RE: [Asterisk-Users] Cisco 7940 Phones as paging system?

2004-05-08 Thread Joe Antkowiak








Got so many people asking for it, heres
what I used for the intercom announce:



http://www.jsci.net/asterisk/intercom-tone.gsm



Its not great, but it does the
job. Actually trying to find something better



-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Antkowiak
Sent: Friday, May 07,
 2004 4:30 PM
To: [EMAIL PROTECTED]
Cc:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
Cisco 7940 Phones as paging system?



This is what we have for
this customer. They have five phones right now. Their normal
extensions are 610x, but for intercom its 510x:



exten =
5101,1,Dial(SIP/5101,10,tA(intercom-tone))

exten =
5101,2,Congestion



If you want the wav file,
let me know. If you make your own, be sure to put a 1-2 second pause in
the beginning, because when the cisco answers it takes a second or to before it
will send any audio to the speaker.



-Original Message-
From: mitchel
[mailto:[EMAIL PROTECTED] 
Sent: Friday, May 07,
 2004 4:16 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
Cisco 7940 Phones as paging system?





Hey Joe,











Could I get a sample config for playing some intro
tones on the intercom? I have the same thing but nobody is using it now because
they are afraid of having someone call in and listen in so we need
some way to announce the incoming intercom call.











Thanks,





Mitchel

Joe Antkowiak 
wrote:





I am currently using 7960's with *, and line 6 is set
to auto answer. Works
great, customer is happy. As far as an intro-tone, you can set the dial
command to play a sound (using the announce option) before the call is
connected. I grabbed a simple tone wav file, and made it play that. Now,
when the intercom ext is called, it plays the tone on the destination phone,
and wa-la, intercom

So it works. Let me know if you need sample configs.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp von
Klitzing
Sent: Friday, May 07, 2004 12:57 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7940 Phones as paging system?

Hi!

 able to support intercom/paging. Having searched the archives, it 
 appears that this question was asked about 6 months ago, and the answer 
 was that the Cisco phones support this using SCCP and having one line 
 set to auto-answer, but at the time this was not supported in the SIP 
 image. Is this still the case?

Dunno about Cisco, but wanted to let you know that the recent Grandstream 
firmware (.55 and later) now also has an auto-answer option. Still I 
guess I should mention that the microphone of the GS phones in 
speakerphone mode is far from a brilliant implementation (- echo for the 
remote speaker talker, and too thin sound from the person in the room).

Cheers, Philipp


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Re: [Asterisk-Users] Cisco 7940 Phones as paging system?

2004-05-08 Thread Ian A. Underwood
Joe Antkowiak wrote:

exten = 5101,1,Dial(SIP/5101,10,tA(intercom-tone))
exten = 5101,2,Congestion
That's not too bad, but how do you page a group of phones...like a real 
intercom?  That's what I'm dying to know!

--
/* Ian A. Underwood - [EMAIL PROTECTED] - http://www.agentgreen.org
   There are 4 boxes to use in the defense of liberty:
   soap, ballot, jury, ammo. Use in that order. Starting now. */
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Re: [Asterisk-Users] Cisco 7940 Phones as paging system?

2004-05-08 Thread Vic Cross
On Fri, 7 May 2004, Ian A. Underwood wrote:

 Joe Antkowiak wrote:
 
  exten = 5101,1,Dial(SIP/5101,10,tA(intercom-tone))
  exten = 5101,2,Congestion
 
 That's not too bad, but how do you page a group of phones...like a real 
 intercom?  That's what I'm dying to know!

in extensions.conf:

[globals]
INTERCOMLINES=SIP/Alice6SIP/Bob6SIP/Chuck6...

Then the extension is as per Joe's example, but replacing SIP/5101 with 
${INTERCOMLINES}.

Extending this, you could set up various intercom numbers for different
parts of the office...

[globals]
SALESINTERCOM=SIP/Sales1-6SIP/Sales2-6...
MKTGINTERCOM=SIP/Marketing1-6...
...

