similar to: Anyone using * in a live production environment?

Displaying 20 results from an estimated 7000 matches similar to: "Anyone using * in a live production environment?"

2005 Mar 24
2
Fun with CAPI
Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension
2003 Nov 05
10
Reasons why I shouldn't use Asterisk?
It would seem an odd question, but I'm trying to put together a little presentation on 'Why Asterisk?' and need to list Pros and Cons.... I've got plenty of Pros (including the availability of commercial support), but the only Con I can think of is 'Relatively few installations worldwide' Can anyone think of any others? Cheeres, Gavin.
2004 Apr 19
2
Advanced queueing
Hullo :) Please be gentle with me, I don't have a working * install, and am just looking for background information. I'm always impressed by companies who implement a queue like "You are now number N in the queue. There are currently M agents answering calls, and your call should be answered in approx. O minutes" I've seen on
2004 Apr 23
2
zaprtc on 2.6
Hullo. Having found http://bugs.digium.com/bug_view_page.php?bug_id=0000875 I grabbed the original 0.0.1 and Dan's patch, and whilst it didn't apply, I was able to patch the zaprtc.c manually - the Makefile has changed a lot, and I wasn't able to understand the changes. (this is all on a machine that's never had any * on it before, and has a 2.6.5 kernel with a matching
2003 Dec 17
5
Readline & readline-devel installation on RH9
I have a new user question. Sorry I know most of you are Linux experts I am not! I am just getting my feet wet with this. And I am sorry to ask this stupid question. I was following an installation post from Wiki that said when using RH 9 you need to make sure that you have the following installed first and you should check them with the following command. Are there any other items I need to
2004 Apr 25
1
MusicOnHold spawns everlasting mpg123 processes
Hullo :) I'm using CVS-04/23/04-23 from the stable 1.0 branch on kernel 2.6 - since I have no Digium h/w, I've just managed to get the zaprtc module to compile and run, so I thought the best way to test it would be via MoH. The MP3Player application works great .. exten => 6901,1,Answer exten => 6901,2,MP3Player(http://127.0.0.1:85/ES/28) This will play callers BBC Radio 4 from
2004 Jan 22
5
Snom 200 phones not working.
I have 2 Snom 200 and would like to get them to work properly with Asterisk. With the Firmware 2.02t I am able to use the phone. But only one line configured. With there newer firmware 2.03o it will not allow me to make calls. But I can get calls on the unit. Again the 2nd line is not able to be registered. Is this an issue with Asterisk or Snom? I could use some example configuration
2003 Oct 13
4
IAXTEL/ Dial problem
Hello I am still having problems with IAXTELL and FWD configuration. I get the following when I dial 17009965342 which is set as an example to dial to FWD people. 1+700+99+ 5 digit number. I have placed XXXXX where my passwords are. CLI> Executing Dial("Zap/14-1", "IAX/abatista:xxxxxx@iaxtel.com/917009965342@iaxtel") in new stack -- Calling using options
2003 Nov 12
2
Media Negotiation Failed
Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip: 64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome message and resend call to Cisco 3600 that have 4 analog lines connected... but after cisco play welcome message and when
2003 Sep 17
4
Programming 976 numbers from dialing out.
I would like to prevent * from dialing 900 and 976 numbers. I setup the following settings in extensions.conf. But this does not seem to work! I don't know what I am doing wrong please help! exten => 1900XXXXXXX,1,Congestion exten => XXX976XXXX,1,Congestion exten => XXX976XXXX,1,Congestion exten => 1XXX976XXXX,1,Congestion exten => 91900XXXXXXX,1,Congestion exten =>
2003 Nov 24
4
Sip phones!
