similar to: Incoming Bogota DID

Starting around 10:00 AM EST. All services from them whether I connect by IP or DNS (both east coast and west). Anyone else? Fred Posner fred at teamforrest.com Main: +1 (212) 937-7844 Direct: +1 (503) 914-0999 www.teamforrest.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Regards -------------- next part

***** Proceedings by Springer. Indexed by Scopus, ISI, etc. ------------ ICITS'20 - The 2020 International Conference on Information Technology & Systems 5 - 7 February 2020, Bogota, Colombia http://www.icits.me/ <http://www.icits.me/> ------------------------------ ------------------------------ ------------------------------ ------------------------ Scope

***** Proceedings by Springer. Indexed by Scopus, ISI, etc. ------------ ICITS'20 - The 2020 International Conference on Information Technology & Systems 5 - 7 February 2020, Bogota, Colombia http://www.icits.me/ <http://www.icits.me/> ------------------------------ ------------------------------ ------------------------------ ------------------------ Scope

Here's the story... Nortel system set to use g711 @ 30ms payload ... Asterisk box would need to communicate to that box @ 30 ms and another end point at 20 ms. I've seen discussions of setting this to a different size, but seems to be limited to the entire codec and not on a per peer basis. Anyone have luck with this? The Asterisk can be 1.4 or 1.6.x... I've a preference for

On Fri, Jul 30, 2010 at 1:15 PM, Fred Posner <fred at teamforrest.com> wrote: > On Jul 30, 2010, at 5:04 AM, Andra=C5=BE wrote: > > > Ok, problem is another, when I run configure, it write this: > > checking for tds_version in -ltds... no > > configure: *** > > configure: *** The FreeTDS installation on this system appears to be > broken. > >

List, We're working on an upcoming job that may require us to access a web service (WS). I'm curious to hear peoples thoughts on the best way to do this with asterisk. We'll be submitting a single number to the WS and it will return a success or error. One solution would be to write a simple perl script to interface into to the WS, and use SYSTEM() from asterisk to call it.

Hello Asterisk People. Asterisk have a really annoying bug, i use frequently the manager status command and when asterisk decide not to show the "statuscomplete" event, it really don't show the "statuscomplete" string, in fact none of the "AgentsComplete", "QueuesComplete' are shown.... I use it for monitoring a queue, but this is really getting me

Spanish/Espa?ol: Hola a Tod at s, Estoy en la implementaci?n de una soluci?n a medida en la cual requerimos conectarnos v?a SIP a un IMG de DIalogic y salir hacia otras centrales (MSC) v?a SS7, la conexi?n entre Asterisk y el IMG es SIP, pero como requerimiento necesito enviar un NOA (Nature of Address) con valor 8 por requerimientos normativos, hasta ahora no he podido encontrar como puedo

Hi, I used in samba 2.9.0 the parameter postscript for hylafax application. I read that samba 3.0.0 the parameter postscript was removed. I need to know what parameter I can use in samba 3.0 in order to replace postscript. Thanks for your help -- Jazmin Baron Garavito Tel.: (57-1)3212333 Bogota - Colombia

Hi Anton Some days ago I was talking by phone with you from Bogota, Colombia, SouthAmerica. I'am interested in to buy a good and cheap channel bank and I need to know your models and prices. Could you send me an e-mail with your best quotation included the sending price, besides cold you give the Bill like a: Sample unit for developmente propose without comercial value; COST: 25 usd;

Starting playing with asterisk 1.6.1.0 i found the following problems: In the cdr_pgsql, the sql statement is wrong: 2009-06-25 12:17:01 COT LOG: statement: INSERT INTO cdr

Hi! Why do I still get an error message on the try function?: > options(show.error.messages = FALSE) > > trab <-matrix(nrow=6,ncol=6) > trab[,] <-0.0001 > rownames(trab) <- c("A","B","C","D","E","I") > colnames(trab) <-

Hi R-Core, i am using nnet and DEoptim, Xcc=matrix(rnorm(100,0.5,0.08),50,2) Ycr=matrix(rnorm(50,0.2,0.05),50,1) pred_regm1 <- function(A) { A1=A[1] A2=A[2] A3=A[3] regm1 <- nnet(Xcc,Ycr,entropy=T,size=A1,decay=A2,maxit=2000,trace=F,Hess=T,rang=A3,skip=T) dif=sum((predict(regm1,Xcc)-Ycr)^2) return(dif) } somar=DEoptim(pred_regm1,c(1,0.00001,0.01), c(25,0.999,0.95),

I am trying to install the package boss but i am getting error below: Please advice install.packages("boss") --- Please select a CRAN mirror for use in this session --- CRAN mirror 1: 0-Cloud 2: Argentina (La Plata) 3: Argentina (Mendoza) 4: Australia (Canberra) 5: Australia (Melbourne) 6: Austria 7: Belgium 8:

Hello, I've been trying to setup a site library that allows the users to manage the R packages themselves, but am having an issue with permissions. As seen below, when installing a package using install.packages, the umask used is always 022. Instead, I would like it to be 002, allowing any user to update packages that were installed by another user. Can someone explain how to get

Hello Asterisk People, I am having trouble connecting asterisk to a huawei SoftX3000 Switch, i can succesfully connect other softphones like Zoiper, but when it comes to Asterisk SIP Client, the system doesn't authenticate, i have the following configuration: peer: 10.220.0.2 username: 4857768 password: 4857768 the configuration is as follows: in the general section: register =>

I am trying to install the package boss but i am getting error below: Please advice install.packages("boss") --- Please select a CRAN mirror for use in this session --- CRAN mirror 1: 0-Cloud 2: Argentina (La Plata) 3: Argentina (Mendoza) 4: Australia (Canberra) 5: Australia (Melbourne) 6: Austria 7: Belgium 8:

Hi All; How to increase the waiting time between entering the digits for the analoge phone device that is connected to fxs? Is it by DigitTimeout? But how it will be apply for analoge station if the user just pickup the handset and dialed the number? Any help? Regards Bilal

I missed the talk that was given on wideband codecs @ astricon last week. I tried to lookup the speaker on astricon.net, but that website seems horribly broken at the moment, showing only a tmcnet video, whatever page i click on. Would somebody have the contact details for that speaker ? Greetings, Zoa