similar to: Linking Asterisk Boxes with Sip

Martin Maechler <maechler at stat.math.ethz.ch> writes: > Well, Dirk has said to have given his last reply on this thread. > I (as a member of R-core) am glad about people like Dirk who > take some of our load and helpfully answer such > questions/reports on R-devel. I am glad too. Thank you. My ultimate goal is to ensure that R works as well on FreeBSD as it does elsewhere.

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant

Hi, We are running a samba fileserver, access controlled using posix acl (right 770, with users/groups on the filesystem level. Therefore samba shares look like this: [share] path = /srv/academic read only = no writable = yes create mask = 0770 directory mask = 0770 Now certain users complain that they cannot access certain folders, but looking at the folders from the linux fileystem, their

http://grandstream.com/BETATEST/HT488_496_386/

Hola, compañeros. Pido ayuda con algo que sé que tiene que ser simple, pero la presión de tener que sacarmelo de encima me simplifica a mí demasiado y no me doy cuenta. Tengo una matriz de datos en la que tengo características tipo factor, necesito trasponer esa información a una matriz de datos binarios, en función de algunas combinaciones de esas variables. Unas combinaciones tienen que pasar a

Anyone? -------- Original Message -------- Subject: Functional test problem Date: Wed, 14 Jun 2006 19:49:03 +0100 From: Chris T <ctmailinglists@gmail.com> Reply-To: ctmailinglists@gmail.com To: rails@lists.rubyonrails.org Embarrassingly late (apps already well advanced), have decided top get to grips with testing and am having a week hopefully covering everything I''ve

Dear R-users, I am comparing differences in variance, skew, and kurtosis between two groups. For variance the comparison is easy: just var.test(group1, group2) I am using agostino.test() for skew, and anscombe.test() for kurtosis. However, I can't find an equivalent of the F.test or Mood.test for comparing kurtosis or skewness between two samples. Would the test just be a 1 df test on

Hi, I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now having a problem with Originate and chan_local. I'm using the following Manager API action to originate a call: Action: originate Priority: 1 Context: trunk Callerid: 100 Channel: Local/100 at callback/n Exten: 123456789 Variable: USERFIELD=127.0.0.1|USEREXT=123456789 WaitTime: 30 This is intended to first call

Hi, Try this: lines1<- readLines(textConnection("gene1 or1|1234 or3|56 or4|793 gene4 or2|347 gene5 or3|23 or7|123456789")) lines2<-readLines(textConnection(">or1|1234 ATCGGATTCAGG >or2|347 GAACCTATCGGGGGGGGAATTTATATATTTTA >or3|56 ATCGGAGATATAACCAATC >or3|23 AAAATTAACAAGAGAATAGACAAAAAAA >or4|793 ATCTCTCTCCTCTCTCTCTAAAAA >or7|123456789

Hi, what's the current status of the memory-ordered operations described in https://docs.google.com/Doc?docid=0AYWBeVVqyP7dZGRiNG1oeHpfMjJkejVnOThkZA&hl=en.&pli=1 i.e. the ones for "load acquire", "store release" etc. for C++0x atomics, not the older ones for the __sync intrinsics? The specification looks good - is it just waiting to be implemented? Al --

Hello all, I would like to use Asterisk to add/modify SIP headers in the INVITE message, to include Privacy information, if the INVITE includes a *67 prefix (or another predefined prefix). That's an example of the INVITE I get: /INVITE sip:*6700112233445 at 192.168.1.100 SIP/2.0 From: "123456789"<sip:*123456789*@192.168.1.100>;tag=333333333 To: <sip:*6700112233445 at

I am looking to setup paging using the auto answer feature on the Grandstream GXP2000. I am thinking I will follow the method as described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page I will setup the 4th account on the phone to auto answer. Does anyone else have a method that works better? I also looked at the allpage AGI written on Voip-Info. But it seems

Hi, I've migrated from an old samba installation (Samba as PDC) that used TDB backend for password. I've setup a box with ubuntu and samba 3 + ldap and I imported the old users. Old users works fine. I have problems with new users and machines. Old users works but they don't show up with smbldap-usershow command and I've problem in changing their passwords. If I check the ldap

Hello, I have an internal extension, e.g. 1005 which is being called from an external/public number like 123456789. Now when it comes to the spoken voicemail information it says something like "number 1000 not available", however it should say "number 123456789 not available". How can I configure this? I already googled and I guess this is really easy, but I just couldn't

hi! anyone has a idea how to send caller id (incoming number) to shell script like: exten => s,1,wait,2 exten => s,2,System(/bin/script callerid-number) exten =>s,3,answer i need to run script in shell with command line like "/bin/script -a 123456789 " this 123456789 must be callerid (incoming call) number. tnx, Thomas

How can I make it more readable? Name/username 601/601 123456789/123456789 voipbuster/abcd 601 = hotline 123456789 = Peter Pan only voipbuster/abcd is easy read/understandable! bye Ronald Wiplinger

For compiling the xen kernel we use -nostdinc which means we don''t want to rely on any external headers. That also counts vor stdarg.h. Use the external header when we compile the tools. Adapt libelf-loader.c which is compiled in the kernel and with the tools. Signed-off-by: Christoph Egger <Christoph.Egger@amd.com> -- ---to satisfy European Law for business letters: Advanced

On 1 July 2016 at 17:41, Joshua Colp <jcolp at digium.com> wrote: > > >> exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) >> same => n,Dial(PJSIP/phone123, 30) >> > > Your exten line has no priority, is that how it is in your dialplan? > Actually no, I stole that line from an earlier email to this list. Mine has a priority.

On 2019-12-16 05:34, Roger Price wrote: > On Sun, 15 Dec 2019, Phil Stracchino wrote: > >> I switched a while back to a Cyberpower PR3000LCDRTXL2U UPS with an >> external battery chassis. We've lost power twice since I installed it, >> and both times, the UPS has killed power to everything while still >> indicating 80% capacity before NUT has even initiated

I see that Asterisk@home and FreePBX are going along similar lines with web based management interfaces.. My Asterisk box has one analog phone, one analog line, 3 SIP phones, IAX inbound numbers, an IAX outbound trunk, IVR menus and voicemail boxes in different contexts for each of the inbound numbers.. Soon I will be adding one or more IAXy devices.. Would either Asterisk@home's or