I am trying to get the following phones for testing. Is there a distributor in the US that is able to sell me these Sip phone and ATA adapters? I can not afford the Cisco phones there too hard to configure and too expensive! 1 - Sipura SPA-2000 2 - Grandstream Sip phone BT-102 1 - Grandstream HT-286 1 - Snom 105 Sip phone. I have called and emailed chagres but they have not reply. Nor
2005 May 10
1
SIP transfers failing
Hullo :) I'm using Debian's Asterisk 1.0.7 bristuffed (though I'm only using CAPI for ISDN, and not HFC-S cards) and trying to transfer an incoming SIP call from sipgate.co.uk to any other extension. My phones are AT-320s (PA168S 1.43 firmware) whose documentation says to blind transfer, simply dial the number you want to transfer to, and press 'FWD'... This is what
2003 Dec 22
2
Sipura 2000 configuration.
Ok here is another problem I have run into. I have a Sipura 2000 and I have been able to configure line 1 with only one small problem. But I can't get the line 2 working with asterisk. Here are samples of my sip.conf and extensions.conf. If I disable line 1 I can then get line 2 working. Is there a sample configuration for the Sipura to get both ports working with Asterisk. Sip.conf
2003 Oct 07
3
Line going to Zombie
I have a problem that sometimes lines will go into what I call never never land. The Asterisk system will put a line with <Zombi> on it when you type show channels it will make the analog phone line dead. And on the CLI it says: astsvr*CLI>Zap/1-2<ZOMBIE>(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r I have tried to release it with soft hangup Zap/1 & also soft hangup
2003 Dec 05
2
Help with setup IpDialog Sip Phones.
I just got 2 IpDialog phones for use with my Asterisk system. I have been able to get the phones to just dial local extensions but it is not able to register with my system correctly. I would like to know if someone has set these phones up before and how they did it! Is there any examples for use with Asterisk? They seem simple enough to config with there web interface. Thanks
2005 Feb 11
5
Asterisk@home .05 release questions on setup.
Hello, Great job on the Asterisk@home project. Looks great this new version is really nicer looking. But I have a few questions. 1) For the new web access http://localIP/maint how and where do I change the password. 2) Since I don't use the Amp section for setup the .conf files I use my own. How do I get the asterisk server running status up. I have it running and works but shows up as not
2003 Sep 22
2
Re: Anyone looking for IP Phones?
---------- Original Message ---------------------------------- From: Louis-David Mitterrand <vindex@apartia.org> Reply-To: asterisk-users@lists.digium.com Date: Mon, 22 Sep 2003 22:28:40 +0200 >On Mon, Sep 22, 2003 at 03:25:00PM -0400, Sales wrote: >> My company has approx. 500 Cisco CP-7960G IP Phones that are coming out of >> service. They were deployed for about 6
2006 Apr 27
5
Xen 3.0.2 on AMD64 - and initrd fun :)
Mm, I have a big Quad-Opteron.. thing.. that I''m trying to get Xen onto. I''ve used the 3.0.2 binary-install mode, updated menu.lst as per the README, but I need an initrd which contains the HP cciss RAID driver, and no Xen initrd image was installed into /boot. Now I notice xen-3.0.2-2-install/install/lib/modules/2.6.16-xen/kernel/drivers /block/cciss.ko But I
2003 Nov 12
3
DIAX 0.93 with some sound improvements and not only...
Hi all, DIAX 0.9.3 is available for download from the same place: http://www.laser.com/dante or http://www.geocities.com/tdanro The new DLL contain the latest updates made by Steve in the iaxclient library. Still just IAX1 is supported (for the moment). What's new in 0.9.3? - accept blank passwords; - accept for registration/calls host names, not only IP Address; - password no
2004 Aug 29
2
Sip device not login or register calls to that device go to busy voicemail not un-available
I feel this is in error some place. If I call a sip device that is not registered or not connected at the time. Asterisk will send that call to voicemail to busy not unavailable. Is there a way to correct this? Ariel Batista Kasi International - Computer Networking Ph: 305-574-6721 Fx: 305-574-0212 -------------- next part -------------- An HTML attachment was scrubbed... URL: