search for: falsy

Hi everyone The following behavior (in R 3.6.1 and R-devel r77040) caught me by surprise today: truthy <- c(TRUE, FALSE) falsy <- c(FALSE, TRUE, FALSE) if (truthy) "check" #> Warning in if (truthy) "check": the condition has length > 1 and only the #> first element will be used #> [1] "check" if (falsy) "check" #> Warning in if (falsy) "check": the condit...

...t; \donttest{ > ## Set colors from colorspace package with a fallback > col <- try(colorspace::rainbow_hcl(5), silent = TRUE) %||% rainbow(5) > } > > The problem is that this makes R CMD check freak out > (http://www.r-project.org/nosvn/R.check/r-devel-linux-x86_64-debian-clang/falsy-00check.html) > if the colorspace package is not declared as a dependency. But the > point of the example is to define a meaningful behavior when > colorspace is missing.... so I don't really want to put this into > \dontrun{}... Why not declare colorspace as a "Suggests:&quot...

...an idea how I could fix this? \donttest{ ## Set colors from colorspace package with a fallback col <- try(colorspace::rainbow_hcl(5), silent = TRUE) %||% rainbow(5) } The problem is that this makes R CMD check freak out (http://www.r-project.org/nosvn/R.check/r-devel-linux-x86_64-debian-clang/falsy-00check.html) if the colorspace package is not declared as a dependency. But the point of the example is to define a meaningful behavior when colorspace is missing.... so I don't really want to put this into \dontrun{}... I vaguely remember that there was also \dontcheck{}, but it is not in ...

On Mon, Dec 8, 2014 at 3:32 PM, Duncan Murdoch <murdoch.duncan at gmail.com> wrote: [...] > Why not declare colorspace as a "Suggests:" kind of dependency? I guess that is a solution. :/ In another example in the 'disposables' package I have: \donttest{ pkg <- make_packages( foo1 = { f <- function() print("hello!") ; d <- 1:10 }, foo2 = { f

On Mon, Dec 8, 2014 at 3:43 PM, Duncan Murdoch <murdoch.duncan at gmail.com> wrote: [...] > In this case, I don't see a way to include that code but avoid warnings. So > you could ask for forgiveness when you submit it to CRAN (and perhaps they > would be forgiving), or you could use some other code in your example. Just to clarify, the code does not give any errors, or even

The Asterisk Development Team would like to announce the release of Asterisk 17.5.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 16.11.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.11.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 16.11.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.11.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 17.5.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

In my setup, which is FreeBSD, using pjsip 2.5.5 as sip backend I am observing a regression when testing the latest Release Candidate. Any calls get refused and the following error shown on console: [Nov 22 10:49:26] WARNING[101105]: res_rtp_asterisk.c:2400 int create_new_socket(const char *, int): Unable to allocate RTP socket: Protocol not supported [Nov 22 10:49:26] WARNING[101105]:

The Asterisk Development Team would like to announce the release of Asterisk 13.34.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.34.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 13.34.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.34.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Zitat von Guido Falsi <mad at madpilot.net>: > So, trying to bind authentication to originate calls to registrations is > conceptually wrong in the SIP world. Maybe you can do that but that's > not the way the protocols have been engineered to work. Hi Guido, thanks for your answer. Well, I decided to do that, since I have my Asterisk reachable from Internet just for my

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: > On Thursday 11 Jun 2015, Luca Bertoncello wrote: >> Now my problem is to check in my dialplan if the peer, that originate >> the call, is reachable, and if not, to give an error... >> >> Is there any function to know if the peer is reachable? > > The peer that *originated* the call *must* be

The Asterisk Development Team would like to announce the release of Asterisk 13.27.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.27.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

According to the instructions given at https://www.asterisksounds.org/de I converted and installed German prompts successfully and for numbers, I can successfully listen to a German female voice counting or telling the date/time. But unlikily, somehow the voicemail prompt is still English, although my general language settings are "de". I use pjsip.conf, not sip.conf. In

The Asterisk Development Team would like to announce the release of Asterisk 16.4.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.4.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Dear all i'm using an "old" Asterisk 1.6.2.9-2+squeeze12, and want to know if is possible to configure a scenario like this: 1) receive a call and put it on-hold in a queue (OK) 2) monitor the queue and trigger an outbound call to a remote number using AMI, setting the channel of the on-hold on a specific var named channel2Link (OK) 3) when the remote number answer, trigger an

On Tue, Nov 29, 2016, at 07:15 AM, Barry Flanagan wrote: > On 29 November 2016 at 10:56, Jonathan H <lardconcepts at gmail.com> wrote: > > > Any ideas why a VPS, cloned from another instance (DigitalOcean > > "droplets" if it matters), won't run on the new instance? > > > > Everything else is the same; region, memory, disk, hypervisor version etc.

The Asterisk Development Team would like to announce the release of Asterisk 14.6.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.6.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: