similar to: X100p Problem, randomly hungup pstn line

Hi there, I have a cablemodem, ARRIS brand, model tm502G. It has two FXS ports. I was wondering if anyone has details about the correct signalling of these FXS ports when connected to original X100p. Tests: fxsks on the zapata.conf and zaptel.conf files. From my cellphone I call the ARRIS, it starts ringing but the zap channel sees no call coming in. fxsls on the zapata.conf and zaptel.conf

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 have a P3-800 with two IDE drives in a software RAID1 configuration. Each drive is on a separate IDE channel. Now anytime there is HD activity, I hear "beeps" and "cutting out" on a call using the X100P card. I ran the zttest program, and discovered HD activity would drop the accuracy down to between 2% and 50%. However I noticed if I disabled one drive in the RAID1

Hi All After lots of try I was successfull in connecting to PSTN to make and recevice calls , I used AMP for this purpose , now I wanted to try out this Asterisk server answers the call , ask for the extensions and then after the extension entered the call is forwarded /transfered to the extension no , I use Asterisk 1.2.4, configured using AMP , on RHEL3 I did some configuration for my

When plotting Binary Trees (ctree) from the party package, is there a way to adjust the font sizes of the leaves? require(party) irisct <- ctree(Species ~ ., data = iris) plot(irisct) I want to adjust the font sizes for "Node 2", "Node 5", etc. I'd also like to be able to adjust the font sizes for the x-axis and y-axis labels of the histograms. Thanks,

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

On 2/3/2017 3:35 AM, Steve Clark wrote: > > What kind of cable modem/gateway do you have? Just wondering because > my 12 year old Toshiba finally > crapped out and Spectrum gave me a new one. Its and ARRIS TG1682G and > it only gives me a private IP not > like the old one which gave me the public IP so I can't ssh to home > from work anymore, so I am wondering > how

Hi there, I'm currently trying to set up SAMBA on SuSE 6.3 but I just can't get it to let my Windows 98 PC in. I created a share and I can access it using smbclient but when I try to click on the System in the Network Neighbourhood or by typing \\Arris\share in Start -> Run all I get is "\\Arris is not available the computer or sharename cannot be found ..." or "The

Hi This may be off topic because it involve cable. I am testing with Arris cable modem / MTA I have 2 models, one older and one newer. With older one, everything works fine With the new one, I can register, make a call and I hear the other person but he can't hear me The config is the same with both units except for the username of course, Anybody ever worked with these units ? Our

In the panic of replacing our firewall(s) earlier in the week, we ended up moving our original shorewall 1.4 config onto a machine with 2.0.10 already installed, overwriting all the 2.0.10 config files. Most things seem to work fine, except for our masq entries. I''ve examined the default 2.0.10 files compared with our 1.4 files, and can''t spot the problem. What am I missing?

> -----Original Message----- > From: CentOS [mailto:centos-bounces at centos.org] On Behalf Of Steve Clark > Sent: Friday, February 3, 2017 6:36 AM > To: CentOS mailing list > Subject: Re: [CentOS] Spotty internet connection > > On 02/02/2017 10:12 PM, TE Dukes wrote: > > > >> -----Original Message----- > >> From: CentOS [mailto:centos-bounces at

Hi all, Sorry for the repost. If I should be asking somewhere else please tell me where. I woke up a few days ago to find this on one of my machines. Below is hopefully the revelent output from dmesg and /var/log/messages. I have had this b4 but this is the first time I have been able to get any useful information from it. The machine usually locks and all I see is the assertion failure on

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

Hi all, With a great help from Richard Alexander (thanks Richard!) I have now a functional CallerID on my X100P. This is what I have done: - update to the latest CVS (as today at 5:00pm GMT) - modify the callerid.c file in the asterisk source like that. original : /* MDMF */ /* Go through each element and process */

Hi, I have the following basic configuration: - one X100P card - two IP Phones (connected to an ATA186) When an incomming call on X100P, both phones connected to an ATA186 rings (configured like that in extensions.conf). When a call is established between the two ATA phones and a call occurs from the PSTN line, the callerid is correctly displayed on both phones (as internal callwaiting

I have a different Unix userid for each project I'm working on, so I need to provide a different userid/password combination for each drive I map to my NT 4.0 SP3 system. But, whenever I try to supply a different userid in the "Connect As" box and hit "OK", I get the message: The credentials supplied conflict with an existing set of credentials Something I read in the

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

Yo all, I have a problem with a Dutch (KPN) PSTN-line, connected to an X100P. The call wil sound OK at first, but after 10-20 minutes, the audio will start to crackle. Soon after that, this crackle turns into a continuous noise and the parties won't be able to hear eachother anymore. It also sometimes happens that the party on the TDM400P hears a very loud, short-delay echo of themselves,