[yourcontext]
exten = 5101,1,Dial(${SALESINTERCOM},10,tA(tone))
exten = 5102,1,Dial(${MKTGINTERCOM},10,tA(tone))
...
exten = 5110,1,Dial(${SALESINTERCOM}${MKTGINTERCOM}${...},10,tA(tone))


Cheers,
Vic Cross
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Re: [Asterisk-Users] Cisco 7940 Phones as paging system?

2004-05-08 Thread Billy Huddleston
That won't work.. That'll DIAL multiple phones/extensions, but will only
bridge 1 of them when it auto-answers..

What we need is a way to have something like meetme call multiple extensions
and bridge them to a meetme confrence (all of them muted but the admin of
course, as it's a one way page) and then we would have a true paging
system..


- Original Message -
From: Vic Cross [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 08, 2004 10:11 AM
Subject: Re: [Asterisk-Users] Cisco 7940 Phones as paging system?


 On Fri, 7 May 2004, Ian A. Underwood wrote:

  Joe Antkowiak wrote:
 
   exten = 5101,1,Dial(SIP/5101,10,tA(intercom-tone))
   exten = 5101,2,Congestion
 
  That's not too bad, but how do you page a group of phones...like a real
  intercom?  That's what I'm dying to know!

 in extensions.conf:

 [globals]
 INTERCOMLINES=SIP/Alice6SIP/Bob6SIP/Chuck6...

 Then the extension is as per Joe's example, but replacing SIP/5101 with
 ${INTERCOMLINES}.

 Extending this, you could set up various intercom numbers for different
 parts of the office...

 [globals]
 SALESINTERCOM=SIP/Sales1-6SIP/Sales2-6...
 MKTGINTERCOM=SIP/Marketing1-6...
 ...

 [yourcontext]
 exten = 5101,1,Dial(${SALESINTERCOM},10,tA(tone))
 exten = 5102,1,Dial(${MKTGINTERCOM},10,tA(tone))
 ...
 exten = 5110,1,Dial(${SALESINTERCOM}${MKTGINTERCOM}${...},10,tA(tone))


 Cheers,
 Vic Cross
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Re: [Asterisk-Users] Cisco 7940 Phones as paging system?

2004-05-08 Thread John Todd
At 10:31 AM -0400 5/8/04, Billy Huddleston wrote:
That won't work.. That'll DIAL multiple phones/extensions, but will only
bridge 1 of them when it auto-answers..
What we need is a way to have something like meetme call multiple extensions
and bridge them to a meetme confrence (all of them muted but the admin of
course, as it's a one way page) and then we would have a true paging
system..
OK, I typically would badger people into looking in Google for this, 
but I'll be darned if I can't find this post on Google myself (search 
for Office-wide paging with Asterisk or AGI(callall) so I'll 
re-post here.  This is a terrible hack.  Someone _please_ make this 
cleaner.

I'm looking at how to add this to the Wiki, but I don't see anything 
that's obviously marked as start new thread or similar links.  If 
anyone is feeling ambitious, please add the stuff below.

JT



Date: Sun, 18 Jan 2004 17:22:11 -0700
To: asterisk-users-lists.digium.com
From: John Todd [EMAIL PROTECTED]
Subject: Office-wide paging with Asterisk and Cisco 7960 7940 phones
I spoke the other day about my preliminary tests with office-wide 
paging with Cisco phones using the new SIP 6.1 image which supports 
auto-answer.  I've got a small and crude recipe for those of you who 
want to experiment and hopefully create some better and more 
complete examples than the one I've thrown together below.

Create a new line on each of the Cisco phones, and put the 
configuration into sip.conf as you normally would.  I figure you 
have enough clue to create a new line in sip.conf and on your Cisco 
phones at this point.  Go into settings - Call Preferences - Auto 
Answer (intercom)  and then make the new line you've just created 
as auto-answer.  (I wish there was a way to do this via the 
configuration file!  Having to set this while sitting in front of 
the phone is silly and wasteful.)

