similar to: inbound call voip providers

Hello everyone! I 've looking for carriers which can terminate my international calls. They must accept payments from Argentina and give me interconection to my Asterisk. I'd appreciate your help or recomendations. Regards. -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 Share your knowledge, use free software.

Hi all I sold new TDM800P card with 8 FXO ports, someone know if can be use this card on AsteriskNOW or trixbox? What can i do for use this card? Thanks. ---------- RafaelCanchola Product Development Engineer, FonetGlobal Inc. rcm at fonetglobal.com http://www.fonetglobal.com Ph. + 52 800 022 10 21 ext. 214 + 52 442 167 08 00 VoIP 523663899 d00d! cyberalph -------------- next part

I'm developing a small linux/speex-based audio streaming (VoIP-like) application, and am looking for a binary windows app that can play speex content (at least from a file, preferably from an IP data stream) out through the windows sound system. This would be very helpful as a test tool. Any recommendations? I'd prefer a binary as I don't have development environment available for

Dear all, I have been using Speex during last year over PC with very good results. Now I'm trying to move my code to the DSP TI C6713. It's the first time I work with a DSP and with Code Composer. I use speex quite well but I'm not very good at compiling for different platforms. After reading all manuals I tried to compile the speex library with Code Composer and the speex options for

Interested in building a speex codec (basically audio <-> speex <-> data stream) using TI's small MSP430 microcontroller. Is there any way to estimate feasibility based on CPU requirements? Example - speex is happily encoding on an old Pentium-1 processor (166 MHz) using about half the CPU (as reported under Linux); the TI microcontrollers are much slower yet (8-16-25 MHz) and

Hello all, I'm trying to figure out if there is some way I can log agents in and out by just having them call an extension. Ideally I'd like to have it set up where each agent just dials an extension to log in and the same one or possibly another one to dial out. I got the logging in portion to work by not using a password in agents.conf , but the manual log out has me stumped. Any input

Greetings, fellow R'ers. How can I get this frame in R: ID meas ID.1 meas.1 1 1.1 3 1.2 2 2.1 4 2.2 to look like this (stacking): ID meas 1 1.1 2 2.1 3 1.2 4 2.2 It's not really the reshape function (or is it?) because we can consider the additional columns, viz., ID.1 and meas.1, as independent of ID and meas so it is basically a stacking

I have a problem where I need to loop over the total combinations of vectors (combined once chosen via combinatorics). Here is a simplification of the problem: STEP 1: Define three vectors a, b, c. STEP 2: Combine all possible pairwise vectors (i.e., 3 choose 2 = 3 possible pairs of vectors: ab,ac, bc) NOTE: the actual problem has 8 choose 4, 8 choose 5 and 8 choose 6 combinations. STEP

Please help. Asterisk 1: Sip.conf [VoipDirect777821] type=friend host=dfvvd.dyndns.org username=VoipDirect777821 secret=xxxxxxxxxxxx accountcode=5260477782 amaflags=billing context=Incoming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes Asterisk 2 sip.conf GNU nano 1.3.12

Is there a simple function or process that will create a matrix of permutations with replacement? I know that using the combinat package ###### begin R code ###### > library(combinat) > m <- t(array(unlist(permn(3)), dim = c(3, 6))) # we can get the permutations, for example 3!=6 # gives us > m [,1] [,2] [,3] [1,] 1 2 3 [2,] 1 3 2 [3,] 3 1 2 [4,]

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-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the

I'm using R on Windows XP Professional SP2. I figured out the fix for my buffer overflow problem that did not allow me to use Rgui.exe for about two weeks. I would keep getting "console not found" and even when I uninstalled and re-installed got the same problem. Here's how the problem started. In Rgui.exe, I attempted to get more visible output in the buffer (I wanted

I have developed a flexible arithmetic coder implementation and I am now looking to incorporate it into a compression system for some proper testing. It is my intention to incorporate the coder into the Vorbis codec to attempt to improve on the current Huffman tree system. It would be useful if someone could point me at the main c and h files that deal with the entropy coding as I am currently

Hey all, Wondering if this is possible.. Incoming call is answered through X100P, then an extension is dialed using the same X100P card. Basically I want to dial in, enter 9 + <phone#> and have it do a flash then have it dial *08 <the same phone number> + # on the same PSTN line to have it transfer my call to another phone number. I realize this isn't very safe, but I would

I need capabilities, for my data analysis, like the Pinheiro & Bates S-Plus/R package nlme() but with binomial family and logit link. I need multiple crossed, possibly interacting fixed effects (age cohort of twin when entered study, sex of twin, sampling method used to acquire twin pair, and twin zygosity), a couple of random effects other than the cluster variable, and the ability to

Does anyone know if sip phones from any of the major IP phone vendors support 802.1x authentication? Any feedback would be greatly appreciated. Thanks in advance. ====================== Jeronimo Romero EUS Networks Email: jromero at euscorp.com Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com ====================== -------------- next part -------------- An HTML

Hi, I succesfully install spandsp chan_misdn and digium card. the rxfax works fine and I get the fax result by email. I would like to do the same using a Patton gw + zaptel but I can't receive fax anymore, the call comes in from ISDN in the Patton gw, patton sends it to asterisk, asterisk run a macro to make a tif file using rxfax, the tif file is correctly created but with a 0 size the call

Hi Folks, I'm trying to test out Asterisk overall. I'm having some problems with DTMF. Currently I'm playing with DISA, but I'm worried this will happen when I get to implementing AAs etc. I have a free SIP trunk from IPKall that I'm trying to make work. I'm able to receive calls, and I've now setup and extension with DISA and a password. I connect ok from the