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). Polycom phones 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 of the

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 SIP extension? Googling didnt return any valuable info.

Would appreciate if you can clarify the ring_down setup and how to go with this solution in Polycom 330 with Mediant 1000 FXO ports available.


Thanks in advance.

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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]>
Date:
Thu, 22 Jan 2009 09:39:21 -0500
To:
"Tony Graziano" <[email protected]>, <[email protected]>

To:
"Tony Graziano" <[email protected]>, <[email protected]>

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Right...

Did you check Keri too?

-----Original Message-----
From: Tony Graziano [mailto:[email protected]]
Sent: Thursday, January 22, 2009 9:34 AM
To: Picher, Michael; [email protected]
Subject: Re: [sipx-users] Integration with door access control systems

Yes, but I wanted to be able to press a button on the phone that
answers to trigger the door to open. A standard gateway doesn't have
the relay built in which is why I was looking at products that are
made
to open doors at the same time.
-----Original Message-----
From: "Picher, Michael" <[email protected]>
To: Tony Graziano <[email protected]>
To:  <[email protected]>

Sent: 1/22/2009 9:26:29 AM
Subject: RE: [sipx-users] Integration with door access control systems

You can also do automatic ring-down from a Polycom phone or from an
analog port on a Patton gateway...



From: [email protected]
[mailto:[email protected]] On Behalf Of Tony
Graziano
Sent: Thursday, January 22, 2009 9:09 AM
To: [email protected]
Subject: [sipx-users] Integration with door access control systems



I have been looking at a couple of different products that have a sip
based "call" button for use at an external door that could call a hunt
group (UPS delivery, etc.) and allow someone to pickup the phone and
press a button which would trigger a relay to open/unlock a door.

I have looked at both the Cyberdata and Aleen products:

http://www.cyberdata.net/products/voip/voip-intercom.html

http://www.its-tel.com/info.asp?id=48

Has anyone done either of these? Does anyone have other success with
this type of integration?

Tony



------------------------------------------------------------------------

Subject:
Re: [sipx-users] Integration with door access control systems
From:
"Picher, Michael" <[email protected]>
Date:
Thu, 22 Jan 2009 09:47:42 -0500
To:
"Picher, Michael" <[email protected]>, "Tony Graziano" <[email protected]>, <[email protected]>

To:
"Picher, Michael" <[email protected]>, "Tony Graziano" <[email protected]>, <[email protected]>

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2


Actually meant Valcom...

http://www.valcom.com/Home_links/sipdoorintercom.htm

I don't think Keri has a native SIP solution.

If you have analog ports on a gateway available, almost any of the
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.

Mike

-----Original Message-----
From: [email protected] [mailto:sipx-users-
[email protected]] On Behalf Of Picher, Michael
Sent: Thursday, January 22, 2009 9:39 AM
To: Tony Graziano; [email protected]
Subject: Re: [sipx-users] Integration with door access control systems

Right...

Did you check Keri too?

-----Original Message-----
From: Tony Graziano [mailto:[email protected]]
Sent: Thursday, January 22, 2009 9:34 AM
To: Picher, Michael; [email protected]
Subject: Re: [sipx-users] Integration with door access control
systems
Yes, but I wanted to be able to press a button on the phone that
answers to trigger the door to open. A standard gateway doesn't have
the relay built in which is why I was looking at products that are
made
to open doors at the same time.
-----Original Message-----
From: "Picher, Michael" <[email protected]>
To: Tony Graziano <[email protected]>
To:  <[email protected]>

Sent: 1/22/2009 9:26:29 AM
Subject: RE: [sipx-users] Integration with door access control
systems
You can also do automatic ring-down from a Polycom phone or from an
analog port on a Patton gateway...



From: [email protected]
[mailto:[email protected]] On Behalf Of Tony
Graziano
Sent: Thursday, January 22, 2009 9:09 AM
To: [email protected]
Subject: [sipx-users] Integration with door access control systems



I have been looking at a couple of different products that have a
sip
based "call" button for use at an external door that could call a
hunt
group (UPS delivery, etc.) and allow someone to pickup the phone and
press a button which would trigger a relay to open/unlock a door.