Now that you have created a valid Asterisk-capable SIP line that 
auto-answers, here's how you get the paging features to work:

Here's what I have in extensions.conf:

[conference]
exten = ,1,AbsoluteTimeout(21)
exten = ,2,AGI(callall)
exten = ,3,MeetMe(,dq)
exten = ,4,Hangup
exten = t,1,Hangup
exten = T,1,Hangup
exten = h,1,Hangup
;
[add-to-conference]
exten = start,1,AbsoluteTimeout(20)
exten = start,2,MeetMe(,dmq)
exten = h,1,Hangup
exten = t,1,Hangup
exten = T,1,Hangup
Here are the contents of /var/lib/asterisk/agi-bin/callall

#!/bin/sh
cp /var/lib/asterisk/agi-bin/*conf /var/spool/asterisk/outgoing
Make sure to make the script executable.

And then for every extension I have as an auto-answer, I have a file 
like this in /var/lib/asterisk/agi-bin :

Channel: SIP/2006
Context: add-to-conference
Extension: start
Priority: 1
CallerID: Office Pager 
So, I have three lines that are configured for automatic answering - 
SIP/2006, SIP/2007, SIP/2008.  I have three files named 2006-conf, 
2007-conf, 2008-conf in /var/lib/asterisk/agi-bin that get copied 
into the outgoing call spool directory every time I call extension 
.   These three lines are the auto-answer lines on each of the 
three phone devices I'm experimenting with.

Now, dial  from any phone and you should have one-way paging. 
Voila!  People who use the pager may have to get used to waiting 1-2 
seconds before speaking to allow all the phones to catch up with the 
audio stream.  All of the phones hang up after 20 seconds, 
regardless of if the person originating the page has stopped 
talking.  Change the AbsoluteTimeout values to increase this 
interval.

If you want a really confusing loud mess, then change the dmq 
options to dq and you'll get an N-way conversation going with 
everyone who has a phone.  Bad.

If you want a really interesting office surveillance tool, change 
the dmq to dt and you'll suddenly be listening to all of the 
extensions in the office, like some kind of mega-snoop tool. Useful 
for after-hours listening throughout the entire office.

Someone should improve my scripts with the following changes:
 1) AGI should automatically show the caller ID of the person 
originating the call instead of a generic pager address
 2) The AGI should take arguments of what extensions to call and 
then dynamically create the list of files that get copied out to the 
/var/spool/asterisk/outgoing directory

JT

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Re: [Asterisk-Users] Cisco 7940 Phones as paging system?

2004-05-08 Thread John Baker
This hack is a tiny bit better:

http://lists.digium.com/pipermail/asterisk-users/2004-March/040186.html

John Baker

John Todd wrote:
At 10:31 AM -0400 5/8/04, Billy Huddleston wrote:

That won't work.. That'll DIAL multiple phones/extensions, but will only
bridge 1 of them when it auto-answers..
What we need is a way to have something like meetme call multiple 
extensions
and bridge them to a meetme confrence (all of them muted but the admin of
course, as it's a one way page) and then we would have a true paging
system..


OK, I typically would badger people into looking in Google for this, but 
I'll be darned if I can't find this post on Google myself (search for 
Office-wide paging with Asterisk or AGI(callall) so I'll re-post 
here.  This is a terrible hack.  Someone _please_ make this cleaner.

I'm looking at how to add this to the Wiki, but I don't see anything 
that's obviously marked as start new thread or similar links.  If 
anyone is feeling ambitious, please add the stuff below.

JT



Date: Sun, 18 Jan 2004 17:22:11 -0700
To: asterisk-users-lists.digium.com
From: John Todd [EMAIL PROTECTED]
Subject: Office-wide paging with Asterisk and Cisco 7960 7940 phones
I spoke the other day about my preliminary tests with office-wide 
paging with Cisco phones using the new SIP 6.1 image which supports 
auto-answer.  I've got a small and crude recipe for those of you who 
want to experiment and hopefully create some better and more complete 
examples than the one I've thrown together below.

