Re: [Asterisk-Users] Call Center with No TDM components

2006-04-19 Thread Begumisa Gerald M
On Wed, 19 Apr 2006, Abhimanyu Rapria wrote: Transcoding and Recording is being done at VICIDIAL/ASTERISK Dialer and load average is 1.5 for 12 agents and pacing of 1.1 to 1.2 What is the average CPU utilization you observe with these load averages? Regards, Gerald.

Re: [Asterisk-Users] te110p and interrupts

2006-04-17 Thread Begumisa Gerald M
Hi Paul, Thanks for the message! On Sun, 16 Apr 2006, Paul Hewlett wrote: [...] I am curious.. Have you tried disabling CPU1 by setting isolcpus=1 on the kernel command line ? This will make the kernel ignore the second CPU - you can then run

Re: [Asterisk-Users] te110p and interrupts

2006-04-17 Thread Begumisa Gerald M
On Mon, 17 Apr 2006, stoffell wrote: Interesting. Now 'why' do they suggest it, is it because older IO-APIC are 'broken' on some boards? I'm very curious as to 'why', [...] Most likely this is why. Regards, Gerald ___ --Bandwidth

Re: [Asterisk-Users] Digium cards, so disappointing !

2006-04-15 Thread Begumisa Gerald M
Hi Steve, Thank you for your very enlightening message! On Sat, 15 Apr 2006, Steve Underwood wrote: [...] modem it must be applied end to end by the modems themselves. The real killer, though, is imperfect timing. [...] and its not always always available within

Re: [Asterisk-Users] te110p and interrupts

2006-04-11 Thread Begumisa Gerald M
On Tue, 11 Apr 2006, Andrew Kohlsmith wrote: Please do not open your mouth to spout nonsense if you do not know what you're talking about. [...] Again, if the IO-APIC is reporting that the card is on its own IRQ, it really, truly, honestly *IS* on its own IRQ. The

Re: [Asterisk-Users] te110p and interrupts

2006-04-11 Thread Begumisa Gerald M
Hi, I've been battling with a similar issue: a) I wrote a script to periodically run the command cat /proc/interrupts and figure out the interrupts per second. I run this script for over 24 hours and never once did the difference between the preceeding and succeeding interrupt counts go below

Re: [Asterisk-Users] Asterisk 1.2 and weird ZAP interface behaviour

2005-12-03 Thread Begumisa Gerald M
On Sat, 3 Dec 2005, Remco Barende wrote: Whenever I pick up that phone I get on the console: Dec 3 16:37:36 WARNING[19551]: pbx.c:2347 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' Have

Re: [Asterisk-Users] Asterisk 1.2 and weird ZAP interface behaviour

2005-12-03 Thread Begumisa Gerald M
On Sat, 3 Dec 2005, Remco Barende wrote: I have but only for the phone line, it is immediately after: signalling=fxs_ks immediate=yes What I actually meant is that you should turn this off if you don't need the functionality. Most likely you are defining the extension

Re: [Asterisk-Users] New Application: Broadcast

2005-10-13 Thread Begumisa Gerald M
Hi Steve, On Thu, 13 Oct 2005, Steve Daniels wrote: What excatly does it do? What messages does it send out? And what software needs to be configured to listen for these messages? Bret explained mostly what the software does in a basic use case where you would like a nice window

[Asterisk-Users] New Application: Broadcast

2005-10-12 Thread Begumisa Gerald M
Hello, I've released an Asterisk application under the terms of the GNU GPL. You may find it here: http://psg.com/~begg/projects/ A short exerpt from the README: -- Broadcast is an Asterisk (http://www.asterisk.org) application which you may use to send a generic message over TCP/IP to any

Re: [Asterisk-Users] RE: Getting phpconfig to work?

2005-03-03 Thread Begumisa Gerald M
Hi, When I do click on the phpconfig.php link from http://ip-of-machine/phpconfig/, it returns a page with the actual contents of that file (phpconfig.php) and doesn't load the page. See some of the output below; It's quite likely that your Apache+PHP installation is

Re: [Asterisk-Users] wctdm and two tdm cards

2005-03-02 Thread Begumisa Gerald M
If I reboot the system with reset button, ctrl alt del, or 'reboot' the TDM04P does not get detected. To completely reset the TDM cards before they can be reliably detected again, you may have to completely power down the machine - even to the extent of pulling out the power plug and

Re: [Asterisk-Users] Digium Card Problems

2005-02-28 Thread Begumisa Gerald M
Hi Mark, On Mon, 28 Feb 2005, Mark Kidd wrote: modprobe zaptel - no problems [EMAIL PROTECTED] root]# modprobe wcfxo I'm just curious, did 'modprobe wcfxo' ever work? I seem to remember that for the TDM400P suite, the module to load was (rather confusingly) 'wcfxs', even

Re: [Asterisk-Users] IPCB

2005-02-24 Thread Begumisa Gerald M
On Thu, 24 Feb 2005, HASS, JOHN wrote: type=peer For some reason type=friend seemed to solve a similar problem I had (not with IP Clearing Board, though). I was kinda too busy to figure out why it solved the problem, actually [sorry] but it *may* be worth checking out. Then, just to

Re: [Asterisk-Users] how to manage Digium TDM04B outgoing calls correctly

2005-01-20 Thread Begumisa Gerald M
So if you think the server can handle 5 TDM400P cards let me know. I've done an installation with 5 TDM400P cards - 4 PSTN lines and 12 analog phones. There are no outstanding issues that havent been solved by tweaking a particular config option (e.g echo, callprogress issues etc...).

Re: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server

2005-01-15 Thread Begumisa Gerald M
Yup, I found their support very unhelpful and unwilling to go the extra (or even the first) mile.. Might ACPI (not APIC) have anything to do with this condition? I once had a hard time with a bunch of cards which were not taking interrupts. I disabled ACPI interrupt routing (from

[Asterisk-Users] TDM31B has no interrupts?

2004-11-16 Thread Begumisa Gerald M
Hi, I've installed a TDM31B card successfully but had a few problems making calls through it - summary is below: o Calls cannot be placed using an analog phone o The interrupts count value in /proc/interrupts remains at zero (see below) CPU0 0: 7495 XT-PIC

Re: [Asterisk-Users] TDM31B has no interrupts?

2004-11-16 Thread Begumisa Gerald M
Hi, Thanks for taking time to answer. Not enough info in the above to hint at the problem. What linux distro, SuSE Linux 8.2 2.4.20-4GB what does your /etc/zaptel look like, For the TDM22B card: fxoks=1-2 fxsks=3-4 loadzone = uk defaultzone=uk zapata.conf

[Asterisk-Users] TDM31B Interrupt Issue SOLVED! :-)

2004-11-16 Thread Begumisa Gerald M
-- My apologies if this gets posted twice. I made a mistake with my from address. -- -- Hi All, Many thanks to everyone that gave input on the above issue. I'm glad to announce its been solved. The trick: -- TURN OFF ACPI! -- With SuSE you can do this by setting the boot option pci=noacpi.

[Asterisk-Users] 3 - TDM31B Card Installation Difficulty

2004-11-14 Thread Begumisa Gerald M
Hi, o I purchased 3 TDM31B cards and fixed them in my computer (in 3 PCI slots) o I downloaded the latest Zaptel source from CVS, compiled it and loaded modules zaptel.o and wctdm.o. o I successfully configured them from /etc/zaptel.conf as shown in the information below. ztcfg returned

Re: [Asterisk-Users] 3 - TDM31B Card Installation Difficulty

2004-11-14 Thread Begumisa Gerald M
Hi Steve, If you call from X-Lite to the demo menus can you hear them clearly (no choppy sound)? Actually I can't - the sound is still choppy! Interesting. When I unload the zaptel and wctdm modules the problem goes away (I can hear the demo files quite clearly from the X-Lite

Re: [Asterisk-Users] Dialing Out through Provider with Authentication

2004-09-10 Thread Begumisa Gerald M
On Thu, 9 Sep 2004, Karl Brose wrote: In order to dial out to a sip provider, you need to configure that provider in your sip.conf file as a peer with your proper username and secret, etc. Cool! Just found that in the handbook too a second or two ago :-) Thanks for taking

Re: [Asterisk-Users] Dialing Out through Provider with Authentication

2004-09-10 Thread Begumisa Gerald M
On Fri, 10 Sep 2004, Johannes Hollerer wrote: I tried to make a call to extension 2001 with the setting [EMAIL PROTECTED] (Detailed: exten = _7.,2,Dial(SIP/[EMAIL PROTECTED]/${EXTEN:1}) which does not work at all - i always get the failure message: No such host

Re: [Asterisk-Users] Asterisk testbed for teaching connecting to a PRI-ISDN

2004-09-10 Thread Begumisa Gerald M
On Fri, 10 Sep 2004, Francesco Delfino wrote: [...]One of the box will represent the Telco, the other two, the two companies PBX. I would like to know if it is needed something between the point-point connections or it is possible to just cross-connect them. As more

Re: [Asterisk-Users] Dialing Out through Provider with Authentication

2004-09-09 Thread Begumisa Gerald M
On Thu, 9 Sep 2004, Johannes Hollerer wrote: I try to dial out through a Provider, but for that i need to be authenticated - it actually does not work !. For my tests I did not need to be authenticated. This is what I used in asterisk: exten =

Re: [Asterisk-Users] Dialing Out through Provider with Authentication

2004-09-09 Thread Begumisa Gerald M
But the provider also has a gateway to provide the possibility to call to the pstn (and the pstn number exists) - so what i tried to achive is to call an external pstn number thru that gateway. This works if i connect the xlite client directly to the provider - then i can

Re: [Asterisk-Users] Placing Asterisk between existing PBX and PSTN

2004-09-07 Thread Begumisa Gerald M
2004, Begumisa Gerald M wrote: Hi, I've read through the Asterisk handbook and I'd just like clarification from someone that's implemented the above before. Lets imagine I want to use the CallingCard application and don't want to tell a client to buy a channelbank

RE: [Asterisk-Users] Placing Asterisk between existing PBX and PSTN

2004-09-07 Thread Begumisa Gerald M
Hi Jens, Thanks alot for your input, I do appreciate it! [...] I would like to suggest that you don't try this with analogue lines (fxo) and extensions (fxs) - you will not be able to monitor call progress and lose all (possible) DDI information. Imagine my original setup was

[Asterisk-Users] Placing Asterisk between existing PBX and PSTN

2004-09-06 Thread Begumisa Gerald M
Hi, I've read through the Asterisk handbook and I'd just like clarification from someone that's implemented the above before. Lets imagine I want to use the CallingCard application and don't want to tell a client to buy a channelbank (the analog extensions are too many to connect to FXS cards),