similar to: Bulk Reverse Phone Lookup

I'm trying to understand something that just doesn't seem to compute. How can companies like Cisco justify selling their hard phones for as much as they do? I know there is a matter of recouping R&D costs but when you look at the iPhone with all its amazing features for less than $500.00 it just doesn't make sense. Am I the only one that thinks this? Roy Anciso Director of

I've been evaluating Asterisk for a while, and things seem to be going very well. The issue of redundancy and automatic fail-over is now on my mind. I searched the archives and googled for solutions, but didn't really come up with much. We'll be using queues (modified), which precludes some of the standard redundancy solutions, since the queue needs to know all the agents

On Mar 19, 2008, at 1:00 PM, asterisk-users-request at lists.digium.com wrote: > Am I expecting too much? Perhaps. I think the hardware on which we run Asterisk can be much more reliable than the software, which is often the case. We have a bunch of HP servers with RAID and have never lost anything. A HD may fail, but the RAID keeps it going until we pop a new drive in there. A

I am attempting to establish SIP peering between Asterisk and an AltiGen soft PBX. This is my first experience with SIP peering. I can successfully make both inbound and outbound calls to/from a softphone on the AltiGen system (network access is provided by a PRI on the Asterisk system), but they are disconnected unexpectedly. The attachment is a redirect of the Asterisk CLI during a call that

What does it take to get ztdummy to work correctly? I have a new laptop HP HDX9200. I am running asterisk 1.4.19 and zaptel 1.4.9.2 Zaptel compiles fine. asterisk compiles fine. ztdummy loads asterisk runs. Problem is playback() does not work. So then I stop zaptel, asterisk runs and playback() now works. However, meetme()'s dont work. I need ztdummy I'm pretty sure for that. I am

Hello All, We have been experiencing some ongoing reliability problems with Asterisk for quite some time, and I am trying to find out if anyone else has experienced the same problems. We are running asterisk 1.4.17~dfsg-2+b1 on Debian Lenny, with a Digium PRI card, and have approximately 120 sip peers, mostly Snom 360s, with a few Grandstream GXP2000 and a handful of Handytone 486 units. The

Does anyone know of an online tool or program that would call our phone numbers to confirm that they are up? What I'm imagining is something like pingdom.com (a ping checker for web services), except for the phone system. It would call us, confirm an answer, then hang up. If no answer, it would send an e-mail to our cell phones so we knew about it right away. It seems like there would be a

Job Description: Asterisk MySQL Support Engineer Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD

All, I recently got a request to deploy VOIP helplines with a bank's ATM site locations. The purpose of this solution was to be able to resolve cash dispensing errors as quickly as possible. My problem is that i do not have any idea about outdoor type VOIP equipments. Does anyone have ideas???

Has anyone had good results with an on-line database that returns a LATA based on NPA NXX? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100518/d77d09b3/attachment.htm

Hi * users, I would like to extract the information present in the SIP REFER message that comes to asterisk. Would SIP_HEADER() allow me to do that ? I have used SIP_HEADER() for extracting the to and from SIP headers previously. Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998

I suppose this is rather an informative e-mail than a question. However if people had similar experiences or could comment what the differences are in other countries or with business analog lines, it would be interesting. It took me a week until a BT engineer was sent to my home home, since BT tech support was unable to provide information about the problem. Problem: Calling party controls how

Anyone using this feature of asterisk's voicemail? I'd never heard of ADSI, and saw it as I was perusing the voicemail source this morning. Is it some kind of visual way of managing voicemail on your phone's display, or does it require a terminal of some kind? Thanks! j

While browsing about, found http://www.voip-info.org/wiki/view/TDM400P, where I found this comment: "Here's a tip passed on from an old telephone engineer. Where your copper 2-wire cable approaches the building, underground, finish with several large loops, about a metre in diameter, laid on top of each other. Fast moving, high energy spikes will spin off the outside of the loop as they

I'm interested in taking a persons spoken recorded name (First, Last) and converting the two spoken words to text. Is there any solutions out there that would make this possible? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090130/1e90ba97/attachment.htm

I've tried timing faxes two ways:

In conjunction with Mark Spencer's visit to our Paris office, we'll be kicking off Asterisk TV (http://asterisktv.com) live during the weekls Asterisk Users Conference. I believe someone from Lumenvox will be back with us on the conference, now that I've had a chance to play with their speech recognition product. I think we're starting to get some great info from the user

I've had it with Dell server garbage. They seem to change RAID controllers as much as I change socks, and then the controllers don't work with Linux, unless you load a new driver. They sell servers with a PCI-e slot in them, but then you get it and find out the RAID controller is using the PCI-e slot! Their sales folks are dumber than rocks, and they change them more often than I

Hi List; I compiled Zaptel 1.4 and Asterisk 1.4 after downloading them using svn, but when I checked the file zaptel.conf under etc/asterisk, I did not find this file. Any help? By the way: How can I know the asterisk and zaptel version extactly that I compiled them? In other words, asterisk 1.4.... and zaptel 1.4.... ? Regards ------------- ITS IP Telephony and Contact Center Engineer Eng.

hi i have configured an asterisk server which i have tested locally with x-lite and that's ok but when i wanted to access to it since internet that hasent taken place knowing that my server has access to internet by a wifi router that has a public ip address (e-g a.b.d.c) and asterisk server has a private ip 192.168.1.111 (the firewall is disabled) can some one tellme how i should configure