Create a new line on each of the Cisco phones, and put the 
configuration into sip.conf as you normally would.  I figure you have 
enough clue to create a new line in sip.conf and on your Cisco phones 
at this point.  Go into settings - Call Preferences - Auto Answer 
(intercom)  and then make the new line you've just created as 
auto-answer.  (I wish there was a way to do this via the configuration 
file!  Having to set this while sitting in front of the phone is silly 
and wasteful.)

Now that you have created a valid Asterisk-capable SIP line that 
auto-answers, here's how you get the paging features to work:

Here's what I have in extensions.conf:

[conference]
exten = ,1,AbsoluteTimeout(21)
exten = ,2,AGI(callall)
exten = ,3,MeetMe(,dq)
exten = ,4,Hangup
exten = t,1,Hangup
exten = T,1,Hangup
exten = h,1,Hangup
;
[add-to-conference]
exten = start,1,AbsoluteTimeout(20)
exten = start,2,MeetMe(,dmq)
exten = h,1,Hangup
exten = t,1,Hangup
exten = T,1,Hangup
Here are the contents of /var/lib/asterisk/agi-bin/callall

#!/bin/sh
cp /var/lib/asterisk/agi-bin/*conf /var/spool/asterisk/outgoing
Make sure to make the script executable.

And then for every extension I have as an auto-answer, I have a file 
like this in /var/lib/asterisk/agi-bin :

Channel: SIP/2006
Context: add-to-conference
Extension: start
Priority: 1
CallerID: Office Pager 
So, I have three lines that are configured for automatic answering - 
SIP/2006, SIP/2007, SIP/2008.  I have three files named 2006-conf, 
2007-conf, 2008-conf in /var/lib/asterisk/agi-bin that get copied into 
the outgoing call spool directory every time I call extension .   
These three lines are the auto-answer lines on each of the three phone 
devices I'm experimenting with.

Now, dial  from any phone and you should have one-way paging. 
Voila!  People who use the pager may have to get used to waiting 1-2 
seconds before speaking to allow all the phones to catch up with the 
audio stream.  All of the phones hang up after 20 seconds, regardless 
of if the person originating the page has stopped talking.  Change the 
AbsoluteTimeout values to increase this interval.

If you want a really confusing loud mess, then change the dmq 
options to dq and you'll get an N-way conversation going with 
everyone who has a phone.  Bad.

If you want a really interesting office surveillance tool, change the 
dmq to dt and you'll suddenly be listening to all of the 
extensions in the office, like some kind of mega-snoop tool. Useful 
for after-hours listening throughout the entire office.

Someone should improve my scripts with the following changes:
 1) AGI should automatically show the caller ID of the person 
originating the call instead of a generic pager address
 2) The AGI should take arguments of what extensions to call and then 
dynamically create the list of files that get copied out to the 
/var/spool/asterisk/outgoing directory

JT

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RE: [Asterisk-Users] Cisco 7940 Phones as paging system?

2004-05-07 Thread Carlton J. O'Riley
The SIP 6.1 image has auto answer available, which would function the same
as the SCCP implementation. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Travers
Sent: Friday, May 07, 2004 12:02 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco 7940 Phones as paging system?

Hi all;

I have been searching for an answer to a question that a customer asked me
and I have only found a few older answers.  So, wanting to find out if
anyone has any experience with this issue and can help provide me with some
advice.

I have a customer which is strongly interested in using Asterisk as a PBX.
One of the core requirements, however, is that the system MUST be able to
support intercom/paging.  Having searched the archives, it appears that this
question was asked about 6 months ago, and the answer was that the Cisco
phones support this using SCCP and having one line set to auto-answer, but
at the time this was not supported in the SIP image.  Is this still the
case?

Although I know that SIP is the preferred protocol for connecting these
phones with Asterisk, how stable/reliable are the skinny channels?  Is there
any reason I should be rethinking this solution?

Best Wishes,
Chris Travers
Metatron Technology Consulting
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Re: [Asterisk-Users] Cisco 7940 Phones as paging system?

2004-05-07 Thread Rich Adamson
 I have been searching for an answer to a question that a customer asked 
 me and I have only found a few older answers.  So, wanting to find out 
 if anyone has any experience with this issue and can help provide me 
 with some advice.
 
