I thought it could be something related to NAT, but I don't have this problem when using VoipUser or Asterisk2PSTN, for example. ? I tried with different codecs: gsm, alaw and ulaw but no change. ? So, now?I

Hi - I need to configure Asterisk (Trixbox 2.2) and my ATA-186 with both ports. I need to be able to call one port from the other-- the idea is to have two phones in two different locations that _can_ call each other. So, in reading the Asterisk Wiki and other sites, the best documentation I found was this: http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt

Was messing with a polycom 501 and changed the IP from dhcp to static. Working with a user remotely. Now, the user says the phone does not show anything on the LCD and does not respond to any buttons. When rebooting, there is text shown as it proceeds. ?? Is there a way to reset this to a default? Does not respond to ping on the address we set. joe a.

Is Zaptel no longer available? I returned to a long shelved project (using TDM400P and a customized, canned version of *) and, getting to the configuration, find wctdm is not there. I recall the authors where very "enterprise" oriented and focused on T1 cards. So they left analog support out. Anyway, before I abandon all hope and dive into the "new stuff", I thought I would

Since the beginning (of my Asterisk life) I have an install that is, supposedly, set up for call waiting. Using a TDM400p, with FXO and FXS modules. On the Analog phones, I can hear the Incoming call (call waiting) tone, but the system does not respond to a "hook flash", to place the current call on hold and answer the incoming call. I have not attempted, nor research how/if this can

Mark Leeds <mleeds at mlp.com> wrote (to S-News): > does anyone know the R equivalent of the SPlus rowVars function ? Andy Liaw <andy_liaw at merck.com> replied: > More seriously, I seem to recall David Brahms at one time had created an R > package with these dimensional summary statistics, using C code. (And I > pointed him to the `two-pass' algorithm for variance.)

Getting "no service" display on aastra 480i. Sip debug shows an "unathorized" blub when the aastra tries to register. Some reading indicates that 1.4 firmware wants aastra.cfg and mac.cfg in /tftpboot/. There are none. Anyone have basic config files? Or can point me to a good link? All links I have tried, that purport to have config files, are either dead or error out.

This is a follow up to an earlier post. Looking for a means to "individualize" incoming FAX, so as to distribute them to the intended recipient. While the PBX is based on Asterisk, it is not possible for me to "enter" the box to modify things, to any great degree. I thank those who mentioned IAXMODEM, earlier, but that seems a no go. Currently, there is a dedicated T1 into

I wonder whether the functions for skewness and kurtosis in the e1071 package are based on correct formulas. The functions in the package e1071 are: # -------------------------------------------- skewness <- function (x, na.rm = FALSE) { if (na.rm) x <- x[!is.na(x)] sum((x - mean(x))^3)/(length(x) * sd(x)^3) } # -------------------------------------------- and #

How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ? /HHA

Hello, I'm very new with * and I would really appreciate some help to implement a SIP to PSTN Gateway. My current scenario includes an * box with a TE405P board. I have a 1.5Mb connection to the outside world (using a router with firewall capabilities) and channel banks that allow me to connect the T1s coming out of the TE405 board to PSTN network (carrier). I need to configure * to

I'm looking for recommendations for UK-based VoIP-PSTN gateways. They should ideally offer: - IAX connection - Multiple simultaneous calls on a single account - Lower call rates than BT Business - Auto-top up or invoicing in arrears I can find several that offer one of these facilities, but none that offer all. Thanks! -- David Gurr Congruity Ltd. Hemel Hempstead, UK

I'm about to purchase an adaptor for a POTS data modem and was looking at the Sipura line of adaptors (SPA-1000, SPA-1001, SPA-2000, SPA-3000). Do these work well? Anyone have a suggestion on which model of the Sipura I should get? Does one work better with * than the others? Are there other adaptors that work better that I should get? Thanks, -Dave -------------- next part