similar to: Effect.Appear/Hide and numbered lists

For anybody interested in tracking when this is resolved for mainline samba4 there is now a bugzilla record here: https://bugzilla.samba.org/show_bug.cgi?id=10980 On 2 December 2014 at 11:52, Kirin van der Veer < kirin.vanderveer at planetinnovation.com.au> wrote: > Is there a corresponding bug number for the Samba bugzilla so that I can > track when this is fixed for vanilla samba?

Hello ! In order to chase after a problem I implemented the following dialplan to have an answertime of exactly one minute: exten => xxxxxxxxxxx,1,NoOp(Test wait) exten => xxxxxxxxxxx,n,Answer exten => xxxxxxxxxxx,n,NoOp(Current timestamp: ${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}) exten => xxxxxxxxxxx,n,Wait(60) exten => xxxxxxxxxxx,n,NoOp(Current timestamp:

Return-path: <xxxxxx-xxxxxxxx-xxxxxxxxx-xxxxxxx-xxxxxxx-xxx at xxxxxx.xxxxxxxxx.xx.xxx> Envelope-to: xxxxx at xxxxxxxxx Delivery-date: xxx, xx xxx xxxx xx:xx:xx +xxxx Received: xxxx [xxx.x.x.x] (xxxx=xxxxxxxxx) xx xxxxxxxxx.xxxxxxxxxxxx.xx xxxx xxxxx (xxxx x.xx) (xxxxxxxx-xxxx <xxxxxx-xxxxxxxx-xxxxxxxxx-xxxxxxx-xxxxxxx-xxx at xxxxxx.xxxxxxxxx.xx.xxx>) xx xxxxxx-xxxxxx-xx xxx xxxxx

Hello all, I am working at the FORTH institute in Crete and it's been a long now that I am trying to reproduce the results of the paper : "Gene expression profiling predits clinical outcome of breast cancer", by Van't Veer et al. It has been published in NATURE, vol 415, 31 January 2002. http://www.nature.com/nature/journal/v415/n6871/full/415530a.html I am facing some

mbox messages gets header corruption caused by an extra linefeed after Content-Length Users sees their mails in Sent mbox folder without the from and to fields, without attachments and with the date of 1/1/1970 Diego. --- Here is an anonymized header: >From xxxxxxxx at xxxxxx.xxxxxx.xxxxx.xx.xx Tue Jun 03 09:14:33 2008 Message-ID: <xxxxxxxx.xxxxxxx at xxxxxx.xxxxx.xx.xx> X-UID: 3913

Hi, I've got a problem with my asterisk set up which has been going on for a while (months). I'm currently running 1.2.7.1 on a gentoo box with the topology below: +-----+ PSTN ---------+ * +------------- Service Provider (wctdm400p) +-+-+-+ IAX | | | | FXS --+ +-- SIP (cisco 7940)

Hello colleagues, I have tried to use the package biglm. I want to specify a multivariate regression with a weight. I have imported a large dataset with the library(bigmemory). I load the library (biglm) and specified a regression with a weight. But I get everytime a error message like ?object not found? or ?`weights' must be a formula? or "error in eval(expr, envir, enclos)". I

Hi, I need to create something like this: <?xml version="1.0" encoding="UTF-8"?> <Container> <id>aQlfVHX+qPM</id> <lifetime>2009-09-19T08:14:55Z</lifetime> </Container> The response should contain the next headers: Content-Type=`application/vnd.3gpp+xml`

View Project Issue [1] Dear Gluster Users group, We are looking from feedbacks, advise and production experience running GlusterFS in cloud infrastructure environment. *--? Main aim ---* Using GlusterFS as a mechanism to ensure data application sourcecode replication in order to run a web based Python Framework

Hi, I have an ISDN phone connected to a hfc-s card. I use it to phone via an iax provider to foreign countries. Inside my country it works reliable, but to other country it happens very often that the other side hears ringing and before it can take the phone the line is dropped. What makes me wonder is that I hear no ringing at all. With asterisk -vvvvc I get this: Asterisk Ready. *CLI> ==

Hi, I am trying to dial a sip user via his IP:PORT Combination. i am using XYZ as target user which is registered. Asterisk server IP: 70.118.x.x calling user IP: 117.58.x.x called user IP: 117.58.x.x:5062 First I dialed my registered user in normal way like this, Dial(SIP/XYZ,30,rtT) and during conversation audio was OK in both ways. Then I dialed the registered user via

We regularly get questions about making pie charts, and frequently veer off into the philosophical and practical issues surrounding that form of data graphic: human perception of angles and so on. So while this isn't an R issue, I thought some of you might appreciate this brief discussion of the efficacy of pie charts: http://fakescience.tumblr.com/post/6042293374/understanding-pie-charts

-----Original Message----- From: asterisk-users-bounces at lists.digium.com on behalf of Steve Edwards Sent: Fri 7/9/2010 5:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] False answer() being sent by cellphone providers On Fri, 9 Jul 2010, Mike Ely wrote: > (off list) Continuing to veer off-topic... > Yes indeed we do. The telcos

Recently, I got the following error messages in CLI periodically. Jan 20 17:43:18 ERROR[8641]: chan_sip.c:11002 handle_request_subscribe: Got SUBSCRIBE for extension XXXXXXXXXXX@from-int from 192.168.0.123, but there is no hint for that extension I have no idea what the error message tell me. I am sure I haven't that account XXXXXXXXXXX in my database and there is no hint extensions in dial

I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1

Has anyone configured ast_fax (sending faxes via asterisk) with sendmail? The creation of rules to trap all numbers <mailto:XXXXXXXXXXX@domain.com> XXXXXXXXXXX@domain.com seems too complicated. Does anyone have setup details to share? (I don't want to switch MTA's). As a workaround, I could launch the app automatically from sendmail using an alias like: fax: "|

hallo all, could sombody please help me, i dont know why nativ bridging is not working when i choose the ilbc codec, with speex it is working,?? iaxcomm (ilbc) ---> asterisk --> ( asterisk2 --> sip grandstream (alaw) ) \-----------------native bridge------------------/ 1. if i use on iaxcomm as default speex, nativ bridging between iaxcomm and my sip phone is working 2.

Hello. I'm in need of some help to shed some light on a strange problem. The box is running Samba 4.1.7 on FreeBSD 9.2/amd64 and I suspect (though I'm not sure) all troubles started after the upgrade from 4.1.6. The setup is quite simple: no AD, no LDAP, only one server. A command as simple as "smbclient -U user //SERVER/SHARE", if fired up several times in a row on the

Folks, I've been trying to handle the problem where blocked callerids appear as coming from asterisk <asterisk> on the email notification, and the message envelope simply doesn't say anything (does not actually play the vm-unknown message). So, following the tip provided by several previous posters, I tried putting this in my extensions.conf (the xx's are my DID,

Below is a copy of my log, zapata.conf & extensions.conf that relate to the ZAP lines. Basically when we dial out it takes on 10-12 seconds before the ZAP line actaully picks up. I'm hoping to find out what the cause is for this as it's causing user grief with extremely long connect times, and I believe it may be causing issues of cross lines (an outgoing call gets mixed with an