similar to: CPU utilization in general

Greetings to all. I am hoping someone can help me out with a problem I am having getting my Cisco phones, 7960s and 7940s, to download the appropriate files from our TFTP server. The TFTP server is running on Fedora Core 4. The TFTP server appears to be setup properly: service tftp { socket_type = dgram protocol = udp wait =

Greetings to all, I ma having a problem with channel variables on a couple of our Asterisk boxes. Here is the setup. Asterisk on customer's site (1.2.5), using IAX to our external GW (1.2.5), IAX to PSTN GW (1.0.10), E1/PRI to PSTN. On the External GW, we also have an IAX trunk to a VOIP provider if for some reason the E1 is down. If the DIALSTATUS is CHANUNAVAIL, which should be

I was just looking through the Wiki for some info on how to retain the original caller's callerid when make transfers to internal extensions, and I came across the parameter below: useincomingcalleridonzaptransfer=yes There is nothing in the zapata.conf file from vers. 1.2.7 so I am wondering if this is still a valid parameter. If not, does anyone know how I can do this? Thanks, Joe

currently we are able to use our USA sip phone to conenct into the E1 box, but still unable to dial out to chinese phone numbers. They said from their ISDN switch console, it shows D channel not connected to the voip server yet. here si the sip debug msg, we got a Message type: DISCONNECT (69) and unable to dial any numbers. Jul 12 12:56:26 WARNING[1523]: chan_zap.c:1931 pri_find_dchan: No

Geetings to all. I am having a hell of a time getting incoming SIP connections to work properly, and am hoping that someone can help me. Here is what I am using as a guide (from the wiki): "Incoming SIP Connections When Asterisk receives an incoming SIP call, the SIP Channel Module first tries to find a [user] section matching the caller name (From: username), then tries to find a [peer]

Anybody know the proceedure to factory reset the a 7960 phone running 6.3 SIP software? I've tried holding # when booting the phone and nothing, i can do that on my 8.2 phone but this phone i just got with 6.3 isnt working. Also **# doesnt work either.. -- ~Shaun

Hi: i have a problem. Mi extensions.conf: exten => _N.,1,Setvar(VOICEMAILREQ=${EXTEN}) exten => _N.,2,SetAccount(${customer}) exten => _N.,3,SetCDRUserField(${VOICEMAILREQ:1}) exten => _N.,4,ResponseTimeout(5) exten => _N.,5,Background(ifyou) exten => _N.,6,Background(silence/1) exten => _N.,7,Background(ifyou) exten => _N.,8,Background(silence/5) exten

I have a major problem with SIP redirects, or maybe just do not understand how they are supposed to work. We are using Cisco 7940s and 7960s with SIP version 6.3. Asterisk 1.2.5. A call come in to extension 944 over the IAX trunk. Extension 944 has forward all to extension 904 set on the phone. According to the dialplan. extension 904 should ring for 90 seconds, then ring another extension, and

Kemal, It sounds like you are doing something wrong. I strongly recommend that you profile your application to see exactly where the CPU time is being spent. AMD happens to have a nice profiler called CodeAnalyst that they give away for free. And it's plenty usable on Intel CPUs as well. http://www.amd.com/us-en/Processors/DevelopWithAMD/0,,30_2252_3604,00.html Make sure you test a

Hi all, We are using Speex codec for our P2P Audio conferencing application. Thus, a node may serve multiple neighbors, requiring multiple encode/decode at a time. I have observed that Speex consumes alot of CPU power; one machine is Pentium 3 800MHz, 256MB RAM with XP. When this PC serves the other two faster PC's, CPU utilization soars to maximum. I have tried various versions of speex;

On 4/8/06, Joe <jrothstein@comcentrixs.com> wrote: > Remove the SIP /400 entry from the Asterisk DB. > > Database del .... At asterisk prompt. > > Or look at the wiki for info on how to remove it. > > Or make sure the SIP/500 uses a different IP address than the old SIP/400. > > Joe > > > > thanks for your reply i've tried to remove the entry in the

Greetings to all, I am getting the following error on my PRI-connected Asterisk box, and am just wondering if anyone else has seen this, and if so how they solved it. Jan 18 10:10:23 NOTICE[6070]: chan_zap.c:7428 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Jan 18 10:10:24 NOTICE[6070]: chan_zap.c:7428 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary

Does anyone have an example of hoe to use these two commands? I have read he documentation, and I am still unclear on where this command goes, as part of extensions.conf or where? If someone could post a working example it would be most helpful. Regards to all, Joe

Grretings to all, I am having a problem with a customer's queue setup that I don't really understand. Background: Customer has 5+ queues and is using dynamic login to the queues based on SIP/XXX for example. There is a litle script that runs that allows agents to log into particular queues via the keypad. The user can log in to any queue that he wants, including multiple queues. The

Hello, I would like to determine overall CPU utilization on a server that''s running xen, and log this information to a file - such as sar can do when run in a non-xen environment. I understand that xenmon shows utilization for each domain, but how can I massage this data to get an overall host server utilization? I''m looking for a single value that describes how much of the host

I have setup several queues for a customer. Their periodic announcement says please wait for the next available agent, or press * to leave a voicemail. This does not work when the message is playing. The message stops, but the user is left in the queue. Q-exit with * works the rest of the time fine. Has anyone seen this or know if it shoudl actually work differently? Regards to all, Joe

Hi, I am currently testing running Openstack instance on Cinder volume with libgfapi. This instance is the Windows instance and i found that when running random 4k write workload, the CPU utilization is very high, 90% CPU utilization with about 86% in privileged time. I also tested the workload with volume from NFS and the CPU utilization is only around 5%. For gluster fuse, the CPU utilization

Hi, I am trying to get the CPU utilization of a node and I am using virNodeGetCPUStats(...) Here is my code snippet. The error checking is omitted for brevity. /* get the number of params */ retval = virNodeGetCPUStats (conn, VIR_NODE_CPU_STATS_ALL_CPUS,\ NULL, &nparams, 0); printf ("nparams: %d\n", nparams); /* allocate space for the params */ params =

  Hi, How i can calculate cpu utilization percentage for host using python libvirt API. Which function can i use for calulating cpu percentage of host and Virtual Machines? 1)Host cpu utilization  ====================  I tried getCPUStats() function,it gives the following output.Is there any way to calculate cpu percentage from the following output?  Is there any other function availabe for

Greetings to all. I am currently implementing call queues for a customer and have come across several "problems". The customer is an airline representative, and will be using call queues for different airline reservations. The customer requires that any agent be able to login to any number of queues. This means that queue members have to be dynamic, not using "member =>