similar to: Numbering plan for incoming call CLID on chan_zap (PRI)

Hi all, I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID. I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls. Thanks for any

hi, on our incoming E1-PRI from german telco Arcor the leading 0 for the (area access code in europe) and the 00 (country accescode in europe) are missing on incoming callerids. only prepending a single 0 is not the solution as suggested by some writers on this list, because there is no way to differ between national and international callerids and it's not possible to make the decission

Hi, I'm looking for a mean to send digital data over an E1 line, just like isdn4linux or Capi via AVM's FritzCard is able to do it with BRI lines (e.g. for PPP or ISDN raw connections). I'm not looking for modulated audio data representing digital data, like fax or the analogue modems of former times. I want an interface to the ISDN raw data, with an outgoing call marked as

hi, after several problems getting the right callerid on a E1-PRI there is (so far) only one problem left: when receiving calls over the telephone network from another E1-PRI that has a "Caller ID no screen" capability (e.g. a bank and a customer of us), asterisk does not get the callerid that is set up by the calling PBX, but the callerid of the trunk of the calling PRI. no matter

Is it possible to send CLID NAME on a PRI? The numbers we send out are being received by telco and propagated, but the names we send out are not showing up. Is this a feature in PRI? Do we need to set PRI_NET instead of PRI_CPE? Is this just not possible? Is this a telco config issue? Thanks for your help... I've read voip-info, and various other sources, and search engines, and google...

Hi: I'm trying to plot dates on the x-axis of a code, but the legend is not being displayed. I receive the following error: Error in match.arg(x, c("bottomright", "bottom", "bottomleft", "left", : 'arg' should be one of bottomright, bottom, bottomleft, left, topleft, top, topright, right, center In addition: Warning message: longer

Hi, how can I disable event_log in order to reduce hard disk activity? I can't find any hints in conf files. Must I hack the source code or even use brutal methods like creating a dir called event_log in the log dir, in order to prevent asterisk from creating an event_log file? (Just chmod a-w event_log does not work, unfortunately.) Thanks for any hints! Roger.

Hi, there is a feature, which I would like to use with asterisk, and I assume it exists. Unfortunately I don't know how to say it in english. In german it's "einen Ruf heranholen". It means: The phone set of my collegue is ringing, and I'm hearing the ringing. I know, that my collegue is not at his desk, and now I want to answer the call at my phone (instead of running to

I am experimenting with oh323 channels and h.323 gateways and a Cisco CallManager. I am not using a gatekeeper at this time. Is it possible to place calls coming into Asterisk from specific peers into specific contexts? In iax.conf eaxh peer has a context in which I can specify the context an inbound call will be placed in. I don't see anything like this in the oh323.conf file or the oh323

Hi, I have unpleasent short audio gaps when a perl based agi scripts starts. Thus, I now started to put all those things in C programmed daemons for fast-agi. Anyway I'm looking for another mean, which would help me more quickly. I noticed, that all agi scripts are running with system priority -11, like asterisk does. This is really waste of priority. I would like to have the AGI scripts

I launched www.goiax.com last week, which is intended to promote the use of IAX as a free and open source alternative to products like skype. There is no charge for the service. Right now I have free outbound to united states toll-free and us domestic numbers working. Currently the site hands out a virtual 87820-xxxxxxx number but I intend to add the ability to get a free United States DID

Hi, I am plotting a policy function (result from a dynamic stochastic optimization problem, discretized approximation). The policy function maps from an 2 x 2 x 2 x 3 x B x F state space to a B x F state space (B and F are usually between 4-6, and represent domestic and foreign savings. The other variables are income (Y), inflation (Pi), domestic and foreign interest rates (R and Z)). I

Hi, some days ago we discused here the need for an analyzer for the 1000 Hz tone, as opposite application to Milliwatt. Here it is: Mwanalyze http://planinternet.net/download/voip/asterisk/app_mwanalyze.c It performs a Fourier analysis for a fixed frequency and tells the amplitude. The frequency is not limited to 1000 Hz, but can be passed as argument. The periode duration must be a mulitple

Hi, I put reinvite=yes in my sip.conf. For testing, I restricted the codecs to alaw. I have no modifiers in my dial command. Thus, there should be no reason not to reinvite. Call (sip, authenticated) comes in and is forward via SIP (not authenticated) to another asterisk box. Unfortunately, media path still passes through the asterisk box in the middle. Using sip debug I even can't find

Hi, some months ago, I had the problem with an asterisk-1.4.x- Version, that some calls (but not all) were interrupted 64 seconds after connect (a call limit of 86400 seconds was installed using the S()-parameter). It was just a test machine, and later, I switched to callweaver, and the problem had gone. Thus, I never investigated this problem. Now, I upgraded a machine for production use to

I am downloading time series data from FRED. I have a working download, but I do not want to write out the download for all 50 states likes this: IDRGSP <- read.table('http://research.stlouisfed.org/fred2/data/IDRGSP.txt', skip=11, header=TRUE) IDRGSP$DATE <- as.Date(IDRGSP$DATE, '%Y-%m-%d') IDRGSP$SERIES <- 'IDRGSP' IDRGSP$DESC <- "Real Total Gross

I'm having problems getting a SAMBA share on a remote (Windows XP) machine to allow me write access to files. I can copy/paste whole files OK from my CentOS 5 box, but can't do any editing of files on the Windows XP machine, which I use as a file server for most of my domestic data (historical reasons for using Windows!). I've tried re-setting permissions to allow write access to the

I have a bit of a weird problem that I'm having great trouble debugging. I have a TDM400P PCI card with two FXO and two FXS modules. Both FXO modules are connected to BT lines here in the UK. Both BT lines have V23 Caller-ID, which works fine with Asterisk. Both asterisk and zaptel are fresh from CVS. Both FXO modules (channels 3 and 4) are in "group 1" for outgoing calls. My

Hi all, many thanks for the replies! On 14/05/18 19:05, Parke wrote: > On Mon, May 14, 2018 at 4:44 AM, Keith Whyte <keith at rhizomatica.org> wrote: >> but then I read that no, each host much have the key of >> the other to establish the direct connection. But I am looking at >> tcpdump right now in the terminal and seeing the UDP tunnel packets >> flowing from

Hi, My question is how do I gain control over what values the X and Y axis show. Below is a sample plot I have made and want the X axis to represent a time vector with values taking the form Q1.60, Q2.60, Q3.60...Q1.90..etc...Currently the X axis starts with value 0 and increases by 1 through the end of the sample. win.graph() matplot(v.0,log.diff.v.6,type="l",lty=1, col=2,