similar to: Cisco 7940 Phones as paging system?

There was a thread a few months ago that tossed around some ideas for using a cisco phone for intercom or paging. I don't have any ip phones, and wondered if anyone had any luck getting intercom or paging to work on the cisco units. Do any of the (cheaper) ip phones have a way to support intercom or paging? I presume that it's not part of the SIP or IAX protocols. Chris.

> > No, this is not the case currently with any of the Cisco SIP software > loads that I am aware of. If you find this to be incorrect, please > let the list know. Cisco has not deployed much of the featureset in > their SCCP phones (such as paging/intercom) into the SIP phones due > to lack of standards/interest/political capital. > > JT Ok, after further

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

I've been digging around with some cisco engineers for about a week & I finally got an encouraging response to the Auto-Answer issue with the SIP Phones. Here is their reply: =============== == FROM CISCO == =============== Auto-Answer feature is introduced in SIP IP Phone 6.0 version. This software version is expected to be available for customers shortly. Please let me know if you

Is there anyway to force an autopickup on a cisco 7940 / 60 from the dialplan ? My problem is that I am originating a call from the AMI, with the internal user being called first, and then connecting to external user. However, sometimes the internal user doesn't pick up the phone, so the call is never placed. I need to know the results of the call so I need to be able to either a) get

Hi, Does anybody knows how to get this work? I have been searching all over and can?t find the way. Using chan_sccp tge phone can log in and has line tone. But the problem is that it is not able to receive nor dial out. Any help would be apprecieted. Thanks Javier --------------------------------- Sell on Yahoo! Auctions - No fees. Bid on great items. -------------- next part

Hi ! We have Cisco 7912 phones, and the doc says that I can "create up to four speed dial buttons on my phone using the Cisco CallManager". Does anyone knows which protocol is used to configure speed dials (Is it documented somewhere) ? Did someone tried to reverse engineer the protocol ? It would be cool, not having to pay $15000 just for configuring speed dials on those phones ;-D

I want to implement a one way announcement and paging facility using Asterisk and Sipura phones. The wiki says Sipura phones only support Auto Answer using the Call-Info header which is no lone shipped with asterisk stable since 1.0.4. I would like to ask if anyone has implemented a similiar facility using Sipura SPA-841 or any other SIP phones. If I could take a look at how

What models? Jeff Noxon (jeff-asterisk at planetfall.com) wrote*: > >I just purchased a bunch of Nortel Meridian POTS phones that support >intercom on the 3rd pair. I intend to get it working with Asterisk. >The phones support MWI, have a 3-line display, callerID, call waiting >callerID, 2 lines...very nice. > >On Thu, Feb 27, 2003 at 01:07:19AM -1000, James H. Thompson

Hello, I'm considering using Asterisk with some type of Cisco phone, and currently considering either the 7940 or 7960 because of its more-complete functionality (compared to the 7905). I'm currently wondering: Do all the expected functions (transfer, conference, voice mail, message waiting indicator, etc.) work normally with Asterisk over SIP? What caveats are known about using

I am using chan_sccp on bristuffed asterisk (0.2RC3 on asterisk 1.0.3). Things seem fine but I am seeing some weird stuff. I have a Kirk IP600 connecting to * with 2 handsets. The weird thing is that for incoming calls the handset that is put second as my dialstring, never rings. This is my dial string: exten => 666666666,4,Dial(${PHONE1}&${DECT1}&${DECT2}),25,tm) Where DECT1

Hello all, I noticed that Cisco claims that you can do station to station intercom with the 7940/7960 phones, and the Cisco Call Manager. Does anyone have an example SIP header that shows this in action? Or is it something else that triggers the intercom? I would like to add this in to *. Thanks, Mike

Greetings- As many of your are Polycom "experienced", I was hoping some kind soul could provide direction on a specific issue. On a system running Asterisk 11.11.0 (and latest FreePBX), I'm finding an instance where, using intercom/paging functionality of FreePBX, I need to override an end user's 'Do Not Disturb' selection on the handset. By default, DND simply

Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot, plus asterisk would be taking a single input stream and exploding it out to 500 endpoints. - There are 500

Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it's worth my time

I see a lot of chatter in the archives about intercom and paging, but has anyone addressed zone paging? Each of the 50 telephones in a large clinic would be members of one or more paging zones. Someone could then page Dr. X in zone #1. Would this be possible with analog phones? SIP? Thanks, Mike

I am having a problem with Polycom auto-answer. I have the auto-answer working between PhoneA and PhoneB, but when I try to use the intercom between more then one phone I start having problems. PhoneA dials *3 which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only one will pick up, the rest will hang up and I get this error on Asterisk: Got SIP response 500 "Internal Server

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've been able to page to a specific phone (intercom type of thing), but I'd like to have a macro or agi that pages all phones but first checks if their on the phone. It looked like there used to be a pageall.agi type of script on the wiki, but that link isn't valid anymore. Does anyone have that script, or something else that would work? I would just do SIP/1000&SIP/1001, but

Hi, I'm testing paging using snom 360. Can someone correct my dialplan? Regards, Jason. ================================================== ;exten => _99XXXX,1,SIPAddHeader(Call-Info: Answer-After=0) ;exten => _99XXXX,n,SIPAddHeader(Call-Info: <sip:192.168.1.113>\;answer-after=0) ;exten => _99XXXX,n,Dial(SIP/${EXTEN:2}) exten => _99XXXX,1,Set(__SIPADDHEADER=Call-Info: