similar to: Google Puts the Final Nail in the Google Voice Coffin

On Thu, 5 Oct 2017, hw wrote: > me at tdiehl.org wrote: >> On Wed, 4 Oct 2017, hw wrote: >> >>> A SPA122 ATA from Cisco might be useful as a gateway, they are cheap. >>> You?d be using it kinda in reverse, but I don?t see why that shouldn?t >>> be possible. >>> >>> Other than that, specialized cards have come down in prices, probably

me at tdiehl.org wrote: > On Wed, 4 Oct 2017, hw wrote: > >> Jose Maria Terry Jimenez wrote: >>> El 4/10/17 a las 17:45, david escribi?: >>> >>>> Folks >>>> >>>> A have a PCIe modem (Conexant ChipSet, PCI id = 14f1:2f83. It interfaces to my land-line (POTS) telephone line in the United States. On Windows, I had a good answering

At 10:20 AM 10/4/2017, you wrote: >On Wed, 4 Oct 2017, Jose Maria Terry Jimenez wrote: > >>El 4/10/17 a las 17:45, david escribi?: >> >>>Folks >>>A have a PCIe modem (Conexant ChipSet, PCI id = 14f1:2f83.? It >>>interfaces to my land-line (POTS) telephone line in the United >>>States.? On Windows, I had a good answering machine package

On Wed, 4 Oct 2017, Jose Maria Terry Jimenez wrote: > El 4/10/17 a las 17:45, david escribi?: > >> Folks >> >> A have a PCIe modem (Conexant ChipSet, PCI id = 14f1:2f83.? It interfaces >> to my land-line (POTS) telephone line in the United States.? On Windows, I >> had a good answering machine package (Ventafax) that reported CallerID, >> recorded

On Wed, 4 Oct 2017, hw wrote: > Jose Maria Terry Jimenez wrote: >> El 4/10/17 a las 17:45, david escribi?: >> >>> Folks >>> >>> A have a PCIe modem (Conexant ChipSet, PCI id = 14f1:2f83. It interfaces >>> to my land-line (POTS) telephone line in the United States. On Windows, I >>> had a good answering machine package (Ventafax)

Hi all, I managed to get all my VMs (5 in total) up and running in bridged mode with public IP addresses from my ISP. All are running Debian Etch in a Dom) of Debian Etch with Xen 3.1 using the instructions here: http://www.howtoforge.com/debian_etch_xen_3.1 But ideally I only want two of the DomUs on the public internet with the other three only on an internal network and the two

Putting the Genie Back in the Bottle: More RedHat Legal Shenanigans with CentOS http://nerdvittles.com/?p=8721 Closing the Book on CentOS: [...] http://nerdvittles.com/ Thanks to the author of the above articles. jb

I just got cepstal working fine in the dial plan using code like: exten => 511,5,AGI(cepstral.pl|Welcome to my house finder. At the beep enter your zip code.) The php script it calls is based on the nerdvittles weather one so it calls a webpage which prints to the screen, the nerdvittles code uses system to generate the .wav file then has the dial plan call it via: //php script $retcode2 =

Hi, I don't know how to call this functionality, but what I want to do is join an already established communication between PSTN---FXS_connected_phone using my SIP phone (I have an asterisk v11 with digium TDM400P at home) Is it possible? What I don't want is using the conference sound and menu.... It's just a normal call between to channels that I have to join for few minutes.

Actually I can start and receive SIP calls (PC client, iphone client) but?I have an issue with calling external number throught PSTN (certified-asterisk-1.8.11-cert2). I'm having this ?error when making a call: *CLI> ? == Using SIP RTP CoS mark 5 ? ? -- Executing [9000 at local:1] Dial("SIP/3000-00000006", "DAHDI/1/4384019357,10") in new stack [Jun 23 16:18:09]

Package: logcheck Version: 1.3.5 Severity: minor Hi, reading logcheck source it seems that it requires nail for MAILATTACH to work, however it is not suggested/recommended. (JFTR it is debatable if nail is appropriate or something else should be used) thanks, filippo -- System Information: Debian Release: squeeze/sid APT prefers unstable APT policy: (990, 'unstable'), (500,

Dear All I'm having problem with some script which worked a few months ago (on a different computer that might well have had a different version of R installed, so perhaps it has to do with the old version of R?): library(reshape) Loading required package: plyr > tble.data <- melt.array(interp, varnames=c("tme","lon","lat")) > > allyrs.interp <-

Thanks I had this line in my /etc/asterisk/chan_dahdi.conf : include=/etc/asterisk/dahdi-channels.conf the file /etc/asterisk/dahdi-channels.conf was generated by /usr/sbin/dahdi_genconf I simply did that : cat /etc/asterisk/dahdi-channels.conf >> /etc/asterisk/chan_dahdi.conf It works now. May be the option "include" is not supported within the file chan_dahdi.conf

Hi, Can anyone tell me if the Motorola Q has its Bluetooth always on like the IPhone? I want to use the Motorola Q in a Proximity Detection setup like that described on nerdvittles.com. I know the Treo 650 does not work well since the display must be on for the bluetooth to be on and this eats power. Thanks Chuck Bunn

"... we need a baseline to work from - one standard format that (if all else fails) everything can fall back to. This doesn?t need to be the most complex format, or the most advertised format, or even the format with the most companies involved in its creation. All it needs to do is to be available, everywhere. The codec in the frame for this is Ogg Theora..." :)

Hi, Plenty to talk about today. Dave Nelsen, a founder of Talkshoe, has a lot of experience in the telecommunications space and he joins us today to chat about its current state, conferencing and whatever else comes to mind. So we have a meta conference aout conferencing, it won't be the first time :) You probably saw John Todd's message on one of the lists: Skype for Asterisk is in open

Hi, Plenty to talk about today. Dave Nelsen, a founder of Talkshoe, has a lot of experience in the telecommunications space and he joins us today to chat about its current state, conferencing and whatever else comes to mind. So we have a meta conference aout conferencing, it won't be the first time :) You probably saw John Todd's message on one of the lists: Skype for Asterisk is in open

I am trying to get ListPro from Ilium Software to register. According to their tech support, it needs to write a file out to C:\Documents and Settings\All Users\Application Data\Ilium Software\ListPro However, the error the program gives is "Registation Information could not be saved". I created the directory within the drive_c directory. I have also verified the spelling of the sub

Hi, voicemail plays after hitting "#" as final file "auth-thankyou". Is there any possibility to change this behaviour? Custom soundfile or disable it perhaps? Thanks for your answer(s)! -Thorsten-

Hi, Today at 12 Non EDT, Dan York will be with us to talk about the recent on and off moments of Google Voice SIP URI calling. Like Skype + Asterisk (or any SIP), Google Voice and SIP compose the other shoe waiting to drop. We're following this with interest. So GV turned on SIP URI and then a few days later, turned it off. Why? Did the geeks (like us) jump on this too quickly or too heavily?