similar to: Echo Cancelation on TE110P

Has anyone got any information on bulk provisioning of Linksys SPA-941/94s? There is an overview in the admin guide but it refers to a different provisioning guide that I haven't found anywhere. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com

Is there any setting in the voicemail that will send the voicemail file in a type that is recognized on a Blackberry? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com <http://www.techdatapros.com/> http://www.techdatapros.com --------------

I am ready to pull my hair out. I cannot seem to get the Polycoms to read the time properly. Regardless of the server they are pointed to our the offset, i am getting the correct time, but 24 hours ahead. So for today it is showing Friday April 28 but with the correct time. Any clues? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider

I dont recall the SPA-941 playing a stutter tone in the previous firmware but it is driving me nuts, anyone know where to turn it off? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com <http://www.techdatapros.com/> http://www.techdatapros.com

We are trying to setup a sip connection behind a Watchguard Firebox 1000 and it is simply put...not working. The ports are all forwarded but the packets are not going out. It is as if the firewall simply ignores SIP packets. Has anyone seen this or have any idea what the issue could be? Watchguard so far has been of zero help. Kerry Garrison Director of Technical Services Tech Data Pros -

i have a client that wants a function that will automatically logout an agent from a queue if they do not answer a call. This would prevent future calls from being sent to that phone if the agent forgot to logout. Any ideas? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com>

Has anyone had any experience connecting Asterisk to Cbeyond's SIPconnect service (http://www.sipconnect.info)? Any opinions? Thanks, -Brian

This is a system for our lab. I have no problem getting rid of X100P clones. But I am just curious why can they work. Even the drivers are not loading correctly. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kerry Garrison Sent: Tuesday, May 02, 2006 10:51 AM To: 'Asterisk Users Mailing List -

Has anyone actually used these USB speakerphones http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_ W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewIte m Seems to get a pretty good review here http://voipspeak.net/index.php?option=com_content&task=view&id=39&Itemid =27 But looking for real world feedback. Cheers,

Hi, I am renovating my house completely and installing new cabling for communication. I'm not to into this PBX thing but I would like to have a simple one for my house, to have different phone numbers for my family members, some kind of integration with my Broadband telephone, and possibly Skype.... Browsing the digium website didn't make me much wiser. I have a linux box that I can

I am trying to implement a follow-me script (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) but I am having a brain fart as I haven't a clue where to get started with what to do with this. From my main menu, I want the extension 300 to execute the script as follows: exten => 300,1,dial(sip/200,20) exten => 300,2,playback(pls-wait-connect-call) exten =>

I am planning to deploy Asterisk Business edition. Does this edition have a web module administration? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060510/7295b78f/attachment.htm

Hi, I have lookedaround and don't see this anywhere. Is there a way to tell the ip500 to not make the aural MWI blips?

Does anyone know if a TDM2400 will fit into a Dell 2850? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com <http://www.techdatapros.com/> http://www.techdatapros.com -------------- next part -------------- An HTML attachment was scrubbed...

On a TDM2400 with 3 FXO modules, is there a way to split each line into basically being its own trunk or another way to pull off the following scenerio: PBX has 12 inbound PSTN lines 1,3,5,7 are the 714 phone number hunt group 2,4,6,8 are the 888 phone number hunt group 9-12 are fax lines Customer wants outbound calls to go out in the following order: 8,7,6,5,4,3,2,1,12,10,11,9 Kerry

I have written up an guide on how to do bulk provisioning of the Linksys phones and ATAs. http://voipspeak.net/index.php?option=com_content <http://voipspeak.net/index.php?option=com_content&task=view&id=73> &task=view&id=73 Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 -

I'm attempting to use callprogress in my system, and I'm having trouble. Callprogress always can tell if the line is busy or ringing, but when the line is answered, the call does not get bridged. Messages showing that "line is ringing" stop in the console and if the called party hangs up, asterisk reports the line is busy. Are there any settings that I could use to help with

Hi, I'm new in asterisk world. I have questions. For example I have my server with public IP address, but two customer with softphone in a private network. How can I do to make them work with the asterisk server? Best Regards -- Ever Zalazar -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All Has anyone come across a handset that can somehow replace FOP? Some users don't like FOP unless it is on a dedicated PC. Thanks Garth

Hi, I'm interested in developing an automated phone survey and am curious if Asterisk could be configured to run such a system.. My idea is to record a message and a series of sub-questions. The system would call each number on a list and play the message, Depending on the touch tone response another message would be played. Is it possible for asterisk to manage a survey like this?