similar to: call waiting, * and FXO

Hi R-users, I am using arima to fit a time series. Now I would like to include an intervention component "It (0 before intervention, 1 after)" using different types of impacts, that is, not only trying the simple abrupt permanent impact (yt = w It ) with the xreg option but also trying with a gradual permanent impact (yt= d * yt-1 + w * It ), following the filosophy of Box and Tiao

Hi I'm trying to measure the effect of a policy intervention (Box and Tiao, 1975). This query has to do with the coding of the model rather than with the particulars of my dataset, so I'm not providing the actual dataset (or a simulated one) in this case, apart from some general description. The time series are of length n=34 (annual observations between 1977 and 2010). The policy

Hi - I am doing some monte carlo simulations comparing bayesian (using Plummer's jags) and maximum likelihood (using lmer from package lme4 by Bates et al). I would like to know if there is a way I can flag nonconvergence and exceptions. Currently the simulations just stop and the output reads things like: Error in optim(.Call("lmer_coef", x, 2, PACKAGE = "Matrix"), fn,

I have been looking for the answer to this question for a while. Google-ing and reading the archives of Asterisk-Users has not enlightened me. It seems that this question has been asked many times, and many times it has gone unanswered. I have call waiting and three way calling on my PSTN line from Verizon (the local telco). This is connected to a FXO port on a TDM400P. I also have

I had a similar issue with sending a flash to the PSTN for call waiting. I found that my dial plan in the extensions.conf file was not allowing me to dial *xx. Once I corrected my dial plan I was able to dial *0, *69, *78, *79, etc. Training the wife on how to actually use it was an entirely different issue. I have not tried to enable 3-way calling via the PSTN. "Jeff Otterson"

if you have anyone questions about your service you can contact us at the support 978-418-7300 James Jones Broadvoice Technical Support ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Ben Wern Sent: Sat 8/28/2004 4:34 PM To: Asterisk Users Subject: [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting Can anyone who is using Asterisk

We asked Henning Schulzrinne (the author of RTP and currently the RTP/AVG maintainer) about getting a static RTP payload type code assigned for Speex, since we meet the criteria in rfc1890. He said no. Here is his answer: On Fri, 22 Nov 2002, Henning Schulzrinne wrote: > Sorry, this is not going to happen, regardless of the codec or its > merit. See the new draft which will replace RFC

Anyone interested in this feature? I have a version 0.1 patch, which is currently against 1.2.25-bristuffed, but which should port trivially to almost any version. I am away until Tuesday 8th April, but if there is enough interest, I will open a "new-feature" ticket and upload the patch to the bugtracker so that more capable programmers can laugh at it ;-) It should work reasonably on

Not sure if it's alright to talk about this here??? compiled the STUN server from Vovida on RedHat 7.3. Looks simple to configure. It isn't starting...it tries to for a long time and then just craps out. Here is my config:/etc/sysconfig/stund #!/bin/echo Not to execute. # Path to stund STUND=/usr/sbin/stund # Set the required args for STUND STUNDPRIMARYHOSTNAME=208.x.x.x # The hostname

What is the best free stun server out there? The one that I have looked at from vovida requires two NICs. Is this neccessary?

Folks, I love Asterisk, have been using it for a while now. I'd like to know if anyone has some good comparison points on Asterisk vs. Vocal (Vovida) vs. GNU Bayonne. I know only a little about the later two. Also, one drawback I've hard about Asterisk (not for me, but for general consumption/deployment) is easy of configuration -- people like GUIs. They want point-n-click. I'm a

One of my clients today asked me about TLS support for encryption of SIP payloads, and I didn't have an adequate answer as to why it wasn't supported or even discussed. Some archive searching finds scant mention of this in reference to Asterisk. Of course, encrypting the SIP payload is only 1/2 the problem; the payload itself is the next problem. I understand that IAX solves these

Hi all, I want to deploy a SIP Proxy but I just don't know which one to choose. Researching in the Internet I found the following ones: * SIP Express Router <http://www.voip-info.org/wiki/view/SIP+Express+Router>: SER is used by many SIP providers standalone or in conjunction with Asterisk * Vovida.org <http://www.voip-info.org/wiki/view/Vovida.org> * sipX

Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the "very inexpensive" Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to

Hey there I have two vectors: y<- c(0.4, 0.0, 0.2, -0.2, -0.6, 0.2, 0.0, 0.0, 0.4, 0.4, 0.2) In the vector y, I want to access (in the order given) all of the values in between each of the specific values of given. I understand subsetting with y[i], but how do I get to ssomewhere in between -0.6 and 0.2? Thanks Eric Jennings matheric at myuw.net

any way the you can get * to send a NOTIFY SIP message to all SIP phones? to have the SIP sets recheck thier configs etc?? Like this? NOTIFY sip:sip@192.168.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP ipaddress:5060;branch=1 Via: SIP/2.0/UDP ipaddress From: <sip:webadim@192.168.0.1> To: <sip:sip@192.168.0.3> Event: check-sync Date: Mon, 10 Jul 2000 16:28:53 -0700 Call-ID: test@192.168.0.1

Hi, I run a local exchange server and would like asterisk to send voicemail notification messages via exchange. I have had a look at the voicemail.conf file, but I can't see how I would go about configuring it to use an account set up on exchange ? The exchange account would have both POP3 and IMAP access, so how can I tell Asterisk to use the exchange account rather then sendmail ?

Hello! Some time back, I added the Speex protocol to my version of VOCAL (www.vovida.org, VOIP tool). Recently, I also added PESQ (automated voice quality testing algorithm) to my tool and have been running some tests on a clean network. The source file is a woman reading some phrases meant to test various aspects of codecs... Speex has a respectable result of 3.67 Some other codecs I've

Hi friends ! I want to add stun functionality in asterisk. can anybody give me some hint that how can i start that. thanks in advance Deepak Dhiman

Hi, I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear