search for: faried

Can anyone suggest where I might begin looking for an answer please? I have just installed a TDM400P (2x FXS and 1x FXO modules installed) The first problem is that it does not seem to be able to detect if the remote party has hung up when a call comes through on the FXO. For example, if someone calls in, and then hangs up at any time after it starts ringing, Asterisk carries on as though the

Signed-off-by: Faris Alsalama <farisbenbrahem at gmail.com> --- drivers/gpu/drm/nouveau/nouveau_backlight.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/gpu/drm/nouveau/nouveau_backlight.c b/drivers/gpu/drm/nouveau/nouveau_backlight.c index 89eb460..dd1cc9b 100644 --- a/drivers/gpu/drm/nouveau/nouveau_backlight.c +++ b/drivers/gpu/drm/nouveau/nouveau_backlight.c @@ -232,6

I have a really baffling problem. A couple of months ago I purchased a pair of GrandStream GSX-2000 phones for use with Asterisk. At first all was well. But recently I've noticed terrible sound quality problems. Basically the sound will "glitch" or stutter randomly from time to time. Now, what is interesting is that this happens even with the phone totally disconnected from any

I've somehow managed to battle may way through hinting issues with type=peer type=friend and various other oddities and now have presence working correctly on my Polycom 600 and Grandstream GXP-2000 phones. However, on the Polycom I have to press the Buddies softkey in order to see if an extension with a hints priority is in use or not. I've spent all day going through google and my

(I hope this isn't html - Thunderbird is so annoying) I'm new to using hints/subscriptions on * so please be patient with me. I have two * systems in different geographic locations, connected via IAX Location1 has a Polycom 600 and a GXP-2000 phone Location 2 has a single GXP-2000. With the latest GS firmware, at Location1 I've managed to get an LED to light up on the GS phone

I've having a big problem after having upgraded to 1.2.4 and chan_capi-cm 0.6.4 When making outgoing calls I don't seem to have any control over the CLI that is presented to the called party -- it can be any one of the MSNs allocated to the line, allocated on what seems to be a random basis. This is on a BT Business Highway line (which is essentially an ISDN2e line with two built-in

Can anybody point me in the right direction please? I'm having some issues getting iaxmodem and hylafax to talk to each other. I have no doubt that someone has had this type of issue before but I can't find anything useful in the archives or on Google. Under RH9, with chan_capi 7.1, Asterisk 1.2.24, with an AVI Fritz ISDN2e BRI card all working perfectly... I've downloaded and

Hello, I was planning on adding multiple physical functions using the built in SR-IOV features offered by libvirt to a create a bigger pool of virtual functions using the <forward mode='hostdev' managed='yes'> heading (I want to be able to use more virtual machines then there are virtual functions available on an ethernet port). As soon as I try to define multiple

Can anyone help point me in the right direction please? I'm based in the UK and I want to start using a Premium Rate number with Asterisk - I think the equivalent in the US would be a "900 number". Effectively the caller pays much more to call such a number than a normal national or local call. The problem with these is that I don't want Asterisk to actually signal to the

I'm having some mysterious problems installing 1.6.2.10 on Centos 4.8 (totally up to date). I can't see anything on Google or the list regarding this issue, which I find a bit odd considering 1.6.2.10 was released a few days ago. I'm therefore assuming there's something weird about my setup, even though there shouldn't be! I had no problems with 1.6.2.7 or any other release.

Been using mpg123 for moh for the last two years or so. However, when I have * config errors, often times get a endless stream of console messages and need to kill the two mpg123 processes. Is there an alternative to mpg123 that eliminates that issue? I see references in musiconhold.conf relative to madplay, native file format, asterisk-addons, etc. Not sure why the asterisk-addon approach

Hi, my name is Paula. Im trying to learn how to build an R package in Windows XP using a simple example which includes a C source code. The foo.c file is located in mypkg/src/foo.c. I used the command Rcmd build --force --binary [pkgpath]. The problem is when I install the zip file in R the function foo doesn?t exist. The result a recieved during the building process is copied below. I really

Good Day I am the Head of Finance at Sullah Finance FZE here in Abu Dhabi, UAE. I am looking at investing over US$M900, 000,000.00 that has been placed for investments by a private owner. If you are in need of funds to expand existing businesses or to start up a new project, then look no further as we would be more than delighted to work with you. We are driven by a project's credibility

Changing the default behavior of regmatches would break its use with gregexpr, where the number of matches per input element faries, so a zero-length character vector makes more sense than NA_character_. > x <- c("John Doe", "e e cummings", "Juan de la Madrid") > m <- gregexpr("[A-Z]", x) > regmatches(x,m) [[1]] [1] "J"

I installed the game like it is described in the WineHQ application database. The installation worked without problems but running the game produced the following error message: Cannot create DirectSound object (-2147467259) I got around this by switching off OSS sound and using ALSA instead, but know I have the same problem that Erich Hoover mentioned in the AppDB. The music starts to play but

I installed this card, everything work, i can make call and receive call with no echo and great sound quality, but after between 5 to 50 secs the call disconnect by itself, in the log i don't see nothing revelant. I don't share any IRQ, zttest show me values between 99.98 and 100. The only thing i see, the pstn line is not realy a true pstn line, it's plugged in an Arris Telephony

Hello everybody, I'm a new user of * and I just bought a Sipura SPA-3000 to make a home voip installation. I actually have a problem when a PSTN user calls and hangs up. The disconnect tone is not detected by the SPA, the the call continues and, for example, leaves an empty message on the voicemail before hanging up (because * hangs up). How could resolve this problem?. I set, Detect

A client asked me to come with a system that will pass certification with PPP in the UK. Google is not being helpful :( It has something to do with recording calls in case the PPP requests a copy. Supposedly their rules were relaxed on August 1st if that helps. Any clues (links?) will be appreciated. -- Thanks in advance,

in wine 0418, the strupr and strlwr defines were removed from tchar.h does anyone know why? ------------------------------------------------------------ Get your FREE web-based e-mail and newsgroup access at: http://MailAndNews.com Create a new mailbox, or access your existing IMAP4 or POP3 mailbox from anywhere with just a web browser.

https://bugzilla.samba.org/show_bug.cgi?id=1599 Summary: copy-unsafe-links cann't take affect Product: rsync Version: 2.6.2 Platform: x86 OS/Version: Linux Status: NEW Severity: normal Priority: P3 Component: core AssignedTo: wayned@samba.org ReportedBy: faris.xiao@haoxi.com