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88fingerslukee

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Are there plans to offer phone-to-phone paging/intercom? I'm referring specifically to the ability to dial the extension and have the speakerphone automatically answer.

It seems somebody else asked about this in February but there was no response.

thanks!
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88fingerslukee

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Well, I looked into this a bit more and it seems that this is a phone-level functionality. The Snom 8XX versions support this and those are the ones we are going with.

Sorry for the waste of time.
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PC_MAC

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Does this Paging work for only Phone-to-phone, or have you found a way to have it work with Phone-to-Speakers somewhere in the building? I'd like a way to tie it into the speaker system we have in the office, as our old phone system used to be able to do.

Any thoughts guys?
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Vladimir Toncar (Kerio)

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Hi,

Have a look at Snom PA1, it's a public address device with a SIP client.

Vladimir
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PC_MAC

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Thanks, That might just do it.

Adam
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sourceminer

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I know that Asterisk has this feature built into the product.
I would post a url but I cant being I have not posted more than 5 messages LOL.

Freepbx has this, Etc.

Its not just the phone to phone part its the lack of paging groups and announce groups.
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sourceminer

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I know this has been a feature request since 1.0. Would be great to use this product in place of the free ones Wink
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Vladimir Toncar (Kerio)

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@sourceminer,

Are you talking about sending instant messages to phones or groups of phones? I think you could send the URL to me in a private message.

Vladimir
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sourceminer

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Not sure about the term "Instant Message"
I am talking about paging/Intercom.

Asterisk has this function built into the version 1.8

So here is the feed I am reading on voip-info

New in Asterisk 1.8: A new RTP engine and channel driver have been added which supports Multicast RTP.
The channel driver can be used with the Page application to perform multicast RTP paging. The dial string format is:

MulticastRTP/<type>/<destination>/<control address>

Type can be either basic or linksys. Destination is the IP address and port for the RTP packets. Control address is specific to the linksys type and is used for sending the control packets unique to them


Example

[macro-page]
; Paging macro:
; Check to see if SIP device is in use and DO NOT PAGE if they are
; ${ARG1} - Device to page
;
exten => s,1,ChanIsAvail(${ARG1}|js) ; j is for dump and s is for ANY call
exten => s,2,Set(_ALERT_INFO="RA") ; This is for the PolyComs
exten => s,3,SIPAddHeader(Call Info
Answer-After=0) ; This is for the Snoms and Others
exten => s,3,SIPAddHeader,Call-Info
sip:192.168.20.1\;answer-after=0
exten => s,3,SIPAddHeader(Call-Info:<sip:domain>\;answer-after=0) ; enter your domain
exten => s,4,NoOp() ; Add others here
exten => s,5,Dial(${ARG1}||)
exten => s,6,Hangup
exten => s,102,Hangup

[page] ; Paging context
exten => 202,Macro(page,SIP/polycom)
exten => 208,Macro(page,SIP/cisoo1aa)
exten => _X.,1,Macro(page,SIP/${EXTEN})

The line below goes in the context where you have your extensions:

exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten => 7999,2,Page(Local/202@page&Local/208@page&Local/210@page/n&Local/interal 223@page|)


Example2

This works for Linksys SPAXXX and Snom phones. (confirmed working with Asterisk 1.2.7.1, Linksys SPA941, SPA942 & Snom 360. May 29, 2006)

It implements both paging and intercom. Other phones would work as well but you would have to adjust the SIPAddHeaders for your brand of phone. NOTE: The Linksys SPAXXX phones already have *96 assigned so if you are going use *96 in Asterisk don't forget to first remove *96 from the phones Advanced Regional settings first! (The built in Paging feature of the Linksys phones only works with the SPA9000 so its safe to reuse it)

How to use it: Users pick up phone and dial *96. They hear a beep. Dial the extension of the person you want to intercom with OR dial * to page all phones.

exten => *96,1,Goto(intercom,s,1)

[intercom]
exten => s,1,Answer
exten => s,2,Playback(beep)
exten => s,3,Set(TIMEOUT(digit)=5)
exten => s,4,WaitExten(10)

exten => *,1,SIPAddHeader(Call-Info: <sip:10.1.1.171>\;answer-after=0) ; Change 10.1.1.171 to your Asterisk server's IP
exten => *,2,Page(SIP/3218x1&SIP/3219x1&SIP/3220x1) ; add all your devices here

exten => _XXXX,1,SIPAddHeader(Call-Info: <sip:10.1.1.171>\;answer-after=0) ; 4 digit extensions
exten => _XXXX,2,Dial(SIP/${EXTEN})


Here is how I got this to work for my polycom phones.

[page] ; if you cut and paste this make sure you include page under the context where your phones are
exten => *96,1,Goto(intercom,s,1)

[intercom]
exten => s,1,Answer
exten => s,2,Playback(beep)
exten => s,3,Set(TIMEOUT(digit)=5)
exten => s,4,WaitExten(10)

exten => *,1,SIPAddHeader(Alert-Info: Ring Answer)
exten => *,2,Page(SIP/202&SIP/231&SIP/207) ;add all extensions here

exten => _XXX,1,SIPAddHeader(Alert-Info: Ring Answer)
exten => _XXX,2,Dial(SIP/${EXTEN})

I also had to make these changes to the sip.conf file (actually I found this is in the sip.cfg polycom provisioning file, not the asterisk sip.conf file GTM)

<alertInfo voIpProt.SIP.alertInfo.1.value="Ring Answer" voIpProt.SIP.alertInfo.1./>
and
<RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="100" se.rt.4.ringer="11" ; you could also use 7 here

Note

As of Aug 16 2006, the following firmware versions seem to work when using SIPAddHeader(Call-Info: sip:\;answer-after=0) for auto-answer. While using SIPAddHeader(Call-Info: answer-after=0) does work for Grandstream it does not for Aastra or Snom;
Aastra - 480i - 1.4
Grandstream - GXP2000 - 1.1.0.16
Snom - 360 - 6.2.3

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sourceminer

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Freepbx has this code in there version of Asterisk as well they use the feature code *80xxx (xxx being the extension, or group) We use the SPA500 series, and have been using FreePBX but recently started using Operator and really miss this feature!!
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Vladimir Toncar (Kerio)

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Thank you for clarifying. I thought you asked about broadcasting text messages to the displays of SIP phones, that's why I asked about IM. You may propose this as a new feature using the "Suggest idea" button at the splashscreen of Operator's admin GUI.

Vladimir
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88fingerslukee

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I can't get my snom phones to auto-answer the intercom. Supposedly an Alert-info string is passed when you set the Snom Function Key to Intercom but I'm not sure if KO is passing this correctly. The phone still rings on the other end when I press the function key.
See the following website:

http://wiki.snom.com/Category:HowTo:Intercom
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88fingerslukee

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Yeah, I can't get this to work at all. I've tried manually editing extensions.conf to include the lines that sourceminer provided but I can't get them to persist across reboots. They don't seem to do anything when I don't reboot the device and reboot the phones.

Nothing works. I must say, this is a pretty critical feature for us.
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Vladimir Toncar (Kerio)

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The Asterisk configuration files are overwritten after each boot or after you change the configuration so editing the extensions.conf is not a way. You need to wait till we support the intercom or provide a method for user modifications in Asterisk files.

Vladimir
88fingerslukee

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Can you guys please confirm that you are working on the addition of this functionality?
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