similar to: Server recommendation help

I am trying to tweak my dial plan and I am running into a problem. Sometimes my VoIP out bound calls do not complete on overseas calls(busy or just a hang-up). Is there a way in the dial plan to automatically dial out of my PRI when something like this happens. Either by time limit by a failure event? Any point in the right direction would be great Thanks, CLI output (cleansed to protect the

Hello, Digium uses the Dell PE 2850 for their testing. This site says that 3.3V PCI slot. http://www.voip-info.org/wiki/view/Asterisk+hardware We are planning on purchasing a Dell PE 2850 and putting a TE205P card on it. However, the needs a 5V PCI slot. Does Dell PE 2850 has a 5V PCI slot? A person in our group tried to call Dell's customer support but they do not seem to know.

hi i'm trying to configure a TE205P on asterisk@home i've edited /etc/sysconfig/zaptel adding this line: MODULES="$MODULES wct2xxp" now, when the system is loading, i can see that the wct2xxp module is loaded correctly but if i try the command: /usr/local/sbin/genzaptelconf i get: STOPPING ASTERISK STOPPING FOP SERVER Generating '/etc/zaptel.conf' Generating

Hi List!! Thanks for the colaboration, especially to Richard Cavanna who gave me the necessary support. I followed your indications and the comunication was better for the test users. The warning indication is no jumping anymore and the voice is not delayed. This is my sip.conf: [general] context=default ;allowguest=no ;realm=mydomain.tld bindport=5060 bindaddr=0.0.0.0 srvlookup=yes

Hi, Has anyone had success using a TE410P card in an HP-Compaq DL380 G4 server? For me the card is detected fine, but the system just never sees an interrupt from the card. I've tried everything I can think of. The card definitely works. Its Fedora Core, but we also tried a stock 2.6.10 kernel. We tried with and without Hyperthreading, with "noapic", we disabled all the

Hi, I am having problems with an Asterisk with a Digium TE205P card. The issue is that the Ring Back Tone is noisy. I am making modem's calls and this noise influences on the initial negotiation protocol, so modems have to recall. My configuration is: Asterisk version: Asterisk 1.4.21.2 Linux version: CentOS release 5.2 (Final) Card: Digium TE205P

Hi guys, Currently I just purchased the TE205P for the installation of my ISDN30 setup. I compiled, setup the card correctly I think. When I do the ztcfg -vvv it shows all 31 channels clear, with 30 bchannels using for voice ranging from 1-15, 17-31, and 16 is the dchannel. However, when I do a cat /proc/zaptel/1 I see only 24 channels configured, and then, when I do a cat /proc/zaptel/2 I see

I have a TE205P, jumpered for E1, added the missing wct4xxp-line to /etc/modprobe.d/zaptel, zaptel.conf is just span=1,1,0,ccs,hdb3,crc4,yellow span=2,2,0,ccs,hdb3,crc4,yellow bchan=1-15 dchan=16 bchan=17-31 bchan=32-46 dchan=47 bchan=48-62 Which, according to my reading of the documentation I could find, should be correct. ztcfg doesn't complain about anything, yet all I ever get is RED

I'm looking at ordering a server from HP. I checked around on Google and found in the Wiki that the ProLiant DL380 is supposed to be known to work with *. I'm going to get a price quote on the following setup: HP ProLiant DL380 G4 Server w/ the following options: Intel Xeon 3.20GHz/1MB 2GB REG PC2-3200 (2 X 1GB) HP ProLiant Battery Backed Write Cache Enabler for SA6i RAID 1 drive set HP

Hi, I have an asterisk box with a QuadBRI. I am close to make it work like I want: Incoming calls to SIP are OK, outgoing calls are quiet working thanks to Michael Sandee advices. My real problem is that my * box is freezing very often when I probe qozap (bri-stuff 0.0.2). I have tried two distribs (Debian Kernel 2.6, Mandrake 9.1 Kernel 2.4) and it always work for a few minutes and freezes. At

Hi all, I'm confused, in digium website, it says: TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a total of 24 lines. 6 plus 6 is 12, how come it's 24? if I have 24 PSTN lines, i'll be needing 24 FXOs. Pls. elaborate. thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

All, I am setting up asterisk on a nslu2 (Linksys) using unslug. Everything is working great except that I have a polycom 301 and I cannot get the message indicator to work. I have created the users and mailbox in users.conf and I can manually dial the mailbox (*986000). Last thing is I am not using config files for the polycom just web browser. Can anyone point me in the right direction I

I have a TE205P that has been working fine for 2 years. power outage yesterday took out my everything for over an hour. Everything has come back up except the PRI. My provider has checked it to the box and says everything looks good on their end. I get this message: [Jul 9 12:40:32] WARNING[13709] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! ztcfg

Hi I am in the middle of move a small business over from legacy PABX + PSTN lines to VOIP infrastructure. I borrowed a spa9000 to place between the PABX and the PSTN lines. I have had this going for a while (>5 months) and it has been working fine (some issues with echo and other minor things), which is why I am moving to asterisk. I bought a tdm410 with 3 fxo + fxs. The fxs is connected to

Dear list users, let me ask you this trivial question. I worked on that for a long time, by now. Suppose to have a data frame with NAs and to sum some columns with rowSums: df <- data.frame(A = runif(10), B = runif(10), C = rnorm(10)) df[1, ] <- NA rowSums(df[ , which(names(df) %in% c("A","B"))], na.rm=T) If all the elements of the selected columns are NA, rowSums

New never used bought by mistake ( 5V pci slot ) and stored... pls make offer, thanks and regards, Jean-louis Curty The TE207P is a bundling of our leading TE205P product and our new VPMOCT064 Octasic DSP-based echo cancellation module. The TE207P is Digium's and the industry's first two-port digital card featuring hardware-based echo cancellation. The new module enables users to

Hi everyone! I have a Digium Card TDM410 But, it appear for me this massege chan_dahdi.c:8061 handle_alarms: Detected alarm on channel 3: Red Alarm But my line is ok! But sometimes it back sig_analog.c:3807 analog_handle_init_event: Alarm cleared on channel 2 But it again back to red alarm. What can be happen? My lines is all ok! But when I put on Digium Card TDM410 is very inconsistent

> On 20 Jul 2018, at 15.20, Fineware PM <postmaster at fineware.it> wrote: > > if I dump the index of inbox : > "doveadm -v dump example.com/user/idx/mailboxes/INBOX/dovecot.index" > > I don't find the message. > I think is corrupted the index > > try then running: doveadm force-resync -u user "*" Sami -------------- next part

Hi all, I've setup a PBX in production environment last week but we have immediatly find out drop conversations and lot of errors (in asterisk logs) like: Apr 10 17:30:48 NOTICE[25154] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 My configuration is: Motherboard: SuperMicro P8SCT (Intel E7221 chipset) Hypertrading and APIC enabled CPU: P4 3GHz Digium TE205P

After speaking with several people, and even participating in a forum of several other people with echo issues, I thought I'd share what we've done (well actually what our chief R&D engineer, Brett Bourn has done...) First let me say that normal cheapy PC hardware couldn't be made to function with out echo. We tried on both the single port Digium T1 card and the 4 port Digium T1