similar to: Dropped Call Problem -- Looking for ideas and a consultant.

Hello, I just installed a PRI and when I make a local (seven digit) call, I get Code 28 back from the telco, (I believe code 28 means "Invalid Number") and I hear a fast busy on the phone. Here is the output: -- Executing Dial("SIP/marke-17b1", "ZAP/G1/4967171") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G1/4967171

Hi, I have a Dell Power Edge server & planning yo buy Sangoma A101D card. To configure with my Asterisk 1.2.18 & zaptel-1.2.17.1 & Free-PBX setup. So I wanted to know the steps & any issue which I may come accross if any. I have googled & have some docs handy wrt Trixbox-2.2. Just wanted to get some notes from user with custom install setup when used with

Hi Guys, Anyone might know why this error keeps showing up and inbound/outbound is not working on a Bell PRI with Sangoma A101D? -- Got SABME from network peer. Sending Unnumbered Acknowledgement No calls can be made inbound/outbound. Keeps repeating. No alarms ON and no changes been made to the system. Stopped all a sudden. Asterisk CLI doesn't show anything with full verbose for both

Hi Folks, Prior to upgrading my Zaptel sources everything was working fine. I have a X100P connected to my analogue line. The handset port of the X100P is connected to my desk phone's line 2 input. When the analogue line rings I see the CID on my line 2 but not from Asterisk on line 1 via the Cicso ATA. This used to work fine until I upgraded the sources. I get this when watching the

Dear, I have an Asterisk PBX with two E1 cards: Digium TE122 and Sangoma A101D. Sangoma card has SNMP support but Digium card has not, and also SNMP does't give me ral time information. Within CLI Asterisk I execute "dahdi show channels" but I don't get information about channels usage. What is the best way to have real time monitoring of E1 channels usage and status ???

We will be connecting our Asterisk server to ISDN 30 and intend using the Sangoma A101 card. The install location is in London (UK). Sangoma card at Voipon http://www.voipon.co.uk/sangoma-a101-pri-isdn-card-p-132.html?gclid=CI32vJz22I0CFQXklAodIgjHaA I would be grateful to hear if this is the right choice of card. Usage reports would be helpful. Regards Rory -- Rory Campbell-Lange

Hi Guys, Running the following with a Sangoma A101D PRI card: *Asterisk 1.4.21.2* *LibPRI version: 1.4.10* No inbound or outbound calls can be made. In fact Asterisk CLI doesn't show any activity. Problem goes away on restart of the system or maybe asterisk. I see post about upgrading Libpri to 1.4.10.2 and then I see posts that even that didn't work. Anyone can weigh in this please?

Hi all, I am wondering what people are doing for security when registering IP phone's remotely if you do not have the equipment to do a VPN tunnel at the remote site. The phone I would be working with mainly is the Polycom lineup. Thanks, Connor Spiess -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am using R on a Windows XP professional platform. The following code is part of a bigger one CODE press=function(y,x){ library(qpcR) models.press=numeric(0) cat("\n") dep=y print(dep) indep=log(x) print(indep) yfit=dep-PRESS(lm(dep~indep))[[2]] cat("\n yfit\n") print(yfit) yfit.orig=yfit presid=y-yfit.orig press=sum(presid^2)

Hello All, I have a setup of ABE on rPath linux,Sangoma A101D, and a T-1 line (Not PRI) which is all happily coexisting and all lights are green. The T-1 comes in from the world into a "Shark Box" which splits the T into 384K data and 6 channels voice. The data side is working great. The voice side, not so great. It was originally broken out to 6 pots line and Verizon came back

> I have a * box with 2 nics in the following setup: > > Internet > | > 192.168.5.253 (firewall) > | > 192.168.5.xxx network (gw 192.168.5.253) > | > 192.168.5.10 (* nic 1) > 192.168.6.10 (* nic 2) > | > 192.168.6.xxx network > > The netmask for both networks is 255.255.255.0 > > The 192.168.6.xxx networks has a 48 port switch solely for the use

Sorry to ask such a basic question: I have a * box with 2 nics in the following setup: Internet | 192.168.5.253 (firewall) | 192.168.5.xxx network (gw 192.168.5.253) | 192.168.5.10 (* nic 1) 192.168.6.10 (* nic 2) | 192.168.6.xxx network The netmask for both networks is 255.255.255.0 The 192.168.6.xxx networks has a 48 port switch solely for the use of cicso 7940 phones, the 192.168.5.xxx is

Dear R-helpers, does anyone know of some package/function that can build a network from the functions that are implemented in a package, i.e. visualize the cross-references from one function to another in the same or some dependent package? An example would be a function like 'nls' on top of the hierarchy and then a network of nodes from the functions that are called within 'nls'

Ich werde au?er Haus sein von 18-06-99 Bis 28-06-99. Ich werde Ihre Nachrichten nach meiner R?ckkehr beantworten. Bitte wenden Sie sich in dringenden F?llen an: Frank R?seler, Tel: MON 3855 Uwe Spiess, Tel: MON 3743 Guido Zimmermann, Tel: MON 3962 Ferdinand Gro?, Tel: MON 2368 lieber Gru? Axel Hellmig

Dear R-helpers, I´m in the context of writing a general function for error propagation in R. There are somehow a few questions I would like to ask (discuss), as my statistical knowledge is somewhat restricted. Below is the function I wrote, the questions are marked. Many thanks in advance. propagate <- function(expr, varList, type = c("stat", "raw"), cov = TRUE) {

Hi everyone, I'm trying to get a Sangoma A101D-X card talking to a Sasktel PRI (Megalink) circuit and having some trouble getting it to handshake. Thanks for any help or suggestions to diagnose this problem. The problem is that Asterisk has trouble bringing up the link. I see the following lines every couple of minutes: == Primary D-Channel on span 1 up == Primary D-Channel on

Hello List, We purchased a TE120P card from Digium and it works great. The only problem is that we are still experiencing echo on some calls. I've tried various echo cancellers (right now we are using OSLEC) and still no luck. My question has anyone gone from the TE120P to a Sangoma A101D-X Single Port T1/E1/J1 w/ echo cancellation? Have you noticed a difference? Also I called

Hi all, i have been using asterisk for a few years but i am about to do my first t1 setup. After terrible quality issues between two business locations, we have decided to purchase a point to point t1 from the local phone co. The internet is too crappy, too much lag, queing and jitter. Most calls were dropped. I was about to order two cisco routers with csu cards and remembered our wonderful

Hi, I'm using Cicso - Linksys SPA3102 to connect to asterisk. I have followed the official user manual and the blog post here http://www.skelleton.net/2012/08/02/linksys-spa-3102/ When I call an extension say 225 from the analog phone, I can get the IVR I have setup in my dialplan. But when I Call the analog phone extension using a sip phone I get the following error message: Unable to

I just got a Cisco 7960. Of course, it has the Call Manager image on it, and I don't have access to a SIP image. So, I read that Asterisk has 'some' support for the skinny protocol. I played around with it, and I actually got the Cisco phone to display line 1 on it. I can call another phone (SIP) from the cisco, but I cannot hear on the other phone. The cisco, however, can hear the