[Asterisk-Users] Overhead Paging - app notes
John Todd
jtodd at loligo.com
Fri Nov 14 14:39:50 MST 2003
[apologies for the -dev cross-post, but I feel this might be worthy
of examination by the residents of that less-flooded list]
Multicast is very elegant, but usually unworkable, and mostly
unsupported for SIP devices, so I would avoid going further down that
path.
I think that Dustin has the right idea: an application needs to be
written for both paging and intercom. The "Intercom" application is
really just an app that does what I've been doing with really ugly
call spooling - it creates a semi-dynamic MeetMe room with a bit more
control. The pager is a slight modification on the same application.
I would suggest, however, that it is written not as a specific
solution for SIP but as a solution for all channel types, since
limiting ourselves to just SIP seems to be a bit heavy-handed and not
in the spirit of Asterisk's multi-channel philosophy. By allowing
any channel type, these two apps can support paging over multiple
Asterisk platforms by allowing IAX connections to chain the audio
streams to multiple remote hosts, thus using a "tree" approach to
localize audio storms across WAN segments. Here are my ideal
first-cut manual pages for the two apps...
Page(Technology/resource[&Technology2/resource2...][|options])
Requests one or more channels and places calls to those channels in
a broadcast-only mode. Devices are responsible for ensuring that
appropriate methods are used to transmit the audio to a suitable
broadcast medium, if desired. All resources are dialed, and as soon
as the last resource has answered, a high/low "ready" one will be
played to the caller, indicating that all participants are ready for
the page announcement. If at least one channel/resource answers,
then if either side hangs up the line the application will exit with
-1.
Options:
'A(x)' -- Plays an attention-getting file (usually a set of
tones) over the pager channel before playing the "ready" tones to the
caller.
'q' -- Do not play tones
'w(x)' -- Wait time, in milliseconds, that the application will
wait for the last resource to pick up the line. If w milliseconds
expires but at least one resource has answered, then a low/low/high
"ready" tone series will be played for the caller to indicate that
not all callers are participating in the bridge. If no resources
answer, a low/low/low "not ready" tone will be played, and the
channel will hang up with status 0. Calls that answer after w
milliseconds are added to the audio stream as they answer, but will
not hear the whole announcement.
Intercom(Technology/resource[&Technology2/resource2...][|options])
Requests one or more channels and places calls to those channels in
a multi-way chat. All resources are dialed simultaneously, and as
soon as channels answer, a low/high tone is played in the
"conference" indicating that a new user has joined the conference.
Options:
'A(x)' -- Plays an attention-getting file (usually a set of tones)
over the pager channel before playing the "ready" tones to the caller.
'q' -- Do not play tones as callers enter/exit the conference bridge
'w(x)' -- Wait time, in seconds, for at least one resource to
answer. If expired, exit with 0.
JT
At 8:53 AM -0800 11/14/03, Chris Albertson wrote:
>
>I'd hate to see conference bridging use for paging. A lot of
>wasted CPU and bandwidth. Could you "multicast" the UDP packets?
>
>We assume you don't need to page across multiple Asterisk servers
>but if you did the software wuld need to be smart enough to
>"know" which groups of extensions could be in a multicast and
>whci need to be bridged. Basically check to see if the SIP phone
>are on the same subnet.
>
>
>--- DUSTIN WILDES <dwildes at pabbankshares.com> wrote:
>> I feel this needs to be a separate application in Asterisk, like
>> app_sipintercom
>> The application would connect to all available auto-answer SIP
>> phones, play a short frequency tone for the intercom alert, only
> > allow one-way streaming to the phones, then disconnect all phones
>> whenever the originator hangs up.
>>
>> Same is true for a paging application, app_sippage
>> The application should work the same as intercom, but allow two-way
>> audio streaming.
>>
>> I was starting the design of these two applications unless anyone
>> else has a better idea or has already begun work?
>> Feedback welcome....
>>
>>
>>
>>
>>
>> -----Original Message-----
>> From: Jerry Gibson [mailto:jerrygib at ix.netcom.com]
>> Sent: Friday, November 14, 2003 10:41 AM
>> To: asterisk-users at lists.digium.com
>> Subject: RE: [Asterisk-Users] Overhead Paging
>>
>>
>> Scott:
>>
>> The operation you describe with multiple phones set to ring on the
>> same extension is correct. The first one that answers, gets it.
>> However, the meetme setup you describe also works great. I have that
>> set up for a conference bridge where one person sets up a conference
>> with one click on a web page which calls multiple Snom phones into a
>> conference. This is a full conference where everyone can talk to
>> everyone. However, The Snom phone also allows you set it up with the
>> mic permanently muted, which would work great for paging.
>>
>> Jerry
>>
>>
>> -----Original Message-----
>> From: asterisk-users-admin at lists.digium.com
>> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Bisker,
>> Scott (7805)
>> Sent: Friday, November 14, 2003 9:25 AM
>> To: 'asterisk-users at lists.digium.com'
>> Subject: RE: [Asterisk-Users] Overhead Paging
>>
>>
>> Jerry,
>>
>> Do you have it setup so that multiple phones answer one extension? I
>> tried that setup with two Cisco phones, however, only the quickest
>> responding phone answered. If you have a config that rings multiple
>> phones and all of the phones answer the same call, I'd be interested
>> to see the config. I guess theway to do it would be to setup a
>> meetme conference and then dial all parties into the conference then
>> speak....
>>
>> -sb
>>
>> -----Original Message-----
>> From: Jerry Gibson [mailto:jerrygib at ix.netcom.com]
>> Sent: Friday, November 14, 2003 8:52 AM
> > To: asterisk-users at lists.digium.com
> > Subject: RE: [Asterisk-Users] Overhead Paging
> >
> >
> > We do the same thing with the Snom phones. They can be set up for
> > auto-answer, and they have a speaker jack in the back that is the
>> same levels as a sound card on a PC. And the Snom phone automaticly
>> hangs up when the caller hang up is detected (the SIP BYE message).
>>
>> Jerry
>>
>> -----Original Message-----
>> From: asterisk-users-admin at lists.digium.com
>> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Bisker,
>> Scott (7805)
>> Sent: Thursday, November 13, 2003 6:17 PM
>> To: 'asterisk-users at lists.digium.com'
>> Subject: RE: [Asterisk-Users] Overhead Paging
>>
>>
>> Our setup is to set the OSS device to autoanswer. The output of the
>> soundcard feeds into a bank of overhead speakers. If the channel is
>> in use, then the call gets put in a queue until the OSS device is
>> free.
>>
>> -sb
>>
>>
>>
>>
>> -----Original Message-----
>> From: Johnson, Randy [mailto:rjohnson at Spang.com]
>> Sent: Thursday, November 13, 2003 5:34 PM
>> To: 'asterisk-users at lists.digium.com'
>> Subject: [Asterisk-Users] Overhead Paging
>>
>>
>>
>> Does anyone have any recommendations for overhead paging systems for
>> use with Asterisk?
>>
>> Thanks,
>> Randy Johnson
>>
>>
>
>
>=====
>Chris Albertson
> Home: 310-376-1029 chrisalbertson90278 at yahoo.com
> Cell: 310-990-7550
> Office: 310-336-5189 Christopher.J.Albertson at aero.org
> KG6OMK
>
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