 I have a customer which is strongly interested in using Asterisk as a 
 PBX.  One of the core requirements, however, is that the system MUST be 
 able to support intercom/paging.  Having searched the archives, it 
 appears that this question was asked about 6 months ago, and the answer 
 was that the Cisco phones support this using SCCP and having one line 
 set to auto-answer, but at the time this was not supported in the SIP 
 image.  Is this still the case?
 
 Although I know that SIP is the preferred protocol for connecting these 
 phones with Asterisk, how stable/reliable are the skinny channels?  Is 
 there any reason I should be rethinking this solution?

Apparently your search didn't find several other postings on the subject.

The cisco v6.x sip releases also include the ability to auto-answer a
call (required for phone paging), however some folks tend to suggest that
is a security problem as anyone can call that autoanswer extn number
and listen in on whatever is going on around the phone. There is no
beep or other indication the phone/microphone is open.

If your customer is looking for overhead (as in PA) paging, there are lots
of postings relative to add-on hardware, etc, to do that. Use of the 
sound card within the * box has historically been hit/miss as not all
sound cards are supported, etc.

Rich


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Re: [Asterisk-Users] Cisco 7940 Phones as paging system?

2004-05-07 Thread Philipp von Klitzing
Hi!

 able to support intercom/paging.  Having searched the archives, it 
 appears that this question was asked about 6 months ago, and the answer 
 was that the Cisco phones support this using SCCP and having one line 
 set to auto-answer, but at the time this was not supported in the SIP 
 image.  Is this still the case?

Dunno about Cisco, but wanted to let you know that the recent Grandstream 
firmware (.55 and later) now also has an auto-answer option. Still I 
guess I should mention that the microphone of the GS phones in 
speakerphone mode is far from a brilliant implementation (- echo for the 
remote speaker talker, and too thin sound from the person in the room).

Cheers, Philipp


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Re: [Asterisk-Users] Cisco 7940 Phones as paging system?

2004-05-07 Thread Shane Young
Quoting Rich Adamson [EMAIL PROTECTED]:

 The cisco v6.x sip releases also include the ability to auto-answer a
 call (required for phone paging), however some folks tend to suggest that
 is a security problem as anyone can call that autoanswer extn number
 and listen in on whatever is going on around the phone. There is no
 beep or other indication the phone/microphone is open.

We are on Cisco's beta program where we get to try out the new sip software before 
it's released, 
report bugs, suggestions, etc.  This was one of the things I pointed out to them.  I 
suggested 
adding a beep or something just before it answers.  At the very least, they could make 
the icon of 
that line different than the regular idle icon.  

They didn't add the beep, but they did add the icon so when the phone is idle, you can 
see that it 
could auto-answer.

I'll try suggesting it again for 7.X of the SIP image.

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RE: [Asterisk-Users] Cisco 7940 Phones as paging system?

2004-05-07 Thread Joe Antkowiak
I am currently using 7960's with *, and line 6 is set to auto answer.  Works
great, customer is happy.  As far as an intro-tone, you can set the dial
command to play a sound (using the announce option) before the call is
connected.  I grabbed a simple tone wav file, and made it play that.  Now,
when the intercom ext is called, it plays the tone on the destination phone,
and wa-la, intercom

So it works.  Let me know if you need sample configs.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp von
Klitzing
Sent: Friday, May 07, 2004 12:57 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7940 Phones as paging system?

Hi!

 able to support intercom/paging.  Having searched the archives, it 
 appears that this question was asked about 6 months ago, and the answer 
 was that the Cisco phones support this using SCCP and having one line 
 set to auto-answer, but at the time this was not supported in the SIP 
 image.  Is this still the case?

Dunno about Cisco, but wanted to let you know that the recent Grandstream 
firmware (.55 and later) now also has an auto-answer option. Still I 
guess I should mention that the microphone of the GS phones in 
speakerphone mode is far from a brilliant implementation (- echo for the 
remote speaker talker, and too thin sound from the person in the room).

