similar to: need * consultant in houston area

Does anyone know if sip phones from any of the major IP phone vendors support 802.1x authentication? Any feedback would be greatly appreciated. Thanks in advance. ====================== Jeronimo Romero EUS Networks Email: jromero at euscorp.com Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com ====================== -------------- next part -------------- An HTML

Running Asterisk 1.2.9. I just installed a TE110P card and configured zaptel.conf & zapata.conf. The config files look right to me but I'm getting the following error when trying to start asterisk: Asterisk died with code 1. Automatically restarting Asterisk. Does anyone have any idea what is wrong with this configuration?? Thanks in advance!!! Here's my config files: zaptel.conf

We are doing a medium sized office in NYC with 80 phones. The customer originally requested Polycom 601 phones. The COO also authorized us to purchase 2 Grandstream GXP2000 phones for the mail room. We find these phones much easier to configure and work with asterisk . They support BLF & intercom right out of the box. They can also be centrally managed and provisioned. They also sound great

speech rec what works ? anything out there with established dictionary, eg medical ? don't want to pay $3-4K for Nuance API thx in advance _________________________________________________________________ Is your PC infected? Get a FREE online computer virus scan from McAfee® Security. http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963

I'm trying to set on-demand call recording. Here's a snippet of the pertinent dialplan. The purpose of this is to allow one user in particular to be able to receive an email recording of the call everytime he dials *91 + number. The problem is that the email is not going out or being generated when I use the ${CALLFILENAME} variable. When I use the actual file name of the gsm recording,

i have a customer that needs to plug the phones into the pc's using the pass-through rj45 available on most sip phones the question they are asking me is how to keep the data network separate from / secure from the voip network i understand they can set up vlans but i am hazy on a few details 1 since the phones are plugged into the pc's how will the phones be segmented into their own

Hi, I was planning on getting a Dell PowerEdge 2950 for our new Asterisk configuration. But while searching for documentation about it and/or reported issues, I found this: http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, which has been known to cause random locksup - if you plan on using a Dell server, disable the

Hi List, Does anyone know of a good dual wan router that can handle SIP well and can failover between connections if there is a SIP issue on one of the lines (meaning there still is a connection however there isnt enough bandwith or sip packets arent going thru etc.) ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've been searching the Internet for information regarding the replacement of astdb with a modern sql engine. There are several reasons one would like to do this. First of all, external applications have a hard time reading/writing to the now-old astdb format. Also (and this is what interests me most), the sql astdb could easily be clustered throughout several servers (I'm looking for a

Asterisk Users, I am running Asterisk-1.4.11 on a Debian "Etch" system. On an occasion, when customer calls into my Asterisk Box, I get this error messagefrom Asterisk "CallerID returned with error on channel Zap/3-1" , causing all my zap channels to be busy. So, I cannot make any calls in, nor out. I am located in the United States. Is there any other suggestions,

I'm trying to find a way to get remote connections from my Linux Desktop machine to a remote Linux box which is hosting a MySQL database, via SSH port 22, and then once connection is established, log in to the database port 3306 and have either an SSH and or SCP connection established so I can securely do edits and queries. I am able to connect this way from my Windows machine to remote

Hey everyone, just wondering if there are any other people on the list in Omaha, NE that would be interested in getting together and chatting about rails. Cheers, - Matt -- Matt Secoske http://www.secosoft.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://wrath.rubyonrails.org/pipermail/rails/attachments/20060226/93c1467b/attachment.html

fit lwr upr 1 218.4332 90.51019 346.3561 2 218.3906 90.46133 346.3198 3 218.3906 90.46133 346.3198 4 161.3982 44.85702 277.9394 5 192.4450 68.39903 316.4909 6 179.8056 56.49540 303.1158 7 219.5406 91.52707 347.5542 8 162.6761 46.65760 278.6945 9 193.8506 70.59838 317.1029 10 181.3816 58.11305 304.6502 11 221.2871 92.14366 350.4305 12 164.2947 47.91081 280.6785 13

Hi all:I found some great demonstrations of interactive presentation graphics generated in R with the SVGAnnotation package, here:http://www.omegahat.org/SVGAnnotation/http://www.omegahat.org/SVGAnnotation/SVGAnnotationPaper/SVGAnnotationPaper.htmlI tried to install the package available at that website (it's not on CRAN) and am getting some pretty uninformative errors (see below). My best

The best place for Astri Con 2006 would definatly be Omaha, Nebraska! ;) very central ...ah one could hope. __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

Running Asterisk 1.2.8 kernel 2.6.13.4-1. Everything in my dialplan seems to be working well except for one problem. When calls are blind transferred to an invalid extension I would like the call to go to the operator on ext 1000? What is the best way to do this? Thanks in advance Here's a snippet of my extensions.conf [default] exten=>_10XX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})

Hello! Here are my specs. Slackware 7.1 OpenSSL 0.9.6a Samba 2.2.0 downloaded today It claims that it cant find ssl.h Here is my configure line: ./configure --prefix=/usr --with-ssl --with-sslinc=/usr/local/ssl/include/openssl --with-smbmount --enable-debug I verified that ssl.h is in /usr/local/ssl/include/openssl Any ideas? Thank you -- Terry Davis Systems Administrator BirdDog

Is there any technical difference between a T1 cable and a cat5e patch cable as far as using them with Digium T1/E1 cards? Can PRI circuits terminating at a smart jack connect successfully to Digium cards using straight through CAT5e cables? If so, are they using all of the pins in the cable? Thanks in advance

How does one move from a rpm based samba to tar.gz? In other words how do I upgrade from samba-2.2.1a-4 to samba-2.2.6.tar.gz. Do I remove samba : rpm -e samba? And what do I do with the other samba rpm packages: samba-client-2.2.1a-4 & samba-common-2.2.1a-4? Are they included with the tar package? Using RH7.2. Roger -- ********************************** Roger Schmeits System

The Houston RoR group meeting is off for tonight (Aug. 8) due to some issues with the venue for the meeting that came up at the last minute. We are rescheduling for next Tuesday, the 15th. Sorry for the inconvenience and late notice. Keith Lancaster http://groups.yahoo.com/group/Houston-RoR/ -- Posted via http://www.ruby-forum.com/.