similar to: RE: Is the 2.6 Linux kernel ready for production * environment

uname -a Linux asterisk 2.6.10-1.1771_FC2smp # 1 SMP Mon Mar 20 01:10:51 EST i686 i686 i386 GNU/Linux Processing hundreds of calls & faxes a day. No problem. If you are going to use a Fedora box make sure you yum update right after install then the kernel sources symbolic link is not horribly broken as it used to be. Otherwise, just go. -----Original Message----- From: canuck15 To:

Hello all, I have Asterisk 1.2.1 and a TE110P connected to a Nortel Meridian Option 81C system. The PRI line is currently setup as DMS100. Here are the relevant lines from zaptel.conf and zapata.conf: zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone = us zapata.conf: [channels] language=en context=from-internal musiconhold=default switchtype=dms100

Hi, i have one Asterisk with a Digium E1 card, and a Meridian Nortel Release 11. I need to connect both of them. We are using MFC/R2 for this.. The Diagram: [ NORTEL ] ( AMI ) ------------------------------------------------ (DIGIUM) [ ASTERISK] we have green light at the digium card, and at asterisk we see all 31 channels as idle. But when i want to recive a call from the Nortel to the

Hi R-helpers, I would like to subset my dataframe, keeping only those rows which satisfy the following conditions: 1) the string "dnv" is found in at least one column; 2) the value in the column previous to the one "dnv" is found in is not "0" Here's what my data look like: ??? POND_ID 2009-05-07 2009-05-15 2009-05-21 2009-05-28 2009-06-04 4 ? ? ? 101 ? ? ?

Hi R-helpers, I'm trying to use this code > pvh_dnv<-pvh[sapply(pvh=="dnv"),] to make a new dataframe containing the rows from pvh that contain the value of "dnv" in ANY column. But, it's not working. I get this error Error in match.fun(FUN) : element 1 is empty; the part of the args list of 'is.function' being evaluated was: (FUN) which, to

Hi, I successfully connected 2 servers via IAX but I'm pulling my hair to connect 2 extra servers , Anyone connected 3 or 4 servers together ? is it possible ? I d like to share the dialplan so _2XXXX goes to server A _3xxxx goes to serverB _4xxxxx goes to server C etc from the 4 servers any example of which one is peer, which one is user or friend would help me :-) thanks jl

We've been trying unsuccessfully to connect our Meridian Option 81C to a TE110P via PRI. We've followed the directions in asterisk-meridian-a1.pdf (link on http://www.voip-info.org/wiki/view/Asterisk+legacy+integration), but it doesn't seem to work on our 81C (even though many, many users report it works very well on Option 11's). Has anyone had any success in getting the above

I remember seeing something like this on the list a while ago, but I'm darned if I can find it. We have a number of Polycom IP501 phones, some of which have more than one registration on them. When a voicemail is left for a phone with only one registration, the MWI lights up and stays lit until the voicemail is listened to. However, on our phones with more than one registration, the MWI

Hi there, I'm getting a bunch of these errors from Polycom phones in 1.2.1: ERROR[24301]: chan_sip.c:10790 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 4003 in context internal I've searched the Wiki and archives to no avail - what do these errors mean? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web:

Does anyone out there have a recommendation for tools that will monitor the quality of VoIP systems? I am looking for jitter and MOS monitoring. I have a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms but I am looking for a little more detail. I would not be against writing something in Perl for Nagios to do but I don't really know where to start on measuring jitter

Guys. This idea has been banging my headfor days now and I feel the need to share with you. Imagine this scenario: all calls come in thru a receptionist, asterisk records all incoming calls, the receptionist's work is to transfer the calls to internal people but some of them are bosses and you know how bosses are, they don't want their calls to be recorded, so, I have been trying to

Does anyone know how to maximize music on hold quality on calls inbound from PSTN? I know that it is common to have choppy and static sounding music on hold when connecting via PSTN but how can that be minimized? I assume that the bitrates, type of music, etc can minimize the effects. Does anyone have any experience in this area? Do you know where I should look for more information?

I have a couple SIP phones on a PIII 1Ghz 256MB * server with a TDM01B connected to the PSTN. Calls between SIP phones are clear. Calls to the PSTN are quite noisy. The other person does not hear noise but I hear quite a bit. It is not an annoying sound but definitely much noisier than typical PSTN or even cell phone calls. I believe I have a TDM400P REV H card. I definitely don't

Trying to figure out if a bug report should be submitted. Can anyone on 1.2.x verify of this has been corrected? I am on CVS 8/2005 If a call comes in to an extension that dials more than one channel (rings at more than one phone) both calls in the CDR show a status of answered when only one is answered, the source channel is bridged to only one of the two destination channels, but both CDRs

We have a te410p, with the following connections: span 1 connected to a 32 Channel EuroISDN span 2 connected to a card in a legacy pbx (Meridian) span 3 connected to a 10 Channel EuroISDN span 4 connected to a card in a legacy pbx (Meridian) We have no need for the meridian now, and decided to turn it off. I did not change the zaptel.conf settings, nor the zapata.conf settings. When the

Hi, Some time ago, there was a discussion about the inability of nortel meridian pbx to dial analog tones thru an meridian ATA, and the work arround was to enable 808 function that makes the dtmf tones long for the current call. The nortel meridian is connected via a nortel ATA to a TDM400 to a FXO port. Anyone can say me who to actually use that function (you dial something or is pbx

Has anyone integrated Asterisk for Voicemail with a Nortel Meridian phone system or know if this could be done? Thanks, John Haigh

Hi, I'm on a Dell VEP 1405 running Debian 9.11 and I'm running a few tests with various interfaces given in PCI passthrough to a qemu/KVM Virtual Machine also running Debian 9.11. I noticed that only one of the four I350 network controllers can be used in PCI passthrough. The available interfaces are: *# dpdk-devbind.py --status Network devices using kernel driver

Hi there, I have a *really* weird issue with rspec on rails: Given a time meridian formatted using %p When I run `rake spec` the time meridian is converted to lower case: expected: "Today 08:00 PM", got: "Today 08:00 pm" Yet when I run `rake spec:models` and `spec -cfs app/models` the specs pass, ie the the time meridian is in upper case. Anyone got any ideas how I

How does one go about connecting Asterisk to a Meridian PBX to handle voicemail? I have a customer who is out of capacity on their voicemail system (which connects to their meridian via several FXS cards) and I would like to see if I could use Asterisk to handle their voicemail. -Jonathan