Cheers, Philipp


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Re: [Asterisk-Users] Cisco 7940 Phones as paging system?

2004-05-07 Thread Jeff Roberts
Joe Antkowiak wrote:

I am currently using 7960's with *, and line 6 is set to auto answer.  Works
great, customer is happy.  As far as an intro-tone, you can set the dial
command to play a sound (using the announce option) before the call is
connected.  I grabbed a simple tone wav file, and made it play that.  Now,
when the intercom ext is called, it plays the tone on the destination phone,
and wa-la, intercom
So it works.  Let me know if you need sample configs.

 

I'd like to see them, maybe you could put them on the wiki.
-Jeff
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RE: [Asterisk-Users] Cisco 7940 Phones as paging system?

2004-05-07 Thread Joe Antkowiak








This is what we have for this
customer. They have five phones right now. Their normal extensions
are 610x, but for intercom its 510x:



exten =
5101,1,Dial(SIP/5101,10,tA(intercom-tone))

exten = 5101,2,Congestion



If you want the wav file, let me
know. If you make your own, be sure to put a 1-2 second pause in the
beginning, because when the cisco answers it takes a second or to before it
will send any audio to the speaker.



-Original Message-
From: mitchel
[mailto:[EMAIL PROTECTED] 
Sent: Friday, May 07, 2004 4:16 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
Cisco 7940 Phones as paging system?





Hey Joe,











Could I get a sample config for playing some intro
tones on the intercom? I have the same thing but nobody is using it now because
they are afraid of having someone call in and listen in so we need
some way to announce the incoming intercom call.











Thanks,





Mitchel

Joe Antkowiak 
wrote:





I am currently using 7960's with *, and line 6 is set
to auto answer. Works
great, customer is happy. As far as an intro-tone, you can set the dial
command to play a sound (using the announce option) before the call is
connected. I grabbed a simple tone wav file, and made it play that. Now,
when the intercom ext is called, it plays the tone on the destination phone,
and wa-la, intercom

So it works. Let me know if you need sample configs.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp von
Klitzing
Sent: Friday, May 07, 2004 12:57 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7940 Phones as paging system?

Hi!

 able to support intercom/paging. Having searched the archives, it 
 appears that this question was asked about 6 months ago, and the answer 
 was that the Cisco phones support this using SCCP and having one line 
 set to auto-answer, but at the time this was not supported in the SIP 
 image. Is this still the case?

Dunno about Cisco, but wanted to let you know that the recent Grandstream 
firmware (.55 and later) now also has an auto-answer option. Still I 
guess I should mention that the microphone of the GS phones in 
speakerphone mode is far from a brilliant implementation (- echo for the 
remote speaker talker, and too thin sound from the person in the room).

Cheers, Philipp


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Re: [Asterisk-Users] Cisco 7940 Phones as paging system?

2004-05-07 Thread Billy Huddleston



hey, can you send me the tone?


  - Original Message - 
  From: 
  Joe Antkowiak 
  To: [EMAIL PROTECTED] 
  Cc: [EMAIL PROTECTED] 
  
  Sent: Friday, May 07, 2004 4:30 PM
  Subject: RE: [Asterisk-Users] Cisco 7940 
  Phones as paging system?
  
  
  This is what we have 
  for this customer. They have five phones right now. Their normal 
  extensions are 610x, but for intercom its 510x:
  
  exten = 
  5101,1,Dial(SIP/5101,10,tA(intercom-tone))
  exten = 
  5101,2,Congestion
  
  If you want the wav 
  file, let me know. If you make your own, be sure to put a 1-2 second 
  pause in the beginning, because when the cisco answers it takes a second or to 
  before it will send any audio to the speaker.
  
  -Original 
  Message-From: mitchel 
  [mailto:[EMAIL PROTECTED] Sent: Friday, May 07, 2004 4:16 
  PMTo: 
  [EMAIL PROTECTED]Subject: RE: 
  [Asterisk-Users] Cisco 7940 Phones as paging system?
  
