You would already have a door entry system with a relay for the door to be control in-place.
Once that is done, you could add a native sip device (in my case I also want it to be POE). It would register as a local line in sipx. There are different models available from various vendors. Cyberdata claims to be compatible with sipXecs on their website. The device is programmed from its own web-ui, and is programmed to dial a specific number when the call button is pressed (in my case it might just be a hung group within the building). Someone picks up and there would be a conversation where the caller would say "UPS with a delivery for someone.", at which point the person who answered the call could press a pre-programmed number (like "9") which unlocks the door. The outside device would receive the DTMF as RFC-2833, recognize "9" as the legitimate code, and tell the relay, which is in the call box outside and signal the door controller system to unlock the door. I was just looking to see if anyone had a success story I could follow already before I jumped into it blind. >>> Cuneyt M <[email protected]> 01/22/09 10:58 AM >>> Hi Mike and Tony, The door-relay idea sounds very interesting. We have polycom phones and Mediant 1000 with ISDN and analog interfaces(where most analog interfaces (fxo) are currently not used). Polycomphones dial out and receive calls from ISDN lines. How do you do a ring-down in such setup with a polycom (say 330) phone? >>If you have analog ports on a gateway available, almost any ofthe </[email protected]><pre wrap="">>>analog solutions should work. Press the button, the phone goes>>off-hook, have the gateway do automatic ring-down to a SIP extension.>>Once in a conversation usually pressing # will send DTMF which the Door>>Phone device will hear and fire a relay. How do you setup the gateway to do automatic ring-down to a SIPextension? Googling didnt return any valuable info. Would appreciate if you can clarify the ring_down setup and how to gowith this solution in Polycom 330 with Mediant 1000 FXO ports available. Thanks in advance. Today's Topics: 1. Re: Integration with door access control systems (Picher, Michael) 2. Re: Integration with door access control systems (Picher, Michael) 3. 3.11.9 questions (dimitris(yahoo)) 4. Dial plan issue - internal extensions (Mitchell, Kenny (Ineos)) 5. Re: Dial plan issue - internal extensions (Picher, Michael) Subject: Re: [sipx-users] Integration with door access control systems From: "Picher, Michael" <[email protected]> Content-Type: text/plain; charset="us-ascii" Message: 1 Subject: Re: [sipx-users] Integration with door access control systems From: "Picher, Michael" <[email protected]> Content-Type: text/plain; charset="us-ascii" Message: 2 Subject: [sipx-users] 3.11.9 questions From: "dimitris\(yahoo\)" <[email protected]> Content-Type: multipart/alternative;boundary="----=_NextPart_000_002F_01C97CB6.798BD8B0" Message: 3 Hi everybody,<o:p></o:p> I haveinstalled (just for curiosity reasons), the release3.11.9 of sipXecs and I have noticed the following :<o:p></o:p> 1. After anormal registration of a hard phone orsoftphone , the operator or the AA cannot be reached , but thevoicemailfunction is operational.<o:p></o:p> 2. After myregistration and deregistration of acouple of hard – soft phones many times, I realized that someextensions wereregistered more than once with different times of expiration…<o:p></o:p> 3. Are NATproblems solved in this release??? BecauseI registered with softphones behind different NAT’s , and onlysometimesthe signaling was functional , not the RTP….<o:p></o:p> <o:p> </o:p> automatic remotephone configurationthrough ftp ?<o:p></o:p> <o:p> </o:p> PS. The work that is done tillnow regarding sipXecsis just GREAT.<o:p></o:p> Subject: [sipx-users] Dial plan issue - internal extensions From: "Mitchell, Kenny \(Ineos\)" <2780c2f0c716ad4080c32e7ae34b6b41015a2...@in1cplvex003.in1.ad.innovene.com> Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C97CA5.F1917E0C" Message: 4 I'm currently trying to set up a Sip-X3.10.2 system at work to evaluate the possibility of using it toreplace our traditional switches. We have a number of Siemens and iSDXswitches on site, so I am using a Mediant 1000 gateway with an FXO cardin it to give myself a tap into the local extensions. I havedownloaded the UK English localization pack and installed it, thenenabled the dial-plans that come with it and I can quite happily callthe outside world. Since I'm going through a hard-wiredswitch I have to dial a zero for an outside line, which is no hardship,but I want to be able to call local extensions on our internal switcheswithout the need for a prefix. I currently have to dial a 9 to get theswitch to route to the gateway and the internal 4-digit extensions, andI have no clue which rule is handling this. I have tried to set up aplan whereby any 4-digit number starting with a 3,5,6,7 or 8 is routedto the gateway, but all I get is the dreaded "that extension is notvalid". How do I get Sip-X to send a 4-digit number to the gateway viaa dial-plan, as none of mine appear to work. Help! Kenny Mitchell Infrastructure Analyst Ineos Refinery 01324 478305 Subject: Re: [sipx-users] Dial plan issue - internal extensions From: "Picher, Michael" <[email protected]> Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C97CA6.E2438A9B" Message: 5 <o:shapedefaults v:ext="edit" spidmax="1026" /> <![endif]--> <o:shapelayout v:ext="edit"> <o:idmap v:ext="edit" data="1" /> </o:shapelayout><![endif]--> HiKenny,<o:p></o:p> <o:p> </o:p> Youcan setup a custom dial rule to make this happen. In thecustom dial rule make sure you have your gateway in there and have thedail planentry enabled..<o:p></o:p> <o:p> </o:p> Afteryou create the dial plan entry, don’t forget to activatethe dial plan again.<o:p></o:p> <o:p> </o:p> Mike<o:p></o:p> <o:p> </o:p> mailto:[email protected]] On Behalf Of Mitchell,Kenny (Ineos) Sent: Thursday, January 22, 2009 10:28 AM To: [email protected] Subject: [sipx-users] Dial plan issue - internal extensions<o:p></o:p> <o:p> </o:p> <o:p> </o:p> I'mcurrently trying to set up a Sip-X 3.10.2 system at work to evaluatethepossibility of using it to replace our traditional switches. We have anumber of Siemens and iSDX switches on site, so I am using a Mediant1000gateway with an FXO card in it to give myself a tap into the localextensions. I have downloaded the UK English localization pack andinstalled it, then enabled the dial-plans that come with it and I canquitehappily call the outside world.<o:p></o:p> SinceI'mgoing through a hard-wired switch I have to dial a zero for an outsideline,which is no hardship, but I want to be able to call local extensions onourinternal switches without the need for a prefix. I currently have todiala 9 to get the switch to route to the gateway and the internal 4-digitextensions, and I have no clue which rule is handling this. I havetriedto set up a plan whereby any 4-digit number starting with a 3,5,6,7 or8 isrouted to the gateway, but all I get is the dreaded "that extension isnotvalid". How do I get Sip-X to send a 4-digit number to the gatewayvia a dial-plan, as none of mine appear to work. Help!<o:p></o:p> KennyMitchell InfrastructureAnalyst IneosRefinery 01324478305 <o:p></o:p> http://list.sipfoundry.org/mailman/listinfo/sipx-users
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