I have looked at both the Cyberdata and Aleen products:

http://www.cyberdata.net/products/voip/voip-intercom.html

http://www.its-tel.com/info.asp?id=48

Has anyone done either of these? Does anyone have other success with
this type of integration?

Tony

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------------------------------------------------------------------------

Subject:
[sipx-users] 3.11.9 questions
From:
"dimitris\(yahoo\)" <[email protected]>
Date:
Thu, 22 Jan 2009 17:25:48 +0200
To:
<[email protected]>

To:
<[email protected]>

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Message:
3


Hi everybody,

I have installed (just for curiosity reasons), the release 3.11.9 of sipXecs and I have noticed the following :

1. After a normal registration of a hard phone or softphone , the operator or the AA cannot be reached , but the voicemail function is operational.

2. After my registration and deregistration of a couple of hard -- soft phones many times, I realized that some extensions were registered more than once with different times of expiration...

3. Are NAT problems solved in this release??? Because I registered with softphones behind different NAT's , and only sometimes the signaling was functional , not the RTP....

And a general question : Is there any big problem , if I use only DHCP server , without using DNS resolution ? Is there going to be any problem with the automatic remote phone configuration through ftp ?

PS. The work that is done till now regarding sipXecs is just GREAT.


------------------------------------------------------------------------

Subject:
[sipx-users] Dial plan issue - internal extensions
From:
"Mitchell, Kenny \(Ineos\)" <[email protected]>
Date:
Thu, 22 Jan 2009 09:27:31 -0600
To:
<[email protected]>

To:
<[email protected]>

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Message:
4



I'm currently trying to set up a Sip-X 3.10.2 system at work to evaluate the possibility of using it to replace our traditional switches. We have a number of Siemens and iSDX switches on site, so I am using a Mediant 1000 gateway with an FXO card in it to give myself a tap into the local extensions. I have downloaded the UK English localization pack and installed it, then enabled the dial-plans that come with it and I can quite happily call the outside world.

Since I'm going through a hard-wired switch 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 switches without the need for a prefix. I currently have to dial a 9 to get the switch to route to the gateway and the internal 4-digit extensions, and I have no clue which rule is handling this. I have tried to set up a plan whereby any 4-digit number starting with a 3,5,6,7 or 8 is routed to the gateway, but all I get is the dreaded "that extension is not valid". How do I get Sip-X to send a 4-digit number to the gateway via a 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]>
Date:
Thu, 22 Jan 2009 10:34:14 -0500
To:
"Mitchell, Kenny (Ineos)" <[email protected]>, <[email protected]>

To:
"Mitchell, Kenny (Ineos)" <[email protected]>, <[email protected]>

Precedence:
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References:
<2780c2f0c716ad4080c32e7ae34b6b41015a2...@in1cplvex003.in1.ad.innovene.com>
In-Reply-To:
<2780c2f0c716ad4080c32e7ae34b6b41015a2...@in1cplvex003.in1.ad.innovene.com>
Message-ID:
<[email protected]>
Content-Type:
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5


Hi Kenny,

You can setup a custom dial rule to make this happen. In the custom dial rule make sure you have your gateway in there and have the dail plan entry enabled..

After you create the dial plan entry, don't forget to activate the dial plan again.

Mike

*From:* [email protected] [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

I'm currently trying to set up a Sip-X 3.10.2 system at work to evaluate the possibility of using it to replace our traditional switches. We have a number of Siemens and iSDX switches on site, so I am using a Mediant 1000 gateway with an FXO card in it to give myself a tap into the local extensions. I have downloaded the UK English localization pack and installed it, then enabled the dial-plans that come with it and I can quite happily call the outside world.

Since I'm going through a hard-wired switch 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 switches without the need for a prefix. I currently have to dial a 9 to get the switch to route to the gateway and the internal 4-digit extensions, and I have no clue which rule is handling this. I have tried to set up a plan whereby any 4-digit number starting with a 3,5,6,7 or 8 is routed to the gateway, but all I get is the dreaded "that extension is not valid". How do I get Sip-X to send a 4-digit number to the gateway via a dial-plan, as none of mine appear to work. Help!

*Kenny Mitchell*
/Infrastructure Analyst/
/Ineos Refinery/
/01324 478305/

------------------------------------------------------------------------

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