similar to: Zap - 256 format frames

This just started happening today. I've got 1 queue and 6 agents. All logged in. I tell the service people to ignore my call if they see my caller id. I call the queue and watch as asterisk bounces me around the phones. Our agent ring time is 5 second timeout and a 5 second wait time before trying next agent. I get the same message in console for each agent attempt: -- Executing

I'm not a PRI expert and therefore don't know what this debug stuff means for PRI, so if anyone can help me here... I'm running the latest libpri and zaptel from CVS. Keep in mind that everything works fine when using the STABLE libpri and zaptel. I am NOT running CVS asterisk. I am running 1.0.6. asterisk*CLI> pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 T203

The setup: Inc SIP Call -> Asterisk 1 -- IAX --> Asterisk 2 -- SIP --> phone (3044) Asterisk 1 shows the following: (1.0.3) -- Executing Goto("SIP/XX.XX.XX.XX-0819f590", "cytel-internal|3044|1") in new stack -- Goto (cytel-internal,3044,1) -- Executing Dial("SIP/XX.XX.XX.XX-0819f590",

Why are there so many attempts to native bridge? The call is actually up and working by attempt #1 so what is it doing on all those other attempts? We are only allowing G729 so it can't be codec negotiation. -Matthew -- Executing Goto("SIP/3013-3dfa", "cytel-outgoing|917034439032|1") in new stack -- Goto (cytel-outgoing,917034439032,1) -- Executing

Hey gang, There must be any easy solution for this but my mind is frazzled on compiling 2.4 with RTC as module. Bleh. Currently extension 9000 is our VoicemailMain(@company) line. Some employee's are complaining that the old system was better because you didn't have to enter your mailbox number and that instead the old system took you right to it. I figured there was something similar

Well, it is clear enough that the problem is in interpreting the documentation. However, when you claim you tested something, and found it inconsistent with tables, it would be advisable to back it up with examples! The description in the help files and in the sources is admittedly confusing. The original paper has this, rather more clear, description in the abstract: "We consider the

hi, im looking at deploying asterisk in a small corporate enviroment which will have approx. 1200 IP Phones and an average of about 100 to 200 calls at any given time. The calls will be sent out SIP to my Cisco Gateway. Im running Asterisk on a Dell Dual P3 1.2ghz running Fedora. Is there a calculator or a spreadsheet which could tell me about how many calls I will be able to make through *

Hello R help members. I came across today with an article on Barnard's exact test (http://www.cytel.com/Papers/twobinomials.pdf), that is supposed to give a more powerful fisher.test - Because it doesn't assume that we know the row and column totals are in advance. Any pointers to such a function ? Thanks, Tal -- ---------------------------------------------- My contact information:

Dear R folks, We completed a matched case-control study that was analyzed using conditional logistic regression. Because of the small sample size we need to calculate exact confidence intervals. The quick solution is to purchase LogExact by Cytel. However, we'd like to do this in R. Anyone have experience with this? Many thanks, Tomas ____________________________________ Tomas Aragon, MD,

Hi all. I have a strange problem, I've got a AS5350 hooked up to a telco using two trunked E1's The 5350 should only act as a GW to a sipproxyserver. THe thing is it seems to be only oneway audio? There are no firewall at all, and the audio still only get one-way When I call from pstn --> as5350 --> sip-sip-phone I can here the sip-phone ,, but the sipphone cannot her the

Another example: Jerry Friedman's MART is available in R from Salford for the same price as the stand-alone TreeNet, even though they don't advertise it on their web site. Andy > -----Original Message----- > From: rossini at blindglobe.net [mailto:rossini at blindglobe.net] > Sent: Sunday, August 17, 2003 9:50 PM > To: rhelp > Cc: pralay at cytel.com > Subject: [R]

At JSM, I spent a bit of time with old friends at the Cytel booth (makers of StatXact/LogXact). They were wondering whether it was both feasible and of interest to create a package of the StatXact compute engine for R (to be commercially licensed, not for free!), similar to what they've done for SAS. As far as I know, it's feasible, (this is not the first commercial external package,

We cannot help you understand what you are doing if you do not show us what you are doing. Here are some discussions about how to communicate questions about R [1][2][3]. [1] http://stackoverflow.com/questions/5963269/how-to-make-a-great-r-reproducible-example [2] http://adv-r.had.co.nz/Reproducibility.html [3] https://cran.r-project.org/web/packages/reprex/index.html -- Sent from my phone.

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

I am having some problems with g729 with SIP and ZAP channels. 1) I have two g729 licences. Very frequetnly (I don?t know what triggers the error) I get the following warnings and error when I try to place a call via SIP to my X100P. The only way to get out of this is through a restart of *. When the error ocurrs there are no other calls in place. Any ideas? Error Opening channel:2 not

Does anyone on-list use the Zultys 4x4 or 4x5 ip phones? I'd like to hear some opinion before I buy a few. I'm especially interested in the PSTN interface on the 4x5. Does it relay to * for VM when an incomming call is not answered by the phone? Thanks, Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist

Dear all, I wanted to compare Bonferroni vs TukeyHSD correction over a range of groups and group sizes, and wanted to use the function qtukey. In the help documentation it says qtukey(p, nmeans, df, nranges = 1, lower.tail = TRUE, log.p = FALSE) Arguments q vector of quantiles. p vector of probabilities. nmeans sample size for range (same for each group). df degrees of freedom for s (see

Here is the full "transaction" -- outgoing agentcall, to agent '1001', on 'Local/3044@local-4fee,1' -- Called Agent/1001 -- Executing Macro("Local/3044@local-4fee,2", "sipline|3044") in new stack -- Executing Dial("Local/3044@local-4fee,2", "SIP/3044|20|t") in new stack -- Called 3044 -- SIP/3044-ea92 is

I am getting this on the console once people call in -- outgoing agentcall, to agent '1001', on 'Local/3044@local-fd6d,1' -- Called Agent/1001 -- Executing Macro("Local/3044@local-fd6d,2", "sipline|3044") in new stack -- Executing Dial("Local/3044@local-fd6d,2", "SIP/3044|20|t") in new stack -- Called 3044 --

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Conditional logistic regression and small sample sizes: what to do? Dear R-Gurus, [I run R 2.8.0 on WinXP. I have no formal training in statistics.] Please feel free to skip the 'Blah blah part' for questions below. - ---Blah blah part-------- Why I use clogit (package 'survival')?: I am looking for risk factors for cases during a