  
  Hey Joe,
  
  
  
  Could I get a sample config for playing 
  some intro tones on the intercom? I have the same thing but nobody is using it 
  now because they are afraid of having someone call in and "listen in" so we 
  need some way to announce the incoming intercom call.
  
  
  
  Thanks,
  
  MitchelJoe Antkowiak 
   wrote:
  
I am currently using 7960's with *, and 
line 6 is set to auto answer. Worksgreat, customer is happy. As far as 
an intro-tone, you can set the dialcommand to play a sound (using the 
announce option) before the call isconnected. I grabbed a simple tone 
wav file, and made it play that. Now,when the intercom ext is called, it 
plays the tone on the destination phone,and wa-la, intercomSo it 
works. Let me know if you need sample configs.-Original 
Message-From: 
[EMAIL PROTECTED][mailto:[EMAIL PROTECTED] 
On Behalf Of Philipp vonKlitzingSent: Friday, May 07, 2004 12:57 
PMTo: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] 
Cisco 7940 Phones as paging system?Hi! able to support 
intercom/paging. Having searched the archives, it  appears that this 
question was asked about 6 months ago, and the answer  was that the 
Cisco phones support this using SCCP and having one line  set to 
auto-answer, but at the time this was not supported in the SIP  
image. Is this still the case?Dunno about Cisco, but wanted to let 
you know that the recent Grandstream firmware (.55 and later) now also 
has an auto-answer option. Still I guess I should mention that the 
microphone of the GS phones in speakerphone mode is far from a brilliant 
implementation (- echo for the remote speaker talker, and too thin 
sound from the person in the room).Cheers, 
Philipp___Asterisk-Users 
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RE: [Asterisk-Users] Cisco 7940 Phones as paging system?

2004-05-07 Thread stevek
I would apprechiate the wav tone...

Regards, 


Steve Kalcevich


Quoting Joe Antkowiak [EMAIL PROTECTED]:

 This is what we have for this customer.  They have five phones right now.
 Their normal extensions are 610x, but for intercom its 510x:
 
  
 
 exten = 5101,1,Dial(SIP/5101,10,tA(intercom-tone))
 
 exten = 5101,2,Congestion
 
  
 
 If you want the wav file, let me know.  If you make your own, be sure to put
 a 1-2 second pause in the beginning, because when the cisco answers it takes
 a second or to before it will send any audio to the speaker.
 
  
 
 -Original Message-
 From: mitchel [mailto:[EMAIL PROTECTED] 
 Sent: Friday, May 07, 2004 4:16 PM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Cisco 7940 Phones as paging system?
 
  
 
 Hey Joe,
 
  
 
 Could I get a sample config for playing some intro tones on the intercom? I
 have the same thing but nobody is using it now because they are afraid of
 having someone call in and listen in so we need some way to announce the
 incoming intercom call.
 
  
 
 Thanks,
 
 Mitchel
 
 Joe Antkowiak  wrote:
 
 I am currently using 7960's with *, and line 6 is set to auto answer. Works
 great, customer is happy. As far as an intro-tone, you can set the dial
 command to play a sound (using the announce option) before the call is
 connected. I grabbed a simple tone wav file, and made it play that. Now,
 when the intercom ext is called, it plays the tone on the destination phone,
 and wa-la, intercom
 
 So it works. Let me know if you need sample configs.
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von
 Klitzing
 Sent: Friday, May 07, 2004 12:57 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Cisco 7940 Phones as paging system?
 
 Hi!
 
  able to support intercom/paging. Having searched the archives, it 
  appears that this question was asked about 6 months ago, and the answer 
  was that the Cisco phones support this using SCCP and having one line 
  set to auto-answer, but at the time this was not supported in the SIP 
  image. Is this still the case?
 
 Dunno about Cisco, but wanted to let you know that the recent Grandstream 
 firmware (.55 and later) now also has an auto-answer option. Still I 
 guess I should mention that the microphone of the GS phones in 
 speakerphone mode is far from a brilliant implementation (- echo for the 
 remote speaker talker, and too thin sound from the person in the room).
 
 Cheers, Philipp
 
 
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