similar to: voicemail-hangup issue

Here is my system: POWER SUPPLY 450W ALLIED ATX450P4 R(41) MB NFORCE2 A7N8X DELUXE ASUS RTL(Standard) CPU AMD|2500/333 ATHLON XP BARTON R(Standard) DDRAM 256M|DDR333 PC-2700 -K %(Standard) HD 40GB|WD 7200RPM 8MB WD400JB%(70) VGA ASUS|V8170MAGICII/T 64M MX440SE(58) CD ROM 56X|AOPEN CD-956 RTL(22) Besides the above parts, i have 2 X100P FXO PCI cards, and 1 TDM40B I bought 2 phones from radio

I'm rsyncing from an linux box to a win2k box with a precompiled version of rsync for win2k. durring the procces he gives me a lotte of the following errors (with different filenames): rename realother/Maildir/cur/.1005882149.8001_0.nyx.wildape,S=3080:2,.001002 -> realother/Maildir/cur/1005882149.8001_0.nyx.wildape,S=3080:2, : No such file or directory I user the command: rsync -cru

I know the recommendation it to not run more than 2 T/E100P's in a system but what about X100P's.. Usually there are 5 PCI slots in a system, has anyone tried 5 x X100P's in a system? Later.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

The cheap ones on EBay won't work with the SC420 server. I have one and can't make any of the clones work. I do have one TDM40B card for analog stations that works well. The problem with the SC420 is that it won't let you set the interrupts yourself and you end up with interrupts being shared. =============================================================== Message: 26 Date:

Hi. Im trying to set intel_iommu=on on kernel parameters at grub but for some reason it doesnt work. I edit /etc/default/grub file and i add the parameter. then i run grub2-mkconfig -o /boot/efi/EFI/centos/grub.cfg and then i reboot. when i run virt-host-validate i get QEMU: Checking if IOMMU is enabled by kernel: WARN (IOMMU appears to be disabled in kernel. Add intel_iommu=on to kernel

I have a couple of X100P's in my system and while on calls they just randomly hang up for no reason. I have tried messing with the busydetect and callprogress setting them to yes and no same and still random hangups. Is there another setting I should be looking at? My zap config looks like. context = inbound-work include => extensions signalling

sorry to bother with this trivial issue, but i am loosing all my hair ;-) got 2 x100p's and * on a slakware box x-lite to x-lite works fine! i also have: #ztcfg -vvv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. and in extensions.conf i got: [locals] exten

We are using icecast as stream-relay for a true narrowcast solution we've developed. The idea behind the system is that each individual client gets it's own mountpoint. Basically we have an application that creates streams. Its written in c++ and uses libshout to handle connecting to icecast. Everything works fine, as long as we keep the number of mountpoints below 347. As soon as we

Hi folks, this is specifically directed to Australia Asterisk users...... We are having a roblem with x100p 's when dialing STD. Upon receipt of the approximately the 5th (out of the ten) PIP's asterisk will hang up.... Now I am wondering if others are suffering the same problem ?? Any ideas ?? (it might exist on other cards, but so far I have only noticed the problem on x100p's).

Just installed the new 4-port FXO card and moved two pstn lines from existing x100p cards to ports on the FXO card. All zapata.conf entries that were functional on the x100p's were copied to the new FXO channels (including callprogress=no). Observations thus far: 1. asterisk will spontanously decide a pstn call has arrived, and ring the sip phone designated in the dailplan. Verified

Ok here is one for you. I know we all do the this for 911: exten => _911,1,Dial(Zap/1/911) exten => _9911,1,Dial(Zap/1/911) And this probably is more then acceptable for most of us. However I have a system setup that uses SIP for most calls and 1 POTS line. We use a "least cost" routing that uses the POTS line for local calls AND SIP when appropiate. What I want to do is

I'm running the a very recent CVS version of asterisk on an RH9 machine. My problem is that my x100p takes about 10 seconds to detect a hangup. After that it takes about 10 more seconds for the the zaptel device to release the line. Here's an example of my console report: == Parsing '/var/spool/asterisk/voicemail/default/101/INBOX/msg0000.txt': == Parsing

Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO 4-port gateway? The archives tend to suggest the box is not very straight forward, and possibly lacks some basic pstn interaction features. Thinking about trying one in place of a pair of x100p's (functioning fine now). CallerId, etc, supported on this gateway? Rich

hello, while starting shorewall4.5.20 on debian7 I get the following from touch touch: cannot touch `/var/lock/subsys/shorewall'': No such file or directory The needed file can not be created because the directory subsys is not present! If I creat the folder manuallly touch does not complain when shorewall start! Thus it would be nice if durring start shorewall could check if the

Dear All ~ I have * setup & running ok (with two Wildcard X100P's to PSTN). I also have two analog phones connected into same through a SIPURA 2000. These work fine, except that when I call out through PSTN & try to send DTMF tones to (say) a remote PBX to dial an extension, the gain seems to go wild (high), and the DTMF tones are not recognized at the other end. I tried setting the

I have a friend I am building a phone system for, I would like to use * He owns a "Take Out" restaurant with 4 Phone lines and 4 Phones. I'm going to use Cisco 7960's for the phones; is there a better phone I should be using? I need to know if this is possible... On each phone program the appearances of 4 "Extensions" that are really the 4 phone lines? If Phone

Thought I would post this experience to the list so it's archived for posterity... My company is deploying Asterisk-based PBX's to all of our branch offices. Each office has 2 analog Voice lines and a fax line. We didn't want to go to the expense of using TDM400's in the servers (which run asterisk and Hylafax) so we opted for 2 X100P cards in each box. So far they have

Hello, I am using a simple script, which is based on prio. The point is, that it is not possible to use htb on wifi networks, so I thought that prio will work fine, but it does almost nothing. All I wanted was to make the important packets like icmp, games, VoIP,... to go first, and to slow the things like FTP data transfer, etc. When I use $TC -s qdisc show dev ${IFACE}, I see, that the

Hello all, I have a question. I have an asterisk box setup with an x100p card installed. I have about 3 VoIP phones connected to it. The x100p is connected to my pots line. I also have regular phones connected the the pots line as well like most normal people would. When I get an incomming call all the regular phones ring and my VoIP phones ring. This is good. My question is this. Say

If you have echo on the X100P's Mark setup chan_zap to pretrain the echo can, but it had a few issues until today which Mark nailed down the bug that caused the DTMF to be unreliable. Ok here is how you would do it: in your zapata.conf.sample: ; In some cases, the echo canceller doesn't train quickly enough and there ; is echo at the beginning of the call. Enabling echo training will