similar to: 2 way intercom recommendation for restaurant kitchens

On Tue, 4 Jan 2011, Andy Graybeal wrote: >> The Polycom 321 has not been EOL'd and supports VLAN. It is, however, >> lacking a 2nd ethernet port if you were to go that route. >> >> -M >> > Thanks for the response Mark. I see the 331 has two ports and the same > features as the 321. > > I'm wondering what phone would be best being used as an

Greetings, I mailed the list regarding an intercom system some months ago and we came to the conclusion that I should purchase a Polycom 501 phone. I'm now considering the purchase for this year, and I'm now wondering between the Polycom 501 and the 320 for the intercom. I don't need the spare ethernet on the phone because I would like to have my voice network separate from my

Hi, I have two objects: Contacts Lists I want to be able to add multiple contacts to multiple lists. I''ve created a linking table and a form that allows me to select the contacts using checkboxes, but I''m having some problems with the controller and model. I receive the following error message when I submit the form: Mysql::Error: Cannot add or update a child row: a

Wickes Kitchens , Go to www[.]wickeskitchensreviews[.]co[.]uk Wickes Kitchens ________________________ www[.]wickeskitchensreviews[.]co[.]uk -- Posted via http://www.ruby-forum.com/. -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To unsubscribe from this group and stop receiving emails from it, send an email to

Do you need a place to stay? Food to eat? 4000+ Shelters and soup kitchens in all 50 states: http://groups.google.com/group/homeless_shelters?hl=en --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to

is anyone using a multi-seated configuration with CentOS? I'm thinking about buying a two 2 headed nvidia card to run 4 separate monitors and 4 separate keyboards and mice. I don't mind using either proprietary or opensource nvidia drivers. I'm looking for more information regarding centos and multiseated configurations. I've read a bit about fedora and ubuntu with

I am attempting to port an operating system project to use Clang/LLVM instead of GCC, but I'm having issues with Compiler-RT. Right now, the OS is being cross-compiled on an x86_64 host targeting ARMv7a. I built the LLVM toolchain according to the instructions, though I limited the supported targets to arm, x86, and x86_64 to reduce compilation time. I'm finding that compilation proceeds

Hi, Does anybody know of a VoIP compatible doorbell or door intercom unit? I've contemplated buying a cheap SIP phone, ripping it apart, and putting it inside an IP66 sealed unit... It would need: - At least one speed-dial key, or some way to make every button dial the same extension number - PoE (power over ethernet), so I can power it off the central switch - cheap enough to rip apart

Hello: I would like to know if anybody has test various usb phones with an usb hub conected to an asterisk server. The idea is to build a public telephony site with one or 2 analog cards 4 or 8 FXO, least cost routing and billing and usb handsets with dialpad for users, so I really don't need a graphical interface for the soft phone ( the users just dial from dialpad). Any has done this

Hello list, this works : exten => _*XXX*,n,SIPAddHeader("Call-Info:\; answer-after=0") exten => _*XXX*,n,Dial(SIP/${SIPACCOUNT}) The phone auto-answers the call... this does not work : exten => _*XXX*,n,SIPAddHeader("Call-Info:\; answer-after=0") exten => _*XXX*,n,Page(SIP/${SIPACCOUNT}) The phone rings and does not auto-answer the call... Can you tell me

Hey, Hi, All, We have a few dozen of the Kirk (ie: Polycom bought this European brand) out there & most all work very well & work very well with most all versions of Asterisk. But we have been tripped-up by one combination of firmware & version & configuration variables. We are running Asterisk 1.4.23.1 (TrixBox CE). We are running latest stable firmware on the handsets. Most

A friend pointed out this url http://www.cisco.com/univercd/cc/td/doc/pcat/clmn32.htm where it lists intercom/auto-answer as being a feature in Cisco Call Manager (which as I understand it, uses SIP predominately for handsets). I've come across comment somewhere that intercom isn't supported in the SIP spec. Does anyone know if the apparent capability of Intercom being available in SIP

How would I go about setting this up. I have a few 7960's with an extension set to autoanswer. How do I let all extensions answer and be active? Thanks, Will -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031110/f17a2971/attachment.htm

I wasn't thinking of using the conference system as the basis. I was thinking more along the lines of: 1) Setup a second extension on the Cisco phone named "INTERCOM" enabled for auto-answer 2) Create a call group on asterisk to dial that "INTERCOM" extension on every phone that will participate 3) Add a feature code that would dial the intercom extension and connect

iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X systems. Sources included in the iaxclient library: http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz Precompiled binaries at: http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz Features: * Register with multiple servers (ie enterprise server and iaxtel).

Hi there, Just wondering if there is a way to speak to the person you are transferring a call to before actually connecting them to the incoming call. E.g. "Hi, Colleague, I've got Bill from Microsoft on the line here... putting you through now" Then actually transfer the call. Does that make sense!? -- Steve Foy | http://www.unite.net UNITE Solutions | Tel: 028 9077

Hey guys, I have run into one last issue before I do my full * conversion this evening. I can't seem to get paging to work. I have the chan_oss module loaded as per the wiki, and I have the following in my dial plan ;here is our intercom exten => 6000,1,Dial,console/dsp when I dial it here is the output from the console -- Executing Dial("SIP/3062-4f07",

I'm replacing a Merlin for a client and they have a PagePal Intercom that I would like to reuse. Here is what I know about it: It has a screw-down wires that goto rj-11 (This was told to me over the phone) that went into one of the Merlin ports. I tried bring it up with fxo_ks and fxo_ls (assuming it was analog and autoanswered) but no luck. I would be happy to replace if anyone knows of

I am using Console/DSP for an intercom. I want to play my MP3 collection over it when no one is using it, like when they do in the supermarket. Can anyone help me with this. Any suggestions will be appreciated. -- Christopher Dobbs

OK, I got my Cisco 7960's to auto-answer on the second line but I can't get the Asterisk to call all the lines at one time. I have 4 phones I would like all of then to answer when I dial x300. Any help would be great Thanks Tuska extensions.conf [conference] exten => 300,1,AGI(callall) exten => 300,2,MeetMe(300,dtqp) ; press # to exit the conference exten =>