similar to: Tracking how many open files a process has?

Hello List, I need to access some data a MS-SQL server ie run a stored procedure or user-defined function or a select every so often maybe 3 times a day. Have centos 4.5 on intel xeon. Is there an obvious certain-to-succeed approach? perl DBI + ODBC spings to mind, have not tried it yet - cannot access the systems yet - need to make a good plan. Am a bit concerned cos I dont know of an

Hi Centos folks! I'm trying to get a file with wget from Apache on Centos 3 and it doesnt succed. What could be the cause of this what is the resource that is not available, whats your oppinion? strace short output --------------------------- read(14, "GET /centos/3/os/i386/heade"..., 8000) = 133 read(5, 0xbfff99f3, 1) = -1 EAGAIN (Resource temporarily unavailable)

I'm planning for new Centos 4 computers using the 64bit distribution and is wondering if there is a difference on how much RAM memory you should use on a 32bit versus 64bit operating system? Will processes such as for example KDE, Firefox use more memory on the 64bit architecture than on the 32bit and if so by how much as a rule of thumb? Kind regards, Christian -------------- next part

In article <E02FA554-9D6D-4E7D-8A78-5FBDE1DE939D at kicp.uchicago.edu>, Valeri Galtsev <galtsev at kicp.uchicago.edu> wrote: > > > > On Sep 26, 2020, at 8:05 AM, Jerry Geis <jerry.geis at gmail.com> wrote: > > > > I have a disk that is flagging errors, attempting to rescue the data. > > > > I tried dd first - if gets about 117G of 320G disk

In article <20151125133008.6369360.14455.17239 at gmail.com>, Israel Gottlieb <isrlgb at gmail.com> wrote: > Try putting progress instead of answer Yes, I tried Progress already, and it didn't help. But thanks for the suggestion! Tony > I have a puzzling situation, and would be grateful for any insight. > > I have a dialplan that forwards an incoming call out to

has anyone done caller announce in MeetMe's before? Dave P >>> brian@bkw.org 5/18/2004 5:50:49 PM >>> With multiple parking lots you can give each person their own lot thus exten 800 for everyone will connect them with just their call passing the lot name which you know for X customer. bkw ----- Original Message ----- From: "Andrew Kohlsmith"

I've just set up a new box with FC1+updates and the latest Stable Asterisk from CVS. Asterisk is started with the default safe_asterisk script with a console on TTY9. The coloured text on this console is made up of weird characters instead of normal. Please see http://www.softins.co.uk/dsc00018.jpg for an example. If I do "asterisk -rvvvvv" on a normal login, either via the

Hello List, I have seen from the following link that, for SIP channels there is no audio communication possible in MeetMe with AGI_BACKGROUND. http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe Currently we are using asterisk-1.6.2 and the problem still persists. Is there any solution available to overcome this problem? According to our requirement, we have to run an AGI script in MeetMe.

I have been writing a lot of AGI programs in C with good success. I would like somehow to have an AGI program continue in the background while the pbx execution returns to the dialplan and continues. Is this possible? I was thinking that perhaps I could fork or create another thread within the AGI prog. The reason I want to do so is in order to monitor external information (e.g. credit limit and

I am having problems getting the latest CVS right now. A cvs checkout asterisk -t gets to this part and sits forever: S-> server_register(fpm-world-mix.mp3, 1.1, , , , , ) S-> Register(fpm-world-mix.mp3, 1.1, , , ) Anyone know how I can just skip the file? Travis Conway EFS, Inc. Information Technology Desk:?? (334) 215-6551 Mobile: (334) 391-4450 mailto:travis@homeoffice.quikpawn.com

I've noticed that with FastAGI (and maybe AGI) that when you sequentially send a sequence of dial commands, if the call is picked up, that after the call ends, the Fast AGI script keeps executing the commands! Is there anyway to stop execution once a call is picked up? I think looking at the result codes after the Dial to determine if the call was picked up or not is not a good idea... if it

I have a puzzling situation, and would be grateful for any insight. I have a dialplan that forwards an incoming call out to another number via the same SIP trunk as it came in on. e.g. [from-siptrunk] exten => 0123456789,1,NoOp exten => 0123456789,n,Dial(SIP/siptrunk/0987654321) Now, if I use a different SIP trunk for the outbound call, than the inbound call came on, the call is set up

Does anyone know how a program, script or shell user can best determine whether the machine is running on bare metal or is a VMware guest? Cheers Tony -- Tony Mountifield Work: tony at softins.co.uk - http://www.softins.co.uk Play: tony at mountifield.org - http://tony.mountifield.org

Thanks for the feedback. I do agree with having multiple smaller servers. When I was first approached with this task I mentioned as much. However, the current desire is to work with already existing hardware. That is out of my hands at the moment unless it just can't be done. I will explore Freeswitch a bit soon to compare it as well. I am struggling to find what the bottle neck is in

I have a small number of boxes in different locations, and currently have a fairly crude cron job running on each, which does a ping of one or more of the other boxes, and if the ping fails, it emails me to say the other box might be down. It then emails me again the next time the other box appears to be up. Of course, this can't distinguish between the remote box really being down and there

Can anyone de-cypher the second entry for me? --------------------- httpd Begin ------------------------ Requests with error response codes 403 Forbidden /: 9 Time(s) /?c=4e5e5d7364f443e28fbf0d3ae744a59a: 3 Time(s) I have found the string via Google but have not located any explanation. -- *** e-Mail is NOT a SECURE channel *** Do NOT transmit

Thanks for the suggestion Tony, I installed each codec for MoH, core sounds, and extra sound packages. Unfortunately the tests produce the same results. [Sep 1 20:36:45] ERROR[10081][C-00007fe5]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x20380b0 ( continuously for a while followed by a [Sep 1 20:36:46] WARNING[7761][C-0000770d]:

Hi all, Does anyone know how to get the channel ID on the other side of the call? For example: When SIP/50 calls SIP/21, and the call is answered by SIP/21 I get: SIP/21-6735 answered SIP/50-b456 ${CHANNEL} will show me SIP/50-b456. Is there a parameter or a workaround to get the SIP/21-6735 part? Thanks. Assaf Benharoosh -------------- next part -------------- An HTML attachment was

On 06/09/2015 12:48 AM, Nicolas Thierry-Mieg wrote: > On 06/09/2015 12:33 AM, tony at softins.co.uk (Tony Mountifield) wrote: >> In article <ml1jnh$afr$1 at softins.softins.co.uk>, >> Tony Mountifield <tony at softins.co.uk> wrote: >>> When I started using CentOS 6 instead of CentOS 5, I discovered that >>> "less" no longer understood \<

Can someone tell me why this is not valid... [start] exten => 1000,1,Answer exten => 1000,2,Wait,1 exten => 1000,3,AgentcallbackLogin(1000||2000@Local) exten => 2000,1,Macro(DialProxy,115551212) exten => 3000,1,Queue(testq||||45) while this is: [start] exten => 1000,1,Answer exten => 1000,2,Wait,1 exten => 1000,3,AgentcallbackLogin(1000||2000@start) exten =>